Nec Univerge Neax 2000 Ips Users Manual 2006007RFP

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UNIVERGE NEAX 2000 IPS
INTERNET PROTOCOL SERVER
Request For Proposal (RFP)
Reference Guide

May 2006

NEC Unified Solutions, Inc.

LIABILITY DISCLAIMER
NEC Unified Solutions, Inc. reserves the right to change the specifications, functions, or
features, at any time, without notice.
NEC Unified Solutions, Inc. has prepared this document for use by its employees and
customers. The information contained herein is the property of NEC Unified Solutions, Inc. and
shall not be reproduced without prior written approval from NEC Unified Solutions, Inc.

UNIVERGE®, NEAX® and Dterm® are registered trademarks of NEC Corporation. Microsoft® is

a registered trademark of Microsoft Corporation. Intel® is a registered trademark of Intel
Corporation. All other product references and/or company references are registered
trademarks or trademarked for their respective products and/or company.

The UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide has been
developed to provide technical information on the UNIVERGE NEAX 2000 IPS, which is
required when responding to the majority of questions asked on Request For Proposals. The
information provided has been compiled from a variety of available documentation and has
been consolidated into a single manual. For this reason, it may be necessary to gather
additional information from other supporting documentation to fulfill the Request For Proposal
(RFP) requirements.
Information concerning questions not covered in this guide, corrections and/or comments are
most welcome and should be sent to:
NEAX2000@necunifiedsolutions.com
NEC Unified Solutions, Inc.

UNIVERGE NEAX 2000 IPS Product Management
6555 North State Highway 161
Irving, TX 75039-2402
http://www.necunifiedsolutions.com
Copyright 2006
NEC Unified Solutions, Inc.
Printed in the U.S.A

Chapter 1 Introduction .......................................................................................................................................... 1

Overview of NEC ...................................................................................................................................1
Components ..........................................................................................................................................3
Communications ....................................................................................................................................4
Social Contributions...............................................................................................................................4
Global Resources ..................................................................................................................................4
NEC Unified Solutions, Inc. ...................................................................................................................6
Vertical Markets .....................................................................................................................................8
Competitive Benefits and Advantages ...................................................................................................8
UNIVERGE NEAX 2000 IPS ...............................................................................................................11
UNIVERGE NEAX IPS DM ..................................................................................................................14
UNIVERGE NEAX IPS DML ................................................................................................................15
UNIVERGE NEAX IPS DMR ...............................................................................................................15
Stand Alone System Capacity .............................................................................................................17
IP Remote Network Capacity...............................................................................................................23
Chapter 2 System Architecture ............................................................................................................................ 1

Hardware Architecture ...........................................................................................................................1
UNIVERGE NEAX 2000 IPS System Configuration ..............................................................................4
UNIVERGE NEAX IPS DM/IPS DML/IPS DMR System Configuration ...............................................11
Software Architecture ..........................................................................................................................15
Chapter 3 System Highlights................................................................................................................................ 1

Processors.............................................................................................................................................1
Reliability and Availability ......................................................................................................................4
System Traffic........................................................................................................................................8
System Specifications..........................................................................................................................16
System Compliance.............................................................................................................................23
Chapter 4 Equipment List ..................................................................................................................................... 1

Module/Installation Hardware ................................................................................................................1
Circuit Cards ..........................................................................................................................................3
Chapter 5 Station Equipment ............................................................................................................................... 1

Terminal Line Up ...................................................................................................................................1
Analog Terminals...................................................................................................................................2
Dterm Series i (IP) Terminals ................................................................................................................3
Dterm Series i (TDM) Multi-line Digital Terminals..................................................................................5
Dterm Series E (Multi-line Digital Terminals) .........................................................................................8
Dterm Cordless Terminals ...................................................................................................................15
INASET................................................................................................................................................18
Dterm PS III Wireless Handset ............................................................................................................20
Dterm SP30 SoftPhone .......................................................................................................................21
Dterm Extenders..................................................................................................................................26
Chapter 6 Trunking................................................................................................................................................ 1

Type of Trunks.......................................................................................................................................1
Trunk Card Specifications......................................................................................................................2
System Trunking Capacity.....................................................................................................................6
Least Cost Routing (LCR)......................................................................................................................8
Chapter 7 Attendant Answering Position ........................................................................................................... 1

Attendant Console (SN716)...................................................................................................................1
Business Attendant System (BAS) ........................................................................................................8

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Chapter 8 System Administration........................................................................................................................ 1

System Administration ...........................................................................................................................1
Customer Administration Terminal (CAT) ..............................................................................................1
Maintenance Administration Terminal (MAT) ........................................................................................2
System Diagnostics ...............................................................................................................................5
Self Diagnostic/System Messages ........................................................................................................5
Remote Maintenance.............................................................................................................................5
MP Program Download..........................................................................................................................6
MA4000 Management System ..............................................................................................................8
Chapter 9 Feature Descriptions ........................................................................................................................... 1

Business/Hotel/Data Feature List ..........................................................................................................1
Business/Hotel/Data Feature Descriptions ............................................................................................5
CCIS Feature List ................................................................................................................................34
CCIS Features Descriptions ................................................................................................................35
ISDN Feature List ................................................................................................................................41
ISDN Feature Descriptions ..................................................................................................................42
Q-SIG Feature List...............................................................................................................................44
Q-SIG Feature Descriptions ................................................................................................................44
Wireless Feature List ...........................................................................................................................45
Wireless Feature Descriptions.............................................................................................................47
Chapter 10 System Input/Output (I/O) Interfaces ................................................................................................. 1

Station Message Detail Recording (SMDR) ..........................................................................................1
Message Center Interface (MCI) ...........................................................................................................6
Property Management System (PMS) ...................................................................................................8
Chapter 11 Open Application Interface ............................................................................................................... 1

System Outline ......................................................................................................................................1
OpenWorX: Attendant Statistics ............................................................................................................1
OpenWorX: Business Attendant System (BAS) ....................................................................................1
OpenWorX: Business Receptionist (BR) ...............................................................................................6
OpenWorX: Communications Portal......................................................................................................7
OpenWorX: Dialer..................................................................................................................................9
OpenWorX: Location Status Information (LSI) ....................................................................................10
OpenWorX: Message Reader (MR).....................................................................................................10
OpenWorX: Short Text Messaging ......................................................................................................11
OpenWorX: Incoming Call Assistant (ICA) ..........................................................................................12
OpenWorX: Group Call Forward Control (GCFC) ...............................................................................13
OpenWorX: Personal Call Assistant (PCA) .........................................................................................14
OpenWorX: Name Display...................................................................................................................16
OpenWorX: Multiple CCIS Node Configuration ...................................................................................17
OAI Application Software Development...............................................................................................17
System Specifications..........................................................................................................................17
Chapter 12 Automatic Call Distribution (ACD) with MIS...................................................................................... 1

Automatic Call Distribution (ACD)..........................................................................................................1
Basic ACD .............................................................................................................................................1
Basic ACD Features ..............................................................................................................................2
CallCenterWorX ACD 3.0 for Business .................................................................................................3
CallCenterWorX MIS .............................................................................................................................6
Q-Master 3.1........................................................................................................................................10
QueWorX4.0 ........................................................................................................................................15
Professional Services for QueWorX ....................................................................................................18
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Chapter 13 Voice over IP (VoIP) ........................................................................................................................... 1

VoIP Solutions .......................................................................................................................................1
Extended Enterprise IP Solution............................................................................................................1
IP Station ...............................................................................................................................................3
CCIS Networking via IP .......................................................................................................................14
H.323 Connection ................................................................................................................................18
FAX and Modem over IP .....................................................................................................................21
Remote PIM over IP ............................................................................................................................25
Planning and Installation......................................................................................................................30
System Conditions/Limitations (Peer-to-Peer IP) ................................................................................36
Chapter 14 Common Channel Inter-Office Signaling (CCIS) .............................................................................. 1

CCIS and ISDN......................................................................................................................................3
Digital and Analog CCIS ........................................................................................................................4
IP CCIS..................................................................................................................................................5
Centralized Billing ..................................................................................................................................8
Centralized E911 – CCIS.....................................................................................................................10
Call Set Up Times................................................................................................................................11
Look Ahead Routing ............................................................................................................................11
Shared Trunk Facilities and Alternate Routing ....................................................................................12
Centralized System Maintenance and Administration .........................................................................13
Centralized Call Accounting/Billing Systems .......................................................................................14
Centralized Voice Processing / Messaging .........................................................................................15
Centralized Attendant Consoles ..........................................................................................................15
Uniform Numbering Plans....................................................................................................................16
CCIS Feature Chart .............................................................................................................................16
System Capacity..................................................................................................................................18
Required Equipment ............................................................................................................................19
IP Specifications ..................................................................................................................................20
Chapter 15 Integrated Services Data Network (ISDN)........................................................................................ 1

ISDN Primary Rate Interface .................................................................................................................1
PRI Services & Features .......................................................................................................................1
Event Based CCIS.................................................................................................................................7
Business Feature List ............................................................................................................................8
ISDN Network Requirements for Layer One (1) ....................................................................................9
Supported Network Services (Trunk provisioned only)..........................................................................9
ISDN PRI Specifications ......................................................................................................................10
ISDN Basic Rate Interface (BRI) .........................................................................................................17
ISDN Basic Rate Interface (BRI) .........................................................................................................18
Documentation.....................................................................................................................................21
Chapter 16 Wireless System ................................................................................................................................ 1

UNIVERGE NEAX 2000 IPS Wireless Communication System (WCS) ................................................1
Wireless Roaming..................................................................................................................................3
Wireless – Short Text Message Notification (OAI) ................................................................................7
System Description..............................................................................................................................12
Wireless Specifications........................................................................................................................15
Wireless LAN .......................................................................................................................................18
Chapter 17 Hotel/Motel System............................................................................................................................ 1

Features.................................................................................................................................................4
Hotel System Capacity ........................................................................................................................13
System Specifications..........................................................................................................................13
Station Equipment ...............................................................................................................................14
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Chapter 18 Call Accounting.................................................................................................................................. 1

AIMWorX™............................................................................................................................................1
Configurations........................................................................................................................................4
The AIMWorX Manager .........................................................................................................................5
Integrating AIMWorX modules...............................................................................................................5
Additional Reference Material................................................................................................................8
Chapter 19 Voice Messaging Systems ................................................................................................................ 1

Voice Mail Integration ............................................................................................................................1
Message Center Interface (MCI) ...........................................................................................................3
NEAXMail AD-120 .................................................................................................................................5
NEAXMail AD-64 .................................................................................................................................10
NEAXMail IM-16 LX.............................................................................................................................24
Chapter 20 System Documentation ..................................................................................................................... 1

UNIVERGE NEAX 2000 IPS Documentation List..................................................................................1

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Chapter 1 Introduction
Overview of NEC
NEC Corporation was founded in 1899. NEC worldwide is built on a strong tradition and global
heritage. NEC was created out of a joint venture between Western Electric from America and a
Japanese investment group. Japan’s first joint venture combined the technology and
manufacturing techniques of the Bell Company with the vision of Japanese investors to form
one of the oldest, established telecommunications companies in the world. NEC is one of only
a small number of companies within the world that have successfully pioneered technology
and delivered products within every major evolution of business communications systems.
NEC, recognized as a worldwide leader in high technology, is one of the few companies
capable of offering a full spectrum of products and systems in computers, communications,
and semiconductor devices.
NEC first established a United States presence over 40 years ago when it opened a sales
office in New York in 1963. Since that time, NEC has broadened its operations in the United
States by expanding into manufacturing, research and software development operations, by
employing 7,000 people and by establishing extensive marketing, sales and service networks
nationwide with revenues exceeding $5.9 billion.
In 1993, NEC introduced a new corporate logo:
. The logo represented the beginning of
a dynamic era in the life of a company with a distinguished past. The changing dynamics of
domestic and international markets mandate that the world’s leading companies evolve with
those changes. In response to this evolution, NEC is building a more responsive business
based on the steadfast foundation of the traditional NEC business philosophy. NEC's
organization has become a recognized leader in linking people and information through
technology. The logo evolved from the simple abbreviation of Nippon Electric Company, in the
late 1890.s, into a dynamic symbol featuring the traditional NEC letters with bold, sharp lines
expressing technology and confidence, combined with gentle curves that are associated with
humanity and friendliness. The logo has a holistic design that each letter expresses an
independent character and at the same time represents the harmony of the logo as a whole or
unit. NEC blue has been designated as the corporate color. This subtle blue was selected to
symbolize human intelligence, life, and the protection of the environment.

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The Invention Age
NEC began its solid tradition with some of the first manual telecommunications systems in Japan.
During the Invention Age, NEC had achieved the following:
In 1900 - NEC began manufacturing its own products
In 1903 - NEC manufactured the first battery phone in Japan
In 1923 - NEC entered the radio transmission field
In 1927 - NEC began automatic switching manufacturing
In 1938 - NEC began manufacturing crossbar switching systems

The Industrial Age
The Industrial Age in America brought great strides in business communication systems with the
development of the electro-mechanical system. NEC successfully delivered step-by-step, crossbar,
and cross-reed technology to the world marketplace. During this era;
In 1950 - NEC began manufacturing some of the world’s first semiconductors
In 1952 - NEC won the Deming prize awarded in communications
In 1956 - NEC introduced step-by-step, cross-reed technology, and crossbar
telephone switching systems to the world marketplace.
In 1959 - NEC developed one of the first transistorized computers
In 1963 - January 17, 1963 NEC entered the American marketplace with a sales
office in New York
In 1965 - NEC became a world innovator in digital transmission equipment

The Electronic Age
NEC entered the Electronic Age with solid-state technology, stored program control, and digital
switching throughout the 1970’s and 1980’s.
In 1972 - NEC developed the world’s first 10K single element LSI chip
In 1976 - NEC introduced the first skinny wire system
In 1978 - NEC introduced the first digital hybrid, the Electra-100
In 1979 - NEC entered the personal computer market
In 1980 - NEC patented its Distributor Processor design (US Patent # 4,210,782).
This little known patented process, invented by Kazunori Fujita, ushered in a new
era of business communications systems through the use of distributed processing,
modular building block concept, and fully integrated voice and data switching. Out of
this patent came NEC’s premier flagship product, the NEAX 2400 IMS
In 1983 - NEC introduced the NEAX 2400 IMS
In 1983 - NEC introduced one of the world’s first super computers
In 1985 - NEC introduced the Electra IMS
In 1986 - NEC created one of the first 4MBit processor chips
In 1988 - NEC opened the Software Development Center in the U.S. market
In 1989 - NEC introduced the NEAX1400 IMS

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The Information Age
Once again, NEC is delivering powerful communications systems. Systems that interface
today’s computers and telephones deliver a broad spectrum of integrated answers to today’s
modern communication challenges.
In 1993 - NEC introduced the world’s first 64M DRAM samples
In 1993 - NEC introduced the NEAX 2400 ICS
In 1994 - NEC introduced the NEAX 2000 IVS
In 1997 - NEC introduced the NEAX 1000 IVS
In 1998 - NEC introduced the NEAX 2400 IMX
In 1999 - NEC introduced the NEAX 2000 IVS2
In 1999 - NEC introduced the NEAX EXPRESS
In 2000 - NEC introduced the NEAX 2400 IPX
In 2001 - NEC introduced the NEAX 2000 IPS
In 2002 - NEC introduced the NEAX 2000 IPS DM
In 2002 - NEC introduced the NEAX 2400 IPXi
In 2003 - NEC introduced the NEAX 2000 IPS DMR
In 2004 - NEC introduced the UNIVERGE SV7000
In 2005 - NEC introduced the UNIVERGE NEAX 2000 IPS
In 2005 - NEC introduced the UNIVERGE NEAX 2000 IPS DML
In 2005 - NEC introduced the UNIVERGE NEAX 2400 IPX
NEC continues to focus its activities on the integration of computers and communications. NEC
research and development, production, marketing, and service based on this integration
positioned it to meet diversifying needs in worldwide markets.

Worldwide Leadership
NEC is committed to the linking people and information through technology, with over 15,000
different products distributed worldwide. Innovative, award-winning products featured in top
publications form the foundation for NEC’s leadership in these high-tech industries.

Components
NEC is the world leader in integrated components with billions of dollars in annual sales. NEC
provides major processors and component chips to other leading manufacturers of
communications, computers and electronic equipment.
Very Large Scale Integrated (VLSI) memories
Micro and Mini computers
Application Specific Integrated Circuits (ASIC)
Semiconductors
A broad spectrum of other electronic components

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Communications
NEC is one of the world’s top suppliers in the production and distribution of communications
equipment. With billions of dollars in sales, NEC is not the largest manufacturer of
communications products, but offers the broadest spectrum of products and a demonstrated
commitment to the total integration and networking of those products.
Telephone company switching systems
Network transmission systems
Business communication systems
Broadcast equipment
Space electronics

Social Contributions
As a global corporation, NEC aims at achieving a higher level of coexistence with society. We
will also strive to become an outstanding corporation respected by all those related to us and
fully capable of offering high standards. In order to achieve this, we appreciate the great
importance of contributing to society as a good corporate citizen and walk hand in hand with
the world toward a better society.

Global Resources
The primary elements in a company’s ability to respond to marketplace, buyer, and customer
demands are the scope of its resources and strategy by which it uses those resources. NEC
has established a worldwide network of human, physical, financial, and intellectual resources.
These are deployed through localized operations, strategically located, professionally staffed,
and thoroughly equipped. This enables NEC to deliver leading edge products and innovative
services to businesses with hundreds of locations across the country or one location across
town.
People
NEC has a global network of dedicated employees with offices in many countries. Our
experienced management, engineers, installers, trainers, and staff pride themselves in a high
level of job performance and customer support. More than 7,000 employees in the U.S. help
maintain NEC’s technological leadership with fresh ideas and the energy to turn them into
reality.

Facilities
The physical resources of the company are the tangible assets that allow it to respond to your
system’s needs and your communication demands in a timely and effective manner. NEC takes
pride in maintaining smooth working facilities with all resources required to provide the highest
level of system service and client support. NEC has invested millions of dollars in facilities,
tools, equipment, and inventory located in hundreds of corporate locations, direct offices,
distributors, and affiliates worldwide to create one of the broadest multi-level support systems in
the industry.
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Assets
NEC’s ability to manufacture and market its offerings depends in large part on its ability to
remain profitable. With a substantial revenue base worldwide, NEC has built a strong financial
and operational base -- a strong foundation of support our customers can depend on.

Innovation
Top training, knowledge, experience, and creativity are the hallmarks of intellectual excellence
and of NEC’s professional staff. Our telecom experts combine the skills and desire needed to
provide the highest level of responsiveness to a system’s service needs and your company’s
communication support requirements.

Community
NEC and its employees are active in worldwide, national, regional, and local community affairs
through the dedication of their time and resources. NEC affiliates and their employees donate
time, money, and equipment to help meet everyday and emergency needs of the healthcare
and other social communities. In addition, the NEC businesses and their people give freely of
their time, effort, and contributions to help meet the needs of local business groups, fraternal,
and other social organizations.

Education
Through worldwide efforts, NEC soundly supports the education of citizens and nations around
the world through financial grants, volunteer work, donations of equipment, and the sharing of
knowledge. Specifically, the NEC Foundation of America is dedicated to encouraging and
supporting the efforts of creative and effective non-profit organizations in the United States and
their staffs, clients, and volunteers who, like NEC, work to develop and share the benefits of
technology with society in order to help improve the quality of human life.

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Chapter 1 Introduction

NEC Unified Solutions, Inc.
Established
(PRIVATE)

1963, as a subsidiary of NEC Corporation, Tokyo, Japan

Headquarters

6555 North State Highway 161
Irving, Texas 75039-2402
PHONE: (214) 262-2000

Revenues
Employees

$3.7 billion
2,800+

Sales & Marketing
Network

65 offices in 20 states throughout the United States, Puerto Rico
and Canada.

Manufacturing
Facilities

HILLSBORO, OREGON
Fiber Optic Transport Systems
Digital Loop Carrier Systems
Very Small Aperture Terminals (Satellite) (VSATs)
Asynchronous Transfer Mode (ATM) Switching Systems
Digital PBXs
Digital Multiplex Systems
Automotive Electronic Systems
Cellular Telephone and Pagers
TECNOLOGIAS NEC DE MEXICO, S.A. DE C.V.
Cellular Telephones and Pagers

Development
Activities

DALLAS, TEXAS
Advanced Switching
Computer Telephony Software
Radio & Satellite Communications Software
Wireless Communication Products
HERNDON, VIRGINIA
Transmission Software & Firmware
SAN JOSE, CALIFORNIA
Transmission Software
HILLSBORO, OREGON
Transmission Hardware & Software

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Major Product
Areas

PUBLIC NETWORKING
Asynchronous Transfer Mode (ATM) Switching
Broadband Access Systems
Digital Loop Carrier Systems
Fiber Optic Transmission Systems
Microwave Radio Systems
Network Management Systems
Satellite Communications Systems
SONET Transport Systems
Internet Access Systems
Narrowband Wave Division Multiplexing Systems
Personal Access Communications System (PACS) Infrastructure
Equipment
CORPORATE NETWORKING
Asynchronous Transfer Mode (ATM) Switches
Audio/Video Teleconference Systems
Telemedicine Products
Distance Learning Products
Data Communications Products
Key Telephone Systems
Local Area Network (LAN) interconnectivity
Private Branch Exchange (PBX) Systems, Software and
Peripheral Products
COMMUNICATIONS TERMINALS
Analog & Digital Telephones
Cellular Mobile Telephones
Facsimile Equipment
Narrowband & Broadband Personal Communications Services
(PCS) Products
Numeric & Alphanumeric Pagers
Enhanced Wireless Products (messaging, voice, data &
accessories)

Scope Of Business

Development, manufacturing, sale, service and support of
Telecommunications Products and Network Management
Systems to the public and private sectors.
Engineering and Marketing of Integrated Computer and
Communications (C&C) Systems and Networks including
Broadband Multimedia Products.
Customization of ATM Switching Systems, Business
Communication Systems, Data Communications, Radio and
Transmission Equipment.

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Vertical Markets
The UNIVERGE NEAX 2000 IPS is a full-featured IP based communications system that
addresses the telecommunications market by providing a rich feature set of both Key and PBX
features. The UNIVERGE NEAX 2000 IPS is positioned to provide a more cost effective solution
for the small to medium size business, hotel property or networked environment. The
UNIVERGE NEAX 2000 IPS provides unparalleled investment protection by allowing the user to
take full advantage of today and tomorrow’s technology advancements. The following is a list
of some of the markets the IPS easily adapts to fit customer requirements.
Business
Telemarketing
Communication
Tenant Services
Education
Health Care

Legal
Hospitality
Financial
Government
Transportation

Competitive Benefits and Advantages
Advanced Technology (complete IP system on one card)
Pentium Equivalent AMD CPU
PSTN Gateway (requires trunk cards)
Supports Both IP & TDM
Both TDM & IP Totally Non-Blocking
H.323 Connectivity
Smaller Foot Print
Low Power Consumption
Wide Selection of IP and TDM Terminals
Economical Incremental Licensing Costs
Reduced Hardware Costs
Survivability at the Remote Site
Mobility and Continuance with in a Remote PIM Network

The UNIVERGE NEAX 2000 IPS provides a unique set of advantages to users who seek an
advanced information system that is both flexible and dependable. Through the use of state-ofthe-art computer controlled telecommunications technology, NEC is able to provide the
following advantages:
1.) Full-Featured System – Station users have access to more than 300 service features that
enhance user productivity, reduce operating costs, and improve communication efficiency. In
addition, the control, network and interface positions can accommodate features, services and
subsystems as required by specific applications.
2.) Network Integration - The UNIVERGE NEAX 2000 IPS offers business, industries, hospitals and
hotel/motels the ability to access an extensive array of information processing and management
services by serving as the central controller of an integrated information network. The UNIVERGE
NEAX 2000 IPS reflects the philosophy of NEC Corporation to integrate C&C technology, and can
provide many voice and non-voice services.
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3.) Flexible Line Size - Innovative Modular hardware and software design allows from 64 ports up to
1020 ports in a main plus remote environment. Modularity gives the system the ability to expand
from its minimum configuration to its maximum capacity as the need arises. This unique expansion
capability allows the system to grow in a cost affective manner as the user requirements expand.
Your initial system investment is protected through growth capability.
4.) Energy Saving and Space Savings - Through employment of state-of-the-art technology in the
system circuitry design, NEC has been able to reduce power consumption. As a result, the current
consumption of the system has been reduced to 50% of that of any conventional electronic PBX
systems. This energy saving oriented system design allows for the use of much smaller capacity
main power equipment and air conditioning equipment. In parallel with the energy savings, the
space requirement for the system has been reduced to one third when compared with that of a
conventional electronic PBX system.
5.) Building Block Configuration - In a conventional switching system, various kinds of equipment
are mounted in a cabinet group and are connected to each other by use of connecting cables. The
UNIVERGE NEAX 2000 IPS, however, uses a building block modular design. When installing the
system, the required blocks are placed on top of each other in a building block formation and
interconnected by round bus cables.
6.) Flexible Interface Ports - The UNIVERGE NEAX 2000 IPS employs a Universal Port architecture
that has the flexibility to accommodate station terminal equipment, trunks and adjunct processors.
This universality allows the UNIVERGE NEAX 2000 IPS to optimize use of slot space and lower
expansion costs.
7.) High Reliability - The UNIVERGE NEAX 2000 IPS is designed and manufactured to provide the
highest level of system reliability. The UNIVERGE NEAX 2000 IPS is designed with such features as:
remote maintenance, distributed call processing, error-correcting memory, battery backup, and
automatic system alarm indications to insure unsurpassed reliability. Only the finest components
have been used. In addition, through the employment of LSI and custom LSI and VLSI technology,
the number of component parts has been greatly reduced, thus lessening possible failures and
insuring continuous operation.
8.) Intelligent Attendant Console - The UNIVERGE NEAX 2000 IPS attendant console, SN716 Desk
Console, is a compact, desktop unit equipped with non-locking keys, Light Emitting Diodes, and a 4
x 40 character Liquid Crystal Display. The LED’s provide continuous information relative to the
status of calls in progress. The display provides station and trunk identification, class of service and
the number of calls waiting. The LCD changes with different call states and instructs the user which
Multifunction keys are available for each state. Keys such as Busy Verify, DND Override, etc. only
appear when needed. The Multi-function Keys reduce the number of different buttons and greatly
simply operation.

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9.) Intelligent Digital Multifunction Terminal - In addition to supporting conventional station
equipment, the UNIVERGE NEAX 2000 IPS can be equipped with the Dterm series digital electronic
multifunction terminals. The Dterm terminals are intelligent microprocessor controlled terminals,
which enhance the feature capabilities offered by the system and provide the service of
conventional key telephones over 1-pair wiring. The Dterm instrument may be equipped with an
interface adapter to allow simultaneous voice and data switching, without compromising the voice
communication system.
10.) Ease of Installation - Because the UNIVERGE NEAX 2000 IPS uses pre-assembled modules
and plug-in type circuits packs, it is easy to install. Wiring connections, both internal and external,
are made through simple-to-use standard plug-ended cables. In addition, with each unit and system
having been fully factory tested prior to shipment, potential obstacles to easy installation have been
held to a minimum.
11.) Ease of Maintenance - Because the system is constructed with first quality components,
reliability is high and operation is trouble free. However, if a minor fault occurs, the self-diagnostic
programs will detect the fault, and automatically make the needed corrections. If the problem is
beyond the internal correction capabilities of the system, the self-diagnostic programs will
automatically print the nature of the fault and the involved unit is identified on the man-machine
interface equipment. The faulty plug-in unit can then be quickly replaced with little or no interruption
of service.
12.) Flexible Numbering Plan - The UNIVERGE NEAX 2000 IPS provides flexible numbering
assignment to meet all forms of network integration service.
13.) Future Capabilities - Because the UNIVERGE NEAX 2000 IPS uses a stored program control,
performance enhancements and new features can be easily incorporated by simple changes in
software. The UNIVERGE NEAX 2000 IPS can be upgraded by using the most current software
release and, when necessary, additional hardware. This means that the system will not become
obsolete.
14.) Cost Controls - With telecommunications costs growing, it is becoming increasingly important
to control them. The UNIVERGE NEAX 2000 IPS makes it possible for you to get a firm grasp on
telecommunications costs. Through the use of such features as least cost routing, class of service
and detail call recording, cost reduction and control are possible.

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Chapter 1 Introduction

UNIVERGE NEAX 2000 IPS
In the modern information-based economy, timely access to accurate information is an
essential resource for every successful organization. In many cases it is the primary
commodity being created, bought, or sold and anything less than the best communication
system puts your organization at a disadvantage. The complex interconnections of voice,
print, and data communications demand a powerful but elegant system capable of taking you
through the 21st century.
The UNIVERGE NEAX 2000 IPS (Internet Protocol Server) is a full-featured IP based
communications system providing a rich feature set with pure Voice over IP (VoIP)
communications (peer to peer connections), across corporate Local and Wide Area Networks
(LAN and WAN). DtermIP telephones are designed to provide a converged infrastructure at
the desktop, with a 100 Base T Ethernet connection to the LAN and built-in hub for a PC
connection to the telephone itself. The system can provide peer-to-peer connections between
DtermIP telephones with voice compression, offering existing Dterm Series i telephone
features. On the WAN side, the system can provide peer-to-peer connections over IP
networks with the voice compression, on a CCIS basis (CCIS over IP) or Remote PIM (Remote
PIM over IP).
The UNIVERGE NEAX 2000 IPS can provide legacy station/trunk interfaces to support the
existing Time Division Multiplexing (TDM) based infrastructure, such as analog telephones,
analog networks, and digital networks (T1/E1, ISDN etc.). At maximum configuration, the
system can provide 1020 ports for IP and legacy devices, and 256 ports for Application cards.
Communications between legacy stations/trunks and DtermIP telephones/IP networks are
made via IP PAD, which converts packet-based voice data to TDM-based voice data, and vice
versa. Both peer-to-peer connections and TDM-based connections are controlled by the Main
Processor (MP) card. The MP card incorporates a built-in Device Registration Server (DRS)
and a single interface point of IP connection to IP telephone, MATWorX, Voice Mail and
OAI/ACD servers.

UNIVERGE NEAX 2000 IPS users have access to hundreds of service features that are used in
building unique telephony applications that enhance productivity, reduce operating costs and
improve communications efficiently. The innovative modular hardware and software design
allows efficient, effective growth within each module from its minimum to its maximum
configuration. The UNIVERGE NEAX 2000 IPS software design is as advanced as its hardware.
It ensures the system will support evolving applications and have the reliability needed to
compete in today's world and into tomorrow's. The software is designed with modularity in
mind. Together, these modular building blocks allow customers to initially buy what they need
and add capacity and capabilities as the business demands, resulting in a greater degree of
cost control for new installations and for upgrades to features, capacities and the software
versions.

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The illustration below shows a typical system layout.
UNIVERGE NEAX 2000 IPS

In-skin VMS

UNIVERGE NEAX 2000 IPS
ZT

MATWorX
(Via IP: CCIS)

In-skin Router

PSIII

PSTN
CCIS over IP
LC/DLC

COT

Intranet

IP-PAD

MATWorX
(Via RS232C)

PSTN
MP

FTP
Server

Switching Hub

IPT
(H.323
Handler)

H.323 GK
Router

Router
UNIVERGE
DMR
NEAX IPS

(100Mbps)

DtermIP
Switching
Hub
Router

Client PC

MATWorX
(Via LAN)

Dterm
OAI
DHCP
Server Assistant Server

DtermIP
Remote PIM over IP
with Survivability

DtermIP
Client PC

Dterm
INASET

Dterm SP30

Station to Station Connection
For DtermIP to DtermIP connection (Peer to Peer connection), the voice data is transmitted and
received directly between DtermIPs on the LAN. For Dterm Legacy terminal connection, the IPPAD card and VCT card are required to transmit and receive the voice data. These cards are used
to control and convert the voice data. The MP card in either of the connections above manages the
control signals.

CCIS Connection
DtermIP to DtermIP connection (Peer to Peer connection) via CCIS is available only when the
destination office is UNIVERGE NEAX 2000 IPS or UNIVERGE NEAX 2400 IPX. The system
provides only Point to Multipoint connection.

Maintenance
MATWorX IPS is used as the maintenance program for the UNIVERGE NEAX 2000 IPS. Direct
connection (RS-232C), Modem connection and LAN (TCP/IP) connections are available to connect
to the MAT (Maintenance Administration Terminal).
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Chapter 1 Introduction

Hybrid System of IP (peer-to-peer connection) and TDM Switching
The UNIVERGE NEAX 2000 IPS supports both pure IP switching (peer-to-peer connections) and Time
Division Switching (TDM). The pure IP switching is provided for communications between DtermIPs
and for CCIS/Remote PIM connections with another UNIVERGE NEAX 2000 IPS/IPS DM/IPS
DML/2400 IPX (CCIS over IP or Remote PIM over IP). On the other hand, the TDM switching is
provided for communications between legacy stations/trunks. Connections between DtermIP/CCIS or
Remote PIM over IP and legacy stations/trunks are made via IP PADs, which converts packet-based
voice data to TDM-based voice data, and vice versa.

Built-in DRS (Device Registration Server) on MP
The UNIVERGE NEAX 2000 IPS incorporates DRS (Device Registration Server) on MP, which provides
a Log-in/Log-out management of DtermIP including Registration, Authentication. Also the built-in DRS
can be inter-worked with DHCP server to provide easy administration of IP address.

Reduced Hardware with IP based Architecture
The DtermIPs accommodated in a LAN do not require a DLC card because they can be interfaced
directly with the LAN and connected on a peer-to-peer basis. When a DtermIP is connected with
station/trunk, which is accommodated in TSW, the speech path between LAN and TSW is made via IP
PAD under the call processing control of MP. The number of DtermIPs can be simply expanded by
adding the terminal itself, an IP Seat License and IP PADs if traffic volume is increased. With this
system architecture, the hardware such as DLC, PIM, Power Supply etc. is reduced and easy cost
effective move, add, change is realized.

Office Data Backup Enhancement
The office data of UNIVERGE NEAX 2000 IPS is stored in Flash ROM.

Various Installation Methods
To meet the specific needs of the customers’ environment, the UNIVERGE NEAX 2000 IPS provides the
following installation methods:
Floor Standing Installation
Wall-mounting Installation
IEC standard 19-inch Rack-mounting Installation

Unified Circuit Card Size
All the circuit cards for UNIVERGE NEAX 2000 IPS are designed in one size (PN-type), and installed in
the PIM. This maximizes the efficiency of slot utilization of the PIM.

High Density Line/Trunk Cards
Major line/trunk cards used in UNIVERGE NEAX 2000 IPS are provided with 8 circuits per card. This
allows the physical system size to be compact.

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Universal Slot
One PIM provides 12 card slots for Line/Trunk (LT). Also, these card slots can be used for Application
Processor (AP) cards without complicated limitations. As this makes easy quotation and installation,
more number of AP cards can be mounted in one PIM.

DC/DC Power Supply for –48V
The PIM houses optional DC/DC Power Supply for the cards which require –48V power such as CSI
card used for interface of Cell Station (CS) of wireless system. Since this power supply is mounted in
the space under the AC/DC power, additional Power Module/card slots are not required.

Extended Application Processor (AP) Port Capacity
The UNIVERGE NEAX 2000 IPS provides a maximum of 256 AP ports and it is independent of the 512
physical ports for legacy Line/Trunk (LT), therefore, more AP cards can be used in the system such as
T1/E1 digital link, etc.

Dual MP System
The system complies with dual control system on Main Processor.
Note: Since the system employs Cold Standby processing in MP changeover, the calls in progress are
terminated as a result of the MP changeover. Also, during the MP changeover, the call originating/receiving and
service feature access are not effective. (It takes about 30 to 60 seconds to complete the MP changeover.)

UNIVERGE NEAX IPS DM
The UNIVERGE NEAX IPS DM (Internet Protocol Server Distributed Model) is equipped with all
the features and functions of the UNIVERGE NEAX 2000 IPS, with a smaller space
requirement. It is a full-featured PBX that supports advanced networking, pure peer-to-peer IP
telephony connectivity and traditional TDM switching capabilities. Designed primarily for pure
converged IP networks, the UNIVERGE NEAX IPS DM can also accommodate a mixed (i.e.,
TDM and IP) converged IP network or standalone solution.
The UNIVERGE NEAX IPS DM supports up to 952 peer-to-peer IP stations and 56 TDM ports
in a single modular chassis. Up to two chassis can be stacked providing maximum capacity of
112 legacy TDM ports while still supporting as many as 888 peer-to-peer IP stations or more
depending on the amount of TDM stations used. It uses the same CPU, line/trunk cards,
application processor cards and software of the UNIVERGE NEAX 2000 IPS and comes
equipped for 19” rack mounting. It offers superior port density; each chassis only occupies two
Rack Units (2RU).

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UNIVERGE NEAX IPS DML
The UNIVERGE NEAX IPS DML (Internet Protocol Server Distributed Model Local) is a
UNIVERGE NEAX IPS DM that has been optimized for Small Office Stand Alone IP Solution
with from 10 to 100 IP telephones. The UNIVERGE NEAX IPS DML uses the SPN-CP31C as
the Main Processor. The SPN-CP31C is a cost down CPU to compete with Mitel 3100,
NBX25, and CISCO 2600 Series. This system targets users with up to 112 TDM station, 128
IP stations and can be used as an end point in a peer-to-peer CCIS network. The following
options that are built-in on the CP24 are not available with the CP31:
No built-in DAT.
Only one RS Port.
No built-in DK (external/relay key).
No MN Alarm Indication

UNIVERGE NEAX IPS DMR
The UNIVERGE NEAX IPS DMR (Internet Protocol Server Distributed Model Remote) is an
UNIVERGE NEAX IPS DM that has been optimized for Remote PIM over IP applications. The
UNIVERGE NEAX IPS DMR uses the SPN-CP31C as the Main Processor. The SPN-CP31C
is a cost down CPU to compete with Mitel 3100, NBX25, and CISCO 2600 Series. This
system targets users who have up to 30 relatively small offices that accommodate 10-30
extensions at the Remote Site. The MP card at Remote Site has the same system data as
that at Main Site, because Remote Site automatically gets the data from Main Site at the time
of setup. In normal operation, Main Site automatically copies the system data to Remote Site
through the network once a day.
The following options that are built-in on the CP24 are not available with the CP31 in an
UNIVERGE NEAX IPS DMR:
No built-in DAT.
No built-in Modem
Only one RS Port.
No built-in DK (external/relay key).
No MN Alarm Indication

System Outline for Remote PIM
The MP card at Main Site controls system processing, and Remote Site follows the Main Site.
Remote Site can accommodate most terminals and trunks such as Dterm, Single-Line
telephone, DSS/BLF Console, DtermIP, COT, ISDN, etc. The Attendant Console, Dterm
Attendant position, and Add-on Module are not supported at the Remote Site.
Local Switch (TDSW) at Remote Site controls connections within the Remote Site if possible.
In the case of connections between Main-Remote and Remote-Remote, the voice path is
connected via Peer-to-Peer or IP-PAD.
If the communications between Main-Remote are interrupted, the Remote Site survives by itself
after a system reset.
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Advantages of Remote PIM
The system regards the terminals accommodated in both Main Site and Remote Site as the
extensions in the same office. Therefore, the service transparency is superior to CCIS.
Remote PIM over IP has no limitation of distance between Main and Remote.
Remote Site has a switching function at local. This provides the effective configuration of C.O.
line. In addition, the Remote Site can accommodate AP cards. This is an advantage to
accommodate ISDN lines especially.
The Remote Site survives by itself even if the link between Main and Remote is disconnected.
Therefore, the impact to users at the Remote Site will be smaller if the link between Main and
Remote is disconnected.
This feature can reduce the bandwidth used on the WAN that is connected to CO lines at
Remote Site, rather than DtermIP at remote location or the Media Converter (MC)
accommodation.

Characteristics of the UNIVERGE NEAX IPS DM/IPS DML/IPS DMR
Compact and Small Size MODULAR CHASSIS
One MODULAR CHASSIS provides 8 card slots /56LT ports (8 virtual LT ports are available per
MODULAR CHASSIS in addition to 56LT ports).
2 MODULAR CHASSIS can be used per system for IPS DM/DML/IPS DMR
MP (Main Processor)
PN-CP24A/B/C for IPS DM, the same MP as the UNIVERGE NEAX2000 IPS.
PN-CP31C for IPS DML and PN-CP31A/C for IPS DMR

Power Failure Transfer (PFT)
Power Failure Transfer (PFT) for the IPS DM is provided with PZ-4PFTA card. The PZ-8PFTB is not
available for the IPS DM.
Installation Methods
The UNIVERGE NEAX IPS DM/IPS DML/IPS DMR can be installed on a desktop or into the 19-inch
rack. Wall Mount Installation is not available.
Remote PIM over IP with Survivability
The UNIVERGE NEAX 2000 IPS can have a PIM installed at a remote site through an IP network. At
the main site, the UNIVERGE NEAX IPS/IPS DM is installed and UNIVERGE NEAX 2000 IPS/IPS
DM/IPS DMR or IPS Retro is installed at the remote site. The main site controls call processing and
service feature access for station users located at both the main and remote sites. When the Remote
PIM cannot be connected with main site due to the IP network and/or main PBX failure, the Remote
PIM initializes the system and re-starts operation by its own Main Processor (survival mode). In the
survival mode, almost all service features are provided to the station users accommodated in Remote
PIM. When the IP network/main PBX recovers, the Remote PIM can be restored to normal mode with
a system initialization by manual operation or automatically (Selectable by system data setting).
IPS DMR with CP31A/C MP
IPS PIM with CP24A/B/C or CP27A MP
IPS Retro with CP26 or CP28 MP

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Stand Alone System Capacity

LT Card

IPS System Capacity (Single MP System)
Capacity Per PIM
Note
Item
PIM1 PIM2 PIM3 PIM4 PIM5 PIM6
(No. of Ports)
64
128
192
256
320
384
(No. of Cards)
(No. of Ports)

AP Card
(No. of Cards)
Total Number of Lines (Single Line Tel. + Dterm)
IP PAD
(No. of Channel)

PIM7
448

PIM8
512

Max. 256 ports per system
12
64

128

192

24
320

256
128

384

448
512
256

192

Standard

128

192

256

320

384

448

512

Long

144

192

240

288

336

384

Standard

128

192

256

320

384

448

512

Long

120

144

168

192

Dterm IP/Dterm IP INASET (PTP Connection)
Dterm PS
Cell Station (CS) / Zone Transceiver (ZT)

952

888

824

632

568

504

760
696
512
64
80

112

128

ISDN Station

112

128

Loop Start

128

192

256

DID w/4DID

144

192

240

256

2W E&M

120

144

168

192

4W E&M
CCIS Trunk (Peer to Peer Connection)
1.5M
DTI/CCIS Digital Link
2MI
1.5M/2M (PRT)
ISDN
4BRT (card)
IP Trunk

144

168

192

6
1

12
2

18
3

24
6

24
7

24
8

PFT Connections

Single Line Telephone
(Lines)
Dterm (Lines)

Central Office Trunk (Lines)
Tie Line Trunk (Lines)

3-Party Conference
6-/10-Party Conference

96
120
Max. 127
DTI: 10, CCIS: 8
8
8
24
24
4
5
32

Max. 16 conference groups per system
6-Party

Max. 4 conference groups per system

10-Party

Max. 2 conference groups per system

32-Party Conference

Max. 8 conference group per system

Built-in Router

Max. 8 cards per PIM

DTMF Sender

Max. 32 circuits per system

DTMF Receiver

Attendant Consoles

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IPS System Capacity (Single MP System cont’d)
Capacity Per PIM

Item
PIM1

Attendant Terminal (Dterm ATT Position)

PIM2

PIM3

PIM4

PIM5

Note
PIM6

PIM8

Max. 8 sets per system
Max. 2 Interface ports (RS232)
Max 1 interface port (IP) per system
Max. 1 Interface port (RS232)
Max 1 interface port (IP) per system

SMDR Interface
PMS Interface
ACD / MIS or OAI Interface

Max. 1 interface port per system

Remote PIM over IP (Number of PIM for Remote
Sites)

Up to 30 (depending on network)

DID Dial Conversion

1000

Call Forwarding-Outside Set

496

Authorization. Code / Forced Account Code /
Remote Access to System(DISA)Code

3000

Message Reminder Set

1024

Name Display / Guest Name Display

512

Speed Calling-Station (Station Speed Dial) Set

10000

MP built-in SMDR Call Record

1024

Note: System Capacity is for Main site only.
Capacity.

Page 1-18

PIM7

For Total System Capacity see IP Remote Network System

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IPS System Capacity (Dual MP System)
Capacity Per PIM

Item
LT Card
AP Card

(No. of Ports)
(No. of Cards)
(No. of Ports)
(No. of Cards)

Total Number of Lines (Single Line Tel. + Dterm)
IP PAD
(No. of Channel)
Standard
Single Line Telephone (Lines)
Long
Standard
Dterm (Lines)
Long
Dterm IP/Dterm IP INASET (PTP Connection)
Dterm PS
Cell Station (CS) / Zone Transceiver (ZT)
ISDN Station
Loop Start
Central Office Trunk (Lines)
DID w/4DID
2W E&M
Tie Line Trunk (Lines)
4W E&M
CCIS Trunk (Peer to Peer Connection)
1.5M-AMI
DTI/CCIS Digital Link
2M-AMI
1.5M/2M (PRT)
ISDN
2BRT (card)
4BRT (card)
IP Trunk
PFT Connections

PIM2

PIM3

PIM4

PIM5

PIM6

PIM7

PIM8

64
11

128
23

192
35

256
47

320
59

384
71

448
83

512
95

11
64

23
128

64
44
64

128
92
128

Max. 256 ports per system
24
192
256
320
384
128
192
192
256
320
384
140
188
236
284
192
256
320
384

22
952

46
888

70
824

16
16

32
32

48
48

64
44
22
22

128
92
46
46

192
140
70
70

11
6

23
12

256
256
188
236
94
118
94
118
Max. 127
DTI: 10, CCIS: 8
8
8
24
18
24
24

3-Party Conference
6-/10-Party Conference

Note

PIM1

94
118
760
696
512
64
80
64
80

448
512
256
448
512
332
380
448
512

142
632

166
568

190
504

96
96

112
112

128
128

256
256
142
142

256
256
166
166

256
256
190
190

Max. 16 conference groups per system
6-Party
10-Party

Max. 4 conference groups per system
Max. 2 conference groups per system

32-Party Conference

Max. 8 conference group per system

Built-in Router

Max. 8 cards per PIM

DTMF Sender

Max. 32 circuits per system

DTMF Receiver

Attendant Consoles

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IPS System Capacity (Dual MP System, Cont’d)
Capacity Per PIM

Item
PIM1

Attendant Terminal (Dterm ATT Position)

PIM2

PIM3

PIM4

PIM5

Note
PIM6

PIM7

PIM8

Max. 8 sets per system
Max. 2 Interface ports (RS232)
Max 1 interface port (IP) per system
Max. 1 Interface port (RS232)
Max 1 interface port (IP) per system

SMDR Interface
PMS Interface
ACD / MIS or OAI Interface

Max. 1 interface port per system

Remote PIM over IP (Number of PIM for Remote
Sites)

Up to 30 (depending on network)

DID Dial Conversion

1000

Call Forwarding-Outside Set

496

Authorization. Code / Forced Account Code /
Remote Access to System(DISA)Code

3000

Message Reminder Set

1024

Name Display / Guest Name Display

512

Speed Calling-Station (Station Speed Dial) Set

10000

MP built-in SMDR Call Record

1024

Note: Capacity is for Main site only. For Total System Capacity see IP Remote Network System Capacity.

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IPS DM/IPS DML System Capacity
Capacity Per MC
Number of PHYSICAL MODULAR CHASSIS
1
2
LT card

No. of ports

Note 1

No. of cards
No. of ports

AP card

112

Max. 256 ports per system

No. of cards

Total number of lines (Single Line Telephone + Dterm)

112

IP-PAD

No. of channel
8LC
Long Line

32
56

64
112
Not Available

Standard

112

Long Line

952

888

Single Line Telephone
(Lines)
Dterm(Lines)

DtermIP/DtermIP/INASET/SP30(PTPConnection)Note 3
ISDN Station
Central Office Trunk (Lines)

Loop Start

112

DID w/4DID

2W/4W E&M

CCIS Trunk (Peer to Peer Connection)
DTI/CCIS Digital Link
Note 2
ISDN

Max. 127

1.5M

DTI: 10, CCIS: 8

1.5M/2M(PRT)

4BRT (card)

IP Trunk
PFT Connections
3-Party Conference
6-/10-Party Conference
32-Party Conference

6-Party
10-Party

4
8
Max. 16 conference groups per
system
Max. 4 conference groups per system
Max. 2 conference groups per system
Max. 8 conference
5
groups per system

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Chapter 1 Introduction

IPS DM/IPS DML System Capacity Cont
Capacity Per MC
Number of PHYSICAL MODULAR CHASSIS
1
2
Built-in Router

Max. 1 card per Site

DTMF Sender

Max. 32 circuits per system

DTMF Receiver

SN716 Desk Console

32
8

Attendant Terminal (Dterm ATT Position)
SMDR Interface
PMS Interface
ACD / MIS or OAI Interface
Remote PIM over IP (Number of PIM for Remote
Sites)
Note 3
DID Dial Conversion

Max. 8 per system
Max. 2 Interface ports (RS232)
Max 1 interface port (IP) per system
Max. 1 Interface port (RS232)
Max 1 interface port (IP) per system
Max. 1 Interface port per system
Up to 30 (depending on network)
1000

Call Forwarding-Outside Set
Authorization Code / Forced Account Code / Remote
Access to System(DISA) Code
Message Reminder Set

496
3000

Name Display / Guest Name Display

512

1024

Speed Calling-Station (Station Speed Dial) Set

10000

MP built-in SMDR Call Record

1024

Note 3

Note 1:

Each Modular Chassis has 8 Virtual LT Ports that can only be used to expand the PAD
channels from 24 to 32 using the 8IPLA w/24IPLA.

Note 2:

The total number of trunk line and DTI channel shall be 256 or less. (Each trunk line and DTI
channels are required to assign the “Trunk Number” by system data programming and
maximum number of system parameter for “Trunk Number” is 256.)

Note 3:

IPS DML only supports a maximum of 128 IP stations; IPS DML does not support built-in
SMDR. The IPS DML is a Stand Alone Only solution; Remote PIM’s off the IPS DML are not
supported.

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IP Remote Network Capacity
Total System Capacity (Main plus Remote)
Item

Capacity

LT Ports
AP Ports
Analog Single Line Tel. + Dterm
IP PAD
(No. of Channel)
Dterm IP/Dterm IP INASET (PTP Connection)
Dterm PS
Cell Station (CS) / Zone Transceiver (ZT)
ISDN Station

1020
256
980
256
952
512
128
128

Central Office Trunk (Lines)

256

Tie Line Trunk (Lines)
2W/4W E&M
CCIS Trunk (Peer to Peer Connection)
DTI/CCIS Digital Link
1.5M/2M
1.5M/2M (PRT)
ISDN
4BRT (card)
IP Trunk
PFT Connections
3-Party Conference
6-Party
6-/10-Party Conference
10-Party
32-Party Conference
Built-in Router
DTMF Sender/Receiver
Attendant Consoles
Attendant Terminal (Dterm ATT Position)
SMDR Interface
PMS Interface

192
127
DTI: 10/CCIS: 8 Links
8
24
8
64
Max. 16 conference groups
Max. 4 conference groups
Max. 2 conference groups
Max. 8 conference groups
1 per Site
Max. 32 circuits
8
Max. 8 sets
Max. 2 Interface ports (RS232)
Max 1 interface port (IP) per system
Max. 1 Interface port (RS232)
Max 1 interface port (IP) per system
Max. 1 interface port
Up to 30 (depending on network)
1000
496

ACD / MIS or OAI Interface
Remote PIM over IP Note
DID Dial Conversion
Call Forwarding-Outside Set
Authorization Code / Forced Account Code / Remote
3000
Access to System(DISA)Code
Message Reminder Set
1024
Name Display / Guest Name Display
512
Speed Calling-Station (Station Speed Dial) Set
10000
MP built-in SMDR Call Record
1024
Note: Remote PIM networks using the IPS DML as the main are not supported.

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Page 1-23

Chapter 1 Introduction

IPS DMR Capacity
Number of PHYSICAL MODULAR CHASSIS
LT card

Note 1

AP card
IP-PAD
Single Line Telephone (Lines)
Dterm(Lines)

No. of ports
No. of cards
No. of ports

Capacity Per MC
1
2
56
112
7
14
Max. 256 ports per network

No. of cards

No. of channel
8LC
Standard

32
56
56

64
112
112

Long Line

DtermIP/DtermIP/INASET(Peer to Peer Connection) Note 2
Central Office Trunk (Lines)
DTI
ISDN

128

Loop Start

112

DID w/4DID

2W/4W E&M

1.5M

1.5M(PRT)

4BRT (card)

PFT Connections
IPS PIMMJ (As Remote PIM) Capacity
Number of PHYSICAL PIMS
LT card
AP card
IP-PAD
Single Line Telephone (Lines)
Dterm(Lines)

No. of ports
No. of cards
No. of ports

Capacity Per PIM
1
2
64
128
8
16
Max. 256 ports per network

No. of cards

No. of channel
8LC
Standard

32
64
64

64
128
128

Long Line

DtermIP/DtermIP/INASET(Peer to Peer Connection) Note 2
Central Office Trunk (Lines)
DTI
ISDN

128

Loop Start

128

DID w/4DID

2W/4W E&M

1.5M

1.5M(PRT)

4BRT (card)

PFT Connections
8
16
Note 1: Each Modular Chassis has 24 Virtual LT Ports that can only be used to expand the PAD
channels from 8 to 32 using the 8IPLA w/24IPLA.
Note 2: Remote PIM’s Support up to 2 Virtual PIM’s for assignment of DtermIP/DtermIP/INASET only.

Page 1-24

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
Issue 6

Chapter 2 System Architecture
The UNIVERGE NEAX 2000 IPS family consists of the IPS, IPS DM, IPS DML and IPS DMR.
These UNIVERGE NEAX 2000 Internet Protocol Servers fuse existing NEC technologies with
dynamic advancements in hardware and software to satisfy the most stringent system
requirements. NEC's modular design, sensible migration, comprehensive network solutions,
and the ability to customize functionality to match specific business applications are just a few
of the NEC benchmarks that the new IPS builds upon.

UNIVERGE

NEAX 2000 IPS users have access to hundreds of service features that enhance

productivity, reduce operating costs, and improve communications efficiently. The innovative
modular hardware and software design allows efficient, effective growth within each module
from its minimum to its maximum configuration. This is achieved by a unique building block
architecture allowing growth without the loss of existing hardware or software.
Voice and Data Switching
The UNIVERGE NEAX 2000 IPS systems support advanced applications such as Tandem Networking,
Centralized Attendant Service, Direct Digital Interface and other voice features. The system's nonblocking architecture and distributed processor control hierarchy are designed to support the traffic and
control the load generated by voice and data switching. Station users may perform simultaneous voice
and data transmission at speeds up to 19.2Kbps over universal 1-pair wiring without the use of
modems. Proprietary Digital Instruments (Dterm) may be provided to increase system flexibility and
eliminate the need for conventional multiple line stations with their associated control equipment and
cable plant requirements.

Hardware Architecture
The UNIVERGE NEAX 2000 IPS systems employ the latest advancements in Large Scale
Integration (LSI) circuits and component manufacturing techniques to create a highly reliable
and serviceable communications system. The IPS has changed from NEC's traditional PROM
based CPUs to a Flash ROM based for system software and RAM for data programming. The
UNIVERGE NEAX 2000 IPS architecture consists of three major functional components:
Distributed Controller, Digital Switching Network, and Port Interface.
1. Distributed Controller - The Distributed Controller is composed of distributed
multiprocessing units, generic memory, database instructions, system interface, and
interface ports for system maintenance and administration.
2. Digital Switching Network - The Digital Switching Network consists of a non-blocking
digital time division switch, allowing all ports to be used simultaneously.
3. Port Interface - The Port Interface provides access to the public and private network for
various types of terminal devices, including digital and analog telephones, data terminals,
computers and subsystems such as Voice Mail Systems, Data Switch Networks, and
related communication and information services.
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Chapter 2 System Architecture

Hardware Design
The UNIVERGE NEAX 2000 IPS hardware is designed to provide the best of the following:
Flexibility
Capacity
Reliability
Optimum use of space
Minimal environmental requirements
Ease of installation and maintenance
A unique modular design is employed throughout the UNIVERGE NEAX 2000 IPS line of products. As
additional equipment modules are required, they are stacked on top of each other, rather than mounting
the equipment in conventional frames or cabinets. This innovative method reduces installation time,
avoids the use of bulky frames and provides for manageable future expansion, virtually eliminating the
possibility of outgrowing your UNIVERGE NEAX 2000 IPS.

Fusion of RAM and ROM Programs
The UNIVERGE NEAX 2000 IPS incorporates the flexibility of Random Access Memory (RAM) for Office
Data Memory and Flash ROM that contains the operating system. This fusion of divergent memory
technologies results in high performance 32-bit processing reliability in system operation as well as fast
recovery time in case of system power loss. This unique configuration provides improved efficiency for
updates to the generic program by floppy disk, and simplifies upgrades for system features and
capabilities throughout its life cycle.

Main Processor (MP) with Integrated Functionality
The UNIVERGE NEAX 2000 IPS Main Processor (MP) is a heart of pure IP connections and TDMbased connections. The MP employs a high-speed CPU, which is equivalent with Pentium. With this
processing power and System On Chip (SOC) technology, the MP integrates Device Registration
Server (DRS), AP01 (OAI) functions*, which are provided by an additional card in the previous IVS2.
Also, by means of today's advanced LSI technology, the MP card size is minimized and On-board
Ethernet Interface Card is mounted on MP without using an additional slot space in PIM. This interface
card is linked with LAN for call control processing of DtermIP and inter-work with MATWorX and OAI
server. (*RS-232C interface, Authorization Code service and FLF facility are deleted.)

Application Processors
The processing architecture of the UNIVERGE NEAX 2000 IPS allows the system to be configured with
Application Processors that provide processing capacity for a specific application or multiple
applications. These EPROM-based processors support applications such as:
ACD Processor for call center applications
SMDR for call accounting
Hotel Processor for Hospitality applications
Message Center Interface (MCI)
Open Applications Interface (OAI), which connects to external computing devices to support a
wide range of host computer-supported features.
These application processors provide the power and flexibility to meet specific users’ applications and
traffic needs.
Page 2- 2

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Chapter 2 System Architecture

Integration
In addition to the application processors manufactured by NEC, the UNIVERGE NEAX 2000 IPS
systems also deliver an open architecture for integration with other manufacturers. Interface
processors and specifications for communicating with computing devices support this integration.
These computing devices enable such service features as network and facilities management, call
center applications, call accounting, property management, voice mail, maintenance access terminals,
and management information systems for ACD reports and monitoring.

Unified Circuit Card
All the circuit cards for UNIVERGE NEAX 2000 IPS are designed in one size (PN-type), and installed in
the PIM. This maximizes the efficiency of slot utilization of the PIM.

High Density Line/Trunk Cards
Major line/trunk cards used in UNIVERGE NEAX 2000 IPS are provided with 8 circuits per card. This
allows the physical system size to be compact.

Universal Slot
One PIM provides 12 card slots for Line/Trunk (LT). Also, these card slots can be used for Application
Processor (AP) cards without complicated limitation. As this makes easy quotation and installation,
more number of AP cards can be mounted in one PIM.

DC/DC Power Supply for –48V
The PIM houses an optional DC/DC Power Supply for the cards which require –48V power such as the
CSI card used for interface of Cell Station (CS) of a wireless system. Since this power supply is
mounted in the space under the AC/DC power, additional Power Module/card slots are not required.

SN716 Attendant Console
The SN716 Attendant Console connection is available through one twisted-pair cable when using an
AC adapter. When using the PN-PW00, one twisted-pair cable is required for voice/data and one pair
is required for power from the PW00 to the Console.

Analog/Digital Telephones
The analog/digital telephone connection is available through one twisted-pair cable.

IP Telephones
The IP telephones can communicate with other IP telephones over the LAN, on a peer-to-peer
connection basis. The Main Processor (MP) provides call control with an Ethernet adapter card. Voice
packets are transmitted between IP telephones over the LAN (not through Time Division Switch). Voice
compression of G.729a (8 kbps) and G.723.1 (5.3 kbps/6.3 kbps) is available for these connections.
The IP telephones can communicate with legacy stations and trunks (including H.323 trunks) via an IPPAD that converts voice packet data to PCM signals. Call control signals are transmitted to the MP over
the LAN, while voice packets are transmitted via the IP-PAD. The number of IP-PADs depends on the
traffic volume of connections between the IP Enabled Dterm and legacy stations and trunks.

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Chapter 2 System Architecture

UNIVERGE NEAX 2000 IPS System Configuration
The UNIVERGE NEAX 2000 IPS consists of single or multiple Port Interface Modules (PIM)
depending on the system configuration, and there are two types of PIM’s; “Physical” PIM and
“Virtual” PIM. The Physical PIM is “hardware” PIM which is used to accommodate an MP,
FPs, IP PADs, legacy LT cards, AP cards, and power supply units. One Physical PIM provides
up to 64 LT ports and up to 8 Physical PIM’s can be accommodated in a Stand Alone system.
A Virtual PIM is a “software” PIM and provides up to 64 ports per PIM for use by system
programming as DtermIP telephones, Wireless PS stations or Peer to Peer (PTP) CCIS trunks.
The system consists of up to 16 PIM’s, by the combination of Physical PIM’s and Virtual PIM’s,
thus providing 1020 ports. When the use of Virtual PIM’s exceeds 8 then the number Physical
PIM’s is reduced by one for each additional Virtual PIM required.
The illustration below shows examples of 1020-port configuration by the combination of TDM
LT ports, Dterm IP telephones, Wireless PS stations and Peer to Peer (PTP) CCIS trunks.

Example 1
PIM #3 PIM #7

PIM #11 PIM #15

PIM #3 PIM #7

PIM #11 PIM #15

PIM #2 PIM #6

PIM #10 PIM #14

PIM #2 PIM #6

PIM #10 PIM #14

PIM #1 PIM #5

PIM #9 PIM #13

PIM #1 PIM #5

PIM #9 PIM #13

PIM #0 PIM #4

PIM #8 PIM #12

PIM #0 PIM #4

PIM #8 PIM #12

448 LT ports +
64 IP-PADs

Page 2- 4

Example 2

term

252 D IP + 128
Wireless + 128
PTP CCIS

192 LT ports +
64 IP-PADs

PIM

Physical PIM

Virtual PIM

term

508 D IP + 128
Wireless + 128
PTP CCIS

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
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Chapter 2 System Architecture

Module and Installation Hardware
The UNIVERGE NEAX 2000 IPS is comprised of up to 8 Port Interface Modules (PIM’s). For installing
the PIM’s in various installation methods, additional installation hardware is required.
Modules

1.) Port Interface Module (PIM)
A PIM provides 13 card slots for common control, Line/Trunk (LT), and Application Processor (AP)
cards. It also houses an AC/DC Power Supply, DC/DC Power Supply (for -48V), and batteries for
protection from short-term (about 30 min.) power interruption.
Four champ connectors for Line/Trunk (LTC 0 to 3) are located at the lower front side of the PIM.
A PIM provides a maximum of 12 card slots for Line/Trunk (LT) and Application Processor (AP) cards.
At maximum configuration, the system is comprised of 8 PIM’s.
There are two types of PIM (PIMMJ and PIMMK) depending on the system type as follows.
Type of PIM

Single MP System

PIM MJ
PIM MK

Used for PIM 0-7
Not used

PIM MJ (PIM3) PIM MJ (PIM7)
PIM MJ (PIM2) PIM MJ (PIM6)
PIM MJ (PIM1) PIM MJ (PIM5)
PIM MJ (PIM0) PIM MJ (PIM4)
(Single MP System)

Dual MP System

Used for PIM 1-7
Used for PIM 0
PIM MJ (PIM3) PIM MJ (PIM7)
PIM MJ (PIM2) PIM MJ (PIM6)
PIM MJ (PIM1) PIM MJ (PIM5)
PIM MK (PIM0) PIM MJ (PIM4)
(Dual MP System)

Unit Configuration

2.) Battery Module (BATTMH)
The BATTM is an optional module for installing optional long-term (about 3 hours) backup batteries.
The BATTM is designed to accommodate batteries covering up to a 4-PIM system (2 BATTMs support
maximum system configuration).
The BATTM is available for Floor Standing Installation or 19 inch Rack-mounting. (When the system is
Wall-mounted, the BATTM cannot be installed with the PIM.)
Modules
Abbrev

Description

PIMMJ

SN1729 PIMMJ

PIMMK

SN1730 PIMMK

BATTM

SN1731 BATTMH

Remarks

Single MP System: PIM 0 - PIM 7
Dual MP System: PIM 1 – PIM 7
Single MP System: Not used
Dual MP System: PIM 0
1 per STACK, Max.2 per system

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Chapter 2 System Architecture

Installation Hardware
Base/Top Assembly

The Base/Top Assembly includes a Base Unit and a Top Cover for the PIM. One Base/Top Assembly is
required for each PIM stack. The Base Unit also serves as the AC power distribution panel for up to a
four PIM configuration.
Hanger Assembly

The Hanger Assembly is used for Wall-mounting Installation. One set of Hanger Assembly is required
for each PIM.
19 inch Bracket

The 19-inch Bracket is a set of hardware used for 19-inch Rack-mounting Installation. The 19-INCH
RACK BRACKET (A) is installed on both sides of the PIM. One set of 19 inch Bracket (A) is required
for each PIM. The 19-INCH RACK BRACKET (B) is installed at the BASE of stack. One 19-INCH
BRACKET (B) is required for each stack.
If the system is 2 PIM or more configurations with 19-INCH BRACKET (B), one set of 19-INCH
BRACKET (A) is also required for the topmost PIM.
Optional Brackets

The Mounting Bracket is used for Floor Standing Installation. Without Mounting Bracket, 1.1G
shockproof is provided for 1 to 3-module stack and 0.5G shockproof is provided for 4 or more module
stack. To enhance the shockproof capability to 1.1G, one set of Mounting Bracket is required for each 4
or more module stack and attached to the topmost PIM.
The I/F Bracket is used for Floor Standing Installation to join the neighboring topmost PIM in 6 PIM or
more configurations. One set of I/F Bracket is required for multiple stacks.
The Base Tray Assembly is used for Floor Standing Installation for stationary equipment (UL complied).
One set of Base Tray Assembly is required for each stack.

Abbrev
Top Cover
Base/Top ASSEM
Hanger Assem
19 inch Bracket
Mounting Bracket
I/F Bracket
Base Tray

Page 2- 6

Installation Hardware
Description
Quantity
TOP COVER ASSEM
1/STACK (BASE ASSEM is local supply)
SN1545 BASERE
1/STACK
HANGER ASSEM (UL)
1/PIM (Wall-mounting Installation)
19 INCH RACK BRACKET (A)
1/PIM (19 inch Rack-mounting Installation)
19 INCH RACK BRACKET (B)
1/STACK (19 inch Rack-mounting Installation)
MOUNTING BRACKET
OPTION (1/STACK)
I/F BRACKET ASSEM
OPTION (1/SYSTEM)
BASE TRAY ASSEM
OPTION (1/STACK)

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Chapter 2 System Architecture

UNIVERGE NEAX 2000 IPS SYSTEM POWER SUPPLY
AC/DC Power Supply

The AC/DC Power Card is mounted in the left side of each PIM. The AC/DC Power card provides
power to all circuit cards, which reside in the PIM. AC power requirements are as follows:
Input Voltage: 90 to 132 Vrms or 180 to 264 Vrms (selectable by switch) 50/60 Hz
AC/DC Power Card

Name Code
PZ-PW121

Qty
1 per PIM

Remarks
AC/DC Power Supply Input:100 V/200 V
Output:+5 V,-27 V,+90V, CR

DC/DC Power Unit

The DC/DC Power Unit is mounted under the AC/DC Power Card and generates -48 V power for the
circuit cards that need such power.
DC/DC (-48V) Power Card

Name Code
PZ-PW122

Qty
1 per PIM

Remarks
DC/DC Power Supply, Input: -27 V, Output: -48 V

Battery Backup
Internal Short-term option

For customers requiring battery backup, short-term and/or long-term options are available. Two 3.4AH
batteries are required per PIM, and installed inside of each PIM. Backup time is approx. 30 minutes
when PHS (Wireless PS) is not accommodated and approx. 10 minutes when PHS (Wireless PS) is
accommodated in the system.
Internal Battery Option

Name Code
PWR CA-A
BATT CA INT

Qty
1 per (2) PIM’s
1 per PIM

Remarks
DC Power Cable (-27v input)
Internal Battery Cable

External Long-term option

Two 24AH batteries are required per each 2 PIM’s, and installed inside of Battery Module in a stack
basis. Backup time is approx. 3 hours when PHS (Wireless PS) is not accommodated and approx. 2
hours when PHS (Wireless PS) is accommodated in the system. The batteries are varied depending
on the requested backup time. The battery shall be locally provided.
Name Code

PWR CA-A
BATT CA EXT

External Battery Option
Qty

(No. of PIM) –1= up to 4PIM
(No. of PIM) –2= 5 to 8PIM
1/Stack

Remarks

DC Power Cable (-27v input)
External Battery Cable

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Chapter 2 System Architecture

Cabling
The internal and external cabling for the UNIVERGE NEAX 2000 IPS is as follows:

Internal Cabling
BUS Cable

The BUS Cable provides a connection between PIM’s for I/O Bus, PCM Bus, and Alarm Bus extension
in a multiple-PIM configuration.
Bus Cable

Name Code
48-TW-0.7 CONN CA

Qty
(No. of PIM)-1

Remarks
0.7 m (2.3ft.)

Power Control Cable

The Power Control Cable provides a connection between the MP and the AC/DC Power Supply for
controlling battery backup function and MJ/MN alarm indication.
Power Control Cable

Name Code
PWR CNT CA-D

Qty
1/PIM

PWR CNT CA-E

1/PIM

Remarks
Power control cable (for PIMI-PIM7):
BWB - PZ-PW121/122 (Included in
PIM)
Power control cable (for PIM0)
BWB - PZ-PW121/122 (Included in
Base and PIM for Dual System)

DC Power Cable

The DC Power Cable provides a connection between the AC/DC Power Supplies for extending the 27
VDC input in a multiple-PIM configuration, when battery backup option is required.
Power Cable (Internal Battery Option)

Name Code
PWR CA-A
BATT CA INT

Qty
1/2 PIM
1/PIM

Remarks
DC Power Cable (-27v input)
Internal Battery Cable

Power Cable (External Battery Option)

Name Code
PWR CA-A
BATT CA EXT

Qty

Remarks

(No. of PIM) –1= up to 4PIM
(No. of PIM) –2= 5 to 8PIM
1/Stack

DC Power Cable (-27v input)
External Battery Cable

AC Cord

The AC cord provides a connection between the AC/DC Power Card and Base Unit.
AC Cord

Name Code
AC CORD-B-U
AC CORD-D-U

Page 2- 8

Qty
1/PIM
1/BASE

Remarks
AC power cable (Included in PIM)
AC power cable for BASE (Included in BASE)

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Chapter 2 System Architecture

External Cabling
IP Connection
MP with PZ-M606-A and IP PAD

Name Code

Qty

IP TRK BUS CA

1/VCT

Remarks
IP TRK BUS Cable 0.1m (0.33ft)
(Included in SPN-16VCTA IP PAD)

IP Trunk (H.323, CCIS over IP w/IVS2)

Name Code
IP TRK BUS CA

Qty
1/VCT

Remarks
IP TRK BUS Cable 0.1m (0.33ft)
(Included in SPN-4VCTI-A, or
SPN-4VCTI-B.)

Maintenance Administration Terminal (MAT) (RS-232C)
Direct Connection
MAT Cable (Direct)

Name Code

Qty

MAT CA-T
MAT CA-P

1/MAT
1/MAT

Remarks
RS-232C cable, 2m (6.6ft.)
RS-232C cable, 4m (13.1ft.)

Remote connection

Remote connection is available via an internal modem in the MP or an external modem.
MAT Cable for external modem

Name Code

Qty

RS NORM-4S CA-A

1/MAT

Remarks
RS-232C cable, 4m (13.1ft.)

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Chapter 2 System Architecture

Built-in SMDR/MCI on MP
The MP card provides RS-232C interface ports for SMDR/MCI connection.
No. 0/1 Port: Async. : 1200/2400/4800/9600/19200 bps
Connection to SMDR is made on a port basis by a front cable.
Cable for Built-in SMDR/MCI on MP

Name Code

Qty

RS RVS-4S CA-C

1/SMDR

RS NORM-4S CA-A

1/SMDR

Remarks
RS-232C cable, 4m (13.1ft.)
MP-DTE
RS-232C cable, 4m (13.1ft.)
MP-DCE (MODEM)

SMDR/PMS/Hotel Printer/MCI (with AP00B MRC-C)
The PN-AP00-B card provides 4 RS-232C interface ports for SMDR, PMS, CIS Printer, Hotel Printer,
MCI and Centralized SMDR. Connection to SMDR, PMS or Printer is made on a port basis by a front
cable.
Cable for AP00
Name Code

Qty

RS RVS-4S CA-C

1/SMDR or
PMS
1/SMDR or
PMS
1/Printer

RS NORM-4S CAA
RS PRT-15S CA-A

Remarks
RS-232C cable, 4m (13.1ft.)
PN-AP00 - DTE
RS-232C cable, 4m (13.1ft.)
PN-AP00 - DCE (MODEM)
RS-232C cable, 15m (49.2ft.)
PN-AP00 - Printer

External Alarm Display
The UNIVERGE NEAX 2000 IPS can provide an optional external alarm display unit (ALM DSPP) to
indicate Power-ON/OFF, Major Alarm, and Minor Alarm status.
External Alarm Display Panel

Name Code
ALM DSPP

Qty
1/System

Remarks
External Alarm Display Panel

Digital Trunk Interface (DTI)
Primary Rate Interface (PRI)
CCIS Trunk Interface (CCT)
The DTI/PRI/CCT connection is available through twisted-pair cable via the MDF.
Page 2- 10

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
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Chapter 2 System Architecture

UNIVERGE NEAX IPS DM/IPS DML/IPS DMR System Configuration
IPS DM/IPS DML Modular Chassis (MC)
The UNIVERGE NEAX IPS DM consists of one to three MODULAR CHASSIS depending on the system
configuration. The UNIVERGE NEAX IPS DML consists of one or two MODULAR CHASSIS depending
on the system configuration. The MODULAR CHASSIS provides 56 LT ports in hardware slots and
provides 64 ports in software port allocation (56 LT ports and 8 virtual ports). There are 2 types of
MODULAR CHASSIS; "Physical MODULAR CHASSIS" and "Virtual MODULAR CHASSIS ". The
Physical MODULAR CHASSIS is a “hardware MODULAR CHASSIS” and is used to accommodate an
MP, IP PADs, legacy LT/AP cards, and power supply units. The Virtual MODULAR CHASSIS is a
“software MODULAR CHASSIS” used to accommodate IP stations and Peer-to-Peer CCIS Trunks by
system data programming. The port capacity of the Virtual MODULAR CHASSIS is varied depending
on the number of Physical MODULAR CHASSIS.
One MODULAR CHASSIS provides 8 card slots including one card slot for Main Processor (MP) and
other 7 slots for Line Trunk (LT)/Application Processor (AP) cards; 56 LT ports and 8 virtual LT ports;
AC, LTC, BUS cable connectors and power switch which are located at the rear side of MODULAR
CHASSIS. The following illustration shows MODULAR CHASSIS hardware configurations, software
port allocation, face layout and rear view of MODULAR CHASSIS for IPS DM/IPS DML.
IPS DM MODULAR CHASSIS Hardware Configuration

1 MODULAR CHASSIS

2 MODULAR CHASSIS

Virtual
MODULAR
CHASSIS
(952 IP Ports)

MC-0

56
LT ports

8 Virtual
LT ports

Virtual
MODULAR
CHASSIS
(888 IP Ports)

MC-1

56
LT ports

8 Virtual
LT ports

MC-0

56
LT ports

8Virtual
LT ports

256 AP Ports per System
MODULAR CHASSIS Software Port Allocation

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Chapter 2 System Architecture

IPS DMR Modular Chassis (MC)
There are 2 types of MODULAR CHASSIS; "Physical MODULAR CHASSIS" and "Virtual MODULAR
CHASSIS ". The Physical MODULAR CHASSIS is a “hardware MODULAR CHASSIS” and is used to
accommodate an MP, IP PADs, legacy LT/AP cards, and power supply units. The IPS DMR can
consist of one or two MODULAR CHASSIS depending on the system configuration. The Physical
MODULAR CHASSIS provides 56 LT ports in hardware slots and provides 64 ports in software port
allocation (56LT ports and 8 virtual ports). The Virtual MODULAR CHASSIS is a “software MODULAR
CHASSIS” with a port capacity of 64 ports. A maximum of two Virtual MODULAR CHASSIS can be
assigned per remote site for a total of 128 ports used to accommodate IP stations by system data
programming. The maximum number of Remote Sites is 30. This system allocates the maximum of 64
FP/AP cards per system, at multiple Remote Sites. The number of FP/AP cards accommodated at one
Remote Site should be a maximum of eight including the MP with built-in FP. If more than eight FP/APs
are assigned, the system does not operate normally.
The following illustration shows MODULAR CHASSIS hardware configurations, software port allocation,
face layout and rear view of MODULAR CHASSIS for IPS DMR.

IPS DMR MODULAR CHASSIS Hardware Configuration

1 MODULAR CHASSIS

2 MODULAR CHASSIS

Virtual
MODULAR
CHASSIS
(128 IP Ports)

Virtual
MODULAR
CHASSIS
(128 IP Ports)
MC-1
MC-0

56
LT ports

8 Virtual
LT ports

MC-0

56
LT ports
56
LT ports

8 Virtual
LT ports
8 Virtual
LT ports

256 AP Ports per System
MODULAR CHASSIS Software Port Allocation

Page 2- 12

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Chapter 2 System Architecture

Modular Chassis (MC) Installation
The MODULAR CHASSIS can be installed on the desktop or into the 19-inch rack only.

Modular Chassis (MC) and Bracket
The following tables show the name and specification number of MODULAR CHASSIS and bracket.
The bracket is selected depending on the installation method (Desktop or 19" Rack).
Modular Chassis
Qty

Description
SN-1737 IPSMJ

Modular Chassis With Out RGU(Molded Cover)

Bracket
Qty

Description
RACK MOUNT KIT(U)

JOINT BRACKET KIT(U)

Remarks

1/Modular Chassis for 19” rack-mounting installation
1/Expansion Modular Chassis for desktop-setting
installation

Cabling
Description
BUS-0.4 CA-X

BUS Cable / AC Cord
Qty
1 Expansion MODULAR CHASSIS

Remarks
BUS cable for 8 Slot Chassis
BUS cable for connection between
BUS-0.4 CA-CHG
1 Expansion MODULAR CHASSIS
6 Slot and 8 Slot Chassis
AC CORD-E-U
1 per MODULAR CHASSIS
AC Cord
Note: When expanding an existing 6 slot Chassis only a two chassis configuration is supported
and the 6 slot chassis must be PIM 0.

Description
BATT CA-P5

PFT and Battery cable
Qty
1 per MODULAR CHASSIS

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Remarks
PFT and Battery connection
cable 5m (15ft). Cable for
connecting Battery to 4PFT of
AC/DC Power

Page 2-13

Chapter 2 System Architecture

System Power Supply and Cooling Fan
AC/DC Power Supply

The AC/DC Power Card is mounted in the rear side of each MODULAR CHASSIS. The AC/DC Power
card provides power to all circuit cards, which are resided in the MODULAR CHASSIS.
AC power requirements are as follows:
Input Voltage: 85 to 264VAC, 50/60 Hz (Auto sensing 100V/200V selection)
AC/DC Power Card
Description
PZ-PW131

Qty
1 per MODULAR CHASSIS

Remarks
AC/DC Power Supply
Output:+5 V, 5.5A, -27 V, 3A

Back-up Power

UPS or battery backup (with external battery) is available for backup power for AC power failure. In
case of battery backup, sealed type battery should be used. The open type battery is not available
because of AC/DC power specifications.
Battery Backup Time
10 minutes (Built-in battery)
30 minutes (External battery)
3 hours (External battery)

Required Battery Capacity
Required Battery Capacity
12VDC, 0.8AH x 2
12VDC, 2.2AH x 2
12VDC, 12AH x 2

Ring Generator Unit (RGU)

The RGU provides ringing supply to analog SLT (PN-4LCAA is no longer required); however, it cannot
ring more than 6 analog SLT’s per MODULAR CHASSIS simultaneously. Therefore, the simultaneous
ringing has to be limited by office data setting.
Description
PZ-PW139

Ring Generator Unit (RGU) Card
Remarks
Ring Generator Unit
MODULAR CHASSIS
Ring Voltage: 85/75Vac
Ring Current: 0.05A
Qty

Cooling FAN

The cooling FAN is mounted at the right side of each MODULAR CHASSIS. The FAN should be
replaced every 4 years in order to ensure the cooling system.
Cooling FAN
Description
109P0624H
7D09 FAN

Page 2- 14

Qty

Remarks

1 per MODULAR CHASSIS

Each Modular Chassis is equipped with a
Cooling Fan this part is for replacement only

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 2 System Architecture

Software Architecture
The UNIVERGE NEAX 2000 IPS systems offer very attractive system starting points. Listed
below are the System/Expansion Packages available with the UNIVERGE NEAX 2000 IPS
family.
System Packages
UNIVERGE NEAX 2000 IPS System Packages
Description
UNIV IPS IP 8 Seat System Package
IPS UNIV PIMMJ (UA)
ICS VS BASE-C (UA)
SPN-CP24-C (CPU)
PZ-M606-A
SPN-8IPLA IP PAD-C
64 Port SYS SOFTWARE
KEY KEEPER (FD)
8 SEAT LICENSE
MATWorX IPS

UNIV IPS 64 Port Basic System Package
SPN-CP24-C (CPU)
IPS UNIV PIMMJ (UA)
ICS VS BASE-C (UA)
64 Port SYS SOFTWARE
MATWorX IPS

UNIV IPS Back-Up CPU 8 IP Seat SYS PKG
IPS UNIV PIMMK (UA)
ICS VS BASE-C (UA)
SPN-CP27A (CPU)
SPN-CP27A (CPU)
PZ-M606-A
PZ-M606-A
SPN-8IPLA IP PAD-C
64 Port SYS SOFTWARE
KEY KEEPER (FD)
8 SEAT LICENSE KEY
MATWorX IPS

UNIV IPS Back-Up CPU 64 Port (TDM) SYS
PKG (IPS)
IPS UNIV PIMMK (UA)
ICS VS BASE-C (UA)
SPN-CP27A (CPU)
SPN-CP27A (CPU)
64 Port SYS SOFTWARE
MATWorX IPS

IPS/IVS2 UNIV Upgrade Kit
SPN-CP24-C (CPU)
IPS DM UNIV Front Cover
64 Port SYS SOFTWARE

Remarks

IP Starter Kit Package.
Supports up to 8 IP Dterm’s, 64 LT Ports,
5 T1’s /E1’s, 5 ISDN-PRI DCH’s, 48 ISDN-BRI
Trunks.
NEC Customer Software License Agreement Required.

Supports up to 64 LT Ports & 1 T1.
NEC Customer Software License Agreement Required

Package is equipped with 64 TDM ports and 8
IP. Provides matching CPU cards with the
same CPU number. With matching CPU’s,
only one set of software is needed.
NEC Customer Software License Agreement Required.

Package is equipped with 64 TDM ports
License for each CPU. Provides matching
CPU cards with the same CPU number. With
matching CPU’s, only one set of software is
needed.
NEC Customer Software License Agreement Required.

Upgrades CPU to CP24 C and Adds
UNIVERGE Front Cover
NEC Customer Software License Agreement Required

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 2-15

Chapter 2 System Architecture

UNIVERGE NEAX IPS DM/IPS DML System Packages
Description
UNIV IPS DML IP SYS PKG

Remarks
Provides for up to 48 Legacy ports and 128 IP
stations per system. Comes with 8 IP Seat
Licenses. Additional IP Seats Licenses are
required beyond 8.

IPS DM(E) UNIV PIMMJ
AC CORD-E-U
Rack Mount Kit
SPN-CP31-C (CPU)
PZ-M606-A
SPN-8IPLA IP-PAD-C
64 Port SYS SOFTWARE
Key Keeper (FD)
8 Seat Licenses
24 Port Patch Panel
MATWorX IPS

NEC Customer Software License Agreement Required

Provides for up to 48 Legacy ports and 952 IP
stations per system. Comes with 8 IP Seat
Licenses. Additional IP Seat Licenses are required
beyond 8.

UNIV IPS DM IP SYS PKG-E
IPS DM(E) UNIV PIMMJ
AC CORD-E-U
Rack Mount Kit
SPN-CP24-C (CPU)
PZ-M606-A
SPN-8IPLA IP-PAD-C
64 Port SYS SOFTWARE
Key Keeper (FD)
8 Seat Licenses
24 Port Patch Panel
MATWorX IPS

NEC Customer Software License Agreement Required

UNIV IPS DM/DML Expansion Module 1(E)
IPS DM(E) UNIV PIMMJ
AC CORD-E-U
Rack Mount Kit
Joint Bracket Kit
BUS-0.4 CA-PA
Key Keeper (FD)
LT Port 64
24 Port Patch Panel

DM to UNIV DM Upgrade Kit
SPN-CP24-C (CPU)
IPS DM UNIV Front Mask
64 Port SYS SOFTWARE
MATWorX IPS

Page 2- 16

Adds 56 additional Legacy ports for a total of 112
(DM/DML) and supports up to 888 IP station (DM
Only) per system. DML supports 128 IP stations
max.

Upgrades to CP24C CPU and Adds UNIVERGE
Front cover.

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 2 System Architecture

IP REMOTE PIM PACKAGES
Description
UNIV IPS DMR SYS PKG-E
IPS DM(E) UNIV PIMMJ
SPN-CP31-C (CPU)
PZ-M606-A
SPN-8IPLA IP PAD-C
AC CORD-E-U
Rack Mount Kit
24 Port Patch Panel

UNIV IP REMOTE PIM-B (IPS)
IPS UNIV PIMMJ (UA)
ICS VS BASE-C (UA)
PN-CP24-C
PZ-M606-A
SPN-8IPLA IP PAD-C

UNIV IPS DMR Expansion Module
IPS DM(E) UNIV PIMMJ
AC CORD-E-U
Rack Mount Kit
Joint Bracket Kit
BUS-0.4 CA-PA
24 Port Patch Panel

Remarks
DMR Package for Remote PIM.
Provides DM Chassis for Remote PIM over IP.
The 8IPLA PAD card is provided.
Remote PIM Site License required.

IPS Package for Remote PIM.
Provides IPS PIM that can be used as Remote
PIM over IP.
Remote PIM Site License required.

Adds another DM PIMMF (Modular Chassis) to
provide five card slots for 56 additional TDM
ports.

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 2-17

Chapter 2 System Architecture

Software Keys/Licenses
The UNIVERGE NEAX 2000 IPS systems offer both TDM Software Keys and Peer-to-Peer IP Seat
Licenses, which are designed with modularity in mind. This approach allows customers a greater
degree of cost-control for new installations and for upgrades to features, capacities and software series.
Description
Generic Program
64 Port Sys Software

Remarks
Basic Business/Hotel/Motel Features for:
64 LT Ports, 5 T1’s /E1’s, 5 ISDN-PRI DCH’s, 48 ISDN-BRI
Trunks.
NEC Customer Software License Agreement Required

Key Keeper (Parent Capacity Option)
Key Keeper (FD)

Floppy Disk that holds selected Key files from below

Capacity Option (used w/Key Keeper)
LT 64 Port
Expands LT Ports from 64 to 1020 Ports in increments of 64.
CCIS Link (1)
CCIS Link (4)
CCIS Link (8)

Adds support for one CCIS Link
Adds support for four CCIS Link
Adds support for eight CCIS Link

IPT Card (1)
IPT Card (4)
IPT Card (8)

Adds support for one IP trunk card
Adds support for four IP trunk card
Adds support for eight IP trunk card

Event Based CCIS (ECCIS) Key

Adds Event Based CCIS capability

Wireless Software Key

Adds Wireless with support of 128 ZT’s and 256 PS’s

Wireless Software Key 8 PS

Adds 8 Wireless ports each key above the initial 256

T1/E1 6 to 10

Expands T1/E1 capacity between 144 to 240 channels

ISDN DCH 5 to 8

Expands capacity between 5 DCH Cards and 8 DCH cards

IP Remote PIM 1 Site License

Adds IP Remote Capability and is required for each Remote Site

Dterm IP Software Seat Licenses
DtermIP terminals are controlled by the CPU and do not use
digital line cards, Instead they require DtermIP seat licenses.
The licenses are available in 8 seat increments and are
8 IP Seat License Key
cumulative. For example, if you have 8 existing DtermIP Seats
and need a total of 16, add another 8-seat license for a total of 16
seats.
Each SP-30 4 Seat License can support up to four simultaneous
sessions via the SP-30 Soft-Phone. The SP30 Soft-Phone also
SP-30-4 Seat License
requires IP 8 Seat License. To support eight simultaneous softphone sessions would require eight Soft-Phone Licenses and 8
IP Seat Licenses.

SP-30-4 ACD Seat License

Page 2- 18

Each SP-30 4 ACD Seat License can support up to four
simultaneous sessions via the SP-30 ACD Soft-Phone and can
support either Business or ACD mode. The SP30/SP30 ACD
Soft-Phone also requires IP Seat License. To support eight
simultaneous soft-phone sessions would require eight SoftPhone Licenses plus 8 IP Seat Licenses.

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights
Processors
The UNIVERGE NEAX 2000 IPS, IPS DM, IPS DML and IPS DMR are distributed
multiprocessor systems. Their control system consists of a Main Processor (MP), Firmware
Processors (FP), and Application Processors (AP). Both the FP and APs execute their
predetermined functions under the control of the MP.
Main Processor (MP)
Name Code

Remarks

PN-CP24-C

Main Processor Card for UNIVERGE NEAX 2000 IPS and IPS DM.
One card is required per system.

PN-CP27-A

Main Processor Card for UNIVERGE NEAX 2000 IPS Dual MP System.
One card is required per system.
Main Processor Card for UNIVERGE NEAX IPS DML
One card is required per system.(Stand Alone system or CCIS end point)

PN-CP31-C
Main Processor Card for UNIVERGE NEAX IPS DMR.
One card is required for each Remote Site.

PZ-M606

Ethernet Control Card:
• Mounted on MP card to accommodate the Ethernet and transmit/receive a signal
of TCP/IP protocol.
• 10 BASE-T/100 BASE-TX twisted pair cable is connected directly to this card.

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-1

Chapter 3 System Highlights

Major specifications and functionality of the UNIVERGE NEAX 2000 IPS MPs are shown below:
PN-CP24-C
PN-CP27-A

Item
Central Processing
System Memory
Network Switching
3-Way Conference
DTMF Signal Sender
Music-on-Hold
Mini Jack
Audible Tone Generator (DTG)
Phase Lock Oscillator (PLO)
Built-in SMDR
Built-in MCI
Built-in FP0
BS00 Function
DTMF Receiver
AP01 Function
Built-in DRS
MAT Interface
Direct Connection
Remote Connection
w/Built-in MODEM
External Alarm Indication
DAT
DK00

1 port
1 port

PN-CP31-C

ElanSC520
Flash ROM (16MB), SDRAM (64MB) Note 1
1,024 · 1,024 Time Division Switch
16 sets of 3-way conference circuitry
32 circuits (digit 0 to 9, *, and # are generated)
10 types are available Note 2
1 for External Music Source for Music on Hold Note 2
Available
2 ports (Source/Receiver)
Available
Available
Available
Available
4 circuits
Available
Available
----1 port
1 port Note 3

MJ and MN
MJ only
2 circuits (120 seconds per circuit) Not Available
2 circuits (relay drive x1,
Not Available
external key scan x1)
Application Key Program
In EPROM
In Flash ROM
MP Program Download
Upgrade Program via FTP Server Note 1
Note 1: CP-27-A memory is 8MB Flash and 32MB SDRAM and MP Program Download is not supported.
Note 2: In case of Dterm IP, the synthesized melody is provided by the IP adapter on the phone
Note 3: Modem on the CP31 is for Stand Alone mode IPS DML only

Firmware Processor (FP)
Firmware Processors (FP) are required when more than two PIM’s are used. The FP provides
supervision and status analysis of line/trunk ports, which reside in the PIM. The FP provides the bus
interface for I/O Bus, PCM Bus, and Alarm Bus in a multiple-PIM configuration. The major
specifications of the FP are shown below:
Central Processor Unit: 16-bit (25 MHz)
Memory: Program Area (384 kb), Work Area (384 kb)
BS01 Function
Item

Remarks

PN-CP15

Firmware Processor Card for use with the UNIVERGE NEAX 2000 IPS.

Page 3- 2

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

Application Processor (AP)
Application Processors communicate directly to the MP, bypassing the FP. Features such as SMDR,
Property Management System (PMS), Digital Trunks (T1), CCIS, etc. are Application Processors.
System Capacity for Application Processor Card:
Maximum of 24 cards per system
Maximum of 256 ports per system
There are several different AP cards. The table below provides a list of APs and their functions.
Application Processor (AP) Cards
AP CARD
SPN-24DTAC (AP)
SPN-30DTC (AP)
SPN-24PRTA (AP)
SPN-24PRTA QSIG(AP)
SPN-24CCTA (AP)
SPN-2BRTC (AP)
SPN-4BRTA (AP)
SPN-SC00 CCH (AP)
SPN-SC01 DCH (AP)
SPN-SC03 8ICH (AP)
SPN-SC03 8CSH (AP)
SPN-SC01 QSIG (AP)
SPN-AP00B MRC-C (AP)
SPN-AP00B MRC-F (AP)
SPN-4RSTBA (AP)
SPN-4RSTBA-911 (AP)
SPN-4RSTC (AP)
SPN-IPTB-A (AP)
SPN-IPTB-B (AP)
SPN-CFTC (AP)
SPN-AP00B DBM (AP)
SPN-SC01 DCH-Q (AP)

Function
24 Channel Digital Trunk Interface with built-in CSU
30 Channel Digital Trunk Interface with A/U Law conversion
24 Channel PRI Trunk with built-in D-Channel Handler (DCH)
24 Channel PRI Trunk with built-in QSIG D-Channel Handler (DCH)
24 Channel CCIS Trunk with built-in Common Channel Handler (CCH)
2 circuit BRI (Trunk)
4 circuit BRI (Trunk)
Common Channel Handler for CCIS Signaling
D-Channel Handler for ISDN Primary Rate Access
D-Channel Handler for BRI
CS (ZT) Handler (CSH)
Q-SIG Protocol Handler
SMDR, PMS, H/M Printer & MCI Port
DND Group, PMS Executed History Print out, SMDR (2400 Format) & MCI
4 circuit MF Receiver for T1 trunks with ANI
4 circuit MF Receiver for enhanced E911
4 circuit ID Number Receiver for Analog Caller ID (Class SM)
IP Trunk for VoIP (P-MP)
IP Trunk for VoIP (H.323)
32-Party Conference
Data Base Memory, Wireless Roaming
Q931a Protocol Handler for Wireless Roaming

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-3

Chapter 3 System Highlights

Reliability and Availability
Mean Time Between Failures (MTBF)
The MTBF represents the minimum time frame in which a failure of any common equipment --such as
processors, memory, switching, and etc. - may cause a partial system outage. The following table
shows the MTBF in hours and in years for each card and terminal.

Reliability and Availability Chart
Reliability
Description

Availability

Fit

MTBF
years

MTBF
Minute

MTTR
Minute

ICS VS PIMMJ (BWB)

1868

61.1

32136156

99.999813%

SPN-CP24A MP

11849

9.63

5064995

99.999112%

SPN-CP24B MP

10798

10.6

5575176

99.999193%

SPN-CP31A MP

10700

10.7

5627772

99.999200%

PN-CP27A MP

10798

10.6

5575176

0.3

99.999995%

SPN-CP15 FP

3060

19460520

99.999974%

PZ-M606-A

1269.2

89.94

47304842

99.999979%

PZ-PW121

13850

8.2

4312872

99.999884%

PZ-PW122

5780

19.8

10414008

99.999904%

PZ-PW131

13350

8.6

4523256

99.999668%

PN-8LCAA

1934

31031640

99.999984%

PN-8COTS

3312

34.5

19092348

99.999974%

PN-4DATC

5490

20.79

10934708

99.999954%

PZ-8PFTB

1439

79.3

41708628

99.999988%

PN-CFTB

941

121

63641160

99.999992%

PN-CFTC (AP)

6000

19.8

10414008

99.999952%

PN-IPTB-B (AP)

5757

19.83

10429787

99.999952%

1435.3

79.54

41834858

99.999988%

4467

25.6

13464576

99.999963%

SPN-4VCTI-B
PN-4LCAA

Page 3- 4

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

Reliability and Availability Chart (Cont.)

Reliability
Description

Availability
Fit

MTBF
Years

MTBF
Minute

MTTR
Minute

PN-2ILCA

5849

19.5

10256220

99.999951%

PN-8DLCP

1848

61.77

32488549

99.999985%

PN-2DLCN

856

133.36

70142026

99.999993%

PN-M10

1750

65.23

34308371

99.999985%

PN-DK00

1274

89.6

47126016

99.999989%

PN-4LLCB

1792

63.7

33503652

99.999985%

PN-AP00B MRC-C (AP)

5196

11571120

99.999957%

PN-SC03-B 8ICH(AP)

6816

16.75

8809830

99.999943%

PN-16VCTAA IP PAD-A/B

2674

42.69

22453232

99.999978%

PN-20DTB

2965

38.5

20249460

99.999975%

SPN-32IPLAA IP PAD-C/D

11112

10.3

5417388

99.999908%

SPN-SC00 CCH-D(AP)

8816

12.95

6811182

99.999927%

PN-30DTCC-A(AP)

3504

32.58

17135777

99.999971%

SPN-24PRTA (AP)

4625

24.7

12991212

99.999962%

PN-24CCTA

4625

24.7

12991212

99.999962%

PN-8RSTG

133

858.31

4.51E+08

99.999999%

6566.4

17.39

9146444

99.999945%

4082

14726880

99.999966%

7790.4

14.3

7521228

99.999934%

PZ-M537

346

329.9

17409276

99.999943%

PZ-M542

1464

41024880

99.999976%

PZ-4PFTA

751

152

79945920

99.999987%

13680

8.3

4365468

99.999313%

PN-4BRTA-F(AP)
SPN-24PRTA-QSIG (AP)
SN716 DESK CON-B

IPS DM PIMMF

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-5

Chapter 3 System Highlights

Reliability and Availability Chart (Cont.)
Reliability
Description

MTBF
years

MTBF
Minute

MTTR
Minute

Availability

Fit
DTR-1-1 TEL

1359

44180640

99.999989%

DTR-1HM-1 TEL

2718

22090320

99.999977%

DTR-2DT-1 TEL

9927

11.5

6048540

99.999917%

DTR-8-1 TEL

8154

7363440

99.999932%

DTR-8D-1 TEL

10378

5785560

99.999914%

DTR-16D-1 TEL

10976

10.4

5469984

99.999909%

DTR-32D-1 TEL

12614

9.05

4759938

99.999895%

ITR-8D-2 TEL

14269

4207680

99.999881%

ITR-16D-2 TEL

14269

4207680

99.999881%

DCR-60-1P Console

9593

11.9

6258924

99.999920%

AD(A)-R Unit

2857

39.95

21012102

99.999952%

AP(R)-R Unit

6668

17.12

9004435

99.999889%

AP(A)-R Unit

6406

17.82

9372607

99.999893%

CT(A)-RP Unit

8154

7363440

99.999864%

IP-R Unit

12144

9.4

4944024

99.999798%

DCU-60-1

9593

11.9

6258924

99.999920%

DTP-8-1 TEL

8272

13.8

7258248

99.999931%

DTP-8D-1 TEL

10570

10.8

5680368

99.999912%

DTP-16D-1 TEL

11083

10.3

5417388

99.999908%

DTP-32D-1 TEL

12826

8.9

4681044

99.999893%

DTP-1-2

3110

36.7

19302732

99.999974%

DTP-1HM-2

4494

25.4

13359384

99.999963%

Page 3- 6

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

Mean Time To Repair (MTTR)
The Mean Time To Repair has been studied with the following conditions:
1. A mid to high-level technical person will attend to the site for the detection of the fault and
replacement of the faulty circuit board.
2. All critical circuit boards for replacement are stocked at each Fault Dispatch Center (FDC).
3. All non-critical circuit boards for FDCs are stocked at NEC. The calculation result of the MTTR is
1.0 hour.

Reliability Calculations
The following information is to show you how to calculate the reliability for the system you are
configuring. Remember that these calculations are used to determine a partial system outage.
Reliability Calculation

1FIT
109
MTBF=
=
Total FITS Total FITS

1Defect
1FIT=
1,000,000,000Hours•Parts
System Calculation

To calculate the MTBF of a single system, obtain the sum of the FITs from each component.
For example: Assume you have a complete system that consists of two components:
Item A FIT=31,700, Item B FIT=80,000
Total Fit = 31,700 + 80,000 = 111,700
109
MTBF=

= 8952.55 Hours
111,700
8,952.22 Hours
= 1.02 Years

8,760Hours / Year

Availability Calculations
Availability=

MTBF
MTBF+MTTR

=.9XXXXX

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-7

Chapter 3 System Highlights

System Traffic
In regards to the traffic capacity of electronic PABX, two kinds of descriptions are usually
employed. One is Busy Hour Call Attempts (BHCA), which relates to the ability of the Central
Processor Unit (CPU). Another is the traffic handling capacity per line that relates to the
handling capacity of each station. This technical information herein offers the traffic data for
both cases.
Traffic Load
Traffic load is expressed as the quantity of traffic during the busy hour, the busiest one-hour period of
the day for traffic. Traffic can be measured in minutes, hours, hundreds of call seconds (CCS), or
Erlangs.
CCS– one CCS is equal to 100 seconds of telephone time, or 1/36 Erlang.
Erlang– one Erlang is equal to one hour of telephone time or 36 CCS.
CCS units are convenient when traffic is measured in small increments. For telephone traffic, a typical
business generates about 6 CCS per telephone during the busy hour (1/6 Erlang).

Busy Hour
The hour during the day when traffic is highest is called the busy hour. Since this is the period when
users are most likely to encounter blockage, the traffic load during the busy hour is the load you need to
plan for. Traffic tables show the amount of traffic that can be handled during the busy hour. Within the
busy hour, peaks will occur when blockage will be higher than the design objectives. It's not
economical, however, to design a network to handle absolute peaks. Most networks are designed to
cause some traffic to be blocked or overflow to other services.
One way to establish the load during the busy hour is to schedule traffic studies for a week when you
know the load is high. Or, you can use tables developed by traffic engineers to estimate the busy hour
load.

BHCA (Busy Hour Call Attempts)
BHCA of the UNIVERGE NEAX 2000 IPS is shown below.
Number of PIM’s
Business or
Hotel/Motel
without ACD/OAI

Max. 2500 BHCA
(No FP)

Max. 5000 BHCA
(One FP)

Max. 7500 BHCA
(Two FPs)

Max. 8000 BHCA
(Three FPs)

Business or
Hotel/Motel with
ACD/OAI

Max. 2000 BHCA
(No FP)

Max. 4000 BHCA
(One FP)

Max. 5000 BHCA
(Two FPs)

Max. 5500 BHCA
(Three FPs)

Page 3- 8

1PIM

2PIM

3PIM

4PIM

5PIM

6PIM

7PIM

8PIM

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

Traffic Capacity In Erlang (For TDM Circuits)
Calculation Procedure

TDSW
LC
Internal Traffic
(I-erl)
LC
STA
LC
COT

External Traffic
(E-erl)

C.O.
Total system traffic capacity (T-erl) is obtainable by the following formula.
T (erl) = I (erl) + E (erl)
In the UNIVERGE NEAX 2000 IPS, external traffic capacity is obtained from the Central Office Trunk
(COT) quantity by employing the “Erlang B Table”. There is no limitation for the internal traffic capacity.
The time slot is assigned for individual station and the connection between stations can be made
without limitation through TDSW (Time Division Switch). Consequently, when describing the traffic
capacity in Erlang, it is meaningless unless specifying the ratio of the internal and external traffic. From
the above idea, total system traffic capacity (T-erl) is actually obtained by the following formula:
T (erl) = Internal Traffic (I-erl) + External Traffic (E-erl)
= E (erl) ‚ (Ratio of external traffic)
When assuming the external and internal traffic is even.
T (erl) = E (erl) + 0.5
For example:
System configuration:
• 384 Lines
• 64 C. O. Trunks
• External traffic: Internal traffic: = 7:4
In this system, external traffic capacity is obtained from “Erlang B” table. That is, 64 C. O. Trunks at
grade of service of 0.01 can carry 50.6 erl. Total traffic capacity of this system is obtained from above
50.6 erl and ratio of external traffic as shown below:
T (erl) = E (erl) ÷

(Ratio of external traffic)

= 50.6

7__
7÷3

= 72.3 erl
Traffic capacity per station line
= 72.3erl
÷ 384 lines
= 0.19 erl/L

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-9

Chapter 3 System Highlights

Traffic Capacity per Line (For TDM Circuits)
Traffic capacity per line (erl/L) can be calculated in accordance with the calculation procedure in
Section 3.4. The traffic capacity per line at various line/truck configurations as the calculation result is
shown in the following table. Though the following table does not cover all UNIVERGE NEAX 2000 IPS
line/trunk configurations, it covers the typical TDM line/trunk configurations likely to be required in the
actual system. Grade of service is considered as 0.01. Use the Erlang B Table to find the required
number of DTMF register ports for a specific UNIVERGE NEAX 2000 IPS.
Unit = erl per line

Number of Trunks
Use Erlang B table to determine the number of trunks at a particular grade of service after using the
following formulas:
Incoming: incoming ccs traffic
----------------------- = Erlang B
36 ccs
Outgoing: outgoing ccs traffic
----------------------- = Erlang B
36 ccs
Note: Traffic loads are measured in hundred call seconds, or CCS. Since there are 3,600 seconds in an hour, a
line that's tied up for one hour measures 36 CCS traffic.

Page 3- 10

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

Erlang B Table
Traffic
Distribution
Line
16L

32L

48L

96L

144L

192L

240L

288L

336L

384L

Trunk
4 COT
8 COT
12 COT
4 COT
8 COT
12 COT
16 COT
4 COT
8 COT
12 COT
16 COT
12 COT
16 COT
20 COT
24 COT
12 COT
16 COT
20 COT
24 COT
20 COT
24 COT
28 COT
32 COT
20 COT
24 COT
28 COT
32 COT
20 COT
24 COT
28 COT
32 COT
28 COT
32 COT
36 COT
40 COT
44 COT
28 COT
32 COT
40 COT
44 COT
48 COT
64 COT

Internal
Call: 0.3
External
Call: 0.7
0.078
0.279
0.525
0.039
0.140
0.262
0.396
0.026
0.093
0.175
0.264
0.087
0.132
0.179
0.228
0.058
0.088
0.119
0.152
0.090
0.144
0.139
0.164
0.072
0.091
0.111
0.131
0.060
0.076
0.093
0.109
0.079
0.094
0.108
0.123
0.138
0.069
0.082
0.108
0.121
0.134
0.188

:0.4

:0.5

:0.6

:0.7

:0.6

:0.5

:0.4

:0.3

0.091
0.326
0.612
0.045
0.163
0.306
0.462
0.030
0.109
0.204
0.308
0.102
0.154
0.209
0.266
0.068
0.103
0.139
0.177
0.104
0.133
0.162
0.191
0.084
0.106
0.129
0.153
0.070
0.089
0.108
0.128
0.092
0.109
0.127
0.144
0.161
0.081
0.096
0.126
0.141
0.157
0.220

0.109
0.391
0.735
0.054
0.196
0.367
0.555
0.036
0.130
0.245
0.370
0.122
0.185
0.251
0.319
0.082
0.123
0.167
0.212
0.125
0.159
0.194
0.230
0.100
0.127
0.155
0.184
0.084
0.106
0.129
0.153
0.111
0.131
0.152
0.173
0.194
0.097
0.115
0.151
0.169
0.188
0.264

0.136
0.489
0.918
0.068
0.244
0.459
0.693
0.045
0.163
0.306
0.462
0.153
0.231
0.313
0.398
0.102
0.154
0.209
0.266
0.157
0.199
0.243
0.287
0.125
0.159
0.194
0.230
0.104
0.133
0.162
0.191
0.139
0.164
0.190
0.216
0.242
0.121
0.144
0.189
0.212
0.235
0.329

0.181
0.652
1.224
0.091
0.326
0.612
0.925
0.060
0.217
0.406
0.613
0.204
0.308
0.418
0.531
0.136
0.205
0.278
0.354
0.209
0.266
0.324
0.383
0.167
0.212
0.259
0.306
0.139
0.177
0.216
0.255
0.185
0.219
0.253
0.288
0.323
0.162
0.191
0.252
0.282
0.313
0.439

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-11

Chapter 3 System Highlights

PEG Count
The PEG counter is used for maintenance purposes to verify various data in the system. PEG counters
can be set by system programming to start and stop by month, day, hour and minute. A system reset
will clear all PEG count data. The following data can be measured by PEG count.
Trunk PEG Count

Number of outgoing trunk seizure by Trunk Route
Number of tandem connections established
Number of times a busy station was encountered
Number of all types of calls to Attendant Console
Number of connections giving Dial Tone
Number of station-to-station connections established
Number of failures caused by all senders being busy
Number of failures caused by all registers being busy
Number of failures caused by all ringing trunks being busy
Number of failures caused by all IP-PAD channels being
Number of forced release of communication between station and Trunk/Tandem connection
Number of call forwarding caused by the calling number is not informed from network
Number of rejection of the incoming call the calling is not informed from network
Number of incoming call seizure-Trunk Route
Number of times all trunks found to be busy trunk route
Number of incoming calls terminated to busy tone-Trunk Route
Number of unanswered incoming calls-Trunk Route
Number of register connection on trunk call-Trunk Route
Number of conference calls (Three/Four way Calling)
Number of failures cased by all conference trunks (For three way Calling) being busy
Number of transferred incoming calls to Attendant Console or predetermined station, by Call
Forwarding-Don’t Answer
UCD PEG Count

Number of answered calls on UCD station
Number of incoming calls to UCD Group
Number of call waiting calls for predetermined time in queuing mode on UCD Group
Number of abandoned calls to UCD Group
Number of incoming calls to all busy of UCD Group
Number of incoming calls to UCD Group that were answered
Number of times queuing
Wireless PEG Count

Number of Call Origination
Number of Call Termination
Number of Location Registration
Number of Handovers
Number of Out of Cell
IP Network PEG Count

Number of times that traffic exceeded the limit bandwidth
Number of times that traffic exceeded the warning bandwidth
Maximum bandwidth used
Bandwidth used now
Page 3- 12

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

Number of Voice Channels per IP Trunk
The maximum voice channels per IP Trunk card depends on the payload size as follows (payload size
can be assigned in system programming):

CCIS (Point-to-Multipoint)
Payload Size

G.729a

G.711

G.723.1

20 ms

8 Channel

------

30 ms

16 Channel

40 ms

16 Channel

------

Payload Size

G.729a

G.711

G.723.1

20 ms

6 Channel

5 Channel

---

30 ms

8 Channel

7 Channel

8 Channel

40 ms

12 Channel

10 Channel

---

VoIP (H.323)

Payload size for Virtual IPT
Payload Size

10 ms.
20 ms.
30 ms.
40 ms.

Max. Voice Channels Per IPT

G.729a
4
8
16
16

G.711
4
8
16
16

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

G.723.1
–
–
16
–

Page 3-13

Chapter 3 System Highlights

IP PAD Calculation
IP PAD Calculation
Number of extensions
Legacy: Number of PAD Ports
0%

Number of IP PAD Cards

Legacy: Number of PAD Ports
10%

Number of IP PAD Cards

Legacy: Number of PAD Ports
20%

Number of IP PAD Cards

Legacy: Number of PAD Ports
30%

Number of IP PAD Cards

Legacy: Number of PAD Ports
40%

Number of IP PAD Cards

Legacy: Number of PAD Ports
50%

Number of IP PAD Cards

Legacy: Number of PAD Ports
60%

Number of IP PAD Cards

Legacy: Number of PAD Ports
75%

Number of IP PAD Cards

Legacy: Number of PAD Ports
80%

Number of IP PAD Cards

Legacy: Number of PAD Ports
88%

Number of IP PAD Cards

Legacy: Number of PAD Ports
95%

Number of IP PAD Cards

Legacy: Number of PAD Ports
99.0%

Page 3- 14

Number of IP PAD Cards

100

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

IP PAD Calculation
Legacy:
0%
Legacy:
10%
Legacy:
20%
Legacy:
30%
Legacy:
40%
Legacy:
50%
Legacy:
60%
Legacy:
75%
Legacy:
80%
Legacy:
88%
Legacy:
95%
Legacy:
99.0%

Number of extensions

120

200

208

264

300

320

376

432

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

Number of PAD Ports

Number of IP PAD Cards

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-15

Chapter 3 System Highlights

System Specifications
AC Power Requirements
Description
AC Input Voltage
AC Input Current

Specifications
90 to 132Vac or 180 to 264Vac; 47 to 64Hz
3.5A(at 100V), 2.0A(at 200V)

AC Power Consumption / Thermal Output (Maximum)
Description
1-PIM
2-PIM
3-PIM
4-PIM
5-PIM
6-PIM
7-PIM
8-PIM

AC Power Consumption (KVA)
100V
200V
0.35
0.40
0.70
0.80
1.05
1.20
1.40
1.60
1.75
2.00
2.10
2.40
2.45
2.80
2.80
3.20

Thermal Output (BTU)
100V
200V
1,195
1,365
2,389
2,730
3,584
4,096
4,778
5,461
5,973
6,826
7,167
8,191
8,362
9,556
9,556
10,922

Battery Requirements
Description
Max. Battery Capacity
DC Input Voltage for Battery
Built-in Battery Requirements
Physical Size of Built-in Battery
(one 12V battery)

Specifications
260AH per 4 PIM (65AH (12V) x 8)
-24V
3.4AH (12V) x 2 (approx. 30min. backup)
133(W) x 60(H) x 67(D) mm

Operating Environment
Description
Ambient Temperature
Relative Humidity

Specifications
0° to 40° Celsius
Max. 90% (non-condensing)

Electrical Characteristics (Central Office Trunk)
Description
Insulation Resistance
DC Resistance
Impedance
Leak Current

Page 3- 16

Specifications
15 mega-ohms or more at 100Vdc
On-hook conditions: 30 mega-ohms
Off-hook conditions: 1,700 ohms
On-hook conditions: 20 kilo-ohms (300 to 3,400Hz)
8 kilo-ohms (at 24Hz)
Looped conditions: 600 ohms
0 mA at on-hook conditions

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

Transmission Characteristics (For TDM Circuits)
Description
PCM Coding System
Insertion Loss
Return Loss
Longitudinal Balance
Attenuation/Frequency Distortion
Group Delay Distortion
Total Distortion
Idle Channel Noise
Impulsive Noise
Cross Talk Attenuation
Inter-modulation Products
Spurious In-Band Signals
Signal Attenuation

Specifications
A-law/U-law
0.15 dB at 1KHz
20 dB or more (300 to 3,400Hz) against 600 ohms
59 dB or more (300 to 3,400Hz)
-0.2 dB to +0.7 dB (300 to 3,400Hz)
0 to 0.3msec. (500 to 2,800 Hz)
25 dB (Input signal:-45 dBm0)/40 dB(input signal:0 dBm0)
-67 dBmop or less (psophometric noise)
-50 dBm0 or less (single frequency noise)
0 counts at -35 dBm
90 dB or more
-40 dB or more
-49 dBm0 or less
Attenuation rate: 12 dB per octave or more at 3.4 kHz above
Attenuation level: -40 dBm or less at 3.4 kHz and above
-70 dBm or less at 50 kHz and above

Line Conditions
Description
Specifications
Loop Resistance (including Telephone Set)
Analog Standard Line
Max. 600 ohms
Analog Long Line
Max. 2,500 ohms (DP 10pps), Max. 1,700 ohms (DP 20pps)
Max. 1,200 ohms (DTMF)
Loop Resistance (including Opposite End Resistance)
Central Office Trunk
Max. 1,700 ohms
Tie Line Trunk (Loop Dial)
Max. 2,500 ohms
Tie Line Trunk (E&M)
Max. 900 ohms (only E-wire condition)
Cable Length
Note 1
SN716 Desk Console
8DLC/4DLC/2DLC Card
Max. 350 meters (Max. 300 meters for 8DLC card)
4DLC/2DLC Card with
Max. 1,200 meters
AC Adapter
Dterm Series i/E
8DLC/4DLC Card
Max. 200 meters (Max. 300 meters for Dterm 8 and Dterm 8D)
2DLC Card
Max. 850 meters
4DLC/2DLC Card with
Max. 1,200 meters
AC Adapter
DSS/BLF Console
4DLC/2DLC Card with
Max. 1,200 meters
AC Adapter
Zone Transceiver
ZTII-U (for U-Interface)
Max. 1,210 meters (2-wire), Max. 2,100 meters (4-wire) @-48V
Max. 970 meters (2-wire), Max. 1,700 meters (4-wire) @-45V
ZTII-U with AC Adapter
Max. 3,950 meters
Note 1: Cable length is based on cable with 0.5mm diameter and without lightning arresters

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-17

Chapter 3 System Highlights

Rotary Dial Pulse and DTMF Signaling
(1) Rotary Dial Signal
Specifications

Description

Receiving

Dial Speed

9 to 22 pps

Break Ratio
Inter-Digit Pause
Switch-Hook Flash Detection

55 to 77 %
Min. 256 msec.
384 to 2,300 msec.

Sending
10 pps +/- 0.8pps
20 pps +/- 0.8 pps
67 +/- 3% or 62 +/- 3%
300 to 1,000 msec.(10 pps)
Not applicable

(2) DTMF Signal
Description

Specifications
Receiving
High Frequency Group

Signal Code
Low Frequency Group
Frequency Deviation
Signal Duration
Inter-Digit Pause

+/- 1.8 %
Min. 40 msec.
Min. 40 msec.

Signal Level

-46 to -5 dBm

Unwanted Frequency
Components

Not Applicable

Page 3- 18

Sending
1,209Hz
1,336Hz
1
2
4
5
7
8
*
0

697Hz
770Hz
852Hz
941Hz
+/- 0.8 %
64 or 128 milli-sec.
32 to 240 msec.
-10 dBm (low group)
- 8 dBm (high group)

1,477Hz
3
6
9
#

40 dB below the power of signal frequency

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

Multi-frequency Compelled (MFC) – R2 SIGNAL
(1) MFC Frequency Value
Frequencies

Forward Signals(Hz)

Backward Signals(Hz)

F0
F1
F2
F3
F4
F5

1,380
1,500
1,620
1,740
1,860
1,980

1,140
1,020
900
780
660
540

(2) MFC Combinations
Combination Number

Frequencies

1
2
3
4
5
6
7
8
9
10
11
12
13
14
15

F0 + F1
F0 + F2
F1 + F2
F0 + F3
F1 + F3
F2 + F3
F0 + F4
F1 + F4
F2 + F4
F3 + F4
F0 + F5
F1 + F5
F2 + F5
F3 + F5
F4 + F5

(3) Sender/Receiver Specifications
Description

Sender
Sender Transmitted Level
Frequency Variation
Receiver
Sensitivity Range
Frequency Variation

Specifications

-8 dbm to -11.5 dBm
+/- 2 Hz
-35 dBm to 0 dBm
+/- 12 dBm

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-19

Chapter 3 System Highlights

Audible Tones and Ringing Signal
Audible Tones
Frequency
350 Hz mixed with 440 Hz
350 Hz mixed with 440 Hz
480 Hz mixed with 620 Hz
480 Hz mixed with 620 Hz
2,400 Hz interrupted by 16 Hz
350 Hz mixed with 440 Hz
440 Hz mixed with 480 Hz
480 Hz mixed with 620 Hz

Tone
Dial Tone (DT)
Special Dial Tone (SDT)
Busy Tone (BT)
Reorder Tone (ROT)
Howler Tone (HWT)
Service Set Tone (SST)
Ring Back Tone (RBT)
Hold Tone (HDT)
Second Dial Tone

440 Hz mixed with 480 Hz

Call Waiting Ring back Tone
Message Waiting Indication Tone

440 Hz mixed with 480 Hz
350 Hz mixed with 440 Hz

Interruption
Continuous
0.125 sec. ON, 0.125 sec. OFF
0.5 sec. ON, 0.5 sec. OFF
2.5 sec. ON, 0.25 sec. OFF
Continuous
Continuous
1 sec. ON, 3 sec. OFF
0.25 sec. ON, 0.25 sec. OFF
0.25 sec. ON, 1.25 sec. OFF
0.25 sec. ON, 0.25 sec. OFF
0.25 sec. ON, 1.25 sec. OFF
1 sec. ON, 1 sec. OFF
0.125 sec. ON, 0.125 sec. OFF

Ringing Signal
Description
Specifications
Frequency
20 or 25Hz
Voltage
75 or 90Vrms
Note: The IPS has the capability to detect the above type of signal from Central Office and to transmit the
above type of signal to PBX stations.

Dimension and Weight
UNIVERGE NEAX 2000 IPS
Description

Specifications
Dimensions (W x D x H; mm)

Weight (kg)

Main Equipment
PIM (Fully card-mounted)
BASE
BASE TRAY (for UL)

Approx. 430 x 223 x 353
Approx. 430 x 205.2 x 61.6
Approx. 435 x 224.6 x 66.2

Approx. 11.5
Approx. 3.0
Approx. 1.7

UNIVERGE NEAX IPS DM/IPS DML/IPS DMR
Description
Main Equipment
NEAX IPS DM
NEAX IPS DML
NEAX IPS DMR

Page 3- 20

Specifications
Dimensions (W x D x H; mm)
430(W) x 365(D) x 88(H) mm

Weight (kg)

Approximately 7Kg / MODULAR
CHASSIS (when all slots are
occupied)

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

UNIVERGE NEAX IPS DM/IPS DML/IPS DMR System Specifications
System Specifications
Item

Specifications

System Capacity

LT ports: Max. 56 ports / MODULAR CHASSIS,
(Max. 64 ports including 8 virtual LT ports/MODULAR CHASSIS)
Max. 112 ports / system
AP ports: Max. 256 ports / system
IP ports: Max. 956 ports (IPS DM), Max. 128 ports(IPS DML/DMR) /system
Card slots: 8 slots / MODULAR CHASSIS (including 1 slot for MP/FP card)

Circuit Card
Mounted in
MODULAR CHASSIS

All LT/AP cards of the UNIVERGE NEAX 2000 IPS can be used for the IPS
DM/IPS DML/IPS DMR with the exception of 4LLCB, 4ODTA, and 4CSIA
cards.

Power

AC100V – 240V (automatically adjusted)

Installation Method

Desk top-setting, 19” rack-mounting

Conditions

Temperature: 5°C – 40°C (when the system is operating)
Humidity: 20% - 80% (when the system is operating)

Cooling

Cooling by FAN

Safety Standard

Complied with UL60950, CSA22.2 No. 950, EN60950, AS3260

EMC

Complied with VCCI Class A, FCC Part 15 Class A, EN55022 Class A,
AS/NZS 3548 Class A

Open Industry Standards

Open Industry Standards supported by the
UNIVERGE NEAX 2000 IPS
G.711
IP Precedence
G.726(Fax only)
Differentiated Services
G.729a
RTP
H.323 V2
RTCP
Q.931
Ipv6 (In-skin router card required(RTA))
802.1d
H.225
802.1p
H.245
802.1q
TCP/IP
802.3
UDP/IP
SNMP
DHCP
Fax-Group 3
DNS

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-21

Chapter 3 System Highlights

IP Specifications
Item
Voice Encoding
IP-PAD
FAX
Communication
Feature
DTMF Signal
Inter-office/Intraoffice Signaling

Specifications
G.729a
G.723.1 (5.3 k/6.3 k)
G.711
8 or 32 channels per card
Automatically seized per call
FAX Relay Method (T.30)

Remarks
8 kbps CS-ACELP
MP-MLQ/ACELP
64 kbps PCM
32 IPLA-PAD card is required.
G3 FAX (up to 14.4 kbps)
Super G3 Reciprocal: Not supported
8 IPLA-PAD-B card w/R9 software
required (32 IPLA not supported)
H.323 IPT/IP-PAD/DtermIP
DtermIP to DtermIP connection
DtermIP to IP-PAD connection
DtermIP to IPS connection
Point to Multipoint connection
H.323 IPT card and IP-PAD card
are required

FAX Relay Method
(Pass-thru FAX Method)
H.245
H.245
PROTIMS over IP
CCIS over IP
H.323

Jitter Control
QoS
(Quality of Service)

Dynamic Jitter Buffer
• TOS, IP Precedence
• DiffServ

LAN Interface

10BASE-T/100BASE-TX

Echo Canceller
(IP-PAD)
Payload Size

G.168

Auto Negotiation is available.
100BASE-TX is recommended.

DtermIP/CCIS
Virtual IPT

10 ms-40 ms
(G.723.1: 30 ms unit)

H.323 IPT

20 ms.-40 ms. (10 ms.
increments)
(G.723.1: 30 ms. fixed)

PAD Control

Max. voice channels per card
10 ms: 12 ch
20 ms: 20 ch
30 ms: 30 ch
40 ms: 32 ch
Maximum voice channels per card
G.729a G.711 G.723.1
20 ms.: 6ch
5ch
30 ms.: 8ch
7ch
8ch
40 ms.: 12ch
10ch
-

0 dB to +16 dB (+2 dB unit)
0 dB to –16 dB (–2 dB unit)
0 dB to –16 dB

Setting is available per Location No.
For connection via the IPT card

100 Mbps fixed mode
Full Duplex Half Duplex

10 Mbps fixed mode
Full Duplex Half Duplex

LAN interface transmission modes

Device type
MP (M606)

32IPLA/32IPLA-A
8IPLA

Auto
negotiation
X (Default)
X (Default)
X (Default)

X
X
X

—
X
—

—
—
—

—
X
—

X = available — = not available

Page 3- 22

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 3 System Highlights

System Compliance
HIPAA Summary

The security standards set forth within HIPAA require encryption; authentication and audit trail
measures to safeguard patient medical information during electronic data interchange (EDI)
transactions between healthcare providers and third party reimbursement entities. These measures
impact how patient medical information is exchanged, as well as accessed within a facility’s data
network. Claims transmissions using Internet, intranets, extranets or private data networks are subject
to HIPPA security standards as previously outlined.
HIPAA Compliance

Patient medical data is not accessed, stored or otherwise exchanged over a healthcare facility’s private
branch exchange (PBX). However, an NEC manufactured private branch exchange does provide the
flexibility to create detailed call accounting records that can be utilized to audit or otherwise track
utilization of a facility’s NEC telecommunications system and/or PBX. A NEC PBX can also be
configured to restrict access to designated telephone extensions. Furthermore, an NEC PBX can also
be configured to utilize password access to individual telephones, voice mailboxes, etc. for purposes of
end user authentication.
So, under the guise of the security standards within HIPAA, NEC Private Branch Exchange
equipment is HIPAA compliant and does provide the necessary security measures for use within any
business setting that is subject to administrative rules outlined with the Health Insurance Portability and
Accountability Act.
Department of Defense Compliance

The NEC NEAX 2000 Internet Protocol Switch (IPS) with software release D1.8.20, hereinafter referred
to as the system under test (SUT), meets all of its critical interoperability requirements and is certified
for joint use within the Defense Switched Network (DSN) as a Private Branch Exchange (PBX) 2.
However, since PBX2s do not support the Military Unique Feature Requirements detailed in reference
(c), connectivity to the DSN is not authorized until a waiver is granted by the CJCS. PBXs are Military
Department (MILDEP) controlled elements of the Defense Switched Network (DSN). PBX2 switches
have no military unique features (MUFs) and can only serve Department of Defense (DOD), non-DOD,
non-governmental, and foreign government users having no missions or communications requirement
to ever originate or receive Command and Control (C2) communications. C2 users and Special C2
users are not authorized to be served by a PBX2. This interoperability test status is based on
evaluation of Chairman of the Joint Chiefs of Staff (CJCS) validated Generic Switching Center
Requirements (GSCR) for PBX2s and the overall system interoperability performance. The SUT also
offers a Voice over Internet Protocol capability; however this capability is not covered under this
certification.
This statement was obtained from a Department of Defense document posted on there web site and full
copy of the compliance document can be downloaded from the following link.

http://jitc.fhu.disa.mil/tssi/apl.html

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Page 3-23

Chapter 4 Equipment List
Module/Installation Hardware
This table shows the names and functions of the modules.
Equipment Name

Function Name

Remarks

SN1729 PIMMJ

PIM

Port Interface Module (PIM)
Maximum 64 physical ports per PIM. Houses two batteries for
protection from short power interruption (for 30 minutes).
At maximum configuration, the system consists of eight PIM’s,
and provides a total of 512 physical ports (64 ports × 8).

SN1730 PIMMK

PIM

Port Interface Module (PIM) for Backup CPU System
Maximum 64 physical ports per PIM.
Houses two batteries for protection from short power
interruption (for 30 minutes).
One PIM is required per Backup CPU System.

COVER PARTS
ASSEM-A

COVER PARTS
ASSEM

Cover Parts Assembly
One cover parts assembly is required for each PIM.

SN1545
BASE

BASE/TOP
ASSEM

Base/Top Cover Assembly
One base and top cover assembly is required for each stack.

SN1731
BATTMH

BATTM

Battery Module for housing PIM or CS (ZT) backup batteries
Houses two pairs of batteries for protection from long power
interruption (for 3 hours).

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
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Page 4-1

Chapter 4 Equipment List

This table shows the name and functions of installation hardware.
Equipment Name
HANGER ASSEM (B)

Remarks
Wall Hanger Assembly
One HANGER ASSEM is required per PIM for Wall Mounting
Installation.

MOUNTING BRACKET

Safety Mounting Bracket
Used as an overhead hanger for Floor Standing Installation. Wire,
chain or eyebolts are to be locally provided, to secure the bracket.
To be installed on the top PIM in four or more modules of stack. It
provides 1.1G shockproof construction.

19” RACK BRACKET (A)

19-inch Rack Mounting Bracket Type A
One bracket is required for one PIM configuration.
One bracket is required for the top PIM of multiple module
configurations.

19” RACK BRACKET (B)

19-inch Rack Mounting Bracket Type B
One bracket is required for the bottom module of multiple module
configurations.

I/F BRACKET ASSEM

Inter Frame Bracket Assembly
Used to join the frames in two-stack configuration, for Floor
Standing Installation.

BASE TRAY ASSEM

Base Tray Assembly
One BASE TRAY is required per one frame for Floor Standing
Installation of Stationary Equipment.

Page 4- 2

NEAX2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 4 Equipment List

Circuit Cards
The circuit cards used for UNIVERGE NEAX 2000 IPS systems are divided into the following
types. According to these card types, the mounting locations of card and port allocation of the
Time Division Switch are varied.
Common Control Cards

Main Processor (MP)
Firmware Processor (FP)
Ethernet
Power
Line/Trunk (LT) Cards

IP PAD, Line Circuit (LC), Central Office Trunk (COT), Tie Line Trunk (LDT/ODT), etc.
Application Processor (AP) Cards

SMDR/PMS/MCI/Hotel Printer Interface (AP00)
T1/E1 Digital Trunk Interface (DTI)

Common Control Cards
The following table shows the names and functions of each control card.
Common Control Cards
Equipment
Name

PN-CP24

PN-CP27

Function
Name

Remarks
MP Card for Single MP System
Provides LAN control function, system-based Device Registration Server
(DRS), built-in FP, built-in OAI, built-in SMDR, Virtual IPT, Virtual CSH, 33
MHz PCI BUS, Memory (SDRAM 64 MB, Flash ROM 16 MB), TDSW
(1024CH × 1024CH), 16-line CFT, PB sender, Clock, 2-line PLO (receiver
mode/source mode), two RS-232C ports, 2-line DAT (Recording duration:
Maximum 128 seconds), DK, 4-line PB receiver, Modem for remote
maintenance (33.6 Kbps), Music-on-Hold tone, BUS interface. BUS interface
functions as a driver/receiver of various signals, adjusts gate delay timing,
cable delay timing, monitors I/O Bus and PCM BUS.
Supports MP Program Download.
One card is required per system.
Main Processor Card for Backup CPU system
Provides LAN control function, system-based Device Registration Server
(DRS), Virtual FP, built-in OAI, built-in SMDR, Virtual IPT, Virtual CSH, 33
MHz PCI BUS, Memory (SDRAM 32 MB, Flash ROM 9 MB), TDSW (1024CH
× 1024CH), 16-line CFT, PB sender, Clock, 2-line PLO (receiver mode/source
mode), two RS-232C ports, 2-line DAT (Recording duration: Maximum 128
seconds), DK, 4-line PB receiver, Modem for remote maintenance (33.6
Kbps), Music-on-Hold tone, BUS interface. BUS interface functions as a
driver/receiver of various signals, adjusts gate delay timing, cable delay
timing, monitors I/O Bus and PCM BUS.
Two cards are required per Backup CPU system. One card for active MP
[MP0] and another card for stand by MP [MP1].

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
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Page 4-3

Chapter 4 Equipment List

Common Control Cards (Cont.)

PN-CP31

PN-CP15

PZ-M606

Main Processor Card for Remote PIM (DML/DMR)
Provides LAN control function, System-based Device Registration Server
(DRS), built-in FP, 33 MHz PCI BUS, Memory (SDRAM 64 MB, Flash ROM
16 MB), TDSW (1024CH × 1024CH), 16-line CFT, PB sender, Clock, 2-line
PLO (receiver mode/source mode), one RS-232C port, 4-line PB receiver,
internal Music-on-Hold tone, BUS interface.
BUS interface functions as a driver/receiver of various signals, adjusts gate
delay timing, cable delay timing, monitors I/O Bus and PCM BUS.
Supports MP Program Download.
One card is required per system.

Firmware Processor Card for the IPS.
Provides Line/Trunk interface, Memory (RAM 768 KB), and inter-module BUS
interface. BUS interface functions as a driver/receiver of various signals,
adjusts gate delay timing, cable delay timing, monitors I/O Bus and PCM
BUS. When the system consists of three PIM’s or more, one each of this card
is mounted respectively in PIM2, PIM4, and PIM6.
3/4 PIM System: 1 per System
5/6 PIM System: 2 per System
7/8 PIM System: 3 per System

ETHER

Ethernet Card, 1 per MP
For Peer to Peer Connection and MAT/OAI
Mounted on MP card to accommodate the Ethernet and transmit/receive a
signal of TCP/IP protocol. Provides Auto Negotiation function (Always set to
ON). 10 BASE-T/100 BASE-TX cable is connected directly to this card.
Power Supply Card for DESKCON
Provides –48 V DC power.
Maximum 3 cards per PIM. One card per two Desk cons
Occupies two physical slots width per card.

PN-PW00

EXTPWR

PZ-PW121

AC/DC
PWR

Main Power Supply Card
Input: AC120 V/240 V (50 Hz/60 Hz)
Output: –27 V (4.4 A), +5 V (7.2 A), CR (38 mA), +90 V (80 mA)
One card is pre-installed per PIM.

DC/DC
PWR

Power Supply Card for CS (ZT) and PN-4LLCB for Long Line
Input: –24 V DC
Output: –48 V DC (1.7 A)
One card per PIM.
Maximum 16 ZT’s per card. Requires Ferrite Core for Long Line function

PZ-PW122

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NEAX2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 4 Equipment List

Application Processor Cards
The following table shows a summary of the Application Processor cards for UNIVERGE NEAX 2000
IPS systems.
Maximum 24 cards per system
Maximum 256 ports per system

Application Processor Card Name and Function
Function
Digital Trunk Interface
(DTI/PRT/BRT)

Name
SPN-24DTAC-B (AP)

SPN-30DTC-C (AP)

SPN-24PRTA-D (AP)

SPN-24CCTA-A (AP)

SPN-2BRTC (AP)

SPN-4BRTA-F(AP)

Port

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Remarks
T1 Digital Trunk Interface (1.5 Mbps)
Card:
Accommodates 24-channel PCM digital
lines. (Available up to 256 AP ports)
T1 w/CSU
E1 Digital Trunk Interface (2 Mbps) Card:
Accommodates 30-channel PCM digital
lines. (Available up to 128 AP ports)
ISDN Primary Rate (23B+D) Interface
Card w/CSU:
Provides a built-in D-Channel Handler
(DCH) and built-in CSU
CCIS (1.5 Mbps) Trunk Card:
Provides a built-in Common Channel
Handler (CCH) of CCIS.
2-line Basic Rate (2B+D) Interface Trunk
Card:
Accommodates two two-channel PCM
digital lines.
4-line Basic Rate (2B+D) Interface Trunk
Card:
Accommodates four two-channel PCM
digital lines.
Point to Point Connection Only

Page 4-5

Chapter 4 Equipment List

Application Processor Cards (Cont.)
Function
Common Channel
Handler for NO.7
CCIS

D-Channel Handler
for ISDN PRI

D-Channel Handler
for ISDN Station

ZT Handler

Name
SPN-SC00 CCH-D (AP)

SPN-SC01 DCH-C (AP)

SPN-SC03B 8ICH (AP)

SPN-SC03B 8CSH-C(AP)

SPN-AP00B MRC-C(AP)

SMDR/MCI/PMS/CIS
/Hotel Printer
Interface

SPN-AP00B MRC-F(AP)

PZ-M537
Note 1

Port
1

SPN-4RSTBA-B (AP)

SPN-4RSTBA-911 (AP)

SPN-4RSTC-A (AP)

MFC Receiver/Sender

Page 4- 6

Remarks
Common Channel Handler for CCIS
(1/Trunk Route). Transmits/receives
signals on the common signaling channel
of CCIS.
D-Channel Handler Card:
Transmits/receives signals on the DChannel of ISDN Primary Rate (23B+D)
interface.
ISDN-channel Handler Card:
Provides the D-Channel signaling
interface and controls a maximum of four
ILC cards (Layer 2 and 3).
CS (ZT) Handler Card:
Provides the D-Channel signaling
interface and controls a maximum of two
CSI cards, eight CS (ZT) s.
Supports Dukane Nurse Call system.
Application Package for SMDR (2400
basic and 1400 only/ PMS/ Hotel Printer/
CCIS Centralized SMDR
Default data supported.
Application Package for SMDR (2400
basic and extended only)/MCI/DND
Automatic Set/Reset at Appointed Time
and PMS Executed Task Print out.
Memory Expansion Card for AP00 Card:
• The system capacity is expanded when
PZ-M537 is mounted on PN-AP00-B
(AP00) card.
4-line MF Receiver, MFC
Receiver/Sender Card:
• Used for MF/MFC-R2 Signaling on
DID/DOD trunks.
• A maximum of four cards can be
provided per system, including the
PN-4RSTC card.
4 circuit MF Receiver for Enhanced 911
Requires CAMA Trunks
4 circuit ID Number Received for Caller ID

NEAX2000 IPS Request For Proposal (RFP) Reference Guide
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Chapter 4 Equipment List

Application Processor Cards (Cont.)
Function

Name

Port

SPN-AP00B DBM-C (AP)

Remarks
Database Memory for Wireless Roaming
and OAI Applications using Free Location
Facility (FLF).
1 Card / system

Wireless Roaming
SPN-SC01 DCH-Q (AP)

SPN-SC01 QSIG (AP)

DCH Card for Wireless Roaming
1 Card / Tie Line Route
Installed with DTI Card
QSIG Protocol Handler
ISDN Primary Rate (23B+D) Interface
Card w/CSU and built-in QSIG DChannel Handler (DCH).
Also provides Name Display.

QSIG

32-Party Conference

SPN-24PRTA-QSIG (AP)

SPN-CFTC (AP)

32-party Conference Trunk
8 cards per system max. Note 2

Note1: The Expansion Memory (PZ-M537) is the on-board sub-memory card for the AP00-B. It provides
additional memory capacity as shown below.
Note2: Each CFTC reduces system trunk capacity by 32
No EXPMEM
Type of
PN-AP00-B

Local Office of
Centralized
Billing-CCIS
/Stand-alone is
Provided.

With EXPMEM

Center Office
of Centralized
Billing-CCIS is
Provided.
Local Office of

PN-AP00-B
with MRC-C
program

1600

800

PN-AP00-B
with MRC-F
program

2620

1965

Centralized
Billing-CCIS
/Stand-alone is
Provided.

Center Office
of Centralized
Billing-CCIS is
Provided.

27000: Call Record for CIS is not provided.
26000: Call Record for CIS is provided.

23580

22925

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
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Page 4-7

Chapter 4 Equipment List

Line/Trunk (LT) Cards
The following table shows a summary of the Line/Trunk (LT) cards for UNIVERGE NEAX 2000 IPS. The
LT cards may be installed in slot 00 to 11 of PIM 0-7, with the following conditions:
Line/Trunk (LT) Cards
Function

Name

Port

Analog Station
PN-8LCAA

PN-8LCAD

PN-8LCS

PN-4LCD-A

Page 4- 8

Remarks
8 Line Analog Station Card
Message Waiting Control: 8 circuits
Momentary Open: 8 circuits
Loop Resistance: Max. 600ohms including
telephone set
8 Line Analog Station Card
Caller-ID for Analog Station
Loop Resistance: Max. 600 ohms
Message Waiting Lamp control
Momentary Open for each circuit

8 Line Analog Station Card
Momentary Open: 8 circuits, Message
Waiting notification via Stutter Dial
Tone only.
Loop Resistance: Max. 600 ohms

4 Line Analog Station Card with MW Lamp
control, Momentary Open: 4 Circuits
On-Board power (80V) is included.
Loop Resistance: Max. 600ohms including
telephone set

PN-AUCA

PN-4LLCB

2 Line Analog Universal Trunk
2 Line Long Line Circuit with PFT
Loop resistance: max. 2500ohms (LLC)
Two reverse circuits is included
On Board power (-48V) is included
4 Long Line Analog Station Card
Works in either mode:
Long Line Station (Requires Ferrite Core)
Caller ID (Requires PN-4RSTF-A)
Loop Resistance:
Max. 2500ohms including telephone set
PZ-PW122 is required.
Can not be used in IPS DM or IPS DMR

NEAX2000 IPS Request For Proposal (RFP) Reference Guide
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Chapter 4 Equipment List

Line/Trunk (LT) Cards (Cont.)
Function
ISDN Station

Name

Remarks

2 Line ISDN Line Circuit Card
• Provides a physical interface to ISDN
Terminals.
• Occupies eight time slots per one card.

PN-8DLCL

8-line Digital Line Circuit Card for Dterm Series
i/E/III, Dterm Elite, IPK, DSS Console,
DESKCON
[–27 V version, two-wire type]

PN-8DLCP

8-line Digital Line Circuit Card for Dterm Series
i/E/III, DSS Console, DESKCON
[–27 V version, 2-wire type]

4-line Digital Line Circuit Card for Dterm Series
i/E, Dterm Elite, DSS Console, DESKCON
[–27 V version, 2-wire type]
Provides Line Test function.

4-line Digital Line Circuit Card for Dterm Series
i/E, DSS Console, DESKCON
[–27 V version, 2-wire type]
Provides Line Test function.

4-line Digital Long Line Circuit Card for Dterm
Series i/E, DSS Console
[–48 V version, two-wire type]
PZ-PW122 required

2-line Digital Long Line Circuit Card for Dterm
Series i/E, Dterm Elite, DSS Console
[–48 V version, two-wire type]:
• Provides Line Test function.
• Equipped with –48 V DC-DC on-board power
supply.
Max. 850 m (2788.7 ft.)

SPN-2ILCA

PN-4DLCM

Digital Station

Port

PN-4DLCQ

PN-4DLCT

PN-2DLCN

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Page 4-9

Chapter 4 Equipment List

Line/Trunk (LT) Cards (Cont.)
Function

IP PAD
(IP Station
Peer-to-Peer)

Name

Port

SPN-32IPLAA IP
PAD-E

SPN-8IPLA IP
PAD-C

Remarks
32-channel IP-PAD card:
• Provides Packet Assembly/Disassembly to
accommodate Legacy Line/Trunk interface.
• Used with a maximum of two 16VCT cards.
• Two cards can be accommodated per built-in
FP/FP card, a maximum of eight per system.
• 100 BASE-TX twisted pair cable is connected
directly to this card.
• Supports T.30 or Pass-Through Fax
(G.711/G.726). 16VCT is not required
for G.711, 16VCT is required for
G.726. (R9 enhancement).
• Auto-Negotiation or 100Mbps Full Duplex is
selectable.
8-channel IP-PAD Card
Provides LAN Interface, Packet Assembly/
Disassembly to accommodate Legacy
Line/Trunk interface. And provides voice
compression DSP control functions such as
voice compression control, DTMF relay.
• Voice compression protocols:
G.711 (64 kbps), G.723.1 (5.3 Kbps/6.3 Kbps),
G.729a (8 Kbps)
• FAX protocol: Pass-through (G.711, G.726)
• Auto-Negotiation or 100Mbps Full Duplex is
selectable.
When mounting 24DSP (PZ-24IPLA) card, this
card can provides up to 32-channel of IP-PAD
(When using G.723.1, provides up to 24-channel
of IP-PAD).
Two cards can be accommodated per built-in
FP/FP card, maximum 8 per system.
10BASE-T/100BASE-TX twisted pair cable is
connected directly to this card.

PZ-24IPLA

SPN-16VCTAA IP
PAD-B

Page 4- 10

24-channel IP PAD and VCT daughter board.
Mounts on SPN-8IPLA to provide up to 32
G.711, G.729a and 16 G.723.1 PAD channels.
(T.30/T.38 is not supported)

16-channel CODEC Card for IP-PAD:
• Used with IP-PAD card.
• Two cards can be accommodated per IP-PAD
card, maximum 16 per system.
• Voice compression protocols:
• G.723.1, G.729a, G.711,
• FAX (14.4 Kbps), DTMF signals.

NEAX2000 IPS Request For Proposal (RFP) Reference Guide
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Chapter 4 Equipment List

Line/Trunk (LT) Cards (Cont.)
Function

Name

Port

PN-8COTS

Remarks
8-line Central Office Trunk Card
(Loop Start/ Ground Start Trunk):
Provides loop detection, sending/detecting
ground on Tip/Ring wire.

8-line Central Office Trunk Card
(Loop Start Trunk):
Provides loop detection, receiving/sending the
Caller ID (CLASS SM) signal.

4-line Central Office Trunk Card
(Loop Start/Ground Start Trunk):
Provides loop detection, sending/detecting
ground on Tip/Ring wire.

4-line Central Office Trunk Card
(Loop Start Trunk):
Provides loop detection, receiving/sending the
Caller ID (CLASS SM) signal.

PN-2ODTA

2-line Out Band Dialing Trunk Card:
Used as either a 2-wire E&M trunk or a 4-wire
E&M trunk. Equipped with –48 V DC-DC onboard power supply. Both No. 0 and No. 1
circuits must be set to the same purpose
(2-wire or 4-wire) in one card.

PN-4ODTA

4-circuit analog 2W/4W Tie line card. The PZPW122 (-48V Power Supply) is required in
case of E&M Type-I signaling.

PN-8COTQ
Analog C.O. Trunk
PN-4COTB

PN-4COTG

Analog Tie Line
Trunk

PN-4DIDA

Analog DID Trunk

4-line Direct Inward Dialing Trunk Card:
• Provides loop detection, sending reverse
signal and PB to DP signal conversion.

PN-AUCA

2L ANALOG UNIVERSAL TRUNK
Used as 2-Line Long Line Circuit with PFT or
2-Line DID. Loop Resistance: MAX.2500ohms
(LLC) Two Reverse Circuits Included
On-Board Power (-48VDC) included

DTMF Receiver

PN-8RSTG

8-line PB Receiver Card:
Used for a PB station line, DID or tie line.

Caller ID Sender

PN-4RSTF-A

4-line Sender Card for Caller ID Display on
analog single line telephones:
PN-4LLCB is required. Max. 4/sys.

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Page 4-11

Chapter 4 Equipment List

Line/Trunk (LT) Cards (Cont.)
Function
Conference Trunk

Digital
Announcement
Trunk

Tone Trunk

Power Failure
Transfer

Name

Port

PN-CFTB

6-/10-Party Conference Trunk with Amp
6-party (1 CFTB card): 10 ports
10-party (2 CFTB cards): 20 ports

PN-2DATA

2-circuit Digital Announcement Trunk
(Max. 60 sec. per channel)

PN-4DATC

4-circuit Digital Announcement Trunk
(Max. 120 sec. per channel)

PN-TNTA

2-line Tone/Music Source Interface Card:
• Used for BGM or Music on Hold.
• Provides two jacks for an external tone/music
source.

PZ-8PFTB

PN-AUCA

SPN-IPTB-A

SPN-4VCTI-A

SPN-4VCTI-B

Page 4- 12

8-line Power Failure Transfer Card:
• Mounted in PFT slot of PIM.
• One card per PIM.
2 Line Analog Universal Trunk card that
provides 2 Lines of Power Failure Transfer

IP Trunk card for VoIP (CCIS/ p-mp). This card
is used together with a max of four PN-4VCT
cards (16 channels) for connection with
2000/2400 non peer-to-peer connection.
1 card/PIM, 8 cards (Max. 127ch)/system

4-channel CODEC Card for IP Trunk.
Voice compression protocols:
• G.723.1, G.729a, G.711,
• FAX (14.4 Kbps), DTMF signals.
• Used with PN-IPTB card.

VoIP H.323 IP Trunk card. Supports up to 12
channels. (G.729a. PAYLOAD 40ms). Also
requires CCIS and IPT key per IPTB and NEC
GK1000 software.
Has one RJ45 connectors for 10/100
Ethernet/Fast Ethernet to router.

4-channel voice translator CODEC card for
H.323 includes BUS cable.
Voice compression: G729A /G723/G711.
Max 3 VCT’s per SPN-IPTB-B

IP Trunk

SPN-IPTB-B

Remarks

NEAX2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

Chapter 4 Equipment List

Line/Trunk (LT) Cards (Cont.)
Function
Wireless Zone
Transceiver
interface

Fiber Optical

In-Skin Router

Name

Port
16

4-circuit CS interface for U-interface ZT.
Two slot card, 16 LT ports. The PZ-PW122 is
required.

Optical Fiber Interface Card:
Provides optical fiber interface for T1/E1 Digital
Trunk Interface (1.5 M/2 Mbps) or Remote PIM.
Line length: 10 km (6.2 miles) or less. Line
coding: CMI.

PN-RTA

In-Skin Router Card
10/100BASE-TX:1, 10BASE-T:1, RS2332C(Dsub 9pin)
Maximum 8 cards per PIM

PZ-M649

T1 Digital Trunk Interface (1.5Mbps) Card
Mounted on PN-RTA Card
Provides a built-in CSU.

PZ-M623

Ether Control Card
Mounted on PN-RTA Card
Provides one additional 10BASE-T port

PZ-VM07

4/8

4 or 8 Port Voice Mail card (NEAXmail IM-16LX)
This card can provide up to 16 ports of digital
line circuit interface.
Occupies VM slot and up to one LT slot
One card per PBX is available.

PZ-VM08

8/16

PN-DK00

SPN-4CSIA

PN-M10

NEAXmail IM-16LX

External Relay
Control

Remarks

4 Port Voice Mail card used to expand from 8 to
16 ports. Provides one connector for
Occupies VM slot plus and one LT slots
External Relay Contact Control/Scanning
(8-circuit/card)
Works in either mode:
Scanning
Control
8-circuit
0-circuit
4-circuit
4-circuit
0-circuit
8-circuit

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Chapter 5 Station Equipment
Terminal Line Up
A variety of terminal equipment may be connected to the UNIVERGE NEAX 2000 IPS systems.
Dterm Series i Terminals
Analog
DTR-1-1 (Single-Line Analog)
DTR-1HM-1 (Hospitality Single-Line Analog)
DTH-1-1 (Low End Single Line Analog)
Dterm Series i (IP) Multi-line Digital Terminals
ITR-4D-3 (4-Line display)
ITR-8D-3 (8-Line display)
ITR-16D-3 (16-Line display)
ITR-16LD-3 (16-Line Desi-less display)
ITR-32D-3 (32-line display)
Dterm Series i (TDM) Multi-line Digital Terminals
DTR-2DT-1 (2-Line non display)
DTR-4D-1 (4-Line non display)
DTR-8-2 (8-Line non-display)
DTR-8D-2 (8-Line with display)
DTR-16D(BL)-2 (16-Line with Back lit LCD display)
DTR-16D-2 (16-Line with display)
DTR-16LD-2 (16-Line Desi-less display)
DTR-32D-1 (32-Line with display)
DCR-60-1 Console (Add-On Module/DSS/BLF)
D16(LD)-R ADM (16 button Desi-less Adapter)

Dterm Series E Terminals
Multi-line (TDM) Digital Terminals
DTP-2DT-1 (2-Line Digital Terminal)
DTP-8-1 (8-Line Digital Terminal)
DTP-8D-1 (8-Line Digital Terminal with Display)
DTP-16D-1 (16-Line Digital Terminal with Display)
DTP-32D-1 (32-Line Digital Terminal with Display)
DTP-32DA-1 (32-Line Digital Terminal with Display for ACD)
DCU-60-1 CONSOLE (Add-On Module/DSS/BLF)

Dterm Cordless Terminals
DTR-4R-2 (Dterm Cordless II Terminal)
DTH-4R-1 (Dterm Cordless Lite II Terminal)
Plantronics CT Cordless

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Chapter 5 Station Equipment

Inaset
ITR-240G-1 (3” gray scale LCD and 16 programmable keys)
ITR-320C-1 (5.1” color LCD and 32 programmable keys)
ITR-320G-1 (5.1” gray scale LCD and 32 programmable keys)

Wireless Handset
Dterm PSIII

Dterm IP Soft Phones
Dterm SP20
Dterm SP30

Dterm Extenders
Dterm Extender (Analog)

Analog Terminals
NEC offers three distinct analog terminals.
Descriptions of the Analog Terminals
Description
DTR-1-1 (WH) TEL
DTR-1-1 (BK) TEL
DTR-1HM-1 (WH) TEL
DTR-1HM-1 (BK) TEL
DTH-1-1 (BK) TEL

Remarks
Fully modular with Redial key, Flash key, Message Waiting lamp, Data Jack
and Ring/Handset Receive Volume.
Fully modular with Redial key, 'Flash' key, Message Waiting Lamp, Data Jack,
eight programmable Feature/Speed Dial keys and Ring/Handset Receive
Volume.
Fully modular with Redial key, 'Flash' key, Mute Key, Message Waiting Lamp,
Data Jack, four programmable Feature/Speed Dial keys and Ring/Handset
Receive Volume

Line Conditions of the Analog Terminals
Terminal Type
DTR-1-1
DTR-1HM-1
DTH-1-1

Card Type
PN-4LCD-A (max. 600 ohms loop resistance)

Cable Length* (Cable 0.5/24 AWG)
Approximately: 1.43km (.88 miles)

PN-8LCAA (max. 600 ohms loop resistance)

Approximately: 1.43km (.88 miles)

PN-AUCA (max. 2500 ohms loop resistance)

Approximately 12.29km (7.63 miles)

PN-4LLCB (max. 2500 ohms loop resistance)

Approximately 12.29km (7.63 miles)

Note: *Cable length is based on the diameter of the cable and the terminal impedance.

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Chapter 5 Station Equipment

Specification for Analog Terminals
Item
Size
Color
Dial Pad
Type of Dial
Function Buttons
Message Waiting Lamp
Message Waiting Lamp Operating
Voltage
Speed Dials
Hook flash Timer
Redial Key
Ring Vol. Control
Ring Tone Pitch Control
Handset Receiver & Speaker Volume
Handset
Handset Cord
Directory Card
Data Jack
Data Jack Position
Wall-Mount Unit
Electrostatic Discharge
Approvals

Description
224mm x 165mm +/5mm(Length x Width)
Black or White (DTH-1-1 black only)
12-Key Dial Pad: 4 Rows and 3 Columns; Metropolitan Dial Pad
with Alphabet, * and # buttons; Button 5 has a Raised Dot
DTMF and Dial Pulse
Hook flash, Redial Key,
Speaker Monitor and Hold Key(DTR-1HM Only)
Neon Lamp with Window Design -Glow Through Filter Raised
from Surface with MW and Incoming Ring Indication
Activation Voltage 88V to 108V, Deactivation
Voltage 53V or Less
(DTR-1HM Only) 8 Buttons (2 Columns), Maximum 21 Digits
(DTH-1-1) 4 buttons, Maximum 21 digits
630+/10ms (Fixed)
Maximum of 31 digits
4 Levels (Off Low Middle High) Programmable
3 Levels Programmable
6 Levels (Volume Key)
Hearing Aid Compatible, Dynamic Type Element
12 feet
Large Convenient Directory Card
Dedicated Jack; Used for Connection to Modem
Back of Telephone
Built-in
Can Withstand +/- 20kv Discharge
c-UL (Ul 60950 3rd Edition) FCC part 15, 68 IC

Dterm Series i (IP) Terminals
Dterm IP terminals offer features and benefits that brilliantly capture the essence of form and
functions, helping your business take service and performance to newer, more profitable
heights. The Dterm IP terminals consist of four distinct models, a 4, 8, 16 or 32 feature key
terminals, in order to meet diverse telephone terminal needs.
Descriptions of the Dterm IP Terminals
Description
ITR-4D-1 (BK)
ITR-8D-3 (BK)
ITR-8D-3 (WH)
ITR-8D-3 (BB)
ITR-16D-3(BK)
ITR-16D-3(WH)
ITR-16D-3(BB)
ITR-16LD-3(BK)
ITR-16LD-3(WH)
ITR-32D-3(BK)
ITR-32D-3(WH)
ITR-32D-3(BBK)

Remarks
4 Programmable Line/Feature Keys with Display

Physical Features

8 Programmable Line/Feature keys with Display
16 Programmable Line/Feature keys with Display
16 Programmable DESI-LESS Line/Feature keys
with Display
32 Programmable Line/Feature keys with Display

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•
•
•

Tilt-able LCD Unit
Adjustable Base
Built-in Wall Mount Unit

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Chapter 5 Station Equipment

Call Handling Features
Four Local Soft Key Controls (detail functions are dependent on PBX)
Large Message Waiting LED
24 Character, 3-Line LCD
Built-in Headset Jack Connector Note
Speed Dial/DSS Buttons
Programmable Line Keys with 2-Color LED
Eleven Dedicated Function Keys (Feature, Recall, Conf, Redial, Hold, Transfer, Answer,
Speaker, MIC, Directory and Message Note )
Built-in Half Duplex Hands-free Unit
Note: DTR-4D-1 does not include headset jack or directory and message functions

Convergence Features
Two 10/100 full duplex Ethernet ports- One which connects the Dterm IP to the local Ethernet
Network, the other provides connectivity for a local workstation.
Three types of powering options:
o Local AC adapter
o CDP (Cisco Discovery Protocol) power for those infrastructures with an installed base of
Cisco gear
o Spare pair power across the Ethernet network.
Transportable QoS, which follows the user no matter where they log in.
Multiple Voice Coding support, which automatically negotiates to a common setting.
G.711 providing an international standard for encoding/decoding telephony signals on a
64 Kbps Non-compressed channel.
o It also supports the compression algorithms G.729A (8Kbps) and G.723.1 (5.3/6.3
Kbps).

o
o

Line Conditions
ITR-4D-1
ITR-8D-3
ITR-16D-3
ITR-16LD-3
ITR-32D-3

Page 5- 4

Max. 328 ft. between Ether/IP-PAD and Router/ Switching Hub

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Chapter 5 Station Equipment

Dterm Series i (TDM) Multi-line Digital Terminals
The Dterm Series i, digital terminals have a user-friendly ergonomic design and is available in
six distinct digital models.
Descriptions of Terminals
Description
DTR-2DT-1 (BK) TEL
DTR-2DT-1 (WH) TEL

DTR-4D-1(BK) TEL

Remarks
2 LINE TERMINAL - available in black (BK) or white (WH).
(does not support optional adapters)
Fully modular with 2 Flexible, 2-color LED Line keys, eight Function Keys,
built-in Speakerphone and electronic volume and tone controls.
4 LINE TERMINAL – available in black only. Fully modular with 4 Flexible,
2-color LED Line keys, nine Function Keys, built-in Speakerphone, wall
mount unit, four soft keys, Large LED, Electronic Volume and Tone Controls
and tilt stand. (does not support optional adapters)

DTR-8-2 (BK) TEL
DTR-8-2 (WH)TEL

8 LINE TERMINAL – available in black (BK) or white (WH).
Fully modular with 8 Flexible, 2-color LED Line keys, eleven Function Keys,
built-in Speakerphone, headset jack, wall mount unit, four soft keys, Large
LED, Electronic Volume and Tone Controls and tilt stand.

DTR-8D-2 (BK) TEL
DTR-8D-2 (WH) TEL

8 LINE DISPLAY TERMINAL - available in black (BK) or white (WH).
Fully modular with 8 Flexible, 2-color LED Line keys, eleven Function Keys,
built-in Speakerphone, headset jack, wall mount unit, 24-character by 3-line
display, four soft keys, Large LED, Microphone button, Message button,
Directory button, Electronic Volume and Tone Controls, and tilt stand.

DTR-16D(BL)-2(BK) TEL

16 LINE BACK-LIT DISPLAY TERMINAL – available in black only, 16
flexible 2 color line keys, eleven function keys, built-in speaker phone,
headset jack, wall mount unit, back-lit 24-character 3 line display, four soft
keys, large LED, electronic volume and tone controls and tilt stand

DTR-16D-2 (BK) TEL
DTR-16D-2 (WH) TEL

DTR-16LD-1 (BK) TEL
DTR-16LD-1 (WH) TEL

16 LINE DISPLAY TERMINAL - available in black (BK) or white (WH).
Fully modular with 16 Flexible, 2-color LED Line keys, eleven Function Keys,
built-in Speakerphone, headset jack, wall mount unit, 24-character by 3-line
display, four soft keys, Large LED, Microphone button, Message button,
Directory button, Electronic Volume and Tone Controls and tilt stand.
16 LCD DESI-LESS TERMINAL – 16 Flexible LCD appearances for Speed
Dial/DSS, feature or line appearance, 11 Function Keys, Built-in
Speakerphone (Half Duplex), Headset Jack, Wall Mount Unit, 4 Soft-keys,
In-Line Power and 12 ft. Handset Cord

DTR-32D-1 (BK) TEL
DTR-32D-1 (WH) TEL

32 LINE DISPLAY TERMINAL - available in black (BK) or white (WH).
Fully modular with 32 Flexible, 2-color LED Line keys, eleven Function Keys,
built-in Speakerphone, headset jack, wall mount unit, 24-character by 3-line
display, four soft keys, Large LED, Electronic Volume and Tone Controls
and tilt stand.

DCR-60-1 (BK) CONSOLE
DCR-60-1 (WH) CONSOLE

ATTENDANT ADD-ON CONSOLE - Requires an AC-R ADP (included).
Fully modular with 48 programmable, 2-color LED keys (for station trunk
appearances), 12 Function keys with red LED, and tilt stand. Terminal
available in: black (BK) or white (WH).

D16(LD)-R ADM (BK)
D16(LD)-R ADM (WH)

16 LCD DESI-LESS ADD ON MODULE – available in black (BK) or white
(WH) Provides Speed dial directory and one-touch speed dial.

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Chapter 5 Station Equipment

Standard features
Adjustable Base
Built-in Wall Mount Unit
Four Local Soft Key Controls (detail functions are dependent on PBX, only provided on
terminals with display)
Built-in Half Duplex Hands-free Unit
Large Message Waiting LED
24 Character, 3-Line LCD on display equipped models
Built-in Headset Jack Connector Note
Speed Dial/DSS Buttons
Programmable Line Keys with 2-Color LED
Eleven Dedicated Function Keys (Feature, Recall, Conf, Redial, Hold, Transfer, Answer,
Speaker MIC, Directory and Message Note)
Ringing Sound Variation
LCD indication of volume control
FCC Part.15 Class B Approved
Replaceable Function/Dial Key pad:
o Used for foreign languages, ACD instruments, etc.
Un-tangler compatible handset
The Directory card can swivel 360o
Note: DTR-4D-1 does not include headset jack or directory and message functions

Specification
Item
Curl Cord Length
Weight (no handset)
LCD Display
Ringing Sound Level
Built in Hands Free
Items Provided with Instrument
Handset Cradle
LCD angle
Housing Face Angle
Recommended Headset
Other

Page 5- 6

Description
12ft
510g (min.)
24 digit x 3 line (alphanumeric and some characters)
no back light
Max. 80dBSPL (in output limit condition) max.
86dBSPL (in normal condition)
Half duplex
Line cord, Directory card
K type compatible
14~42.5 deg. (on the desk, no housing tilt) 25~53.5
deg. (on the desk, housing tilt up)
-4.4 deg. (wall mounting)
14~25 deg. (on the desk)
-4.4 deg. (wall mounting)
NEC Headsets
HAC

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Chapter 5 Station Equipment

Terminal Options
Item
WM-R Unit
AD(A)-R Unit

AP( R)-R Unit

AP(A)-R Unit

CT(A)-R Unit

IP-R Unit

AC-R Unit

Description
Series i Multi-line Terminals with an AP (R)-R, AP (A)-R, CT (A)-R, and/or an IPR Unit can be wall mounted using the WM-R Unit.
Provides Series i Multi-line Terminals ability to interface with recording device.
Provides Series i Multi-line Terminals ability to interface with analog device such
as a cordless telephone, facsimile machine, external speakerphone, Automatic
Dialer or modem. Provides ringing to analog device connected. Requires an ACR ADP.
Provides Series i Multi-line Terminals ability to interface with analog device such
as a cordless telephone, facsimile machine, external speakerphone, Automatic
Dialer or modem. No ringing is provided.
Connects a Series i Multi-line Terminal to a PC providing a complete turnkey
package with graphical telephone user interface and call logging. Shipped with
Multi-line Phone Kits software. Supports Serial interface.
A compact plug-and-play device that installs into the base of a Dterm Series i
display terminal. Integrated two-port 10/100baseT Ethernet Pass through hub
that permits using one port to connect the network interface card (NIC) from the
PC to the IP network. The other is plugged directly into a LAN or an IP network
device such as a router, DSL modem or cable modem. Requires an AC-R ADP.
AC adapter for DSS/BLF Console, AP(A), AP(R), or IP-R Unit

Line Conditions
Cable Length Note 1
Dterm Series i
Dterm 4D
Dterm 8 / 8D
Dterm 16/16D/16LD
Dterm 32/32D

DSS/BLF Console
Note 3
Note 1:

Standard
8DLC

300m (984ft)

with AC Adapter
Note 2

4DLC

300m (984ft)

1200m (3937ft)Note 4

2DLC

850m (2789ft)

1200m (3937ft)Note 4

8DLC

200m (656ft)

Note 2

4DLC

200m (656ft)

1200m (3937ft)

2DLC

850m (2789ft)

1200m (3937ft)

8DLC

300m (984ft)

4DLC

1200m (3937ft)

2DLC

1200m (3937ft)

Note 2:

Cable length is based on the following conditions.
Diameter of the cable is 0.5 mm.
The Protection arrester is not inserted between the terminal and PBX.
When using 8DLC card, long line function is not availability, even if it is equipped with AC Adapter.

Note 3:
Note 4:

The DSS/BLF Console requires local AC/DC supply.
The DTR 4D-1 does not support an AC adapter.

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Chapter 5 Station Equipment

Dterm Series E (Multi-line Digital Terminals)
The Dterm Series E terminals were strategically designed to provide ergonomic form and userfriendly functions. With advanced digital circuitry, the Dterm Series E terminals consist of five
distinct telephone models and one DSS/BLF console to meet users’ diverse terminal needs.
The Dterm Series E family of terminals offers adjustable display and non-display units with
menu-driven soft key operation, allowing users to program terminals at the desktop. The
display units are equipped with large LCD panels with three lines of display, each with 24
characters. Each terminal offers an optional full duplex speaker phone operation for two-way
conversation. Standard features include headset jacks, wall mount units and adjustable base
units. Laser printer designation cards are available in Metallic Green (MG), Black (BK) and Soft
White (WH).
Standard features

HOLD – Press key to place an internal or external call on hold.
TRANSFER – Allows the station user to transfer established calls to another station
without attendant assistance.
ANSWER – When LED on this key is lighted, press key to answer a waiting call.
SPEAKER – Controls the built-in speaker that can be used for Hands Free
Dialing/Monitoring and Voice Call.
REDIAL – Press key to activate redial feature. Press redials and scrolls back through
numbers that have been dialed. When desired number is displayed, press the * key
to activate dialing.
CONF – Press key to establish a three-way conversation.
RECALL – Press key to terminate established call and re-seize internal dial tone.
FEATURE – Used to activate terminal setup functions and to program One-Touch
Speed Dial/Feature Keys.
MIC – Press to turn on and off the microphone.

Dterm Series E Soft keys
The Dterm Series E Display Terminals have four Soft keys located just under the Display of each
Terminal. These menu-driven Soft keys allow user’s convenient access too many features. Each Soft
key may have up to 4 features programmed on it. Soft keys will be functional for specific states of the
terminal such as idle, during dialing, ringing, etc. The state of the terminal will determine what Soft key
is available to the user. According to the status of the Multiline Terminal, functions of the Soft keys are
displayed in the third line on the LCD and if the status of the Multiline Terminal changes, the Soft Key
displayed will change automatically. Also if the HELP key is pressed, an explanation of the indicated
Soft key is shown on the LCD. Typical Soft key Applications are Business, Hotel, Multi-tenant and
Networking applications where it is necessary to save key space on the telephone and also provide a
variety of quickly accessible features.

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Chapter 5 Station Equipment

Descriptions of Series E Multi-line Digital Terminals
Description
DTP-2DT – 1 (WH)

DTP-8 – 1 (WH)
DTP-8 – 1 (BK)

DTP-8D – 1(WH)
DTP-8D – 1(BK)

DTP-16D – 1 (WH)
DTP-16D – 1 (BK)

DTP-32D – 1 (WH)
DTP-32D – 1 (BK)

DTP-32DA-1 (WH)
DTP-32DA-1 (BK)

Remarks
This 2-line non-display digital terminal is equipped with 2 programmable
line/feature keys (each with a two-color LED), 8 dedicated function keys, a builtin speakerphone, and large LED to indicate incoming calls and messages.
The terminal comes in Ivory White (WH) only.
This 8-line non-display digital terminal is fully modular with 8 dedicated function
keys, 8 programmable line/feature keys (each with a two-color LED), built-in
speakerphone, built-in headset jack, built-in long loop adapter, built-in wall mount
unit, tone/volume/contrast control, tilt stand, and a large LED to indicate
incoming calls and messages.
The terminal comes in two colors: Black (BK) and Ivory White (WH).
This 8-line display digital terminal is equipped with 8 dedicated function keys, 8
Programmable line/feature keys (each with a two-color LED), built-in
speakerphone, built-in headset jack, built-in long loop adapter, built-in wall mount
unit, tone/volume/contrast control, tilt stand, 3 line by 24 character LCD display
panel (adjustable/detachable), four soft keys, and a Large LED to indicate
incoming calls and messages.
This terminal comes in two colors: Black (BK) and Ivory White (WH).
This 16-line display digital terminal is equipped with 8 dedicated function keys,
16 Programmable line/feature keys (each with a two-color LED), built-in
speakerphone, built-in headset jack, built-in long loop adapter, built-in wall mount
unit, tone/volume/contrast control, tilt stand, 3 line by 24 character LCD display
panel (adjustable/detachable), four soft keys, and a Large LED to indicate
incoming calls and messages.
This terminal comes in two colors: Black (BK) and Ivory White (WH).
This 32-line digital terminal is equipped with 8 dedicated function keys, 32
Programmable line/feature keys (each with a two-color LED), built-in
speakerphone, built-in headset jack, built-in long loop adapter, built-in wall mount
unit, tone/volume/contrast control, tilt stand, 3 line by 24 character LCD display
panel (adjustable/detachable), four soft keys, and a Large LED to indicate
incoming calls and messages.
This terminal comes in two colors: Black (BK) and Ivory White (WH).
See above for basic description of phone. This phone is to be used as an ACD
terminal. The Silk Screening of the function keys is changed for the dedicated
ACD functions.

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Chapter 5 Station Equipment

Description of Series E Multi-line Digital Terminals (Cont)
Description
DCU-60 –1 (WH)
DCU-60 –1 (BK)

Remarks
This DSS/BLF/Add-On Console is equipped with 60 programmable line keys
(each with a two-color LED). These keys can be programmed as Direct Station
keys, function keys, or outside line keys. This terminal comes in two colors:
Black (BK) and Ivory White (WH).
When the console is used as an Add-On Module, an additional 25 lines/trunks
and the existing line/trunks for the Multiline Terminal can be accessed directly
(maximum 41 lines/trunks).
The station speed dialing function can be assigned for all keys on the Add-On
Module unit. Also, one of the last three keys can be used as a Day/Night change
keys.
When the console is used as a DSS/BLF console, up to 60 stations can be
directly accessed in addition to those already appearing on the Multiline
Terminal. A red LED associated with each button indicates busy status for each
station. In addition, the DSS/BLF console can provide the following functions:
Message Waiting – Set/Cancel/Status Display
Do Not Disturb – Set/Cancel/Status Display
Automatic Wake Up No Answer – Status Display/Cancel
Agent Busy Out – UCD – Status Display
Line Lockout – Status Display
Room Cutoff – Set/Cancel/Status

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Chapter 5 Station Equipment

Dterm Series E Accessories & Adapters
Unit
ADA-U

HFU-U

APR-U

Description
Ancillary Device Adapter (ADA-U)
The Ancillary Device Adapter (ADA-U) allows for a direct connection to a tape
recorder for logging/recording telephone calls. A dedicated set of input
connectors is also provided for a recording tone unit to inform the parties that
the call is being recorded. The ADA-U does not require an AC adapter.
External Hands free Unit (HFU-U)
The External Hands free Unit (HFU-U) provides a solution for small office
teleconferencing by improving the sound quality of speakerphone calls via an
external microphone. This unit is ideal for a working environment where hands
free calling is a necessity. In order to provide maximum performance, two
user-adjustable switches are provided which allow the speakerphone to be
configured for a unique environment (quiet room, noisy business environment,
and room with acoustic echo). Furthermore, a push-to-mute button is featured
on the external microphone to add privacy for hands free calling.
Analog Port Adapter with Ringing (APR-U)
The Analog Port Adapter with Ringing (APR-U) provides an interface for
installing single line telephones (SLTs), modems*, and the NEC
VoicePoint/VoicePoint + conferencing unit. By installing the APR-U, calls may
be initiated via a PC modem and switched to voice operation if desired. Data
calls at a maximum speed of 28.8 kbps are also supported with the APR-U.
The APR-U has the added benefit of detecting incoming ringing signals. By
providing ring detection, the user may wish to install a personal fax machine or
an answering machine for convenience. Finally, two user-adjustable switches
are provided on the adapter: one allows for either a 600
or complex
impedance interface to a (modem, SLT, etc.), also a second voice channel to
permit calls from the Multiline Terminal and devices installed to the APR-U
simultaneously*. The APR-U requires an AC adapter.
Note 1*: When a modem is installed to the Multiline Terminal with the APR-U,
the modem and the Multiline Terminal (handset, speakerphone, and headset)
cannot be used simultaneously.
Note 2*: Dual voice channel capability is available and special programming
may be required from the NEC PBX/KTS systems to support two simultaneous
voice channels.

CTA-U

Computer TAPI ADAPTER Unit (CTA-U)
The Computer TAPI Adapter Unit (CTA-U) permits connecting a Dterm Series
E multilane terminal to a PC. Each CTA-U unit is shipped with Multiline Phone
Kits software, a complete turnkey package offering a graphical telephone user
interface and call logging.

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Chapter 5 Station Equipment

Dterm Series E Accessories & Adapters (Cont)
ACA-U

WMU-U

AC Adapter Unit (ACA-U)
The AC adapter provides the Dterm Series E terminal with the voltage required
to power the built-in long loop adapter, and APR units. One AC adapter can
power up to three optional adapters.
External Wall Mount Unit (WMU-U)
The External Wall mount Unit is ONLY required if an option adapter is required
on the Multiline terminal which is being mounted on a wall. It permits the
terminal to be installed to a flat wall surface. With this unit, any option adapter
may be installed to the terminal for high functionality. The built-in wall mount
unit should be used for wall mount operation when no option adapter (except
ADA-U unit) is installed on the Multiline terminal.
Note: This unit is designed to enhance the hands free operation of Multiline
Terminal by providing an echo canceling circuit. However, this unit is primarily
designed for a typical small office environment and not for conference rooms.
Its performance should not be compared to commercial, expensive audio
conference units. The HFU-U requires an AC adapter.

WMU-W
PC Telephony
Adapter

IPW-2U (ELC)

IPW-2U (P-P)

Page 5- 12

Wall Mount Unit for DTP-2DT-1 terminal
PC Telephony Adapter
The PC Telephony Adapter provides an integration link from a desktop PC to
the Dterm Series E telephone. Using Microsoft® Windows™ Telephony API,
Windows applications users can now control or use many of the features,
desktop message notification, calling line identification, and calling functions of
the Dterm Series E telephone from their PC.
IP Adapter for Dterm Series E. For use with 32 Button Display Terminal and
PN-32IPLA ELC station card only. Requires either ACA-U local power or
SN1604 Power Hub 12-Port
The IPW-2U (P-P) adapter is a full duplex switch, which can be installed on
any of the NEC LCD Dterm Series E phones when using Peer-to-Peer IP
stations. Requires either ACA-U local power or SN1604 Power Hub 12-Port

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Dterm Series E Specifications
Feature/Function
Housing color
Two colors LED
Line/Feature keys
One touch keys
Fixed feature keys
Number of characters
per line on display
Number of lines on
display
Volume Control
Speakerphone
Headset
Ring Volume
LCD Contrast
Soft keys
Off hook ring on/off key
Mute on/off key
Microphone on/off key
Headset on/off key
Built-in speakerphone
Built-in headset jack
Built-in wall mount unit
AC adapter required
Data Adapter
Message Waiting Lamp
Add-on console
supported
ADA supported
APR supported
CTA supported
PC Telephony Adapter
supported
HFU supported
HFU operation:
Half Duplex
Full Duplex
Adjustable legs
Long loop supported
Hearing Aid compatible
Cable pairs

Dterm Series E Digital Terminals (DTP-)
2DT
8
8D
16D
32D
White
White or Black
Yes
Yes
Yes
Yes
Yes
2
8
8
16
16
None
None
None
None
16
8
8
8
8
8

DCU-60
Add-On
DSS
No
48 ext.
12
None

No
60
None
None

N/A

Yes
No
Yes
N/A
N/A
No
No
No
No
Yes
No
Optional
Optional
No
Yes

Yes
Yes
Yes
N/A
N/A
No
No
No
No
Yes
Yes
Yes
Optional
No
Yes

Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Optional
No
Yes

N/A

N/A
N/A
N/A
N/A
N/A
N/A
N/A
Yes
Yes
N/A
N/A

Yes

N/A

No
No
No

Yes
Yes
Yes

N/A
N/A
N/A

Yes

N/A

Yes

N/A

Yes
No
No
Yes
Yes
1

Yes
Yes
Yes
Yes
Yes
1

N/A

Yes
N/A
N/A
1 port

Yes
N/A
N/A
2 ports

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Dterm Series E Line Conditions

Line Card

4DLC

8DLC

2DLCB/N

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Loop Resistance and Distance (Long Loop circuit built-in)
Cable Length
Maximum
Terminal
(24 AWG Cable)
Loop Resistance ( )
(Without ACA Unit)
DTP-8-1 TEL
35
1000 ft. (304 m)
DTP-8D-1 TEL
35
1000 ft. (304 m)
DTP-16D-1 TEL
26
650 ft. (198 m)
DTP-32D-1 TEL
21
650 ft. (198 m)
DCU-60-1 Console
N/A
N/A
DTP-8-1 TEL
35
1000 ft. (304 m)
DTP-8D-1 TEL
35
1000 ft. (304 m)
DTP-16D-1 TEL
26
650 ft. (198 m)
DTP-32D-1 TEL
21
650 ft. (198 m)
DCU-60-1 Console
N/A
N/A
DTP-8-1 TEL
35
2800 ft. (853 m)
DTP-8D-1 TEL
35
2800 ft. (853 m)
DTP-16D-1 TEL
26
2800 ft. (853 m)
DTP-32D-1 TEL
21
2800 ft. (853 m)
DCU-60-1 Console
N/A
N/A

Cable Length
(24 AWG Cable)
(With ACA Unit)
4000 ft. (1219 m)
4000 ft. (1219 m)
4000 ft. (1219 m)
4000 ft. (1219 m)
1000 ft. (304 m)
1000 ft. (304 m)
1000 ft. (304 m)
1000 ft. (304 m)
1000 ft. (304 m)
1000 ft. (304 m)
4000 ft. (1219 m)
4000 ft. (1219 m)
4000 ft. (1219 m)
4000 ft. (1219 m)
2800 ft. (853 m)

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Dterm Cordless Terminals
Dterm Cordless II (DTR-4R-2 BK)
The Dterm Cordless II is fully digital and operates in the 900MHz frequency range. It includes
Base Unit, Handset (with a standard battery, 6 hours of Talk Time or 4 days of Standby),
Charging Stand, Belt Clip, Wall Plate Adapter and 2 AC Adapters.
Specifications:

Digital
Up to 350 feet
900 MHz digital spread spectrum
4 programmable line keys
10 Simultaneous channel access
Headset connection
Volume slide switch/6 ring tones
Weight: 8.8 ounces w/battery
Options:
Unit
EXP 9530
EXP 9785
EXP 9702
AD 970
EXP 9704
Leather Carry Case
Wall Mount
BT 2499

Description
Headset
Lightweight headset with foam covered ear and microphone
Mounts over the head
REPLACEMENT CHARGER UNIT w/o AC Adapter
The charging cradle is used to charge the batteries used with the Dterm
Cordless II Terminal
REPLACEMENT SPRING-TYPE BELT CLIP
Use to clip the handset to clothes or belt
REPLACEMENT AC ADAPTER FOR CHARGING UNIT
Provides 9Vdc to the Dterm Cordless II Charging Unit
Requires 120V, 60Hz (14W) AC input from standard AC plug
REPLACEMENT AC ADAPTER FOR BASE UNIT
Provides 10Vdc to the Dterm Cordless II Terminal Unit
Requires 120V, 60Hz (14W) AC input from standard AC plug
LEATHER CARRY CASE
Leather case for DTR-4R-1 (BK) Handset
Replacement Wall Mount Adapter
Used to mount the Dterm Cordless II to a standard wall plate
REPLACEMENT BATTERY
Dterm Cordless II - 6 hours of Talk Time or 4 days of Standby Capacity

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Dterm Cordless Lite II (DTH-4R-1 BK)
The Dterm Cordless Lite II is digital and operates in the 900 MHz narrow band frequency range. It
includes Base Unit, Handset, one telephone cord, 2 AC Adapters, 2 Rechargeable Batteries, Spring
Belt Clip, Battery Charger, and Wall Mounting Plate.

Specifications
Frequency
Security
Voice Channel
Interface Port
Multi Line Capability (Hand Set)
LCD Screen
System Caller ID
Remote Base Unit
2nd Battery Charging system
Auto Talk
Auto Standby
Talk Time
Standby Time
Silent Alert System
Handset Size
Ring On-Off Key
Battery Type
Selectable Ring Tones
Message Waiting Indicator
Belt Clip
Headset Jack
Transfer
Conference
Hold
Mute
Backlit Keypad
Hot Swap Battery

900Mhz FM with ADPCM
Digital
30 Channel
Digital/Analog
4 Line Capability
2x16 Digit (10 Digit)
YES
YES
NO
YES
YES
6 Hours
96 Hours
YES
5½
YES
NiMH Battery (3.6V-700mAh)
3 Tone
YES
YES
Yes-2.5mmö
YES
YES
YES
YES
4 LINE KEY
YES

Options
Unit
EXP9660
AD9500
AD960
BT9000B
EXP9603
EXP9683
EXP9685
EXP9530
EARSET

Page 5- 16

Description
Replacement Wall Mount Unit Use to wall mounts the Dterm Cordless Lite Terminal.
Replacement AC Adapter for Charging Base Unit.
Replacement AC Adapter for Charging Unit. Provides 9Vdc to the Dterm Cordless
Lite Charging Unit. Requires 120V, 60Hz (14W) AC input from standard AC plug.
Replacement Battery 5 hours of Talk Time or 40 hours of Standby Capacity,
500mAH 4.0V
Replacement Spring-Type Belt Clip. Use to clip the handset to cloths or belt.
Leather Carry Case.
Replacement Charging Cradle and AC Adapter.
Dterm Headset. Lightweight headset with foam covered earpiece and microphone.
Dterm EARSET. Lightweight ear set with microphone.

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Plantronics CT Cordless
Package includes Base Unit, Handset (with a standard battery 6 hours of Talk Time or 3½ days of
Standby), Belt Clip, M175 Headset, User Guide and AC Adapter. This compact unit is an Analog single
line device that operates in the 2.4 GHz Digital Spread Spectrum frequency range. This unit is
compatible with and Analog single line port, Analog Port Adapter or Analog telephone line from Telco.
Specifications
2.4 GHz Cordless Headset Phone
Range of Up To 150 Feet
Audible Low Battery Indicator
Single Line Operation
Ultra-Compact Remote Unit with Belt Clip
Variable Range Volume Control
10 Number Speed Dial
Page / Find Feature
Redial / Flash
Mute with Audible Reminder
Talk / Charge / Power Indicator Lights
Built In Headset Stand

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INASET
INASET terminals are members of the Dterm IP family. INASET terminals have a Web browser with a
large color display and a built-in multi-port Ethernet switch for connectivity to the user’s local PC.
INASET terminals bring a wealth of information to the desktop, including short text display messages
and Web pages specifically tailored for the small screen format
The INASET’s basic load includes a graphical telephony application that provides an abundance of
telephony information and desktop control that is easy to use with its menu-based interface. Information
and controls accessible via the soft keys and feature buttons include:
Line status showing a visual icon display for the status of all assigned phone lines and DSS
lines.
Caller information showing a visual text display for things such as time, date and call status
information.
Telephony Directory for storing, searching and dialing different profiles which you can
categorize and store in one of three different groups: corporate, personal and group.
Web access providing browsing capabilities to display HTML web-based information located on
the Internet or Intranet. Also includes support for Java applets.
Virtual keys providing access to features, functions and recent keys activated on the terminal.
The user can program display and functions for how they see fit with limited or no administrative
support necessary.
By converging a company’s voice and data networks, there is only one network to manage. The
INASET includes a built-in switch, so you can use a single Ethernet switch port for the computer (data)
and the INASET. Because it’s an IP based telephone, it can be installed anywhere on a corporate IP
network. The INASET is Dynamic Host Configuration Protocol (DHCP) compatible and doesn’t require
the need to be co-located with the NEAX PBX equipment. INASET supports G.711, G.729a and
G.723.1 audio compression for low-bandwidth requirements.

INASET Applications
The INASET is specifically suited for the enterprise environment, including the following end users:
managers, purchasing agents, consultants and call agents. But this is just the beginning. This
advanced IP business terminal can be programmed to do much more than a standard business phone.
For example, the customizable user interface (that’s developed with the SDK) can replace single line
phones that are typically installed in cafeterias, break rooms, lobbies and manufacturing floors. The
INASET’s programmability is also ideal for traders, stockbrokers, real estate brokers, executives, PR
professionals and any other occupation in which professionals use Internet-related information to
interact with the public.

Additional Applications
Personalization at the desktop: Individual customization can bring flexibility to the desktop.
Software developers can enhance traditional features such as Answer, Redial, Conference,
Recall and Help menus.
Centralized services: Information or features that need to be accessed by numerous
individuals within the organization can be centralized on servers. Directory service functions,
Intranet web site information, customer records, CTI applications and other work group

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applications are examples of centralized services that could improve portability and use
throughout the enterprise network.

INASET Terminal Line UP.
The INASET terminals are available in three models. Each is described below.
Description
ITR-240G-1 (BK)

ITR-320C-1 (BK)

ITR-320G-1 (BK)

Remarks
The INASET 240G has a 3” LCD with a gray scale 240 x 160 pixel display and
16 programmable keys.
Supports Power over Ethernet (POE) (802.3af and Cisco Discovery Protocol).
Available in black only.
The INASET 320C has a 5.1” color display and 32 programmable keys.
Supports Power over Ethernet (POE) (802.3af and Cisco Discovery Protocol).
Available in black only.
The INASET 320G has a 5.1” LCD with gray scale display and 32 programmable
keys. Supports Power over Ethernet (POE) (802.3af and Cisco Discovery
Protocol). Available in black only

INASET Accessories
Description
WM-RL UNIT
INASET Handset Cord 12’
INASET Handset Cord 25’
AC-R
ILPA-R
ADA-2R
MIC-R (BK/WH) UNIT
PSA-R

Remarks
Wall mount for the ITR-LC-1
12 Foot Black Handset Cord
25 Foot Black Handset Cord
AC adapter for the INASET
802.3af power Dongle for INASET Original
Audio recording adapter for IP terminals
External Microphone for INASET 240, 320s
Analog survivable adapter

INASET Basic Network Capability
Dual port 10/100 Mbps Ethernet switch
Internal voice packet prioritization
VLAN support (802.1q & 802.1p frame tagging)
Configurable ToS bits (DiffServ and IP Precedence support)
Powering
o Inline power (unused pairs) or
o Direct powering with external wall-plug adapter

PBX System Requirements
NEAX 2000 IPS R6.1or higher
IP-PAD Card
8 Seat Licenses: Supports up to 8 INASET terminals

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Dterm PS III Wireless Handset
The Dterm PSIII is a wireless handset designed to support the professional’s schedule and the special
needs of people on the run. At less than 3.8oz and with 300 hours of standby time, the Personal
Station continues to offer the size, weight and battery life demanded by the user constantly on the go.
The phone uses digital technology with audio quality approaching that of a digital desk phone. The new
DtermPSIII wireless handset moves freely throughout the building with seamless handover when
moving from one zone transceiver to another. Note
Dterm PS III Features
•
•
•
•

Separate Call Hold & Transfer keys
2-Line Operation
Enhanced Short Text Message (STM)
Reminder tones for unread STM

•
•
•
•

Name Display
Directory Dial
Speed Dial
4 Line 18 digit LCD Display

•
•
•

Last Number Redial
Automatic Answer
Vibration Alert

Dterm PS III OPTIONS
Standard Charger for the Dterm PSIII handset. Single AC adapter is included.
There are three cases available for Dterm PSIII:
Two types of Leather Case provide a Glove like fit for ease of handling and are designed for
indoor use. One includes a swivel belt clip with quick release and the other has a built in belt
clip.
The Weather Case model offers a weather resistant material for users whose job is outdoors a
majority of time.
There are two available headset options:
The Headband type is a classic banded headset with a microphone extension.
The Single Ear Cover type is an ear bud type with an in-line microphone.
Note: For seamless handover ZT must be ver7.2 or higher

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Dterm SP30 SoftPhone
The Dterm® SP30 allows customers to capitalize on the advantages of a converged voice and data
network whether they’re in the office or on the road. The Dterm SP30 combines traditional business
communication needs with the data applications your customers require.
The Dterm SP30 optimally delivers high quality voice via a USB-connected headset. The Dterm SP30
has the ability to use a wireless handset (PS) or Dterm for all voice connections instead of the USB
handset. With a simple drag and drop, the Dterm SP30 allows telephone dialing from other telephone
directory applications such as Microsoft Outlook®, HTML pages and Word® documents, etc. In addition,
the Dterm SP30 provides an interface to Microsoft’s Telephony Application Programming Interface
(TAPI) via NEC OpenWorX integration, allowing TAPI-enabled applications, such as Outlook and ACT,
to make and receive calls. The Dterm SP30 can be displayed in 1 of 4 different colors (black, red, gold
and neon blue) in order to reflect the “personality” of the day. The Dterm SP30 also allows for 3
different modes of operation.
Maximized mode: Access to full line of softphone features such as application sharing, member
lists, conference mode, chatting capabilities, Internet access and many others are just one click
away.
Compact Mode: L-shaped user interface, operating in a small footprint on the PC screen.
Compact view allows the softphone to remain active while another application window such as a
Word document; database file or email is the primary focus on the PC. With the compact view, the
most popular features of the converged softphone are just a click away.
Task Mode: The softphone can be minimized and shown as a task within a Microsoft Operating
System. While operating in this mode, the softphone will output an audio notification to the user
upon receiving an incoming call. It will be up to the user to utilize the hot key in order to activate the
Dterm SP30 application and answer the call.
ACD Mode: The Dterm SP30 ACD client can support two operation modes, ACD and Business.
The Business mode is the default mode. When the ACD mode is configured a new set of function
keys becomes available to the Dterm SP30 ACD user. A popup menu is also available when in the
ACD terminal mode. While in ACD mode the Instant Messaging feature functions to the MY Line of
the terminal, not the ACD Line.

Key Features and Benefits
The Dterm SP30 offers a full complement of station and converged features for an important segment
of mobility and PC-based applications. Even though a majority of the hardware telephone features
function on the softphone, the main focus for the softphone is its ability to deliver PC capabilities to the
telephone.
Features:

Displays call records in Quick Reference List via the Call Log Window:
All calls are recorded in a Call Log. Telephone number, date of the call and identification as to
whether or not the call was received and answered are all logged. You can find at a glance when
and with whom you talked. The Dterm SP30 also offers a call memo function whereby you can
record the key points of the call so that you can visually identify the individual records.
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Links with PC applications:
Now you can collaborate on a white board application (Word® or Excel® document or any other
application) that is operating on your PC and the PC at a distant site. Simply place a call to the farend Dterm SP30 user and activate the application you will be collaborating on. You are now able to
have a more productive conversation.
Internet Link:
The Dterm SP30 can be assigned with a common database link for fast access to a particular site.
This link could be an Internet link that needs to be accessed when receiving or placing a call (i.e.,
you receive a call from a customer and need to look up information that is contained in a database).
Previously, you needed to locate and launch the application before loading the file. Now with the
Dterm SP30, simply select the Internet link and the database file is opened, bringing it all together
in one user interface.
Real Time Communication:
In addition to providing a voice and data collaboration link, you can also chat with a remote Dterm
SP30. This is ideal for the real quick conversation you need with a distant Dterm SP30 user. Maybe
you’re on a call and need to get a quick response from a co-worker. There’s no need to put the
customer on hold and call the co-worker for an answer. Instead, send a chat message and get your
answer in real time. The Dterm SP30 will store all your chat messages in a log so that you can use
them for future reference. Additional features of the Dterm SP30: forwarding control selection for
different call modes, videoconferencing capabilities, automatic downloading of telephony features to
the display and diagnostic capacities for audio problem notification.
Presence/Status
The presence/status functionality allows the user to confirm a buddy’s presence/status with a visual
indication (ICON) and text message on the Dterm SP30 GUI. The presence/status information is
provided by the OpenWorX servers (LSI) package.
8/6 Party Conference Control
When the trunk conferencing card is configured within the voice switch and the voice trunk channel
is configured in the program utility for the Dterm SP30, a user can dynamically setup and save
future dated collaboration conferences. All the conferences can be activated immediately upon
configuration or saved for a future date upon activation. When the conference is activated, the
Dterm SP30, with the help of the Voice sever, places calls to the configured participants based on
the number which are user predefined. Conferences do not require users to be Dterm SP30 users
for voice only conference calls. Only Dterm SP30 users will utilize additional features like
application sharing and messaging during call.
Collaboration
Collaboration of a Dterm or PS handset links capabilities to the Dterm SP30 GUI. In this mode, all
the voice connections will be directed to a Dterm or across the PS infrastructure instead of through
Dterm SP30 USB handset. The Dterm SP30 GUI will give the indication that it has received a call
such as caller ID, ringing information and notification as to the state of the line. The only thing that
changes is voice path being directed to the Dterm or PS handset. The Dterm SP30 will provide data
collaboration interface.
Collaboration Terminal list:
ITN Terminal (NEC SIP)
Dterm
Dterm IP
Analog Terminal
PCS/PHS

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Benefits of this functionality:

User has a reliable voice connection with the voice server.
Dterm SP30 user has a terminal device which can be ported from place to place
without the need for programming call forward functionality on the phone.
The user can be reached by one number, no matter if call forwarding is set or
the user is located next to the Dterm SP30.
User has a physical device which they can use for dialing and answering calls
instead of working with mouse and keyboards.
User is no longer locked to their PC for receiving and answering calls. The user
is as portable as the in building wireless network when using the PS.
Voice Recording
A user will be able record the voice connection and save the wave file on a network storage place
of their choice. It is recommended all voice recording be stored on a local hard drive and not a
network drive. With the activation of the voice recording of the Dterm SP30, no extra recording
equipment other than the Dterm SP30 application with a USB headset is required. For those
specific regions which require the notification to the remote party that a voice connection is being
recorded, the Dterm SP30 provides for the setting of an automatic beep tone in the configuration
menus. Beep tone can also be conditioned to send a tone notification at user selectable intervals.
Application Collaboration
The NEC Dterm SP30 allows users to share ideas, information and programs in a variety of ways
while either in a point-to-point connection or 6/8 party conference mode.
Videoconferencing
The Dterm SP30 audio and videoconferencing feature lets you communicate with anyone on the NEC
Network.
Share ideas, information, and applications using video and audio
Send and receive real-time images using Windows-compatible equipment
Allows for broadcasting of the live video to other Dterm SP30 users which might not have video
transmission capabilities
Use of a video camera to instantly view items, such as hardware devices, road conditions or even
personnel, which are displayed in front of the camera
Whiteboard
The whiteboard lets you collaborate in real time with other Dterm SP30 users via graphic design. With
the whiteboard, you can review, create and update graphic information.
Manipulate contents by clinking, dragging and dropping information on the white board with a
mouse/keyboard.
Copy, cut and paste information from any Windows-based application into the whiteboard.
Use different-colored pointers to easily differentiate participant’s comments.
Save the whiteboard contents either at the local side or distant end location
Load saved whiteboard pages, enabling you to prepare information before a conference, then
drag and drop it into the whiteboard during an audio meeting
Chat
The chat functionality lets you conduct real-time conversations via text with as many Dterm SP30 users
as you like. With chat, you can:
Type text messages to communicate with other co-workers during a conference
All messages are sent in a whisper mode so that they are only received by one party
All messages sent and received are saved automatically in the chat log
Automatic pop up notification when a chat message is received
ICON notification within the chat log identifying different states of the messages

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File Transfer
File transfer lets you send one or more files to distant Dterm SP30 users. With file transfer, you can:
Send a file to other Dterm SP30 users
Accept or reject transferred files
Application Sharing:
Dterm SP30 gives you better control over how shared programs are displayed on your desktop and
give the person sharing the program control over who uses it.
View shared programs in a frame, which makes it easy to distinguish between shared and local
applications on your desktop
Minimize the shared program frame and do other work if you don not need to work in the current
conference program.
Easily switch between shared programs using the shared programs taskbar.
Approve conference participants’ requests to work in the program you introduce.
Allow or prevent others from working in a program using the sharing dialog menu.
Application Collaboration Modes Supported
Applications

2 Party Mode

3 Party without UM4730

3 Party with UM4730

Video Conference

Yes

White Board

Yes

Application Sharing

Yes

File Transfer

Yes

Chat

Yes

One Key Operation
The space bar key on the keyboard for the Dterm SP30 PC can be configured in such a way to
allow fast access to placing calls, receiving calls and termination of calls. You are no longer
required to utilize a combination of the keyboard and mouse to access these telephone features.
Call Log Export
The Dterm SP30 provides for the capability of saving the call log file in a “.cvs” file format in addition
to the administrator/user selecting a network storage place for the file. When opening the file with
application render capable of displaying “*.cvs” files, columns breaking out caller direction, date and
time, caller ID, duration of the call, notification of whether the call was recorded and memo
information are displayed.
Voice Quality Alarm
The Dterm SP30 is able to provide notification to the user when it is perceived to be transmitting
and/or receiving voice problems. When the Dterm SP30 application detects a voice problem an
alarm is lit on the GUI and information about the audio problem is stored in the maintenance log file.
2 Line Display
When the Dterm SP30 is set for a compact mode, the display which is shown is only capable of
displaying one line of information. When configuring the Dterm SP30 to a 2 line display, the display
is able to switch (1 second intervals) between the normal 2 lines of display.
Automatic Idle Return
This function allows for automatic disconnection during ROT or BT tone being heard due to
disconnecting by remote party. Administrator/user can choose to disable or enable this function
within the user configuration. The default for this configuration is disabled.

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PB tone sending
During a call, a PB tone of the dial number which is set on the member button can be sent by
selecting the [send PB tone] with a right-mouse click of speed dial number.
Pause into Dialing string
This feature allows the user to insert a pause, using a comma, into the dialed number during the
following conditions:
Drag/drop into the LCD or telephone ICON.
Copy/Paste
Placing/transferring a call with “Call to:” tag
Placing/transferring a call and sending the PB tone with using the member button.
Last Number Redial

A Dterm SP30 user can do a quick access and call to the most recent call placed by the SP30.
During the idle stage a user can redial the telephone numbers previously dialed by selecting the
[Redial] in pop up menu which appears by right-clicking on a “Call” button.
Redial via Call Log

The Dterm SP30 is capable of displaying the last 32 calling/called telephone numbers. This is done
by depressing the Up/Down arrow key (Keyboard) when the LCD field is highlighted.
First number displayed on LCD screen will be the most recent called/received telephone number.
Pressing the arrow up will cycle from the most recent to the latest calls/received telephone number.
Pressing the arrow down button will cycle from the last to most recent calls received/placed.

One Touch Button in Compact Mode
A button called the “One Touch” button has been added to the compact panel. The “One Touch”
button allows access quick access to your speed dial list (Buddy List) that is conditioned on the
main GUI under the Member buttons. By clicking the “One Touch” button, you can place a call or
transfer a call to any of the member’s numbers that has pre-registered on the main GUI.
Notification of Call Received with No Answer
The Dterm SP30 Call Log button which is located on the main and compact panel will flash in blue
when a call was receive by the Dterm SP30 but not answered. Before the integration of the blinking
call Log button the only way a user was noted that a call was received was if the remote party left a
voice mail. In those case where there was no voicemail message was left, the user had no way of
identifying whether calls were received or not. Users can confirm visual that a call was received and
immediately reply to the caller when they become available.

Client PC specification:
PC
OS
Memory
CPU
Sound Device
Hard Drive
LAN Interface
USB Port
Video Camera

Operation Environment
IBM-PC/AT Compatible machine
Windows XP/2000 w/NetMeeting application and DirectX
8.1software components installed
384Mb or more
Pentium III 900Mhz or more
Direct Sound Application version 8.1 or higher
Full duplex capability."
50Mb or more of empty capacity
10/100M Ethernet
2.0 support for USB headset for audio input/output
Optional

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Chapter 5 Station Equipment

PBX Specification (Must be peer to peer IP enabled)
Operation Environment
NEAX 2000 IPS R8 Series 3300 or higher
SP30-4 software license
SP30-4 ACD software license R10 Series 3500 or higher

Dterm® SP30 Softphone Parts List
Software/Licenses/Accessories
SP30-4 software license
Package of 4 right to use IP softphone clients.
SP30-4 ACD software License Package of 4 right to use IP softphone w/ACD function
8 IP Seat License
Enables 8 IP phones and is required in addition to SP30
software License for SP30/SP30-ACD functionality
Dterm SP30 Software CD
Application CD. One CD can be used for multiple Client
installations. Recommend 1 per 4 clients.
NEC USB Headset (DSP300)
One per SP30
NEC USB Headset (DSP400)
One per SP30

Dterm Extenders
The NEC Dterm Extenders extends the operation of the UNIVERGE
telephone to a remote location.

NEAX

2000 IPS digital

Dterm EXT (Analog)
The Dterm Analog EXT enables telephone users to be a fully functional part of the NEC PBX with a
Dterm Series E (DTP-32DE-1) telephone located any distance off-premises. The Dterm Analog EXT
consists of two modules. One module, identified as the Office Module, connects to a digital Dterm
Series E station port on your PBX. The other module, identified as the Remote Module, connects to
your NEC Dterm Series E (DTP-32DE-1) telephone at your off-premises location. The modules
communicate via a single analog telephone line.
The Dterm Analog EXT is transparent to the user and retains access to the features and functions of
the PBX. In addition, an RS-232 data port is incorporated, allowing the user to connect off-premises
RS-232 equipment to equipment at the Office Module location. The Office Module emulates your
telephone, and the Remote Module emulates your NEC PBX. Each module uses a V.90 internal
modem (operating in V.34 mode) for the transmission of all signals between the two modules. Network
configurations that support V.34 modems should work well.
Through the use of soft keys, the remote user signs on with a secured password. When the extender is
online, a voice/data connection is made through public facilities via an Analog line from the remote
location to the office PBX system. The Extender offers the flexibility for the user to operate in
Continuous or Call-On-Demand modes. The continuous mode maintains the connection until the
remote user signs off. The Call-On-Demand mode helps in avoiding continuous connection charges by
only staying connected when a call is in progress.

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Chapter 5 Station Equipment

Service Conditions for Dterm Analog EXT
911 calls should not be made via remote terminals since the telephone number at the PBX
location is sent to the emergency center. The user should use a separate line that is not
connected to the Dterm EXT system.
If a dedicated line is used, it should be an unlisted number in order to minimize busy signals.
No custom calling features, such as Call Waiting or Call Forwarding, should be ordered for the
line to which you connect the Remote Module.
Note: A Call waiting tone causes an interruption in data transfer between the Office and Remote
Module.

When you use voice and data simultaneously, the voice transmission will have priority over the
data transmission regarding available bandwidth allocation.
Up to 100 passwords can be programmed into each Office Module.
Only one Remote Module at a time can be “online” with an individual Office Module.
If passwords have been enabled on the Office Module, authorized connections require a
password, 8 to 10 digits in length.
Passwords are retained in non-volatile memory and are not affected by power outages.
Each password begins with two digits (00-99). These digits uniquely identify up to 100 users.
The 00 password is reserved for the System Administrator and can be used to change any of
the user’s passwords.
Zone paging and voice paging must be turned off via PBX programming on the ports connected
to the Dterm Analog EXT. The remote telephone should be programmed to ring for all incoming
calls.
If the remote Multiline Terminal is programmed to answer multiple calls, these calls should ring
at the terminal. Configure secondary incoming extensions for ringing.
Analog fax machines and modems are not operable with the DTP-32DE-1 Multiline Terminal
when connected to the APR-U optional adapter.
The NEC Dterm Cordless and Dterm Cordless Lite telephones cannot be used in conjunction
with the DTP-32DE-1 Multiline Terminal and Dterm Analog EXT.
The NEC CTA-U optional adapter cannot be installed to a NEC PBX remote Multiline Terminal
to provide TAPI functionally with the PBX.
The auto answer feature for PBX ACD is not operable while in Call on Demand mode.
The Dterm PC and Dterm PC II are not functional with the Dterm Analog EXT.
The HFU-U hands free adapter is not operable with the DTP-32DE-1 Multiline Terminal for
Dterm Analog EXT operation when used in a home environment due to FCC restrictions.
However, it is permitted for use in an office environment.

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Chapter 6 Trunking
Type of Trunks
The following list reflects the type of trunk and the package that supports it.
LT/AP
Slots
Ports
Central Office, Foreign Exchange,
PN-4COTB
4
1
WATS (Analog)
PN-8COTS
8
1
PN-4COTG
4
1
Central Office Caller ID (Class SM)
PN-8COTQ
8
1
PN-4DIDA
4
1
Direct in Dial (DID)
PN-AUCA
2
1
PN-2ODTA
2
1
Two wire E & M tie trunks
PN-4ODTA
4
1
PN-2ODTA
2
1
Four wire E & M tie trunks
PN-4ODTA
4
1
T-1 Spans
SPN-24DTAC
24
1
E-1 Spans
SPN-30DTC
30
1
SPN-24CCTA
25
1
Digital CCIS
SPN-24DTAC & SC00
25
2
SPN-30DTC & SC00
31
2
PN-2ODTA & SC00
2
2
Analog CCIS
PN-4ODTA & SC00
4
2
IP CCIS (Peer to Peer)
Virtual PIM
1-128
0
SPN-IPTB-A
4
1
IP CCIS (non Peer to Peer)
SPN-4VCTI-A
4
1-6
SPN-24PRTA
25
1
ISDN (PRI)
SPN-24DTAC & SC01
25
2
SPN-30DTC & SC01
31
2
SPN-BRTC
1
1
BRI Trunk
SPN-4BRTA
4
1
PN-4COTB
4
1
Paging
PN-8COTS
8
1
PN-CFTB
10
1
Conference
SPN-CFTC
32
1
PN-2DATA
2
1
Announcement
PN-4DATC
4
1
SPN-IPTB-B
4
1
H.323 Trunk
SPN-4VCTI-B
4
1-6
(DTMF) PN-8RSTG
8
1
(MF) SPN-4RSTBA
4
1
Register/Sender
(Caller ID) SPN-4RSTC-A
4
1
(E911) SPN-4RSTBA-911
4
1
Note: Any combination of the above cards is allowed with a limit of 256 trunks per system.
Type of Trunk

Circuit Package

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Chapter 6 Trunking

Trunk Parameters
Trunk Impedance 600/900 ohms (selectable by programming)
Trunk Resistance Characteristics, Central Office Trunk (1200 maximum)
Trunk Set Up Procedure
o Loop Start
o Ground start
o Wink Start
o Ring Down
o E & M Signaling
Release and Incoming Detect Immediate Start and Timing Variable Start

Trunk Card Specifications
PN-4COTB
This card provides 4 Loop/Ground Start trunks with disconnect supervision, mounts in any PIM
(0 to 7) and supports the following interface mixed in any combination:
Interface for standard Analog trunks/Centrex type lines
Interface card for CO ring down trunks
Interface with 4 paging equipment lines
Interface with external music/announcement source for Music-On-Hold feature

PN-8COTS
This card provides 8 Loop/Ground Start trunks with disconnect supervision and mounts in any PIM (0 to
7) and supports the following interface:
Interface with 8 CO lines
Interface with 8 paging equipment lines
Interface with external music/announcement source for Music-On-Hold feature.

The above interfaces can be mixed on to one card in any combination.
PN-4COTG
This card is used to provide the system with Caller ID (Class SM). This feature requires the SPN4RSTC card to receive the FSK signals for Caller ID. This 4-port card mounts in any PIM (0 to 7). This
card supports Loop Start trunks.

PN-8COTQ
This card is used to provide the system with Caller ID (Class SM). This feature requires the SPN4RSTC card to receive the FSK signals for Caller ID. This 8-port card mounts in any PIM (0 to 7). This
card supports Loop Start trunks.

PN-4DIDA
This circuit card is a DID line trunk card which supports the functions of four DID line trunks. This card
provides the following functions:
This card is equipped with Direct Inward Dialing (DID) function.
This card is equipped with loop detection functions, sending reverse signal, and DTMF to DP
conversion.
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Chapter 6 Trunking

PN-AUCA
This circuit card is used to either control two single line telephones for off-premise extensions or for 2line Direct Inward Dialing trunks. Also, this card will connect to single telephones for Power Failure
Transfer.

PN-2ODTA
The 2ODTA card is a two or four wire tie line card that mounts in any LT slot in PIM 0 to 7. This card
provides analog tie line service for standard PBX networking or CCIS.

SPN-24DTAC
This circuit card is a 24 channel 1.5 Mbps digital interfacing trunk card which also supports the interface
between ISDN lines and CCIS lines with 1.5 Mbps AMI. The following circuit cards may be used in
conjunction with the 24DTA:
SPN-SC01 - This circuit card controls the signal link (D channel) when Integrated Services
Digital Network (ISDN) PRI (Primary Rate Interface - 23B+1D) is employed in the system.
This package will support one signaling link on a single DTA package. A maximum of 8 ISDN
spans are allowed per system.
SPN-SC00 - This circuit card controls signal links when Common Channel Interoffice
Signaling (CCIS) is employed in the system. This package will support one signaling link on a
single DTA package. A maximum of 8 CCIS links per system is allowed

SPN-24PRTA
This circuit card is a 24 channel 1.5 Mbps digital interfacing trunk card that also provides a built-in DChannel Handler (DCH). This package provides one DTI span with ISDN Data Link on a single
package. A maximum of 8 ISDN links per system is allowed per system.

SPN-24CCTA
This circuit card is a 24 channel 1.5 Mbps digital interfacing trunk card that also provides a built-in CChannel Handler (CCH) for CCIS. This package provides one DTI span with CCIS Data Link on a
single. A maximum of 8 CCIS links per system is allowed per system.

SPN-30DTC
This circuit card is a 30 B-channel, 2 Mbps E1 digital interfacing trunk card (2.048 MHz). Typical E1
applications apply to any CCIS, Q-SIG, and TIE Line application via continuous property. Complies with
U-LAW PCM voice coding standard used in North America.
Note: Does not support A-LAW coding used in Europe, Mexico and South America.

The following circuit cards may be used in conjunction with the 30DTC:
SPN-SC01 - This circuit card controls the signal link (D channel) when Integrated Services
Digital Network (ISDN) PRI (Primary Rate Interface - 30B+1D) is employed in the system.
This package will support one signaling link on a single DTA package. A maximum of 4 ISDN
spans are allowed per system.
SPN-SC00 - This circuit card controls signal links when Common Channel Interoffice
Signaling (CCIS) is employed in the system. This package will support one signaling link on a
single DTA package. A maximum of 3 CCIS links per system is allowed.

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Chapter 6 Trunking

SPN-BRTC
This card has one circuit of Basic Rate interface (2B+D) and provides connectivity for Voice, High
Speed Data, Switched Video and G4 Fax calls to be placed and received via ISDN Basic Rate (BRI) to
the ISDN Network. The BRTC card has built-in DCH (D channel handler) and has a ST interface. A
locally provided NT1 device is required to connect to the ISDN Network. One BRI trunk provides two
incoming/outgoing voice connections and (or) two 64k bps channels for high-speed data. This card is
used for BRI trunks from the Telco to the PBX.

PN-4BRTA-F
This card has four circuits of Basic Rate interface (2B+D) and provides two 2-channel PCM digital lines.
This card is used for BRI trunks from the Telco to the PBX.

PN-CFTB
The CFTB card is a 6 Party Conference card. One card can control a conference of up to ten
participants. This card occupies 10 time slots per one card.

PN-CFTC
The CFTC card is a 32 Party Conference Trunk card. One card can control a conference of up to 32
participants. Two Conference methods are available: Group Call and Meet-Me Conference. This card
occupies 32 AP time slots per one card.

PN-2DATA
This circuit card is a trunk card that supports a maximum of two announcements without external
announcement equipment. This package supports only the following announcement arrangements:
Two 60-second announcements.
Single or multiple connections to an announcement is allowed.
Announcements will be recorded from a Dterm, Single line, or Attendant Console.

PN-4DATC
This circuit card is a trunk card that supports the announcement function for a maximum of four
announcements without external announcement equipment. This package supports only the following
announcement arrangements:
Four 120-second announcements.
Single or multiple connections to an announcement is allowed.
Announcements will be recorded from a Dterm, Single line, or Attendant Console.

PN-IPTB (IPT)
The PN-IPTB circuit card is an IP Trunk card.
Accommodates the IP network and transmits/receives compressed voice or signals over an IP network.
Used together with maximum of four PN-4VCTI cards (16 channels).
10 BASE-T/100 BASE-TX twisted pair cable is connected directly to this card.

PN-4VCTI (VCT)
The PN-4VCTI circuit card is a 4-channel CODEC card for IP Trunks; four VCT cards per IPT card.
Voice compression protocols: G723.1, G729A, G711, FAX (14.4kbps), DTMF signals
Used together with PN-IPTB card
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Chapter 6 Trunking

PN-8RSTG (DTMF Receiver)
The RSTA card receives DTMF signals sent by single line telephones, DID lines, and tie lines. This
card mounts in any LT slot in any PIM and provides 8 RST circuits per card. Usually at least one card
per system is required. This card uses 8 time slots. This card is required for a 512 Port System. The
128 Port System’s CPU is equipped with 4 RST circuits.

PN-4RSTBA (MF Receiver for T1 MF ANI)
This MF receiver card is required when using the T1-ANI feature. This card receives the MultiFrequency (MF) signals sent by the CO to ID the calling party. One RSTB card provides four MF
receiver ports. This card mounts in any AP slot in PIM’s 0 to 7.

PN-4RSTC (ID Number Receiver)
The Caller ID (Class) feature requires this card to receive and store Caller ID information from the CO.
Frequency Shift Keying (FSK) is used to send calling number and calling name information to the PBX.
The RSTC card receives this information from the CO and provides flexible options for the use of the
Caller ID messages.

PN-4RSTBA-911 (MF Sender)
This MF Sender card is used in conjunction with a trunk circuit to provide 911 information CAMA type
interface. The trunks to be used for the E-911 System must be ordered from the local telephone
company provider as E-911 trunks corresponding to the CAMA standards.
Note:

Ground Start trunks cannot be utilized for E-911 service.

PZ-M606 (ETHERNET TCP/IP)
Peer-to-Peer IP Stations or Trunks require the PZ-M606 Ether Control Card. This card is mounted on
MP card to accommodate the Ethernet transmit and receive a signal of TCP/IP protocol. 10 BaseT/100 BASE-TX twisted pair cable is connected directly to this card.

SPN-8IPLA IP PAD
The PN-8IPLA IP PAD card is an 8 channel IP to TDM Gateway with integrated G.711 CODEC (64
Kbps) and G.723.1 (5.3 Kbps), G.729a (8 Kbps) compression.

PZ-24IPLA IP PAD
The PZ-24IPLA card can be mounted to the 8IPLA card to provide 32-channels of IP-PAD with
compression.

SPN-32IPLAA IP PAD
The SPN-32IPLAA IP PAD card is a 32 channel IP to TDM Gateway with integrated G.711 CODEC.
Compression (G.729/G.723) requires one PN-16VCTAA IP PAD for 16 channels and two PN-16VCTAA IP
PAD cards for 32 channels.

SPN-16VCTAA IP PAD
This card provides G723 and G729a vice compression for Peer-to-Peer IP Trunking.

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Chapter 6 Trunking

System Trunking Capacity
IPS Trunking Capacity
System Capacity
1PIM 2PIM 3PIM 4PIM 5PIM 6PIM 7PIM 8PIM
Loop Start
64 128 192 256 256 256 256 256
Central Office Trunk (Lines)
DID w/4DID
48
96 144 192 240 256 256 256
2W E&M
24
48
72
96 120 144 168 192
Tie Line Trunk (Lines)
4W E&M
24
48
72
96 120 144 168 192
CCIS Trunk (Peer to Peer Connection)
Max. 127
DTI/CCIS Digital Link
1.5M
DTI: 10, CCIS: 8
Note 1
2MI
8
1.5M/2M (PRT)
8
ISDN
4BRT (card)
6
12
18
24
24
24
24
24
IP PAD
No. Channels
64
128
192
256
IP Trunk
1
2
3
4
5
6
7
8
3-Party Conference
Max. 16 conference groups per system
6-Party
Max. 4 conference groups per system
6-/10-Party Conference
10-Party
Max. 2 conference groups per system
32-Party Conference
Max. 8 conference groups per system
Announcement Trunk Circuits
128
DTMF Sender
Max. 32 circuits per system
DTMF Receiver
16
32
Item

Note 1:

Page 6-6

The total number of trunk line and DTI channel shall be 256 or less.

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Chapter 6 Trunking

IPS DM/DML Trunk Capacity
Number of PHYSICAL MODULAR CHASSIS

Capacity Per MC
2

Loop Start

112

DID w/4DIDA

2W/4W E&M

IP-PAD

No. of channel

Central Office Trunk (Lines)

CCIS Trunk (Peer to Peer Connection)
1.5M

DTI: 10, CCIS: 8

1.5M/2M(PRT)

4BRT (card)

DTI/CCIS Digital Link
Note 1

Max. 127

ISDN
IP Trunk
3-Party Conference
6-/10-Party Conference

Max. 16 conference groups per system
6-Party

Max. 4 conference groups per system

10-Party

Max. 2 conference groups per system
Max. 8 conference
7
groups per system
Max. 32 circuits per system

32-Party Conference
DTMF Sender
DTMF Receiver

IPS DMR Capacity
Number of PHYSICAL MODULAR CHASSIS
IP-PAD
Central Office Trunk (Lines)
DTI
ISDN
Note 1:

Note 1

Capacity Per MC
1

No. of channel

Loop Start

112

DID w/4DIDA

2W/4W E&M

1.5M

1.5M(PRT)

4BRT (card)

The total number of trunk line and DTI channel shall be 256 or less.

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Chapter 6 Trunking

Least Cost Routing (LCR)
The UNIVERGE NEAX 2000 IPS provides extensive Least Cost Routing capabilities. LCR
allows the UNIVERGE NEAX 2000 IPS to be programmed to route outgoing calls over the most
economical facility (WATS, FX, and DDD), based on the area code and office code dialed. In
the case of Specialized Common Carriers, associated NXX patterns can be screened out via
6-digit Toll Restriction if the dialed number is not within the designated service area of that
SCC.
The UNIVERGE NEAX 2000 IPS system provides one of the most flexible route selection
capabilities of any system on the market today. While many systems can add and/or delete
digits on incoming calls, it is the capability of what can be done on outgoing calls that make the
UNIVERGE NEAX 2000 IPS such a powerful and cost effective system.
When multiple routes are to be used for an outgoing call, the system is programmed with route
order priority (RA). Priority 1 usually being the most cost effective route for the call being
made. Up to eight route choices can be assigned for a call. These choices would be put into an
Outgoing Route Pattern (OPR). The system allows for up to 64 different OPRs to be
constructed.
For true cost effective use of outgoing dialing, it may be desired to change the outgoing route
selection during specific times when certain carriers may offer cheaper rates. These OPRs can
be assigned using one of eight different Time of Day Patterns (TDPTN). In TDPTN data, you
assign what pattern is used at what time of day. You then assign the Time Pattern in the OPR
data.
Outgoing calls made using the Least Cost Routing (LCR) can have up to 24 digits added to the
number dialed. In OPR data, the system will ask if and how many digits are to be deleted from
the number dialed. It will then ask if any digits are to be added, and if so, what are the digits to
add in front.
Besides the flexibility of getting a call out of the system, restriction controls make this system
one of the best. There are various types of call restrictions that can be made. Some of them
are as follows:
Six Digit Least Cost Routing
This allows you to restrict specific area code(s) from going out a specific route, even though
the route is included in the OPR constructed. This table lets you choose the TDPTN, the OPR,
the RA, and the area code. Then, designate whether the area code is allowed or not.
Route Restriction Class (RSC)
There are 8 different restriction classes that can be created to allow, deny, or toll restrict
certain users from dialing certain numbers. An RSC is assigned to the stations. The RSC table
is then constructed on a per route basis to determine whether a number dialed from a station
in a specific RSC is to be allowed, denied, or toll restricted.
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Chapter 6 Trunking

Toll Restriction
The toll code restriction data allows the system administrator to program certain restrictions for
Tandem (trunk-to-trunk) and Outgoing calls. Outgoing toll restriction is based on a station
RSC. For outgoing, the parameters request the Route Number, Destination Code, RSC, and
Toll Data Index (TDI). The TDI can be assigned as; Connection is restricted, Connection is
allowed, 3/6 Digit toll restriction, or C.O operator call. When 3/6 Digit toll restriction is used, the
system will ask what digit to check the restriction upon (3rd or 6th).
3/6 Digit Toll Restriction
3/6 Digit Toll allows restrictions to be assigned starting with a specific digit place in the dialed
number. This command will request the Route Number, RSC, Digit Code (3 or 6 digits), and
whether it is allowed or restricted. For example, if I wish to restrict stations with an RSC of "4"
from dialing 555 numbers only if a specific route (Route n) gets chosen in the LCR sequence:
In the Toll Restriction data, assignment would be to check the 6th digit (9 + 1 + 201 + 5XX +
XXXX). In the 3/6 digit toll table assign Route n, RSC4, DC 555, RES: 0 [restricted].
Priority Restriction Class
These provide 8 levels of control over a user's route access eligibility. The system allows a call
to reach its destination if the RSC is greater than or equal to the Priority Restriction Class of
the outgoing trunk group (route).

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Chapter 7 Attendant Answering Position
Attendant Console (SN716)
The Attendant Console (SN716 DESKCON) has an ergonomic design and provides full access
to all PBX Console features. The SN716 DESKCON uses the same interface cards as the
digital phones to connect to the PBX. The SN716 Desk Console uses a 6-core Modular Jack;
inner 1 pair for signal and outer 2 pairs are for power supply to connect to the PBX system.
The SN716 Desk Console operates on a switched-loop basis with a maximum of 6 Attendant
loops terminating at each console on the associated Interface card. The Attendant uses these
loops for answering, originating, holding, extending, and re-entering calls. When Attendant
loop release is used, the number of loops is effectively increased to a maximum of 12 for each
console.

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Chapter 7 Attendant Answering Position

SN716 DESKCON General Features
Character LCD (4 x 40 character)
LCD designation strips
Software-controlled LCD loop key
Full access to PBX features
Headset connectivity
Recorder connectivity

SN716 DESKCON Exclusive Features
While the DESKSON has full access to PBX features the SN716 DESKCON has the following exclusive
features.
DESKCON Exclusive Features
Attendant Assisted Calling
Call Waiting Display
Attendant Camp-on (Full and Semi-automatic)
Common Route Indial
Attendant Called/Calling Name Display
Dialed Number Identification Service (DNIS)
Attendant Called/Calling Number
Incoming Call Identification
Attendant Call Selection
Individual Trunk Access
Attendant Console Lockout-Password
Multi-Function Key
Attendant Do Not Disturb Setup and Cancel
Multiple Console Operation
Attendant Interposition Calling/Transfer
Pushbutton Calling - Attendant Only
Attendant Lamp Check
Serial Call
Attendant Listed Directory Number
Time Display
Attendant Loop Release
Trunk Group Busy Display
Attendant Programming
Unsupervised Trunk-to-Trunk Transfer By Attendant
Attendant Training Jacks
Attendant Delay Announcement
Audible Indication Control
Attendant Lockout
Call Processing Indication
Attendant Overflow
Call Queuing
Attendant Override
Call Splitting
Note: For Detail of each feature refer to Chapter 9 Feature Description.

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Chapter 7 Attendant Answering Position

Face Layout of the SN716 DESKCON

Alarm Position Available

(24)

(25)

Position Busy

Night

(26)

(27)

(28)

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Chapter 7 Attendant Answering Position

Functions and use of Keys and Lamp Indications
Location
Number

Key or Lamp
Designation

Full
Name

Basic
/Option

Key/Lamp

L1-L6

Loop

Basic

Key

L1A-L6A

Loop Lamp
A

Basic

Lamp
(one per loop)

L1B-L6B

Loop Lamp
B

Basic

Lamp
(one per loop)

Push-button Dial

Push-button
Dial

Basic

Key

SRC

Source

Basic

Key & Lamp

DEST

Destination

Basic

Key & Lamp

Talk

Basic

Key

Cancel

Basic

Key

Hold

Basic

Key

Release

Basic

Key

Answer

Basic

Key & Lamp

Start

Basic

Key & Lamp

Page 7-4

Function
The attendant answers the call associated to
the particular loops. Loop keys are usually
used to reenter to held calls, answer
automatic recalls.
Steady green lamp indicates attendant
connected to the loop, or called station has
answered. Flashing green lamp indicates call
waiting to be answered. Steady red lamp
indicates call party busy. Flashing red lamp
indicates call held at the console.
Flashing red lamp indicates automatic recall
has been activated.
Allows the attendant to:
Process incoming calls
Originate calls
Activate various service features
Allows the attendant to speak with the calling
party. The associated lamp lights when the
attendant is connected. The source
trunk/station number will be shown in the
number display field.
Allows the attendant to speak with the called
party. The associated lamp lights when the
attendant is connected. The destination
station/trunk number will be shown in the
number display field.
Allows the attendant to join in a three-way
conference with the calling and called parties.
When connection is established, both SRC
and DEST lamps will light.
Allows the attendant to:
Disconnect the calling (source) or called
party (destination) from the loop.
Disconnect an outgoing trunk or tone
seized by the attendant.
Disconnect the station recalling attendant
for transfer assistance.
Allows the attendant to hold a call at the
console and/or to activate it to serial call
state.
Allows the attendant to release from an
established connection freeing the console for
processing of new calls.
Allows the attendant to answer incoming
calls in the order in which they arrive at the
console.
Allows the attendant to extend an outgoing
call to a station. Completion of outgoing
pulse will be recognized.

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14
15
16
17
18

21
22
23
24
25

Incoming Call Identification: These eight non-locking keys with associated lamps provide attendant access to specific
types of incoming calls. A flashing lamp indicates a call waiting to be answered. A steady lamp indicates a call
answered. The standard arrangement of these keys is shown on the face layout.
LDN
Listed Directory Number
Basic
Key & Lamp
Incoming central office trunk call.
Allows the attendant to answer incoming Tie
TIE
Tie Line
Basic
Key & Lamp
Line calls when the distant station dials
access digit to the attendant.
Allows the attendant to answer incoming
Call Forwarding-Busy
Busy
Basic
Key & Lamp
calls to specified station when the station is
Line
busy.
ATND
Attendant
Basic
Key & Lamp
Incoming station call.
Allows the attendant to answer incoming
Call Forwarding-Don’t
calls to specified station when the station
NANS
Basic
Key & Lamp
Answer
does not answer within the predetermined
time.
Incoming station call for attendant assistance
Recall
Recall
Basic
Key & Lamp
in transferring an established outside call to
another station.
Additional incoming special service calls,
Option
Optional
Option
Key & Lamp
such as FX.
Additional incoming special service calls,
Option
Optional
Option
Key & Lamp
such as FX.
Allows the attendant to connect with pager.
PAGE
Page
Basic
Key & Lamp
(overhead paging)
Allows the attendant to connect with
REC
Record
Basic
Key & Lamp
recorder.
Allows the attendant to answer incoming
EMG
Emergency
Basic
Key & Lamp
calls from the station where the station leaves
the receiver off.
Busy
Allows the attendant to enter into station-toBV
Basic
Key & Lamp
Verification
station connection.
Trunk
Allows the attendant to individually select a
TRKSL
Basic
Key & Lamp
Selection
desired trunk.
Allows the attendant to establish incoming
call to Call Park.
Note: The attendant can connect this call
Call Park
Call Park
Basic
Key & Lamp
once again by dialing a specific number and
individual number of the console from an
ordinary extension telephone.
Allows the attendant to be automatically
recalled when the station user replaces the
Serial call
SC
Basic
Key & Lamp
handset, by depressing the key after
Set
extending a central office incoming call to
the station user.
Allows the attendant to supervise a call by
Supervisory
SVC
Basic
Key & Lamp
depressing the key after extending a central
Call Set
office incoming call to the station user.
Allows the attendant access to various
Option
Optional
Option
Key & Lamp
optional features provided as required. Each
key has an associated lamp.
Allows the attendant to cut off the voice
Mute
Mute
Basic
Key & Lamp
transmission.
Steady lamp indicates trouble conditions in
Alarm
Alarm
Basic
Key
the PBX.
Position
When the attendant position (console) is
Position Available
Basic
Lamp
Available
available to process calls, the lamp lights.

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Chapter 7 Attendant Answering Position

Position Busy

Position
Busy

Night

Basic

Key & Lamp

Up Down

Up and
Down

Basic

Key

Basic

Key & Lamp

LCD

Liquid
Crystal
Display

Basic

Display

Multi-Function
Key

MultiFunction
Key

Basic

Display

Page 7-6

When the key is pressed, the lamp will light,
and the console will become not available.
Note: Press the button if operators leave
their seats.
Allows the console to place in the night
answer mode and lights the associated lamp.
Releasing the key restores the console to
normal operation.
Allows the attendant to adjust volume of the
receiver, ringer and contrast of the LCD.
The following information will be displayed:
1st line:
The kind of party that connects to the
attendant, the number of the waiting calls, the
date and time.
2nd line:
Tenant number, station class of service and
station number belonging to the destination
(called) party.
3rd Line:
Tenant number, station class of service and
station number belonging to the source
(calling) party.
4th Line:
Optional indication, such as trunk busy.
Keys while idle:
Mode—allows
access
to
DAY/NIGHT mode and LKOUT
(Console Lock out mode)
Prog—allows
access
for
programming DISA, System Speed
Dial, Date & Time and Tone
Ringer
Keys while answering or originating:
SPB—Out Pulse Short
LPB—Out Pulse Long
SHF—Flash Over Trunk
Keys while calling a busy station:
B.V—Busy Verification
Keys while calling a DND station:
DDOVR—DND override
Keys while accessing Hotel feature:
RC—Room Cut off
MW—Message Waiting
DD—Do not Disturb
WU—Wake up call
RESET—Reset

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SN716 Specifications
ITEM
SN716
DESKCON

Interface and
Power Options
PN-8DLCP

Q’ty
1/8 ATT

PN-4DLCM

1/4 ATT

PN-4DLCQ

1/4 ATT

PN-2DLCN

1/2 ATT

PN-PW00

1/ ATT

ACA-U Unit

1/ ATT

Remarks
Interface Card (-27V)
Distance: Max. 984.3ft (350m)
Interface Card (-27V)
Distance: Max. 984.3ft (350m)
Interface Card (-27V)
Distance: Max. 984.3ft (350m)
Interface Card (-48V)
Distance: Max. 984.3ft (350m)
DC/DC Converter (-27V to –48V)
Distance with 2/4/8 DLC (350m)
24V AC/DC Adapter
Distance w/8DLC (350m)
Distance w/4DLC or 2DLC (1200m)

The UNIVERGE NEAX 2000 IPS can support up to (8) SN716 Desk Consoles.
Dimensions: 10 inches (25.4 cm) wide x 9 inches (22.9 cm) deep x 4 inches (10.2 cm) high.

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Chapter 7 Attendant Answering Position

Business Attendant System (BAS)
NEC’s Business Attendant System (BAS) is a powerful client/server PC-based tool for full-time
and part-time NEAX PBX systems answering positions. The BAS consists of an interactive
client side application running on Windows 98 SE, Windows 2000 (SP2) or Windows XP (SP1)
operating system that connects to the BAS server via the corporate LAN. The BAS server
software is used for PBX connectivity and runs on a Windows 2000 Server (SP2) or Windows
2003 Server. With a dynamic interactive directory and an Open Application Interface (OAI)
driven call queue, the operators are able to process a high call volume with minimal effort and
time expenditure in today’s sophisticated business environment. For example, an operator can
answer an incoming call from one of many queues; find out to whom the caller wishes to
speak, begin typing that name to select it in the directory, and once selected, hit the
key or click the entry to transfer the call. Additionally the caller could be parked, put on hold, or
sent to voice mail. Several operators can monitor the same queues and share the same
directory for call operation. These few examples barely scratch the surface of the capabilities
and features available for the system operator positions.

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BAS: Benefits
Centralized company database that can link and replicate with other databases, including
AIMWorX.
Sound playback and recordings on the operator telephone.
Provides operators with the ability to do more than answer the phone, since the client resides on a
PC.
Meet-me paging to enhance operator productivity.
Unlimited, global parked calls.
Different greetings/displays available for DID calls to the operator, which is especially useful in a
multi-tenant environment.
Single key call processing.
Directory dialing for accuracy.
Look ahead status from the directory that saves time, key strokes and enhances decision-making.
Sort and searchable directory by key fields.
Configurable screens for each operator.
Time stamped recordings with caller ID and optional note field that can be saved permanently or
used for sampling quality.
Queue types allowing flexibility to answer certain call types first if so desired.
Configurable information about the caller is displayed upon answering the call allowing personalized
service.
MAPI compliant e-mail integration.
Ability to set/cancel call forwarding and DND for other phones within the business.
Delay announcements capabilities to ensure callers are informed and provided other options during
busy call times.
Configurable night station or announcement when operators are not logged in.
Flexible DID number assigned to each operator that follows them regardless of what PC/phone they
use to log in.
Optional statistics package to track queue and operator statistics (next release).
Optional integration with OpenWorX Location Status Information to see an employee's availability,
return time, and alternate phone number.
Optional integration with OpenWorX Dialer providing Dterm users’ access to the same employee
and external database .It is searchable by site, department or name.
Optional integration with OpenWorX Short Text Messaging to notify wireless users of messages.

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Chapter 7 Attendant Answering Position

Business Attendant System: Directory
Both an internal directory and an external directory exist in the BAS system. The internal directory lists
employees within the company, while the external directory stores numbers outside the company.
External numbers are usually numbers that are frequently called, or information about people or
companies that frequently call the main listed number. When answering a call from one of these
external numbers, fields from this entry are displayed to the operator, providing more information than
just the 10-digit caller ID. Below is an example of the internal directory screen.

Many of the title bars at the top of the screen are capable of producing a directory sort by clicking the
mouse on the title bar. The information displayed is configurable by the administrator. There are
several different fields available for display and four of these fields can be user-defined. This window is
also expandable for viewing more fields. The “Search” field searches dynamically as the information is
typed in the field selected.
The directory is also capable of being partitioned by company. The drop down box at the top of the
window displays entries from a particular company, or all companies in the system. This is particularly
useful in shared tenant installations, where a single attendant group serves multiple companies.

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Business Attendant System: Speed Dials Pad
Another versatile screen is the Speed Dial pad shown below:

Up to forty (40) Speed Dial fields are available for each attendant. Dragging an entry from the directory
onto the speed dial button automatically programs these fields. Another great feature of these buttons
is that they act as Busy Lamp Fields (BLF). When that user’s phone is off hook, the speed dial button
becomes red and when that phone is ringing, the speed dial button blinks red. By right clicking the
button, the operator can enable or disable monitoring, as well as change the name, forward the
person’s Dterm, set DND, or initiate a screened, blind, or announced voice call. This is useful for an
operator who also has a set of VIPs for whom they answer calls.

Business Attendant System: Additional Features
As the BAS continues to grow, more features will be added and existing features will be enhanced.
Currently, there are several adjunct applications under development for use directly with the BAS
directory. Apart from the directory, another powerful feature integrated into the BAS client is the ability
to record conversations and playback greetings over the telephone. With the addition of a Dterm
Play/Record Module, the client will be able to automatically record voice conversations and instruct the
software to save the last specified number of conversations. Conversations are saved as .wav files,
which are time; date stamped and rotated in a circular buffer. When it is determined that a call needs to
be saved, the operator simply presses the “Save Recordings” button and the selected call is
automatically saved. Conversely, the operator can save a set of greetings to be played back
immediately every time the phone is answered via the BAS. Greetings playback can be determined
based on “internal” or “external” calls, on groups of caller IDs, or even morning or afternoon. A different
greeting can be recorded for any of the scenarios above allowing the attendant to present a pleasant
and consistent greeting over the telephone.

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Business Attendant System: CCIS Network Information
The following is a diagram of a simple CCIS network using the BAS.

The Business Attendant System can run in the above-pictured environment with the following
limitations.
Calls transferred from the BAS Client to a Dterm on the second PBX will not recall to the BAS
operator.
The Dterm in the second PBX cannot be monitored by the BAS system and the monitored
speed dials and status checks in the directory will not function for those extensions.
The system will force supervised transfers to CCIS extensions to ensure callers do not get
transferred to a busy remote extension. Operators need to listen for ring back tone or busy
tone, and then either completes the call or return back to the source caller as appropriate.
The message waiting lamp cannot be controlled on remote Dterm's.
Using multi-node capabilities, the operator can now view station data via the directory or speed
dials to extensions in other PBXs. The other PBXs must have OAI and a TCP/IP connection
back to the OpenWorX server. Additional OpenWorX software is also required per PBX to be
added on the server (Short Text Message and Nurse Call do not have multi-node capabilities).

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Business Application Environment
Description
Business AE UNIVERGE NEAX 2000

OWX BAS 1 Attendant (12 user Max)
OWX Dialer
OWX Incoming Call Assistant (ICA)
OWX Location Status Information (LSI)
OWX Message Reader
OWX TAPI Link Single PBX
OWX TAPI Link Multi-Switch
OWX Personal Call Assistant (100 user Max)
OWX Group Call Forward Control (GCFC)
OWX Name Display
OWX Short Text Messaging (STM) Email Interface
OWX Short Text Messaging (STM) TAP Interface
OWX Short Text Messaging (STM) Email Upgrade
OWX Short Text Messaging (STM) TAP Upgrade

Hardware Requirements
Description

Qty

PZ-M606-A

Remarks
Ether card 1 per system
For Peer to Peer connection
and MAT/OAI

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Chapter 8 System Administration
System Administration
In this system, the Customer Administration Terminal (CAT) or Maintenance Administration
Terminal (MAT) is used for programming the system data. The CAT is a digital multi-function
telephone (Dterm) which is equipped with function keys, a dial pad and LCD and interfaces
with the system via the MP card. The Maintenance Administration Terminal (MAT) is a
personal computer that provides an interface to the PBX via the system CPU card. The MAT
PC must have the MATWorX program properly installed to communicate with the PBX.
MATWorX is required for system software registration and activation.
Password Entry
In a system with password service, a maintenance person is required to enter an authorization level
number (Password Level) and appropriate password prior to engaging in programming the system data
with the MAT/CAT. A maximum of eight (8) Password Levels can be set up. The number of commands
that the maintenance person can access is determined by the Password Level.

Resident System Program
This resident system program generates system data automatically according to the system hardware
configuration, thereby providing immediate operation and shorter programming time. When activated,
the system scans hardware configuration (such as line/trunk card location) and assigns the system data
(such as station numbers, trunk numbers, etc.) according to a predetermined generic program
assignment.
Service Conditions
This service is applicable for equipment installed in PIM0 through PIM3.
Data for any vacant slot is not assigned.
Virtual stations are not assigned.
Some line/trunk cards (PN-AUCA/PN-DK00/PN-CFTA/PN-CFTB/PN-2AMP/PN-4DAT/PN-4RSTF-A/PN4VCTI/PN-32IPLA) are not assigned, even if mounted.
An application card is not assigned, even if mounted.
No tenant assignment is provided. (Tenant 01 is assigned)

Customer Administration Terminal (CAT)
The Customer Administration Terminal (CAT) is a digital multi-function telephone (Dterm)
which is equipped with function keys, a dial pad and LCD and interfaces with the system via
the MP card. Programming of the system can be done from selected Multiline Terminals with
LCD. The designated Multiline Terminals can be placed in program mode, and system data
can then be changed. To prevent unauthorized changes, password levels are assigned,
providing authorization for access to certain areas of programming and denying access to
others.

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Chapter 8 System administration

Service Conditions
1. Programming from a Customer Administration Terminal can only be used when the system is online.
2. All Multiline Terminals with LCD scanned during initialization will be Customer Administration Terminals.
3. The commands CM00 (Office Data All Clear) and CM01 (Office Data Partial Clear) cannot be accessed from
the CAT. The CAT cannot delete itself from the system program.
4. Only two Customer Administration Terminals can be in program mode at the same time.
5. The data that can be changed from the CAT can be limited by the Password level assigned. There are eight
levels of Passwords that can be assigned in system programming. The relation between Password level and
access to available commands is also assigned in system programming.
6. A password can consist of a maximum of any eight digits with the following limitation: The password cannot
be CCCCCCCC or FFFFFFFF.
7. Caution should be exercised when assigning Passwords to command authorization levels. If a password is
forgotten, access to system programming will be limited and a system initialization with subsequent
programming may be required.
8. When the Customer Administration Terminal is offline for programming, it cannot access normal terminal
functions.

Maintenance Administration Terminal (MAT)
The Maintenance Administration Terminal (MAT) is a personal computer (PC) that is used for
programming and maintenance of the UNIVERGE NEAX 2000 IPS system. The MAT can
provide a Maintenance Printout, Peg Count information and fault message output. Additionally,
the MAT can be used to Remove and Restore to service any station in the system and can
read or save system data from disks. The MAT can assign the Key Data for the Attendant
Console. The MAT requires an IBM or compatible PC running Microsoft Windows 98, NT 4.0,
2000 or XP and MATWorX.
MATWorX is a Graphical User Interface (GUI) program that provides an efficient method for
manipulating the PBX database. This program contains extensive help files, Usage Wizards
and Tool Tips, with hyperlinks imbedded in the text. The hyperlinks provide quick access to the
appropriate Add-In modules. Add-In modules provide a user-friendly, intuitive method for
customizing the PBX database.
MATWorX add-ins makes it easy for you to add or remove PBX features at any time. An add-in
is a modular component that lets you program specific features such as Caller ID, Station
Assignments, Day/Night Modes, and Line Key Assignments.
Because an add-in is modular, you can add, remove, and upgrade them individually from
within MATWorX. An Add-in lets you modify your PBX's features without having to upgrade
the MATWorX application itself. MATWorX also gives you a convenient way to launch other
commonly used applications, such as Microsoft Word or Excel, directly from its Toolbar.
There are three ways to connect your PC to an NEC PBX:
Use a modem to establish a dial-up connection.
Use a serial cable to establish a direct connection.
Use TCP/IP over your Local Area Network (LAN)

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Chapter 8 System Administration

The method you use depends on how you installed and configured the device to which you
want to connect. A serial cable direct connection offers better performance than a modem
connection, but requires that the PC and device be within 50 feet of each other. A TCP/IP
connection offers excellent performance and flexibility but requires a network connection to
both your PC and the device.
PBX Configuration Wizard
The PBX Configuration Wizard is a custom tool in MATWorX that enables you to establish the proper
communication settings between your computer and the UNIVERGE NEAX 2000 IPS. The Wizard asks
you simple questions and then uses the information to automatically configure the connection for the
PC and the PBX.
Service Conditions

1. Connection through modems is available, providing remote maintenance capabilities.
2. MATWorX can be connected to the system either directly or remotely. Direct connection is through
the RS connector on the MP card. Remote connection is available via either an internal modem on
the MP card or an external modem for high speed. Remote connection via the internal modem is
through the COT card. Connection between the modem and the COT is accomplished through
internal switching of the TDSW. Remote connection via an external modem is through the MP card.
3. The following functions can be performed from MATWorX:
System, station, and trunk data entry, change, and copy.
Loading, saving, and verification of system data to and from a disk.
ROM check readout of generic program.
Display of fault/fault cleared messages.
On-site or remote access to the system.
Printout of system data (when printer is connected to PC).
Display and setting of system clock/calendar.
Numbering Plan
Least Cost Routing (LCR)
System initialize
UCD/Station Hunting/Call Pickup – Group

4. The PC used with MATWorX must have an RS-232C port, and cannot be located more than 50 feet
(15m) from the system when connected on premises.
5. When stations or trunks are expanded, moved, or changed, office data for a Multiline Terminal
key/station/ trunk can be copied and multiple assignments of related office data is possible.

MACH Script Editor
This is a powerful timesaving tool that enables you to create a list of UNIVERGE NEAX 2000 IPS
commands that perform tasks in the PBX. This list is referred to as a script, which can be saved and
run at anytime. You can also use the MACH Script Editor to perform many other operations.

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Chapter 8 System administration

Traffic Management
The UNIVERGE NEAX 2000 IPS provides traffic management reports to be used for overall analysis of
system performance. MATWorX is used to request and display the type of report, sample measurement
time period, and time increments of reports. Type of Traffic Measurement Reports:
Number of successful attempts at outgoing access, based on trunk route.
Number of times all trunks were busy, based on trunk route.
Number of incoming calls, based on trunk route.
Number of incoming calls connected to busy tone and then trunk abandoned.
Quantity of incoming calls with no answer and trunk abandoned, based on trunk route.
Number of times a push button register was connected, on a system basis.
Number of times all push button registers were busy, on a system basis.
Number of outgoing connections using modem trunks, based on modem trunk group.
Number of incoming connections using modem trunks, based on modem trunk group.
Number of times all modem trunks were busy, based on modem trunk group.
Number of times the Conference circuits were used, on a system basis.
Number of times Conference circuits were all busy, on a system basis.
Number of times an incoming call was Call Forwarded-No Answer to the Attendant or another station (on
DID, Tie or DIT lines), on a system basis.
Number of Tandem Connections, on a system basis.
Number of times a push button register was connected to a trunk, based on trunk route.
Number of Attendant calls including recalls, on a system basis.
Number of station-to-station calls, on a system basis.
Number of times senders were all busy, on a system basis.
Number of ring generator capacity overflows, on a system basis.
Number of DTE to DTE connections, on a system basis. (UCD Peg Count)
Number of answered calls by UCD group.
Number of incoming calls by UCD group.
Number of waiting calls for a pre-determined time into queuing mode on the UCD group.
Number of incoming calls to all busy stations in the UCD group.
Number of answered calls in the UCD group.

MATWorX PC Requirements
MATWorX requires an IBM or compatible PC running Microsoft Windows®ME, 2000 or XP.
Minimum Requirements

Pentium III 350 MHz processor
128 MB RAM
Available hard-disk space
before installation:

Recommended

Pentium IV 1 GHz processor
256 MB RAM
Available hard-disk space
before installation:

200 MB - Full Installation
200 MB - Minimum installation
MATWorX uses about 100 MB
MATWorX uses about 100 MB
SVGA monitor, 800 x 600
15” SVGA monitor, 1024 x
resolution
768 resolution
3.5” diskette drive, CD-ROM drive, Mouse
Valid NEAX hardware connection
(direct serial, modem, or TCP/IP)

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System Diagnostics
When a fault occurs in the system, an audible and visual indication will be given at the
following units:
External alarm indicating unit
Fault messages reported at MATWorX for remote reporting
Alarm lamps in front of each package mounted in the frame

Self Diagnostic/System Messages
The UNIVERGE NEAX 2000 IPS provides a sophisticated array of self-diagnostic routines that
are continually and automatically being performed. Various system messages are printed
when a fault occurs in a central processor system, switch network processors. Many other
miscellaneous system messages and change of key status messages are also printed.

Remote Maintenance
This feature allows station and trunk changes or reassignments to be performed without a site
visit by service personnel, and can be used to retrieve fault codes prior to visiting a site. One
Remote Maintenance center can service an unlimited amount of systems, thus reducing the
amount of personnel to maintain each site, travel costs and customer billing for each site.
Service Conditions
1. The following additional equipment is required for this feature:
A modem at the maintenance center and one at each remote site. (When the internal modem of the Main
Processor (MP) is used, no modem at each remote site is required)
A cable for connection between the MP and the on-site modem. (When the internal modem of the MP is
used, the above cable is not required)

2. The internal modem of the MP is compatible with the following specifications:
Modem Specification
ITU-T V.22 1200 bps
ITU-T V.22 bis 2400 bps
Bell 212A 1200 bps

ITU-T V.32 4800/9600 bps
ITU-T V.34 19.2 k/33.6 kbps

3. Any one of the following connections are also required for access to the modem:
Required Connection
A dedicated line
Direct Inward System Access (DISA)

Attendant controlled transfer
Direct Inward Termination

4. The following operations can be executed from the Remote Maintenance location:
Retrieval of fault data
Retrieval of Peg Count information
Deletion or addition of system data (line, trunk, etc.) using a preprogrammed security password
Data assignment by device number (stations, trunks, and Attendant Console)
Copying of station data from one station to other stations (when adding sequential stations in groups)
Release / Reconnection of backup batteries
Display of station line status
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Chapter 8 System administration

MP Program Download
The UNIVERGE NEAX 2000 IPS provides Online MP Program Download via IP network using
the CP24C and CP31C. This feature allows an MP upgrade program to be downloaded to the
UNIVERGE NEAX 2000 IPS MP card with the PBX on-line and during the download process all
features and functions are available. The MP Program Download feature is available to a stand
alone system or a Remote system in a Remote PIM network.
The IPS downloads the MP upgrade program from FTP server using MATWorX. MATWorX
and the FTP server can be installed and running on the same PC. A single FTP server or
multiple FTP servers can be used. Immediate or scheduled changeover to the upgrade
program is available. It is also possible to change back to the previous program that was in use
before the changeover (changeback).

MP Download Process

Service Conditions
General Service Conditions

The MP program download can be executed to the PN-CP24-C/PN-CP31-C (MP card) when a
PZ-M606-A is mounted.
For the Retrofit system and Backup CPU system remote download of MP program is not
available.
The call processing is stopped while the system is initialized by the program changeover
execution.
Remote download of AP program is not available.
While the PBX is off-line, you cannot execute the program downloading, or program
changeover/ changeback/ program version matching (immediate). However, it is possible to
read various information and set the schedule (date/time) for program changeover/ changeback/
program version matching even while the PBX is off-line.
When the PN-32IPLA is mounted in the system, the MP program download (FTP) is not
available. When the PN-32IPLA-A or PN-8IPLA is mounted, it is available in the system.
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Chapter 8 System Administration

Program Download Service Conditions

Do not reset the PBX and the MATWorX during MP program downloading.
Do not pull out the LAN cable during MP program downloading.
FTP (File Transfer Protocol) is used as the protocol for the remote download of MP program.
Only the file specified in “File Name” of CHECKSUM.TXT can be downloaded to the PBX.
If you do not set the directory name to save the program files and CHECKSUM.TXT by
MATWorX, they are saved to the FTP directory (default).
If you do not set the password and user ID of the FTP server by MATWorX, login to the FTP
server with anonymous.
If required FTP server information is not set when executing the MP program download, “DATA
NOT SET” is displayed.
For the TCP port, the specified port is used for file transfer (control), and the port in front of it is
used for the file transfer. For example, when the TCP port is set to 3000, Port No. 3000 is used
for the file transfer (control), and Port No. 2999 is used for the file transfer (Port 21-File transfer
(control), Port 20-File transfer: default).
If you read the issue number of the standby side (outdated side) and change the settings of FTP
information during MP program downloading, “WAIT, BUSY NOW” is displayed.
When using a single FTP server to update multiple sites in remote PIM network schedule the file
transfer at least 7 minutes or more apart so that bandwidth and the number of connections to
the FTP server are not exceeded.
Program Changeover Service Conditions

The system is initialized automatically after executing the program changeover any data that is
not backed up will lost. Be sure to execute the office data backup before executing the program
changeover.
If an error is detected during system initialization after executing the program changeover, the
PBX starts automatically with the previous program.
If the program download is executed while the program version matching is being executed, the
program version matching is interrupted, and program download is executed.
If the program changeover (immediate) is executed while the program version matching is being
executed, the program is changed after the program version matching is completed.
The program changeover cannot be executed if the program downloading or program version
matching is interrupted or fails. If the program changeover (immediate) is set, the setting is
invalid.
The program changeover for which the schedule (date/time) has been set is executed only
when the PBX is on-line.
When immediate program changeover is executed after a schedule (date/time) for program
changeover execution is set, the schedule is invalid.

Required Hardware
Equipment

Comments

CP24C/CP31C

Stand Alone or Remote PIM network
Version 11.0.0
See MATWorX PC Requirements above
Note
Windows 2000 Server/Windows 2000
Professional/Windows XP Professional
PC/Work station for FTP
Note
(Internet Information Server (IIS)supplied with
Windows 2000/XPProfessional can be used)
Note: MATWorX and FTP server program can be used on the same PC.

MATWorX IPS
PC for MATWorX

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Chapter 8 System administration

MA4000 Management System
MA4000 Management System is a web based, powerful voice server management
configuration suite. MA4000 offers centralized management for the UNIVERGE NEAX 2000
IPS Voice Server as well as simple and powerful tools for managing moves, adds and
changes. MA4000 also offers flexibility and security and allows the end user to manage their
voice server in the same ways they manage their networks.
Web Browser Operation
Email Alarm Notification
Alarm Client Notification
LDAP Auto Provisioning Service
LDAP Authentication
Windows Authentication
IT Friendly Interface
Flexible Access Levels (Roles)
Application Program Interface (API & SDK)
Security
HTTPS supported
Audit Trail Logging

Customizable Portals
Command Line Interface
Integration to OpenWorX
Centralized Authentication Service (NEC CAS)
System Health Monitoring
Range programming that is schedulable

The MA4000 IPS Assistant gives the power of station management directly to the end-user.
Limited by access rights, an administrator can allow his users to manage their own phones.
Supported features include Button Programming, Call Forwarding and Speed Dial
Programming.

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Chapter 9 Feature Descriptions
Business/Hotel/Data Feature List
Account Code
Add-On Module
Alarm Indications
Alphanumeric Display
Analog Port Adapter
Announcement Service
Answer Key
Attendant Assisted Calling
Attendant Camp-On (Full and Semi-automatic)
Attendant Console
Attendant Called/Calling Name Display
Attendant Called/Calling Number
Attendant Call Selection
Attendant Console Lockout-Password
Attendant Do Not Disturb Setup and Cancel
Attendant Interposition Calling/Transfer
Attendant Lamp Check
Attendant Listed Directory Number
Attendant Loop Release
Attendant Programming
Attendant Training Jacks
Audible Indication Control
Call Processing Indication
Call Queuing
Call Splitting
Call Waiting Display
Common Route Indial
Dialed Number Identification Service (DNIS)
Incoming Call Identification
Individual Trunk Access
Multi-Function Key
Multiple Console Operation
Pushbutton Calling - Attendant Only
Serial Call
Time Display
Trunk Group Busy Display
Unsupervised Trunk-to-Trunk Transfer By
Attendant

Attendant Delay Announcement
Attendant Lockout
Attendant Overflow
Attendant Override
Authorization Code
Automated Attendant
Automatic Call Distribution (ACD)
Busy In/Busy Out - ACD
Call Waiting Indication - ACD
Delay Announcement - ACD
Hunt Past No Answer - ACD
Immediate Overflow - ACD
Priority Queuing - ACD
Queue Size Control - ACD
Silent Monitor - ACD
ACD w/Management Information System (MIS)
Automatic Camp-On
Automatic Change to Daylight Saving Time
Automatic Number Identification (ANI)
Automatic Recall
Automatic Wake-Up
Background Music
Back Up CPU
Bandwidth Control
Boss/Secretary Calling
Broker’s Call
Call Back
Call Forwarding
Attendant Call Forwarding Setup and Cancel
Call Forwarding - All Calls
Call Forwarding - Busy Line
Call Forwarding - No Answer
Call Forwarding - Destination
Multiple Call Forwarding - All Calls
Multiple Call Forwarding - Busy Line
Multiple Call Forwarding - No Answer

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Chapter 9 Feature Descriptions

Business/Hotel/Data Feature List (Cont.)
Split Call Forwarding - All Calls
Split Call Forwarding - Busy Line
Split Call Forwarding - No Answer
Call Forwarding - Logout (DtermIP)
Call Forwarding - Override
Group Diversion
Call Park
Call Park - System
Call Park - Tenant
Call Pickup
Call Pickup - Direct
Call Pickup - Group
Call Pickup - Designated Group
Call Redirect
Call Transfer
Call Transfer - All Calls
Call Transfer - Attendant
Caller ID Class
Caller ID Display
Caller ID - Station
Camp-On
Centrex Compatibility
Check In/Check Out
CID Call Back
CID Call Routing
Class of Service
Code Restriction
Conference (Three/Four Party)
Conference (Six/Ten Party)
Conference (32 Party)
Group Call
Meet-Me Conference
Consecutive Speed Dialing
Consultation Hold
Customer Administration Terminal (CAT)
Data Line Security
Delayed Ringing
Diagnostics
Dial by Name
Dial Conversion

Page 9- 2

Direct Data Entry
Direct Digital Interface
Direct Inward Dialing (DID)
DID Call Waiting
DID Digit Conversion
DID Name Display
Direct Inward System Access (DISA)
Call Forwarding Set by DISA
Direct Inward Termination (DIT)
Direct Outward Dialing (DOD)
Direct Station Selection/Busy Lamp Field
(DSS/BLF) Console
Busy Out Status Console
Do Not Disturb Console
Message Waiting Console
Room Cutoff Console
Wake Up No Answer Console
Distinctive Ringing
Do Not Disturb
Do Not Disturb - Group
Do Not Disturb - Hotel/Motel
Do Not Disturb - System
Dterm Assistant
DtermIP
Elapsed Call Timer
Enhanced 911
Executive Calling
Executive Override
External Paging with Meet-Me
Fax Arrival Indicator
FAX over IP
Feature Activation from Secondary Extension
Flexible Line Key Assignment
Flexible Numbering Plan
Flexible Ringing Assignment
Forced Account Code
Group Call
Automatic Conference (6/10-Party)
2 Way Calling

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Chapter 9 Descriptions

Business/Hotel/Data Feature List (Cont.)
Group Call by Pilot Number Dialing
Group Listening
Hands free Answerback
Hands free Dialing and Monitoring
Hold
Call Hold
Dual Hold
Exclusive Hold
Non-exclusive Hold
Hotel/Motel Attendant Console
Hotel/Motel Front Desk Instrument
Hotline - Inside/Outside
House Phone
Individual Attendant Access
Intercept Announcement
Intercom
Manual Intercom
Automatic Intercom
Dial Intercom
Internal Tone/Voice Signaling
Internal Zone Paging with Meet-Me
IP Enabled Dterm
Last Number Redial
Least Cost Routing - 3/6 Digit
Line Lockout
Line Preselection
Maid Status
Maintenance Administration Terminal (MAT)
Message Center Interface (MCI)
Message Registration
Message Reminder
Message Waiting
Miscellaneous Trunk Access
CCSA Access
Code Calling Equipment Access
Dictation Equipment Access
Foreign Exchange (FX) Access
Radio Paging Equipment Access
Wide Area Telephone Service (WATS)
Access

Mobility Access
Modem over IP
MP Program Download (FTP)
Multiple Language Display
Multiple Terminal Attendant Position
Music On Hold
Night Service
Attendant Night Transfer
Call Rerouting
Day/Night Mode Change by Attendant Console
Day/Night Mode Change by Station Dialing
Day/Night Mode Change by System Clock
Night Connection - Fixed
Night Connection - Flexible
Trunk Answer Any Station (TAS)
Overflow for TAS Queue
Queue Limit for TAS
No CID Call Routing
Off-Hook Alarm
Off-Premises Extensions
Open Application Interface (OAI)
Pad Lock
Periodic Time Indication Tone
Pooled Line Access
Power Failure Transfer
Priority Call
Privacy
Direct Privacy Release
Manual Privacy Release
Private Lines
Property Management System Interface
Proprietary Multiline Terminal
Automatic Idle Return
Called Station Status Display
Calling Name and Number Display
Group Feature Key
Dynamic Dial Pad
Hands free Unit
I-Hold/I-Use Indication

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Chapter 9 Feature Descriptions

Business/Hotel/Data Feature List (Cont.)
Microphone Control
Multiple Line Operation
Mute Key
My Line Number Display
Preset Dialing
Prime Line Pickup
Recall Key
Relay Control Function Key
Ring Frequency Control
Ringing Line Pickup
Soft Keys
Volume Control
Remote Hold
Remote PIM over IP
Reserve Power
Resident System Program
Return Message Schedule Display
Room Cutoff
Room Status
Route Advance
Save and Repeat
Security Alarm
Set Relocation
Short Message Service (SMS)
Single Digit Dialing
Single Digit Feature Access Code
SIP Trunk Interface
SNMP
Software Line Appearance (Virtual Extensions)
Stack Dial
Station Hunting
Station Hunting - Circular
Station Hunting - Terminal
Station Hunting - Secretarial
Station Message Detail Recording (SMDR)

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Station Speed Dialing
Step Call
Supervisory Control of Peripheral Equipment
System Clock Setup by Station Dialing
System Speed Dialing
Tenant Service
Terminal Login via NAT
Tie Lines
Tie Line Tandem Switching
Timed Forced Release
Timed Queue
Timed Reminder
Trunk - Direct Appearances
Trunk Queuing - Outgoing
Trunk-to-Trunk Connection
Uniform Call Distribution (UCD)
Busy In/Busy Out - UCD
Call Waiting Indication - UCD
Delay Announcement - UCD
Hunt Past No Answer - UCD
Immediate Overflow - UCD
Priority Queuing - UCD
Queue Size Control - UCD
Silent Monitor - UCD
Uniform Numbering Plan (UNP) - Voice and Data
Variable Timing Parameters
Voice Guide
Voice Mail Integration
Voice Mail Private Password
Voice Mail Transfer
VoIP
VoIP Log Collection
Whisper Page

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Business/Hotel/Data Feature Descriptions
Account Code
This feature, when used with Station Message
Detail Recording (SMDR), allows station
users and Attendants to enter a cost
accounting or client billing code (up to 16
digits) into the system.
Add-On Module
This feature allows the Add-On Module to be
combined with a legacy Multiline Terminal
when there are insufficient line or trunk keys
provided at the Multiline Terminal. When the
DCU-60-1 unit keys are programmed as
line/trunk keys, the additional 25 lines/trunks
and the existing lines/trunks set for the
Multiline Terminal can be accessed directly
(maximum of 49 lines/trunks). The station
speed dialing function can be assigned for all
keys on the DCU-60-1 unit. Also, one of the
last 3 keys can be used as a Day/Night
change key.
Alarm Indications
Faults are indicated by the Major/Minor
(MJ/MN) lamps located on the AC/DC Power
Supply and, optionally, an external alarm
display unit.
Alphanumeric Display
The DTR-16DC-1 Multiline Terminals are
each equipped with a 3-line, 24-character
Liquid Crystal Display (LCD). These displays
are used to provide alphanumeric information
including clock/calendar and call processing
information.
Station Application All Multiline Terminals with
an LCD display.

Analog Port Adapter
This feature allows an APR-U/AP(R)-R or
APA-U/AP(A)-R unit combined with a legacy
Multiline Terminal to connect to an analog
terminal such as an analog telephone,
Modem, and PC with built-in Modem. There
are two communication modes for the
terminal connected via the Analog Port
Adapter as shown below:
1. Single Port Mode
A Multiline Terminal and an analog terminal share
the same port. In this mode, the Multiline Terminal
and the analog terminal cannot be used
simultaneously.
2. Dual Port Mode
A Multiline Terminal and an analog terminal use
different ports. In this mode, the Multiline Terminal
and the analog terminal can be used
simultaneously.

Announcement Service
This feature allows station users to record
messages on Digital Announcement Trunk
(DAT) cards. When a station user dials the
feature access code for this feature, the user
receives the corresponding message from the
system.
Answer Key
An Answer Key is provided on all Multiline
Terminals. The Answer Key can be used to
answer incoming calls on outside lines, and
primary or secondary extensions. When the
Answer Key is used to answer an incoming
call with a call in progress, the first party is
placed on hold and the second party is
connected. If the Answer Key is depressed
while in a three-party call, the user can
alternate between each party and a Broker's
Call is established.

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Chapter 9 Feature Descriptions

Attendant Assisted Calling
This feature allows a station user to ask an
Attendant for assistance in originating a call.
Three methods are available: non-delay,
delay, and passing dial tone.
Attendant Camp-on (Full and Semiautomatic)
The Full Automatic Camp-on feature permits
the Attendant to hold an incoming call in a
special mode when the desired station for the
transfer is busy. The Attendant sends a
Camp-On tone to the busy station. When that
station becomes idle, it is automatically
alerted and connected to the waiting party.
The Semi-automatic Camp-on feature works
similarly to full automatic except when the
station becomes idle the call on Camp-on
recalls to the Attendant. Once the Attendant
answers the Camp-on recall the station is
then called and the call is connected.
Attendant Console (SN716 DESKCON)
The Attendant Console (SN716 DESKCON)
operates on a switched-loop basis with a
maximum of 6 Attendant loops terminating at
each console on the associated Interface
card. The Attendant uses these loops for
answering, originating, holding, extending,
and reentering calls. When Attendant loop
release is used, the number of loops is
effectively increased to a maximum of 12 for
each console.
Attendant Called/Calling Name Display

This feature provides a display of the
calling/called party's name on the Attendant
Console LCD for Attendant Called/Calling
Name Display. On attendant-to-station calls,
the LCD display the name assigned to the
primary extension of the station. On
attendant-to-trunk calls, the LCD displays the
name assigned to the trunk route of the trunk.

Page 9- 6

Attendant Called/Calling Number

This feature provides a display of the station
number and station name on the Attendant
Console during an Attendant-to-station
connection. During an Attendant-to-trunk
connection, the same display shows the trunk
route designation and a trunk identification code
(4 digits).
Attendant Call Selection

This feature allows assignment of keys on the
Attendant Console to particular types of trunk
routes (such as WATS or FX) and particular types
of service calls (such as Attendant recalls,
intercept calls, etc.). LEDs indicate the type of
incoming call and pressing the associated key
allows the Attendant to answer the calls in any
order.
Attendant Console Lockout-Password

This feature allows the Attendant Console to be
set into a lockout mode. This diables the console
from originating or receiving calls and setting or
resetting service features. To return the Console
to its manual operating condition a password is
required.
Attendant Do Not Disturb Setup And Cancel

The Attendant has the ability to enter and remove
individual stations from Do Not Disturb (DND).
Additionally, the Attendant can set one
preassigned group of stations into, or out of, Do
Not Disturb.
Attendant Interposition Calling/Transfer

This feature allows any Attendant to directly
converse with another Attendant and also allows
Attendants to transfer calls from their console to
another Attendant's console in systems where
Multiple Console Operation has been provided.

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Chapter 9 Descriptions

Attendant Lamp Check

This function is used to check the status of keys,
lamps, and LCDs mounted on the Attendant
Console to verify that various operations of the
Attendant Console are functioning normally. The
check is done by a preset procedure.
Attendant Listed Directory Number

This feature provides a display of the Listed
Directory Number on the Attendant Console when
the operator has answered a Listed Directory
Number call.
Attendant Loop Release

This feature allows an Attendant Console loop to
become available for a second call as soon as the
Attendant has directed the first call to a station,
even if that station does not answer.
Attendant Programming

This function is allowed only for the Attendant
Console (SN716) and is used to execute DISA
code set up, speed dial programming, and system
clock set up operations.
Attendant Training Jacks

The Attendant Console can be equipped with two
headset/handset jacks using an optional adapter.
Two jacks are equipped on the adapter and can
be used for training new operators.
Audible Indication Control

This feature allows the Attendant to adjust the
volume of audible indications received at the
Attendant Console.
Call Processing Indication

This feature provides visual indications of all calls
being processed or awaiting processing at the
Attendant Console.
Call Queuing

This feature provides the Attendant the ability to
handle a series of exchange network calls in the
order of their arrival, (first in, first out) thereby
eliminating unnecessary delays.

Call Splitting

This feature allows the Attendant to confer
privately with one party on an Attendant handled
connection without the other party overhearing.
Call Waiting Display

This feature provides a visual indication to the
Attendant when one or more calls are waiting to
be answered.
Common Route Indial

This feature allows assignment of incoming DID
calls to different Attendant Call Selection keys
based on the last 4 digits dialed into the system.
Up to eight individual Listed Directory Numbers
can be assigned in system programming. When
an incoming call to any of these trunks is
received, an Attendant Call Selection key will flash
and the LCD display will indicate the Listed
Directory Number associated with that trunk route.
Dialed Number Identification Service (DNIS)

This feature provides a display of the company
name on the Attendant Console when the
Attendant has answered a Listed Directory
Number or a Tie Line call.
Incoming Call Identification

Incoming calls are identified by various means.
Refer to Attendant Called/Calling Number,
Attendant Call Selection, Attendant Source Key,
Attendant Listed Directory Number and Common
Route Indial Features and Specifications.
Individual Trunk Access

The Attendant Console is provided with the ability
to access each individual trunk by dialing an
associated identification code. This allows
detection of faulty trunks during regular testing or
after complaints. The Customer Administration
Terminal (CAT) or Maintenance Administration
Terminal (MAT) has the capability to then busy
out the trunk until repair is made.

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Chapter 9 Feature Descriptions

Multi-Function Key

This feature allows the top row of keys on the
Attendant Console to perform and display multiple
functions in accordance with the status of call
processing.
Multiple Console Operation

This feature allows more than one Attendant
Console to operate within the same system.
Pushbutton Calling - Attendant Only

This feature permits an operator to place all calls
over Dual-Tone, Multi-Frequency (DTMF) lines
from the pushbutton keypad on the Attendant
Console.
Serial Call

This feature is activated by the Attendant
when an incoming calling party wishes to
speak with more than one internal party.
When the internal station subsequently
disconnects from the Central Office line call,
the Central Office party automatically rings
back to the same Attendant.
Time Display

This feature provides a digital time display on the
Attendant Console LCD.
Trunk Group Busy Display

A visual indication is supplied to the Attendant
when all trunks in a particular trunk group are
busy.
Unsupervised Trunk-to-Trunk Transfer By Attendant

This feature allows an Attendant to transfer an
incoming or outgoing call on one trunk to an
outgoing trunk and exit the connection before the
called party answers.

Attendant Delay Announcement
This feature provides an announcement, via a
Digital Announcement Trunk Card, to external
calls that are not answered by the attendant
within a predetermined time.
Attendant Lockout
This feature denies an Attendant the ability to
reenter an established trunk or station
connection without being recalled by that
station after the call is put in consultation hold.
Attendant Overflow
When an incoming call, which has terminated
from a trunk to the Attendant Console,
remains unanswered after a predetermined
time period, this feature provides a change to
Night Service for that particular trunk.
Attendant Override
This feature permits an Attendant to enter a
busy connection (station or trunk) using the
Attendant Console. When this feature is
activated, a warning tone is sent to the
connected parties after which, they are
connected with the Attendant in a three-way
bridge.
Authorization Code
An Authorization Code is a numerical code
which will temporarily change a station’s
Class of Service to a Class of Service
assigned to that Authorization Code. This new
Class of Service allows access to trunks,
dialing patterns, and/or features that would
otherwise be restricted.
Automated Attendant
This feature allows the system to answer
incoming trunk calls. The system will supply a
message and/or dial tone to the caller. The
caller can then dial the desired extension
number and be directed to that station.

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Chapter 9 Descriptions

Automatic Call Distribution (ACD)
The Automatic Call Distribution (ACD) feature
permits incoming calls to terminate to a
prearranged group of stations. Calls are
distributed in the order of arrival to idle
terminals within the group, based on which
terminal has been idle the longest period of
time. Stations may log on/log off from the
ACD group. Supervisor stations may monitor
conversations of agents.

Immediate Overflow - ACD

This feature allows a call directed to an ACD
group to immediately overflow to another ACD
group, upon encountering an “all agents busy”
condition.
Priority Queuing - ACD

This feature allows the system to prioritize
incoming calls by trunk route and on a per station
basis, when the call enters an ACD queue. When
a call is considered as a priority, it is placed at the
beginning of the queue.

Busy In/Busy Out - ACD

This feature allows an agent in an ACD group to
log their station into or out of the group. This
allows the system to control whether a call
directed to the pilot number of the ACD group
goes to that station or not. This prevents incoming
calls from being directed to stations at which no
agent is available.
Call Waiting Indication - ACD

This feature provides a visual indication when an
incoming call to an ACD group is placed in queue,
due to an “all agents busy” condition. On external
relay controlled indicator or an LED on a Multiline
Terminal can be used to provide Call Waiting
Indication.
Delay Announcement - ACD

This feature allows the system to provide a
recorded announcement to an incoming caller
placed in queue to an ACD group. A single
announcement, or two separate announcements,
can be provided.
Hunt Past No Answer - ACD

This feature allows calls targeted at an ACD group
to hunt past an agent’s station, after a no answer
condition, if the agent forgets to log off of the
group and the agent is unable (or not available) to
answer the call.

Queue Size Control - ACD

On incoming DID/Tie line calls, the system can be
assigned a threshold which limits the number of
calls in queue. When the queue size threshold is
exceeded, incoming callers are connected to busy
tone.
Silent Monitor - ACD

This feature provides the ACD group supervisor
with the ability to monitor a call to an ACD agent.
The silent monitor function gives no indication (as
an option) to either the agent or the calling party.

Automatic Call Distribution with
Management Information System (MIS)
The Automatic Call Distribution (ACD) with
Management Information System (MIS)
feature provides a management information
system to be used in conjunction with the
built-in ACD features of the system. The MIS
incorporates a supervisor's terminal for realtime monitoring of agent activity, amber and
red alarms, and hard-copy summary reports.
Automatic Camp-on
An incoming Direct Inward Termination (DIT)
call which has been terminated to a busy
station can be Camped-On automatically.
When the busy station becomes idle, the
station is automatically called and connected
to the camped on incoming trunk call.

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Chapter 9 Feature Descriptions

Automatic Change to Daylight Saving Time
This feature allows the 2000 IPS system clock
to automatically change from standard time to
daylight saving time, and vice versa. Schedule
to change to/from daylight saving time is
programmed by system data programming.
Automatic Number Identification (ANI)
This feature receives the calling subscriber's
number automatically sent from T1 network
using MF signaling and displays the calling
number on the LCD of a Multiline Terminal
and an Attendant Console.

Back Up CPU
UNIVERGE NEAX 2000 IPS provides a dual
CPU system with two MP cards. When
Emergency Notification from hardware is
detected, the changeover from an active MP
card to a standby MP card will occur. If the
active MP card becomes out of order for any
reason, the standby MP card starts up
automatically. The standby MP card employs
a Cold Standby System that will restart
initialization by the changeover from the active
MP card.

Bandwidth Control
This feature allows to assign an available
bandwidth threshold for VoIP traffic within a
Location and between Locations, and to
restrict outgoing/incoming calls when the VoIP
traffic exceeds the threshold. The Location is
a group of VoIP devices (IP Enabled Dterm,
IP-PAD, or Peer to Peer IP trunks (built-in IP
trunks)), which the same VoIP
Automatic Wake-up
This feature allows the system to be
communications parameters such as codec
programmed to automatically call guest rooms selection list and ToS field value are
or administration stations at specified times.
assigned.
Upon answering, the guest is connected to a
recorded announcement or music source. A
When the VoIP traffic over CCIS exceeds the
printout of unanswered or blocked Automatic
threshold, the call can be routed to legacy
Wake-Up attempts for each guest room is
trunks (TDM network). When exceeding the
threshold, the system can store fault
provided using the Hotel/Motel printer.
information and provide external alarm
indication.
Background Music
Background Music can be provided on a dialup basis over legacy Multiline Terminal
Boss / Secretary Calling
speakers. Incoming voice announcements,
A secretary with a Multiline Terminal can use
ringing and recalls override Background
an appearance of the boss' extension to
Music. Up to 10 music programs can be
screen calls for that extension, and announce
offered.
and/or transfer calls to that extension.
Additionally, the secretary can call the boss
during a busy condition and send a Message
Waiting Indication to the boss' station.
Automatic Recall
This feature works as a timed reminder. When
a call remains on Hold, Camp-On or ringing
unanswered for a fixed interval after being
transferred, the station that initiated the hold,
transfer, or Camp-On is automatically alerted.

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Broker's Call
This feature allows a Multiline Terminal or
Single Line Telephone user to alternate
between two parties, talking to one party while
the other party remains on Hold on the same
line. The Multiline Terminal user uses the TRF
or ANS key to alternate between the two
parties. The Single Line Telephone user uses
the Hold feature to alternate between the two
parties.
Call Back
This feature allows a calling party to set an
automatic Call Back when a busy or no
answer condition is encountered. When the
busy station becomes idle, the station that set
the Call Back will be called. In case of Call
Back no answer, the Call Back to the setting
station is initiated immediately after the called
station goes on hook after making a call or
accessing a feature.

Call Forwarding
Call Forwarding allows calls directed to a
station to be routed to another station, an
Attendant, an outside number or voice mail
equipment. The types of Call Forwarding
provided are:
Attendant Call Forwarding Setup and Cancel
Call Forwarding - All Calls
Call Forwarding - Busy Line
Call Forwarding - No Answer
Call Forwarding - Destination
Multiple Call Forwarding - All Calls
Multiple Call Forwarding - Busy Line
Multiple Call Forwarding - No Answer
Split Call Forwarding - All Calls
Split Call Forwarding - Busy Line
Split Call Forwarding - No Answer
Call Forwarding – Logout (Dterm IP)
Call Forwarding - Override
Group Diversion
Attendant Call Forwarding Set-up and Cancel

All of the various types of Call Forwarding can be
set up or canceled from both Attendant Consoles.
Call Forwarding - All Calls

This feature allows all calls directed to a particular
extension to be rerouted to an alternate
destination, regardless of the busy or idle status of
the extension. Call Forwarding - All Calls can be
set by an Attendant Console, the individual station
user, a Multiline Terminal with a secondary
appearance of the station's extension, or from
another station (which can program itself to be the
destination of the rerouting).

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Chapter 9 Feature Descriptions

Call Forwarding - Busy Line

Split Call Forwarding - Busy Line

This feature permits a call to a busy extension to
be routed to a pre-designated station, Attendant
Console, or voice mail equipment. Call Forwarding
- Busy Line can be set or canceled by an
Attendant Console, the individual station user, or a
Multiline Terminal with a secondary appearance of
the station's extension.
Call Forwarding - No Answer

This feature reroutes calls to extensions which do
not answer. These calls can be rerouted to
another station, an Attendant Console or voice
mail equipment. Call Forwarding - No Answer can
be set by the individual station user, an Attendant
Console, or by a Multiline Terminal with a
secondary appearance of the station's extension.
Call Forwarding - Destination

This feature allows a station (A) user to set Call
Forwarding - All Calls from another station (B)
within the system, to the user's station (A).
Multiple Call Forwarding - All Calls

When a forwarded call is rerouted to a station that
has also set a Call Forward, the call can be
forwarded to another station. A call can be
forwarded up to a maximum of five times, as
specified in system programming.
Multiple Call Forwarding - Busy Line

This feature permits a call to a busy station to be
forwarded, multiple times, to a pre-designated idle
station.
Multiple Call Forwarding - No Answer

This feature permits a call to an unanswered
station, the ability to be forwarded multiple times
to a pre-designated station that does not have Call
Forwarding - No Answer set or to the Attendant
Console.
Split Call Forwarding - All Calls

This feature allows all internal and external
calls to a busy extension to be rerouted to
different destinations individually, regardless
of the busy or idle status of the extension.

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This feature allows internal and external calls to a
busy extension to be rerouted to separate
destinations. Destinations may be an internal
station, Attendant Console, or voice mail.
Split Call Forwarding - No Answer

This feature allows internal and external calls, to
extensions that do not answer, to be rerouted to
separate destinations individually.
Call Forwarding – Logout (Dterm IP)

This feature allows a call terminated to an IP
station in logout status to be forwarded to a
predestinated station, outside number, Attendant
Console or Digital Announcement Trunk (DAT).
This feature is also applicable to the IP stations
when the LAN cable is pulled out or the power is
off.
Call Forwarding - Override

This feature allows the call forward destination
station to call the station which set call forward.
The call forward setting will be ignored.
Group Diversion

This feature allows all calls terminated to an
extension that are not answered within a
predetermined time to be forwarded to a predesignated station.

Call Park
This feature enables a station user or
attendant to place a call into pre-designated
Call Park locations. The station user or
attendant is then free to process other calls.
This feature is available system wide and for
individual tenants.
Call Park - System

When a call is parked by Call Park-System, the
call can be retrieved from Call Park by any station
in the system.

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Call Park - Tenant

When a call is parked by Call Park - Tenant, the
call can be retrieved from Call Park-Tenant by any
station within the tenant from which the call was
originally parked.

Call Pickup
This feature enables a station user to answer
any call directed to another station, to a
station within the user's own Call Pickup
Group, or to a station within a different Call
Pickup Group. Three Call Pickup methods are
available: Call Pickup - Direct, Call Pickup Group, and Call Pickup - Designated Group.
Call Pickup - Direct

This method permits a station user to pickup a call
to any other station in the system by dialing a
specific Call Pickup feature access code and the
number of the called extension.
Call Pickup - Group

This method permits a station user to answer any
calls directed to other extensions in their preset
pickup group by dialing a Call Pickup - Group
feature access code.
Call Pickup - Designated Group

This method permits a station user to answer an
incoming call directed to another group by dialing
the Call Pick-up - Designated Group feature
access code and any station within the group to
which the ringing station belongs.

Call Transfer
This feature permits a station user to transfer
a call to another station in the system directly,
or with assistance from the attendant.
Call Transfer – All Calls

This feature permits a station user to transfer
incoming or outgoing calls to another station within
the system without attendant assistance.
Call Transfer – Attendant

This feature permits a station user, while
connected to an internal or outside call, to signal
the Attendant and have the Attendant transfer the
call to another station within the system or to an
outside connection.

Caller ID Class
This feature receives the calling subscriber’s
name and number sent from a public network
using a MODEM signal and displays the name
or number on an LCD of a Multiline Terminal
and Attendant Console.
Caller ID Display
Without answering incoming calls or held calls
which terminate to the line keys of a Multiline
Terminal, the calling party's information can
be confirmed by the indications on the LCD.

Caller ID Station
This feature enables a user to connect an
analog telephone with Caller ID display
Call Redirect
Without answering incoming calls or held calls function, and provides the calling party’s
number and name on the display without
that terminate to the line keys of a Multiline
Terminal, the calls can be transferred to a pre- answering incoming calls.
programmed station or Voice Mail System.
Two transferring destination number can be
designated per tenant, in system data
programming. This feature can be used
together with the Caller ID Display feature.

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Chapter 9 Feature Descriptions

Camp-on
This feature provides selected stations or
outside calls with Camp-On capability to a
busy internal station. Two Camp-On methods
are provided. The call waiting method allows a
station or an outside party to camp itself on to
a busy station. The transfer method allows a
transferred outside call to be camped-on to a
busy station.
Centrex Compatibility
A combination of features allows full
integration of the UNIVERGE NEAX 2000 IPS
with Centrex service.
Check In / Check Out
When this feature is activated, the following
operations occur:
• Check In
Room Cutoff is cleared.
• Check Out
Room Status printout is supplied.
Do Not Disturb is reset.
Room Cutoff is set.
Message Waiting is reset.
Automatic Wake Up is cleared.

CID Call Back
When an incoming call is terminated from
station and trunk with Caller ID (calling
number information), and called station does
not answer, the calling number is registered to
the system memory and MW lamp is lit. After
the calling number is registered, by operating
from station, confirmation / delete / call back is
available. Besides, when the called station
answers, the calling number is also registered
to the Last Number Redial memory. The
station can search and call back to that
number by the operations of Last Number
Redial / Stack Dial feature.

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CID Call Routing
This feature allows designating a call
terminating system based on the calling party
number received from the network.
Class of Service
This feature permits all stations to be
assigned a Class of Service in accordance
with the degree of system use desired. The
Class of Service is used to assign restrictions
for trunk access and feature access.
Code Restriction
This feature allows the UNIVERGE NEAX 2000
IPS to be programmed to restrict outgoing
calls according to specific area and/or Central
Office codes. This restriction is controlled on
the basis of a three-digit area code or six-digit
area and office code numbering plan.
Conference (Three/Four Party)
This feature provides a station user the ability
to add-on another party (trunk or station) to a
call already in progress. Single Line
Telephone users can add up to one additional
party and Multiline Terminal users can add up
to two additional parties.

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Conference (Six/Ten Party)
This feature permits a station user or
Attendant (conference leader) to establish a
Conference among as many as six or ten
parties (including the Conference leader).
Conference (32 Party)
This feature permits a station user, Attendant,
or a trunk party to establish a conference
among as many as 32 parties (including the
conference leader). Two Conference methods
are available: Group Call and Meet-Me
Conference.
Group Call
This feature enables a station user (PS,
Multiline Terminal, Single Line Telephone)
within the system or a trunk party to establish
a conference among as many as 32 parties. It
also enables a station user to page a
maximum of 31 parties simultaneously,
excluding the conference leader. Three Group
Call methods are available: Group Call Automatic Conference, Group Call Broadcasting, and Group Call - 2 Way Calling.
Group Call - Automatic Conference

This feature enables a station user to establish a
conference among as many as 32 parties. From a
station or Attendant, a maximum of 31
stations/trunks can be paged simultaneously
except the conference leader.
Group Call - Broadcasting

This feature enables a station user to page a
maximum of 31 parties simultaneously except the
group call leader. After paged parties answer, the
leader can speak to the paged parties (the paged
parties only hear the leader's voice).

Meet Me Conference

This feature enables station users (PS, Multiline
Terminal, Single Line Telephone) within the
system, Attendants, or trunk parties to join a
conference of as many as 32 parties by dialing a
specific access code. The conference participants
are automatically connected to the conference
trunk. Conference participants may call in at
preset time or may be directed to do so by a
conference coordinator.

Consecutive Speed Dialing
For Speed Dialing, all digits are registered as
a Speed Dialing Code. In the case of
Consecutive Speed Dialing, the common
portion of the number is registered as a speed
calling code, and the remaining digits of each
number are dialed by each individual calling
station or by using a Station Speed Dial key
on a Multiline Terminal.
Consultation Hold
This feature permits a station user to hold any
incoming or outgoing CO call, tie line call, or
any intra-office call while originating a call to
another station user within the system.
Customer Administration Terminal (CAT)
In addition to the Maintenance Administration
Terminal (MAT), programming can be done
from selected Multiline Terminals with LCD.
The designated Multiline Terminals can be
placed in program mode, and system data
can then be changed. To prevent
unauthorized changes, password levels are
assigned, providing authorization for access
to certain areas of programming.

Group Call - 2 Way Calling

This feature enables a station user to page a
maximum of 31 parties simultaneously except the
group call leader. After one of the paged parties’
answers, paging becomes 2-way calling between
the leader and the first answered party and
automatically stops paging other parties.

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Chapter 9 Feature Descriptions

Data Line Security
This feature allows line circuits that are used
for data transmission to be protected from
interruptions such as Attendant Camp-On,
Executive Override, and Attendant Override.
Delayed Ringing
This feature enables trunks and station lines
to ring immediately at the terminating station,
but also, after a programmable period of time
has elapsed, to ring at secondary Multiline
Terminals with that trunk or line appearance.
Diagnostics
To assist maintenance personnel, the
UNIVERGE NEAX 2000 IPS provides diagnostic
capabilities such as fault code generation,
device status information and alarm
information recording which can be accessed
from the Maintenance Administration Terminal
(MAT) or Customer Administration Terminal
(CAT).
Dial By Name
This feature allows a Multiline Terminal user
to search for a desired number by name. The
number and name are registered in the
system and they are shown on Multiline
Terminal LCD. The Multiline Terminal user
can search for the desired number by name
using up or down soft keys. When the
Multiline Terminal user finds the desired
number, the call can be originated by pressing
the Line/Trunk key or going off hook.
Dial Conversion
The system can be assigned to use rotary
Dial Pulse (DP) or Dual Tone Multi-frequency
(DTMF) trunks and stations. This feature
provides for the repeating of digits dialed by
the station user onto the C.O. trunks.

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This feature allows an incoming call on a DID
trunk or a tie line to automatically be Camped-On
to the destination station if the destination station
is busy.
DID Digit Conversion

This feature allows the UNIVERGE NEAX 2000 IPS
to convert the digits received from the serving
C.O. to valid station numbers when the C.O.
numbering plan differs from the desired station
numbering plan.
DID Name Display

This feature allows name assignment for a DID
number received from a public network, and
displays the name on an LCD of a Multiline
Terminal or Attendant Console.

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Direct Inward System Access (DISA)
This feature allows an outside caller to access
the system using an exchange network
connection without Attendant or station
assistance. The outside user may originate
calls over any or all of the system's facilities
such as WATS, FX, Tie Line or CCSA. The
outside user can also directly call stations and
access miscellaneous trunks for such features
as dictation access.
Call Forwarding Set by DISA

This feature allows an outside caller to set Call
Forwarding - All Calls by using Direct Inward
System Access (DISA) code.

Direct Inward Termination (DIT)
This feature automatically routes incoming
network exchange calls directly to a preselected station without Attendant assistance.
The call can then be processed by the called
party. Three-party Conference, Call Transfer,
etc., are handled in the same manner as any
normal trunk call.
Direct Outward Dialing (DOD)
This feature permits any station user the
ability to gain access to the exchange network
by dialing an access code and receiving new
dial tone. The user may then proceed to dial
the desired exchange network number.

Direct Station Selection/Busy Lamp Field
(DSS/BLF) Console
This feature allows a DCU-60-1 unit
associated with a legacy Multiline Terminal to
be used as a Direct Station Selection/Busy
Lamp Field (DSS/BLF) Console. When the
buttons on the DCU-60-1 unit are
programmed for Direct Station Selection
(DSS) buttons, up to 60 stations can be
directly accessed in addition to those already
appearing on the Multiline Terminal. Busy
status for each station is indicated by a red
LED associated with each button. In addition,
the DSS console can provide the following
functions:
• Message Waiting - Set/Cancel/Status
Display
• Do Not Disturb - Set/Cancel/Status
Display
• Automatic Wake Up No Answer - Status
Display/Cancel
• Agent Busy Out - UCD - Status Display
• Line Lockout - Status Display
• Room Cutoff - Set/Cancel/Status
Busy Out Status Console

This feature allows a DCU-60-1 unit associated
with a Multiline Terminal to be used as a Busy Out
Status Console. This feature is activated by use of
a Function Mode key on a DSS/BLF Console.
Busy Out Status for each station is indicated by a
red LED associated with each button.
Do Not Disturb Console

This feature allows a DCU-60-1 unit associated
with a Multiline Terminal to be used as a Do Not
Disturb (DND) Console. This feature is activated
by the use of a Function Mode key on a DSS/BLF
Console. DND set status for each station is
indicated by a green LED associated with each
button. In addition, the Multiline Terminal user can
set/cancel the DND status of other stations using
the DND Console.

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Message Waiting Console

This feature allows a DCU-60-1 unit associated
with a Multiline Terminal to be used as a Message
Waiting (MW) Console. This feature is activated
by the use of a Function Mode key on a DSS/BLF
Console. The Message Waiting status for each
station is indicated by a green LED associated
with each button. In addition, the Multiline
Terminal user can set/reset MW status using the
MW Console.
Room Cutoff Console

This feature allows a DCU-60-1 unit associated
with a Multiline Terminal to be used as a Room
Cutoff Console. This feature is activated by the
use of a Function Mode key on a DSS/BLF
Console. The Room Cutoff status for each station
is indicated by a green LED associated with each
button. In addition, the Multiline Terminal user can
set/cancel Room Cutoff to another station using
the Room Cutoff Console.
Wake Up No Answer Console

This feature allows a DCU-60-1 unit associated
with a Multiline Terminal to be used as a Wake Up
No Answer (WU) Console. This feature is
activated by a function mode key on a DSS/BLF
Console. The No Answer status for each station is
indicated by a flashing green LED associated with
each button.
Distinctive Ringing

This feature provides Distinctive Ringing
patterns to the station so that the station user
can distinguish between internal and external
incoming calls. This feature also enables the
LED associated with the line key of the
Multiline Terminal to flash in two colors
according to the kind of incoming call.

Page 9- 18

Do Not Disturb
This feature restricts incoming calls to a
station and can be set by an individual station
or from the Attendant Console. Placing a
station in Do Not Disturb (DND) does not
prevent a station from originating a voice or
data call or from receiving a data call. This
feature also allows a station to ensure privacy
from telephone interruptions while on an
outgoing call. Additionally, the Attendant
Console can place a group of stations in the
Do Not Disturb condition.
Do Not Disturb – Group

This feature allows the system to schedule to
set/cancel Do Not Disturb for a group of stations at
appointed time. The system has up to four
patterns of timetable, and each timetable has time
to set/cancel Do Not Disturb in a day (the time is
programmable by the system). The timetable can
be assigned for day of the week. The different
timetable can also be assigned for specific dates
of the year.
Do Not Disturb - Hotel/Motel

This feature allows the Attendant Console(s),
Hotel/Motel Front Desk Instrument(s), guest
stations or Property Management System (PMS)
terminal(s) to place individual stations into Do Not
Disturb. Calls can be placed from stations set in
DND.
Do Not Disturb-System

This feature simultaneously restricts incoming
calls to a pre-assigned group of stations by
operation from the Hotel/ Motel Front Desk
Instrument(s). Attendant Console(s) and
Hotel/Motel Front Desk Instruments can use the
DND OVR key to override this Do Not Disturb
setting.

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Dterm Assistant
Dterm Assistant is Web-based software,
which resides on the server and provides end
users with the ability to maintain their Dterm
Multiline Terminals and the 2000 IPS
telephony features such as Speed Dialing
from their Web-enabled PCs. The Dterm
Assistant operates in a client – server
environment and can manage multiple 2000
IPS systems over a Local Area Network
(LAN)/Wide Area Network (WAN).
DtermIP
DtermIP is an IP-based Multiline Terminal,
which provides a built-in capability of peer-topeer IP communications. The UNIVERGE
NEAX 2000 IPS system provides the DtermIP
with same IP communications capabilities of
an IP Enabled Dterm (The IP Enabled Dterm
is a Dterm Multiline Terminal with an add-on
IP adapter unit).
Elapsed Call Timer
This feature provides a display of the elapsed
time while a Multiline Terminal with LCD is
connected to any trunk.
Enhanced 911
This feature allows the PBX to transmit a
caller’s emergency service identification
information to an Enhanced 911 Emergency
system.
Executive Calling
This feature allows a station to be assigned a
VIP class. This provides special ringing to a
called station when that station is idle, and
automatic sending of three tone bursts to a
called station when that station is busy,
provided the call was originated from a station
assigned as VIP class.

Executive Override
This feature allows selected users to override
a busy condition on a called station. A
warning tone is transmitted to both stations in
the busy call before the busy condition is
overridden, and a three-party Conference is
then established.
External Paging with Meet-Me
This feature allows a station user or attendant
dial-access to local voice paging equipment
and connects both parties automatically after
the paged party has answered the page by
dialing an access code.
Fax Arrival Indicator
When a call from a C.O. line (Direct-InwardTermination, Direct-Inward-Dialing,
Automated Attendant), station or tie line has
terminated to a facsimile machine, a related
lamp on a pre-designated Multiline Terminal is
caused to light, indicating reception of a
facsimile call.
Fax over IP
This feature allows the system to transmit
facsimile communications over IP network, via
Local Area Networks (LAN) and corporate
Wide Area Network (WAN). Since PBX
regards facsimile equipment as one of
ordinary telephones, IP Packet
Assembler/Disassembler (IPPAD) and Voice
Compression Trunk (VCT) are required for
facsimile uses over IP network same as
legacy stations. The facsimile transmission
procedure (T.30 or G.711/G.726 passthrough) is supported with IP-PAD/
VCT.

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Feature Activation from Secondary
Extension
This feature allows the Multiline Terminal user
to access an appearance of another extension
and program certain features from that
extension.
Flexible Line Key Assignment
Multiline Terminals can have any desired linekey assignment. This feature permits
assignments to be tailored to each individual's
needs. (The terminal's primary extension line
appearance is the only line key that cannot be
reassigned.)
Flexible Numbering Plan
The UNIVERGE NEAX 2000 IPS has a Flexible
Numbering Plan. All access codes and station
numbers and can be assigned in system
programming. Refer also to the Single Digit
Dialing Features and Specifications, which
further increases the flexibility of the system.
Flexible Ringing Assignment
This feature allows lines on Multiline
Terminals to be individually programmed to
ring or not ring.
Forced Account Code
This feature forces the user to enter an
Account Code (up to 8 or 10 digits) for all
outgoing calls. The Account Code must be
dialed before dialing the outgoing number.
Calls are processed only when the dialed
Account Codes are valid.

Group Call Automatic Conference (6/10 Party)
This feature allows a Multiline Terminal user
or single line telephone user within the system
to establish a conference among as many as
six or ten parties. From a Multiline Terminal
/Single Line Telephone, a maximum of 9
stations can be paged simultaneously plus the
originator. The stations are assigned to the
simultaneous paging groups as participants
by the system data beforehand.
Group Call - 2 Way Calling
This feature allows a Multiline Terminal/Single
Line Telephone to page a maximum of fifteen
parties simultaneously including the originator.
After one of paged parties answers, the
paging becomes the 2 Way Calling between
the originator and the first answered party,
automatically stops paging other parties. The
stations are assigned to the simultaneous
paging groups as participants by the system
data beforehand.
Group Call by Pilot Number Dialing
This feature allows a station user (Multiline
Terminal / Single Line Telephone / PS) or a
trunk party to page a group of stations
simultaneously by dialing a pilot number. The
maximum of 32 stations can be assigned to a
paging group, and the paging group is
associated with the pilot number. After one of
the paged stations answers, the paging
becomes a 2 way calling between the calling
party and the first answered station and
automatically stops paging other stations.
Group Listening
When a Multiline Terminal user makes a call
using the handset, pressing the SPKR key will
allow others to listen through the built-in
speaker of the Multiline Terminal. The user
may continue talking on the handset at the
same time.

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Hands-free Answerback
This feature allows the station user to answer
a voice call without lifting the handset.
Hands-free Dialing and Monitoring
This feature allows the station user to dial or
monitor a call without lifting the handset.
Hold
This feature permits a user to Hold a call in
progress. After Hold has been set, the station
user can make or answer new calls.

Hotel/Motel Front Desk Instrument
A Multiline Terminal with LCD can be
programmed to function as a Hotel/Motel
(H/M) Front Desk Instrument. This can be
used to set and cancel standard H/M features
such as Message Waiting, Do Not Disturb,
Automatic Wake Up, and Room Cutoff.
Hotline – Inside/Outside
This feature causes the terminal to place a
call to another station or to an outside party
automatically when the user selects the
Hotline extension.

Call Hold

This feature permits a user to Hold a call in
progress by sending a hookflash and dialing the
Call Hold feature access code, or by pressing the
Call Hold key. This line can then be used for
originating another call or returning to a previously
held call.
Dual Hold

This feature permits a station user who is placed
on Hold by another station to place that station on
Hold also.
Exclusive Hold

This feature allows a Multiline Terminal user to
place a call on Hold and to exclude all other
station users from retrieving the held call.
Non-exclusive Hold

This feature allows a Multiline Terminal user to
place a call on Hold that may be retrieved by any
station that has an appearance of the held line.
Hotel/Motel Attendant Console

The Attendant Console can be programmed
to function as a Hotel/Motel Attendant
Console. In addition to the business features
and functions of the Attendant, the
Hotel/Motel Attendant Console can set Room
Cutoff (individual and group), Automatic Wake
Up, Message Waiting, and Do Not Disturb
(individual and group).

House Phone
This feature allows selected stations to reach
the Attendant simply by going off-hook.
Individual Attendant Access
This feature permits a user to call a specific
Attendant by dialing an Attendant call code.
Intercept Announcement
This feature provides the automatic
interception of Direct Inward Dialing (DID) and
Tie Line calls which cannot be completed due
to unassigned station or level. The caller
hears a recorded Intercept Announcement
that informs the caller that an inoperative
number was reached, and may supply the
number for information.
Intercom
Three types of Intercoms are available:
Manual Intercom, Automatic Intercom, and
Dial Intercom. Each type of Intercom provides
access to a small group of Multiline Terminals
with simplified calling methods.

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Chapter 9 Feature Descriptions

Manual Intercom

The Manual Intercom groups have up to six
Multiline Terminals sharing a common signal path.
Users can call other members of the Manual
Intercom group by pressing a Manual Intercom
key; each press sends a tone burst over the
speakers of all the terminals in the group. When
another user answers the call, a speech path is
activated.
Automatic Intercom

Automatic Intercom provides a path for Voice
Announcement Calls with Handsfree Answerback
between two Multiline Terminals using a line key.
Private conversations can be held by using the
Multiline Terminal handsets. The Busy/Idle status
of the associated Multiline Terminal is displayed
on the Automatic Intercom line key LED.
Dial Intercom

Dial Intercom comprises up to 10 Multiline
Terminals which can call each other using a
dedicated Dial Intercom line key with abbreviated
dialing. Dial Intercom calls can be Voice
Announce with Handsfree Answerback or ringing
calls.

Internal Tone/Voice Signaling
Multiline Terminals can signal incoming
internal calls by Voice Announcement or by
ringing according to the programmed mode
(Voice first or Ring first) of the called terminal.
The caller can dial the digit 1 to change from
Voice Announcement to Ring Tone or vice
versa. The Multiline Terminal assigned this
feature can program the following two modes:
• Voice Mode: allows an incoming call to terminate
with Voice Announcement.
• Tone Mode: allows an incoming call to terminate
with ringing.

Internal Zone Paging with Meet-Me
This feature allows the Attendant Console and
system users to page over the built-in
speakers of the Multiline Terminals within the
assigned zone or all zones.

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IP Enabled Dterm
This feature provides a Dterm Series E/i
Terminal combined with an IP Adapter unit
with the capability to connect UNIVERGE NEAX
2000 IPS via IP networking (IP Enabled
Dterm). The IP Enabled Dterm provides users
with all features currently available in Dterm
Series E terminals.
Last Number Redial
This feature allows users to redial the last
station-to-station or outside number they
dialed using a feature access key or a feature
access code. This is useful when the called
station is busy or does not answer.
Least Cost Routing - 3/6 Digit
This service feature allows the UNIVERGE
NEAX 2000 IPS to be programmed to route
outgoing calls over the most economical
facility (WATS, FX, DDD). Based on the
individual area code and office code dialed (6digit analysis), the system examines the
programmed tables and uses the trunk in the
order specified.
Line Lockout
This feature automatically releases a station
from the common equipment if the station
remains off-hook for longer than a
programmed interval before dialing. Howler
tone may be programmed to be sent to the
station in Line Lockout.
Line Pre-selection
This feature provides the station user with two
ways to select an idle, held, recalling, or ringing
line before going off-hook.

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Maintenance Administration Terminal
The Maintenance Administration Terminal
(MAT) is a personal computer (PC) that
provides an interface to the PBX via the
UNIVERGE NEAX 2000 IPS CPU card. The
MAT PC must have the MATWorX IPS
program properly installed to communicate
with the PBX. MATWorX is required for
system software registration, activation and
system data backups.
Message Center Interface (MCI)
This feature provides an interface with a
customer supplied Voice Mail System (VMS)
which can send Message Waiting lamp
control data to the UNIVERGE NEAX 2000 IPS
using an RS-232C interface.
Message Registration
This feature provides output from the system
to a call accounting system using an RS-232C
interface. This allows the Hotel/Motel clerk to
retrieve the information needed to charge for
local and toll calls.
Message Reminder
This feature allows a user or Attendant to turn
on the message waiting (MW) lamp of a
Single Line Telephone, or the Message
Reminder (MSG) LED of a Multiline Terminal
(if assigned).
Message Waiting
This feature allows the Attendant Console,
Hotel/Motel (H/M) Front Desk Instrument,
administrative station, or Property
Management System (PMS) terminal to light a
lamp (on an uninterrupted or interrupted
basis) on a Single Line Telephone or Multiline
Terminal to indicate a message is waiting.
In addition to the lamp indication control, this
feature also provides the Voice Message
Waiting service that an originating station user
can set to Message Waiting with a
prerecorded message by using the Digital
Announcement Trunk card (PN-2DATA).

Miscellaneous Trunk Access
This feature allows the connection of various
types of external facilities. In addition to Loop
and Ground Start Trunks, the following can
also be interfaced with the UNIVERGE NEAX
2000 IPS: CCSA Lines Code Calling
Equipment, Dictation Equipment, Foreign
Exchange (FX) Lines, Radio Paging
Equipment, and Wide Area Telephone
Service (WATS) lines. Refer to separate
features, Direct Inward Dialing (DID), and Tie
Line Access for more applications of
Miscellaneous Trunk Access.
CCSA Access

This feature allows connection to or from a
Common Control Switching Arrangement (CCSA)
network. A CCSA network is a special, privatelyleased network constructed for one customer's
exclusive use that replaces or augments the public
switched network services.
Code Calling Equipment Access

Code Calling Equipment consists of external
paging units and external dialers requiring dial
access from the UNIVERGE NEAX 2000 IPS.
Dictation Equipment Access

This feature permits dial access to customer
provided Dictation Equipment, and in some

instances allows them to maintain telephone
dial control of normal dictation system
features.
Foreign Exchange (FX) Access

An FX line is a line that is extended and
terminated at a distant Central Office. With this
feature, outgoing calls over the FX line become a
local call at the distant C.O.
Radio Paging Equipment Access

This feature provides station users dial access to
Radio Paging Equipment, and to selectively tone or voice/ tone-alert individuals carrying pocket
paging devices. The paged party (when on
premises) can be connected to the paging party
by going to the nearest station and dialing an
answer back code.

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Chapter 9 Feature Descriptions

Wide Area Telephone Service (WATS) Access

This feature allows any station user direct dial
access to outgoing WATS lines.

Mobility Access
This feature allows a user to make/receive
telephone calls from a remote location using a
Mobile/Cell Phone via the PBX, as if the user
was at his/her desk phone.
Modem over IP
This feature allows the system to transmit
modem communications over IP network, via
Local Area Networks (LAN) and corporate
Wide Area Network (WAN). IP Packet
Assembler/Disassembler (IPPAD) and Voice
Compression Trunk (VCT) are required for
modem over IP network. The modem
transmission procedure (G.711/G.726 passthrough) is supported using the 8IPLA/24IPLA
IP-PAD only.
MP Program Download (FTP)
UNIVERGE NEAX 2000 IPS provides Online MP
Program Download via IP network. The IPS
downloads the MP upgrade program from
FTP server using MATWorX. Immediate or
scheduled changeover to the upgrade
program is available. It is also possible to
change back to the previous program that was
in use before the changeover (changeback).
Multiple Language Display
A language displayed in the LCD of a Multiline
Terminal and Attendant Console can be
selected on a per-station basis or on a peroperator console basis. The available
languages are the same as used on a systemwide basis.

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Multiline Terminal Attendant Position
A Multiline Terminal with LCD can be
programmed to function similar to an
Attendant position. This Attendant position
has limited access to Attendant related
features and functions and can be substituted
where an Attendant is required but an
Attendant Console is not necessary. When an
DCU-60-1 unit is associated with this
Attendant Multiline Terminal enhanced
operation is available.
Music on Hold
This feature plays music when a line is placed
on hold. Music is provided by a circuit board
memory chip, IP adpter, or a local music
source, such as a CD player or a radio.
Night Service
This feature provides a variety of methods for
handling incoming calls when the system is in
night mode. These include:
• Attendant Night Transfer
• Call Rerouting
• Day/Night Mode Change by Attendant
Console
• Day/Night Mode Change by Station Dialing
• Day/Night Mode Change by System Clock
• Night Connection-Fixed
• Night Connection-Flexible
• Trunk Answer Any Station (TAS)
• Overflow for TAS Queue
• Overflow Limit for TAS

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Attendant Night Transfer

When the Attendant Console is in Night Service,
any operator directed calls (dial 0 calls) are
automatically routed to a preprogrammed station.
Priority Calls and Off-Hook Alarms which
terminate to an Attendant are also routed by this
feature.
Call Rerouting

This feature provides flexible reroute capabilities
for a variety of calls when the system is in night
mode.
Day / Night Mode Change by Attendant Console

This feature provides activation of DAY/NIGHT
Mode Change by depressing a predetermined key
from the Attendant Console.
Day / Night Mode Change by Station Dialing

This feature allows selected stations to activate a
change from day mode to night mode by dialing a
special code.
Day / Night Mode Change by System Clock
This feature provides automatic activation of DAY/NIGHT Mode
Change by using System Clock.
Night Connection - Fixed

This feature allows incoming calls normally
terminated to the Attendant to reroute to a
predetermined station when the system is placed
in Night Service.
Night Connection - Flexible

This feature provides incoming calls normally
terminated to the fixed night station to be Call
Forwarded to another station.
Trunk Answer Any Station (TAS)

This feature allows any station, other than one
with incoming restrictions, to answer incoming
calls when the system is in the night mode. When
this feature is activated, incoming exchange
network calls will activate a common alert signal at
the customer premises. By dialing a specified
code, any station may answer the call and then
extend it to any other station by means of the Call
Transfer feature.

Overflow for TAS Queue

If the TAS call is not answered by
predetermined time, the call will be forwarded
to predetermined station/Attendant
Console/Announcement Service.
Queue Limit for TAS

When a DID call is converted to TAS and the
number of using lines reaches queue limit,
this feature provides the system to restrict the
next call terminating.
No CID Call Routing
This feature allows designating a call
terminating system based on the reason for
the absence of calling party number received
from the network.
Off-hook Alarm
This feature allows a station user to call the
Attendant, or a pre-designated station, by
simply staying off-hook for a preprogrammed
period of time. The calling number is
automatically displayed at the Attendant
Console, or the pre-designated station if
equipped with an LCD.
Off-Premises Extensions
This feature allows the connection of a single
line telephone in an off-premises location. The
connection to the Off-Premises Extension can
be through direct copper or through the local
telephone company.
Open Application Interface (OAI)
Provides a computer-to-PBX interface,
allowing a computer to control the function of
the UNIVERGE NEAX 2000 IPS. The IPS can
be customized to accommodate most
customer applications. Application software
can be provided by NECAM, an outside
software house, or a customer.
.

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Chapter 9 Feature Descriptions

Optical Interface
The Optical Interface Card (PN-M10) provides
an internal optical fiber modem to the T1/E1
or Remote PIM network. The PN-M10
provides a bandwidth of 6.3 Mbps and
supports a maximum distance of 10
kilometers (or 6.25 miles) point to point. The
UNIVERGE NEAX 2000 IPS only supports
Single Mode optical fiber
Pad Lock
This feature temporarily restricts telephones
from making unauthorized calls by dialing
special access code when station users are
away from their seats.
Periodic Time Indication Tone
This feature provides a periodic tone to the
station user who has made an outgoing call.
This feature can be allowed or denied for
each station.
Pooled Line Access
A line key can be assigned to access Pooled
Lines. Each line key will allow incoming,
outgoing, or both-way access to a trunk route.

Direct Privacy Release

This feature allows a station user with a secondary
appearance of another extension in the system to
access that extension when it is being used by
someone else. This feature allows for a simplified
method for establishing a conference. In addition,
this feature can be used to emulate PC dialing,
where a single line extension connected to a PC
can appear on a Multiline Terminal and be
accessed by the Multiline Terminal user after the
PC is completed dialing.
Manual Privacy Release

This feature allows a Multiline Terminal user to
enter a conversation on a busy line button if the
Multiline Terminal user already in the conversation
allows them by releasing Privacy.

Private Lines
Only a C.O. trunk assigned to that specific
station is seized when a station user
originates an outgoing C.O. call or when an
incoming C.O. call is terminated at the station
designated by Direct-In-Termination. In this
manner, stations and C.O. trunks are to be
associated on a 1-to-1 basis.

Power Failure Transfer
This feature provides for specified trunks to be
automatically connected to designated Single
Line Telephones in the event of AC power
loss. It is normally used when the system is
not equipped with reserve power.

Property Management System Interface
The UNIVERGE NEAX 2000 IPS provides a
data interface to a locally provided Property
Management System (PMS). This enables
communication between the UNIVERGE NEAX
2000 IPS and the PMS in order to provide
computer control of Hotel/ Motel features.

Priority Call
This feature allows the Attendant to answer a
call before other calls, at the Attendant's
discretion.

Proprietary Multiline Terminal
The are several Multiline Terminals available
which can be used with the UNIVERGE NEAX
2000 IPS.

Privacy
This feature restricts Multiline Terminal users
from depressing a busy line button and
entering a conversation unless permitted by
the Multiline Terminal user currently on that
line button or if the line button is assigned for
Direct Privacy Release.

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•
•
•
•

Legacy: Dterm Series i
Legacy: Dterm Series E (Display/Non-Display)
Dterm IP: Dterm Series E/Series i (Display)
Legacy: Electra Elite(Display/Non-Display)

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Automatic Idle Return

Mute Key

This feature returns a station to the idle state after
3 seconds of reorder tone is received due to the
distant end disconnecting.

This feature allows the distant extension user, of a
station user that presses a mute key during
conversation, not to hear the station user's voice
though the station user can hear the distant
extension user's voice. By pressing the mute key
again, the mute status returns to original
conversation.

Called Station Status Display

This feature provides a display on the status of a
called station on the LCD of the calling Multiline
Terminal.
Calling Name and Number Display

This feature provides a display on the LCD of the
Multiline Terminal receiving a call, indicating the
station number/trunk number of the incoming call.
Group Feature Key

A station user can monitor the status of group
member stations such as idle, busy, hold and
ringing by lamp indication of the multiple line keys.
If a line is idle, a station user can make a call to
the member station by pressing the associated
line key (instead of pressing DSS key). If a line is
ringing or is on hold, a station user can pick up the
incoming call or the call on hold by using My Line,
and the ringing or held line is made idle.

My Line Number Display

This feature allows the display of the Prime
Extension in the LCD display of the Multiline
Terminal during an Idle state.
Preset Dialing

This feature allows a Multiline Terminal user to
prepare and verify a number in the display on the
LCD before dialing. When a wrong number is
entered, the user can correct the number before
originating the call.
Prime Line Pickup

This feature allows a Multiline Terminal user to go
off hook and originate a call from the line assigned
as the Prime Line without depressing the
associated line key.

Dynamic Dial Pad

This feature allows a user to make an outgoing
call by first pressing a key on the keypad of
Multiline Terminal without pressing the Speaker
key or going off-hook.
Handsfree Unit

The built-in Handsfree Unit enables full Handsfree
operation for both internal and external calls (No
optional Handsfree Unit is required).
I-Hold / I-Use Indication

Multiline Terminals provide indication of which line
keys have been placed on Hold, or are in use by
that Multiline Terminal. The LED associated with
the line key will give the appropriate indication.
Microphone Control

All Multiline Terminals are equipped with a
Microphone Control button with an associated
LED.
Multiple Line Operation

This feature allows for the appearance of multiple
lines on the Flexible Line Keys and feature keys of
all Multiline Terminals.

Recall Key

Each Multiline Terminal is equipped with a Recall
Key that is used to generate a hookflash to access
features provided by the outside exchange, or to
abandon a call while retaining the line for
origination of another call.
Relay Control Function Key

This feature provides a Multiline Terminal with the
ability to activate/deactivate relays (on a PNDK00) to control external devices.
Ring Frequency Control

The ring frequency of the Multiline Terminal can
be controlled on a station basis in system
programming (eight ro fourteen frequencies are
available) or by use of a function key on the
Multiline Terminal.
Ringing Line Pickup

This feature provides the ability to answer any call
ringing into a Multiline Terminal by just lifting the
handset.

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Chapter 9 Feature Descriptions

Soft Keys

According to the status of the Multiline Terminal,
function keys (Soft Keys) are displayed in the third
line on the LCD. If the status of Multiline Terminal
changes, the Soft Keys will change automatically.
Also if the Help key is pressed, explanation of
indicated Soft Keys are shown on the LCD.
Volume Control

Multiline Terminals are equipped with common
Volume Control keys for:
•
•
•
•
•

Built-in Speaker / Handset Receiver Volume
Ring Volume
C.O. Transmission Level
LCD Contrast
Ring Tone Frequency

Remote Hold
This feature allows a Multiline Terminal user
or an attendant to place a call on hold on the
line button of the destination terminal, by
pressing the Hold key.
Remote PIM over IP
When IPS DMR or 2000 IPS PIM are installed
at remote site, and connected to a 2000 IPS
or IPS DM at main site over IP network, the
Main Site system controls and maintains the
remote DMR or IPS PIM operation as one
single system. If a communication failure
occurs between the Main Site and Remote
Site, the Remote Site automatically changes
over to a survival mode and operates as a
stand-alone system.
IPS DMR: IPS Distributed Model Remote (with
CP31-A/B/C)
IPS DM: IPS Distributed Model (with CP24-A/B/C)

Reserve Power
This feature provides backup power from a
24V battery source in the event of a
commercial power failure.

Page 9- 28

Resident System Program
This feature provides the installers a simple
procedure to have the system generate
system data according to the system
hardware configuration, thereby providing
immediate operation and shorter
programming time. When activated, the
system scans hardware configuration (such
as line/trunk card slot location) and assigns
system data (such as extension numbers,
trunk numbers, etc.) according to a
predetermined generic program assignment.
Return Message Schedule Display
This feature permits any station user to
register his Return Schedule from his phone
when he leaves his desk or the premises, and
have the Return Schedule displayed on a
calling Multiline Terminal with a Liquid Crystal
Display (LCD) during his absence.
Room Cutoff
This feature allows the Attendant Console,
Hotel/Motel (H/M) Front Desk Instrument, or
Property Management System (PMS)
terminal, or guest room telephones using a
special access code, to temporarily restrict
guest room telephones from making
unauthorized calls when guests are away
from their room, and when rooms are in
Check Out status.
Room Status
This feature provides the Hotel/Motel (H/M)
Front Desk Instrument with a visual display of
the guest's room status. A supplementary
print out (individual and summary) can be
provided.
Route Advance
This feature automatically routes outgoing
calls over alternate facilities when the first
choice trunk group is busy. Users select the
first choice route by dialing the corresponding
access code, and the equipment then
advances through alternate trunk groups only
if the first choice is busy.

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Save and Repeat
This feature allows a Multiline Terminal to
save a specific dialed number and then redial
that number at a later time.
Security Alarm
This feature provides an indication on the
Attendant Console when a contact closure
occurs.
Set Relocation
This feature enables two stations to be moved
from one location to another without
reprogramming station data at MAT.
Single Digit Dialing
This feature provides the station user the
ability to dial single digit codes to access
certain features while still allowing the same
digit dialed to be used as the first digit of
station number.
Single Digit Feature Access Codes
This feature allows stations to access certain
other system features by the direct dialing of a
Single Digit Access Code, while receiving
Busy Tone or Ring Back Tone.
SIP Trunk Interface
2000 IPS supports Session Initiation Protocol
(SIP) based connections
SNMP
Simple Network Management Protocol
(SNMP) is a standard protocol for TCP/IP
network management, which enables a
network management application software to
query a management agent (network device
such as router, PC host, and hub) using a
supported MIB (Management Information
Base). The MIB is a database of network
performance information that is stored on the
network devices. The UNIVERGE NEAX 2000
IPS can support the SNMP standard MIB
(MIB-II, defined in IETF RFC 1213) and
private MIB and TRAP. This feature also
enables the network management system
(SNMP manager) to manage the IPS via
Network Address Translation (NAT).

Software Line Appearance
(Virtual Extensions)
This feature permits assignment of circuits
which do not physically exist, to be used as
secondary extensions on Multiline Terminals.
There are a possible 1020 virtuals and can be
calculated by 1020 – number of Multiline
Terminals (Dterm/DtermIP)=Total Virtuals
available.
Stack Dial
This feature enables a Multiline Terminal or
an Attendant Console to remember the
numbers dialed in the preceding five calls,
including the last number dialed. The stack
dial numbers are sequentially displayed on
the LCD display, thus allowing the station user
to make an outgoing call by selecting the
desired dialed number from the display.
Station Hunting
Three Station Hunting arrangements are
available. Station Hunting - Circular processes
the call no matter which station in the hunt
group is called. Station Hunting - Terminal
initiates a hunt only when the pilot number of
a hunt group is called. Station Hunting Secretarial is initiated when a busy station in a
Station Hunting - Circular group or Station
Hunting - Terminal group is reached.
Station Hunting - Circular

When a busy station in a hunt group is called, this
feature permits the call to be processed
automatically through the hunt group in a
preprogrammed order from that station's position
within the hunt group.
Station Hunting - Terminal

When a pilot number is dialed and that number is
busy, sequential Station Hunting will be initiated.
However, if a number other than the pilot number
is dialed and that number is busy, busy tone will
be provided rather than initiate Station Hunting.

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Chapter 9 Feature Descriptions

Station Hunting - Secretarial

System Speed Dialing
This feature provides all users the ability to
dial frequently called numbers using an
abbreviated call code.
Station Message Detail Recording (SMDR) System Traffic
The UNIVERGE NEAX 2000 IPS is the stored
This feature provides a call record for
outgoing station-to-trunk calls, incoming trunk- program controlled digital electronic PABX
employing PCM time division switching
to-station calls (including Data Call) and
technique. Single stage time division
tandem calls. This facilitates cost control by
switching
is employed for all configurations of
identifying trunk use and misuse by individual
the IPS up to its maximum line and trunk
stations. Station Message Detail Recording
(SMDR) enables call billing to customers and configuration. Traffic capacity is expressed in
Busy Hour Call Attempts (BHCA). This value
clients, and provides a means for checking
is the number of call attempts that can be
local telephone bills.
generated in the busiest hour of the day.
Station Speed Dialing
Tenant Service
This feature allows a station user to dial
This feature provides for more than one
frequently called numbers by dialing an
organization (tenant) to share the same
access code and an abbreviated code, or by
UNIVERGE NEAX 2000 IPS system. Through
depressing a feature key or line key assigned system programming, each organization may
for Station Speed Dialing capability.
be restricted to its own Central Office trunks,
Attendant Consoles and extension group. In
Step Call
addition, incoming calls are directed to the
This feature allows the Attendant or station
user, after calling a busy station, to call an idle specific tenant.
Terminal Login via NAT
station by simply dialing an additional digit.
This feature will operate only if the number of The NEAX 2000 IPS supports network
configuration via a router with NAT. NAT is a
the idle station is identical to that of the busy
technology that translates an internal local IP
station in all respects, except the last digit.
address to a globally unique IP address
Supervisory Control of Peripheral
before sending packets to the outside
Equipment
network. NAT is configured on the router at
When various types of peripheral equipment
the border of the inside network and the
(such as facsimiles, dictation equipment,
outside network.NAT is used for efficient use
Voice Mail, etc.) are connected to the line
of global IP addresses and for security.
circuits of the UNIVERGE NEAX 2000 IPS, this
Tie Lines
feature allows the loop of the line circuit
This feature allows any station user dial
concerned to open for a programmable
access or direct access to an E&M Tie Line.
interval, and send a release signal to the
Tie Line Tandem Switching
peripheral equipment when the calling party
This feature allows trunk-to-trunk connections
disconnects.
through the IPS without the need for any
Attendant assistance or control. The major
System Clock Setup by Station Dialing
use of this feature is in association with a dial
This feature enables a station user to set up the
tandem tie line network to allow tie line
system clock, from Single Line Telephone,
connections and incoming tie line calls
Multiline Terminal, and PS.
automatic access to, and completion of, local
This feature allows assignments to be given to
members of Terminal and Circular Hunting groups
to reroute calls (when their hunting group is all
busy) to a back-up hunting group.

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Central Office calls.
Timed Queue
When a user originates an outgoing trunk call
and the called party is busy or does not
answer, the caller can set the Timed Queue
feature. When this feature is set, the trunk
seizure is repeated and the number is
redialed after a predetermined time interval.
Timed Reminder
This feature allows the system to be
programmed to automatically call stations at
specified times. Upon answering, the station
is connected to a recorded announcement or
music source.
Trunk - Direct Appearances
This feature allows Multiline Terminal users
the ability to access a CO line or E&M Tie
Line without dialing an access code. For this
feature, trunks must be assigned to the line
keys on the Multiline Terminal. Incoming calls
on CO lines can be answered on the
appropriate trunk line appearance.
Trunk Queuing - Outgoing
This allows a station user, upon encountering
a busy signal on a trunk, to dial a feature
access code and enter a first-in, first-out
queue. As soon as an outgoing trunk
becomes available, stations in the queue will
be called back on a first-in, first-out basis.
Trunk-to-Trunk Connection
This feature provides any station user with the
ability to conference together two outside
trunk calls and abandon the connection
without dropping the Trunk-to-Trunk
Connection.

Uniform Call Distribution (UCD)
The Uniform Call Distribution (UCD) feature
permits incoming calls to terminate to a
prearranged group of stations. Calls are
distributed in the order of arrival to idle
terminals within the group, based on which
terminal has been idle the longest period of
time. Stations may log on/log off from the
UCD group. Supervisor stations may monitor
conversations of agents.
Busy In/Busy Out-UCD

This feature allows an agent in a UCD group to log
their station onto or off of the group. This allows
the system to control whether a call directed to the
pilot number of the UCD group goes to that station
or not. This prevents incoming calls from being
directed to stations at which no agent is available.
Call Waiting Indication-UCD

This feature provides a visual indication when an
incoming call to a UCD group is placed in queue,
due to an “all agents busy” condition. An external
relay controlled indicator or an LED on a Multiline
Terminal can be used to provide Call Waiting
Indication.
Delay Announcement-UCD

This feature allows the system to provide a
recorded announcement to an incoming caller
placed in queue to a UCD group. A single
announcement, or two separate announcements,
can be provided.
Hunt Past No Answer-UCD

This feature allows calls targeted at a UCD group
to hunt past an agents station, after a no answer
condition, if the agent forgets to log off of the
group and the agent is unable (or not available) to
answer the call.

Timed Forced Release
This feature will restrict Long duration calls by
forcibly releasing the call after a preset time. The
feature is available for outgoing, incoming and
tandem calls.

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Chapter 9 Feature Descriptions

Immediate Overflow-UCD

This feature allows a call directed to a UCD group
to immediately overflow to another UCD group,
upon encountering an “all agents busy” condition.
Priority Queuing-UCD

This feature allows the system to prioritize
incoming calls by trunk route and on a per station
basis, when the call enters a UCD queue. When a
call is a considered as priority it is placed at the
beginning of the queue.
Queue Size Control-UCD

On incoming DID/Tie line calls, the system can be
assigned a threshold that limits the number of
calls in queue. When the queue size threshold is
exceeded, incoming callers are connected to busy
tone.
Silent Monitor-UCD

This feature provides the UCD group supervisor
with the ability to monitor a call to a UCD agent.
The silent monitor function gives no indication (as
an option) to either the agent or the calling party.

Uniform Numbering Plan (UNP) -Voice and
Data
In the numbering plan for a network to be
configured through the use of Tie Lines, a
Uniform Numbering Plan (UNP) is employed.
When UNP is employed, a station user from
any PBX within the network can call a desired
party by using a uniform dialing method based
on the UNP.

Voice Guide
This feature provides a station user with an
announcement that informs:
1. The result of the operation when the station
user set or canceled the service feature, instead of
service set tone.
2. Which service has been set to the station; such
as, Call Forwarding - All Calls, Do Not Disturb or
Message Waiting, when the station goes off-hook,
instead of special dial tone.

Voice Mail Integration
This feature is used to interface the
UNIVERGE NEAX 2000 IPS with a locally
provided stand-alone type Voice Mail System
(VMS). The VMS, connected to the using
single line circuit (LC), is controlled by
sending/receiving DTMF signals using this
LC.
The VMS’s voice mail feature can be used by
accessing this VMS directly from an
extension. If a station sets its call forwarding
destination to the VMS, calls to this station are
connected to the VMS, and the messages can
be registered according to the VMS
instruction. In addition, the Message Waiting
lamp of the station can be turned on
automatically by the VMS.

Voice Mail Private Password
Voice Mail Password can be prevented from
Variable Timing Parameters
displaying in LCD of Multiline Terminals when
This feature gives the IPS the versatility to
change timing duration using the Maintenance connected to the Voice Mail System.
Administration Terminal (MAT) or the
Customer Administration Terminal (CAT). All
timing parameters are set initially in the
Resident System Program. These timing
parameters can be changed according to the
customer's requirements.

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Voice Mail Transfer
This feature has two functions that provide
streamlined transfer access to voice mail.
1. One touch access to VMS: When an
Attendant transfers an external call to a station, and if the station is busy or unanswered,
the Attendant can transfer the call to a VMS
by dialing “9” or by pressing a function key
provided for this feature.
2. Transferring Camp-On call to VMS: When
an Attendant sets Camp-On to a busy
destination station for an external call, and if
the destination station does not answer by
predetermined time, the call can be
automatically transferred to a VMS.

Voice Over IP (H.323)
Provides the ability for the UNIVERGE NEAX
2000 IPS to connect via IP Trunk to other
non-NEAX systems that are compliant with
H.323. Allows Transparent Dialing and
Calling Station Number Display between the
systems. Gate keeper (GK1000) software
and customer supplied server is required.
VoIP Log Collection
This feature provides Voice over IP (VoIP) log
collection functions for analyzing VoIP related
problems caused by IP device, usage method,
network environment and other effects. A
maintenance person will be able to
understand its cause and figure out how to
solve the problem. The following functions are
supported.
Whisper Page
This feature allows a secretary to interrupt the
boss in a private way. By pressing a feature
key or dialing an Access Code, the secretary
station can voice override the conversation
between the boss and another party (station
or trunk). When the conversation is
interrupted, the boss can hear the secretary
but the other party is unaware of the Voice
Override.

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Chapter 9 Feature Descriptions

CCIS Feature List
Attendant Camp-On with Tone Indication - CCIS
Attendant Controlled Conference - CCIS
Automatic Recall - CCIS
Brokerage - Hot Line - CCIS
Busy Lamp Field (BLF) - CCIS
Busy Verification - CCIS
Call Back - CCIS
Call Forwarding - All Calls - CCIS
Call Forwarding - Busy Line - CCIS
Call Forwarding - Don’t Answer - CCIS
Call Forwarding - Intercept - CCIS
Call Forwarding - Override - CCIS
Call Processing Indication - CCIS
Call Transfer - All Calls - CCIS
Call Transfer - Attendant - CCIS
Called Station Status Display - CCIS
Calling Name Display - CCIS
Calling Number Display - CCIS
CCIS Networking via IP
Centralized Billing - CCIS
Centralized Day/Night Mode Change - CCIS
Centralized E911 - CCIS
Consultation Hold - All Calls - CCIS
Data Line Security - CCIS
Deluxe Traveling Class Mark - CCIS
Dial Access to Attendant - CCIS
Digital Display - Station - CCIS
Digital Display - Trunk - CCIS
Direct-In Termination - CCIS
Distinctive Ringing - CCIS
Do Not Disturb - CCIS
Dual Hold - CCIS
Elapsed Time Display - CCIS
Flexible Numbering of Stations - CCIS
Hands-Free Answerback - CCIS
Hot Line - CCIS
House Phone - CCIS
Incoming Call Identification - CCIS
Individual Attendant Access - CCIS

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LDN Night Connection - CCIS
Link Alarm Display - CCIS
Link Reconnect - CCIS
Message Waiting Lamp Setting - Attendant - CCIS
Message Waiting Lamp Setting - Station - CCIS
Miscellaneous Trunk Access - CCIS
Miscellaneous Trunk Restriction - CCIS
Multiple Call Forwarding - All Calls - CCIS
Multiple Call Forwarding - Busy Line - CCIS
Multiple Call Forwarding - Don’t Answer - CCIS
Multiple Console Operation - CCIS
Network Station Number - CCIS (FCCS)
Night Connection - Fixed - CCIS
Night Connection - Flexible - CCIS
Outgoing Trunk Queuing - CCIS
Paging Access - CCIS
Restriction from Outgoing Calls - CCIS
Service Display - CCIS
Single-Digit Station Calling - CCIS
Station-Controlled Conference - CCIS
Station-to-Station Calling - CCIS
Station-to-Station Calling - Operator Assistance - CCIS
Toll Restriction 3/6 Digits - CCIS
Trunk Answer from Any Station - CCIS
Trunk-to-Trunk Restriction - CCIS
Uniform Numbering Plan - CCIS
Voice Call - CCIS
Voice Mail Integration - CCIS
Voice Mail Live Record - CCIS
Voice Mail Private Password - CCIS

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CCIS Features Descriptions
Attendant Camp-On with Tone Indication – CCIS

This feature permits the Attendant, when the
desired station at another switching office is
busy, to hold an in-coming call in a special
waiting mode. A distinctive Camp-On tone is
sent to the busy station when the Attendant
sets Camp-On. When that station becomes
idle, it is automatically rung and connected to
the waiting trunk party.
Attendant Controlled Conference - CCIS

This feature permits an Attendant (2400 IPX)
to establish a conference, through CCIS, with
up to eight parties of stations and/or trunks
(inside and outside parties).

Busy Verification - CCIS

This feature permits an Attendant, via the
Attendant Console on the 2400 IPX or the
2000 IPS, to interrupt a busy station’s call at
another switching office connected through
CCIS.
Call Back - CCIS

This feature provides inter-office Call Back. A
station that has dialed a busy station at
another office can set Call Back - CCIS by
dialing a feature access code. When this
feature has been set, the setting station will
ring as soon as the busy station becomes
available.

Automatic Recall - CCIS

Call Forwarding - All Calls - CCIS

This service feature works as a time reminder.
When an Attendant-handled call through
CCIS remains on hold, camped-on, or ringing
unanswered for a fixed interval, the Attendant
is automatically alerted.

This feature permits all calls destined for a
particular station to be routed to another
station or to an Attendant Console, in another
office in the CCIS network, regardless of the
status (busy or idle) of the called station. The
activation and cancellation of this feature may
be accomplished by either the station user or
an Attendant.

Brokerage - Hot Line – CCIS

This feature provides a ringdown connection
between two stations, each using a Multiline
Terminal, in different offices in the CCIS
network.
Busy Lamp Field (BLF) - CCIS

This feature provides a busy status indication
of the predetermined stations within the CCIS
network. The visual indication is provided with
a red LED associated with each DSS button
on the DSS/BLF Console and Multiline
Terminal. Pressing the DSS button allows a
direct access to the preprogrammed station
within the CCIS network.

Call Forwarding - Busy Line – CCIS

This feature permits a call to a busy station to
be immediately forwarded to a pre-designated
station or to an Attendant Console in another
office in the CCIS network.
Call Forwarding – Don’t Answer - CCIS

This feature permits a call to an unanswered
station to be forwarded to a pre-designated
station or to an Attendant Console in another
office, when the called station does not
answer after a predetermined time period.

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Chapter 9 Feature Descriptions

Call Forwarding - Intercept - CCIS

Calling Number Display - CCIS

This feature allows calls to an inoperative
number, through a CCIS trunk, to be
intercepted and automatically routed to a
recorded announcement informing the caller
that an inoperative number was dialed and
giving the Listed Directory Number for
information.

This feature permits the number of a calling or
called party at another switching office to be
displayed either on a Multiline Terminal or an
Attendant Console.
CCIS Networking via IP

This feature provides CCIS networks with
Voice over IP (VoIP) capabilities to provide a
Call Forwarding - Override - CCIS
converged infrastructure over corporate Wide
This feature allows a target station user
Area Networks (WAN). The IP Enabled Dterm
(Station A) to call a station (Station B) that has can communicate with other IP Enabled
set Call Forwarding - All Calls - CCIS.
Dterm over the CCIS network (IP based) on a
peer-to-peer connection basis. The legacy
Call Processing Indication - CCIS
terminals (TDM based terminals) can
This feature provides visual indications of all
communicate with other legacy terminals or IP
CCIS calls being processed or waiting
Enabled Dterm terminals over the CCIS
processing at the Attendant Console.
network, via IP Packet Assembler/
Disassembler (IP-PAD). Voice compression of
Call Transfer - All Calls - CCIS
This feature allows a station user to transfer
G.729a (8 kbps) and G.723.1 (5.3 kbps /
incoming or outgoing Central Office, intra6.3 kbps) is available for those connections.
The CCIS Networking via IP provides users
office and inter-office calls to another station
with all TDM-based CCIS functionality, such
in the CCIS network, without Attendant
assistance.
as feature transparency, centralized
management, and centralized facilities.
Call Transfer - Attendant - CCIS

This feature permits a station user, while
connected to a CCIS network call, to transfer
a call to an Attendant Console via the CCIS
network.
Called Station Status Display - CCIS

This service feature provides, on the LCD
display of the calling Multiline Terminal, a
display of the called station status of the
remote office within the CCIS network.
Calling Name Display - CCIS

This feature permits the station name of a
calling or called party at another switching
office, through the CCIS network, to be
displayed either on a Multiline Terminal or an
Attendant Console.

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Centralized Billing - CCIS

This feature is used to collect billing
information from each office within the
network and to direct it to the associated
center office. Billing information is then
forwarded to the central billing centers via
RS232C interfaces.
Centralized Day/Night Mode Change - CCIS

This feature switches the Day/Night mode of a
remote office, linked to the main office (2400
IPX) via CCIS. The mode is switched in
accordance with the Day/Night mode
switching on the Attendant Console at the
main office.

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Centralized E911 – CCIS

This feature allows the system to transmit a
calling party number to the 911 Emergency
system over CCIS tandem connection.
Consultation Hold - All Calls - CCIS

This feature permits a station user, within the
CCIS network, to hold any incoming or
outgoing public network or Tie Line call while
originating a call to another station within the
CCIS network.
Date Line Security- CCIS

This service feature allows the lines, which
are used for data transmission through CCIS
to be, protected from interruptions such as
Attendant Camp - On, Busy Verification CCIS, Executive Right of Way, and Attendant
Override.

Digital Display - Trunk - CCIS

This service feature provides the Attendant
with a visual indication of incoming and
outgoing trunk calls during an attendant-totrunk connection within the CCIS network.
Trunk Group number and trunk identification
code are displayed.
Direct-In Termination - CCIS

This feature automatically routes incoming
exchange calls through CCIS to a preassigned station in the network, without
Attendant assistance.
Distinctive Ringing – CCIS

This feature provides distinctive station ringing
patterns for terminated calls, through the
CCIS network, so that a station user can
distinguish between incoming internal and
external calls.

Deluxe Traveling Class Mark - CCIS

This feature provides outgoing call restrictions
within the CCIS network. The following three
types of restrictions are allowed:
• Deluxe Traveling Class Mark Restriction
• Route Restriction
• Numbering Restriction

Do Not Disturb - CCIS

This feature allows a station user to establish
Do Not Disturb (DND) status on a temporary
basis, during which time access to the station
from CCIS calls will be denied.
Dual Hold – CCIS

Dial Access to Attendant – CCIS

This feature allows a station user to call an
Attendant Console by dialing an operator call
code through the CCIS network.
Digital Display - Station - CCIS

This service feature provides a display of the
station number on the Attendant Console,
during an Attendant-to-station connection,
within the CCIS network.

This feature allows two connected Multiline
Terminals to be placed on hold
simultaneously over the CCIS link. This
enables the held parties to answer or originate
a call from a secondary extension or the idle
primary extension.
Elapsed Time Display – CCIS

This feature provides an LCD, which shows
the duration of time that a Multiline Terminal is
connected to any trunk through the CCIS
network.

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Chapter 9 Feature Descriptions

Flexible Numbering of Stations – CCIS

Link Alarm Display - CCIS

This feature allows voice and data station
numbers to be assigned to any instrument in
the CCIS network, based solely upon
numbering plan limitations.

This feature provides an indication on external
equipment when the CCIS link is
connected/disconnected, when the system is
initialized, or when the CCH is in make-busy.
Link Reconnect - CCIS

Hands-Free Answerback - CCIS

This feature allows a Multiline Terminal station This feature provides the system connected to
CCIS network with the capability to release
user to respond to a voice call, through the
the redundant CCIS link connection and reCCIS network, without lifting the handset.
connect the link within the system for efficient
usage of the CCIS links.
Hot Line – CCIS
This feature allows two stations, at different
Message Waiting Lamp Setting – Attendant – CCIS
nodes in the CCIS network, to be mutually
This feature allows an Attendant, in the 2400
associated on an automatic ringdown basis
IPX, to set or cancel a Message Waiting lamp
through the CCIS network.
indication, through the CCIS network, on a
station in 2000 IPS.

House Phone – CCIS

This feature allows selected stations to call an
Attendant Console, through the CCIS
network, simply by going off hook.
Incoming Call Identification - CCIS

This feature allows an Attendant to visually
identify the type of service and/or trunk group,
which is arriving or waiting to be answered at
the Attendant Console through the CCIS
network.
Individual Attendant Access – CCIS

This feature permits a station user to call a
specific Attendant Console, in the CCIS
network, using an individual Attendant
Identification Number.
LDN Night Connection – CCIS

This feature routes Listed Directory Number
(LDN) calls to a pre-selected station, in the
CCIS network, when the Night mode has
been entered.

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Message Waiting Lamp Setting - Station – CCIS

This feature allows a station user, in the 2400
IPX, to set or cancel a Message Waiting lamp
indication, through the CCIS network, to a
station in 2000 IPS with this feature.
Miscellaneous Trunk Access - CCIS

This feature provides access to all types of
external and customer-provided
equipment/facilities, such as Tie Line and
exchange network, along with Dictation,
Paging Access - CCIS and Code Calling
through the CCIS network.
Miscellaneous Trunk Restriction - CCIS

This feature denies certain stations and dialrepeating tie trunks access to specific trunk
groups, such as Tie Line, exchange network,
Dictation or Paging Access - CCIS through
the CCIS network.

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Multiple Call Forwarding - All Calls - CCIS

This feature allows the last hop of a Multiple
Call Forwarding - All Calls sequence to be
forwarded over a CCIS network to a station in
another office.

Night Connection - Flexible – CCIS

This feature provides an inter-office night
connection service, via the CCIS network,
when the calling station and the night station
belong to different offices.

Multiple Call Forwarding - Busy Line – CCIS

This feature allows the last hop of a Multiple
Call Forwarding - Busy Line sequence to be
forwarded over a CCIS network to a station in
another office.
Multiple Call Forwarding - Don’t Answer - CCIS

Outgoing Trunk Queuing - CCIS

This feature allows a CCIS network station,
upon encountering an all trunk busy signal, to
dial a specified access code and enter a firstin, first-out queue. As soon as a CCIS trunk
becomes available, stations in the queue will
be called back on a first-come, first-served
basis.

This feature allows the last hop of a Multiple
Call Forwarding - Don’t Answer sequence to
be forwarded over a CCIS network to a station
Paging Access - CCIS
in another office.
This feature provides dial access to paging
equipment from an Attendant Console or a
Multiple Console Operation – CCIS
station, through the CCIS network.
This feature provides console operation where
Attendant Consoles are installed in more than
Restriction from Outgoing Calls - CCIS
one node in the CCIS network.
This feature automatically restricts users of
pre-selected stations from placing outgoing
Network Station Number - CCIS (FCCS)
calls and/or certain miscellaneous trunk calls
When 2000 IPS is connected to a 2400 IPX
through CCIS, without Attendant assistance.
via CCIS link, the Network Station Number
can be moved to other office within the
Service Display - CCIS
network by a simple command operation from This feature generates LCD displays on the
the Centralized MAT in the 2400 IPX.
Multiline Terminal corresponding to the
various features as they are initiated.
Night Connection - Fixed – CCIS

This feature routes calls normally directed to
the Attendant Console to a pre-selected
station in another office, through the CCIS
network, when the Night mode has been
entered.

Single-Digit Station Calling - CCIS

This feature allows the assignment of SingleDigit Station numbers.
Station-Controlled Conference - CCIS

This feature allows any station of the 2400
IPX to establish a conference among a
maximum of eight parties of stations and/or
trunks (inside and outside parties) of the 2000
IPS, through CCIS.

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Chapter 9 Feature Descriptions

Station-to-Station Calling – CCIS

Voice Call - CCIS

This feature permits any station user to dial
another station directly, through CCIS, without
Attendant assistance.

This feature provides a voice path, through
the CCIS network, between a Multiline
Terminal in one office and a Multiline Terminal
in another office. This path is established from
the calling party to the called party’s built-in
speaker. If the called party’s MIC lamp is on,
the called party can have a conversation in
hands-free.

Station-to-Station Calling - Operator Assistance – CCIS

This feature allows a station user to call
another station in the CCIS network, with the
assistance of an Attendant Console operator.
Toll Restriction - 3/6 Digits - CCIS

This feature allows the system to be
programmed to restrict outgoing calls, through
CCIS, according to specific Area and/or Office
Codes. This restriction is determined on the
basis of a three-digit Area Code or six-digit
Area and Office Code numbering plan.
Trunk Answer From Any Station - CCIS

This feature allows any station, not restricted
from incoming calls, to answer incoming calls
when the network is in Night mode.

Voice Mail Integration – CCIS

This feature allows any station user in the
CCIS network to utilize the Voice Mail System
(VMS) with the Message Center Interface
(MCI).
Voice Mail Live Record - CCIS

This feature allows a Multiline Terminal user to
record the conversation with another station/trunk
within the CCIS network by the Centralized Voice
Mail. The conversation in the office without Voice
Mail can be recorded to Centralized Voice Mail in
other office via CCIS network.

Trunk-to-Trunk Restriction - CCIS

This feature allows Trunk-to-Trunk tandem
restriction by caller’s information sent from
each office (e.g., caller is a trunk) through the
CCIS network.

Voice Mail Private Password - CCIS

Voice Mail Password can be prevented from
displaying in LCD of Multiline Terminals when
connected to the voice mail system via CCIS.

Uniform Numbering Plan – CCIS

In a CCIS network, a Uniform Numbering Plan
enables a station user to call any other station
in the network. Two alternative numbering
plans are provided. In the first plan, the station
user dials any digit station number from three
to eight. The location of the office is identified
by the first one-, two-, or three-digit of the
station number. In the second plan, the station
user dials a one-, two- or three-digit office
code and any digit station number from two to
eight.

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ISDN Feature List
Addressing
Advice of Charge - Display
Call-By-Call Service Selection
Called Party Recognition Service
(Direct-In Termination (DIT))
Channel Negotiation
Connected Line Identification Presentation (COLP) /
Connected Line Identification Restriction (COLR)
CPN To Network - Present
CPN To Terminating User - Display
DID Addressing
DID and DOD Addressing
Event Based CCIS
Incomplete Number Handling
ISDN Terminal
MEGACOM® Access/WATS
MEGACOM® 800 Service/800 WATS/Ultra WATS
MULTIQUEST®/900 Service
Overlap Receiving
Overlap Sending
Subaddress - Present
Trunk Provisioning Service Selection

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Chapter 9 Feature Descriptions

ISDN Feature Descriptions
Addressing

The UNIVERGE NEAX 2000 IPS supports E.164
numbering plan as Type of Number (TON) /
Numbering Plan Identifier (NPI) combination
in Calling and Called Party Number
Information Element for ISDN calls.
Advice of Charge - Display

When an outgoing connection is made on the
ISDN trunk, this feature provides the call
originator with the charge information form the
network. The information will be displayed on
the calling Dterm LCD for six seconds after
the station has been released, and be output
to the SMDR port.

Connected Line Identification Presentation (COLP) / Connected
Line Identification Restriction (COLR)

Connected Line Identification Presentation
(COLP) provides the 2000 IPS to send the
ISDN number of the connected party in the
2000 IPS, to the calling party. Connected
Line Identification Restriction (COLR)
provides the 2000 IPS to restrict presentation
of the connected party’s ISDN number in the
2000 IPS, to the calling party.
CPN To Network–Present

This feature allows the ISDN network to be
informed of the Calling Party Number (CPN)
when a call originates from a terminal
connected to the UNIVERGE NEAX 2000 IPS.

Call-By-Call Service Selection

Services can be selected on a call-by-call
basis to all channels of a single PRI interface
according to applications. Services may be
used on any available channel, unlike Trunk
Provisioning Service, in which services are
assigned to specific channels.
Called Party Recognition Service (Direct-In Termination (DIT))

This feature provides an incoming Direct-In
Termination (DIT) call via an ISDN trunk to be
connected to a predetermined station. This
application can be used for a station or
modem.
Channel Negotiation

When collisions of data occur between an
outgoing call to ISDN and an incoming call
from ISDN, the B channel that 2000 IPS
specified will be busy on the network and the
network will specify another B channel. This
feature provides negotiation for the selection
of a B-channel between the 2000 IPS and the
network, when the SETUP message is
delivered.

Page 9- 42

CPN To Terminating User–Display

This feature provides a visual display of the
originating station’s number and sub-address
information on a Multiline Terminal for
incoming ISDN calls. This provides the
terminal user with a quick and accurate way to
identify the originating station’s number
(Calling Number).
DID Addressing

This feature allows incoming ISDN-PRI calls
to terminate to stations, Attendant Console,
Automated Attendant, etc., based on the
Called Party number. Direct Inward Dial
trunks will be terminated to programmed
destinations without Attendant assistance.
DID and DOD Addressing

This feature allows the system to use DID and
DOD on the same B-Channels. Trunk
Provisioning Service election is not required.
(B-Channels can be used for DID and DOD
without separating the trunk routes.)

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Event-Based CCIS

Overlap Receiving

This feature allows a PBX customer who does
not have a tie line (or when a customer cannot
use the tie line due to busy or fault of the line),
to use the various CCIS feature by using the
ISDN line as a CCIS virtual tie line, between
the 2400 IPX to 2000 IPS connection or the
2000 IPS to IPS connection.

Overlap Receiving is available for an incoming
call to the 2000 IPS from ISDN network.
Overlap Receiving is a procedure, used in call
establishment of an incoming call, to enable
the network to send called party number digits
to the user in successive messages, as and
when they are made available from the
remote network.

Incomplete Number Handling

When a DID number incoming from ISDN is
incomplete, the call can be transferred to a
predetermined station or attendant console.
ISDN Terminal

This feature provides the system with an
ISDN Terminal or Terminal Adapter (TA).
ISDN Terminal to ISDN Terminal, ISDN
Terminal to ISDN Trunk, ISDN Trunk to ISDN
Terminal, ISDN Terminal to Single Line
Telephone, ISDN Terminal to Multiline
Terminal, and ISDN Terminal to PS
connections are available.
®

MEGACOM Access/WATS
®

AT&T’s MEGACOM (WATS) network, as well
as WATS from other carriers, can be used.
®

MEGACOM 800 Service/800 WATS Ultra WATS
®

AT&T’s MEGACOM 800 (Inward WATS)
network, as well as 800 WATS provided by
other carriers, can be used.

Overlap Sending

Overlap Sending is available for an outgoing
call from the 2000 IPS to ISDN network.
Overlap Sending is a procedure, used in call
establishment of an outgoing call, to enable
the user to send called party number digits to
the network in successive messages.
Sub-Address–Present

This feature allows a Primary Rate Interface
ISDN trunk to transfer the Called Party Subaddress information to a destination ISDN
station when the call is originated by the
system. Dialing the called party station
number and sub-address is required.
Trunk Provisioning Service Selection

Each channel of a PRI interface can be
dedicated to a particular service. Services are
designated to specific channels; once
designated, a channel can be used only for
that service.

MULTIQUEST /900 Service
®

AT&T’s MultiQuest service can be used. (It is
a “900”-type service.) Also, 900 service
provided by other carriers can be used.

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
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Chapter 9 Feature Descriptions

Q-SIG Feature List
Calling/Connected Line ID Presentation (CLIP/COLP)/
Calling/Connected Name ID Presentation (CNIP/CONP)
CCIS Tandem Call-Calling Party Number (CPN) Delivery to
ISDN & Q-SIG Networks
Q-SIG Circuit Switched Basic Call - ETSI Version

Q-SIG Feature Descriptions
Calling/Connected Line ID Presentation (CLIP/COLP)/
Calling/Connected Name ID Presentation (CNIP/CONP)

In between Q-SIG networks, this feature allows calling or called party information, to be
displayed on the Dterm LCD. CLIP/COLP conforms to ETS 300 173 and IS-11572 network.
CNIP/CONP conforms to ETS 300 238 and IS-11572 network.
• Calling Line Identification Presentation (CLIP): Calling party number (ID) is displayed on
the called party's Dterm LCD.
• Connected Line ID Presentation (COLP): Called party number (ID) is displayed on the
calling party's Dterm LCD.
• Calling Name Identification Presentation (CNIP): Calling party information (Name ID) is
displayed on the called party's Dterm LCD.
• Connected Name Identification Presentation (CONP): Called party information (Name ID)
is displayed on the calling party's Dterm LCD.
This feature is also available when networking with a CCIS interface.
CCIS Tandem Call-Calling Party Number
(CPN) Delivery to ISDN & Q-SIG Networks

This feature is provided for the call from tie line (CCIS line or
Q-SIG) or a station, is sent to the public ISDN network (AT&T, NT, NI-2) or Q-SIG network.
Q-SIG Circuit Switched Basic Call - ETSI Version

This feature enables NEC’s PBX to connect to NEC’s PBX or other manufacturer by using
Layer 3 protocol for the signaling for the support of circuit mode bearer services at the Q
reference point.
Q-SIG conforms to ETS 300 172.

Page 9- 44

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
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Chapter 9 Descriptions

Wireless Feature List
Analog PBX Interface
Announcement - PS No Answer / Announcement - PS Out of Zone
Automatic Release - Out of Zone Calls
Call Forwarding - Not Available
Call Forwarding - PS Out of Zone
CCIS Interface
Calling Name Display - PS
Calling Number Display - PS
DTMF Signal Sender
Group Call - Automatic Conference (6/10-Party)
Group Call - 2 Way Calling
Handover
Individual PS Calling
Last Number Redial - PS
Multi-Line Operation - PS
Multi-Site Roaming

UNIVERGE NEAX 2000 IPS Wired for Wireless
Number Sharing
Out of Zone Indication
Overlap Dialing
Preset Dialing
PS Authorization
PS Location Indication
PS Location Registration
Q. 931a Roaming over IP Trunk
Radio Channel Changeover
Short Message Notification (OAI)
Soft Phone Wireless Handset
Speech Encryption
Speed Dial - PS
Station Hunting - Not Available
Voice Mail Indication
WLAN Handset Registration/Service

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Page 9- 45

Chapter 9 Feature Descriptions

Table of Features by Configuration Type
Adjunct Type

Integrated Type

Feature

Analog PBX Interface
Announcement - PS No Answer
Announcement - PS Out of Zone
Automatic Release - Out of Zone Calls
Call Forwarding - Not Available
Call Forwarding - PS Out of Zone
CCIS Interface
Calling Name Display - PS
Calling Number Display - PS
DTMF Signal Sender
Group Call - Automatic Conference(6/10-Party)
Group Calling - 2 Way Calling
Handover
Individual PS Calling
Last Number Redial - PS
Multi-Line Operation - PS
Multi-Site Roaming
UNIVERGE NEAX2000 IPS Wired for Wireless
Number Sharing
Out of Zone Indication
Overlap Dialing
Preset Dialing
PS Authorization
PS Location Indication
PS Location Registration
Q. 931a Roaming over IP Trunk
Radio Channel Changeover
Short Message Notification (OAI)
Soft Phone Wireless Handset
Speech Encryption
Speed Dial - PS
Station Hunting - Not Available
Voice Mail Indication
WLAN Handset Registration/Service
X = available — = not available

Page 9- 46

Analog PBX
Interface

CCIS
Interface

X
X
X
X
–
–
–
–
–
X
–
–
X
X
X
X
–
–
–
X
X
X
X
–
X
–
X
–
–
X
X
–
–
–

–
X
X
X
–
–
X
X
X
X
–
–
X
X
X
X
X
–
–
X
X
X
X
–
X
–
X
–
–
X
X
–
–
X

UNIVERGE NEAX 2000 IPS
Wired for Wireless
–
X
X
X
X
X
–
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
Issue 6

Chapter 9 Descriptions

Wireless Feature Descriptions
Analog PBX Interface

This feature allows the WCS to be integrated
with many types of PBXs and to activate
various PBX services by sending hook flash
signal and DTMF tone.
Announcement - PS No Answer /
Announcement - PS Out of Zone

This feature allows calls to a Dterm PS III
which cannot be paged in a predetermined
period of time to be routed to the
announcement notifying the calling party that
the Dterm PS III cannot answer.
Announcements can be divided between
Announcement-PS No Answer and
Announcement-PS Out of Zone depending on
the Dterm PS III condition.
Automatic Release - Out of Zone Calls

When a Dterm PS III user engaged in a call
moves out of the service area or enters a
radio zone where all the channels are busy
and the communication becomes unavailable,
the call will be disconnected automatically,
and the other party receives reorder tone.
Call Forwarding-Not Available

When a Dterm PS III is powered off or out of
zone, a call directed to the Dterm PS III is
forwarded to a VMS, and a voice mail
message can be recorded to the VMS and
checked from the Dterm PS III. Also the VMS
can page the Dterm PS III automatically after
the voice mail message is recorded.

CCIS Interface

This feature allows the WCS to be integrated
with NEC PBXs with CCIS interface.
Calling Name Display - PS

Without answering incoming calls or hold calls
terminates to the Dterm PS III, the calling
party’s name can be confirmed by the
indications on the LCD.
Calling Number Display - PS

This feature provides a display on the LCD of
a Dterm PS III receiving a call, indicating the
station number.
DTMF Signal Sender

This feature allows a Dterm PS III user to
send the DTMF signal (tone) to the called
party (terminal, voice mail system, etc.) while
engaged in communication.
Group Call-Automatic Conference

(6/10-Party)

This feature permits a Dterm PS III user,
Multiline Terminal user or Single Line
Telephone user within the system to establish
a conference among as many as six or ten
parties. From a Dterm PS II/D term /Single
Line Telephone, a maximum of 9 PSs can be
paged simultaneously except the conference
leader. The PSs are assigned to the
simultaneous paging groups as participants
by the system data beforehand.

Call Forwarding - PS Out of Zone

This feature allows calls directed to a Dterm
PS III, which is power off or out of zone to be
routed to an another station, an Attendant, an
outside number or voice mail equipment. The
call forwarding destination can be set/
cancelled from the Dterm PS III station or
MAT/CAT.

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Page 9- 47

Chapter 9 Feature Descriptions

Group Call – 2Way Calling

Multi-Site Roaming

This feature permits a Dterm PS III, Multiline
Terminal and Single Line Telephone to page a
maximum of fifteen parties simultaneously,
except the group call leader. After one of
paged parties answers, the paging becomes
the 2 Way Calling between the leader and the
first answered party, and paging other parties
stops automatically. The Dterm PS III is
assigned through system data as participants
in the simultaneous paging groups.

UNIVERGE NEAX 2000 IPS Wired for Wireless

This feature allows the UNIVERGE NEAX 2000
IPS to have the Wireless PBX feature. The
IPS provides several integrated features
adding on the adjunctive configuration.
Number Sharing

Handover

When the signal transmission quality
becomes inferior, a Dterm PS III re-originates
a call automatically and seizes another radio
channel, making the WCS handover the call
to another zone transceiver to maintain the
speech quality.
Individual PS Calling

This feature allows the calling party to page
the individual Dterm PS III.
Last Number Redial – PS

This feature enables a Dterm PS III to store
the numbers dialed in the previous five calls
including the last number dialed. The stack
dial numbers are sequentially displayed on
the LCD, allowing the station user to make an
outgoing call by selecting the desired dialed
number from the display.
Multi-Line Operation – PS

Dterm PS III equipped with two line keys, L1
key and L2 key, and different station numbers
can be assigned to each of two lines. The
number assigned to L1 key of the PS is called
My Line and the other number assigned to L2
key is called Sub Line. My Line and Sub Line
of a Dterm PS III can be assigned to appear
on the Flexible Line Keys of a Multiline
Terminal and the Multiline Terminal can share
the Dterm PS III lines.

Page 9- 48

Dterm PS II user can originate or receive a
call in any place of a network provided by
multiple PBXs which communicate through
JT-Q931-a.

This feature allows the Multiline Terminal user
to have a Dterm PS III as Sub Station and to
get service with one telephone number. In
case that one user has both a Multiline
Terminal and Dterm PS III, with this feature
used the user is not required to have two
separate telephone numbers.
-When a user is at his desk, a call is
terminated to his Dterm.
-When a user leaves their desk with Dterm PS
III, a call is automatically terminated to the
Dterm PS III.
In this feature, the Multiline Terminal and
Dterm PS III are referred as Main Station and
Sub Station, respectively.
Out of Zone Indication

When a Dterm PS III user moves out of the
service area and the electric field strength
becomes weak, this feature notifies the user
with a warning tone and an LCD display.
Overlap Dialing

This feature allows a Dterm PS III user to
receive dial tone and dial the desired number
to originate a call.
Preset Dialing

This feature allows a Dterm PS III user to
confirm the number to be dialed before
originating a call.

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
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Chapter 9 Descriptions

PS Authorization

This feature is to confirm the identity of a
Dterm PS III to avoid an unauthorized Dterm
PS III from accessing the system.
PS Location Indication

Soft Phone Wireless Handset

This feature allows a Soft Phone (Dterm
SP30) user to use a PS terminal as the
handset of the Soft Phone. The user can
select a handset (Soft Phone Handset or PS)
by pressing the Handset Switch key on the
Soft Phone.

When a Dterm PS III calls a Multiline
Terminal/Attendant Console or vice versa, this
Speech Encryption
feature allows to indicate the location name of This feature protects a call from being tapped.
the ZT II that is connected to the Dterm PS III
Speed Dial-PS
on the LCD of the Multiline terminal.
This feature allows a Dterm PS III user to dial
the certain frequently called numbers using
PS Location Registration
This feature allows the IPS-WCS to supervise two-digit abbreviated call codes.
the location of each Dterm PS III, upon
Station Hunting - Not Available
receiving the location registration request, to
This feature allows a call placed to a Dterm
allow call termination.
PS III station, which is out of zone or power
Q931a Roaming Over IP Trunk

Dterm PS III user can originate or receive a
call in any place of a network provided by
multiple PBXs, which are interfaced by JTQ931-a protocol over IP Trunks.
Radio Channel Changeover

This feature is to supervise and changeover
the channel automatically to avoid the
interference and to maintain the speech
quality.

off to be forwarded to an idle station in a hunt
group. Two Station Hunting arrangements are
available. Station Hunting - Circular processes
the call regardless of which station in the hunt
group is called. Station Hunting - Terminal
initiates a hunt only when the pilot number of
a hunt group is called.
Voice Mail Indication

When a message is mailed in the Dterm PS
III, an indication of the envelope icon is
displayed in the LCD of the Dterm PS III.

Short Message Notification (OAI)

This feature enables a Short Text Message
(STM), once arrived at a mail box of the STM
Server (external equipment), to be
automatically distributed to the addressee
Dterm PS III via Open Application Interface
(OAI). This feature also provides “TM Full”
notification on the display (LCD) of the
address D term PSII that is busy. If the Dterm
PS III is out of zone, the feature makes
retransmission of the STM after the PS
returns to the zone.

WLAN Handset Registration/Service

This feature allows the 2000 IPS to
accommodate the WLAN Handset as a
station. A terminal is connected to the 2000
IPS through Access Point and Gateway and
operates under the control of the virtual CS on
the Gateway. Station-to-Station calls between
WLAN stations and calls between WLAN
stations and existing stations/trunks are
available.

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Page 9- 49

Chapter 10 System Input/Output (I/O) Interfaces
Station Message Detail Recording (SMDR)
This feature provides a call record for outgoing Station-to-Trunk calls, incoming Trunk-toStation calls (including Data Call), and tandem calls. This facilitates cost control by identifying
trunk use and misuse by individual stations. Station Message Detail Recording (SMDR)
enables call billing to customers and clients, and provides a means for checking local
telephone bills. The PBX provides three kinds of Station Message Detail Recording (SMDR)
formats; 2400 IMS Format, 2400 IMS Extended Format and 1400 IMS Format.
2400 and 2400 Extended Format
The following call information is provided to the SMDR terminal upon completion of each outgoing or
incoming C.O. call.
Type of Record
Trunk Number (000-255)
Trunk Route Number (00-63)
Calling/Called Party Information
- Calling/Called Party I.D.
- Tenant Number (01-63)
- Station Number (Max. 5 digits)
- Attendant Number (00-07)
- Trunk Route (00-63) + Trunk Number (000-255)
Time of Start Conversation-Month, Day, Hour, Minute, Second
Time of Call Completion-Month, Day, Hour, Minute, Second
Account Code (Max. 10 digits)
Condition Code
Route Advance Information
Called Number (Max. 24 digits)*
Call Metering Information
Calling Party Office Number (Max. 4 digits)
Billing Office Number (Max. 4 digits)
ANI Information (Former: Max. 16 digits/Extended: Max. 32 digits)*
Authorization Code (Max. 10 digits)*
Call Start Year
Call End Year
Condition Code for Advice of Charge (AOC)*
Advice of Charge (AOC)*
Note: The information marked by “*” is only for outgoing C.O. calls.
The information marked by “**” is only for incoming calls.

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Page 10-1

Chapter 10 System Input/Output (I/O) Interfaces

1400 IMS Format
1400 IMS Format is generally used to print call records on a serial printer and provides an easy to read
format for printouts of outgoing / incoming call activity.
Each record is given a serial number from 0000 to 9999 to identify each call.
Calling Station Number
Called Number (26 digits maximum)
Route Number and Trunk Number
Start of Call Time (from month down to second)
Duration of Call (hours, minutes, seconds, rounded to nearest .5 sec.)
Forced Account / Authorization Code
Account Code
Attendant Handled
Trunk Route Overflow (Route Advance)
CPN/ANI up to (16 digits)
Six options for 80/136 character printers

SMDR Terminal Interface
A Personal Computer or a Host Computer is used as the SMDR terminal for receiving and processing
the call information that arrives at the MP. The transmission interface specification by which the AP or
MP sends out call information to the SMDR terminal are as follows:
SMDR with AP00/MP built-in SMDR on RS-232C
ITEM

SMDR w/AP00

Specifications
MP Built-in SMDR on RS-232C

Physical Interface
Synchronization
Protocol
Transmission Speed

RS-232C
RS-232C
Asynchronous
Asynchronous
Non protocol (Free Wheel)
Non protocol (Free Wheel)
1200/2400/4800/9600 bps
1200/2400/4800/9600/19200 bps
(for PN-AP00-B with MRC-C program)
Note 1
300/1200/2400/4800 /9600/19200 bps
(for PN-AP00-B with MRC-F program)
Stop Bit
1/1.5/2 bits
1/2 bits
I/O Port
No. 0-3 port of AP00-B card
RS port of MP card
Terminal Busy
Data Carrier Detect (DCD) signal
Data Carrier Detect (DCD) signal
Detecting Method
ON/OFF (terminal ready/busy)
ON/OFF (terminal ready/busy)
Note 1: For the port 1 and 3 of AP00-B card with MRC-C program, data speed cannot be set to 9600 bps.
Note 2: Upon confirming that the status of the DCD signal from the SMDR terminal is ON, the system sends out
call information to the SMDR terminal. When the status of the DCD signal is OFF, the system does not
send out call information but temporarily stores the information until the SMDR terminal becomes ready
to receive call information, in other words, until the status of the DCD signal changes to ON. If the status
of the DCD signal has changed from ON to OFF while transmission of specific call information is in
progress, the next call information is not sent out but stored into the system temporarily. The same
applies to CTS and DSR signals.
Note 3: For output to MP built-in SMDR, one message is sent at intervals of one second unidirectional.

Page 10-2

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Chapter 10 System Input/Output (I/O) Interfaces

MP built-in SMDR on IP
ITEM

Specifications
Ethernet

Physical Layer
Connection Layer
TCP/IP Protocol
Socket Interface
Transport Protocol
Application Port Number
Number of Connections
Client/Server

The Ethernet packet format complies with the DIX standard
ARP, IP, ICMP, UDP, TCP
Complies with 4.3 BSD socket interface
TCP stream type protocol
60010 (fixed)
1
Client: SMDR terminal
Server: PBX
Transmission Code
7-bit ASCII code
Quasi-normal Restriction Condition
1. When connection is closed
2. Status monitoring text
Note 1: The MP card in main site communicates with the SMDR terminal. Therefore, in the communication
settings at SMDR terminal side, set the IP address to connected to the address specified by office data
(CM0B Y=00>00 or CM0B Y=02>03), and application port number shown in the above table.
Note 2: One message is sent each time a request is received from SMDR terminal.

Combinations between SMDR and SMDR Output Format

SMDR

2400 IMS
Format

Extended
2400 IMS
Format

1400 IMS
Format

—

SMDR with AP00
(PN-AP00-B with MRC-C program)
SMDR with AP00
(PN-AP00-B with MRC-F program)
MP built-in SMDR on RS-232C
MP built-in SMDR on IP
X: Available —: Not available

Combinations available at the same time

SMDR
SMDR with AP00
(PN-AP00-B with MRC-C program)
SMDR with AP00
(PN-AP00-B with MRC-F program)
MP built-in SMDR on RS-232C
MP built-in SMDR on IP

Pattern A

Pattern B

Pattern C

Pattern D

Pattern E

—

X: Available —: Not available

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
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Page 10-3

Chapter 10 System Input/Output (I/O) Interfaces

Combinations for Center Office of Centralized Billing

SMDR

Former 2400 IMS Format

Extended 2400 IMS
Format

—

SMDR with AP00
(PN-AP00-B with MRC-C program)
SMDR with AP00
(PN-AP00-B with MRC-F program)
MP built-in SMDR on RS-232C
MP built-in SMDR on IP
X: Available —: Not available
Note: Only one output can be used at the center office.

Combinations for Local Office of Centralized Billing

Former 2400 IMS Format

Extended 2400 IMS
Format

—

MP built-in SMDR on RS-232C

MP built-in SMDR on IP
X: Available —: Not available
Note: Only one output can be used at the local office.

—

SMDR
SMDR with AP00
(PN-AP00-B with MRC-C program)
SMDR with AP00
(PN-AP00-B with MRC-F program)

Service Conditions
SMDR can be programmed to record all outgoing calls or toll calls only, depending on the
customers' requirements.
When customer provided computer equipment is connected using the RS-232C interface, SMDR
information will be transmitted directly to the equipment as each call record is completed.
If the outgoing call is directed to a trunk, which does not supply answer supervision, SMDR will start
recording the call approximately 10 seconds after the last digit has been dialed.
Supervision of the status of the external RS-232C terminal is not supplied.
There are two kinds of SMDR features as follows:
a. Built-in SMDR
The MP card has two RS-232C ports for SMDR. This card provides memory for a maximum of
1024 calls, and provides a record of a maximum of 255 trunk calls simultaneously for a
maximum of 6-digit extension number.
Note: In case of built-in SMDR, if the station number is a 7- or 8- digit number, the upper one or
two digits are sent as Calling office Number and the last six digits are sent as Calling Number in
the call record format.
Page 10-4

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Chapter 10 System Input/Output (I/O) Interfaces

b. SMDR with AP00
o
o
o
o

AP00 without expansion memory (standard): up to 1600 calls (if central office feature is used, up to
800 calls)
AP00 with expansion memory (optional): up to 27000 calls (if central office feature is used, up to
26000 calls)
When a call is completed, the record is sent to the output device and is removed from memory.
If the amount of calls exceed the SMDR memory, those overflow calls will not be recorded.

The built-in SMDR cannot be provided when AP00B with MRC-C program is mounted.
Account Codes, Forced Account Codes, Authorization Codes and DISA Codes on Tandem
connections (SMDR with AP00 only) are reported in the applicable call record.
SMDR with AP00 provides a record of incoming Tie Line tandem calls where another trunk is dial
accessed.
For details of SMDR format, data stream, memory buffer, etc., refer to the Installation Procedure
Manual/Feature Programming Manual.
Up to 16 digits of the calling subscriber’s number for Automatic Number Identification (ANI) and
CPN (ISDN) can be recorded in the SMDR by system programming.

Data Transmission Sequence for Ethernet Interface
Establishing a connection, outputting billing Information, and releasing a connection are performed
between the PBX and the SMDR terminal in the following timings.
Timing to Establish a Connection
o A connection is established when a connection request is received from the SMDR
terminal.
Timing to Output Billing Information
o Billing information is output when a polling request is received from the SMDR terminal
connected to LAN.
Timing to Release the Connection
o The connection is released when the SMDR terminal does not receive data from the
PBX in a predetermined time interval. Also the connection is released when the PBX
receives connection release text from the SMDR terminal or does not receive data from
the SMDR terminal within a given time interval.

System Capacity
MP Built-in SMDR
Number of Simultaneous
Trunk Calls
255

Maximum No. Call Records
1024

SMDR with AP00
No EXPMEM card on AP00
EXPMEM card on AP00

When not a center office of
Centralized Billing-CCIS
1600
27000

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
Issue 6

When center office of
Centralized Billing-CCIS
800
26000

Page 10-5

Chapter 10 System Input/Output (I/O) Interfaces

Hardware Required for SMDR
Description

Qty

SPN-AP00B MRC-C (AP)

1 per system

SPN-AP00B MRC-F (AP)

1 per system

Remarks
Control card for SMDR/PMS/H/M Printer/CCIS
Centralized SMDR
Note
Control card for SMDR in 2400 &2400 Extended
format only, Centralized SMDR
Note
MP
MP Mounted card One per MP

SPN-CP24-C MP (UA)
1 per system
PZ-M606-A (ETHER)
1 per system
RS RVS-4S CA-C
1 per SMDR, PMS
RS NORM-4S CA-A
1 per SMDR, PMS
RS PRT-15S CA-A
1 per Printer
SMDR Terminal
1 per system
Printer
as Required
Modem
as required
Note: Only one AP00 can be mounted at on time in the system.

Message Center Interface (MCI)
This feature provides an interface with a customer supplied Voice Mail System (VMS) which
can send Message Waiting lamp control data to the system.
The Message Center Interface (MCI) can provide the following operations:
When terminating the call to the VMS, the system sends call connection status
information to the VMS through the MCI.
The VMS sends the Message Waiting Lamp on data to the MCI.
The system, upon receiving this control data from the MCI, illuminates the Message
Waiting lamp of the corresponding station.
The VMS, upon receiving retrieved message information, will send the Message
Waiting lamp control data requesting the system to extinguish the Message Waiting
lamp of the corresponding station.
Service Conditions
The Voice Mail System (VMS) is interfaced to the system through the 4LCD/8LC card. (The
4LCD/8LC provides disconnect supervision in the form of a momentary loop open.)
The UCD or Station Hunting feature is usually provided with the VMS station.
One RS-232C port on the MP card or the Application Processor (AP00) is required to make a data
link with a customer supplied VMS.
Messages can be retrieved from any Multiline Terminal, DTMF telephone, or the Attendant
Console, but not from DP telephones.
The MCI is available to a direct call or a forwarded call from a station/trunk/Attendant to the VMS.

Page 10-6

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Chapter 10 System Input/Output (I/O) Interfaces

Stations can set Call Forwarding or Split Call Forwarding - All Calls, No Answer, and Busy Line to
the VMS. The system sends out incoming call information to the VMS. A call to a station that has
Call For-warding set to the VMS is automatically answered by the VMS.
The MCI can control the LCD display of a Multiline Terminal for “MESSAGE” Indication. The
number of messages is not displayed.
When the Message Waiting lamp control is activated with the MCI, the lamp control from the
following equipment will not be provided:
From the Property Management System (PMS)
From the station (by dialing the access code)
From the Direct Station Selection (DSS) Console
From the Hotel/Motel Attendant Console
From the Hotel/Motel Front Desk Instrument
From the Attendant Console
Only one system should be programmed (via system programming) to control Message Waiting
lamps through the CCIS network.
The system controls Message Waiting lamps normally when the time interval between messages is
a minimum of 350 msec. or more.
When the VMS interface line does not answer, all of the messages are sent out from the I/O port of
the MP/AP00.
If the VMS is not ready for information receiving (Busy Status), the AP00 can temporarily store up to
16 call records in its internal memory. If the maximum of 16 call records is stored and a 17th is
generated, the system will write over the oldest stored record. When the RS port on the MP card is
used for the data link to the VMS, the MP can store up to 15 call records. If a 16th call record is
generated when the MP stores 15 call records, the system will write over the oldest stored record.
The Voice Mail Integration (In band) feature can be combined with voice mail through the MCI in the
system. The Voice Mail Integration (In band) feature and MCI feature can coexist in one system
and either can be selected per VMS (VMS station number) by system programming.
When terminating a call with the ANI information to the VMS through the MCI, the system can send
the ANI information to the VMS, if required. This is not available through CCIS interface.

Hardware Required for MCI
Description
Qty
Remarks
SPN-CP24-C MP (UA)
1 per system
MP
SPN-AP00B MRC-C (AP)
1 per system
Control card for MCI
Note
SPN-AP00B MRC-F (AP)
1 per system
Control card for MCI
Note
RS RVS-4S CA-C
1 per VMS
AP00-VMS, 13.1ft (4.0m)
RS NORM-4S CA-A
1 per VMS
AP00-MODEM, 13.1ft (4.0m)
PN-8LCAA
As required
Interface for VMS (with MW lamps)
VMS
1 per system
MODEM
As required
Note: Only one AP00 can be mounted at on time in the system.

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Property Management System (PMS)
The system provides a data interface to a locally provided Property Management System
(PMS). This enables communication between the system and the PMS in order to provide
computer control of Hotel/Motel features.
The available PMS feature codes:

Feature

Feature
Code

Message Waiting
Station Message Detail

11
12
13
14

Restriction Control

Maid Status

Check In/Check Out
16
Room Data Image
17
Wake Up
19
Room Change/Room
20
Swap/Room Copy
Room Occupancy/
21
Room Data Change
Data Connection
50
Maintenance
Message Waiting
53
Station Message Detail
54
Check In/Check Out/Room
56
Change
Room Recovery
57
Direct Data Entry
59
Room Data Change
61
Administrative
65
Station Name
Status Inquiry
70
X: Available —: Not available

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Type of Information
Maid status set up by guest room telephones
Maid status set up by the Front Desk Terminal
Message waiting lamp status changes
Local/toll call details on completion of calls
Changes in telephone calling restrictions
placed on room station numbers
Room check in/check out status
Complete status information for a room
Wake up status

PMS
AP00
IP
X

X
X

—

X
X
X

Room change/room swap status

Room Occupancy/Room Data Changes

—

Batch processing status

—

Controlling message waiting lamps
Station message detail information for each call

—
—

X
X

Setting check in/check out and room change

—

Room information at system recovery
Data entered from a guest room station
Room data change information

—
X
—

X
X
X

Setting administrative station name

—

Data link maintenance

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PMS Interface
PMS with AP00-B on an RS-232C Interface
Item

Description
1200, 2400, 4800, 9600 bps, asynchronous, software
selectable

Data Rate

Note 1

Operating Mode
Electrical Interface
Characteristic
Signal Form
Interface Distance

Half Duplex
EIA RS-232C electrical standard interface
EIA RS-404
Max. 15 m (49.2 ft.) between PBX and PMS (without modem)
Note 2

Parity LRC

10 bits (1 start, 7 data, 1 parity, 1 stop) or 11 bits (1 start, 7
data, 1 parity,2 stop)
No parity, even parity, odd parity; selected by PBX system
data
Exclusive OR of message text

Frame Contents

US ASCII 7-bit codes

Word Framing
Parity VRC

Control

Contention
Primary office: PBX
Priority Sequence
Secondary office: PMS
Note 1: For the Port 1 and Port 3, data speed 9600 bps cannot be used.
Note 2: When modems are used, full duplex asynchronous type modems are required.

Built in PMS over IP
Item

Description

Physical layer

Ethernet

Connection Layer

The Ethernet packet format complies with the DIX standard

TCP/IP core protocol

ARP, IP, ICMP, UDP, and TCP

Socket interface
Transport Protocol

Complies with the 4.3BSD socket interface
TCP stream-type protocol

Application Port Number

60050 (Fixed)

Number of Connections

1 connection
Server: PBX
Client: PMS
US ASCII 7-bit codes
1. At connection release
2. Status monitoring text

Client/Server
Frame Contents
Pseudo Normal Restriction
Condition

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Service Conditions
1. Only one Interface Port can be provided for PMS.
2. The system sends information relating to the following features to the PMS upon request from the
PMS:
Do Not Disturb
Room Cut-off
3. The UNIVERGE NEAX 2000 IPS system can support PMS Model-60 via AP00-B or PMS Model
90/120 via TCP/IP connection.
4. The AP00-B with MRC-F program is used for Executed task Print out Request for MP Built-in PMS.

Hardware Required for PMS
Description
Qty
Remarks
SPN-CP24-C MP (UA)
1 per system
MP
SPN-AP00B MRC-C (AP)
1 per system
Control card for PMS Note
SPN-AP00B MRC-F (AP)
1 per system
Executed Task Print Out for MP PMS Note
PZ-M606-A (ETHER)
1 per system
MP Mounted card One per MP
RS RVS-4S CA-C
1 per SMDR/PMS AP00-VMS, 13.1ft (4.0m)
RS NORM-4S CA-A
1 per SMDR/PMS AP00-MODEM, 13.1ft (4.0m)
RS PRT-4S CA-A
1 per Printer
AP00 Printer
PMS Terminal Serial
1 per system
Model 60 only
PMS Terminal over IP
1 per system
Model 90/Model 120
SMDR Terminal
1 per system
Printer
As required
MODEM
As required
Note: Only one AP00 at a time can be mounted in the system.

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System Outline
The UNIVERGE NEAX 2000 IPS is equipped with an Open Application Interface (OAI) to
provide user applications through an external processor, which transmits and receives the
control signals via Ethernet interface on the MP. The interface is supports TCP/IP protocol.

OpenWorX: Attendant Statistics
OpenWorX Statistics operates in conjunction with the Business Attendant System in order to
provide near real time statistics about the operators and the queues. A variety of information is
gathered and stored in a database that allows for a multitude of differing reports. With the
power of Crystal Reports® the user can then view the data with a web interface allowing for
HTML, PDF, and RPT formats. An additional purchase of Crystal Reports Viewer® is required
for the RPT format. Below is a list of the types of reports the user can run.
Attendant Reports
Attendant Call Summary
Attendant Answered Calls
Attendant Outbound Calls
Attendant Interval Summary
Attendant Inbound Interval
Attendant Outbound Interval
Total Attendant Summary
Attendant Login/Logout Detail
Queue Reports
Queue Summary
Queue Abandon Calls
Queue Answered Calls
Queue Interval
Total Queue Summary

OpenWorX: Business Attendant System (BAS)
NEC’s Business Attendant System (BAS) is a powerful client/server PC-based tool for full-time
and part-time UNIVERGE NEAX 2000 IPS system answering position. The BAS consists of an
interactive client side application running on Windows 2000 or Windows XP operating system
that connects to the BAS server via the corporate LAN. The BAS server software is used for
UNIVERGE NEAX 2000 IPS connectivity and runs on a Microsoft Windows 2000 Server or
Windows 2003 Server.

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With a dynamic interactive directory and an Open Application Interface (OAI) driven call queue,
the operators are able to process a high call volume with minimal effort and time expenditure in
today’s sophisticated business environment. For example, an operator can answer an
incoming call from one of many queues; find out to whom the caller wishes to speak, begin
typing that name to select it in the directory, and once selected, hit the key or click the
entry to transfer the call. Additionally the caller could be parked, put on hold, or sent to voice
mail. Several operators can monitor the same queues and share the same directory for call
operation.
BAS: Benefits
Centralized company database that can link and replicate with other databases, including
AIMWorX.
Greeting playback and call recording.
PC-based call processing.
Meet-me paging.
Unlimited, global parked calls.
Different greetings/displays available for DIDs.
Single key call processing.
Directory dialing for accuracy.
Look ahead status from the directory.
Sort and searchable directory by key fields.
Configurable screens for each operator.
Queue types allowing flexibility to answer certain call types.
Configurable screen pop information.
MAPI compliant e-mail integration.
Set/cancel call-forwarding and DND for other phones.
Delay announcements capabilities to ensure callers are informed and provided other options
during busy call times.
Configurable night station or announcement when operators are not logged in.
Flexible DID number assigned to each operator that follows them regardless of what PC/phone
they use.
Optional statistics package to track queue and operator statistics.
Optional integration with OpenWorX Location Status Information to see an employee's
availability, return time, and alternate phone number.
Optional integration with OpenWorX Dialer providing Dterm users’ access to the same employee
and external database.
Optional integration with OpenWorX Message Reader providing Dterm users access OpenWorX
text messages.
Optional integration with OpenWorX Desktop providing PC users all the benefits of the standard
Dterm -based applications from a graphical user interface.
FASTER and MORE EFFICIENT OPERATORS!

BAS: Call Control Functionality
Call control for the BAS is a queue-oriented system. Just like the NEC business attendant
console, the BAS provides the standard familiar call queues and information. However
additional information about incoming calls is added to better empower the Attendant.
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Brief Definition of Queues:
Internal – extension to attendant calls within the PBX.
External – calls coming into PBX from an undefined trunk type.
CCSA, FX, TIE, WATS – calls coming into the PBX from trunks in CCSA, FX, TIE or WATS
Trunk Groups.
Priority – calls that must be answered before all other calls.
Paged – people who have been paged to a meet-me paging number, but the parked person is
no longer waiting to be joined.
Vacant – vacant-level intercept when a call goes to an invalid number.
Park T.O. – calls that were parked and have timed out will recall here.
NANS T.O. – internal calls blind transferred but not answered will recall here.
Station Transfer – transferred to the attendant queue by another station or voice mail.
Attendant Transfer – calls transferred from other attendants will arrive on this queue.
Me – This queue is for the attendant’s personal monitored number or calls made directly to their
extension.
Brief Description of Call Control Function:
Answer – answers the oldest call in queue or in the priority queue.
Park – allows the operator to park a caller on an orbit number. Once there, dialing that orbit
from any telephone in the system retrieves the caller. This function is also known as “Meet-me
Paging”.
Dial – allows the operator to make an outgoing call from his or her extension.
Retrieve – enables the attendants to retrieve parked calls by selecting a caller to retrieve from a
menu of callers that are parked.
Hold – place a caller on hold on one of the loop keys or “un-hold” them as well. The Dterm will
become idle.
Transfer – initiates a supervised transfer once a number has been entered or a directory entry
selected.
Blind Transfer – initiates an un-supervised transfer. As with Transfer, a number must be
selected beforehand.
Join Park – allows the operator to join an existing caller with a parked caller that is selected from
the parked call menu.
Conference – a conference between the source caller, the destination caller, and the operator.
Source/Destination – used when a supervised transfer is initiated. Before the transfer is
completed, the operator may switch voice path back and forth between the existing source and
destination connections.
Camp – camps a caller onto a busy extension.
Cancel – disconnects a call or cancels an existing supervised transfer.
Release – completes a supervised transfer or conference.
Override – allows the operator to perform an executive override of a busy call.
The above functions are mapped to Function Keys (F1 – F12) at the top of the keyboard. The
spacebar can be used to answer the first call in queue and the Enter key will activate dialing or
transfer based on the current call state.
Other Functions and Features Associated With This Screen:
Loop keys allow the attendant to hold up to 10 different callers simultaneously.
The boxes directly below the loop keys are the source and destination boxes. The Source is on
the left and is fully configurable to show the caller name and caller ID if known. Additionally, up
to three lines of information chosen from the directory can be displayed below the name and
number. The destination box is configurable in the same way and shows available information

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when the operator transfers the call to another party, or when a station transfers a call to the
operator.
The Number Entered box allows the operator to enter a known number for either a direct dial or
a transfer.
The number shown in the bottom hand portion of the screen represents the number of calls
waiting in queue at the queue screen, or total queue depth.
The time shown left of the queue depth describes how long the current caller has been
connected. Also if the operator hovers the mouse pointer over a loop key with a caller on hold,
the time that caller has been holding will display.
Left of the timer is an information screen that displays the name of the person holding while
hovering the mouse pointer over a held loop. Also, if a caller is recalled from a park or a blind
transfer, their identification, and the party to which they were transferred or for which they were
parked will be shown here.

BAS: Speed Dials Pad
Up to forty speed-dial fields are available for each individual attendant. Dragging an entry from
the directory onto the speed dial button automatically programs these fields. Another great
feature of these buttons is that they act as a Busy Lamp Field (BLF). When that user’s phone
is off hook, the speed dial button becomes red and when that phone is ringing, the speed dial
button blinks red. By right clicking the button, the operator can enable or disable monitoring,
as well as change the name, forward the person’s Dterm, set DND, or initiate a screened,
blind, or announced voice call. This is useful for an operator who also has a set of VIPs for
whom they answer calls.
BAS: Directory
Both an internal directory and an external directory exist in the BAS system. The internal
directory lists employees within the company, while the external directory stores numbers
outside the company. External numbers are usually numbers that are frequently called, or
information about people or companies that frequently call the main listed number. When
answering a call from one of these external numbers, fields from this entry are displayed to the
operator, providing more information than just the 10-digit caller ID. There are several different
fields available for display and four of these fields can be user-defined. The “Search” field
searches dynamically as the information is typed in the field selected. The directory is also
capable of being partitioned by company. The drop down box at the top of the window
displays entries from a particular company, or all companies in the system. This is particularly
useful in shared tenant installations, where a single attendant group serves multiple
organizations.
DID for Company Lookup
A DID number can be associated with each company selection in the database. With this function
enabled, when a call comes in on a specific DID for that company, the operator directory will
automatically default to that company and only show those employees that are in the company. A
screen pop will also occur letting the operator know which DID was called.

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BAS: Overhead Paging
Most overhead paging is done by the creation of a direct trunk access key or a trunk access
code. The BAS software does not have the ability to utilize these methods. Because of this, if
the user wishes to overhead page from BAS they must manually initiate it from their Dterm.
However the user may create a special speed dial key instead. In the PBX, the user must
create a trunk access dial string. Using LCR, a dial string can directly access the trunk and
provide voice path to the paging system. For example, the user creates the number 19 in their
LCR to access trunk 1 in route 1. This trunk is terminated to the overhead paging. Now when
the attendant hits their speed dial to dial 19, they will be able to overhead page.
BAS: Call Recording and Playback
Apart from the directory, another powerful feature integrated into the BAS client is the ability to
record conversations and playback greetings over the telephone. With the addition of a Dterm
Play/Record Module, the client will be able to automatically record voice conversations and
instruct the software to save the last specified number of conversations. Conversations are
saved as .wav files, which are time; date stamped and rotated in a circular buffer. When it is
determined that a call needs to be saved, the operator simply presses the “Save Recordings”
button and the selected call is automatically saved. Conversely, the operator can save a set of
greetings to be played back immediately every time the phone is answered via the BAS.
Greetings playback can be determined based on “internal” or “external” calls, on groups of
caller IDs, or even morning or afternoon. A different greeting can be recorded for any of the
scenarios above allowing the attendant to present a pleasant and consistent greeting.
BAS: Offline Mode (Pseudo Night Mode)
Night Mode allows for calls to be handled differently during off business hours. When no BAS
attendants are logged into the system, the OWX server is still monitoring calls to the
attendants. Since there are no attendants to send the calls, the server will instead send the
calls to a configurable number, such as the voice mail system, or play a night announcement.
In the case where the server itself goes down, it is recommended that the user set up a UCD
degrade. Since the OAI cannot route the calls, it will instead send the calls to a UCD group,
which includes the operators. Typically a separate set of virtual sub-lines are put into a UCD
group and assigned on each attendant’s Dterm and the pilot number of this UCD group is
associated with the BAS queues to be used in the event of failure.
BAS: Authorization and Account Code Dialing
BAS supports dialing with Account Codes or Authorization Codes. However, it uses a slightly different
number format for the dial string. To denote an account code an exclamation point (!) is used and for
authorization codes a dollar sign ($) is used. Here are some example strings.
Account Code: 2145551212! 1234567890
Authorization Code: 2145551212$ 1234567890
The exclamation point (!) or the dollar sign ($) delineate to BAS where the dial-able number ends and
where the code begins. In the account code example, 2145551212 is the phone number and
1234567890 is the account code.
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OpenWorX: Business Receptionist (BR)
The OpenWorX® Business Receptionist is a PC-based call-processing application designed
primarily for departmental executive assistants. The OpenWorX® Business Receptionist
enables multiple users to answer calls, search the directory for a desired party, check the
Dterm status, and perform call processing functions including making, answering, transferring,
and parking calls. The OpenWorX® Business Receptionist is a line-based system unlike the
Business Attendant System (BAS) which is queue-based. Therefore, the OpenWorX®
Business Receptionist is also an effective tool for other Dterm users such as help desks,
security desks, etc. The OpenWorX Business Receptionist, like all OpenWorX® applications,
uses the common OpenWorX® database and is integrated with a multitude of OpenWorX®
applications.
BR: Speed Dials Pad
The BR speed dial pad is modeled off the BAS speed dial. See the previous section on BAS for details
of operation.
Note: In order for the BR operator to monitor the speed dials, they must be given TAPI rights to the lines they
wish to monitor

BR: Directory
Both an internal directory and an external directory exist in the Business Receptionist system. The
internal directory lists employees within the company, while the external directory stores numbers
outside the company. External numbers are usually numbers that are frequently called, or information
about people or companies that frequently call the main listed number. When answering a call from
one of these external numbers, fields from this entry are displayed to the operator, providing more
information than just the 10-digit caller ID.
DID for Company Lookup
The directory is also capable of being partitioned by company. The drop down box at the top of the window
displays entries from a particular company, or all companies in the system. This is particularly useful in
shared tenant installations, where a single attendant group serves multiple organizations. Additionally, a DID
number can be associated with each company selection in the database. With this function enabled, when a
call comes in on a specific DID for that company, the operator directory will automatically default to that
company and only show those employees that are in the company.

BR: Overhead Paging
Most overhead paging is done by the creation of a direct trunk access key or a trunk access code. The
Business Receptionist software does not have the ability to utilize these methods. Because of this, if
the user wishes to overhead page from Business Receptionist they must manually initiate it from their
Dterm. However the user may create a special speed dial key instead.
In the PBX, the user must create a trunk access dial string. Using LCR, a dial string can directly access
the trunk and provide voice path to the paging system. For example, the user creates the number 19 in
their LCR to access trunk 1 in route 1. This trunk is terminated to the overhead paging. Now when the
attendant hits their speed dial, to dial 19, they will be able to overhead page

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BAS: Call Recording and Playback
Apart from the directory, another powerful feature integrated into the Business Receptionist client is the
ability to record conversations and playback greetings over the telephone. With the addition of a Dterm
Play/Record Module, the client will be able to automatically record voice conversations and instruct the
software to save the last specified number of conversations. Conversations are saved as .wav files,
which are time & date stamped and rotated in a circular buffer. When it is determined that a call needs
to be saved, the operator simply presses the “Save Recordings” button and the selected call is
automatically saved. Conversely, the operator can save a set of greetings to be played back
immediately every time the phone is answered via the Business Receptionist. Greetings playback can
be determined based on “internal” or “external” calls, on groups of caller IDs, or even morning or
afternoon. A different greeting can be recorded for any of the scenarios above allowing the attendant to
present a pleasant and consistent greeting over the telephone.

BR: Authorization and Account Code Dialing
Business Receptionist supports dialing with Account Codes or Authorization Codes. See the previous
section on BAS for details of operation.

BR: Attendant PC Minimum Hardware Requirements
Business Receptionist hardware requirements are the same as the BAS requirements, see previous
section for details.

OpenWorX: Communications Portal
Business communications in the 21st century is no longer defined strictly as phone calls and
documents. It has become pervasive in the business day, taking many forms – phone calls to
the desk, phone calls to cell phones, emails, voice mail, faxes and instant messaging.
The OpenWorX® Communications Portal provides the added value of including communication
tools within the portal. This enables information technology organizations to provide their
users with a single interface for multiple communication applications. The consistent interface
across the organization can streamline administration, management and user training while
providing the greatest level of functionality to users.
Many daily business activities rely on communications, but at the same time, business people
can easily be overwhelmed by the volume of contacts – and the variety of communication
forms enables anyone to interrupt at any time. Today’s business professional must be
empowered to manage their communications. OpenWorX Communications Portal integrates
all of these channels into a unified, web accessible portal that puts the individual back in
control of their own communications. All contacts can be simply and easily managed from a
single application – reducing the complexity of dealing with multiple communication types and
increasing personal productivity.
Today’s business person is mobile, yet they are dependent on communications to maintain the
flow of their business. Most business communication attempts fail on the first attempt because
the receiving individual is away from their desk or is otherwise engaged. The OpenWorX
Communications Portal will understand the individual’s status. Using flexible rules and
capitalizing on the integration of voice and data, it will selectively route contacts and messages
based on the identity of the sender as well as the status of the individual. This allows the
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business user to define who they want to interact with in real-time – to complete those critical
projects – and who they would prefer to “get back to” at a later time.
Communications Portal: Functionality
Portal
The Portal functionality was designed to grant the individual user access to enterprise web-based backend
systems such as databases, HR applications, and existing business portals. The Portal supports custom or
company created portlets, so tools can be provided to users to help them better perform their daily business
tasks. Users can also add Internet sites and applications, personalize their settings, edit screen layouts, and
change color and title schemes.
My Contacts
Having the ability to find the people you need to contact is imperative. The “My Call List” combines call logs,
a custom contact or buddy list – this list can be comprised of individuals or groups – along with personal and
corporate directories in one application. This makes it fast and easy to find that important contact. The
personal directory can be synchronized with Outlook contacts, and the corporate directory links with the
OpenWorX directory via LDAP.
Once you locate a contact, “My Call List” makes it quick and easy for you to connect. With a single click you
can make a phone call, launch an email, start a voice, video or collaboration session, or initiate an instant
message. You simply choose the contact vehicle and launch it directly from a custom made contact list, or
from any directory.
Communications Portal logs incoming and outgoing calls, whether dialed using the Communications Portal, or
by hand. You can sort all call logs by Name Number or Time, and easily return the call by clicking on the
number.

My Controls
The “always available” tool bar interface of “My Controls” gives the user the flexibility to change status and
contact numbers on the fly. The “always available” tool bar interface of “My Controls” gives the user the
flexibility to change status and contact numbers on the fly.
The user can make a call using the number entry box in the “My Control” toolbar. They don’t have to enter 9
or 1, the OpenWorX dialing rules take care of it automatically. These intuitive controls give the user the
functionality they need to stay in touch, and be more productive in a fast paced business environment.

Contact Rules and Methods
Communications Portal employs user based contact rules and methods. Users can create methods for each
status and by the device they want to be contacted by, depending on who is calling them. Contact rules and
methods can be customized by the user to reflect how they wish to be contacted. Up to 6 different methods
can be established for any status and each rule can be assigned to any caller or groups of callers. Outside
calls are automatically routed to the next device, and inside callers can forwarded the call to the next
available device.

My Outlook
The portal mail client is integrated with MS Outlook. This gives the user the ability to manage messages and
other personal information without leaving the portal. Calendar link controls are built-in to automate the user’s
status by scheduled events. “My Outlook” synchronizes with additional Outlook functionality, making it easy to
initiate tasks and manage your mail.

Communications Portal: Benefits
Improve efficiency by combining all telephone and messaging activities within a single
application

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OpenWorX® Communications Portal allows staff members to manage all of their communications
needs within a single window. This allows individuals to place calls, return calls, send and receive
email, manage their contact information and check messages without having to switch
applications. Fewer applications streamline work processes and minimize confusion.

Enhance customer satisfaction with call management
o

Staff members can identify calls from priority customers and have them routed based on the
source, giving the individual the flexibility to handle these calls quickly, even when away from the
office. Key individuals remain available to callers regardless of their location, so mobility isn’t a
hindrance to productivity.

Improve organizational productivity
o

The ability of the OpenWorX Communications Portal to track location status can help the entire
organization stay better connected. The ability to locate key personnel who are often away from
their desks for a variety of reasons for a period of time can keep processes within the
organization moving to enhance everyone's productivity.

Improve communications for remote workers
o

OpenWorX Communications Portal is browser-based and requires network access for full
operation; it is an ideal application to help keep remote staff members in touch with the rest of the
office. Remote users can have all the same capabilities as local users, including managing
contacts, setting and viewing location status information and use of both corporate and personal
directories.

OpenWorX: Dialer
NEC’s OpenWorX Dialer allows any user in the UNIVERGE NEAX 2000 IPS to use his or her
Dterm to electronically search for the extension of another person in the UNIVERGE NEAX
2000 IPS, even while the user is on the phone. Once the listing is located, a simple button
press is all it takes to make the call or initiate a transfer. Instead of thumbing through a paper
directory, the user now accesses the online up to date directory by pressing a function key on
their Dterm. Centralizing the database also ensures that all extensions are easily added and/or
updated; no matter how often updates are made. Also, the user can query using division and
department such as would occur in a multi-site corporation.
Dialer: Database
The database used for Dialer is the same as the Business Attendant System database, which allows
Dialer users to have the same information available to the attendants.

Dialer: Benefits
Saves the attendant time by curtailing the need for employees to dial ‘0’ to get a phone number
or extension of someone in the company directory.
Provides a centralized company phone directory.
Eliminates outdated printed copies of the company directory.

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OpenWorX: Location Status Information (LSI)
LSI is a useful service that allows other system users to obtain information regarding your
current location and status when you are away from your office. This can be done through any
Dterm phone that has the associated function key programmed. The user can:
Set, Modify, or Delete their location information.
View the location information of your colleagues.
Query an extension to determine whether location information is set, and call that extension, if
the person is available.
Check the status of other users simply by dialing their extension, if configured to do so.

LSI: Database
LSI uses the OpenWorX database. This common database allows for other applications such as BAS,
INASET or Desktop to access LSI information.

LSI: Benefits
Never have to guess about someone’s whereabouts.
Know when someone is expected to return and how to contact them while they are away.
Securely set your own location through any Dterm phone in the office.
BAS integration allows attendants to check status via their directory.
Desktop integration allows all employees to view the status of other employees via their web
browser.
INASET integration allows INASET users to see employee statuses in their view of the
corporate directory.

OpenWorX: Message Reader (MR)
Message Reader is a convenient application that allows users to retrieve their messages taken
by the attendants from any Dterm phone set enabled with the proper OAI function key. The
messages can be sent from BAS, INASET or Desktop.
MR: Functionality
Message Reader is quite easy to use. Simply go to any Dterm phone set that has the Message Reader
function key configured, then press the key and enter the extension for which you wish to read
messages. In the event more than one user is assigned to a phone, the user may choose from a list of
employees to designate for whom to read messages, and then enter the proper password, if required. If
no extension number is entered, Message Reader will default to the extension you are using for this
connection. Once having entered the extension and the password, the user may easily peruse all their
messages and after a message is read, it may be deleted or saved. If the person leaving the message
left a return phone number, Message Reader can even dial the call back number automatically!

MR: Database
MR uses the OpenWorX database. This common database allows for other applications such
as BAS, INASET or Desktop to access LSI information.

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MR: Benefits
Read your messages even when away from your desk, from any Dterm phone set.
Save valuable operator time because they no longer have to read the message back to the
employee.
Automatically dial call back numbers, preventing paper clutter or misdialed numbers.

OpenWorX: Short Text Messaging
Short Text Messaging (STM) is a service running on the OpenWorX Server designed to
facilitate the delivery of text messages to NEAX Wireless Dterm PSIII handsets on the
UNIVERGE NEAX 2000 IPS. This is accomplished with either, Simple Mail Transfer Protocol
(SMTP) on a Local Area Network (LAN), or Telelocator Alphanumeric Protocol (TAP) Version
1.8. This application turns your PSIII handsets into one-way internal text pagers in addition to
their normal functionality. The interface allows for an audible alert when messages arrive.
Users should be aware that the volume for the text message arrival notice is the same as the
ringer volume. In addition, if the PSIII is set to vibrate, it will also vibrate when a text message
arrives. Messages can be stored for later reading as well as deleted. Messages cannot be
received while the user is on the phone.
STM: Functionality in an SMTP Environment
The STM server can be installed on your enterprise network as an SMTP mail gateway. In this
configuration, users simply send messages up to 128 characters (always the limit of text messages, no
matter what SMTP set up used) to the address PagerID@domain.com. 2000@wireless.user.com is a
good example. The server will then parse the Pager ID out of the heading and send the message to
the wireless handset with that extension. NEC has verified this functionality on Microsoft Exchange®
and Lotus Notes®.

STM: Functionality in a TAP Environment
STM can also be configured to use an RS-232c TAP interface for sending or receiving alphanumeric
messages to the user’s wireless handsets. Thus a TAP enabled pager program can be used to send a
message.

STM: Benefits
Allow users to send messages directly to users when not available for call transfer.
Short messages can be left as text pages instead of voice mails.
Internal Text messages can be sent directly to wireless handsets via email.
Facilitates quick call back. “Call me at 555-1212, Bob.” Can immediately be returned via the
same device.
Automated reminders can be set up via your email to be sent directly to your PSII.

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Chapter 11 Open Application Interface

OpenWorX: Incoming Call Assistant (ICA)
Incoming Call Assistant (ICA) is a service that gives Dterm users a powerful tool to manage
incoming and outgoing calls. These functions can be accessed with the simple touch of an
associated function key.
ICA: Functionality
The user can:
View the caller’s name and number on the LCD display for any line or sub-line on the user’s
Dterm, even while currently on another phone call. The name lookup comes from the Personal,
External, or Corporate Directory in OpenWorX. The above order is also the search order
priority.
Check for past or current calls to the user’s Dterm on any Dterm in the system with an ICA
enabled function key.
Look up past callers and return their calls with the information stored in ICA’s database. This is
known as the inbound call log. It will store up to 99 entries.
Check list of calls made by the user. This is called the Outbound Call Log. Up to 99 entries will
be stored.
Supports use of * and # in the user extension with the limitation that the user must be at their
physical device to log in if their extension contains either a * or a #.
Determine if a call was forwarded or transferred.

ICA: Functionality
ICA uses the same database as the entire OpenWorX suite of applications. The following fields from
the OpenWorX Database are accessible.
Caller name (inbound and outbound call log)
Caller number (inbound and outbound call log)
Date and time of each call (inbound and outbound call log)
Whether or not the call was transferred and by whom (inbound call log)
Whether or not the call was forwarded and by whom (inbound call log)
The direct number dialed by the caller (inbound call log)

ICA: Benefits
Able to identify callers on prime lines and sub-lines before they are answered, even while the
user is still on another call.
Identify past callers quickly without having to listen to voice mail.
Identify past callers when you are not at your desk.
Return calls quickly with dial back feature.

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ICA: Comparison of ICA and IPS Functionality
Feature
Call Log Capacity

OpenWorX ICA
99 Inbound max
99 Outbound max
(System Level Setting)
Manual, TAPI, OAI, LNR

2000 Caller ID
24 Inbound, or
19 Inbound + 4 Outbound

Call Log Information
displayed

Number, Name, Date, Time,
Transferring/Forwarding
number, Line rung

Number only

Ringing Line
Information
Displayed

Automatic, Name & Number

Prime Line- Automatic Sub
line – Manual, key press
shows Name & Number

Caller ID “Name”

Searches Personal,
Corporate, and External
directories for matching
name to Caller ID number

Displays Caller ID name and
number form Telco. CID
name can be overwritten by
system defined name for
1500 numbers

Dialed call types
saved in Call Log

Manual

ICA: ICA vs. Camp On
It should be noted that ICA and Camp On have similar end goals. With camp on, the user
receives notification of a second call that is “camped” to their extension. They have the choice
of answering this call and putting the original caller on hold. However, Camp On is limits the
user to 2 total callers.
ICA can achieve the same thing through multiple sub lines. The user rolls their forwarding of
busy calls to the next sub line. ICA then gives notification of the information of the incoming
caller. From there the user can decide whether or not to answer the call or let it continue and
roll to voice mail.

OpenWorX: Group Call Forward Control (GCFC)
Group Call Forward Control is an application that allows forwarding patterns of individuals or
user defined groups to be forwarded based on a schedule. Complex schedules are supported,
that allow for recurring forwarding on a daily, weekly, monthly or even yearly basis. GCFC
supports call forward busy, call forward no answer and call forward all calls. It is possible to
forward entire divisions within a company, such as after 5:00 PM all calls are forwarded directly
to voice mail.

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GCFC: Functionality
The user can navigate completely through all the tasks required of the GCFC administrator with a
simple point and click of the mouse. The administrator can also see a summary of what is already
configured within the application. The administrator can modify, define and delete user accounts here.
There are several levels of users.
Administrator – can access and maintain all user accounts, groups and call forwarding
schedules within GCFC.
Premium User – can create and maintain sets of groups. Including adding, modifying and
deleting group members, as well as controlling their call forwarding schedules.
Regular User – can view group and call forwarding jobs within GCFC, as well as maintain a
group if given permission by the group owner
The users of the groups can also be given or denied permission to group level. There is no limit to the
number of users to each individual group, nor is there a limit to the number of people who can modify
the group members or call forwarding patterns. Jobs can be configured to occur only once or recur at
specific time intervals. These time intervals can be as small as forwarding all calls to voice mail at 5:00
PM every weekday, to as broad as forwarding all calls to an automatic announcement for holidays that
occur once a year

GCFC: Functionality
Allows calls for individuals or groups of people based on schedules.
No more forgetting to un-forward your phone after returning from vacation or a business trip.
Insure privacy for users, such as in a hospital or dormitory.

OpenWorX: Personal Call Assistant (PCA)
In NEC’s continuing effort to bring better efficiency to the standard telephony user’s desktop,
Personal Call Assistant was created. PCA includes functionality like Desktop and INASET in a
much more convenient form. PCA replaces Clipboard Dialer (the ability to dial from any
application that supports the Windows Clipboard). PCA brings screen pops to the user
desktop, via text to speech agent, small pop box, or specially written customized pops. The
text to speech agent and small pop box are standard features. The customized pops must be
specially written. It is recommended that PCA’s server reside as an intranet application with
clients treating it as a Trusted Site in their internet options.
The icon is the same as Clipboard Dialer, a small yellow phone in the system tray. This icon
will turn red when the user has missed calls. Hovering over the red icon with your mouse will
list the number of calls that have been missed.
The user can then access their inbound call log to see a list of the calls that have come in to
their phone The name of the caller will be there if that person is in the OWX database. The
inbound number, whether extension or caller id, is recorded as well as the time and date of the
call. A day of the week will be shown for the last 7 days of calls. Calls going back more than 7
days will show a date such as 1/10. Clicking on the call number will return the call with the
correct dialing rules. The user can also check their outbound call log in the same way.

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If the user simply wishes to find someone internal to the company they can use the corporate
directory. Clicking on the telephone icon next to the person will dial that person. Dialing rules
are applied automatically. Other numbers can be stored and displayed this way, as well as
email and pager. These fields just like Desktop can be masked from certain users. Searching
can be done by entering a letter combination and hitting GO. You can search by Full Name,
Last Name, First Name, or Extension. In addition to the corporate directory, each user gets a
Personal Directory. This directory is stored on the server, but is only accessible by that user
and operates the same way as the Corporate Directory. The Personal Directory will also
support importing from a CSV (comma separated values) text file. PCA also includes the
ability to synchronize the user’s Personal Directory with their Outlook Contacts. The mappings
of the fields are controlled by an XML file (mappings.xml in the PCA directory) which is easily
modified to suit the individual’s needs.
PCA: Limiting User Views
Occasionally it is necessary to make certain fields un-available except by certain users. For example,
the company may not wish for non-executives to have access to employee home phones. Because of
this, the OpenWorX administrator can put the login Ids of Desktop users into groups that do not allow
the viewing of specific fields.

PCA: OpenWorX Desktop
Desktop is now a part of Personal Call Assistant as a standard feature. Desktop is designed for
general day-to-day UNIVERGE NEAX 2000 IPS users who have a PC with a web browser. No
application installation needs to take place on the PC. Access is directly via the web.
Desktop Directory
The user can look up individuals within the company directory, and access their speed dials. From each, they
can dial, and from within the corporate directory, a user may send a Short Text Message (if STM is installed),
email, or OpenWorX Message.

Personal Directory
In addition to the standard OpenWorX corporate directory, Desktop gives users an additional Personal
Directory. The Personal Directory is different for each individual user, but the Corporate Directory is common
among all users. For example, the user could have an entry for their Mother, and it would only be for use with
their individual phone.

Desktop Call History
The lists of calls that have been received by the phone are displayed in the “Inbound” Section of this screen.
The number, Name, and Date are included. The Name is only included if there is a match for the number in
the database. The user can also click on the telephone handset icon next to the number to return the call
from their Dterm set.
In addition to searching the corporate directory, ICA will now search the Personal Directory of that station as
well. For example, if 214-555-1111 calls one individual who has “Mom” defined in their Personal Directory for
that number, the ICA pop and record in Desktop will say “Mom”. For another person who does not have
“Mom” defined in their Personal Directory, no name will be attached, or can also have a different number
defined for “Mom. ICA also indicates the phone type found in the Personal contact, using appending “-H” for
home, “-M” for mobile or wireless, and “-W” for any business line.
The “Outbound” section is similar to the “Inbound” section, except it records calls made (by Manual Dialing,
OpenWorX Dterm Applications, Desktop, or TAPI) at the telephone, not calls received.

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Desktop Location Status
With Desktop, the user can set their status, including return date, return time, and alternate number with the
click of a mouse. In addition, Desktop allows for the use of Custom statuses, instead of only the 10 predefined messages available via Dterm operation.

Desktop Message Reader
Previous to Desktop, the only way for an individual to check messages sent by Message Reader was by
using the key on the Dterm, or by calling the attendant and having the message read back to them. Now, the
user can use a Web Browser with Desktop to check their messages. Also, as mentioned in the Directory
section, the user may send these messages to other users of Message Reader. No longer is it just the
attendants who can send messages. The message can also include a number and the Desktop user simply
needs to click on the telephone handset icon next to the number to return the call from their Dterm set.

PCA: OpenWorX INASET
The INASET is now a part of Personal Call Assistant as a standard feature. NEC’s INASET is a
powerful IP terminal and is enhanced even further by the addition of the OpenWorX application suite.
This suite enables INASET Application Services mode, adding a great deal of features and
functionality.
Home Application
This is the base application that is loaded directly to the INASET’s memory. It allows for the basic
functionality expected from a standard IP Dterm. The main difference a user will notice with this screen is the
different application control buttons at the bottom of the INASET screen. Without OpenWorX, these buttons
are static and controlled by the INASET’s local memory. With the addition of the OpenWorX Network
Services mode, the server configures these buttons. The modification of a simple XML file on the server is all
that is required to move, add, or delete additional application buttons. These additional application buttons
can be used for many things, such as custom developed applications and direct links to web sites.

Corporate Directory
The Corporate Directory pulls the information directly from the OpenWorX Employee database. The user can
search for specific people and dial using this directory. Once found in the directory the user may also send a
text message (if Message Reader is installed), send a Short Text Message to their wireless PSIII (if Short
Text Messaging is installed), or simply view the entry to see other information or alternate dial-able numbers.

Location Status
This interface allows the user to set their status in the OpenWorX database. It is then accessible by the
operator. Also when another Dterm user calls the person, they will receive their status on their Dterm display,
along with Return Time, and Alternate Number if provided.

Message Reader
Message Reader allows the user to read messages sent to them by the operator or other users.

OpenWorX: Name Display
Name Display is an application that synchronizes the UNIVERGE NEAX 2000 IPS name
database with the employee portion of the OpenWorX Database. This sync is real time.
In the case of two people sharing a single extension, the application will search through,
Department, Division, and Company in that order. The first one that matches for both
employees will be the one displayed.

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OpenWorX: Multiple CCIS Node Configuration
OpenWorX is capable of hosting applications for multiple UNIVERGE NEAX 2000 IPS systems
on a single server. All instances of this require a viable OAI connection between the
OpenWorX server and the UNIVERGE NEAX 2000 IPS. See the bandwidth requirements list
previously in this chapter to make sure your connection is sufficient.
The applications once equipped with the remote TCP/IP connection and multi-node software
operate as if they are in a single UNIVERGE NEAX 2000 IPS, except where noted below.
BAS: CCIS Network Information
The Business Attendant System can run in a CCIS network environment with the following limitations.
Calls transferred from the BAS Client across a CCIS Link will not recall to the BAS operator.
Using multi-node capabilities, the operator can now view station data via the directory or speed
dials to extensions in other IPS systems. The other IPS systems must have OAI and a TCP/IP
connection back to the OpenWorX server. Additional OpenWorX software is also required per
IPS to be added on the server.

Business Receptionist: CCIS Network Information
The Business Receptionist can run in a CCIS network environment with the following limitations.
Calls transferred from the Business Receptionist across a CCIS Link will not recall to the
operator.
The Dterm across a CCIS Link cannot be monitored by the Business Receptionist system and
the monitored speed dials and status checks in the directory will not function for those
extensions.
The system will force supervised transfers to CCIS extensions to ensure callers do not get
transferred to a busy remote extension.
The message-waiting lamp cannot be controlled on a remote Dterm.
Note: Business Receptionist does not have the same multi-node capability that BAS does

OAI Application Software Development
For users wishing to develop their own applications, NEC offers a complete library of
development tools. For further information, contact the OAI Product Manager.

System Specifications
Required Equipment for OAI
Qty

Remarks

PN-CP24-C (MP)

Equipment

MP Card

PZ-M606-A (ETHER)

Ethernet Control Card

• 10BASE-T Cable

10BASE-T/100BASE-TX
Cable between Ethernet
Card and LAN

One card per system Mount on
PIM0
Mount on MP card One card per
MP
Cable length:
Maximum 100 m (328 ft.)

• 100BASE-TX Cable

Description

As
required

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Chapter 12 Automatic Call Distribution (ACD)
with MIS
Automatic Call Distribution (ACD)
An Automatic Call Distribution (ACD) feature permits incoming calls to terminate to a
prearranged group of stations. Calls are distributed in the order of arrival to idle terminals
within the group, based on which terminal has been idle the longest period of time. Stations
may log on/log off from the ACD group. Supervisor stations may monitor conversations of
agents.
In addition to basic ACD, NEC has a Microsoft® Windows 2000 Professional, Windows 2000
Advanced Server or Windows 2003 Server ACD/MIS system (CallCenterWorX ACD/MIS)
that enables a user to efficiently manage a call center. NEC offers an Infolink Application
(QueWorX ) that provides IVR, Internet and Desktop integration for UNIVERGE NEAX 2000
IPS Call Centers. Finally, Q-Master which offers a modular call center environment allowing
for more flexible options by choosing just the application needed.
The MIS system analyses statistical data received from the ACD, displays real-time activity
screens, and generates and prints graphs and reports of the activity.

Basic ACD
The UNIVERGE
package.

NEAX

2000 IPS has basic ACD that comes standard with any basic system

Service Conditions
1. A maximum of 16 ACD groups can be assigned per system. Each ACD group is assigned a pilot
number. Calls directed to the pilot number are directed to that ACD group.
2. The maximum number of stations in an ACD group is 60. The maximum number of ACD groups in
the system is 16. If ACD-MIS is used, the maximum number of ACD stations is 60.
3. Assignment of ACD groups is performed from the Maintenance Administration Terminal (MAT) or
Customer Administration Terminal (CAT).
4. ACD groups consist of a pilot station and one or more member stations. Hunting is initiated in a
circular fashion, and then based on which member has been idle the longest period of time.
5. If all stations within the ACD group are busy, incoming calls may be serviced in the following ways:
remain in queue until an agent becomes available (Ring back Tone provided)
immediately overflow to another group, to a station, or to the Attendant
remain in queue until an agent becomes available (Delay Announcement or Music on Hold
provided)
remain in queue for a preset time (Ring back Tone, Delay Announcement, or Music on Hold
provided), and then overflow to another group, to a station, or to the Attendant
6. When the pilot station has set Call Forwarding – All Calls, incoming calls to the ACD group will be
transferred to the destination of that Call Forwarding – All Calls setting

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Service Conditions (Cont)
7. An ACD group number can be used as the destination station of Direct Inward Termination (DIT), or
as a designated Night Service station.
8. An ACD group number can be assigned as the destination station of Off-Hook Alarms, Priority
Calls, and Attendant Night Transfer.
9. ACD group pilot numbers should not be placed in Station Hunting groups. The Station Hunting
feature would take priority over the ACD function.
10. Two types of traffic measurements can be provided for ACD:
a. ACD group Peg Count
Count of incoming calls
Count of answered calls
Count of abandoned calls
Count of waiting calls
Count of all busy calls
b. ACD station Peg count
Count of answered calls

11. Upon initial installation, or after a system initialization (reset), each agent must lift and
restore handset (of their station) to begin receiving calls for the ACD group.

Basic ACD Features
Busy In/Busy Out – ACD
This feature allows an agent in an ACD group to log their station onto or off of the group. This
allows the system to control whether a call directed to the pilot number of the ACD group goes
to that station or not. This prevents incoming calls from being directed to stations at which no
agent is available.
Call Waiting Indication – ACD
This feature provides a visual indication when an incoming call to an ACD group is placed in
queue, due to an “all agents busy” condition. On external relay controlled indicator or an LED
on a Multiline Terminal can be used to provide Call Waiting Indication.
Delay Announcement – ACD
This feature allows the system to provide a recorded announcement to an incoming caller
placed in queue to an ACD group. A single announcement, or two separate announcements,
can be provided.
Hunt Past No Answer – ACD
This feature allows calls targeted at an ACD group to hunt past an agents station, after a no
answer condition, if the agent forgets to log off of the group and the agent is unable (or not
available) to answer the call.

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Immediate Overflow – ACD
This feature allows a call directed to an ACD group to immediately overflow to another ACD group,
upon encountering an “all agents busy” condition.

Priority Queuing – ACD
This feature allows the system to prioritize incoming calls by trunk route and on a per station basis,
when the call enters an ACD queue. When a call is a considered as priority it is placed at the beginning
of the queue.

Queue Size Control–ACD
On incoming DID/Tie line calls the system can be assigned a threshold that limits the number of calls in
queue. When the queue size threshold is exceeded, incoming callers are connected to busy tone.

Silent Monitor – ACD
This feature provides the ACD group supervisor with the ability to monitor a call to an ACD agent. The
silent monitor function gives no indication (as an option) to either the agent or the calling party.
Note: The use of monitoring, recording, or listening devices to eavesdrop, monitor, retrieve, or record telephone
conversations or other sound activities, whether or not contemporaneous with its transmission, may be illegal in
certain circumstances under federal or state laws. Legal advice should be sought prior to implementing any
practice that monitors or records any telephone conversation. Some federal and state laws require some form of
notification to all parties to the telephone conversation, such as using a beep tone or other notification methods or
require the consent of all parties to the telephone conversation, prior to monitoring or recording a telephone
conversation. Some of these laws incorporate strict penalties.

CallCenterWorX ACD 3.0 for Business
CallCenterWorX ACD 3.0 for Business is a suite of hardware and software solutions that are
focused on customer care. Included are the Automatic Call Distribution (ACD) system and the
Management Information System (MIS), elements that streamline call flow and workflow. Its
routing capabilities allow businesses to connect callers with the most appropriately skilled
agents in the shortest time possible, minimizing a caller’s wait for assistance and improving
customer satisfaction. Its real-time and historical reporting abilities provide call center
managers and supervisors a tool for maximizing agent productivity, assuring quality
performance in handling the incoming and outgoing call volume in the call center.
CallCenterWorX-Business helps businesses make a good first impression on new customers,
better manage existing customer relationships, and inspire customer loyalty, leading to
increased revenue potential.
CallCenterWorX ACD 3.0 for Business is a Microsoft Windows® based software system that
enables you to efficiently manage your call center. It does so by providing a complete and
flexible feature set, allowing you to customize the ACD based on your particular business
needs. In addition, it provides a powerful Graphical User Interface (GUI) to a UNIVERGE NEAX
2000 IPS ACD System that allows you to program additions, deletions, and changes via your
computer. You can also query and print ACD data from CallCenterWorX.
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The CallCenterWorX ACD 3.0 for Business suite includes the Automatic Call Distribution
(ACD) system and the Management Information System (MIS 3.0). The ACD system
supplements the call processing capabilities of the PBX, with typical call routing designed to
connect each caller with the most appropriate agent to handle their call in the shortest time
possible.
When the CallCenterWorX ACD system accepts a call, it can be configured to play a message
to the caller and route the call in a pre-specified order to a queue of waiting calls for a
particular split (group of agents). The queued call is then answered in sequence by the next
available agent in that split. The Call Control Vectors of the CallCenterWorX ACD system allow
great flexibility in routing calls, permitting alternate solutions to fit almost any situation.
As each call progresses through the system, the CallCenterWorX ACD system provides data
to the CallCenterWorX MIS 3.0, which translates the call-handling data into meaningful
statistics. The MIS uses these statistics to offer call center managers and supervisors access
to real-time and historical data concerning incoming and outgoing call volume.
The CallCenterWorX MIS Status Screens (Real-time Screens) provide color-coded, up-to-thesecond views of agent, queue, and system activity. Statistics on the call volume and agent
effectiveness are computed in real-time and are displayed for the current hour and day on
supervisors’ screens. Optional wall display boards can show current statistics and messages
to call center personnel.
The call statistics are also collected in the MIS database and used to generate reports in text
or graphical format. These reports and graphs can be printed for a view of activities ranging
from a graph updated in the last second to a retrospective report ranging back up to 1.5GB
worth of data.
CallCenterWorX ACD Features
1. Runs on the Windows 2000 Professional operating system using Client/Server architecture, making
information and processing directly available from the desktop interface.
The MAT Client as a remote PC runs on Windows 2000 Professional or Windows XP.
2. A capacity of 10 to 120 positions, depending upon licensing, application, and PBX platform.
3. Can support up to nine tenants.
4. Uses a GUI interface to make commands and controls easily accessible to the user.
5. Routes incoming calls using:
User-defined sets of call handling instructions named Call Control Vectors (CCVs), week
schedules, and holiday schedules.
A system of priorities and call overflows designed to deliver calls rapidly and efficiently to the
appropriate split or agent.

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6. Has dock able toolbars, which can be dragged to other locations on the working screen. The user
can also build a personal toolbar of most frequently used icons.
7. Provides toolbar icons and tool tips for all MAT commands, for opening report views, and for online
Help and context-sensitive Help.
8. Provides data entry screens with a choice of background color pattern for each screen.
Data entry screens can accept new data or revise previously entered data with immediate
confirmation of the action. These screens list operational data in drop-down menus.
Each data entry screen can open to a report view showing the currently programmed data.
The font style and size and the column headings in the report can be reset to suit your
preference.
The report view can open to a print preview showing the report as it would be printed, and
can send the report to a printer. A report can be saved to a data file, saved to a Microsoft
Excel file, or exported to a Microsoft Access database.
9. An optional workbook area showing a tab for each open screen with the MAT command icon for
that screen. A report icon on top of the command icon for those screens open to the report view.
Multiple screens can be open simultaneously. A screen is made the active screen by
selecting the appropriate Workbook tab.
10. Has an optional status bar, which reflects the state of the active screen.
11. Supports 6 languages (English, Japanese, French, German, Italian and Spanish) for display to the
agent’s Dterm position.
12. Uses a Trace Client to view ACD call processing on screen. CCV programming diagnostics let you
monitor the call flow in real-time to design or re-engineer your installation.
13. ACD client MAT provides the ability to program multiple CallCenterWorX ACD`s from one GUI in a
network
14. List-up includes the ability to view multiple reports and to export these reports in MS Access or MS
Excel formats
15. Configuration of trunk group data and holiday calendar performed by single point and click interface
16. System Data configuration allows the ACD to be customized for specific features including Info link
and MIS message generation, debugging tools, and phone-set operation
17. Trace Client view provides the ability to see ACD call processing on screen. Call Control Vector
(CCV) programming diagnostics allow the call flow to be monitored in real-time, aiding in the design
or re-engineering of call center installations
18. A Windows® event viewer provides access to critical ACD event messages such as ACD state
changes and helps technicians identify other critical events such as an inability to connect to the
PBX or indicating that the system security key is not attached
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Chapter 12 Automatic Call Distribution with MIS

CallCenterWorX MIS
The CallCenterWorX Management Information System (MIS 3.0) offers call center managers’
access to the most important real-time and historical data available for workgroups. With this
tool, you are able to maximize agent productivity and assure quality performance in handling
the incoming and outgoing call volume of the center.
CallCenterWorX MIS has a menu-driven user interface. All of the menus show a list of choices
available under a particular topic. Each choice is preceded by a number, which is entered at
the Enter Selection prompt, followed by pressing the Enter key. CallCenterWorX MIS responds
by displaying the requested menu screen, by using one of its prompting methods to ask for
more information, or by displaying the appropriate data screen.
CallCenterWorX ACD 3.0 for Business Turnkey Software Only is available packaged for 10,
20, 30, 40, 60, 80, 100, and 120 Agents. Each combines both the ACD and MIS software on a
CD ROM and comes with a required security key for the main CallCenterWorX PC. The
security key attached to the main CallCenterWorX PC provides the appropriate agent capacity
as well as the main CallCenterWorX supervisor position. Individual security keys can also be
purchased for applications requiring additional supervisor PCs.
CallCenterWorX ACD 3.0 for Business Turnkey Packages

Business Turnkey Packages
CallCenterWorX-B ACD (3.00) (10 Agent Pkg)
CallCenterWorX-B ACD (3.00) (20 Agent Pkg)
CallCenterWorX-B ACD (3.00) (30 Agent Pkg)
CallCenterWorX-B ACD (3.00) (40 Agent Pkg)
CallCenterWorX-B ACD (3.00) (60 Agent Pkg)
CallCenterWorX-B ACD (3.00) (80 Agent Pkg)
CallCenterWorX-B ACD (3.00) (100 Agent Pkg)
CallCenterWorX-B ACD (3.00) (120 Agent Pkg)
CallCenterWorX ACD 3.0 for Business Software Only Packages

Business Software Only Packages
CallCenterWorX-B ACD (3.00) (10 Agent) SW Only
CallCenterWorX-B ACD (3.00) (20 Agent) SW Only
CallCenterWorX-B ACD (3.00) (30 Agent) SW Only
CallCenterWorX-B ACD (3.00) (40 Agent) SW Only
CallCenterWorX-B ACD (3.00) (60 Agent) SW Only
CallCenterWorX-B ACD (3.00) (80 Agent) SW Only
CallCenterWorX-B ACD (3.00) (100 Agent) SW Only
CallCenterWorX-B ACD (3.00) (120 Agent) SW Only

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CallCenterWorX MIS 3.0
CallCenterWorX MIS 3.0 runs on the Windows XP and Windows 2000 operating systems. It provides
MIS services for both CallCenterWorX – Business and CallCenterWorX Enterprise.
The CallCenterWorX ACD 3.0 for Business suite includes the Automatic Call Distribution (ACD) system
and the Management Information System (MIS). The CallCenterWorX ACD system supplements the
call processing capabilities of the platform being used. Typical call routing is designed to connect each
caller with the most appropriate agent to handle the call in the shortest time possible, avoiding long wait
times and getting the caller directly to someone who can deal with their request.
As each call progresses through the system, the ACD system provides call-handling data to the MIS,
which translates this data into meaningful statistics. The MIS uses these statistics to offer call center
managers and supervisors access to real-time and historical data concerning incoming and outgoing
call volume in the call center.
The CallCenterWorX MIS Status Screens (Real-time Screens) provide color-coded, up-to-the-second
views of agent, queue, and system activity. Statistics on the call volume, and on the effectiveness of the
agents handling those calls, are computed in real-time and can be displayed up-to-the-second on
supervisors’ screens. Optional wall display boards can show current statistics and messages to call
center personnel. The call statistics are also collected in the MIS database and used to generate
reports in text or graphical format. These reports and graphs can be printed for views of activity ranging
from a graph of today’s activity to a retrospective report covering previous months.
The following features are available with the purchase of CallCenterWorX MIS 3.0.

•
•
•
•
•
•
•
•
•
•
•

True Multi-Tenant Support
Increased Multi-Split Agents (from 4 to 16 split monitoring)
Microsoft MSDE
ODBC Compliant
Configurable in 15 min. / 30 min. / 1 hr Increments
Conversion utility to migrate proprietary database to ODBC database
Backup and Restore
1.5 GB database storage
New "System Agent Detail Status" Screen
Configurable Agent State Colors
Retains Real-Time Screen Configuration

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Chapter 12 Automatic Call Distribution with MIS

Business System Capacities
Item ACD

Capacities
CallCenterWorX
CallCenterWorX
ACD 3.0
MIS 3.0

Tenants per System
9
Splits per ACD Tenant
Note 1
Note 1
Splits per System
150
150
Splits per Logon ID
32
32
Active ACD Agents per
120 Note 2/ Note 4
120 Note 2
System
Trunks per System
255 Note 4
255
Maximum Queue Depth
300
–
Priority Levels 1
250
Note 3
Maximum Calls in
300
–
Queue per System
Transfer to PBX
120
–
Numbers
Agent Logon ID Codes
400
400
per System
Agent Logon ID Code
9
9
Digits
Statistics Report History
–
1.5GB
Limit
MIS per System
8
8
Pilot Numbers
300 Note 4
300
Tally Code Names per
–
100
Split
Tally Code Names per
–
3000
System
Busy Hour Call Attempts
–
Note 5
(BHCA)
Note 1: The total number of splits per system (150) is shared among the total number of tenants (up to 9).
Note 2: Agent positions available depend on the installed security key’s capacity for logged-on agents. The
total number of agents is shared among the total number of tenants (up to 9). Security keys are available in
the following capacities: 10, 20, 30, 40, 60, 80, 100, & 120.
Note 3: The MIS tracks Split queue calls as “High” and “Standard.”
Note 4: Dependent upon the CallCenterWorX configuration formula based on the OAI monitoring limitations:
(Agents x 2) + (Pilots and/or Personal Pilots) + Trunks + IVR Ports must be equal to or less than 256.

Note 5:

BHCA (Busy Hour Call Attempts)
Number of PIM’s 1PIM
2PIM
3PIM
4PIM
Business or
Hotel/Motel
with ACD/OAI

Page 12- 8

Max. 2000 BHCA
(No FP)

Max. 4000 BHCA
(One FP)

5PIM

6PIM

Max. 5000 BHCA
(Two FPs)

7PIM

8PIM

Max. 5500 BHCA
(Three FPs)

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PBX Requirements
PBX Requirements
PZ-M606-A

Remarks
On board Ethernet Interface card (Ethernet TCP/IP)

Related Documents
Title
CallCenterWorX System Manual
CallCenterWorX Features and Specifications Manual
Dterm Series E Supervisor Console User Guide
Dterm Series E Agent Console User Guide
CallCenterWorX Sales Slick
ICS ACD System Manual
Abacus User’s Guide
Navigator Configuration Guide
Navigator Reports Manual
Navigator Real Time Screen Manual
CallCenterWorX ACD User Guide

ACD Terminals
With the Dterm Series i/E ACD terminals enjoy the convenience of having the most frequently
used ACD function keys silk-screened right on the terminal.
Availability

The 32-button Series E ACD terminals are available in soft white or black. User Guides and
spare parts are also available.
DTP-32DA-1 (WH)
DTP-32DA-1 (BK)
Dterm Series E Supervisor User Guide (10-pack)
Dterm Series E Agent Console User Guide (10-pack)

Detachable dial-pads are available for Dterm Series I and Dterm IP instruments in an ACD
environment.
BS(ACD)-R

Dial pad for ACD (Available in black and white)

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Q-Master 3.1
Q-Master is a scalable contact center solution offering high-end functionality at a reasonable
price. Q-Master is completely modular, allowing contact center management to select the
components that best meet operational and budgetary requirements while retaining the
flexibility to add additional modules in the future. One of the key advantages of this is the
range and scalability of the solutions. This is the first contact center application available that is
capable of providing the same functionality to end users regardless of whether they deploy a
small, medium or large PBX or Key system using any of the UNIVERGE Family of systems or
the Elite IPK. This creates a powerful migration path never before available to our customers.
Q-Master Contact Center Solutions
Contact Routing (Q-Control)

Q-Control is the core module of Q-Master. It lets users maximize contact center efficiency by providing
dynamic control over the delivery of phone, email and web initiated contacts. Since labor is a contact
center's largest expense, this efficiency equates to significant operational cost savings. Q-Control uses
skills-based routing to match agent knowledge to the needs of different callers or groups of callers;
agents handle the inquiries they are most equipped to deal with. When all the agents in a specific
queue are busy, Q-Control automatically overflows calls to the next most suitable agent in another
queue, by matching agent skill sets to a variety of different queues using a predetermined call delivery
matrix. List of features included in Contact Routing (Q-Control):
Individual agent IDs - Agents log on and off from their telephone or PC with a personalized ID
number - the agent, not the extension is tracked.
Wrap-up codes - prompts agents to supply a Wrap-up code, providing detailed real-time call
handling information.
Work-time allowance - provides each agent with post-call processing time.
Inter-Queue Transfer - even when a call is transferred from one queue to another, callers
maintain their position in the queue based on total time waited.
Wallboard Interface - access real-time stats displayed on an electronic wallboard. A quick
glance delivers an accurate picture of current performance levels.
Specialized features for supervisors - a number of specialist features are also available for
supervisors including:
o on-demand call delivery
o remote log in/out option
o conversation monitoring/intrusion
o emergency queue re-routing
Simplified Administration – The wizards and a graphical user interface help to administer all
aspects of the system. Adjustments to call delivery patterns, queue setup, agent assignment
and other parameters can be made in a matter of seconds as updates are performed in realtime.

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Reports (Q-Control)
Because access to timely and relevant information is critical for effective contact center management, a
comprehensive reports package comes standard with every Q-Master solution, ensuring that users
have the tools to run a contact center at peak efficiency. An onboard database holds statistics on every
facet of every contact from the time it arrives at the telephony switch through to its termination (cradle
to grave), allowing managers to report on areas such as volume, agent activity, abandonment rates,
wrap-up codes and service levels. More than 120 reports are available detailing all aspects of the
contact center operation. List of feature included with Reports (Q-Control):
Detailed queue and agent reporting analysis
Business reports on wrap-ups and service levels
Ability to reformat data into graphs and tables
Single integrated reports package covering telephony, email, web-based and outbound contacts
Automatic report scheduling Simplified Wizards to configure and run reports data export to
compatible packages
Historical data archiving and retrieval
Announcements (Q-Announce)
Holding for any length of time is a frustrating experience for callers and projects the wrong image for a
business. With Q-Announce users can configure each queue to play informative, specific messages
that a hold a caller’s attention and reduce the chance that they will abandon the call. With Calling Line
ID enabled Q-Announce can recognize and play customized messages to a particular customer.
Perhaps the most powerful feature of Q-Announce is its ability to keep callers informed of their position
in the queue and the estimated wait time. This improves customer satisfaction and ensures that callers
won't hang up just before they reach the front of the queue. List of features included with
Announcements (Q-Announce):
Progress Announcements - this simple technique significantly reduces the number of
abandoned calls. Our results have proved that informed callers are 30% more likely to wait on
hold.
Automated Attendant - splits a single telephone number into numerous queues. Callers are
then prompted to select the appropriate destination from a list, reducing wait times and
connecting customers to the correct agents.
Multi-language Capability – record each queue announcement in multiple languages. The
language played to the caller is determined by the inbound number they dialed, the menu option
they selected or their calling line ID.
Audiotext - pre-recorded information menus provide answers to commonly asked questions.
Audiotext saves valuable agent time that could be more effectively spent handling callers with
complex queries.
Customer Query - identify callers based on their calling line ID or their response to an
automated query (such as “please enter your customer number”) and use this information to
play customized messages, direct the call to a specific agent or escalate their priority within the
queue.

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Agent Desktop (Q-Desktop)
As the importance of contact centers increases, so does the need to monitor operational parameters.
Q-Desktop lets managers and agents view real-time information on queue and agent performance
across all contact media, whether phone, email or the web, providing supervisors and agents with the
tools for making the right decisions at the right time.
Agents have access to a wide range of contact center functionality directly from their PC desktop,
ensuring that they can operate at optimal efficiency. Using the Agent toolbar they can:
Log into and out of queues
Request post call work time
Take a break
Wrap up or resolve a call
Raise an alert for assistance
Q-Desktop also gives agents control over phone functions (e.g., answer, transfer, conference and hold)
from their PC and identifies incoming calls by screen popping prudent information such as ANI, wait
time, queue name, as well as caller's name account number and any other desired information from QMaster or external database. Any actions taken by the agent in the contact center are reflected in realtime by Q-Desktop. These views give agents and supervisors a lot of valuable information at a glance.
List of features included with Agent Desktop (Q-Desktop):
Real-time information - view the status of each agent and queue. Supervisors can monitor
contact center service levels and act immediately when issues arise unexpectedly. Experience
has shown that agent commitment increases when staff can view each other's performance.
Your agents become increasingly goal oriented and require less supervision, resulting in
increased productivity and accountability.
Caller information display- Q-Desktop screen-pops calls as the phone or email is delivered to
the agent. Using Caller ID, users can view the caller's name and number before answering the
call, saving valuable seconds. Use the Phonebook, Q-Desktop's own database, to store special
details for a caller, which may be screen-popped with the call.
Multiple views – users can make their view of the contact center as simple or as detailed as
they wish, and flick from one view to another with a single mouse-click. They may choose to
view:
o the queues and which agents are logged into
o the individual agents and the type of call they are on
o both queues and agents
Precise and convenient dialing - point-and-click directories ensure fast and accurate dialing.
Agents can also dial directly from their PC keyboards.
Specialized features for supervisors - supervisors will always have a current picture of
contact center activity: queue status, service level, abandonment rates, traffic volumes and
agent statistics. A number of specialist features are also available for supervisors, including:
o On-demand call delivery
o Remote log in/out option to control agent status
o Conversation monitoring and intrusion
o Direct access to Q-Master's administration package - for dynamic changes
o Emergency queue re-routing
o At-a-glance icon indications of call and agent status

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Callbacks (Q-Callback)
Callback allows callers who are waiting in the queue to leave a message and then hang up. The
callback request retains its position in the queue, so when an agent becomes available the caller is
automatically contacted as if they'd stayed on the line.
Gives Your Callers Control - while some callers are content to wait on the line, many are not.
Q-Callback invites callers to enter their contact number, leave a message and hang up, without
losing their position in the queue. Callers using the Callback feature are handled just as quickly
as if they had waited on the line.
Lower abandonment rates - waiting on hold can be frustrating, costly and time consuming for
your customer. You may also pay the price for leaving your caller on hold for "too long," causing
them to hang up and take their business elsewhere.
Makes It Simple for Agents - Callback is very flexible and can be configured on a per queue
basis. Use the Callback schedule to determine times of day that you want Callback to be
available, and specify thresholds when it should apply. Agents and Supervisors can see the
number of callback requests in the queue and make the adjustments to operating parameters as
required.
Lower your network costs - benefit from substantial telecommunications cost savings, as QCallback cuts the number of calls holding on your free phone numbers.
Answers Your Callers 24 Hours/Day - Q-Callback can act as an automated answering
service, for instance, after-hours. When agents log in first thing in the morning, calls from the
night before are immediately presented to their workstation for handling. No voice mailbox
needs to be cleared and no calls have to be requested.
Multi-Media (Q-Email, Q-Chat, Q-Fax)
Use Q-Master's multimedia modules to manage these communication media within your existing
contact center infrastructure - in much the same way as you currently manage your telephony calls.
Just as phone calls, emails and web interactions are also delivered to agents via a queue as soon as
they become available. Contacts are fairly distributed among agents and monitoring and reporting on
agent performance and response time is available across all media. These modules are fully integrated
into Q-Master so individual agents can receive requests initiated via telephone, web or email. The
properties of intelligent call delivery and skills-based routing are applied when distributing any type of
contact to agents. List of feature included with Multi-Media (Q-Email, Q-Chat, and Q-Fax):
Q-Email - integrating email communication into a contact center environment provides
customers with accurate, timely responses to their inquiries in a consistently professional
manner. Monitoring and reporting of the incoming email traffic also provides a total picture of all
inbound communications.
Q-Web Callback - lets potential customers request a Callback from within a website. These
requests are routed to the most suitable agent in your contact center. The agent receives a
screen-pop with the details of the request. When they click the dial button, Q-Master
automatically establishes the call.
Q-Web Chat - offers web users real-time online interaction with a contact center via text-based
chat sessions with agents. Agents have the ability to "push" links directly to online customers.
The possibilities are endless and the impact on customer service is significant.

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Outbound Dialing (Q-Outdial)
Why wait for your customers to call? Your customers can generate additional revenue, proactively
manage customers and improve the productivity of every outbound customer contact by intelligently
managing outbound call campaigns from within the existing Q-MasterEX solution.
Blend Inbound and Outbound Calls - agents can be set up to work in dedicated outbound
queues or take a mix of inbound and outbound calls. By sharing agent resources between
inbound and outbound call activity, contact center managers can optimize agent productivity in
periods of low inbound activity.
Flexible Call Delivery Options - two types of call delivery modes are available; Power Dial and
Preview Dial.
Information on Call Outcomes - resolution and wrap-up codes allow agents to record the
outcome of every call, indicating the status and commercial outcome of a call.
Build and Manage Campaigns Easily - Q-Outdial is administered and managed using an
easy-to-use administration interface.
Fully Integrated Solution - Q-Outdial utilizes a common interface to manage and deliver both
inbound and outbound calls. Agents and supervisors can perform all required call delivery tasks
in this environment, minimizing operational complexity and training, and maximizing labor
efficiency.
Reports - allows the contact center manager to view campaign information and details of
specific campaigns in an easy-to-read format that helps manage outbound campaigns.
Interactive Voice Response (Q-IVR)
Q-IVR is fully customizable, interactive voice response technology that allows callers to interact directly
to a database through the key pad on their phone to place orders, manage their account, check the
status of an existing order or have other routine questions answered. What is unique about Q-IVR is
that it is fully integrated with the call center. List of features included with Interactive Voice Response
(Q-IVR):
Reduce call traffic to agents – enable customer to place orders, manage their account, check
the status of their order and get answers for routine questions automatically thought the use of
the IVR. By enabling this feature, agents can address more complex issues, better utilize agent
time and reduce customer wait time throughout the organization by eliminating time consuming
simple tasks that customers can now handle on their own.
Make it a 24 x 7 operation – after normal business hours customers can still utilize the IVR
capabilities, allowing automated services 24 hours, seven days a week.
Reduce call abandonment - since agents are now handling the more difficult questions, call
volume will most likely drop significantly, therefore wait times and call abandonment will
decrease.

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QueWorX4.0
QueWorX® 4.0 is a suite of customer service and productivity-oriented applications. QueWorX
utilizes NEC’s CTI link, “Infolink” to deliver advanced applications to contact centers
These advanced applications are tightly integrated and specially designed to work in
conjunction with NEC’s CallCenterWorX ACD applications for both UNIVERGE NEAX 2000 IPS
and UNIVERGE NEAX 2400 IPX systems. QueWorX integrates with CallCenterWorXEnterprise (I), & CallCenterWorX ACD 3.0 for Business & Enterprise. QueWorX provides the
same features and functionality regardless of the ACD platform it accompanies. In addition,
QueWorX may be installed on the same server as the CallCenterWorX ACD 3.0 for Business
or Enterprise.
QueWorX 4.0 Licensed Applications

Automated Attendant
Immediate Callback
Scheduled Callback
Web (Internet) Callback
Estimated Time to Answer
Queue Depth
ANI/Are Code Routing
Account Code Routing
Screen Pop
Soft Phone
QueWorX 4.0 Features:

Modular Licensing
Multilingual Announcements
Global Repeat Verification
Customer Profile Records Database
Unlimited Multi-level Automated Attendants
Customizable Announcements
Customizable Reports
The above items are detailed below:
Auto Attendant

The Auto-Attendant provides basic call routing functions to the caller. The caller receives a message
with up to ten call routing options and the caller presses a key to initiate a selection. The caller’s
selection is processed as a transfer to a destination. Additional menu choices are available for
customers with the multi-level automated attendant. The caller may be prompted for Account Code or
User ID, which will then be displayed on the agent’s Dterm phone as the call arrives. Alternatively, the
Account Code or User ID may appear in the QueWorX Agent Screen Pop. This allows the agent to
begin accessing a customer’s profile while greeting the caller.

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Immediate Customer Callback

This feature allows callers, based on their estimated time to answer, to elect to receive a callback from
an agent instead of waiting in queue. By entering their telephone number and recording a brief
message, their call will remain in queue until an agent becomes available, even though they have hung
up. The QueWorX system will automatically dial the number entered by the caller when the agent
receives the call. Bonus: The contact center does not incur toll charges for the time call remains in
queue awaiting a callback.
Scheduled Callbacks

This feature allows callers to determine when an agent calls them back. This feature brings a new level
of flexibility to the immediate callback feature, allowing customers to determine interactions with the
contact center based on their timetable.
Internet Initiated Callbacks

This feature allows customers to request a callback from an agent while browsing the organization's
Web site. Requests for callback will be placed into queue as are voice calls. When agents become
available, they will receive a screen pop with collected customer information and URL information.
Estimated Time to Answer (ETA)

QueWorX can provide a caller with information on the wait time expected before an agent answers the
call. An announcement will only be provided if the actual ETA falls within the programmed parameter
thresholds. Announcements can be provided in three variations stating a number of minutes and
seconds (“approximately 2 minutes and 45 seconds”), only a number of minutes (“approximately 2
minutes”) or a maximum number of minutes (“less than 3 minutes”).
Queue Depth

QueWorX can provide an announcement indicating how many callers are ahead of a caller in the
queue. Threshold settings are set within the system to determine when to offer this announcement to
callers. For example the 2nd through 15th caller may hear their position within the queue, such as “The
are5 callers ahead of you”. Any caller falling outside of this range would not hear the announcement.
ANI / Area Code Routing

This feature allows calls to be routed based on one or a combination of the following factors – ANI
(Automatic Number Identification), Account Code, Area Code. This feature allows callers to be routed
based on a match of any of these criteria. For example, callers may be prompted to enter their account
number as they enter the system. Based on a match by the QueWorX system, their call may be
redirected to a group of agents (or their personal agent) at a higher priority.
Account Code Routing

This feature allows the system to route a caller to a specific split or agent based on Account Code.

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Screen Pop

Provides an application that is installed at each agent’s desktop. Agents can log into QueWorX and the
CallCenterWorX system via this application. Customer information provided in the Customer Profile
Records Database can be displayed in the form of a ‘pop-up’ at each agent’s desktop as calls are
received. Additionally, the softphone capabilities of this application can be utilized to allow agents to
enter break mode, ready mode, enter tally codes, etc. right from the computers.
Soft Phone

Using the agent Soft phone application from the agent client PC, the agent has the ability to:
Answer the incoming Call
Release (disconnect) the Call
Set Break Mode
Tally the Call
Transfer the Call
Conference the Call
Place the Call on Hold
Retrieve a Call on Hold
Multilingual Announcements

Up to seven (7) language sets can be configured and recorded in QueWorX. Each set is selected
based on pilot number to accommodate multilingual applications. For example, a pilot may be defined
to send calls to a group of agents who support Spanish-speaking customers.
Global Repeat Verification

This feature allows callers to verify their customer input, such as account number of callback number.
This feature helps ensure the proper handling of customer calls.
Customer Profile Records Database

The profile contains information specific to each customer. This information is used to display pertinent
customer information to agents via the QueWorX Agent screen pop application.
Unlimited Multi-Level Automated Attendant

Up to 10 menu options and unlimited levels are recorded and offered to customers. With this option,
customers can provide more options for callers. Multiple levels allow callers to provide more detailed
routing. For example, when the PBX option is chosen from the menu, callers may receive additional
selection choices, such as “For NEAX 2000 press 1, for NEAX 2400, press 2”, and so on up to 5 levels
deep. Up to 10 distinct automated attendants can be configured to accommodate multiple customer
menus. Calls can be routed via the automated attendant to selected departments or groups of agents.

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Chapter 12 Automatic Call Distribution with MIS

Customized Announcements

QueWorX can be configured with up to 89 custom announcements. These announcements are
recorded in QueWorX and invoke utilizing the Call Control Vectors of the CallCenterWorX application
with the “IVR Announcement #” command.
Customizable Reports

The Reports can be customized using the default reports as templates. The modified reports can be
saved for later use.

Professional Services for QueWorX
While QueWorX provides an advanced set of features, some customers will require
applications specific to their contact center that are outside the scope of QueWorX. With
hundreds of successful solutions deployed Professional Services has developed many custom
solutions that allow us to provide Call Center, CTI and speech applications that meet the
customer’s business needs and demands. Requests for these applications can be forwarded
to your NEC representative who will work with NEC Professional Services to help define the
requirements of unique applications or you may contact Professional Services via
ProServ@necunified.com.

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Chapter 13 Voice over IP (VoIP)
VoIP Solutions
The UNIVERGE NEAX 2000 IPS (Internet Protocol Server) is a communication system that can
convert voice signals into IP packets and transmit them through the same data communication
line as IP network (Intranet). Using this system, both voice and data communication lines are
integrated into one communication line, and communication cost can be reduced.
The following VoIP solutions are available with the UNIVERGE NEAX 2000 IPS system.
VoIP Solutions
Extended Enterprise IP Solution
H.323 Connection
IP Station (Peer-to-Peer)
Fax over IP
CCIS Networking via IP
Modem over IP
Remote PIM over IP

Extended Enterprise IP Solution
Teleworkers, usually working from an off-site location, can place a call through the corporate PBX/KTS
by simply dialing the number as if they were in their office. The Dterm phones can be connected to
either a multiple-user unit (called the Dterm IP Branch), or a single-user device (called the Dterm IP
Adapter). The Dterm IP Gateway, Dterm IP branch and Dterm IP Adapter units are linked through a
wide variety of network devices (TA, FRAD, and CSU/DSC) and network types (ISDN, IP, Frame Relay,
T1, Fractional T1, ATM).
Dterm IP Gateway: is a Multiline module, available in 8 or 12 port units. On one side it has an AMP
connector that is used to interface with Digital station ports. On the other side it connects to an IP
packet network though a 10BaseT Ethernet port or circuit network through a DB25 WAN port. It
communicates over the IP network to the Dterm IP Adapter(s) or over an IP network or circuit network
to Dterm IP Branch unit(s).
Dterm IP Adapter: is a compact plug-and-play device that installs into the base of a Dterm display
terminal. It has an integrated two-port 10BaseT Ethernet pass-though hub that permits using one port
to connect the network interface card (NIC) from the PC to the IP network and the other port is plugged
directly into a LAN or into an IP network device such as a router, DSL modem or cable modem.
Dterm IP Branch: Designed for small branch offices the Dterm IP Branch resides at the distant office.
On one side it has an AMP connector that is used to interface with Dterm terminals. On the other side it
connects to an IP packet network though a 10BaseT Ethernet port or circuit network though a DB25
WAN port. It communicates over the IP network or circuit network to the Dterm IP Gateway units.

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Chapter 13 Voice Over IP (VoIP)

Extended Enterprise IP Solution (Cont)

Typical IP Gateway, Branch and Adapter configurations

Required Equipment for NEC’s Extended Enterprise IP Solution
The table below shows the equipment list for NEC’s Extended Enterprise IP Solution.
Equipment Name

Function

Dterm IP Gateway
12 Port Unit

The Dterm IP Gateway is a rack-mountable (or desktop) digital voice
termination device that connects to the Dterm station ports of the NEAX or
Elite system. The Dterm IP Gateway’s main function is to extend Dterm
station ports off of the corporate NEAX or Elite system to users over the LAN
or WAN, such as telecommuters or small branch office workers.

Dterm IP Gateway
8 Port Unit

Dterm IP Adapter
(IPA-U10 Unit)

The Dterm IP Adapter connects to a Dterm Series E display terminal to
provide voice access to a corporate NEAX or Elite system for a single user or
local user. Voice and signaling traffic generated by the Dterm Series E will
be placed in IP packets and sent out to the LAN via a 10Base T Ethernet
connection to the Dterm IP Gateway.
Used for connecting a single user.

Dterm IP Branch
12 Port Unit

The Dterm IP Branch converts NEAX or Elite systems voice and signaling
into packets, which can be sent and received through a network connection.
The Dterm IP Branch supports up to 8 or 12 users per stackable unit.

Dterm IP Branch
8 Port Unit

Used for connecting multiple users from a single location.

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IP Station
IP station is a DtermIP terminal, Dterm Series E/Series i with an IP Adapter Unit, INASET, and
Dterm SP30/SP30 ACD Softphone. IP stations provide a converged infrastructure at the
desktop, with a 10Base-T/100Base-TX Ethernet connection to corporate Local Area Networks
(LAN). The IP station can communicate with other IP stations or CCIS network (IP based) on
a peer-to-peer connection basis and the IP stations can communicate with legacy stations and
trunks (TDM based) via IPPAD (IP Packet Assembler/Disassembler). The IP stations provide
users with all features currently available in Dterm Series E/Series i terminals.
The following figure shows a typical network configuration using IP stations.

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Peer-to Peer Connections between IP Enabled Dterm
The IP stations can communicate with other IP stations over the LAN, on a peer-to-peer connection
basis. Call control is provided by the Main Processor (MP) with Ethernet adapter card, and voice
packets are transmitted between IP stations over the LAN (not through Time Division Switch). Voice
compression of G.729a (8 kbps) and G.723.1 (5.3 kbps/6.3 kbps) is available for its connections.

Peer-to-Peer Connections over CCIS Networking via IP
The IP stations can communicate with other 2000 IPS/2400 IPX systems over the CCIS networking via
IP, on a peer-to-peer connection basis. MP controls a connection between the IP station and IP trunk
built into the MP. The built-in IP trunk communicates with IP trunks of the distant systems (2000
IPS/2400 IPX) over CCIS networks via IP. Voice packets are transmitted over the LAN and WAN (via
router). Voice compression of G.729a (8 kbps) and G.723.1 (5.3 kbps/6.3 kbps) is available for its
connections.

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Connections to Legacy Stations and Trunks
The IP station can communicate with legacy stations and trunks (including H.323 trunks) via IP-PAD
that converts voice packet data to PCM signals. Call control signals are transmitted to MP over the
LAN, while voice packets are transmitted via IP-PAD. The number of IP-PADs depends on the traffic
volume of connections between the IP station and legacy stations and trunks.

Remote Connections of IP Stations
The IP stations can be located at remote sites over an intranet/internet/PSTN/ISDN via routers. IP
stations can be installed behind routers that are configured to use NAT (Network Address Translation)
at the remote location. This allows IP station users at remote sites to use the UNIVERGE NEAX 2000
IPS features as same as those at local site. The number portability among the sites is available, so that
the same IP station user can login at any site, using the single login code and password. The IP station
has two operating modes: Local Connection Mode and Remote Connection Mode. Each operating
mode can have individual VoIP communication parameters such as CODEC selection list and usable
bandwidth, so that the appropriate communication is available even if the IP station user logs in at
either local site or remote site.

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Automatic Program Download for IP Enabled Dterm
This feature provides the method to download the latest firmware program of IP Enabled Dterm from
the FTP/TFTP server automatically by system programming. The following three patterns of program
download are available.
Program Download at Appointed Time
Program Download at Login Time
Program Download for Designated Terminals

Service Conditions
1. Up to 956 IP terminals can be accommodated per system.
2. Total number of ports for IP terminals, legacy stations/trunks (except for AP cards) and IP PADs is
up to 1020 per system. See chapter 1 for capacities of stand alone and network configurations.
Note: Legacy stations/trunks means the ones connected to the Time Division Switch (TDSW). Examples of
legacy station/trunk cards are LC, DLC, COT, PRT, CCT, DTI, 8RST (DTMF receivers), CFTB (Conference
trunk), etc.

3. IP terminals are accommodated in the Virtual PIM’s (Virtual FPs) so that the hardware-PIM’s are
not required to accommodate IP Enabled Dterm terminals. The Virtual PIM is software-defined PIM
used for data assignment for IP Enabled Dterm terminals and Virtual IP trunks (for CCIS
Networking via IP). Maximum nine (15) Virtual PIM’s are provided per system. Total number of the
Virtual PIM’s (for IP terminals) and hardware-PIM’s is up to 16 per system.
4. Minimum one hardware-PIM is required per system to accommodate the MP card. The number of
hardware- PIM’s depends on the number of legacy stations/trunks and IP-PADs. Up to two
hardware-PIM’s (128 LT ports) can be controlled by the MP without the addition of an FP card. With
three FP cards, up to eight hardware-PIM’s (512 LT ports) can be controlled [One FP card controls
up to two hardware- PIM’s (128 LT ports)].
5. The IP-PAD is required for connections between IP terminals and legacy stations/trunks. The IPPAD converts voice packet data to PCM signals. The number of IP-PADs depends on the traffic
volume of those connections. Up to eight IP-PADs can be accommodated per system, thus
providing 256 channels. Up to two IP-PADs can be controlled by the MP without FP cards, and
additional IP-PADs can be controlled by the FP cards (two IP-PADs per FP card).
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8IPLA/24IPLA
PN-8IPLA card provides 8 channels of IP-PAD with VCT function per card. Adding PZ-24IPLA
daughter-card on the 8IPLA card provides up to 32 channels of IP-PAD with VCT function.
Maximum of 8 IP-PAD cards can be accommodated per system, thus providing 256 channels.
By system data setting, the used ports for IP-PAD can be changed to 8/16/24/32 ports.
Note: Up to 24 channels of IP-PAD are provided when using G.723.1 codec.

6. The following types of connections are available on a peer-to-peer basis:
Connections between IP terminals
Connections for CCIS networking via IP from/to IP terminals (2000 IPS or 2400 IPX)
7. The IP-PADs are required for the following connections/statuses:
Connections between IP terminals and legacy stations/trunks/IP trunks(H.323)
Connections for CCIS networking via IP from/to legacy stations/trunks
While IP terminals are on hold (Consultation Hold, Call Transfer, Music-on-Hold, etc.)
When any override service is activated (Executive Override, etc.) including IP terminals.
Three/four-party conference including IP terminals
8. Voice compression is available for following connections.
Connections between IP terminals
Connections via IP-PAD (for CCIS networking via IP from/to legacy stations/trunks)
Voice compression can be assigned on a call basis or terminal basis, by system programming.
The following type of voice compression is available:
G.711 (64 kbps)
G.729a (8 kbps)
G.723.1 (5.3 kbps/6.3 kbps)
9. The Device Registration Server (DRS) is built into the MP card (System-based DRS). The Systembased DRS can provide log-in/log-out function and registration authorization function of IP
terminals. Up to 956 IP terminals can be managed by the System-based DRS.
10. Payload size of voice packet can be assigned from 10 milliseconds to 40 milliseconds in 10
milliseconds increments by system programming. However, available payload size depends on the
type of voice codec.
Codec Type

G.711
G.729a
G.723.1

Payload Size

Available Range
10 milliseconds to 40 milliseconds
10 milliseconds to 40 milliseconds
30 milliseconds (fixed)

Default Value
40 milliseconds
40 milliseconds
30 milliseconds

11. Maximum number of voice channels per IP-PAD card are as follows:
8IPLA/24IPLA
Payload Size

10 ms
20 ms
30 ms
40 ms

Maximum Voice Channels per IP-PAD Card (8IPLA/24IPLA)
G.711
G.729a
G.723.1

20
32
32
32

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12. Following parameters can be assigned on a Location basis. The Location is a kind of group (similar
to Tenant in a PBX) that is defined for bandwidth management and parameter settings for LAN
traffic based on the location of IP terminals. Maximum 64 Location numbers can be assigned per
system.
Type of Codec selection list
Type of Service (ToS)
PAD control pattern
Echo canceller-ON/OFF

Options Per Location
DiffServ
Setting Area/Country
IP-PAD Group number
Usable Bandwidth

Operation when over usable
bandwidth threshold.
Jitter buffer

13. When the IP adapter unit is attached with a Dterm Series E/Series i terminal, other optional
adapters (except for AD/DC adapter unit) cannot be attached with the Dterm Series E/Series i
terminal.
14. Dynamic Host Configuration Protocol (DHCP) cannot be used to assign an IP address to an MP.
15. Login is restricted in following conditions.
When designating a non-existent station number.
("Unregistered" is displayed on the LCD.)
When designating a station number that is not assigned to Multiline Terminal.
("Illegal LEN" is displayed on the LCD.)
16. Logout should be allowed by a station class of service. If not allowed, reorder tone will be heard and
"Restrict" will be displayed on the LCD.
17. Intranet to support Quality of Service (QoS) is preferable to connect IP Stations, to reduce the delay
of connection and voice, and the inferior grade of voice quality.
18. IP Stations operate on 10 Mbps or 100 Mbps LAN based on IEEE 802. (100Mbps LAN is
recommended.)
19. IP Stations support Dynamic Host Configuration Protocol (DHCP). (External DHCP server is
required.)
20. Network-based DRS (external DRS) 3.0 is not supported.
21. Switching hub(s) are required.
22. Spanning Tree (IEEE 802.1d) function is not available.
23. LACP (Link Aggregation Control Protocol, IEEE 802.3ad) function is not available.
24. The UNIVERGE NEAX 2000 IPS can configure the speed mode of the LAN interface for MP/IP-PAD
card, in addition to Auto Negotiation (default).
The speed mode for the MP card can be set to 100 Mbps full duplex in system data
programming. System initialization is required for the mode to take effect.
The speed mode for the 32IPLA/32IPLA-A card can be set to 100 Mbps full duplex, 100 Mbps
half duplex, or 10 Mbps half duplex, using the rotary switch on the card. The card initialization is
required for the mode to take effect.
The speed mode for the 8IPLA card can be set to 100 Mbps full duplex using the DIP switch on
the card. The card initialization is required for the mode to take effect.
25. When the QoS Display function key is pressed on IP Enabled Dterm, the QoS information (number
of lost packets, codec type and payload size) is displayed.
A function key on IP Enabled Dterm should be assigned as QoS Display key in system data
programming.
No other LCD information from PBX such as Clock/Calendar, calling station number and name
is displayed during QoS display.
To stop QoS display, press the EXIT soft key (not EXIT button). Even if the call ends and the
Dterm user goes on-hook, the QoS display remains unless pressing the EXIT soft key.

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Service Conditions on Legacy Service Features
1. A DSS/BLF Console can be associated with the IP Enabled Dterm terminal, but the console must
be connected to a DLC card.
2. Multiple Line Operation with normal Multiline Terminal is available.
3. Service features requiring continuous voice transmission, such as the Background Music feature,
cannot be used because this traffic may reduce overall performance of the Local Area Network
(LAN).
4. CAT mode is available.
5. Soft Key and Help indication is supported.
6. Set Relocations is not available between IP Station and TDM Multiline terminal.
7. DTMF signal is sent for 112ms ~ 128ms when Key Pad is pressed.
9. IP terminals do not support Add-on Module feature.
10. IP Enabled Dterm uses only one synthesized melody (Minuet), built into an IP adapter unit. Music
on Hold using external source is not available for the IP Enabled Dterm.
11. Internal Zone Page from the IP terminals has the following conditions:
IP Enabled Dterm can page up to 3 Dterm/IP Enabled Dterm terminals.
Legacy Dterm can page up to 16 Dterm/IP Enabled Dterm terminals.
IP stations cannot provide all zone paging (receive reorder tone).

Service Conditions on System Registration
1. The IP Station can be registered in the UNIVERGE NEAX 2000 IPS in Login method (with password
protected) or Automatic Login method (MAC address authentication). The registration method can
be assigned in a station class of service.
Note: the DtermSP20/SP30 (soft phone) is registered in Automatic Login method only.

2. The Login method allows the IP station user to be registered in the system by entering its own login
code (station number) and password. The station user can login to the system from any IP Enabled
Dterm terminal in the system, which is assigned to Login method. The Automatic Login method
(MAC address authentication) allows the IP Enabled Dterm to be registered in the system at the
installation time by entering its own login code (station number) and a special password for
installation engineers.
Once the terminal is registered, the station user does not have to login/logout to use the IP Station.
3. Up to 8-digit password can be assigned by system programming. 0 to 9, A and B can be used as a
password. The password can be masked by “*” on the LCD of IP Enabled Dterm.
4. The registration data by Automatic Login method (MAC address authentication) can be backed up
in the flash ROM of the MP by MAT operation or automatic system data back up at a designated
time. Therefore, the installation engineer does not have to re-register the IP Enabled Dterm when
the system reset should occur. The registration data by Login method (with password protected)
cannot be backed up.
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5. When the IP terminal with a call in progress has the LAN cable extracted or a power-off occurs and
restores, in this situation, “Double Assignment” is displayed on the LCD. When the terminal is
registered by Automatic Login method the terminal can be used automatically after about two
minutes. When the terminal is registered by Login method the station user can login from the same
terminal after about two minutes, or login from another terminal (Override).
6. Logout operation from the IP terminal registered in Automatic Login method should not be done in
the normal operation. If the logout occurs, re-registration is required.
7. The registration data by Automatic Login method can be saved from the MAT by designating the
area number 80 (normal system data saving operation does not save the registration data).

Service Conditions on Encryption in System Registration
1. When the IP Enabled Dterm is registered in the UNIVERGE NEAX 2000 IPS, the login code (station
number) and password entered from the terminal can be encrypted.
Login code: Proprietary algorithm
Password: Proprietary algorithm or MD5 algorithm
MD5 is an algorithm defined in RFC 1321 from the IETF.
Encryption algorithm can be assigned on a system-wide basis (no encryption is also available).
2. Encryption is available for Login method (password protected) and Automatic Login method (MAC
address authentication).
3. Up to 8-digit password can be assigned. 0-9, A and B can be used as a password.
4. When the password is not assigned, the registration is not allowed and “Unauthorized” is
displayed on the terminal.
5. When the encrypted password is manipulated in the network, the registration is not allowed and
“Unregistered” is displayed on the terminal.

Service Conditions on VLAN
1. UNIVERGE NEAX 2000 IPS supports VLAN based on IEEE 802.1Q (Tag VLAN).
2. VLAN can be assigned to IP port(s) of MP, IP-PAD and IP Enabled Dterm, and cannot be assigned
to IP ports of IP trunks and IPELC. The VLAN assignment to the IP Enabled Dterm is provided by
Configuration Mode.
3. One IP port can have one VLAN ID. Multiple VLAN IDs cannot be assigned to the same IP port.
4. VLAN ID can be assigned from 1 to 4094. VLAN ID = 0 is handled as Null VLAN ID and is effective
to assign priority only.
5. Switching hub must support VLAN. If the switching hub does not support VLAN, the VLAN function
is not effective even when the system data of the UNIVERGE NEAX 2000 IPS is assigned.

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Service Conditions on Remote Connections
1. The network between NEAX 2000 IPS and IP Enabled Dterm must meet the following requirement:
Waiting time for ACK signal: maximum 600ms (300ms recommended)
To check the above requirement, send Ping command from a PC to a remote IP Enabled Dterm
and see the result of the command (“Time” shows the time to receive ACK signal.). If the result
does not meet the above requirement, abnormal operation such as connection unavailable,
deterioration of voice quality or abnormal state of IP terminal may occur.
2. The remote IP Enabled Dterm can be connected to the network via either of the following devices:
Router
Dial-up router
Modem
Cable Modem
ADSL Splitter
3. The IP terminal can be assigned to Local Connection Mode or Remote Connection Mode by
entering the Administrator Mode of the IP terminal. After the operation mode of the terminal is
changed (Local to Remote, or vice versa), login operation is required again.
4. The differences between Local Connection Mode and Remote Connection Mode are as follows:
Item

Local Connection Mode

Confirmation to Connect
IP Terminal

1 time per minute

Clock Control

2000 IPS sends clock signal to
IP terminal on a minute basis.

Remote Connection Mode
1 time per day
(10 minutes from 00:00 a.m.)
2000 IPS sends clock signal to IP
Enabled Dterm when the IP
Enabled Dterm is logged in or the
clock data is entered by CM02.

5. Individual Location number can be assigned for Local Connection Mode and Remote Connection
Mode, on a station number basis, by system programming. The IP Enabled Dterm can work based
on the location number in each mode. When the Location number for the station is changed, login
operation is required again.
6. NAT (Network Address Translation) is supported using remote Dterm IP terminals only. For NAT
support a Global IP Address programmed in the NEAX 2000 IPS CPU is required.

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7. Because the IP station can have only two-operation mode (Location number), when the number of
remote sites (Location numbers) is over two, there is a case that the appropriate communications
may not be available. Refer to Example 2 for details.
Location number in each operation mode for each IP station:

Remote Connection Mode: Location #1
Local Connection Mode: Location #0

Remote Connection Mode: Location #1
Local Connection Mode: Location #0 or #2

Remote Connection Mode: Location #1
Local Connection Mode: Location #2

Operation mode for each IP terminal:

Local Connection Mode

Remote Connection Mode

Local Connection Mode

Example 1: When Sta. 200 logs in Location #1
STA. 200 logs out from Location #0, and then logs in from IP terminals in Location #1. STA 200 works
in Remote Connection Mode (because IP terminals in Location # 1 are assigned to Remote Connection
Mode.), under the conditions of Location #1 (because STA. 200 in Remote Connection Mode is
assigned to operate under the condition of Location #1.). In this case, the appropriate communication is
available because the operating condition of STA. 200 matches the actual Location number (Location
#1).

Example 2: When Sta. 200 logs in Location #2
STA. 200 logs out from Location #0, and then logs in from IP terminals in Location #2. STA. 200 works
in Local Connection Mode (because IP terminals in Location # 2 are assigned to Locale Connection
Mode.), under the conditions of Location #0 (because STA. 200 in Local Connection Mode is assigned
to operate under the condition of Location #0.). In this case, the appropriate communication may not be
available because the operating condition of STA. 200 (Location #0) does not match the actual
Location number (Location #2).

8. Following features are not available for IP terminals at remote sites:
Multiple Line Operation
Assignment of feature keys which the lamp indication and ringer is controlled by other IP
terminals
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Service Conditions on Automatic Program Download
1. For program download, the FTP or TFTP server is required.
2. This service is available for the IP Enabled Dterm with Peer-to-Peer connections. Terminals with
IPELC connections are not supported.
3. Maximum of four terminals can be downloaded at the same time regardless of the type of
download.
4. Firmware version filed in FTP/TFTP server is assigned by the system data (CM0C YY=00-07,
1st=02). When the latest firmware version is filed in the server, be sure to update the system data.
5. When the FTP server is used for program download, login from the IP Enabled Dterm with a login
name "anonymous" and password "2000 IPS" (authentication by FTP server is required).
6. Conditions on Program Download at Appointed Time
Program Download at Appointed Time service is available by the system data assignment
(CM15YYY=482) on a station class of service basis.
By the same command mentioned above, program download can be retried just one time.
When the download results in failure two times in succession, the IP Enabled Dterm operates
with previous firmware.
When this service is executed, neither Program Download at Login Time nor Program
Download for Designated Terminal can be executed.
PBX system checks the firmware version of each terminal starting from the terminal with the
smallest LEN number and executes this service if required.
This service is executed only for the idle terminals. This service is not executed for the terminals
in Logout or busy status.
The actual starting time of the Program Download at Appointed Time may be varied with a
range of +/--1 minute. When 448 of IP Enabled Dterm have to be upgraded, for example, it
takes about 450 minutes to be completed, without retry.
CMFA YY=50 1st=01/02 counts the number of terminals that are successfully downloaded and
the number of terminals that result in download failure only when the PBX sends the latest
program download message to the terminals. If the terminals are busy or logged out, CMFA
YY=50 1st=02 does not count the number of terminals in download failure because the PBX
does not send the download message to the terminals.
Program Download at Appointed Time can be suspended by CMFA YY=50 1st =00: CCC (clear)
during executing the download. However, you cannot suspend the download of the terminal
with "Downloading..." displayed.
7. Conditions on Program Download at Login Time
Program Download at Login Time service can be assigned on a system basis by CM0C YY=90
1st =02.
When this service is assigned, retry in download failure status is not available.
8. Conditions on Program Download for Designated Terminals
Program Download for Designated Terminals is assigned and executed on the Primary
Extension number basis.
This service is executed only for idle terminals. This service is not executed for the terminals in
Logout status.
When this service is assigned, retry in download failure status is not available.
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Required Equipment for IP Station
Equipment Name
PZ-M606-A
SPN-32IPLAA IP PAD
SPN-16VCTAA IP PAD
SPN-8IPLA IP PAD
PZ-24IPLA
IPW-2U Unit
IP-R Unit
AC-R Unit
ACA-U Unit
IP Terminals

Power

Function
On board Ethernet Interface card. One per system
32ch IP Interface card for IP PAD
16ch Codec card for IP PAD Compression
Max. 2 per 32IPLA Card (optional)
8ch IP PAD card w/compression
24ch Expansion card for IP PAD. Mounts on 8IPLA.
IP Adapter Unit for Dterm Series E terminals
IP Adapter Unit for Dterm Series i terminals
AC/DC Adapter for Dterm Series i terminals
AC/DC Adapter for Dterm Series E terminals
DtermIP terminals
Dterm Series i/E terminals w/IP adapter
INASET terminals
SP30/SP30 ACD Softphones
For Powering the IP stations use one of the following:
AC power supply (ACA-U)
SN1604 Power Hub
Cisco Catalyst switch power feed

CCIS Networking via IP
This feature provides CCIS networks with Voice over IP (VoIP) capabilities to provide a
converged infrastructure over corporate Wide Area Networks (WAN). The IP Enabled Dterm
can communicate with other IP Enabled Dterm over the CCIS network (IP based) on a peer-topeer connection basis. The legacy terminals (TDM based terminals) can communicate with
other legacy terminals or IP Enabled Dterm terminals over the CCIS network via IP Packet
Assembler/Disassembler (IP-PAD). Voice compression of G.729a (8 kbps) and G.723.1 (5.3
kbps / 6.3 kbps) is available for those connections. The CCIS Networking via IP provides users
with all TDM-based CCIS functionality, such as feature transparency, centralized
management, and centralized facilities.
There are two types of connections available for CCIS Networking via IP:

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Chapter 13 Voice Over IP (VoIP)

CCIS Networking via IP (Peer-to-Peer Connections Basis)
When the distant systems are 2000 IPS, the systems are connected on a peer-to-peer basis. The CCIS
call control signals are transmitted between the built-in IP trunks (CCIS Handler; CCH) on the MP card,
over the Local Area Networks and Wide Area Networks (LAN and WAN). For connections between IP
Enabled Dterm terminals, voice signals are also transmitted over the LAN and WAN. For connections
between legacy terminals, voice signals are transmitted via IP-PADs. This connection is also available
when the distant systems are 2400 IPX that supports peer-to-peer connections.

Example IP Enabled Terminals Peer-to-Peer Connection

Example IP between Legacy Terminals via Peer-to-Peer Connection

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Chapter 13 Voice Over IP (VoIP)

CCIS Networking via IP (Non Peer-to-Peer Connections Basis)
IP trunk connections over CCIS (No.7 CCIS Common Channel Inter-Office Signaling) Networking via IP
is used to provide telephony services between the 2000 IPS, IVS2, IPS Retro or 2400 IMX/IPX and
Electra Elite IPK/IPKII on opposite side which do not support peer-to-peer connections. The IP-PAD is
required for connections between IP terminals and IP trunks. A maximum of one IP trunk card can be
accommodated per PIM with a maximum of eight IP trunk cards per system. Voice compression of
G.729a (8 kbps) and G.723.1 (5.3 kbps / 6.3 kbps) can be provided by the IP trunks with VCT cards.
Communications cost can be reduced, by utilizing IP Trunks in the system to convert voice signals into
IP packets and transmit them over the data network. All CCIS centralized and transparent features can
be used connecting up to 255 NEAX systems in a network for a total of 130,560 station ports.

Point-to-Multipoint Connection
Point-to-Multipoint Connection is a connection type, which designates multiple destination IP
addresses, per IP trunk, as shown below. One IP trunk can be connected to multiple opposite offices in
Point-to-Multipoint connection.

IP Trunk: Point-to-Multipoint Connection
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Chapter 13 Voice Over IP (VoIP)

Service Conditions (IP CCIS via IP Trunk cards)
1. Maximum voice channels per IP trunk card depends on the payload period as follows (payload
period can be assigned in system programming):
Payload Period
10 ms
20 ms
30 ms
40 ms

Maximum Voice Channels per IP Trunk
4
8
16
16

2. Maximum 127 voice channels can be provided per system (point-to-multipoint connection).
3. IP trunk cards must be connected with the intranet via router.

IP Trunk Card Specifications
Description
QoS signaling
Speech encoding/decoding
Payload period
FAX communications protocol
FAX mode detection timer
PAD control
Tandem connection
IPT connection mode
Dynamic jitter buffer

Specifications
IP Precedence/DiffServ
G.711/G.729a/G.723.1
G.711/G.729a : 20 ms, 30 ms, 40 ms
G.723.1 : 30 ms (fixed)
T.30
0 to 5 min. (1 min. increment)
Controlled in a node basis
IPT to IPT/IPT to ACIS: available (but voice quality may deteriorate)
IPT to ISDN : available
Fixed Connection Mode or Automatic Disconnect Mode can be
assigned per IPT trunk or destination node in system programming.
10 to 600 ms (10 ms increment)

Required Equipment for IP CCIS via IP Trunk cards
Equipment Name
SPN-IPTB-A
PN-IPT
SPN-4VCTI-A
W/Cable

Functional Name
IPT

VCT

Function
IP Trunk Card Accommodates the IP network and
transmits/receives compressed voice or signals over IP
network. 10 BASE-T/100 BASE-TX twisted pair cable is
connected directly to this card.
4-channel CODEC Card for IP Trunk (max of four per IPTB)
Voice compression protocols:
G723.1, G729A, G711, FAX (14.4 kbps),DTMF signals
Used together with PN-IPTB card.

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Chapter 13 Voice Over IP (VoIP)

H.323 Connection
VoIP (Voice over IP) allows the system to transmit voice conversations over a corporate
Intranet using ITU-T H.323 protocol. For DtermIP-to-DtermIP connection via the IP network
with H.323 protocol, the IPT card and IP-PAD card are required to transmit and receive the
control signal and voice data. For voice compression, the 16VCT card and 4VCT card are
required. For Legacy terminal connection via the IP network with H.323 protocol, the IPT card
is required.

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Chapter 13 Voice Over IP (VoIP)

H.323 Features
The following service features are available when connecting with H.323 terminal (NEC product). For
other vender products, confirmation test is required.

H.323 Features
Call Forwarding-All Calls

Speed Calling-Station

Call Forwarding-Don’t Answer

Bearer Service

Call Forwarding-Busy Line

Tandem Switching of TIE Trunks-2/4-Wire

Split Call Forwarding

The Line Connection with PAD Control

Call Transfer-All Calls

Outgoing Trunk Busy Announcement

Hotline

Brokerage Hot Line-Outside

Do not Disturb-Dterm

Least Cost Routing-3/6-Digit

Calling/Called Number Display

LCR-Time of Day Routing

Digital Display-Trunk

Attendant Manual Override

Speed Calling-System

Echo Canceller Control

Service Conditions for H.323
1.
2.
3.
4.
5.
6.

The data network used for H.323 must be a corporate Intranet.
Gatekeeper is required for H.323 network.
IP trunk cards must be connected with a corporate Intranet via a router.
CCIS feature key FD is required.
Maximum number of nodes in the network depends on the capability of the Gatekeeper.
Maximum one IP trunk card can be accommodated per PIM and maximum eight IP trunk cards per
system.
7. Maximum voice channels per IP trunk card depends on the payload period as follows (payload
period can be assigned in system programming):
Maximum Voice Channels per IP Trunk
G.729a
G.711
G.723.1
20 ms
6
5
–
30 ms
8
7
8
40 ms
12
10
–
Note: Maximum 96 channels can be provided per system
Payload Period

8. The same IP trunk card cannot be shared with CCIS network via IP and H.323.
9. Tone signals to an extension are provided by the system.
10. When a LAN cable is unplugged from the IP trunk card, the associated calls are disconnected.
11. When connected with the system by other manufacturer, a pre-connection test is required.
12. DTMF Relay is supported.

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Chapter 13 Voice Over IP (VoIP)

H.323 IP Trunk Specifications
Description
QoS signaling
Speech encoding/decoding
Payload size
Call model
H.323 procedure
Supplementary service (defined in H.450)
Registration on Gatekeeper
DHCP server
FAX communications
PAD control
Echo canceller

Tandem connection

IPT connection mode
Dynamic jitter buffer

Specifications
IP Precedence/DiffServ
(assigned on an IP trunk basis)
G.711/G.729a/G.723.1
G.711/G.729a : 20 ms, 30 ms, 40 ms
G.723.1 : 30 ms (fixed)
Gatekeeper Routed/Direct
Normal Connect/Fast Connect
Not available
Static registration
Not supported
Not available
Controlled in a node basis
Controlled in a node basis
IPT to IPT/IPT to ACIS
: available (but voice quality may deteriorate)
IPT to CCIS : available
IPT to ISDN : available
IPT to analog C.O. : available
IPT to MFC : not available
Point to Multipoint
10 to 600 ms (10 ms increment)

Required Equipment for H.323
Equipment Name
SPN-IPTB-A
PN-IPT
SPN-4VCTI-A
W/Cable

Page 13- 20

Functional Name
IPT

VCT

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Chapter 13 Voice Over IP (VoIP)

FAX and Modem over IP
This feature allows the system to transmit facsimile or modem communications over IP
network, via Local Area Networks (LAN) and corporate Wide Area Network (WAN). Since PBX
regards facsimile and modem equipment as one of ordinary telephones, IP Packet
Assembler/Disassembler (IP-PAD) and Voice Compression Trunk (VCT) are required for
facsimile uses over IP network same as legacy stations. The facsimile transmission procedure
(T.30 or G.711/G.726 pass-through) is supported with IP-PAD/VCT. The following figure shows
a typical configuration of facsimile or modem use on Peer-to-Peer CCIS network.

IPS

MP
LC

IP-PAD

FAX

LAN or
WAN

IP-PAD

FAX
PC built-in Modem

Example of FAX and Modem over IP connection

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Chapter 13 Voice Over IP (VoIP)

Service Conditions
1. IP-PAD and VCT are required for facsimile use on Peer-to-Peer CCIS network or Remote PIM over
IP function.
2. PN-32IPLA (IP-PAD) card and PN-16VCTA (16VCT) card do not support FAX or Modem over IP.
PN-32IPLA-A (IP-PAD) does not support Modem over IP.
3. IP-PAD card and 16VCT card support Fax over IP with the following protocol.
Protocol
Cards

G.711
Pass-Through
X
(Series 3300
software or later)
X
(Series 3300
software or later)
X
(Series 3300
software or later)
X
(Series 3400
software or later)
X
(Series 3300
software or later)

G.726
Pass-through
X
(Series 3300
software or later)
X
(Series 3300
software or later)

SPN-32IPLA IP PAD-E +
SPN-16VCTAA IP PAD-A

—

SPN-32IPLA IP PAD-C +
SPN-16VCTAA IP PAD-A

—

SPN-8IPLA IP PAD-C

SPN-8IPLA IP PAD-B

SPN-8IPLA IP PAD-A

SPN-32IPLA IP PAD-E
SPN-32IPLA IP PAD-E +
SPN-16VCTAA IP PAD-B

T.30
—

—

X
(Series 3400
software or later)

X
(Series 3200
software or later)
X
(Series 3200
software or later)
X
(Series 3200
software or later)

X: Available —: Not available

4. The analog Media Converter used by 2400 IPX cannot be accommodated in 2000 IPS. It is possible
to connect facsimiles between 2400 IPX and 2000 IPS.
5. A problem may occur such as “expected transmission speed is not obtained” or “not connectable”,
depending on a facsimile model even if it supports T.30.
6. If a Super G3 facsimile is used, the transmission speed will be equivalent to G3.
7. If a facsimile with Error Correction Mode (ECM) function is used, ECM does not work.
8. Connection speed for Modem over IP are:
a. G.711 Pass-through 24kbps – 14kbps
b. G726 Pass-though 14.4kbps – 9.6kbps
Note: Due to many factors these speed are not guaranteed and an evaluation is suggested before actual use.

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Chapter 13 Voice Over IP (VoIP)

Required Equipment for FAX or Modem over IP
Item

Description

Stock #

Qty

PZ-M606-A

151492

1/Sys.

SPN-32IPLAA IP PAD-E

153158

2/FP

SPN-16VCTAA IP PAD-B

153136

SPN-8IPLA IP PAD-C

153154

PZ-24IPLA

151254

Remarks
On board Ethernet Interface Card
(ETHERNET TCP/IP)
IP Interface Card for IP PAD w/o FP: Max 2
w/ FP: Max 2/FP, Max 8/Sys.

2/32IPLA 16ch Codec Card for IP PAD (Option)
Max. 2/32IPLA Card
2/FP
IP Interface Card for IP PAD w/VCT
1/8IPLA Expansion Internet Protocol Line
Circuit for IP PAD w/VCT

Connectable combinations between facsimile stations are shown below.
Connectable Combinations
Source

Via

Remarks
Destination

IPS FAX station

Local

IPS FAX station

Peer-to-Peer CCIS

IPS FAX station

Peer-to-Peer CCIS

IPS DM FAX station

IPS FAX station

Peer-to-Peer Local

IPS DMR FAX
station

IPS FAX station

Peer-to-Peer CCIS

IPX FAX station

IPS FAX station

Peer-to-Peer CCIS

IPX MC FAX station

Station-to-Station connection

Station-to-Station connection
(Remote PIM over IP)

Opposite Office (2000 IPS)

Own Office (2000 IPS)

8IPLA IPPAD-B
8IPLA
IP-PAD-B
8IPLA
IP-PAD-A
32IPLA
IP-PAD-E
32IPLA
IP-PAD-E +
16VCTAA
IP-PAD-B
32IPLA IPPAD-E +
16VCTAA
IP-PAD-A
32IPLA IPPAD-C +
16VCTAA
IP-PAD-A

G.711
G.726

32IPLA IP- 32IPLA IP- 32IPLA IP8IPLA IP- 32IPLA IPPAD-E +
PAD-E +
PAD-C +
PAD-A
PAD-E
16VCTAA IP- 16VCTAA IP- 16VCTAA IPPAD-B
PAD-A
PAD-A
G.711
G.711
G.711
N/A
N/A
G.726

Opposite Office
(2400/MC)
G.711

G.711

N/A

G.711

N/A

G.711

G.711
G.726

G.711

G.711
G.726
T.30

T.30

G.711
T.30

N/A

T.30

N/A

T.30

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Chapter 13 Voice Over IP (VoIP)

Required bandwidth for FAX Connection
• 32IPLA-A/16VCTA-A
Connection Conditions

Required Bandwidth (One-way)

T.30, G711, Payload=40ms

150 kbps (estimated)

T.30, G729a, Payload=40ms, Communication
speed=14.4Kbps
(No IP Header compression in Router)

23.6 Kbps (FAX Payload=78byte)

T.30, G729a, Payload=40ms, Communication
speed=14.4Kbps
(with IP Header compression in Router)

16.6 Kbps (FAX Payload=78byte)

• 8IPLA/24IPLA
Connection Conditions

Required Bandwidth (One-way)

G.711 pass-through, Payload=40 ms
(No IP Header compression in Router)

72 kbps (FAX Payload=320 byte)

G.711 pass-through, Payload=40 ms
(with IP Header compression in Router)

65 kbps (FAX Payload=320 byte)

G.726 pass-through, Payload=40 ms
(No IP Header compression in Router)

40 kbps (FAX Payload=160 byte)

G.726 pass-through, Payload=40 ms
(with IP Header compression in Router)

33 kbps (FAX Payload=160 byte)

Note: The data in the above tables does not include MAC Header.

Required bandwidth for Modem Connection

Connection Conditions

G.711 pass-through

G.726 pass-through

G.711 pass-through

G.726 pass-through

Page 13- 24

Payload
length
40 ms
30 ms
20 ms
10 ms
40 ms
30 ms
20 ms
10 ms
40 ms
30 ms
20 ms
10 ms
40 ms
30 ms
20 ms
10 ms

IP Header
compression in
Router

Yes

Required
Bandwidth (Oneway)
72 kbps
74.67 kbps
80 kbps
96 kbps
40 kbps
42.67 kbps
48 kbps
64 kbps
65 kbps
65.33 kbps
66 kbps
68 kbps
33 kbps
33.33 kbps
34 kbps
36 kbps

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Chapter 13 Voice Over IP (VoIP)

Remote PIM over IP
Remote PIM over IP targets users who have 1-30 relatively small offices that accommodate
10-30 extensions at the Remote Site. When UNIVERGE NEAX IPS DMR or UNIVERGE NEAX
2000 IPS PIM are installed at remote site, and connected to an IPS or IPS DM at main site
over IP network, the Main Site system controls and maintains the Remote PIM operation as
one single system. If a communication failure occurs between the Main Site and Remote Site,
the Remote Site automatically changes over to a survival mode and operates as a stand-alone
system.

UNIVERGE NEAX IPS DMR: IPS Distributed Model Remote (with CP31-A/C)
UNIVERGE NEAX 2000 IPS: IPS PIMMK (with CP24-A/B/C)
The UNIVERGE NEAX IPS DMR is designed primarily for distributed IP networking but also
supports traditional analog and digital trunks for connection to the Public Switched Telephone
Network (PSTN). The UNIVERGE NEAX IPS DMR supports up to 128 peer-to-peer IP stations
and 56 TDM ports in a single modular chassis. Up to two chassis can be stacked providing
maximum capacity of 112 TDM ports while still supporting as many as 128 peer-to-peer IP
stations.
Note: The MP card at Remote Site has the same system data as the CPU at the Host Site; the Host Site
automatically downloads system data to the Remote Site at the time of setup. In normal operation, Main Site
automatically downloads a copy the system data to Remote Site through the network once a day.

Because the CP31 as a DMR is designed as a Remote PIM CPU, the following options that
are built-in on the CP24 are not available with the CP31:
No built-in Modem
No built-in DAT.
Only one RS Port.
No built-in DK (external/relay key).
No MN Alarm Indication

Network Conditions and Payload
Item

Requirement

Protocol

TCP/IP transparent

Maximum
Delay Time

120ms(one way)/240ms(return)Recommended
150ms(one way)/300ms(return) Maximum

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
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Remarks
Support the quality class A,
B of IP Telephone

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Chapter 13 Voice Over IP (VoIP)

Bandwidth Requirement
With G7.23.1
(5.3k/6.3k)
Compression

With G729a ( 8k)
Compression

Without Compression
(G.711)

Control

4.1 Kbps

Voice

31.8/37.8 Kbps

48 Kbps

432 Kbps

Control

4.3 Kbps

Voice

42.4/50.4 Kbps

64 Kbps

576 Kbps

Control

4.3 Kbps

Voice

63.6/75.6 Kbps

96 Kbps

864 Kbps

Control

4.5 Kbps

Voice

84.8/100.8 Kbps

128 Kbps

1152 Kbps

Control

4.5 Kbps

Voice

127.2/151.2 Kbps

192 Kbps

1728 Kbps

Control

4.9 Kbps

Voice

169.6/201.6 Kbps

256 Kbps

2304 Kbps

Control

4.9 Kbps

Voice

254.4/302.4 Kbps

384 Kbps

3456 Kbps

Control

5.8 Kbps

Voice

339.2/403.2 Kbps

512 Kbps

4608 Kbps

Control

5.8 Kbps

5.8Kbps

Voice

381.6/453.6 Kbps

576 Kbps

5184 Kbps

Control

6.7 Kbps

Voice

508.8/604.8 Kbps

768 Kbps

6912 Kbps

Established
Voice Calls
6
8
12
16
24
32
48
64
72
96

Note: This information is an estimation based on an established call. Slightly Higher Control values will occur
at time of call origination and termination.
Base values
Originating from a station: 9.6 Kbps/Call (estimated)
Terminating to a station: 5.76 Kbps /Call (estimated)
Originating to C.O: 11.5 Kbps/Call (estimated)
Terminating from C.O: 5.76 Kbps/Call (estimated)
Keep Alive to Remote Site: 0.032Kbps (estimated)
Other control packets for Remote Site: 4Kbps (estimated)
G.723.1 voice: 5.3Kbps (one-way)
G.729a voice: 8Kbps (one-way)
G.711 voice: 64Kbps (one-way)
The above base values are primarily used for call setup with the exception of keep alive; 0.032Kbps with no voice
traffic. Connections between IP PAD are half duplex, established call utilization is G.711 voice: 64Kbps, G.723.1
voice: 5.3/6.3Kbps, or G729a voice: 8Kbps. Peer-to-Peer IP station calls are full duplex, compression can be
specified by location numbers in system data. Peer-to Peer IP station calls even though full duplex will utilize
one-way for Bi-directional networks such as T1. Peer-to Peer IP station calls over Asymmetrical networks such as
ADSL may realize higher bandwidth utilization, compression can be specified by location numbers in system data.
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Chapter 13 Voice Over IP (VoIP)

Advantages
The system regards the terminals accommodated in both Host Site and Remote Site as the extensions
in the same office. Feature transparency is superior to CCIS.
The Digital Remote PIM cannot accommodate some AP cards; Remote PIM over IP can accommodate
AP cards such as ISDN PRI and T1.
This feature can reduce the bandwidth used on the WAN that is connected to CO lines at Remote Site,
rather than DtermIP at remote locations.
Since all Remote PIM over IP sites are treated as extensions in the same office, software and
applications only have to be implemented in the host site. This provides centralized use of application
for example distributing ACD agents in the DMR locations. CCIS requires each location to have
separate software and applications.
CCIS over IP can be combined with Remote PIM over IP to accommodate larger network
configurations. Up to 255 host sites can be connected via CCIS, each host site can have up to 15
Remote PIM over IP locations.
User Mobility/User Continuance allows a user to relocate from one site to another then login using the
same station number and password. User Mobility/User Continuance is only available to the 2000 IPS
in a Remote PIM network using a DtermIP terminal.

Service Conditions
1. It is a requirement that the same software version be loaded in the Main Site and All Remote
Sites.
2. Host site can be an IPS, IPS DM or IPS Retro system. Main plus Remote PIM over IP is
available in any combination of the following CPUs.
Main Site: CP24-A/B/C, CP27-A, CP26-A, CP28-A
Remote Site: CP31-A/B/C, CP24-A/B/C, CP27-A, CP26-A, CP28-A
Note: The CP27/CP28 can not be used as Backup CPU system in a Remote PIM.

3. Software and Key FD for the whole system must be loaded at the Main Site. No software or
key’s can be loaded into the Remote Site.
4. All system data changes for the whole system must be performed in the Main Site. No system
data changes can be done in the Remote Site.
5. A TCP/IP network is required between the Main Site and Remote Site. A closed and bandwidth
guaranteed network is preferable, such as IP-VPN (layer 3 VPN) or wide area Ethernet service
(layer 2 VPN). Permissible Delay with in these types of networks is 100ms one way (120ms
Max) or 200ms both way (240ms Max).
6. The CPU card at Remote Site has the same system data as the CPU at Main Site; the Host Site
automatically downloads its system data to the Remote Site at the time of setup. In normal
operation, Host Site automatically copies the system data to Remote Site through the network
once a day.

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Chapter 13 Voice Over IP (VoIP)

7. Remote Site automatically operates by itself (survival mode) when Keep Alive signal (sent every
30 sec) between the Host Site and Remote Sits is interrupted. When Keep Alive is interrupted
the Remote Site is reset to change the operation from normal mode to survival mode.
8. Remote Site in survival mode checks at 30 seconds intervals if the communications to Main Site
are possible. When Keep Alive is detected, the Remote Site automatically is reset to change the
operation from survival mode to normal mode.
9. When unstable conditions occur in the network, the Remote Site can be manually set to
survivable mode (override automatic) until stability in the network is established. This prevents
the Remote Site from resetting normal mode to survivable mode etc.
10. User Mobility/User Continuance is only available to IP enabled Dterm, DtermIP and Soft Phone
Dterm SP30 terminals using Protected Login Mode. The SP30 is not supported when used in
collaboration mode with a Dterm or wireless PS.
11. For User Mobility/User Continuance to operate at the visitor site (Main or Remote site), a virtual
station port for each visitor station is required. Each visitor port assigned reduces the total
number of virtual FP ports available to that site (Main or Remote Site), which reduces the total
number DtermIP terminals.
12. While logged in as a visitor the location number, tenant number and service features depend on
the setting of the site that the terminal is logged into. At the time if login at the visitor site stack
dial and last number redial are erased.

Required Hardware and Software
Host Site
Equipment Name
PZ-M606-A
SPN-8IPLA IP PAD
PZ-24IPLA
SPN-32IPLAA IP PAD

Remarks
On board Ethernet Interface card
8 Port PAD with built-in compression
24 Port PAD Expansion, mounts on SPN-8IPLA
Provides Packet assembly/disassembly to accommodate
Legacy Line/Trunk interface.
SPN-16VCTAA IP PAD-B
for compression or T.30 FAX (Optional)
R-PIM 1 Site License
1 required for each Remote site
Note: Registration of Host CPU and software required
Remote Site
Equipment Name
PZ-M606-A
SPN-8IPLA IP PAD
PZ-24IPLA
SPN-32IPLAA IP PAD

Remarks
On board Ethernet Interface card
8 Port PAD with built-in compression
24 Port PAD Expansion, mounts on SPN-8IPLA
Provides Packet assembly/disassembly to accommodate
Legacy Line/Trunk interface.
SPN-16VCTAA IP PAD-B
for compression or T.30 FAX (Optional)
Note: Registration of software is “not” required. Core software of remote site should be the same
version as main site

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Chapter 13 Voice Over IP (VoIP)

List of Service Features Operation
X: Available ─: Not Available
Remote site
Service Features
Main Site
Normal Mode
Survival Mode
Multiline Terminal, SLT operation
X
X
X
IP Enabled Dterm operation
X
X
X
COT, LDT, ODT
X
X
X
Attendant Console
X
─
─
Add-On Module
X
─
─
DSS Console
X
X
X
ISDN Terminal
X
─
─
IPT (CCIS, H.323)
X
─
─
Built-in IPT
X
─
─
OAI client
X
X
─
PS, ZT
X
─
─
Trunk IC/OG call (CO, Tie line)
X
X
X
Caller ID Class
X
─
─
Caller ID Display (MFC, T1-ANI)
X
─
─
Caller ID Display (ISDN)
X
X
X
Caller ID-Station
X
─
─
Enhanced 911
X
─
─
CCIS (Digital, Analog)
X
X
X
SMDR
X
─
─
MAT (RS232C)
X
X
X
MAT (LAN)
X
X
X
MAT (Modem)
X
─
─
System Data change by MAT
X
─
─
SNMP
X
─
─
VLAN
X
X
X
Fault Message registration
X
─
X
Announcement Service
X
─
─
Automated Attendant
X
─
─
Background Music
X
─
─
Conference (Three/Four Party)
X
X
X
Conference (Six/Ten Party)
X
─
─
Conference (32 Party)
X
X(Note 2)
─
Internal Zone Paging
X
─
─
User Mobility/User Continuance Note1
X
X
X
Note1: User will be required to login in again once remote system starts in survival mode.
Note2: Available using CFTC card in the Main Site.

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Chapter 13 Voice Over IP (VoIP)

Planning and Installation
Vendor Support for Open System Standards
The vendor should be committed to supporting open system industry standards, such as
H.323, 802.1p and 802.1q, MGCP, TAPI, JTAPI, etc.
Voice Messaging:
A scalable and cost-effective voice messaging solution that supports industry standards such
as AMIS-A, VPIM, LDAP and IMAP.
Bandwidth requirements
One of the most important factors in the success of your IP network is to determine the
necessary network bandwidth needed for your application. Determining the appropriate
bandwidth is truly a function of two factors:
Number of Users (Phones)
Voice Compression for each user
Number of Users: The number of users is the number of simultaneous users (digital phones)
that will be required.
Voice Compression: NEC’s IP solutions deploy voice compression in order to extend multiple
users across fewer data channels. The following compression algorithms are supported:
G.711, G.723.1 and G.729A. Depending on the voice compression algorithm selected and the
number of voice frames encapsulated in one data packet.
Select the Proper Voice Compression: The best voice quality is achieved by using the noncompression voice algorithm G.711. The maximum quality comes at the expense of the
highest utilized bandwidth. The largest vice compression is achieved by using G.729A. If you
are using this algorithm, you will save on bandwidth and still achieve voice quality that is
regarded as near toll. If absolute conversation quality is your focus and bandwidth is no
object, you probably want to select G.711. If bandwidth is a priority you will need to employ
G.729A.

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Chapter 13 Voice Over IP (VoIP)

Network Consideration
There are several network considerations that must be met before deployment of IP
Telephony. These items are:
The network must support the Internet Protocol and the local interface must meet 802.3
Ethernet standards.
It is recommended that the managed network be based on Switched, Fast Ethernet
environment for IP telephony.
Packet Delay or Latency, one-way should be less than 100 milliseconds optimal, with a
maximum of 250 milliseconds one-way.
Jitter, or out of sequence packets, should be between 10 -150 milliseconds optimal with
a maximum of 300 milliseconds while packet loss of 1% or less is recommended with
the maximum of 3% from end to end.
Network Bandwidth
Network bandwidth requirements are based on the total number of IP trunks or IP stations
installed in the PBX. The multiple algorithms available the bandwidth needed for a voice call
out to an IP network can range from 10.7 Kbps to 96 Kbps.
The most common algorithm used for NEAX IP trunks is G.729A, which allows the voice to be
compressed to 8KBps. Once the layer 3 (IP) overhead is added to the voice payload, the
approximate bandwidth is 16KBps for a single voice stream out of the PBX to an IP network.
This bandwidth calculation does not include layer 2 overhead and will vary depending on the
type of transport (Frame Relay, ATM, Ethernet etc.). Depending on what type of algorithm is in
place and how many devices are deployed dictates the amount of bandwidth needed.
The bandwidth for call setup for the CCIS signaling channel over IP is 7.2 Kbps per call and
the FCCS signaling channel is 83.2Kbps per call. Call teardown requires the same amount of
bandwidth. This number varies and 7.2 Kbps and 83.2 Kbps are averaged values from the
start sequence until setup or teardown is accomplished.
This means to make an FCCS or CCIS call from PBX A to PBX B, PBX A will transmit 83.2
Kbps (FCCS) or 7.2 Kbps (CCIS) as the F/C channel until the call is established. Once the call
is established, PBX A will stop sending the 83.2 Kbps/7.2 Kbps setup information and will
begin sending the appropriate voice payload per the encoding/decoding algorithm selected
(G.711, G.729A, G.723.1).
To make multiple calls, the system will send 7.2 Kbps/83.2 Kbps for each call until all the calls
are established. During tear down 7.2 Kbps/83.2 Kbps will be transmitted for each call again.
If an IP station has been idle for up to 4 seconds, a 7.6Kbps “Keep Alive” packet will be sent to
those respective devices. A “Keep Alive” packet is not generated to an idle station if it is
configured across a WAN. This same process occurs on IP trunk routes. A “Keep Alive”
packet is sent to the trunk route for each system or Point Code in the network. If traffic is
present on either IP stations or trunk routes, no “Keep Alive” packet is generated.
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Chapter 13 Voice Over IP (VoIP)

Bandwidth utilization
The following chart shows the amount of bandwidth for IP overhead per the fill times set in the
PBX system. The higher the fill time used, better performance from the PBX and IP network
can be realized due to the smaller number, yet larger (in size) packets generated.
Codec

10 msec
32 Kbps
32 Kbps
N/A
N/A

G.711 (64 Kbps)
G.729a (8 Kbps)
G.723.1 (6.3 Kbps)
G.723.1 (5.3 Kbps)

Filler Time
20 msec
30 msec
16 Kbps
10.67 Kbps
16 Kbps
10.67 Kbps
N/A
10.67 Kbps
N/A
10.67 Kbps

40 msec
8 Kbps
8 Kbps
N/A
N/A

This chart shows the total amount of bandwidth in layer 3 per voice call during the transmit
stream from an IP trunk or IP station. The value shown is based on the fill time from the table
above and the bandwidth required by the encoding/decoding algorithm that is used (codec).
Total Bandwidth per transmit stream
10 msec
20 msec
30 msec
96 Kbps
80 Kbps
74.67 Kbps
40 Kbps
24 Kbps
18.67 Kbps
N/A
N/A
16.97 Kbps
N/A
N/A
15.97 Kbps

Codec
G.711 (64 Kbps)
G.729a (8 Kbps)
G.723.1 (6.3 Kbps)
G.723.1 (5.3 Kbps)

40 msec
72 Kbps
16 Kbps
N/A
N/A

This chart shows bandwidth usage (G.729A) based on an idle state followed by a keep alive
message with a call setup and voice usage next. Time is not to scale.
Bandwidth per Process

Call Setup

Voice Usage

16
Keep Alive
14
12
10
Bandwidth in
8
Kbps
6
4
2
0

Page 13- 32

Time (not to scale)

Call
Teardown

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Chapter 13 Voice Over IP (VoIP)

Quality of Service (QoS)
In order to obtain quality of service (QoS) for the voice traffic, routers or layer 3 switches in the
network must support a queuing mechanism. The queuing mechanism allows for the
prioritization of the voice packets to pass through the router before the routine data traffic. The
NEAX PBX system allows for customizing the priority of the voice traffic out of the PBX. This
prioritization occurs with the IP precedence bits inside the IP header. With a higher level of
prioritization, the voice traffic will not accumulate as much delay as traffic with lower levels of
priority. With Diff-Serve (Differentiated Services), prioritization can be made if the routing
devices support it. NEAX PBX IP trunks support Diff-Serve.
QoS should be pushed as close to the network edge as possible. At the edge, this is
considered CoS (Class of Service). A layer 2 switched environment is recommended to obtain
full throughput to a routing device for NEAX IP telephony and for CoS management.
Specific ports on the data switch may be prioritized for QoS or CoS. If this is available, the port
that is connected to a NEAX IP telephony device should be prioritized in this manner.
Additionally in the network, the port that has been prioritized, in the above manner, must also
be set as a trusted value in the network. When CoS is used on the edge, map it to a higher
level (layer 3) QoS value. This mapping allows for a consistent prioritization throughout the
entire network. By using the prioritization methods described for the PBX and the network
optimal data transfer in the core and edge devices throughout the network may be
accomplished.
Device Registration Server (DRS) and DHCP
The Device Registration Server (DRS) is built into the MP card (System-based DRS). The
System-based DRS can provide log-in/log-out function and registration authorization function
of IP Enabled Dterm terminals. Up to 956 IP Enabled Dterm terminals can be managed by the
System-based DRS. Once the Device Registration Server (DRS) is located, depending on the
configuration chosen, the IP Enabled Dterm will prompt for a login and password. Networkbased DRS (external DRS) 3.0 is not supported.
DHCP (Dynamic Host Configuration Protocol) can be used for ease of administration for IP
Enabled Dterm terminals in the network. The IP adapter can be set to use a DHCP server to
supply its IP address and the address of the DRS. Network administrators can set up the
option in the DHCP server to give the IP Enabled Dterm terminal the information needed to
find the DRS.

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Chapter 13 Voice Over IP (VoIP)

TCP/UDP/RTP PORT NUMBER CONDITIONS
Tables below show the default TCP/UDP/RTP port number and range of the available TCP/UDP/RTP
port numbers used in 2000 IPS
For Dterm IP
Port Type

Port of Login to
DRS (UDP)

Port for
PROTIMS
(UDP)

Destination

Default Port No.

Remarks

2000 IPS
(MP Built-in DRS of
Main Site)

3456

1024-65535
Assign by CM0B Y=10>60

2000 IPS
(MP Built-in DRS of
Remote Site)

3456

Fixed

Dterm IP

3455

1024-65535
Assign by Dterm IP set up
(User Menu)

2000 IPS

50000 + 64 x Virtual
FP No. + Dterm IP
Port No.

Base Port: 1024-65534
Assign by CM0B Y=10>00-59
(Virtual FP No.)
(Default: 50000 + 64 x Virtual FP No.)

Dterm IP

3458

1024-65534
Assign by Dterm IP set up
(User Menu)

For Voice Control/Voice Packet
Port Type

Destination

Default Port No.

Remarks

Dterm IP

4000

1024-65535
Assign by Dterm IP set up
(User Menu)

IP PAD

50000

1024-65534
Assign by CM0A Y=10-17/
30-37/100-115>90

Dterm IP

1024-65534
Assign by Round Robin method
(Use From the lowest in blank port)

6000-7024

Base Port: 1024-65534
Assign by CM0A Y=10-17/
30-37/100-115>92
(Set every 1025 port)

Port for Voice
Control (UDP)

Port for H.245
Voice Control
(TCP)

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IP PAD

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Chapter 13 Voice Over IP (VoIP)

For Voice Control/Voice Packet
Port Type

Port for Voice
Packet (RTP)

Destination

Default Port No.

Remarks

Dterm IP

3462

1024-65534
Assign by Round Robin method
(Use From the lowest in blank port)

IP PAD

9998-10317

Base Port: 1024-65534
Assign by CM0A Y=10-17/
30-37/100-115>93

IPT (CCIS)

56000-56062

Base Port: Fixed to 56000

IPT (H.323)

56000-56047

Base Port: 20000-65000
Assign by CMBA Y=34
(Default: 56000)

For CCIS Virtual IPT
Port Type

Destination

Default Port No.

Remarks

Port for Server
PBX (TCP)

Server PBX

57000

1024-65534
Assign by CM0B Y=20>00

58000-59023

Base Port: 1024-65534
Assign by CM0B Y=20>01
Select by Round Robin method
(Use from lowest in blank port)

Port for Client
PBX (UDP)

Client PBX

For H.323 IP Trunk
Port Type

Destination

Default Port No.

Remarks

H.255.0 RAS
Port (UDP)

IPT

1719

Fixed to 1719

H.255.0 Control
Port (UDP)

IPT

1720

Fixed to 1720

For Voice Control/Voice Packet
Port Type

Destination
2000 IPS
(Main Site)

Port for
Remote PIM
Control (UDP)

2000 IPS
(Main Site)

Default Port No.

Remarks

3300

1024-65534
Assign by CM0B Y=10>70
DHM Self Port (Communication port
between the Main Site and Remote
Site), Common Port

3300 + Remote Site
No.

Port per Remote Site (MP-MP)
Port per Remote Site:
Setting data of CM0B Y=10>70 + Site
No.

50000 + FP/AP No.
of Remote Site x 64

1024-65534
Assign by CM0B Y=10>00-59
(Virtual FP No. of Remote Site)
Port per Remote Site (MP-FP)

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Chapter 13 Voice Over IP (VoIP)

System Conditions/Limitations (Peer-to-Peer IP)
Limitations
Add-on-Module:
Dterm IP does not support Add-on Module feature.
Analog Port Adapters:
When the IPW-2U unit is attached with a Dterm Series E terminal, other optional adapters (except for
ADA-U Unit) cannot be attached with the Dterm Series E terminal.

Back Ground Music (BGM):
Service features requiring continuous voice transmission such as Background Music feature cannot be
used because the traffic may reduce overall performance of the Local Area Network (LAN).

Internal Zone Paging:
There is a maximum of three DtermIPs per Internal Zone Page Group. Internal All Zone Paging will not
access IP Dterm.

Set Relocation:
Set Relocation is not available between Dterm IP terminals. This feature is for legacy terminals only.
Note: Legacy stations/trunks means the ones connected to the Time Division Switch (TDSW). Examples of
legacy station/trunk cards are LC, DLC, COT, PRT, CCT, DTI, 8RST (DTMF receivers), CFTB (conference trunk),
etc.

Conditions
DSS/BLF:
A DSS/BLF Console can be associated with the Dterm IP and can be used in a Remote PIM, but
the DSS/BLF console must be connected to DLC card.

IP-PAD:
The IP-PADs are required for the following connections/statuses:
Connections between Dterm IP terminals and legacy stations/trunks
Connections between Dterm IP terminals and IP trunks (H.323)
Connections between legacy stations/trunks and IP trunks (H.323)
Connections for CCIS networking via IP from/to legacy stations/trunks
While Dterm IP terminals are on hold (Consultation Hold, Call Transfer, Music-on-Hold, etc.)
When any override service is activated (Executive Override, etc.)
Three/four-party conference including Dterm IP terminals

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Chapter 14 Common Channel Inter-Office
Signaling (CCIS)
The UNIVERGE NEAX 2000 IPS offers a very powerful proprietary networking feature called
Common Channel Inter-Office Signaling (CCIS). This functional and extremely flexible
“intelligent network” capability allows two or more UNIVERGE telecommunications systems to
be networked together to provide feature transparency and to centralize many important
telecommunications functions such as Attendant Consoles, Call Accounting / Billing Systems,
Voice Processing Systems, Management systems, and Trunk Facilities, to name a few.
A variety of NEC’s UNIVERGE family of telecommunications systems can be networked
together using CCIS, creating a proprietary corporate network. This network can serve multiple
company facilities in the same building or town, in different towns, in different states, and even
in different countries. Centralized services provide a great cost reduction in the initial system
hardware costs, ongoing maintenance costs, trunk and tie line costs and overall operations
costs. Feature transparency increases user productivity by providing a common numbering
plan for simplified on-net calling and a common feature package. Feature transparency allows
most features to be accessed using similar procedures throughout the local, national, or global
corporate network.
Alternate Routing capabilities provide increased network reliability, additional cost savings, and
increased user productivity. Based on the industry standard CCITT Signaling System #7,
which is the frame work for ISDN, CCIS offers 64k “Clear” channel voice/data transmissions.
The CCIS “Intelligent Network” can be configured to provide redundant signaling channels and
multiple alternate routing schemes for maximum network reliability. CCIS is so flexible it can be
used with standard digital T1 spans, analog tie lines, or satellite transmission systems.
The CCIS architecture is similar to ISDN. Both use 23B + D type trunks. Both networking types
conform to and comply with the Open Systems Interconnection (OSI) 7 Layer Model.
The UNIVERGE NEAX 2000 IPS can be interfaced to another UNIVERGE NEAX PBX by No. 7
CCIS Signaling (Analog/Digital), CCIS Networking via IP (non Peer-to-Peer connection), or via
IP (Peer-to-Peer connection). For adding No. 7 CCIS to the system, it is necessary to install
the 24 channel DTI (Digital Trunk Interface) for a digital network or LDT (Loop Dial Trunk)/ODT
(2 wire E&M or 4 wire E&M Trunk) for an analog network via a MODEM, and a PLO (Phase
Locked Oscillator) for network synchronization. Also, it is necessary to install a CCH (Common
Channel Handler). The CCH receives/transmits common signaling data to/from the distant
office. In each local office, the PBX can provide Centralized Billing function in addition to a
variety of inter-office service features. For addition of the Centralized Billing function, an AP
(Application Processor) is required when the UNIVERGE NEAX 2000 IPS is the center office
and Built-in SMDR when the UNIVERGE NEAX 2000 IPS is used as a tandem or local office.
Note: Centralized Day/Night Mode Change and Centralized Fault Message require CCIS No.7 networking with
the IPX.
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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

For DtermIP to DtermIP connection via CCIS (Peer to Peer connection), the voice data is
transmitted and received directly between DtermIPs via Intranet (CCIS via IP). For Dterm IP to
Legacy terminal connection via CCIS, the IP-PAD card is required to transmit and receive the
voice data. This card is used to control and convert the voice to data. The control signals are
managed by the MP card in either of the connection above.
The illustration below shows a system outline of CCIS Connection.

System Outline of a CCIS Connection

Page 14-2

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Common Channel Interoffice Signaling
(1) 64 Kbps D Channel
Office

Office

B
(23) - 64 Kbps Clear Channel Bearer Channels

Signaling

Addressing
Information

CCIS uses a common channel to provide the signaling information between two
systems (nodes). The information transmitted in the signaling channel between two
systems includes addressing information, supervisory information, and centralized
network information. This signaling information is to be transmitted between all
network nodes for a fully integrated network.
This information includes dialed digits (called number) and calling party
identification. CCIS provides the ability to provide calling number identification or
the name of the calling party to any display station across the network. Users can
screen calls by looking at the display prior to answering the call.

Supervisory
Information

This information includes station status, call set up and termination information.

Network
Information

This information includes centralized billing information, system fault data
messages, message waiting indication control information, voice processing
integration information, and data related to other network feature control.

CCIS and ISDN
Since the first CCIS system was installed in 1985, CCIS has proven to be the most reliable and
versatile proprietary network system available. The delays in achieving nation wide ISDN
compatibility, and the fact that a total global ISDN network is not yet available, exemplify the
true values of an intelligent proprietary CCIS network. Additionally, it is easy to understand that
a global network that only needs to be compatible with one family of telecommunications
products can easily offer much more feature transparency and centralization than a global
network that is intended to be compatible with all telecommunications systems. For this reason
alone, the benefits of a CCIS network are much greater than those of an ISDN network. And
with CCIS, it is possible to create a mixed network of CCIS and ISDN.

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Digital and Analog CCIS
The PBX can provide No. 7 CCIS via either a digital network or an analog network. Regardless
of the relevant network being a digital network or an analog network, CCH (Common Channel
Handler) to control the common signaling channel is required.
Digital CCIS
Digital CCIS uses standard 24 channel digital T1 facilities to provide 23 B (voice/data) channels and 1
D (signaling) channel for the first CCIS dedicated T1 span. Additional T1 spans can be added and all
24 channels of each additional T1 span could be used for voice and/or data transmission. In this case,
the D channel of the first T1 span is using non-facilitated associated signaling for the additional T1
spans. A second D channel of the first T1 span can be configured to provide a redundant signaling link
between nodes to increase network reliability. To provide even more reliability, one channel of a
second T1 span can be programmed to serve as the redundant signaling link. In this configuration,
network reliability is at a maximum because if the first T1 span fails, the 23 B channels and the
redundant D channel of the second T1 span would keep the network operational.
The D channel can be programmed to transmit the signaling information at rates of 48K, 56K, or 64K
bits per second. At these high transmission rates, one D channel can easily handle the signaling
requirements for multiple T1 spans (non-facilitated associated signaling).
When using a Satellite system as the transmission medium, the NEAX Family of systems are usually
configured for digital CCIS, however, it should be noted that many Satellite systems can be configured
to accept multiple analog tie lines which would be compatible with analog CCIS.

Analog CCIS
For applications requiring analog tie lines, CCIS requires the use of modems at each end of the
dedicated signaling tie line. These modems can operate as high as 56K baud rate. The voice and data
lines (B channels) also use standard analog tie lines. If the customer requires 7 voice / data channels
between nodes, 8 tie lines are ordered providing one extra tie line to be dedicated as the signaling
channel. This may appear to require more tie lines then a non-CCIS network but actually less tie lines
are required when CCIS is used thanks to the intelligent Look Ahead Routing feature.

Page 14-4

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

IP CCIS
Peer-to-Peer Connections
Peer-to-peer connections over CCIS Networking via IP are available when the distant systems
2000 IPS or 2400 IPX supporting peer-to-peer connections. The built-in IP trunks (CCH) on the
card are assigned with the Virtual Application Processors (AP), and can support up to 127 trunks
system. One Virtual AP can support up to 64 trunks, and up to two Virtual APs can be assigned
system.

are
MP
per
per

The 8 IPLA IP PAD is required for connections between legacy stations/trunks, IP terminals and legacy
stations/trunks over the CCIS networking via IP. The IP-PAD converts voice packet data to PCM
signals, and one IP-PAD can provide 8 PCM channels or one 8 IPLA plus one 24 IPLA can provide 32
PCM channels. The number of IP-PADs depends on the traffic volume of those connections, and up to
eight 8 IPLA IP PADs with 24 IPLA daughter boards can be accommodated per system, thus providing
256 PCM channels in total. Up to two IP-PADs can be controlled by the MP without FP cards, and
additional IP-PADs can be controlled by the FP cards (two IP-PADs per FP card).
Note: Legacy stations/trunks means the ones connected to the Time Division Switch (TDSW). Examples of
legacy station/trunk cards are LC, DLC, COT, PRT, CCT, DTI, etc

Voice Compression
Voice compression is available for CCIS networking via IP to/from legacy stations/trunks.
The following type of voice compression is available:
G.711 (64Kbps)
G.729a (8Kbps)
G.723.1 (5.3Kbps/6.3Kbps)
Voice compression can be assigned on a call basis or terminal basis, by system programming.
The 8 IPLA IP-PAD and the 24 IPLA daughter board both have built-in VCT capability to provide voice
compression which is required for G.729a and G.723.1.

Service Conditions
Intranet must be used for a CCIS network via IP. (Internet is not supported.)
Maximum number of nodes in the CCIS network via IP is 255.
All CCIS features are available in the CCIS networking via IP.
When T1 timeout of IAI is occurred, an outgoing CCIS call can be routed over alternate trunk
route. This alternate routing can be allowed or denied on a trunk route basis by system
programming.
Note: The alternate routing is not available for tandem calls.

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

CCIS Networking via IP between IP Terminals

CCIS Networking via IP between Legacy Terminals

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

CCIS Networking via IP (Non Peer-to-Peer)
When the distant PBX system does not support peer-to-peer, the systems are connected with IP trunks
(including Voice Compression Trunks (VCT)), via Time Division Switch (TDSW). Voice signals of IP
terminals are transmitted via IP-PADs, while those of legacy terminals are directly connected to the IP
trunks. Call control signals between the systems are also transmitted over the IP trunks. Voice
compression of G.729a (8Kbps) and G.723.1 (5.3Kbps/6.3Kbps) can be provided by the IP trunks with
VCT cards.

Service Conditions on Non Peer-to-Peer Connections
IP trunk connections over CCIS Networking via IP are available when the distant systems are
NEAX 2000 IVS2, NEAX 2400 IMX/IPX or Electra Elite IPK that does not support peer-to-peer
connections.
The IP-PAD is required for connections between IP Enabled Dterm terminals and IP trunks.
Maximum one IP trunk card can be accommodated per PIM. Maximum eight IP trunk cards can
be accommodated in the system
Maximum voice channels per IP trunk card depends on the payload period as follows (payload
period can be assigned in system programming):
Payload Period
10 ms
20 ms
30 ms
40 ms

Maximum Voice Channels
per IP Trunk
4
8
16
16

Maximum 127 voice channels can be provided per system (point-to-multipoint connection).
IP trunk cards must be connected with the intranet via router.

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Centralized Billing
This feature is used to collect billing information from each office within the network and direct
it to the associated center office. Billing information is then forwarded to the central billing
centers via RS232C interfaces.
Service Conditions
1. The Centralized Billing system is composed of local office, tandem office, and center office.

2. The UNIVERGE NEAX 2000 IPS is generally used as a local office; however, it can be used as a
center office for centralized billing if all the PBXs within the network are IPS.
3. When the UNIVERGE NEAX 2000 IPS is used as a center office, the billing information is sent to a
SMDR system using the NEAX 2400 format and an AP00.
4. When a UNIVERGE NEAX 2400 IPX is used as a center office, either the Sub Line or the My Line
number can be selected by system programming to be sent to the SMDR terminal. When the
UNIVERGE NEAX 2000 IPS is used as a center office, only the My Line number can be sent to
the SMDR terminal. When a call is originated from a station (Station A) of a local office (without
AP00), via a COT of a center office (with AP00), a message is sent to the center office from the
local office via CCIS. According to the station number of the calling party, the AP00 of the center
office generates billing information to the SMDR terminal.
5. Maximum digits of station number is as follows:
Local office/Tandem office:
using Internal SMDR: six digits (SMDR over IP is not supported)
using AP00 : four digits (if fifth digit of all stations uses a common leading digit then five
digits)
Center office:
using Internal SMDR: six digits
using AP00:four digits (if fifth digit of all stations uses a common leading digit then five
digits)

Page 14-8

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

6. Center office
Maximum of eight local offices can be accommodated.
Maximum of 3,600 calls per hour can be received. If the center office exceeds 3600, billing
information is stored at local office.
7. Local office
By predetermined office data, local office number of the calling station can be forwarded to
the center office, and output to the SMDR.
In case of using internal SMDR
a. The billing information cannot be output at local office.
b. When a failure of center office occurs or CCIS link is disconnected, the billing
information is stored up to 1,024 calls at local office. System initializing loses
stored information.
c. Maximum of 3,600 calls per hour can be sent to center office/tandem office.
In case of using AP00
a. Internal SMDR cannot be used.
b. Maximum of 3,600 calls per hour can be sent to center office/tandem office.
c. The billing information can be sent to center office/tandem office and at the
same time it can be output at local office.
The billing information can be stored as follows and system initializing does not lose stored
information.
a.
b.
c.
d.

AP00B MRC-C(AP) without expansion memory: up to 800 calls
AP00B MRC-C(AP) with expansion memory: up to 27000 calls
AP00B MRC-F(AP) without expansion memory: up to 1965 calls
AP00B MRC-F(AP) with expansion memory: up to 22925 calls

8. Tandem Office
One center office and maximum of seven local offices can be accommodated.
Maximum of 3,600 calls per hour can be received from local office. If the center office exceeds
3600, billing information is stored at local office.
Maximum of 3,600 calls per hour can be sent to center office.
Tandem office also can function as a local office.

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Page 14-9

Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Centralized E911 – CCIS
This feature allows the system to transmit a calling party number to the 911 Emergency
system over CCIS tandem connection.

Service Conditions
1. The Calling Party Information Transferring Service must be provided between calling office and
tandem office.
2. The calling through CCIS must be Least Cost Routing (LCR) calling.

Page 14-10

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Call Set Up Times
Standard tie line networks typically require from two to eight seconds to complete the set up of
a call and return ring back tone to the caller. With a CCIS network, call set up times are much
lower averaging from 100 msec. to two seconds depending on the number of systems in the
network.

Look Ahead Routing
In a standard non-CCIS network using tie lines to connect two nodes, when a station in node A
calls a station in node B a tie line is seized and the call set up information is transmitted (in
band) to attempt to ring down the called station. If that station is busy, busy tone is heard
through the tie line from the destination system. This acceptable practice has been used for
many years in spite of how inefficient it may be. The above scenario illustrates why many nonCCIS tie line networks require tie line usage for calls that never get completed. This is not very
efficient and often results in customers needing to purchase more voice/data tie lines then
what would really be needed if an intelligent network were employed.
The intelligent CCIS network provides a Look Ahead Routing feature that helps to reduce the
load on network tie lines (voice/data channels) thereby reducing the total number of voice/data
channels (tie lines) required between two nodes. When an on-net call is placed to a remote
node, the system will “Look Ahead” to the destination system to determine if the called station
is available. This is accomplished by communications between the systems over the D
channel. If the called station is not available, busy tone is provided to the calling party locally
and a voice/data channel (tie line) is not used. This provides a decrease in the overall number
of tie lines needed to carry the traffic loads between network nodes, further reducing ongoing
operation costs and system hardware requirements.
The benefits of Look Ahead Routing are also realized in applications where trunk facilities are
strategically distributed throughout the network. CCIS will “Look Ahead” when a user in node A
tries to access a trunk facility from node B. A tie line from node A to node B would only be
used if the trunk facility requested is available.

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Shared Trunk Facilities and Alternate Routing
Many network applications dictate a need to strategically distribute trunk facilities throughout
the network. The practice of sharing trunk facilities across a network is beneficial because it
reduces the overall network trunking requirements and provides additional lower cost route
choices. When combining this practice with the Alternate Routing capabilities of CCIS, a large
increase in network reliability and user productivity can be realized while reducing network
operations costs.
The flexibility surrounding CCIS allows multiple alternate routing schemes, which can be
created to enhance network reliability while they are specifically designed to consider Least
Cost Routing (LCR) benefits. This provides the user with multiple alternate routes that are
automatically selected with call costs as the primary factor in route selection. As shown in the
following diagram, CCIS intelligence will also automatically use the pre-configured alternate
route for the D channel (if configured for link redundancy) in the event that the primary D
channel route is disabled.
Alternate Routing

LINK FAILURE

VOICE/DATA
NETWORK

ALTERNATE ROUTE

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Centralized System Maintenance and Administration
A centralized system maintenance and administration facility can be designated at one node in
the network allowing all nodes in the CCIS network to be maintained from a single location.
This reduces travel costs and maintenance times. All system messages, fault data, and cyclic
diagnostic reports will be transmitted over the D channel to the designated central facility as
shown in the following diagram. The individual system messages and diagnostic data of each
node can be easily identified, since each system node has an assigned name allowing
messages from multiple sites to be separated. Additionally, each node will buffer and output
system messages and diagnostic data to a local printer at the node generating the information.
This feature applies to the 2400s sharing a network with 2000s. The 2000 IPS will pass fault
information originating from the 2400s through to other 2400s. When a UNIVERGE NEAX 2400
IPX is in the network, the 2000 IPS will send its fault data to the 2400 IPX designated as
Central Fault MAT.
System administration also benefits from the ability to perform high speed adds, moves or
changes from the centralized facility for any node in the network without the addition of costly
peripheral equipment. The costs savings in this area are immediately realized.
Centralized Maintenance Facility

RS-232C Interface
NEAX
2400

NEAX2400

FAULT RECORD

CENTRALIZED
MANAGEMENT CENTER
FAULT RECORD

NEAX2000

FAULT RECORD

CCIS LINK

NEAX2400 or
NEAX2000

FAULT RECORD

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Centralized Call Accounting/Billing Systems
Standard tie line networks require independent call accounting / billing equipment for each
node in the network. This results in a costly investment of peripheral equipment, higher
maintenance costs, and the need to service this equipment at each node.
In a CCIS network, it is not necessary to purchase independent call accounting / billing
equipment for each node in the network. One call accounting / billing system can be centrally
located, connected to any one system in the network, and the intelligent CCIS network will
transmit the SMDR data from each system in the network over the D-channel to the
designated centralized collection point, as shown in the following diagram. This increases
reliability by minimizing hardware requirements, simplifies call costing data analysis, and
provides a large reduction in operation costs.
Centralized Call Accounting / Billing

RS-232C Interface
NEAX
2000

CALL RECORD

NEAX2400

CALL RECORD

NEAX2000

CENTRALIZED BILLING
CENTER

CALL RECORD

CCIS LINK

NEAX2400 or
NEAX2000

Page 14-14

CALL RECORD

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Centralized Voice Processing / Messaging
Voice Processing Systems such as voice mail, fax mail, and Integrated Voice Response (IVR)
systems can be centralized in a CCIS network. This eliminates the need for costly independent
systems to be purchased and maintained for each node in the network. Maintenance is
simplified by locating the system at the centralized maintenance facility or any other facility
within the CCIS network. The information required for full integration from any of these types of
systems will be transmitted over the D-channel to the required node as required by the
integration logic of the voice processing system. Increased network reliability and substantial
operating cost savings can be realized.
CCIS Centralized Voice Mail

NEAX2400/NEAX2000
NEAX2400/NEAX2000

NEAX2400

VOICE MAIL
SYSTEM
CCIS LINK

NEAX2400/NEAX2000
NEAX2400/NEAX2000

Centralized Attendant Consoles
CCIS allows the centralization of attendant consoles to service calls from any system in the
network. By centralizing attendant services, a greater number of calls can be handled with
fewer consoles based on economies of scale. System hardware requirements can be reduced
and overall operations costs are lower. CCIS uses the Look Ahead feature discussed earlier to
minimize tie line usage as both on-net and off-net calls are centrally serviced and distributed
throughout the network.

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Uniform Numbering Plans
CCIS offers complete flexibility when designing network numbering plans. Two primary
numbering schemes can be used.
One scheme uses a four- or five-digit station numbering plan with no system node identifier. In
this scheme, station numbers are not repeated throughout the network and usually the first
digit is used as an indicator of which node the station is at. For example; a three-node system
might use station numbers 1000-2999 at node 1; 3000-4999 at node 2; and 5000-6999 at node
3. The intelligence of CCIS and LCR will automatically route the call to the correct node.
The second scheme uses a one, two or three digit node identification code followed by free
station numbering. In this scheme station numbers can be repeated at each node in the
network, if desired. This scheme is especially beneficial to those customers who are upgrading
a network or who are networking multiple facilities for the first time and wish to maintain their
existing numbering plans at each facility. It is also very useful in very large networks where it
may be necessary to repeat station numbers at different facilities due to the total number of
stations in the network.

CCIS Feature Chart
CCIS Feature Name
Attendant Camp-On with Tone Indication-CCIS
Attendant Controlled Conference-CCIS
Automatic Recall-CCIS
Brokerage-Hot Line-CCIS
Busy Lamp Field (BLF)-CCIS
Busy Verification-CCIS
Call Back-CCIS
Call Forwarding-All Calls-CCIS
Call Forwarding-Busy Line-CCIS
Call Forwarding-Don’t Answer-CCIS
Call Forwarding-Intercept-CCIS
Call Forwarding-Override-CCIS
Call Processing Indication-CCIS
Call Transfer-All Calls-CCIS
Call Transfer-Attendant-CCIS
Calling Station Status Display-CCIS
Calling Name Display-CCIS
Calling Number Display-CCIS
Call Transfer-All Calls-CCIS
Centralized Billing-CCIS
Centralized Day/Night Mode Change-CCIS
Centralized E911-CCIS
Consultation Hold-All Calls-CCIS
Deluxe Traveling Class Mark-CCIS
Dial Access to Attendant-CCIS
Direct-in Termination-CCIS

Page 14-16

NEAX 2400 Required
YES

YES

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

CCIS FEATURE CHART (Cont.)
CCIS Feature Name
NEAX 2400 Required
Distinctive Ringing-CCIS
Do Not Disturb-CCIS
Dual Hold-CCIS
Elapsed Time Display-CCIS
Flexible Numbering of Stations-CCIS
Hands-Free-Answer Back-CCIS
House-Phone-CCIS
Hot Line-CCIS
Incoming Call Identification-CCIS
Individual Attendant Access-CCIS
LDN Night Connection-CCIS
Link Alarm Display-CCIS
Link Reconnect-CCIS
Message Waiting Lamp Setting-Attendant-CCIS
Message Waiting Lamp Setting-Station-CCIS
Miscellaneous Trunk Access-CCIS
Miscellaneous Trunk Restriction-CCIS
Multiple Call Forwarding-All Calls-CCIS
Multiple Call Forwarding-Busy Line-CCIS
Multiple Call Forwarding-Don’t Answer-CCIS
Multiple Console Operation-CCIS
Night Connection Fixed-CCIS
Night Connection Flexible-CCIS
Outgoing Trunk Queuing-CCIS
Paging Access-CCIS
Restriction from Outgoing Calls-CCIS
Service Display-CCIS
Single Digit Station Calling-CCIS
Station Controlled Conference-CCIS
YES
Station to Station Calling-CCIS
Station to Station Calling-Operator Assistance-CCIS
Toll Restriction-3/6 Digit-CCIS
Trunk Answer from Any Station-CCIS
Trunk to Trunk Restriction-CCIS
Uniform Number Plan-CCIS
Voice Call-CCIS
Voice Mail Integration-CCIS
Voice Mail Live Record-CCIS Note
Voice Mail Private Password-CCIS
Asynchronous Data Switching – CCIS
Data Interface - Automatic Answer – CCIS
Data Line Security – CCIS
Data Transparency – CCIS
Simultaneous Voice and Data Transmission – CCIS
Synchronous Data Switching – CCIS
Note: Live Record Over CCIS is not supported to a NEAX 2400.

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

System Capacity
System Capacity for CCIS with Digital Interface
Description

DTI Card
CCH Card
AP00 Card (for Centralized Billing)
Trunks for DTI
CCIS Routes
Trunks per DTI Card
CCT Card
Trunks for CCT
Trunks per CCT Card
M10 Card

24DTI

30DTI

8
8
1
192
8
24
—
—
—
4

8
8
1
248
8
31
—
—
—
4

Capacity
24CCT

—
—
1
—
8
—
8
200
25
4

30CCT

—
—
1
—
8
—
8
248
32
4

System Capacity for CCIS with Analog Interface
Description

Capacity

CCH Card
AP00 Card (for Centralized Billing)
ODT Card
ODT Trunks

8
1
128
254

System Capacity for CCIS Networking via IP
Description
CCIS Trunks (IP Trunk/ Peer to Peer)
CCIS via IP (non Peer-to-Peer)
SPN-IPTB
SPN-4VCTI
CCIS via IP (Peer-to-Peer)
PZ-M606-A
SPN-8IPLA IP PAD-C
PZ-24IPLA

Page 14-18

Capacity
255
8
32
1
8
8

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

Required Equipment
Required Equipment for Digital CCIS
EQUIPMENT
NAME
PN-24CCTA

FUNCTIONAL
NAME
CCT

FUNCTION
CCIS (1.5 Mbps) Trunk Card
Provides a built-in Common Channel Handler (CCH) of CCIS.

PN-24DTA-C

DTI

T1 Digital Trunk Interface (1.5 Mbps) Card
Accommodates 24-channel PCM digital lines.

PN-30DTC-C

DTI

E1 Digital Trunk Interface (2 Mbps) Card
Accommodates 30-channel PCM digital lines.

PN-SC00

CCH

Common Channel Handler Card
Transmits/receives signals on the common signaling channel of
CCIS.

PN-M10

M10

Optical Fiber Interface Card
Provides optical fiber interface for T1/E1 Digital Trunk Interface
(1.5 M/2 Mbps) Line length: 6.2 miles (10 km) or less
Line coding: CMI

Required Equipment for Analog CCIS
EQUIPMENT
NAME
PN-SC00

FUNCTIONAL
NAME
CCH

PN-2ODTA

ODT

FUNCTION
Common Channel Handler Card
Transmits/receives signals on the common signaling
channel of CCIS.
2-line Out Band Dialing Trunk Card
Used as either a 2-wire E&M trunk or a 4-wire E&M
trunk.
Equipped with –48V DC-DC on-board power supply.
Both Circuits 0 and 1 must be set to the same
purpose
(2-wire or 4-wire) in one card.

Required Equipment for CCIS Networking via IP
Description
CCIS via IP (non Peer-to-Peer)
SPN-IPTB
SPN-4VCTI
CCIS via IP (Peer-to-Peer)
PZ-M606-A
SPN-8IPLA IP PAD-C
PZ-24IPLA

Remarks
Network Interface:
ETHER/FAST ETHER
Codec Card for IP Trunk
On board Ethernet Interface Card
(ETHERNET TCP/IP)
IP Interface card for IP Pad
24-channel daughter board for 8IPLA

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Page 14-19

Chapter 14 Common Channel Inter-Office Signaling (CCIS)

IP Specifications
Item
Voice Encoding
IP-PAD
FAX Communication
Feature

DTMF Signal
Inter-office/Intraoffice
Signaling

Specifications
G.729a
G.723.1
G.711
8/32 channels per card
Automatically seized per call
FAX Relay Method (T.30)

FAX Relay Method
(Pass-through (G.711/G.726))
H.245
H.245
PROTIMS over IP
CCIS over IP
H.323

Jitter Control
Quality of Service
(QoS)
LAN Interface

Dynamic Jitter Buffer
• TOS, IP Precedence
• DiffServ
10BASE-T/100BASE-TX

Echo Canceller (IPPAD)
Payload DtermIP/
Size
CCIS
Virtual
IPT
(32IPLA +
16VCT)
DtermIP/
CCIS
Virtual
IPT
(8IPLA +
24IPLA)
H.323
IPT

G.168 (64 ms.)

PAD Control

Page 14-20

10 ms.-40 ms.
(G.723.1: 30 ms. fixed)

20 ms.-40 ms. (10 ms. increments)
(G.723.1: 30 ms. fixed)

0 dB to +16 dB (+2 dB increments)
0 dB to –16 dB (–2 dB increments)
0 dB to –16 dB (For North America)
0 dB to –12 dB (Outside North
America)

Remarks
8 Kbps CS-ACELP
5.3/6.3 Kbps MP-MLQ/ACELP
64 Kbps PCM
G3 FAX (up to 14.4 Kbps)
Super G3 Reciprocal: Not allowed
FAX communication with H.323:
Not available
PN-32IPLA (IPPAD) card
(PN-8IPLA (IP-PAD) card is not available.)
PN-8IPLA (IP-PAD) card
(PN-32IPLA (IPPAD) card is not available.)
H.323 IPT/IP-PAD/DtermIP
DtermIP-to-DtermIP connection
DtermIP-to-IP-PAD connection
DtermIP-to-2000 IPS connection
Point-to-Multipoint connection
H.323 IPT/4VCT card
and IP PAD card are required

Auto Negotiation is available.
100BASE-TX is recommended.
Maximum voice channels per card
G.729a G.711 G.723.1
10 ms.: 12 ch
12ch
_
20 ms.: 20 ch
20ch
_
30 ms.: 30 ch
30ch
24ch
40 ms.: 32 ch
32ch
_
Maximum voice channels per card
G.729a G.711 G.723.1
10 ms.: 20ch
20ch
20 ms.: 32ch
32ch
30 ms.: 32ch
32ch
24ch
40 ms.: 32ch
32ch
Maximum voice channels per card
G.729a G.711 G.723.1
20 ms.: 6ch
5ch
30 ms.: 8ch
7ch
8ch
40 ms.: 12ch
10ch
Setting is available per Location No.
For connection between DtermIPs
For connection via the IPT card

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Chapter 14 Common Channel Inter-Office Signaling (CCIS)

DTI Specifications
Characteristics

24-Channel

30-Channel

Line Rate
Line Code
Line Impedance

1.544 Mbps – 50 ppm
AMI with ZCS/B8ZS*
100 ohms

Pulse Amplitude
(Base to Peak)

3 volts – 0.6 volts

Pulse Width

324 ns – 30 ns

2.048 Mbps – 50 ppm
HDB3 (High Density Bipolar 3)
75 ohms + 100 mH
(Coaxial Cable)
120 ohms + 160 mH
2.37 volts nominal
(Coaxial Cable)
3 volts nominal
(Twisted-Pair Cable)
244 ns nominal

Output

Input

2.048 Mbps – 50 ppm
1.5 volts – 2.7
(Coaxial Cable)
1.5 volts – 3.3 volts
(Twisted-Pair Cable)

Line Rate
Pulse Amplitude
(Base to Peak)

1.544 Mbps – 2000 bps (130 ppm)
1.5 volts – 3 volts

Frame Synchronization
Pattern

100011011100

Input Jitter

ITU-T Fig. 1/G743

Wander

+138UI, -193UI or
-138UI, +193UI

ITU G823

Cable Length from
PBX to MDF or
External Equipment

Max. 200 m (656.2 ft.)
[With 0.65 gauge (22 ABAM)
twisted pair cable]

Maximum 400 m (1312.4 ft.)
[With 0.65 gauge (22 ABAM)
twisted pair cable]

*AMI: Alternate Mark Inversion
ZCS: Zero Code Suppression
B8ZS: Bipolar Eight Zero Substitution.

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Page 14-21

Chapter 15 Integrated Services Data Network
(ISDN)
ISDN Primary Rate Interface
ISDN is becoming a basic requirement for any PBX system sold today. The UNIVERGE NEAX
2000 IPS supports many ISDN BRI/PRI features and services. AT&T and Northern Telecom
specifications are both supported by the IPS. ISDN PRI trunks can use all of the same features
of the UNIVERGE NEAX 2000 IPS that any DID or DOD type trunk can use. For example, ISDN
PRI trunks can terminate to the PBX and use features such as Internal Automated Attendant,
DISA, Attendant Overflow, etc. In a typical installation, ISDN trunks will be terminated as DID
type lines, but the IPS also allows ISDN trunks to terminate as ring down type lines. As this
section will convey, the UNIVERGE NEAX 2000 IPS provides a very attractive ISDN PRI
package.

ISDN FEATURE LIST

CALL-BY-CALL SERVICE SELECTION
CALLING PARTY RECOGNITION SERVICE
(DIRECT-IN TERMINATION (DIT))
CPN TO NETWORK—PRESENT
CPN TO TERMINATING USER—DISPLAY
DID ADDRESSING
DID AND DOD ADDRESSING
EVENT BASED CCIS
ISDN TERMINAL
MEGACOM® ACCESS/WATS
MEGACOM® 800 SERVICE/800 WATS/ULTRA WATS
MULTIQUEST® /900 SERVICE
SUB-ADDRESS—PRESENT
TRUNK PROVISIONING SERVICE SELECTION

PRI Services & Features
Services provided by ISDN carriers are separate and different from features supported by
ISDN carriers. Services, in general terms, are often entire packages provided by the carriers,
such as AT&T’s Megacom®, which provides routing options and many reporting features.
Features of ISDN trunks provide the user with items such as Calling Party Number (CPN) and
Sub-Addresses. The distinction between services and features should be recognized to a
certain extent. This section will explain which features and services the UNIVERGE NEAX 2000
IPS supports.
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Page 15-1

Chapter 15 Integrated Services Data Network (ISDN)

Call-By-Call Service Selection
Services can be selected on a call-by-call basis to all channels of a single PRI interface according to
applications. That is, unlike Trunk Provisioning Service in which services are assigned to specific
channels, services may be used on any available channel.
Service Conditions
1. The services that can be designated include ACCUNET, MEGACOM, MEGACOM 800, INTERNATIONAL
800, SDN, MULTIQUEST (AT&T), and PRIVATE, INWATS, OUTWATS, FX, TIE (Northern Telecom).
2. Channel selection is possible by the LCR function only.
Note: During call termination, there is no indication of which service is being used.
3. Up to 5-digit Network ID of Network-Specific Facilities Information Element can be sent to ISDN Network.
4. Transit Network Selection Information Element can be sent to ISDN Network.
5. Service conditions #3 and #4 are available with the following firmware of PN-24PRTA:
SC-2795 IXS PRT PROG-A1: Issue 2.01 or later
SC-2888 IXS PRT PROG-A2: Issue 1.01 or later
When the firmware issue is earlier than above, up to 3-digit Network ID can be sent to the ISDN Network.

Called Party Recognition Service (Direct-In Termination (DIT))
This feature provides an incoming Direct-In Termination (DIT) call via an ISDN trunk to be connected to
a predetermined station. This application can be used for a station or modem. For incoming calls in a
Primary Rate Interface trunk, this service feature should be used only when DID trunks are not desired.
Note: Refer to the Business Features and Specifications manual for details on service Conditions

CPN To Network–Present
This feature allows the ISDN network to be informed of the calling party number (CPN) when a call
originates from a terminal connected to the System.
Service Conditions
1. The data provided as the calling station number is assigned via MAT/CAT for each ISDN circuit or station.
In addition, if no data has been assigned as the calling station number, the system will not provide any
information to the network.
2. A maximum of 16 digits can be assigned as the calling station number.
3. The maximum number of area codes and office code patterns is 15.
4. Transmission of CPN to the ISDN network can be programmed by the Class of Service. Programming is
required for each station.
5. The delivery of CPN information is subject to local regulations.

Page 15-2

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Chapter 15 Integrated Services Data Network (ISDN)

CPN To Terminating User–Display
This feature provides a visual display of the originating station’s number and subaddress information on
a Multiline Terminal or an Attendant Console for incoming ISDN calls. This provides the terminal user
with a quick and accurate way to identify the originating station’s number (Calling Party Number).
Service Conditions
1. This feature is available on the Multiline Terminal with Display, Attendant Console and Dterm PSIII.
2. A maximum of 16 digits forming the originating party’s number (CPN), including the PBX access code, can
be displayed. For sub-address, a maximum of 8 digits can be displayed. If the subaddress exceeds 8 digits,
the first 8 are displayed.
3. The CPN is flashing while the Multiline Terminal is ringing on its Prime Line. The duration of display after
the call is answered can be selected by system data programming (display for 6 seconds or continuously).
4. If the ISDN network provides Name Display service (NI-2/DMS100), the calling party name can be
displayed in place of the CPN (A maximum of 16 characters).
5. The calling name from ISDN can be received by terminating systems (CM30 YY=02, 03) such as Trunk
Direct Appearance (02), Trunk Direct Appearance + TAS (03), Direct-In Termination (04), Automated
Attendant(09), ISDN Indial (18), and Attendant Console (14). In addition, Sub-address termination is
available.
6. The 24PRT card is required.
7. When Call Forwarding or Call Transfer routes an incoming call from ISDN to a Multiline Terminal/PS in the
own office, the calling name is displayed on the destination station of Call Forwarding or Call Transfer.
8. Calling Name Display on ISDN Terminals is not supported.
9. Calling Name Display or Calling Number Display can be selected as the initial display in station Class of
Service.
10. The display can be changed to the calling name or calling number by pressing the display selection key on
a Multiline Terminal or an Attendant Console while the number or name is displayed.
11. When the terminating system (CM30 YY=02, 03) is set to Trunk Direct Appearance or Attendant Console,
the calling name is displayed only after the called party answers the call.
12. Conditions on Tandem Connection
a. The calling name received from ISDN is relayed, only via CCIS, with tandem connection. (A
maximum of 16 characters)
b. When Call forwarding routes an incoming call from ISDN to CCIS, the calling name is relayed via
CCIS to the destination of Call Forwarding.
c. When the calling name from NI-2 is received as a FACILITY message, the name cannot be relayed
to CCIS. Only when the calling name from NI-2 is received as a SETUP message, the name can
be relayed to CCIS.
d. Only when the calling name from DMS100 is received as a SETUP message, the name can be
relayed to CCIS.
13. Conditions with Other Services
a. The calling name received from ISDN has priority over the calling name registered to Station
Speed Dialing by CM74, so the former is displayed even though the latter has been assigned by
CM74.
b. If the calling name is not received from ISDN, the calling name assigned by CM74 is displayed if
available.
c. When a call is relayed to CCIS with tandem connection, the calling name is not relayed to CCIS, if
the name is not received from NI-2, even though the calling name has been assigned by CM74.
However, when Call Forwarding - No Answer routes an incoming call from ISDN to CCIS with
tandem connection, the calling name assigned by CM74 is relayed to CCIS.

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Chapter 15 Integrated Services Data Network (ISDN)

DID Addressing
This feature allows incoming ISDN-PRI calls to terminate to stations, Attendant Console, Automated
Attendant, etc., based on the Called Party number. Direct Inward Dial trunks will be terminated to
preprogrammed destinations without Attendant assistance.
Service Conditions
1. If the network is notified that the destination station for the DID call is busy or a connection-controlled
station, the network gives the calling station a Busy Tone. (This depends upon call forwarding services
being in service.)
2. If the called station is nonexistent, the DID call can be routed to the Attendant Console, another predesignated station, or to receive Reorder Tone.
Note: Refer to the Business Features and Specifications manual for more details of DID service.

DID and DOD Addressing
This feature allows the system to use DID and DOD on the same B channels. Trunk Provisioning
Service Selection is not required. (B-channels can be used for DID and DOD without separating the
trunk routes.)
Service Conditions
1. Confirm the DID and DOD selection with the customer’s local exchange carrier prior to installation.
2. The DID/DOD is supported without service provisioning, since no facility indication to the network is
required.
Note: For more details, refer to “Direct Inward Dialing” and “Direct Outward Dialing” in the Business Features
and Specifications manual.

ISDN Terminal
This feature provides the system with an ISDN Terminal or Terminal Adapter (TA). ISDN Terminal to
ISDN Terminal, ISDN Terminal to ISDN Trunk, ISDN Trunk to ISDN Terminal, ISDN Terminal to Single
Line Telephone, ISDN Terminal to Multiline Terminal, and ISDN Terminal to PS connections are
available.
Service Conditions
1. The ILC (ISDN Line Circuit) card and the ICH (ISDN Channel handler) card are required.
2. The ISDN Terminal must be locally powered.
3. The following connections are available:
Point to Point connection
Point to Multipoint connection
National ISDN 1
AT&T
4. The following features are available:
Individual Terminal Calling (Point to Multipoint connection)
Group Calling (Point to Multipoint connection)
Called Party Recognition Service (DIT)
DID Addressing
DID and DOD Addressing
CPN to Network-Present
CPN to Terminating User-Display
Sub Address-Present
Direct Outward Dialing (DOD)
Restriction from outgoing call
Toll restriction
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Chapter 15 Integrated Services Data Network (ISDN)
Station to Station Calling
Station Hunting
Simultaneous voice and data transmission
Asynchronous Data Switching
Synchronous Data Switching
Half/Full Duplex Switchover
5. The ISDN Terminal can provide Voice and data communication via the CCIS network. The CCIS network
must be digital.
6. During communication via the CCIS network, the link reconnection is not available.
7. The conditions about station services are as follows:
Station Hunting
When a Single Line Telephone/Multiline Terminal or an ISDN Terminal calls a busy Single Line Telephone/Multiline Terminal, another Single Line Telephone/Multiline Terminal in Station Hunting group is
called. When a Single Line Telephone/Multiline Terminal or an ISDN Terminal calls a busy ISDN
Terminal, an-other ISDN Terminal in Station Hunting group is called.
Note: An ISDN Terminal cannot be included in the same Station Hunting group as Single Line
Telephone/Multi-line Terminals.
Call Forwarding
An ISDN Terminal cannot set Call Forwarding - All Calls/- Busy Line/-Don’t Answer, and cannot be the
destination of a Call Forwarding. When an ISDN Terminal is a calling station, it can be forwarded to
another Single Line Telephone/Mul-tiline Terminal but cannot be forwarded to central office trunk or tie
line trunk.
Call Transfer
A Single Line Telephone/Multiline Terminal cannot transfer a call in progress with an ISDN Terminal to
another station. In the same way, an ISDN Terminal cannot transfer a call in progress with a Single
Line Telephone/Mul-tiline Terminal to another station. While a Single Line Telephone/Multiline
Terminal converses with CCIS trunk and ISDN trunk, the Single Line Telephone/Multiline Terminal
cannot transfer the call to an ISDN Terminal.
Call Pickup, Call Pickup - Designated group
An ISDN Terminal cannot be assigned to Call Pickup group and cannot pickup a call to another station.
A Single Line Telephone/Multiline Terminal cannot pickup a call to an ISDN Terminal.
Executive Override
During voice communication between a Single Line Telephone/Multiline Terminal and an ISDN
Terminal, another Single Line Telephone/Multiline Terminal can interrupt into only Single Line
Telephone/Multiline Terminal. During data communication with an ISDN Terminal, Executive Override
is not allowed.
Busy Service (Step Call, Call Back)
When a Single Line Telephone/Multiline Terminal calls a busy ISDN Terminal, busy service such as
Step Call and Call Back cannot be provided to the ISDN Terminal. It is the same when an ISDN
Terminal calls a busy Single Line Telephone/Multiline Terminal.
8. When a Single Line Telephone is calling an ISDN Terminal or an ISDN Terminal is busy, hooking service is
not available. Therefore, at this time other services are not available.
9. A Multiline Terminal can call an ISDN Terminal via Primary Extension or Sub Line. And a Multiline Terminal
can be called from an ISDN Terminal via Primary Extension or Sub Line.
10. When a Single Line Telephone/Multiline Terminal calls an ISDN Terminal, it can send the calling station
number to the ISDN Terminal. At this time, ISDN trunk number and local number are also attached to be
sent.
11. Station to Station calling between an attendant console and an ISDN Terminal is not available.

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Chapter 15 Integrated Services Data Network (ISDN)

12. In case of Point to Multipoint connection, a Single Line Telephone/Multiline Terminal must dial ISDN Multipoint station number assigned by CM1B, not the ISDN Terminal number assigned by CM10.
13. When a Multiline Terminal calls an ISDN Terminal or an ISDN Terminal calls a Multiline Terminal, a calling
station number is displayed to the calling Multiline Terminal or the ISDN Terminal. In case of Point to
Multipoint connection, the calling station number displayed is the ISDN station number assigned by CM10.
14. Station to Station calling between a PS station and an ISDN Terminal is available.
15. Only Preset Dialing can be used from an ISDN Terminal, Overlap dialing is not available.
16. For E-CCIS you must have 64K unrestricted digital data connections (circuit mode). The Telephone
Company cannot overflow these ISDN calls to analog lines.

17. Group Call, This feature terminates a call to all ISDN Terminals or Terminal Adapters
accommodated on the same bus. Group Call is available for following connections:
ISDN trunk to ISDN Terminal
ISDN Terminal to ISDN Terminal
Single Line Telephone to ISDN Terminal (Series 3200 R6.2 enhancement)
Multiline Terminal to ISDN Terminal (Series 3200 R6.2 enhancement)

MEGACOM ® Access/WATS
AT&T’s MEGACOM ® (WATS) network, as well as WATS from other carriers, can be used.
Service Conditions
1. The available WATS service is limited to MSB (Maximal Subscribed WATS Band).
2. No specific band will be indicated to the Network.

MEGACOM ® 800 Service/800 WATS Ultra WATS
AT&T’s MEGACOM ® 800 (Inward WATS) network, as well as 800 WATS provided by other carriers,
can be used.
Service Conditions
1. Command 76 can be used to convert indialed digits.
2. The Dialed Number Identification Service (DNIS) must match the station numbering plan (CM20).
3. Multiple DNIS are supported.

MULTIQUEST ® /900 Service
AT&T’s MultiQuest ® service can be used. (It is a “900”-type service.) Also, 900 service provided by
other carriers can be used.
Service Conditions
1. The DNIS (Dialed Number Identification Service) must match the numbering plan (CM20) of the 2000 IPS.
2. Multiple DNIS numbers are supported.
3. Command CM76 can be used to convert DNIS digits received from the ISDN Network.

Subaddress–Present
This feature allows a primary rate interface ISDN trunk to transfer the called party subaddress
information to a destination ISDN station when the call is originated by the system. Dialing the called
party station number and subaddress is required.
Service Conditions
1. If the calling party fails to dial the called party subaddress, ISDN cannot transfer any called party
subaddress information to the destination party.
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Chapter 15 Integrated Services Data Network (ISDN)
2. If a calling party does not wish to provide a called party subaddress, the call must terminate with the # key
(Immediate Start). If the # key is not entered, a Timing Start operation begins. The Timing Start uses the
interdigit timeout operation.
3. Subaddress dialing is available only on those telephone terminals that can generate push-button (DTMF)
signals.
4. The called party subaddress must not exceed 8 digits.
5. The called party subaddress can be sent with trunk direct dial access.
6. This feature cannot be used when a call is originated to ISDN using Speed Dialing or Call Forwarding
features.

Trunk Provisioning Service Selection
Each channel of a PRI interface can be dedicated to a particular service. Services are designated to
specific channels; once designated, a channel can be used only for that service.
Service Conditions
1. The services that can be designated include MEGACOM ® /WATS and MEGACOM ® 800/800WATS.
2. Arrangement with the carrier at the time of provisioning is needed.

Event Based CCIS
This feature allows a PBX customer that does not have a tie line (or when a customer cannot
use the tie line due to the busy or fault of the line), to use the various CCIS feature by using
the ISDN line as a CCIS virtual tie line, on the 2400 IPX - 2000 IPS connection or the 2000 IPS
- 2000 IPS connection.
Service Conditions
1. Event Based CCIS is available between 2000 IPS to 2400 IPX and 2000 IPS to 2000 IPS.
2. The maximum number of virtual tie lines is 16 channels per system. This includes the common signaling
channels and voice channels.
3. The ISDN line used for the virtual tie line can also be used as a regular ISDN line.
4. This feature supports voice calls only. (Supported object at PBX transmission side: Single Line Telephone,
Multiline Terminal, DID/E&M/Ring Down (analog/T1) tandem calls). The data calls are transmitted via the
regular ISDN network.
5. The Peg Count (the number of originating call from the ISDN trunk) is counted when using the ISDN line for
the virtual tie line by route basis.
6. Billing information of the virtual tie line using the ISDN line is treated as regular tie line calls.
7. Billing information of the virtual tie line using the ISDN line is treated on tandem calls.
8. The voice channel of the virtual tie line is released at a programmable time after the call finishes.
9. The CCH (Common Channel Handler) card is required for providing Event Based CCIS.
10. For Event Based CCIS using PRI, the following services must be ordered from Telco on the PRI Span:
Each B channel must be capable of making and receiving 64K voice calls.
Each B channel must be capable of making and receiving 64k Unrestricted Digital Data calls.
The ISDN PRI span must be ordered with ISDN Indial (DID) service.
11. The PRI trunks should be ordered with the following components included:
Protocol must be National ISDN-1
Calling and Called Party Sub-Address
NIUF Capability Package S
Two B-channels capable of Voice and Data simultaneously on both B-channels
Two Directory numbers, each with a SPID
12. For any feature that requires a tandem connection (Call-Forward CCIS, etc.), additional Telco BRI circuits
are required. More than one BRI trunk is required to allow calls to go back to the PBX that transferred or
forwarded a call.
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Chapter 15 Integrated Services Data Network (ISDN)

Business Feature List
Business Feature List
Account Code
Elapse Call Timer
Answer Key
Forced Account Code
Attendant Delay Announcement
Group Listening
Attendant Overflow
Hold - Call
Authorization Code
Hold - Exclusive
Automated Attendant
Hold - Nonexclusive
Broker’s Call
Last Number Redial
Call Forwarding
Least Cost Routing 3/6 Digit
Call Forwarding - All
Maintenance Administration Terminal(MAT)
Calls Call Forwarding - Busy Line
Fault Messages
Call Forwarding - No Answer
Peg Count
Call Forwarding - Outside
Night Service
Multiple Call Forwarding - All
Call Rerouting
Multiple Call Forwarding - Busy
Day/Night Mode Change - ATTCON
Multiple Call Forwarding - No Answer
Day/Night Mode Change - Station
Split Call Forwarding - Busy Line
Night Connection - Fixed
Call Forwarding - Override
Night Connection - Flexible
Group Division
Trunk Answer from Any Station(TAS)
Call Park - System
Route Advance
Call Pickup - Direct
Save and Repeat
Call Pickup - Group
Stack Dialing
Class of Service
Station Hunting - Circular
Code Restriction
Station Hunting - Terminal
Conference
Station Hunting - Secretarial
Consecutive Speed Dial
Station Message Detail Recording(SMDR)
Consultation Hold
Station Speed Dialing
Delayed Ringing
System Speed Dialing
Direct Inward Termination(DID)
Tenant Service
DID Call Waiting
Trunk Queuing - Outgoing
DID Digit Conversion
Trunk to Trunk Connection
Direct Inward System Access(DISA)
Uniform Call Distribution(UCD)
Direct Inward Termination(DIT)
"Silent Monitor"
Distinctive Ringing
Voice Mail Integration
Do Not Disturb

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Chapter 15 Integrated Services Data Network (ISDN)

ISDN Network Requirements for Layer One (1)
Interface:
Framing:
Line Coding:
D-Channel Rate:
D-Channel:

T1 (1.544 M) only
ESF (24 Multi-frame) only
B8ZS only
64 kbps only
T1 Channel 24 (24th channel)

Note: Individual B-Channels cannot be looped back, local or remote.

Supported Network Services (Trunk provisioned only)
AT&T
Software Defined Network
Global Software Defined Network
International 800 Service
International MEGACOM
AT&T’s Alternate Destination Call Redirection feature is supported by the IPS.

NT
Incoming Note
- Private
-Tie
- WATS
- 800 WATS
Outgoing Note
- DID/DOD
Note: Services that do not require NSF can be supported by NEAX 2000 IPS.

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Chapter 15 Integrated Services Data Network (ISDN)

ISDN PRI Specifications
DTI
The Digital Trunk Interface (DTI) interfaces the PBX directly to 24/30-channel PCM transmission line.
The DTI has the following functions.
For 24DTI:

•
•
•
•
•
•

Two Directory numbers, each with a SPID
Unipolar/Bipolar Conversion (AMI/B8ZS Format)
Alarm Detection/Insertion
Digital PAD on Voice Signal Transmission
Loop-Back Test (Local/Remote Loop Back)
Cyclic Redundancy Checking (based on ITU-T Rec. G704)

For 30DTI:
• Unipolar/Bipolar Conversion (HDB3 Format)

•
•
•
•

Alarm Detection/Insertion
Digital PAD on Voice Signal Transmission
Loop-Back Test (Local/Remote Loop Back)
Cyclic Redundancy Checking (based on ITU-T Rec. G704)

For connections of 24DTI and transmission line, twisted-pair cable can be used. For connection of
30DTI and transmission line, either coaxial cable or twisted pair cable can be used.

DCH
The D-Channel Handler (DCH) provides the D-Channel signaling interface through the DTI to an ISDN
exchange, and it is responsible for signaling between the PBX and the ISDN exchange under control of
the system MP.

PRT
The Primary Rate Interface Trunk (PRT) provides the ISDN Primary Rate Interface (1.5 Mbps PCM-23B
+ D/2 Mbps PCM-30B + D) and a built-in D-Channel Handler (DCH). The PRT has the following
functions.
For 24PRT:

Unipolar/Bipolar Conversion (AMI/B8ZS Format)
Alarm Detection/Insertion
Digital PAD on Voice Signal Transmission
Loop-Back Test (Local/Remote Loop Back)
Cyclic Redundancy Checking (based on ITU-T Rec. G704)
For 30PRT:

Unipolar/Bipolar Conversion (HDB3 Format)
Alarm Detection/Insertion
Digital PAD on Voice Signal Transmission
Loop-Back Test (Local/Remote Loop Back)
Cyclic Redundancy Checking (based on ITU-T Rec. G704)
Page 15-10

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
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Chapter 15 Integrated Services Data Network (ISDN)

For connections of 24PRT and transmission line, twisted-pair cable can be used. For connection of
30PRT and transmission line, either coaxial cable or twisted pair cable can be used.
Note: ISDN requires B8ZS Line coding with Extended Superframe (ESF) format.

BRT
The Basic Rate Interface Trunk (BRT) provides one or two physical interface to the ISDN-Basic Rate
Interface service (192 kbps PCM-2B + D).
The BRT has the following functions.
Unipolar/Bipolar Conversion (AMI/B8ZS Format) (S/T Interface)
Signaling Insertion/Extraction
Frame Synchronization
Digital PAD on Voice Signal Transmission
For connections of BRT and transmission line, twisted-pair cables can be used.
Note: We recommend the point-to-point connection when connecting the system to the public network using the
BRT card (Set the second data of CM35 Y=79 to 0). For the point-to-multipoint connection using the BRT card,
when the system is established far from the public network, a communication error occurs easily because the
ISDN signal fades away.

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
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Page 15-11

Chapter 15 Integrated Services Data Network (ISDN)

Feature Compatibility Table
Inter-Exchange Carrier Network Voice Services and Features /
UNIVERGE NEAX 2000 IPS Compatibility
SERVICES AND FEATURES
INTER-EXCHANGE CARRIERS
CALL BY CALL SERVICE
SELECTION
Basic Call Note 1
WATS
800 Service
Virtual Private Network
900 Service Note 2
900 Service Flexible Billing
International WATS
International 800 Service
Foreign Exchange (FX)
Tie Line
DEDICATED ACCESS
PROVISIONING
Basic Call Note 1
WATS
800 Service
Virtual Private Network
900 Service (Note 2)
900 Service Flexible Billing
International WATS
International 800 Service
Foreign Exchange (FX)
Tie Line
FACILITY MANAGEMENT
B-Channel Negotiation
D-Channel Back-Up
Non-Facility Associated Signaling
Release Link Trunks

AT&T AT&T
4ESS NET
X
X
X
X
X
X
X
-

X
X
X
X
X
X
X
-

AT&T AT&T
4ESS NET
X
X
X
X
X
X
X
-

X
X
X
X
X
X
X
-

AT&T AT&T
4ESS NET

X
X
X
X
X
X
X = Feature Available / - = NOT Available

Page 15-12

NTI
MCI SPRINT
DMS
NET
NET
250
X
Note 1 Note 1
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
NTI
MCI SPRINT
DMS
NET
NET
250
X
Note 1 Note 1
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
NTI
MCI SPRINT
DMS
NET
NET
250
X
X
X
X
X
X
X
X
X

UNIVERGE NEAX 2000
IPS
Note 1
X
X
X
X
X
X
X

UNIVERGE NEAX 2000
IPS
Note 1
X
X
X
X
X
X
X
-

UNIVERGE NEAX 2000
IPS
-

UNIVERGE NEAX 2000 IPS Request for Proposal (RFP) Reference Guide
Issue 6

Chapter 15 Integrated Services Data Network (ISDN)

VOICE FEATURES
CPN to Network
Presentation Restrict/Allow
CPN/BPN Delivery
Provisioned by Service
Call by Call Request
While All Trunks Busy
CPN Sub-Address
Calling Party Name
Dialed Number ID Service (DNIS)
Original CPN
Original CPName
Redirecting Number
Redirecting Name
Redirection Number
Redirection Name
Called Party Sub-Address
Connected Party Number
Connected Party Name
User To User
Message Associated UUI
Call Associated TSC
Non-Call Associated TSC
Network Ring Again
Customer Group Info. Transport
Alternate Destination
Account Code Prompt and Send

AT&T AT&T
4ESS NET
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X

X
X
X
X
X
X
X
X
X
X
X
X
X
X
X

NTI
DMS
250
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X

MCI
NET

SPRINT
NET

UNIVERGE NEAX 2000

X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X

X
X
X
X
X
X
X
X
-

IPS

X = Feature Available / - = NOT Available
Note 1: The use of a Basic Call type of call is unique to NTI in the direct connection to the long distance
network. This is sometimes referred to by the IEX carriers as "Megalink" service.
Note 2: AT&T and NTI have used different signaling to indicate the "900 service" selection. The NEAX2000
IVS supports the AT&T defined signaling.

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Issue 6

Page 15-13

Chapter 15 Integrated Services Data Network (ISDN)

Local Exchange Carrier Network Voice Services and Features/
UNIVERGE NEAX 2000 IPS Compatibility
SERVICES AND FEATURES
INTER-EXCHANGE CARRIERS
CALL BY CALL SERVICE
NTI
AT&T AT&T
MCI SPRINT UNIVERGE NEAX 2000
SELECTION
DMS
IPS
4ESS NET
NET
NET
SWITCHED DATA
250
64 kbps Clear Channel
X
X
X
X
X
X
64 kbps Restricted
X
X
X
X
X
56 kbps Restricted
X
X
X
X
X
International 64 kbps Clear
X
X
X
X
X
X
International 64 kbps Restricted
X
X
X
X
X
International 56 kbps Restricted
X
X
X
X
X
HO 384 kbps
X
X
X
X
H11 1526 kbps
X
X
X
X
N x 64 kbps
X
X
NTI
AT&T AT&T
MCI SPRINT UNIVERGE NEAX 2000
VIRTUAL PRIVATE NETWORK
DMS
4ESS NET
NET
NET
IPS
250
64 kbps Clear Channel
X
X
X
X
X
64 kbps Restricted
X
X
X
X
X
56 kbps Restricted
X
X
X
X
X
International 64 kbps Clear
X
X
X
X
X
International 64 kbps Restricted
X
X
X
X
X
International 56 kbps Restricted
X
X
X
X
X
HO 384 kbps
X
X
X
X
H11 1526 kbps
X
X
X
X
N x 64 kbps
X
X
DEDICATED ACCESS
NTI
AT&T AT&T
MCI SPRINT UNIVERGE NEAX 2000
PROVISIONING
DMS
4ESS NET
NET
NET
IPS
SWITCHED DATA
250
64 kbps Clear Channel
X
X
X
X
X
X
64 kbps Restricted
X
X
X
X
X
56 kbps Restricted
X
X
X
X
X
International 64 kbps Clear
X
X
X
X
X
X
International 64 kbps Restricted
X
X
X
X
X
International 56 kbps Restricted
X
X
X
X
X
HO 384 kbps
X
X
X
X
H11 1526 kbps
X
X
X
X
N x 64 kbps
X
X
X = Feature Available / - = NOT Available

Page 15-14

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Issue 6

Chapter 15 Integrated Services Data Network (ISDN)

VIRTUAL PRIVATE NETWORK
64 kbps Clear Channel
64 kbps Restricted
56 kbps Restricted
International 64 kbps Clear
International 64 kbps Restricted
International 56 kbps Restricted
HO 384 kbps
H11 1526 kbps
N x 64 kbps

AT&T AT&T
4ESS NET
X
X
X
X
X
X
X
X
-

X
X
X
X
X
X
X
X
-

NTI
DMS
250
X
X
X
X
X
X
X
X
X

MCI
NET
X
X
X
X
X
X
X
X
X

SPRINT UNIVERGE NEAX 2000
NET
IPS
X
X
X
X
X
X
-

Local Exchange Carrier Network Voice Services and Features
UNIVERGE NEAX 2000 IPS Compatibility
SERVICES AND FEATURES
LEC CARRIERS
UNIVERGE NEAX 2000 IPS
CALL BY CALL SERVICE
AT&T
NTI
SIEMENS
SELECTION
5ESS DMS100
EWSD
Basic Call
X
X
X
X
WATS
Maximal Band
X
X
X
X
Banded
X
X
X
800 Service (INWATS)
X
X
X
X
FX LINE
X
TIE LINE
X
Virtual Private Network
X
X
X
Electronic Tandem Network
X
Operator Access (Note 2)
LEC Operator
X
X
X
X
Default IEX Operator
X
X
X
X
Other IEX Operator
X
DEDICATED ACCESS
AT&T
NTI
PROVISIONING
5ESS DMS100
Basic Call
X
X
WATS
Maximal Band
X
X
800 Service (INWATS)
X
X
FX LINE
X
TIE LINE
X
Virtual Private Network
X
X
Electronic Tandem Network
X
X = Feature Available / - = NOT Available

SIEMENS
EWSD
X
-

UNIVERGE NEAX 2000 IPS Request For Proposal (RFP) Reference Guide
Issue 6

UNIVERGE NEAX 2000 IPS
X
X
X
X
-

Page 15-15

Chapter 15 Integrated Services Data Network (ISDN)

AT&T
NTI
5ESS DMS100
Operator Access (Note 2)
LEC Operator
Default IEX Operator
Other IEX Operator
FACILITY MANAGEMENT
D-Channel Back-Up
Non-Facility Associated Signaling
VOICE FEATURES
CPN to Network
Presentation Restrict/Allow
CPN/BPN Delivery
Calling Party Name
Original CPN
Original CPName
Redirecting Number
Redirecting Name
Redirection Number
Redirection Name
Connected Party Number
Connected Party Name
User To User
Message Associated UUI
Network Ring Again
Network Message Waiting
Message Waiting Indicator
Message Waiting Activation

SIEMENS
EWSD

UNIVERGE NEAX 2000 IPS

X
X
-

X
X
X
-

X
X
-

X
X

AT&T
NTI
5ESS DMS100
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X

SIEMENS
EWSD
X
X
X
-

UNIVERGE NEAX 2000 IPS
X
X
X
-

X
-

X = Feature Available / - = NOT Available
Note 1: The Siemens EWSD offers PRI support via both a National ISDN (Bellcore) and an AT&T
"Custom" implementation. The above information pertains to the "Custom" implementation only (NTI has no
National ISDN support). Although this interface is based on the AT&T 5ESS, NEC products have not been
tested on the EWSD. Please contact NEC Product Management for any potential sales requiring PRI
connection via the EWSD "Custom" interface.
Note 2: Support for Operator Access by the AT&T and NTI systems are realized in different ways. The
NEAX2000 supports the AT&T implementation.

Page 15-16

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Issue 6

Chapter 15 Integrated Services Data Network (ISDN)

Local Exchange Carrier Network Data Services and Features
UNIVERGE NEAX 2000 IPS Compatibility
SERVICES AND FEATURES
LEC CARRIERS
Call By Call Service Selection
SWITCHED DATA
64 kbps Clear Channel
64 kbps Restricted
56 kbps Restricted
HO 384 kbps
H11 1526 kbps
N x 64 kbps

AT&T
NTI
SIEMENS
5ESS DMS100
EWSD
X
X
X
-

X
X
X
X
X
X

X
X
X
-

64 kbps Clear Channel
64 kbps Restricted
56 kbps Restricted
HO 384 kbps
H11 1526 kbps
N x 64 kbps

AT&T
NTI
SIEMENS
5ESS DMS100
EWSD
X
X
X
X
X
X
X
X
X
-

DEDICATED ACCESS
PROVISIONING

AT&T
NTI
SIEMENS
5ESS DMS100
EWSD

VIRTUAL PRIVATE NETWORK

SWITCHED DATA
64 kbps Clear Channel
X
64 kbps Restricted
X
56 kbps Restricted
X
HO 384 kbps
H11 1526 kbps
N x 64 kbps
VIRTUAL PRIVATE NETWORK
64 kbps Clear Channel
X
64 kbps Restricted
X
56 kbps Restricted
X
HO 384 kbps
H11 1526 kbps
N x 64 kbps
X = Feature Available / - = NOT Available

UNIVERGE NEAX 2000 IPS
X
-

UNIVERGE NEAX 2000 IPS
-

UNIVERGE NEAX 2000 IPS

X
X
X
X
X
X

X
X
X
-

X
-

X
X
X
X
X
X

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Page 15-17

Chapter 15 Integrated Services Data Network (ISDN)

ISDN Basic Rate Interface (BRI)
The UNIVERGE NEAX 2000 IPS systems equipped with Basic Rate Interface (BRI)
provides the user with a vehicle for circuit switched data at rates of 64 or 128 kbps to
the desktop. ISDN BRI connects BRI terminals (stations) to the PBX. ISDN PRI is used
to connect the PBX to the public network. Circuit switched data is the focus of the IPS
Basic Rate Interface. Features such as Group Four Fax, video conferencing, and data
calls are the main objective of BRI on the UNIVERGE NEAX 2000 IPS.
Applications
The UNIVERGE NEAX 2000 IPS supports National ISDN - 1 and AT&T 5ESS specifications for
Basic Rate Interface.
Desktop Video Applications
From the desktop, users can place and receive 128 K video conference calls.

PSTN
BRI

PRI

BRI

NT- 1
Desktop Video

Desktop Video

UNIVERGE NEAX2000 IPS

Video Conferencing for large conference rooms or classrooms.
Using an I-MUX to receive 3 BRI ports, a bandwidth of 384 kbps is possible. This provides good quality
video for large conference rooms.

Video CODEC

I-MUX

Video CODEC

PRI
BRI x 3

PSTN
I-MUX
BRI x 3

Page 15-18

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Chapter 15 Integrated Services Data Network (ISDN)
ISDN Data Modems
ISDN Data Modems will provide users with access to the Internet at speeds of 64 or 128 kbps. ISDN
modems can place and receive ISDN calls at 56, 64, and 128 kbps. Data calls within the PBX or external
using ISDN PRI can greatly increase the user’s productivity by using the higher rates of ISDN BRI. NEC
America has successfully tested the IBM Waverunner series of ISDN modems.
Group four Fax
Fax machines capable of G4 can send faxes at speeds of 56 kbps using ISDN BRI.

System Requirements
BRI stations within the IPS can place calls to other BRI stations or outside the PBX by using ISDN PRI
lines to the public network. Each IPS system can support up to 24 BRI Terminal ports.
The PN-2ILCA card is used for to connect the BRI terminal to the PBX. An SPN-SC03 card is required
to control D channel activity for the BRI terminals. One SC03 card supports four 2ILCA cards. The
UNIVERGE NEAX 2000 IPS can support 48 ILCA cards that mount in LT slots and 12 SC03 cards that
mount in AP slots.

SPN-SC03

ONE SC03 SUPPORTS
FOUR 2ILCA

PN-2ILCA

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Page 15-19

Chapter 15 Integrated Services Data Network (ISDN)

ISDN Equipment List
Functional
Name
BRT

Card Name
PN-BRTA
PN-4BRTA-F

BRT

PN-24DTA-C

DTI

PN-2ILCA

ILC

PN-24PRTA

PRT

PN-SC00

CCH

PN-SC01

DCH

PN-SC03

ICH

Function
1-line Basic Rate (2B + D) Interface Trunk Card
Accommodates one 2-channel PCM digital lines.
4-line Basic Rate (2B + D) Interface Trunk Card
accommodates four 2-channel PCM digital lines.
Digital Trunk Interface (23B + D, 1.5 Mbps) Card
accommodates 24-channel PCM digital lines.
2-line ISDN Line Circuit Card provides a physical
interface to ISDN Terminals. Occupies 8 time slots
per one card.
ISDN Primary Rate (23B + D) Interface Card
provides a built-in D-channel Handler (DCH)
Common Channel Handler Card transmits/receives
signal on the common signaling channel of No. 7
CCIC.
D-channel Handler Card transmits/receives signal
on the D-channel of ISDN Primary Rate (23B + D)
interface or WCS Roaming interface.
ISDN-channel Handler Card provides the Dchannel signaling interface and controls. Maximum
four ILC cards (Layer 2 and 3)

ISDN PRI Required Equipment
Equipment
PN-24DTA-C
PN-24PRTA
PN-SC01 (DCH)

Description
24-Channel DTI Card
24-Channel PRT Card
D-Channel Handler Card

Qty
1-8
1-8
1-8

Remarks
1 DTI/Card

ISDN BRI Required Equipment
Equipment
PN-BRTA (BRT)
PN-4BRTA-F (BRT)

Description
1-Line BRT Card
4-Line BRT Card

Qty
1-12
1-24

Remarks

ISDN Terminal Required Equipment
Equipment
PN-SC03
PN-2ILCA

Description
ISDN Channel Handler Card
ISDN Line Circuit Card

Qty
1-24
1-48

Remarks
Supports 4 ILC Cards
Supports 2 Terminals

Event Based CCIS Required Equipment
Equipment
PN-SC00

Page 15-20

Description
Common Channel Handler Card

Qty
1-8

Remarks
Supports 1 DTI Card

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Chapter 15 Integrated Services Data Network (ISDN)

Capacity for ISDN-PRI
Capacity
Description
DTI Card
DCH Card
Trunks for DTI
PRT Card
Trunks for PRT
ISDN Routes
ICH Card
ILC Card
Port per DTI Card
Port per DCH Card
Port per PRT Card
Port per ICH Card
Port per ILC Card

24DTI
8
8
192
8
24
48
24
1
4
8

24PRT
8
192
8
24
48
24 + 1 (DCH)
4
8

System Capacity for ISDN-BRI
Description
BRT Card (BRTA/4BRTA-F)
Trunks for BRT (BRTA/4BRTA-F)
ICH Card (SC03)
ILC Card (2ILCA)
Port per BRT Card (BRTA/4BRTA-F)
Port per ICH Card (SC03)
Port per ILC Card (2ILCA)

Capacity
12/24
24/192
24
48
2/8
4
8

Documentation
More information regarding the ISDN capabilities of the UNIVERGE NEAX 2000 IPS can
be found in the UNIVERGE NEAX 2000 IPS ISDN System Manual and the UNIVERGE
NEAX 2000 IPS ISDN Features and Specifications.

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Page 15-21

Chapter 16 Wireless System
UNIVERGE NEAX 2000 IPS Wireless Communication System
(WCS)
NEC’s wireless technology offers high system capacity, proven system reliability, high voice
quality, network integration, investment protection into the future, and extensive user mobility.
The UNIVERGE NEAX 2000 IPS Wireless Communication System (WCS) provides interface for
Personal Station (PS) via a Zone Transceiver (ZT). When installing the WCS, there are three
types of Configuration Applications:
1. Integrated Type: Wired for Wireless
The UNIVERGE NEAX 2000 IPS provides both PBX and WCS functions together.
2. Adjunct Type [1]: Analog PBX Interface
The WCS is positioned as a behind system of the existing PBX linked by LC-COT connection.
3. Adjunct Type [2]: CCIS Interface
The WCS is positioned as a behind system of the existing PBX linked by CCIS.

Wired for Wireless
This feature allows the UNIVERGE NEAX 2000 IPS to have the Wireless PBX feature. The IPS
provides several integrated features adding on the adjunctive configuration.
Service Conditions

1. The service conditions of each feature are the same as for the Single Line Telephone, except for
the following:
Calling Number Display – PS
Last Number Redial – PS
Speed Dial – PS
2. The station number of a Dterm PS III can be assigned flexibly. The limitation of the number and
maximum digits depends on the UNIVERGE NEAX 2000 IPS.

Analog PBX Interface
This feature allows the WCS to be integrated with many types of PBXs and to activate various PBX
services by sending hook flash signal and DTMF tone.
Service Conditions

1. The signal type of the following cases can be programmed on a route-by-route basis between the
WCS and the PBX.
To dial for placing a call.
To dial for accessing to the PBX service.
To dial after sending hook flash.

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Page 16-1

Chapter 16 Wireless System

2. The signal type of dial while a call is established is DTMF. It is not programmable.
3. If the PBX requires pause timing between a trunk access code and an external number, the WCS
can automatically insert the pause timing.
4. The called/calling party number is not displayed on the LCD of a Dterm PS III.
5. The interface between a PBX and the WCS is a LC and a COT. The same number of LC and COT
as Dterm PS III are required.
6. Code Restriction and Class of Service are provided by the PBX.
7. Any type of call is originated via the PBX, even a Dterm PS III-to-Dterm PS III call.
8. The trunk circuit of the WCS can recognize a momentary open signal as a release signal, if the PBX
provides it.
9. The station number of a Dterm PS III can be assigned flexibly. The limitation of the number and
maximum digits depends on the UNIVERGE NEAX 2000 IPS.

CCIS Interface
This feature allows the WCS to be integrated with NEC PBXs with CCIS interface.
Service Conditions

1. The service conditions of each feature are the same as those of the Single Line Telephone of CCIS
features, except the following features:
Calling Number Display – PS
Last Number Redial – PS
Speed Dial – PS
2. If the PBX requires pause timing between a trunk access code and an external number, the WCS
can automatically insert the pause timing.
3. The interface between a PBX and the WCS is CCIS. The number of voice link depends on the
traffic except PS-to-PS calls.
4. Code Restriction and Class of Service are provided by the PBX.
5. Any type of call is originated via the PBX, even a Dterm PS III-to-Dterm PS III call.
6. The station number of Dterm PS III can be assigned flexibly. The limitation of the number and
maximum digits depends on the UNIVERGE NEAX 2000 IPS.
7. If the SMDR is needed, AP00 card is required and the upper PBX must have a center node feature
of Centralized Billing - CCIS.

Page 16-2

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Chapter 16 Wireless System

Wireless Roaming
The UNIVERGE NEAX 2000 IPS supports the JT-Q931a protocol and JT-11582 for signaling at
Q-reference point between PBXs on the private network. By supporting this protocol, the PSs
can be used in any Calling Area on the private network.
Roaming can be executed only on trunk connection between PBXs based on JT-Q931a
protocol or IP. To each trunk route of JT-Q931a digital trunks/IP trunks, it can be specified
whether Roaming is provided or not. The JT-Q931a digital trunks/IP trunks can be used by
single line telephone stations and Dterm stations for originating or receiving calls as same as
common trunks.
The Data Base Module (DBM) card (PN-AP00-B) is required per PBX. The DBM card cannot
be used as billing application processor (for SMDR, MCI, PMS or Hotel printer). System data
stored in the memory of the DBM card can be saved, loaded and verified from a MAT. A
Roaming network can consist of maximum 512 PBXs. Visitor Location Register (VLR)
information for maximum 512 Visitor PSs can be recorded to a system. VLR information is the
various information of Visitor PS and is made in the memory of DBM on the Visitor PBX when
the PS is roaming. When the VLR information exceeds for more than 512 PSs, DBM
overwrites the oldest VLR information.
One of the following two Roaming service is available in the PBX.
Roaming service via D channel private line using Q931a digital trunks; Multi-Site Roaming
Roaming service via IP network using Virtual IPT; Q931a Roaming over IP Trunk

The following sections describe the system outline and system configuration of Multi-Site
Roaming/ Q931a Roaming over IP Trunk.

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Page 16-3

Chapter 16 Wireless System

Multi-Site Roaming
In a mixed network of both UNIVERGE NEAX 2400 IPX and UNIVERGE NEAX 2000 IPS systems or a
network of UNIVERGE NEAX 2000 IPS systems supporting wireless users in a private network, some
users may desire the ability to take their handsets from one site to the other and still remain in touch
with those who are calling. Dterm PS III users can originate or receive a call in any place of a network
provided by the plural PBXs which are interfaced by JT-Q931-a, common wireless roaming protocol for
network wide roaming capabilities.

Configuration Application: Integrated Type and Adjunct Type with CCIS interface
Service Conditions

1.
2.
3.
4.

A roaming service network can be provided by maximum 512 PBXs.
One AP00B card for Data Base Module (DBM) is required per PBX.
An AP00B DBM-C is required when accommodating 257 or more PS’s per site.
Visitor Location Register (VLR) information for maximum 512 Visitor PSs can be recorded to a
system. When the VLR information exceeds for more than 512 Visitor PSs, AP00 deletes the
oldest VLR information in the system.
5. When a call terminates to a Multiline Terminal from a PS, the ringing of the Multiline Terminal is as
follows, regardless of Home PS or Visitor PS:
When the PS is under control of the same PBX: Ringing for Internal call
When the PS is under control of the other PBX: Ringing for External call
6. When the Home PS is roaming, the Home PBX cannot route Announcement-PS No Answer /-PS
Out of Zone to the calling party. If the Visitor PBX allowed it will inform the Home PBX that the PS
cannot answer, the Home PBX activates Call Forwarding-Not Available.
7. Billing of Home PS is executed to its Virtual LC Number.
8. While the Home PS is roaming, billing information cannot be recorded.
9. The Visitor PS can allow following service features.
Originating a call to a station of the same office
Originating a call to a station of the different office
Terminating a call from a roaming trunk
PS Exclusive Hold
Call Forwarding
10. The visitor PS cannot allow Calling Name Display.
11. Maximum 64 Visitor PSs can originate or receive a call in a network at the same time.
12. The Visitor PS can answer a call only from the roaming trunk.
13. Billing of Visitor PS is executed to its Virtual LC Number.
14. Billing information for the Visitor PS can be recorded only when the Visitor PS originates a call.

Page 16-4

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Chapter 16 Wireless System

Q. 931a Roaming over IP Trunk
Dterm PS III user can originate or receive a call in any place of a network provided by the plural PBXs,
which are interfaced by JT-Q931-a protocol over IP Trunks.
Q931a Roaming over IP Trunk is
available only on the Point-to-Multipoint connection. Roaming by Q931a digital private line and
Roaming by IP network cannot be used together in one PBX. A Roaming IP network consists of
maximum 255 PBXs. Roaming via Virtual IPT and Roaming via Physical IPT cannot be used together
in one PBX. IP-PAD card is required for the Roaming via Virtual IPT, when the called/calling station is
the legacy terminal (Dterm/Single Line telephone/ATTCON, etc.).

Configuration Application: Integrated Type and Adjunct Type with CCIS interface
Service Conditions

1.
2.
3.
4.

A roaming service network can be provided by maximum 512 PBXs.
One AP00 card for Data Base Module (DBM) is required per PBX.
An AP00B DBM-C is required when accommodating 257 or more PS’s per site.
Both Multi-Site Roaming (using DTI) and this feature cannot be provided in the same system. i.e.
the roaming network must be configured in only DTI trunks or IP trunks.
5. Visitor Location Register (VLR) information for maximum 512 Visitor PSs can be recorded to a
system. When the VLR information exceeds for more than 512 Visitor PSs, AP00 deletes the
oldest VLR information in the system.
6. When a call terminates to a Multiline Terminal from a PS, the ringing of the Multiline Terminal is as
follows, regardless of Home PS or Visitor PS:
When the PS is under control of the same PBX: Ringing for Internal call
When the PS is under control of the other PBX: Ringing for External call
7. When the Home PS is roaming, the Home PBX cannot route Announcement-PS No Answer /-PS
Out of Zone to the calling party. If the Visitor PBX allowed it will inform the Home PBX that the PS
cannot answer, the Home PBX activates Call Forwarding-Not Available.
8. Billing of Home PS is executed to its Virtual LC Number.
9. While the Home PS is roaming, billing information cannot be recorded.
10. The Visitor PS can allow following service features.
Originating a call to a station of the same office
Originating a call to a station of the different office
Terminating a call from a roaming trunk
PS Exclusive Hold
Call Forwarding
11. The visitor PS cannot allow Calling Name Display.
12. Maximum 64 Visitor PSs can originate or receive a call in a network at the same time.
13. The Visitor PS can answer a call only from the roaming trunk.
14. Billing of Visitor PS is executed to its Virtual LC Number.
15. Billing information for the Visitor PS can be recorded only when the Visitor PS originates a call.

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Page 16-5

Chapter 16 Wireless System

Wireless Definitions
Virtual Station:

Virtual Station exists only on the system data, provided via non-hardware
supported LEN. The Virtual station must be assigned by the system data
programming for operating Home PSs and Visitor PSs used for Roaming.

Individual PS number:

Individual PS number is assigned to a PS to identify the PS on the Roaming
network. It must be a unique number in the network.

Network ID method:

Network ID method is one method to operate Roaming. A Roaming PS must
have two SYS-ID on the Network ID method. One is main SYS-ID for Home
PBX, and another is Network ID for Roaming network. The Network ID is
used to define whether the PS can operate under the control of PBXs on the
Roaming network. The network ID must be the same for all PBXs within the
same network.

Visitor PBX:

When a PS leaves control of a PBX to which it belongs originally, and is
operating in a zone of another PBX, the PBX is called “Visitor PBX”.

Visitor PS:

When a PS leaves control of a PBX to which it belongs originally, and is
operating in a zone of another PBX, the PS is called “Visitor PS”.

Home PBX:

Home PBX is a PBX to which a PS ordinarily belongs.

Home PBX ID:

Home PBX ID is a unique number to identify the PBX on the Roaming
network.

Home PS:

When a PS operates under control of a PBX to which the PS originally
belongs, the PS is called “Home PS”.

Roaming number:

Roaming number is assigned to a Visitor PS temporarily, when the PS is
roaming to a Visitor PBX. The actual Roaming number is Virtual station
number assigned as a pilot station of Station Hunting group on the Visitor
PBX.

HLR:

Home Location Register. A database to store the location registration data of
the Home PS.

VLR:

Page 16-6

Visitor Location Register. A database to store the location registration data of
the Visitor PS temporarily, when the Visitor PS is in the zone of another PBX.

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Chapter 16 Wireless System

Wireless – Short Text Message Notification (OAI)
This feature enables a Short Text Message (hereafter, STM), once arrived at a mail box of the
STM Server (external equipment), to be automatically distributed to the addressee Dterm PS
III via Open Application Interface (OAI). This feature also provides “TM Full” notification on the
display (LCD) of the address Dterm PS III that is busy, and if the Dterm PS III is out of zone,
the feature makes retransmission of the STM after the PS returns to the zone.
Note: This describes STM Facilities available for external STM OAI application. Actual functionality depends on
the application.

Configuration Application: Integrated Type
Service Conditions

The feature is available only for My Line (L1) of a two-line Dterm PS III (not provided for Sub Line (L2)).
1. STM transmission route is only one way: STM server fi Dterm PS III
2. One STM can contain 128 byte data at the maximum.
3. The following features are not available at this feature.
Call Forwarding - All Calls
Call Forwarding - No Answer
Call Forwarding - Busy Line
Call Forwarding - Not Available
4. If the following occur, STM is not distributed, but stays in the mail box of the STM server. The STM
is safely transmitted later by this feature’s retransmission function.
Addressee Dterm PS III is out of zone.
Addressee Dterm PS III is power off.
The line of addressee Dterm PS III is busy, on hold, or ringing.
Dterm PS III is in a state of hand-over. Note
5. With this feature, PCS Roaming is not available.
Note: When hand-over unexpectedly starts while the message transmission is partially in progress, this feature
sees this as a squelch disconnection and performs message retransmission after the hand-over is completed.

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Page 16-7

Chapter 16 Wireless System

Features
Feature Matrix by Configuration Type
Adjunct Type

Feature
Analog PBX Interface
Announcement - PS No Answer
Announcement - PS Out of Cell
Automatic Release - Out of Cell Calls
Call Forwarding - Not Available
CCIS Interface
Calling Name Display - PS
Calling Number Display - PS
DTMF Signal Sender
Group Call Automatic Conference (6/10 Party)
Group Calling - 2 Way Calling
Handover
Individual PS Calling
Last Number Redial - PS
Multi Line Operation - PS
Multi Site Roaming

UNIVERGE NEAX 2000 IPS Wired for Wireless
Number Sharing
Out of Zone Indication
Overlap Dialing
PS Authorization
PS Location Registration
Preset Dialing
Q. 931a Roaming over IP Trunk
Radio Channel Changeover
Short Message Notification (OAI)
Speech Encryption
Speed Dial - PS
Station Hunting - Not Available
Voice Mail Indication
X = available
– = not available

Page 16-8

Integrated Type

Analog PBX
Interface(1)

CCIS
Interface(2)

X
X
X
X
–
–
–
–
X
–
–
X
X
X
X
–
–
–
X
X
X
X
X
–
X
–
X
X
–
–

–
X
X
X
–
X
X
X
X
–
–
X
X
X
X
X
–
–
X
X
X
X
X
–
X
–
X
X
–
–

UNIVERGE NEAX
2000 IPS
Wired for Wireless
–
X
X
X
X
–
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X
X

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Chapter 16 Wireless System

Feature Descriptions
Feature

Description

Analog PBX
Interface

This feature allows the IPS/WCS to be integrated with many types of
PBXs and to activate various PBS services by sending hook flash
signal and DTMF tone.

Announcement
- PS No Answer
/Announcement
- PS Out of
Zone

This feature allows calls to a Dterm PS III, which cannot be paged, in
a predetermined period of time to be routed to the announcement
notifying the calling party that the Dterm PS III cannot answer.
Announcements can be divided between Announcement - PS No
Answer and Announcement - PS Out of Zone depending on the
Dterm PS III condition.

Automatic
Release - Out
of Zone Calls

When a Dterm PS III user engaged in a call moves out of the service
area or enters a radio zone where all the channels are busy and the
communication becomes unavailable, the call will be disconnected
automatically, and the other party receives reorder tone.

When a PS is power off or out of zone, a call directed to the PS is
Call Forwarding forwarded to a VMS, and a voice mail message can be recorded to
the VMS and checked from the PS. Also the VMS can page the PS
- Not Available
automatically after the voice mail message is recorded.
CCIS Interface

This feature allows the IPS/WCS to be integrated with NEC PBXs
with CCIS interface.

Calling Name
Display – PS

Without answering incoming calls or hold calls terminates to the
Dterm PS III, the calling party’s name can be confirmed by the
indications on the LCD.

Calling Number
Display – PS

This feature provides a display on the LCD of a Dterm PS III
receiving a call, indicating the station number.

DTMF Signal
Sender

This feature allows a Dterm PS III user to send the DTMF signal
(tone) to the called party (terminal, voice mail system, etc.) while
engaged in communication.

Group Call Automatic
Conference
(6/10-Party)

This feature permits a Dterm PS III user, Dterm user or Single Line
Telephone user within the system to establish a conference among
as many as six or ten parties. From a Dterm PS III/Dterm /Single Line
Telephone, a maximum of 9 PSs can be paged simultaneously
except the conference leader. The PSs are assigned to the
simultaneous paging groups as a participant by the system data
beforehand.

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Chapter 16 Wireless System

Group Call - 2
Way Calling

This feature permits a Dterm PS III/Dterm /Single Line Telephone
can page a maximum of fifteen parties simultaneously except the
group call leader. After one of paged parties answers, the paging
becomes the 2 Way Calling between the leader and the first
answered party, and paging other parties stops automatically. The
PSs are assigned to the simultaneous paging group as participants
by the system data beforehand.

Handover

When the signal transmission quality becomes inferior, a Dterm PS III
re-originates a call automatically and seizes another radio channel,
making the WCS handover the call to another zone transceiver to
maintain the speech quality.

Individual PS
Calling

This feature allows the calling party to page the individual Dterm PS
III.

Last Number
Redial – PS

This feature enables a Dterm PS III to store the numbers dialed in the
previous five calls including the last number dialed. The stack dial
numbers are sequentially displayed on the LCD, allowing the station
user to make an out-going call by selecting the desired dialed
number from the display.

Multi-Line
Operation – PS

Dterm PS III equipped with two line keys, L1 key and L2 key, and
different station numbers can be assigned to each of two lines. The
number assigned to L1 key of the PS is called My Line and the other
number assigned to L2 key is called Sub Line. My Line and Sub Line
of a Dterm PS III can be assigned to appear on the Flexible Line
Keys of a Dterm and the Dterm can share the Dterm PS III lines.

Multi-Site
Roaming

Dterm PS III users can originate or receive a call in any place of a
network provided by the multiple PBXs that are interfaced by JTQ931a.

UNIVERGE
NEAX 2000

Wired for
Wireless

IPS

This feature allows the UNIVERGE NEAX 2000 IPS to have the
Wireless PBX feature. UNIVERGE NEAX 2000 IPS provides several
integrated features adding on the adjunctive configuration.
This feature allows the Dterm user to have a Dterm PS III as Sub
Station and to get service with one telephone number. In case that
one user has both Dterm and Dterm PS III, with this feature used, the
user is not required to have separate two telephone numbers.

Number
Sharing

Page 16-10

When user is at his desk, a call is terminated to Dterm.
When user leaves his desk with Dterm PS III, a call is
automatically terminated to Dterm PS III. In this feature, the
Dterm and Dterm PS III are referred as Main Station and Sub
Station, respectively.

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Chapter 16 Wireless System

Out of Zone
Indication

When a Dterm PS III user moves out of the service area and the
electric field strength becomes weak, this feature notifies it to the
user with the warning tone and the LCD display.

Overlap Dialing

This feature allows a Dterm PS III user to receive dial tone and dial
the desired number to originate a call.

PS
Authorization

This feature is to confirm the identity of a Dterm PS III to avoid an
unauthorized Dterm PS III from accessing the system.

PS Location
Registration

This feature allows the WCS to supervise the location of each Dterm
PS III, upon receiving the location registration request, to allow call
termination.

Preset Dialing

This feature allows a Dterm PS III user to confirm the number to be
dialed before originating a call.

Q.931a
Roaming over
IP Trunk

Dterm PS III user can originate or receive a call in any place of a
network provided by the plural PBXs, which are interfaced by JTQ931-a protocol over IP Trunks.

Radio Channel
Changeover

This feature is to supervise and changeover the channel
automatically to avoid the interference and to maintain the speech
quality.

Speech
Encryption

This feature protects a call from being tapped.

Speed Dial – PS

This feature allows a Dterm PS III user to dial the certain frequently
called numbers using two-digit abbreviated call codes.
This feature allows a call placed to a Dterm PS III station, which is
out of zone or power off to be forwarded to an idle station in a hunt
group. Two Station Hunting arrangements are available.

Station Hunting
- Not Available

Voice Mail
Indication

Station Hunting - Circular processes the call no matter which
station in the hunt group is called.
Station Hunting - Terminal initiates a hunt only when the pilot
number of a hunt group is called.
When a message is mailed in the Dterm PS III, an indication of the
envelope icon is displayed in the LCD of the Dterm PS III.

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Chapter 16 Wireless System

System Description
Dterm PS III
The user terminal for the UNIVERGE NEAX 2000 IPS Wireless system is the Dterm PS III. This terminal
is specifically designed for the mobile user, offering portability, durability and functionality.
Portability has two aspects to it. First, the handset must be easily carried by the mobile user. At slightly
more than 3.8 ounces with dimensions of 5” x 1.5” x 7/8” and with a variety of carrying case options,
users can carry the Dterm PS III easily. Second, to be truly portable the handset must also operate
over a long period without having to “return to base” for a recharge. With Talk Time at 6 hours and
Standby Time at 300 hours, the Dterm PS III will serve through most, if not all, of a typical day on the
job.
Durability is another important feature of the Dterm PS III. Once the cord is cut and the user is carrying
the handset with them, it will be dropped, banged into cabinets, and generally suffer a fate not
experienced by business desktop terminals. Drop-tested to 6 feet, designed to endure hostile
environments (temperatures to 122o F), and adaptable to outdoors use with a special case, this
handset can meet the needs of most wireless users.
A wireless handset is no good to the user if it does not deliver the features and functions needed for
effective communication. The Dterm PS III supports a wide range of timesaving, convenient business
features such as:
•
•
•
•
•

Call Hold & Transfer
Directory Dial-by-name (100 entries)
Station Speed Dial (10 with name)
Last Number Redial (20)
Calling Party Number redial (20)

•
•
•
•
•

Automatic Answer (immediate or delayed)
Headset Operation
4 Line by 18 Digit LCD Display (with scrolling)
Modem Data (up to 19.2 kbps)
Vibration Alert

The basic Dterm PS III package includes the Dterm PS III Handset and one Hand Strap. A User Card
explaining basic operation and a comprehensive user guide is also available.

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Chapter 16 Wireless System

Dterm PS III Accessories
To enhance the use of the Dterm PS III, NEC Unified Solutions, Inc. offers several choices in each area
of handset accessories. They are:
Lithium-Ion Battery
(PS III Battery Pack) that provides up to 6 hours of continuous talk time and 300 hours on standby. Additional
batteries may be required for extended use applications that require “around the clock” coverage.
Battery Charger
Battery Charger for the Dterm PS III includes the Dterm PS III charging slot only. Recharging time is a maximum
of 4 hours.
Headsets
Headsets for the Dterm PS III come in two configurations. The first is a classic banded headset with the
microphone extension. The second is an ear bud type with an in-line microphone. The banded headset is
typically used in cases of prolonged active use (e.g., customer service agent) while the ear set is for occasional
use or where concealment is desired (e.g., hotel security).
Dterm Leather Cases
Dterm Leather Cases are available in three models. (1)A Leather Case for carrying the PS III when the large
majority of use is indoors. It utilizes a unique integrated swivel clip that allows the handset to rotate out of the way
(e.g., in case of sitting), and a quick release mechanism to allow for easy retrieval for use. (2)A Leather Case for
carrying the PS III when the large majority of the use is indoors and has an integrated belt clip that doesn’t swivel.
(3)The Weather Case model offers a weather resistant material for users whose job is outdoors a majority of time.

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Chapter 16 Wireless System

Zone Transceiver II (ZT)
Access to the communications network by the Dterm PS III user is gained through radio connection
with one of the several Zone Transceivers installed on the premise. ZT’s are placed strategically
around the establishment in an overlapping fashion to provide continuous radio coverage throughout
the specified area. The number of Zone Transceivers needed to provide service for any single
establishment is defined by several factors:
The total area requiring wireless service, which is a function of the area of operation of the users of the
Dterm PS III handset. Even in the case of providing a small number of handsets, if staff whose job
requires wireless service throughout the whole facility uses them, the number of ZT’s is determined by the
size of the facility, not the number of users.
The configuration, environment and construction of the building(s) in the areas requiring coverage have
considerable impact on the effective coverage area of any one Zone Transceiver. Spacing of ZT’s is
adjusted according to these conditions resulting in considerably different quantities needed to cover equal
amounts of square footage when compared from site to site.
Special conditions, such as coverage inside elevators and stairwells (both are typically not covered). In
addition, accommodating high traffic areas will result in additional Zone Transceivers. Typical
deployments provide access to at least 2 ZT’s from most locations identified for coverage. By placing
ZT’s in closer proximity to each other than normal spacing, access to more ZT’s in any one spot is
supported. In any one zone (an area covered by one ZT, but potentially equipped with more), up to 36
simultaneous calls can be supported within the given 10MHz of the U-PCS band.
Each Zone Transceiver provides for up to three simultaneous connections to Dterm PS III users within the
covered area. As users communicating through a ZT move out of its range and into an area covered by
another ZT, the connection is automatically switched (called a handover) to the new ZT (noticeable by a
momentary disruption in talk path (Note)). With the proper placement of ZT’s throughout the facility,
users can move about the area while communicating (or idle) and have their calls follow them.
The ZT is supplied with mounting hardware and can be installed on walls, mounted on solid ceilings or
placed above drop ceilings. The Zone Transceiver II is also offered with a weather resistant outdoor
enclosure, mountable on outside building walls or poles. Options for both fan cooled or heated
enclosures are also available.
ZT IIU can be deployed using 1 pair or 2 pair wiring.
Note: With ZT’s version 7.2 or higher seamless handover can be achieved.

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Chapter 16 Wireless System

Wireless Specifications
Zone Transceiver II Specifications
ITEM
• ZT II U
Output Power
Type of Interface
Number of Simultaneous Speech Path
Power Supply
Power Consumption
Operating Conditions
Dimensions (unit: mm)
Weight
Wire Diameter
PBX Power
Supply at -48 V

PBX Power
Supply at -45 V

PBX Power
Supply at -43 V

SPECIFICATIONS
6.8 mW
U Interface
Maximum 3
• PBX Power Supply (DC –48 V)
• Local Power Supply (AC 100 V/200 V)
Maximum 3 W
Ambient Temperature: –10 50 °C
Relative Humidity : 45 - 85 % (non-condensing)
Installation Location : Indoor
160 (W) x139 (D) x40 (H) (Excluding Antenna)
Approx. 400 g

26 AWG
2500 ft.
(762 m)

24 AWG
4000 ft.
(1219 m)

22 AWG
5500 ft.
(1676 m)

19AWG
8000 ft.
(2438 m)

4300 ft.
(1310 m)

6900 ft.
(2103 m)

9500 ft.
(2895 m)

13000 ft.
(3958 m)

2000 ft.
(609 m)

3200 ft.
(975 m)

4700 ft.
(1432 m)

7000 ft.
(2131 m)

3500 ft.
(1066 m)
1800 ft.
(548 m)

5600 ft.
(1706 m)
2900 ft.
(883 m)

8200 ft.
(2499 m)
4400 ft.
(1340 m)

12200 ft.
(3718 m)
6400 ft.
(1948 m)

3100 ft.
(944 m)

5000ft.
(1524 m)

7600 ft.
(2316 m)

11000 ft.
(3352 m)

11000 ft.
(3350 m)

13000 ft.
(3958 m)

Local Power Supply

System Capacity
Capacity
Description

Zone Transceiver (ZT)
Dterm PS Note
CSI
4CSIA-A
CSH
PS Simultaneous Connections
Calling Area
ZT per Calling Area

Integrated
1PIM 2PIM
16
32

Adjunct (Analog) Adjunct (CCIS)
3PIM 4PIM 5PIM 6PIM 7PIM 8PIM
48
64
80
96
112
128
512
32
24
216
32
128

Note: All existing CSH cards in a system should be SPN-SC03B 8CSH-C (AP) when accommodating
257 or more PSs.
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Chapter 16 Wireless System

WCS Required Equipment
EQUIPMENT

DESCRIPTION

QTY

PZ-PW122
(DC/DC PWR)

–48V Power supply card for ZT
One per PIM
16 ZT powered/card

1-8

PN-4CSIA-A (CSI)

ZT Interface card
4 per PIM

1-32

PN-SC03-B (CSH)

ZT Handler card
1 per 2 4CSIA cards

1-16

PN-4COT
PN-8COT

COT for Analog Interface
4COT: 4 PS per card
8COT: 8 PS per card

SPN-24CCTA

1.5M AMI (24ch) CCIS Interface Trunk

PN-24DTA-C(24DTI)

24-channel DTI card

1-8

PN-30DTC-C (30DTI)

30-channel DTI card

1-8

PN-SC01 (DCH)

D-channel Handler card

1-8

PN-AP00-B (DBM)

Roaming Data Base Module card

PZ-M606-A (ETHER)

Ethernet Control card

PN-32IPLA (IP-PAD)

32-channel IP-PAD card

1-8

PN-16VCTA (16VCT)

16-channel CODEC card

1-16

IPTRK BUS CA

Set of 3 BUS Cables between IP-PAD
and 16VCT cards
Length: 0.1 m (0.33 ft.)

PN-8IPLA (IP-PAD)

8-channel IP-PAD card

1-8

PN-24IPLA (24DSP)

24-channel expansion DSP card for 8channel IP-PAD card

1-8

REMARKS
Analog Interface
CCIS Interface
Wired for Wireless

4COT: 64
8COT:32
As req.
Max. 8 per sys
For Multi-Site
Roaming Note

For Q931a Roaming
over IP Trunk
(Virtual IPT)

1-8 sets

Note: The SPN-AP00B DBM-C is required when accommodating 257 or more PSs.

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Chapter 16 Wireless System

WCS Required Equipment (Cont.)
EQUIPMENT
• 10BASE-T Cable
(TIA/EIA category
3 or larger)

DESCRIPTION
10BASE-T/100BASE-TX Cable
between ETHER card and Ethernet
Cable Length:
Maximum 100 m (328 ft.)

QTY
As required

REMARKS
For Q931a Roaming
over IP Trunk
(Virtual IPT) Note

• 100BASE-TX
Cable (TIA/EIA
category 5)
Router

As required

PN-AP00-B (DBM)

Roaming Data Base Module card

PN-IPTB (IPT)

IP Trunk card

1-8

PN-4VCTI (4VCT)

4-channel CODEC card

1-32

IPTRK BUS CA

Set of 3 BUS Cables between IPT and
4VCT cards
Length: 0.1 m (0.33 ft.)

1-32 sets

• 10BASE-T Cable
(TIA/EIA category
3 or larger)

10BASE-T/100BASE-TX Cable
between IPT card and Ethernet
Cable Length:
Maximum 100 m (328 ft.)

As required

For Q931a Roaming
over IP Trunk
(Physical IPT)Note

• 100BASE-TX
Cable (TIA/EIA
category 5)
Router

As required

PN-AP00-B (DBM)

Roaming Data Base Module card

PZ-M606-A (ETHER)

Ethernet Control Card

• 10BASE-T Cable
(TIA/EIA category
3 or larger)

10BASE-T/100BASE-TX Cable
between ETHER card and Ethernet
Cable length:
Maximum 100 m (328 ft.)

For Short Message
Notification (OAI)

• 100BASE-TX
Cable (TIA/EIA
category 5)
Note: The SPN-AP00B DBM-C is required when accommodating 257 or more PSs.

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Chapter 16 Wireless System

Wireless LAN
The current Wireless LAN supported interface to the UNIVERGE NEAX 2000 IPS is the
UNIVERGE WL System Telephony Gateway. The UNIVERGE WL System Telephony
Gateway is a wireless telephony product that provides high quality packetized voice
communications using Internet Protocol (IP). This configuration uses a Digital Interface from
the UNIVERGE NEAX 2000 IPS to integrate with the mobile handsets to emulate digital
telephone sets to deliver advanced capabilities such as multiple line appearances and LCD
display features.
Wireless LAN Handset
The MH110 and MH120 Wireless Telephones are mobile handsets for workplace telephone systems.
The Wireless Telephones operates an 802.11b (802.11 for FH) wireless Ethernet LAN providing users
a wireless voice over IP (VoIP) extension. By seamlessly integrating with the Telephone Gateway and
the site PBX, Wireless Telephone users are provided with high-quality mobile voce communication
throughout the workplace. The Wireless Telephones gives users the freedom to roam throughout the
workplace while providing all the features and functionality of a desk phone.
The Wireless Telephones reside on the wireless LAN with other wireless devices using Direct
Sequence (DSSS) or Frequency Hoping (FHSS) Spread Spectrum radio technology. The handset
radio transmits and receives packets at up to 11Mb/s.
The Wireless Telephone supports Wired Equivalent Privacy (WEP) as defined by the 802.11
specification. NEC Unified Solutions, Inc. offers the product with both 40-bit and 128-bit encryption.
WEP increases the security of the wireless LAN to the level similar to a wired Ethernet LAN.

Wireless LAN Documentation
For details on Wireless LAN related hardware and configurations refer to system documentation online
at the following links. Some or all links may require a valid login and password for access.
http://www.ilibrary.com/cng/docs.nsf (See System Documentation)
http://www.necunifiedsolutions.com/main/Products/ProProductCategoryMain.asp?catid=161

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Chapter 17 Hotel/Motel System
Hospitality features are included in the basic system software of the UNIVERGE NEAX 2000
IPS. Additional software is not required. Any Business system can use the Hospitality
features and any Hospitality system can use any Business features as necessary.
The UNIVERGE NEAX 2000 IPS Hotel/Motel features allow the hotel’s administrative staff to
control the telephone activity and check status of guest rooms. Service staff can also
communicate the condition of the guest rooms back to the front desk. By dialing access codes
from the guest room stations, the service staff can report to the front desk area that a room
Needs repair, Repairs are complete, Cleaning in progress, Cleaning complete, Room has been
checked, and other important service functions. By dialing access codes from the guest
rooms, the time spent dialing, waiting for answer, and communicating the message to
someone is eliminated.
The UNIVERGE NEAX 2000 IPS Hospitality package offers the features that will be described in
detail in the following sections.
Hotel Feature List
Feature Name

Guest

Automatic Wake Up
X
Check In/Check Out Note
–
Direct Data Entry
X
Do Not Disturb-Hotel/Motel
X
Do Not Disturb-System
–
Hotel/Motel Attendant Console
–
Hotel/Motel Front Desk
–
Instrument
House Phone
X
Maid Status
X
Message Registration
–
Message Waiting
–
Property Management System
–
(PMS) Interface
Room Cutoff
–
Room Status
–
Single Digit Dialing
X
X: Applicable –: Not applicable

Application
Front Desk
Administrative
Terminal
X
X
–
X
–
–
X
X
–
X
–
–

Hotel
Console
X
–
–
X
X
X

PMS
X
X
X
X
–
–

–

X
X
–
X

X
X
X
X

X
–
–
X

–
X
X
X

–

–
–
X

X
X
X

X
–
–

Note: When MP built-in PMS on IP is provided, you can set and cancel this hotel feature only from PMS.

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Chapter 17 Hotel/Motel System

Application Processor
PMS with AP00 on RS-232C
The Application Processor (AP) manages guest or administration room status and stores call
information on each guest or administrative station. The AP also provides interface ports for a
Property Management System (PMS), Station Message Detail Recording (SMDR) and a Hotel
Printer.
MP built-in PMS over IP
The Main Processor (MP) manages guest or administration room status and stores call
information on each guest or administrative station. The MP also provides a LAN interface port
for a Property Management System (PMS) terminal or a Station Message Detail Recording
(SMDR) terminal. Adding an AP00B MRC-F with PMS over IP allows the print out of an
execution history of Automatic Wake Up, Do not Disturb, Room Cutoff, Check-in/Check-out
service, Message Waiting and PMS connect or disconnect status.

Hotel Console
The DESKCON is programmed as a Hotel Console. The Hotel Console can access Room Cutoff
(individual and group), Automatic Wake Up, Message Waiting, or Do Not Disturb (individual and group)
with the function keys; in addition to the attendant features and functions.

Hotel/Motel Front Desk Terminal
The Dterm can be programmed to function as a Hotel/Motel Front Desk Terminal. This allows setting
and canceling of the following hotel features:
Automatic Wake Up
Check In/Check Out Note
Do Not Disturb
Do Not Disturb-Override
Message Waiting
Room Cutoff
Room Status Note
Note: When MP built-in PMS on IP is provided, you can set and cancel these hotel features only from PMS.

Station Message Detail Recording (SMDR)
The Station Message Detail Recording (SMDR) sends out the outgoing/incoming C.O. call information
to an external SMDR terminal (Personal Computer). The SMDR is usually used in conjunction with the
PMS and used for the following purposes.
Management of guest/administrative station call
o The PMS does not manage the guest/administrative station call
Backup of guest/administrative station call for a PMS failure
Management of either guest or administrative station call
o For example, the SMDR manages an administrative station call, and the PMS
manages a guest station call

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Chapter 17 Hotel/Motel System

Hotel Printer
When PMS with AP00 on RS-232C is provided, the various system messages and the guest room
status can be obtained through a locally provided Hotel Printer. The following information is
automatically printed out as a system message:
Wake Up attempts whether successful or not
Remaining messages for the station which are set to Check Out
Codes and quantities of the goods requested from a guest room by Direct Data Entry
If the print out function key is provided on the Front Desk Terminal, the status of the following features
are printed out when the feature is set or reset and Room Status print out is activated:
Automatic Wake Up
Check In/Check Out
Do Not Disturb
Message Waiting
Room Cutoff
Room Status-individual guest station/all guest stations

Executed Task Print Out
Using the built in PMS on the MP and an AP00B MRC-F will allow the print out of an execution
history for Automatic Wake Up, Do not Disturb, Room Cutoff, Check-in/Check-out service,
Message Waiting and PMS connect or disconnect status.
In the following cases, a history can be output:
When Automatic Wake Up is set/cancelled
When Automatic Wake Up from an administrative station or console is set/canceled
When Automatic Wake Up has been executed
When Do Not Disturb is set/cancelled
When Room Cutoff is set/canceled
When Message Waiting is set/canceled
When a check-in/checkout/checkout cancellation is performed from PMS
When PBX-PMS communication is started/disconnected

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Chapter 17 Hotel/Motel System

Features
Automatic Wake-up
This feature allows the system to be programmed to automatically call guest rooms or administration
stations at specified times. Upon answering, the guest is connected to a recorded announcement or
music source. A printout of Unanswered, Busy or Blocked Automatic Wake-Up attempts for each guest
room is provided using the Hotel/Motel printer.
Setting and canceling can be initiated from the following:
Attendant Console
Hotel/Motel Front Desk Instrument
Guest Station
Administrative Station
Property Management System (PMS) terminal
1. The time is entered on a 24-hour system in one-minute increments.
2. A maximum of 32 stations can be set for the same Automatic Wake-Up time. When the setting
exceeds 32, the excess stations will be set to an earlier time in five-minute intervals.
3. Wake-Up attempts, whether successful or not, can be printed out at a locally provided printer. The
results of execution of Automatic Wake-Up set and cancel are also printed.
4. The Automatic Wake-Up call will ring a station in Do Not Disturb.
5. When Automatic Wake-Up call is received, the station receives music or an announcement. A DAT
or COT card (with locally provided music or announcement source) is required. As an option, a PNDK00 card can be programmed to provide a contact closure for starting the external announcement
or music source when used with a COT card.
6. Data for Automatic Wake-Up programming is canceled by Check Out operation.
7. The number of Automatic Wake-Up call attempts is programmable from one to five times.
8. If the station does not answer, is busy, in Line Lockout, or ringing, recalling is initiated one minute
later. Recalling is repeated up to five times. Each call that fails is printed. When the final attempt
results in failure, a buzzer is sounded at the printer and a flashing LED on the DSS/BLF Console is
available to indicate which station does not answer.
9. Call Forwarding and other features will not reroute Automatic Wake-Up calls.

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Chapter 17 Hotel/Motel System

Check In / Check Out
When this feature is activated, the following operations occur:
Check In:

Room Cutoff is cleared.
Check Out:

Room Status printout is supplied.
Do Not Disturb is reset.
Room Cutoff is set.
Message Waiting is reset.
Automatic Wake Up is cleared.
Multiline Terminals with LCD can be assigned as Hotel/Motel Front Desk Instruments. This feature can
only be activated from the Hotel/Motel Front Desk Instrument or the Property Management System
(PMS). This feature only applies to guest room stations. Refer to the Hotel/Motel Front Desk
Instrument and Property Management System Features and Specifications for more information on
Check In/Check Out.
Note: When MP built-in PMS on IP is provided, you can set and cancel these hotel features only from PMS.

Direct Data Entry
This feature allows a maid or other hotel personnel to enter numeric data to the Property Management
System (PMS), using the guest room station for entry through dial operation. The same numerical data
can be output to a Hotel/Motel Printer by system data programming.
This feature is activated from a guest room telephone.
The digits 0, 1, 2, 3, 4, 5, 6, 7, 8, and 9 can be used for the input data.
The input data can be sent out to the PMS and/or Hotel/Motel Printer. The output option can be
selected by system data programming.
A maximum of 30 digits can be entered in one operation to the PMS and/or Hotel/Motel Printer. If data
exceeds 30 digits, the guest room station receives reorder tone.
If the input data to be sent is less than 30 digits, “#” can be used to end the digit string. (The “#” is not
sent to the PMS and Hotel/Motel Printer.)
Example: Access code + 1234567890#
When the Hotel/Motel Printer is used, “*” can be used to separate the data entry (“*” is not printed out,
but is sent out to the PMS). In this case, the input data including “*” is up to 30 digits.
Example: Access code + 001*002*12345*3#
Reorder tone will be received instead of service set tone if the PMS does not respond within 15
seconds or sends a negative answer to the UNIVERGE NEAX 2000 IPS.

Do Not Disturb Console
This feature allows a DSS/BLF Console associated with a Multiline Terminal to be used as a Do Not
Disturb (DND) Console. This feature is activated by the use of a Function Mode key on a DSS/BLF
Console. DND set status for each station is indicated by a green LED associated with each button. In
addition, the Multiline Terminal user can set/cancel the DND status of other stations using the DND
Console.
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Chapter 17 Hotel/Motel System

Do Not Disturb-Hotel/Motel
This feature allows the Attendant Consoles, Hotel/Motel Front Desk Instruments, guest stations or
Property Management System (PMS) terminals to place individual stations into Do Not Disturb. Calls
can be placed from stations set in DND.
Automatic Wake Up and Timed Reminder will override Do Not Disturb.
A station in Do Not Disturb can be called from the Attendant Console or the Hotel/Motel Front
Desk Instrument using the DNDOVR key.
Do Not Disturb is automatically cleared when Check Out is performed.
Depending on system programming, an incoming call addressed to a station in DND condition is
routed to one of the following on a per tenant basis:
o Reorder tone
o Attendant Console
o A pre-assigned station
Call Forwarding - Busy settings by stations in DND will result in calls being forwarded, even if
the stations are idle.

Do Not Disturb-System
This feature simultaneously restricts incoming calls to a pre-assigned group of stations by operation
from the Hotel/Motel Front Desk Instruments. Attendant Consoles and Hotel/Motel Front Desk
Instruments can use the DND OVR key to override this Do Not Disturb setting.
Stations are assigned to the Do Not Disturb (DND) Group in Class of Service.
Calls to extensions whose stations are in Do Not Disturb will receive reorder tone.
The station in Do Not Disturb can originate calls in the normal manner.
Verification of stations in Do Not Disturb is possible from the Hotel/Motel Front Desk Instrument
and Attendant Consoles.
Only Hotel/Motel Front Desk Instruments and Attendant Consoles have the ability to place a
group of stations in Do Not Disturb. There is only one group available, and the stations within
the group are programmed in system data. There is no limitation on the number of stations in
the group.
A station included in a DND group retains the ability to place that particular station in DND.
When the Hotel/Motel Front Desk Instrument places a group in DND, an individual station within
the group can cancel the DND setting to that station.

Hotel/Motel Attendant Console
The Attendant Console can be programmed to function as a Hotel/Motel Attendant Console. In addition
to the business features and functions of the Attendant, the Hotel/Motel Attendant Console can set
Room Cutoff (individual and group), Automatic Wake Up, Message Waiting, and Do Not Disturb
(individual and group).

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Chapter 17 Hotel/Motel System

Hotel/Motel Front Desk Instrument
A Multiline Terminal with LCD can be programmed to function as a Hotel/Motel (H/M) Front Desk
Instrument. This can be used to set and cancel standard H/M features such as the following:
Do Not Disturb
Room Cutoff
Message Waiting
Print Out
Check In/Check Out
Automatic Wake up
Room Status
Do Not Disturb Override
For each feature desired, a dedicated line key on the Hotel/Motel Front Desk Instrument must be
assigned.
The Room Cutoff and Do Not Disturb features are applicable to a group of stations and individual
stations. There is one group available for each feature and the stations in the groups are programmed
on a per-station basis in system data.
The Print Out function provides a hard copy on the status of the following features when the feature is
set or reset:
Do Not Disturb
Room Cutoff
Message Waiting
Automatic Wake up
Room Status
Check In/Check Out
Successful and unsuccessful Wake Up attempts are printed out.
The Print Out function allows selection of output based on individual station numbers (except for Room
Status).
When Check Out is done, the following functions are set or cleared:
DND – cleared
Message Wait – cleared
Room Cutoff – set
Automatic Wake Up - cleared
Room Status Codes are totally flexible, and the user determines the meaning for each code. The
system will print the maid ID or station number (if other than guest room) that set the specific code. Up
to eight codes (1-8) are available.

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Chapter 17 Hotel/Motel System

House Phone
This feature allows selected stations to reach the Attendant simply by going off-hook.
House Phones may be equipped without dials.
House Phone assignments are programmed into system data from the Maintenance
Administration Terminal (MAT) or the Customer Administration Terminal (CAT).
There is no limit to the number of House Phones permitted in the system provided the maximum
amount of available ports is not exceeded.
The Attendant has the option to process the call using the delay or non-delay operation or
passing dial tone to House Phones with dials.
The system’s response to a hook flash from a House Phone can be assigned to provide feature
dial tone or recall the Attendant.
Four house phone groups are available.
Each house phone group can be assigned to automatically call the Attendant Console or a
predestinated station.

Maid Status
This feature allows the Hotel/Motel (H/M) Front Desk Instrument, Property Management System (PMS)
terminal, or guest room station (using special access code) to register the condition of each guest
room. There are a maximum of eight types of room conditions.
Although the system provides for a maximum of eight types of room conditions, the eight codes (1-8)
provided are not fixed for a given condition. This provides the end user with the flexibility to select any
code (1-8) to serve as the Maid Status code for any condition. When Maid Status is displayed, or
printed out, the digit(s) dialed as the code when the Maid Status was last changed are displayed or
printed.
When dialing from a guest station, a Maid Identification Code (maid ID code) can be provided (up to
two digits). This is allowed or denied in system programming. Status number Room condition (example;
see below)
1. Check In
2. Check Out
3. Under Cleaning
4. Cleaning Finished
5. Check Finished
6. Out of Service
7. Needs Maintenance
8. Maintenance Complete

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Chapter 17 Hotel/Motel System

Message Registration
This feature provides output from the IPS to the Property Management System (PMS). This allows the
PMS system to receive call record reports for guest room phone calls. The PMS system can combine
the phone charges with the room charges, allowing for a presentable final bill to the guest upon check
out.
Message registration is a condensed SMDR record, designed to provide only the necessary call
information to the PMS system.
This feature is a function of the PMS feature and is optional. The standard SMDR output can be used
along with this feature.

Message Waiting
This feature allows the Attendant Console, Hotel/Motel (H/M) Front Desk Instrument, administrative
station, or Property Management System (PMS) terminal to light a lamp (on an uninterrupted or
interrupted basis) on a Single Line Telephone or Multiline Terminal to indicate a message is waiting.
In addition to the lamp indication control, this feature also provides the Voice Message Waiting service
that an originating station user can set the Message Waiting with a prerecorded message by using the
Digital Announcement Trunk card (PN-2DATA)

Voice Message Waiting - System
An originating station user can choose the prerecorded message to be set by dialing the message
number associated. The predetermined station records the messages.

Voice Message Waiting - Individual
When setting Message Waiting, an originating station user announces the message to be recorded
after dialing the station number.

Message Waiting Console
This feature allows a DSS/BLF Console associated with a Multiline Terminal to be used as a Message
Waiting (MW) Console. This feature is activated by the use of a Function Mode key on a DSS/BLF
Console. The Message Waiting status for each station is indicated by a green LED associated with
each button. In addition, the Multiline Terminal user can set/reset MW status using the MW Console.
The MW Console only displays/cancels the set status for those stations to whom Message Wait was
set, by the station user of the MW Console. If the system is initialized (reset), the Console function
mode will return to DSS/BLF.

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Chapter 17 Hotel/Motel System

Property Management System (PMS)
The UNIVERGE NEAX 2000 IPS provides a data interface to a customer-supplied Property
Management System (PMS) accommodating Hotel Management Features. The PMS (Model 60 only)
can be any computer connected to the PBX via a RS-232C Interface. It communicates with the PBX
using the specified protocols.
The Data Link Interface allows the PMS to accommodate both front and back-office Hotel Management
Features, by providing a means of communication between the PMS and the PBX for features such as
Check In/Check Out, Message Waiting, Station Message Detail Data, and control features such as Do
Not Disturb and Room Cutoff.
The PMS can communicate with the PBX to obtain the following information:
1. Maid Status:
This information can be entered from either a guest room telephone or Front Desk Instrument, and
will automatically be transmitted to the PMS for data update.
2. Message Waiting Lamp Status Change
This information can be entered from the Attendant Console or Front Desk Instrument. It is then
automatically transmitted to the PMS for data update. If the automatic MW lamp off feature is
activated, MW data is cleared and status is sent to PMS.
3. Station Message Detail Data
This information is transmitted to the PMS after completion of each local and toll call.
4. Wake-Up Service
This information can be entered from the Attendant Console, Front Desk Instrument or guest room
station, and is automatically transmitted to the PMS for data update.
5. Do Not Disturb/Room Cutoff
This information can be entered from the Attendant Console or Front Desk Instrument, and is
transmitted to the PMS by request from the PMS.
6. Check-In/Check-Out
When PMS with AP00 on RS-232C is provided, this information can be entered from the Attendant
Console or Front Desk Instrument, and is automatically transmitted to the PMS for status update.
7. Room data image messages indicating requests for database updates and database images.
8. Room change, room swap and room copy for data update. Note
9. Room occupancy change and room data change for data update.
10. Routine activity checks between the PMS and the PBX.
Note: Room copy is available only when MP built-in PMS over IP is provided.

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Chapter 17 Hotel/Motel System

The PMS can send the following information to the PBX:
1.
2.
3.
4.
5.
6.
7.
8.
9.
10.

Maid Status
Message Waiting Lamp Status Changes
Telephone Restriction Status Changes
Check-In/Out Messages
Room Data Image Inquiry
Wake-Up Status Changes
Room Change, Room Swap and Room Copy Note
Room Occupancy and Room Data Change
Status inquiry for routine activity checks
Guest Name and Guest Information to be displayed on Administration Station, Front Desk
Instrument or Attendant Console Note

Note: Room copy and Guest Room Information is available only when MP built-in PMS over IP is provided.

Room Cutoff
This feature allows the Attendant Console, Hotel/Motel (H/M) Front Desk Instrument, or Property
Management System (PMS) terminal, or guest room telephones using a special access code, to
temporarily restrict guest room telephones from making unauthorized calls when guests are away from
their room, and when rooms are in Check Out status. There are two types of Room Cutoff conditions
depending on the type of calls restricted.
External Call Restriction: All outgoing calls from guest room stations are restricted in the Room
Cutoff status.
Toll Call Restriction: All toll calls from guest room stations are restricted during Room Cutoff
status. (Internal and local calls are available.)
Stations in Room Cutoff condition are able to place outgoing calls using the Attendant Assisted Calling
feature. If the station under Room Cutoff status dials a C.O. line access code and/or a special area
code, the station is rerouted to one of the following:

Reorder tone
Attendant Console
Room Cutoff is automatically set by Check Out operation and it is automatically reset by Check In
operation. Station-to-station calling and service feature access (such as Maid Status) are still available.

Room Cutoff Console
This feature allows a DSS/BLF Console associated with a Multiline Terminal to be used as a Room
Cutoff Console. This feature is activated by the use of a Function Mode key on a DSS/BLF Console.
The Room Cutoff status for each station is indicated by a green LED associated with each button. In
addition, the Multiline Terminal user can set/cancel Room Cutoff to another station using the Room
Cutoff Console.
The Room Cutoff Console only displays/cancels the set status for those stations that were set into room
cutoff by the station user of the Room Cutoff Console. Indication is not provided when another Room
Cutoff console, the Attendant Console, a Hotel/Motel Front Desk Instrument, or a PMS changes the RC
status of a station.

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Chapter 17 Hotel/Motel System

Room Status
This feature provides the Hotel/Motel (H/M) Front Desk Instrument with a visual display of the guest’s
room status. A supplementary print out (individual and summary) can be provided.
Items indicated are as follows:

Set status of Do Not Disturb
Set status of Message Waiting
Set status of Room Cutoff
Automatic Wake Up Time if set
Check In/Check Out status
Maid Status
The status of the function is indicated by a green LED associated with each function key:
Check In/Check Out set
Do Not Disturb set
Message Waiting set
Room Cutoff set
Automatic Wake Up set (LCD displays time set)
The Room Status of stations can be printed at the Hotel/Motel printer (if available) by pressing the print
(PR) key on the Hotel/Motel Front Desk Instrument prior to pressing the STS key.

Single Digit Dialing
This feature provides the station user the ability to dial single digit codes to access certain features
while still allowing the same digit dialed to be used as the first digit of guest room station numbers.
This feature is available on a numbering plan basis. Up to four different numbering plans are available
per system. For multiple tenant applications, each tenant can be assigned to one of these four
numbering plans.
When this feature is assigned, digits in the numbering plan can overlap and Single Digit Dialing is
based on a time-out after dialing the first digit. The timing duration before the system stops looking for a
second digit is programmable from two-eight seconds. The default setting is four-five seconds.
Digits 0-9, *, and # can be assigned within each numbering plan for Single Digit Dialing.
The following features can be activated using Single Digit Dialing:
Trunk Answer Any Station
Trunk Access
Single Digit Station Numbering
Operator Calls (Dial 0)

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Chapter 17 Hotel/Motel System

Hotel System Capacity
Description

Capacity

Guest/Administrative Station
Front Desk Terminal
Hotel Console
Hotel Printer
I/O port for PMS/SMDR/Hotel Printer
I/O port for SMDR/PMS via LAN
I/O port for Hotel Printer

1020
8
8
2
2
1
2

System Specifications
PMS/SMDR via RS-232C Interface (Model 60 only) and Hotel Printer Interface
Specifications
PMS/SMDR Interface
Hotel Printer Interface

Item
Physical Interface

RS-232C

Synchronization

Asynchronous

Protocol

IMS Procedure

—

Transmission Speed

1200/2400/4800/9600 bps Note

1200/2400/4800 bps

I/O port

No. 0-3 port of AP00-B card

No. 0/3 port of AP00-B card

Note: For the port 1 and port 3 of AP00-B card with AP00 program, data speed cannot be set to 9600 bps.

PMS/SMDR via LAN Interface (Model 90/120 only)
Item

Specifications

Physical layer

Ethernet

Connection layer

Ethernet packet format complies with the DIX standard.

TCP/IP protocol

ARP, IP, ICMP, UDP, TCP

Socket interface

Complies with 4.3 BSD socket interface

Transport protocol

TCP stream type protocol

Application port number

SMDR: 60010 (fixed)
PMS : 60050 (fixed)

Number of connections

Client/Server

Client : SMDR/PMS terminal
Server: PBX

Transmission code

7-bit ASCII code

Quasi-normal restriction condition

1. When connection is closed
2. Status monitoring text

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Chapter 17 Hotel/Motel System

Station Equipment
Front Desk Instrument:

Dterm Series E/Series i with Display
Terminals with 16 feature keys are recommended

Guest Room Phone:

Analog Single Line
H/M Analog Single Line
Dterm Series E Digital 2-Line

Hotel/Motel Attendant Console:

SN716 DESKCON

House Phone:

Analog Single Line
Dterm Series E/Series i Digital Terminal

Administration Phone:

Analog Single Lines
Dterm Series E/Series i Digital Terminal

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Chapter 18 Call Accounting
AIMWorX™
NEC’s Applications Integration Manager or AIMWorX is an application suite of management
software based on a call accounting foundation. It provides an easy way to cost calls with a
variety of features to use this information. AIMWorX has a wide range of predefined
administrative reports and gives you the ability to create your own.
AIMWorX first gathers Station Message Detail Records (SMDR) from the switch. This data
stream is sent each time a call is completed. It contains data about the call including; trunk and
route information, phone numbers of where the call was placed from and where it went to,
what time it started and ended, and variable information that may include authorization codes,
account codes, or office codes. AIMWorX costs each record using costing data that you have
defined to create Call Detail Records (CDR). These records provide the data to run queries
and reports.
AIMWorX supports many Single Point of Entry (SPE) plug-in modules and add-on applications.
Single Point of Entry modules share AIMWorX databases, allowing user information to be
entered once for many different applications. These modules add interfacing capabilities to
remote functions through tabs added to the AIMWorX screens.
Add-on applications are telecommunication packages that have separate user interfaces
but can use AIMWorX already established database. Examples of add-ons are Interactive
Directory, Message Center, and Visual Console.
All optional features, plug-ins, and applications are delivered on a single CD ROM. Included,
as part of the package is a security mechanism (hardware key) that allows the customer to
register the product, and to activate the desired features for the product that they selected.
Also contained on the CD ROM are online help files for each application, and a copy of all
associated application documentation that can be read by the customer using a shareware tool
included for this specific purpose.
AIMWorX is available as software only for use with customer-provided PC, or as a turnkey
hardware system. These turnkey systems are pre-loaded at our customizing center on servers
and desktop PCs from NEC Computer Systems Division (NEC CSD).
AIMWorX integrates with Property Management Systems when the AIMWorX Hospitality Links
module is purchased.

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Chapter 18 Call Accounting

Multi-Site (Centralized Operations)
Multi-Site operation allows AIMWorX Call Accounting to collect data from more than one SMDR source.
Simply put, a site is a PBX or group of PBXs. If you have several separate PBXs (operating
independently, whether on- or off-premises), Multi-Site operation is required to let AIMWorX collect the
call data and process it for each site. Multi-Site collects SMDR via polling, direct connection, or a
combination of both. How AIMWorX handles each PBX is configured in its PBX setup screens.

AIMWorX databases
AIMWorX supports both Microsoft SQL databases. You may purchase client/server version of
AIMWorX with SQL Server 2000 database or without a database. AIMWorX demonstration systems
may use the free version of SQL Server called Microsoft Database Engine (MSDE). This is included on
the AIMWorX CD.
Select Database Licensing Option
Description
Notes
AIMWorX Standalone DB
AIMWorX Program Disk with
Microsoft SQL 2000 Integrated Disk
1 Client Access License
AIMWorX C/S NEC Provided DB AIMWorX Program Disk with
Microsoft SQL 2000 Integrated Disk
5 Client Access Licenses
AIMWorX C/S Cust. Provided DB

AIMWorX Program Disk
No Database - Customer provided

AIMWorX Standalone DB: The AIMWorX Standalone Database includes the AIMWorX Program disk,
which has all modules on it. The core modules that are enabled are AIMWorX Manager (Call
Accounting) and Alarm Manager (included, is the freely licensable Alarm Client). The standalone
version only allows one person to access the AIMWorX application at a time, from the AIMWorX server.
AIMWorX C/S NEC Provided DB: The AIMWorX Client/Server NEC Provided Database includes the
AIMWorX Program disk, which has all modules on it. The core modules that are enabled are AIMWorX
Manager (Call Accounting) and Alarm Manager (included, is the freely licensable Alarm Client).
Processor licensing allows an unlimited number of clients to access the AIMWorX applications
simultaneously from remote PCs connected to the LAN. Access from PCs over the Internet is also
possible, if the proper TCP/IP port is enabled through an organizations firewall. Refer to AIMWorX
documentation for more information on this.
AIMWorX C/S Customer Provided DB: The AIMWorX Client/Server Customer Provided Database
includes the AIMWorX Program disk, which has all modules on it. The core modules that are enabled
are AIMWorX Manager (Call Accounting) and Alarm Manager (included, is the freely licensable Alarm
Client).

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Chapter 18 Call Accounting

SQL Server licensing
If you purchase SQL Server with AIMWorX, the licensing is provided in multiples of five Client Access
Licenses (CALs). All AIMWorX Client/ Server versions include the first five CALs. Purchase additional
CALs in multiples of five.
You may also choose to license the SQL Server database in processor licenses. You purchase one
processor license for each processor in the AIMWorX server and receive unlimited user login name
assignments. Purchase additional processor licenses from your NEC distributor.

Oracle database
If you choose to use an Oracle database, you must purchase it separately. Oracle 7.3.x versions may
be installed on the AIMWorX server computer or on a computer networked to the AIMWorX server. In
either case, all AIMWorX clients must have Oracle Client 7.3.x installed. This is done automatically

during the AIMWorX client installation.
Licensing
AIMWorX modules are sold and licensed on a per customer basis. You purchase an AIMWorX tier, the
AIMWorX modules you need, and the number of seats for each module. Your security key is
programmed with these variables and gives you access to the appropriate parts of the installed
program.

Seat licenses
Each module you purchase comes with five seats or instances of this module. The license server will let
you have five copies of the module active at a time. Starting a sixth copy will cause an access violation
error. Purchase groups of five seats for any module from your AIMWorX distributor.
SQL Server database licenses are sold separately, based on the database manufacturer’s
requirements.

Security
A security program on the server prohibits AIMWorX usage beyond what you have licensed. This
program keeps track of the number clients using AIMWorX, the modules you have licensed, and the
instances of each module. It interacts with your security key and the application on each client to allow
or disallow you access based on the license you purchased and what is currently running.
The security key plugs into the parallel port of the server computer. It is programmed in combination
with your serial registration number, so cannot be copied or used on other installations.

Information you need before installing AIMWorX
Before you install any of the AIMWorX modules, you need to know some information about the phone
service carrier, the switch, the trunks & routes, the organization, and how you will cost the calls.
AIMWorX supports, but is not limited to:
C/O Line
VPS
DID-incoming only
Conference
Tie line
CAS Line
F/X-Foreign Exchange
Data DM
ISDN
Hot Lines
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Chapter 18 Call Accounting

Configurations
Standalone
This allows the user to only access the AIMWorX applications from the server on which they are
installed (Network Printers and email can still be used - Also the freely licensable Alarm Clients can still
be used from Network PCs). This includes the AIMWorX Program Disk (1 CD) and the AIMWorX
Database Disk (1 CD). The AIMWorX Database Disk provides Microsoft SQL 2000 both Personal and
Standard Editions. The Standard Edition is used with Windows 2000 Professional/Server and with
Windows NT Workstation/Server. The Personal Edition is used with Windows 98. It is HIGHLY
Recommended that you install Standalone AIMWorX Systems on a Microsoft 2000 Professional or
Microsoft XP Professional Operating System.

C/S NEC
This is a Client/Server configuration that includes 5 MS SQL 2000 Client Access Licenses. Additional
Client/Access Licenses may be selected below. This includes the AIMWorX Program Disk (1 CD) and
the AIMWorX Database Disk (1 CD). The AIMWorX Database Disk provides Microsoft SQL 2000 both
Personal and Standard Editions. The Standard Edition is used with Windows 2000 Professional/Server
and with Windows NT Workstation/Server.

C/S CUST
This is a Client/Server configuration. It does not include a database. The Database must be provided by
the customer. Currently supported Databases are Microsoft SQL 2000 (SP1 and above) and Oracle
7.3.4. The customer is responsible for purchasing the database of their choice and the database
licenses they desire. This option includes the AIMWorX Program Disk (1 CD).

Call Record and User Capacity
You can purchase AIMWorX in one of six tiers: Basic, Business, Small Corporate, Corporate,
Enterprise, and Large Enterprise. These tiers differ based on the number of user records each
supports. The smaller tiers can be run as standalone systems, while others operate in a client/server
environment only.
Basic tier
The Basic tier manages up to 200 extensions or users. It can be installed as a standalone system or in a
client/server environment and works with the UNIVERGE NEAX 2000 switch family, the UNIVERGE NEAX
2400 switch family, and some non-NEC switches for SMDR collection only. This system supports AIMWorX
Manager, call accounting features, and most AIMWorX features and modules.

Business tier
The Business tier manages up to 500 extensions or users. It can be installed as a standalone system or in a
client/server environment and works with the UNIVERGE NEAX 2000 switch family, the UNIVERGE NEAX
2400 switch family, and some non-NEC switches for SMDR collection only. This system supports all
AIMWorX features and modules.

Small Corporate tier
The Small Corporate tier manages up to 2,500 extensions or users. It can be installed as a standalone
system or in a client/server environment and works with the UNIVERGE NEAX 2000 switch family, the
UNIVERGE NEAX 2400 switch family, and some non-NEC switches for SMDR collection only. This system
supports all AIMWorX features and modules.
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Chapter 18 Call Accounting

Corporate tier
The Corporate tier manages up to 5,000 extensions or users. It can be installed as a standalone system or in
a client/server environment and works with the UNIVERGE NEAX 2000 switch family, the UNIVERGE
NEAX 2400 switch family, and some non-NEC switches for SMDR collection only. This system supports all
AIMWorX features and modules.

Enterprise tier
The Enterprise tier handles up to 65,000 extensions or users. It must be installed in client/server format. It
works with the UNIVERGE NEAX 2000 switch family, the UNIVERGE NEAX 2400 switch family, and some
non-NEC switches for SMDR collection only. This system supports all AIMWorX features and modules.

Large Enterprise tier
The Large Enterprise tier handles up to 125,000 extensions or users. It must be installed in client/server
format. It works with the UNIVERGE NEAX 2000 switch family, the UNIVERGE NEAX 2400 switch family,
and some non-NEC switches for SMDR collection only. This system supports all AIMWorX features and
modules.

The AIMWorX Manager
The AIMWorX Manager is the foundation of AIMWorX. This allows a user to synchronize
databases and to control multiple modules via a true single point of entry. It also includes a
complete call accounting system offering over 20 reports, each of which can be fully
customized and saved for later use. Reports can be scheduled far in advance to print to file (in
a variety of formats) or to paper. Reports printed to file can be viewed over the network via a
small application loaded on the interested party’s computers. Printing times can be scheduled
to suit the specific needs of the company.

Integrating AIMWorX modules
AIMWorX modules add flexibility to your telemanagement system because you only purchase
the modules that you need. Nearly all of them interface with AIMWorX Manager. Some are
accessed directly through a tab on the Users screen, while others have separate interfaces,
but use the call accounting database. AIMWorX modules seamlessly integrate with each other.
To ensure this integration, certain modules should be purchased together. Some modules
should not be paired.
AIMWorX works optimally if these modules are paired:
Cable Manager and Asset Manager
Work Order and Trouble Ticket
Asset Manager and Billing Manager
NEAX SPE and Voice Mail SPE
NEAX SPE and Auth Code Manager
NEAX SPE, 9-1-1 OSN, and Alarm Manager
AIMWorX Manager and Multi-site
Hospitality Links and Voice Mail SPE
Traffic Analysis and NEAX SPE
These modules are mutually exclusive and should not be purchased together:
Interactive Directory 2000 and Interactive Directory 2400
Interactive Directory 2000/2400 and Message Center
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Work Order should have these three modules at a minimum to work:
NEAX SPE
Asset Manager
Billing Manager
Work Order will best enhance your system’s capabilities when combined with these modules:
Trouble Ticket
Auth Code Manger
9-1-1 SPE
Cable Manger
9-1-1 OSN
Voice Mail SPE
Interactive Directory

9-1-1 SPE
9-1-1 SPE is a module that lets you download specific user location information directly to the phone
company’s ALI system. It appears as a tab on the Users screen in AIMWorX Manager and exports the
9-1-1 user database in NENA format. The 9-1-1 SPE is not involved at the time of an emergency; it
supplies user location information to the ALI system, which then supplies it to the authorities when
needed. 9-1-1 On Site Notification is a feature of Alarm Manager that will notify your own security
department with the same information when someone places a 9-1-1 call.

Alarm Manager
The Alarm Manager feature of AIMWorX provides a service that collects and forwards 9-1-1 OSN,
Traffic, Collector, Auth Code and other alarm notices to Alarm Clients. Alarm Clients must be
networked to AIMWorX, but do not need to be AIMWorX clients. Through the Alarm Manager you can
define which clients receive which alarm notices, and what happens if the alarm is not acknowledged.

Asset Manager
The Asset Manager module of AIMWorX is a way to track stock inventory once it becomes an asset.
Asset Manager lets you record all assets and assign them to categories and subcategories. It track
warranties, vendors, and manufacturers for assets and lets you bill them to a user, department, or
customer. This program works in conjunction with the other AIMWorX modules and has extensive
import/export capabilities in order to share data with your current accounting and inventory programs.

Auth Code Manager
Auth Code Manager is in charge of tracking and monitoring authorization codes as well as granting
permissions at the time of a call. Use authorization codes to define limits and access for specific users.
Auth Code Manager interfaces to the AIMWorX database for user information and appears as a tab on
the user screen.

Billing Manager
Billing Manager is a module of AIMWorX that provides a way to charge a customer, department, or
company for calls, equipment, and services. It is not a full-blown accounting system, but a subsystem of
AIMWorX that will export phone related charges in formats acceptable to your organization's current
accounting package. This module works with the AIMWorX Manager, Asset Manager, and Work Order
to provide a billing solution for your telemanagement needs.

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Bill Reconciler
Bill Reconciler compares monthly billing statements from the telephone service carrier to CDR
generated by AIMWorX. You can also save the results from this comparison and create reports using
this cost, your cost, or the billed cost, or the differences between them.

Cable Manager
Cable Manager lets you maintain information on the cable and wire layout of your facility. It lets you
perform powerful end-to-end searches and recommends a cable path, lets you review it, then reserves
it. You can define each down to the levels of rooms and closets in a drag and drop graphic
representation. On other screens, assign switch nodes and ports, frames, blocks, pairs, crossconnects, locations, and include descriptions. Cable Manager complies with EIA/TIA-568 A&B wiring
standards and the EIA/TIA-606 Wire Records Keeping procedures recommended by the International
Standards Body.

Custom SPE Writer
Custom SPE Writer lets you create AIMWorX SPE modules for applications outside of AIMWorX. Use
your own development tools to create an interface using commands that talk to the AIMWorX Arena.
Each application will then appear as a new tab on the Users screen and you can populate the external
database from inside AIMWorX.

Hospitality Links
Hospitality Links combines the power of your Property Management System (PMS), the flexibility of
your UNIVERGE NEAX 2400 IMX/IPX Hospitality switch, and the control of AIMWorX into one solution for
your Hospitality environment. This module lets you manage your administrative call accounting while
formatting and forwarding guest calls to your PMS for billing. The AIMWorX Suite Master application is
a part of Hospitality Links and lets you manage the Flexible Suiting feature of the UNIVERGE NEAX 2400
IMX/IPX Hospitality switch.

Interactive Directory
Interactive Directory is an electronic phone book and more. It provides a directory of user information
including extension, fax number, phone number, emergency number, e-mail address, and twelve
custom fields of information. The directory can be published from AIMWorX Manager or you can enter
user information directly through the Interactive Directory Manager program.

NEAX 2000 SPE
NEAX SPE is a module that lets you manage some of the switch Maintenance and Administration
Terminal (MAT) functions from your AIMWorX computer. This program is a typical example of the
concept of ‘single point of entry’. Once the user information is in the AIMWorX system, you can
program the switch with information to assign phones, extensions, and other administrative duties. This
is all done from a single screen in AIMWorX Manager.

Traffic Analysis
Traffic Analysis intercepts the UNIVERGE NEAX 2000 IPS statistics, such as peg counts, from the PBX's
MAT port and uses them to analyze your telecommunications needs. This module shows your single or
multi-switch system in a graphic layout. It provides a graph of call traffic between PBXs and lets you
zoom in on specific sections to assess usage of specific paths in 30-minute segments.
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Trouble Ticket
Trouble Ticket simplifies the trouble reporting, resolution, and tracking process. It lets you see a
database of telecom problems and their associated solutions with a simple character search. Suggest
simple solutions over the phone or create work orders, on the same screen, for the more involved
problems.

Voice Mail SPE
Voice Mail SPE is an AIMWorX module that lets you program the voice mail system from a tab on the
AIMWorX Manager user’s screen. This interface lets you avoid the time needed and mistakes made by
entering the duplicate information into two systems. The AIMWorX Voice Mail SPE currently supports
the Octel Overture 250, Baypoint/Centigram C70, 120I, 120R, 120S, or 640, Captaris/AVT CallXpress,
and NEAXmail AD-40 and NEAXmail AD-120 voice mail systems.

Work Order
Work Order lets you create work orders that automate commands from all the other AIMWorX modules.
It uses a separate user interface that lets you schedule work orders, manage technician’s schedules,
track progress through completion, and manage inventory and billing.

Additional Reference Material
All additional documentation is included on the CD ROM. Also included on the CD-ROM is a copy of
Acrobat Reader to view the documentation.

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Voice Mail Integration
Voice mail allows callers to leave a recorded message, in their own voice and exact words, for
another individual or group of individuals. With a voice mail application, people can
communicate precisely and personally without having to be on the same telephone line at the
same time. Voice mail applications allow communication to take place without endless games
of telephone tag. Research has shown that only one-third of all business calls achieve direct
contact, yet 40% of all messages are one way messages that do not require dialogue. Voice
mail saves time by allowing you to leave a message immediately. Voice mail bypasses the
problems of time zones, after-switchboard hours, and other obstacles to communication in this
fast-paced business world. Voice mail allows the people in your office to better manage their
communications and time.
Voice mail features can be used by accessing the VMS directly from an extension. If a station
sets its call forwarding destination to the VMS, calls to this station are connected to the VMS,
and the messages can be registered according to the VMS instruction. In addition, the
Message Waiting lamp of the station can be turned on automatically by the VMS.
The following voice mail systems to integrate to the UNIVERGE NEAX 2000 IPS system.
NEAXMail AD-120: The AD-120 is available in 4 to 72-port configurations and offers both
analog and digital station port integration. . The AD-120 may cluster up to 5 72-port systems
supporting a total of 360 ports. The AD-120 also provides a range of capabilities, from voice
messaging to Unified Messaging.
NEAXMail AD-64: The AD-64 is available in 4 to 96-port configurations and is digitally
integrated to the NEAX 2000 IPS PBX. The NEAXMail AD-64 comes as a Level I or Level II
system and offers a “25 free seat license of TeLANophy” along with other optional features.
NEAXMail IM-16 LX: A card type voice processing system with various features and is
mounted into the LT slot of the PBX. The basic system of the NEAXMail IM-16 LX provides 4
or 8 ports of voice mail. Adding an additional expansion card provides 4 or 8 additional ports
yielding a maximum of 16 voice mail ports.

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Service Conditions

1. The system transfers only DTMF signals to the connected VMS. It cannot transfer dial pulses to the
system.
2. Messages can be retrieved from any Multiline Terminal, DTMF telephone, or the Attendant
Console, but not from DP telephones.
3. When the calling party is connected to the Voice Mail System, only DTMF signals can be sent to the
VMS for registering a message. DP telephones cannot be used.
4. Stations can set Call Forwarding - All Calls, Call Forwarding - No Answer, and Call Forwarding –
Busy Line to the VMS. The system sends out a mail box number to the VMS. Calling a station that
has Call Forwarding set to the VMS is automatically answered by the VMS.
5. The DTMF signal pause, Inter-Digit Pause, and DTMF signal width of the station number
automatically sent out to the VMS from the system are as follows:
a. Pause: Variable from 1 second to 12 seconds in 1 second increments
b. Inter-Digit Pause: Variable from 32 milliseconds to 240 milliseconds
c. DTMF signal width: Fixed at 64 milliseconds or 128 milliseconds
6. A special number of up to 4 digits (including an Inter-Digit Pause) can automatically be added, both
before and after, to the station number that is sent to the VMS from the system. This can be used
for a variety of identification codes as required. Two types of Inter-Digit Pauses can be set per
system. One is fixed at 1.5 seconds; the other is programmable from 1.5 seconds to 16 seconds.
7. The VMS can control the Message Waiting Lamp of the Station set by using the Message
Waiting/Message Reminder feature. The retrieval access code for Message Waiting/Message
Reminder is variable and can be set from 1 to 4 digits, in system programming.
8. When all VMS ports are busy (assuming the following condition):
Station-A (or outside party)....Calling Party
Station-B.Called Party
Station-B sets Call Forwarding - All Calls/Busy Line/No Answer to the VMS.
Station-A (or outside party) makes a call to Station-B.
a. Call Forwarding - All Calls :
a. Station A hears reorder tone.
b. Outside party hears busy tone.
b. Call Forwarding - Busy Line
a. Station-A hears busy tone, and can set any busy service to station-B.
b. Outside party hears busy tone.
c. Call Forwarding - No Answer
a. Station-B continues to ring until the VMS becomes idle regardless of whether the
predetermined time for Call Forwarding - No Answer has elapsed. When the VMS
becomes idle, Station-A is connected to the VMS.
d. Direct access to VMS
a. If station A or outside party accesses the VMS directly, the outside party hears busy tone
and station A can set call back to the VMS.
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9. Multiple Call Forwarding: When the final destination for any combination of Multiple Call Forwarding
is the VMS, calls can be transferred to the VMS. The first forwarded station's number is sent to the
VMS. For example, a call is received by Station A, which is forwarded to Station-B, which is
forwarded to Station-C, which is forwarded to VMS. The number of Station-A is sent to the VMS.
10. Ringing Transfer to an Attendant via the VMS: The system allows the VMS to transfer the station or
outside party to the Attendant and releases before the Attendant answers.
11. When the VMS is recalled, by transferring the call to an unanswered station, the system may be
programmed to send the recalling extension number to the VMS.
12. A maximum of 26 digit extension numbers including DTMF signal after the VMS answered, can be
programmed to a One Touch key of a Multiline Terminal.

Message Center Interface (MCI)
This feature provides an interface to the Voice Mail System (VMS), which can send Message
Waiting lamp and other control data to the system. The Message Center Interface (MCI) can
provide the following operations:
1. When terminating the call to the VMS, the system sends call connection status information to the
VMS through the MCI.
2. The VMS sends the Message Waiting Lamp on data to the MCI.
3. The system, upon receiving this control data from the MCI, illuminates the Message Waiting lamp of
the corresponding station.

4. The VMS, upon receiving retrieved message information, will send the Message Waiting lamp
control data requesting the system to extinguish the Message Waiting lamp of the corresponding
station.
5. The MCI interface is a half-duplex, EIA-RS232C, asynchronous serial data link that operates under
a specific message protocol and format. The characteristics of the link, the protocol used, and the
message text are shown below.
ITEM
Data Rate
Operating Mode:
Electrical Interface
Characteristic:
Signal Form:
Interface Distance:

DESCRIPTION
1200, 2400, 4800, 9600 bps asynchronous, software selectable
Half duplex
EIA RS-232C electrical standard interface
EIA RS-404
Max. 15 m (49.2 ft.) between PBX and VMS (without modem)
Note: When modems are used, full duplex asynchronous type modems
are required.

Word Framing:
Parity VRC:
Frame Contents:
Control:
Priority Sequence:

10 bits (1 start, 7 data, 2 stop)
No parity, even parity, odd parity; selected by PBX system data
US ASCII 7-bit codes
Contention
Primary station: PBX
Secondary station: VMS

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MCI Service Conditions

1. The UCD or Station Hunting feature is usually provided with the VMS station.
2. One RS-232C port on the MP card or the Application Processor (AP00) is required to make a data
link with a customer supplied VMS.
3. Messages can be retrieved from any Multiline Terminal, DTMF telephone, or the Attendant
Console, but not from DP telephones.
4. The MCI is available to a direct call or a forwarded call from a station/trunk/Attendant to the VMS.
For details of the connecting patterns, refer to the Installation Procedure Manual/Feature
Programming Manual.
5. Stations can set Call Forwarding or Split Call Forwarding - All Calls, No Answer, and Busy Line to
the VMS. The system sends out incoming call information to the VMS. A call to a station that has
Call Forwarding set to the VMS is automatically answered by the VMS.
6. The MCI can control the LCD display of a Multiline Terminal for “MESSAGE” Indication. The
number of messages is not displayed.
7. When the Message Waiting lamp control is activated with the MCI, the lamp control from the
following equipment will not be provided:
From the Property Management System (PMS)
From the Station (by dialing the access code)
From the Direct Station Selection (DSS) Console
From the Hotel/Motel Attendant Console
From the Hotel/Motel Front Desk Instrument
From the Attendant Console
8. Only one system should be programmed (via system programming) to control Message Waiting
lamps through the CCIS network.
9. The system controls Message Waiting lamps normally when the time interval between messages is
a minimum of 350 msec. or more.
10. When the VMS interface line does not answer, all of the messages are sent out from the I/O port of
the MP/AP00.
11. If the VMS is not ready for information receiving (Busy Status), the AP00 can temporarily store up to
16 call records in its internal memory. If the maximum of 16 call records is stored and a 17th is
generated, the system will write over the oldest stored record.
12. When the RS port on the MP card is used for the data link to the VMS, the MP can store up to 15
call records. If a 16th call record is generated when the MP stores 15 call records, the system will
write over the oldest stored record.
13. The Voice Mail Integration (In band) feature can be combined with voice mail through the MCI in the
system. The Voice Mail Integration (In band) feature and MCI feature can coexist in one system
and either can be selected per VMS (VMS station number) by system programming.
14. When terminating a call with the ANI information to the VMS through the MCI, the system can send
the ANI information to the VMS, if required. This is not available through CCIS interface.
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NEAXMail AD-120
The NEAXMail AD-120 Product offers the following two choices for system configuration:
Voice Mail Only - provides voice mail only services. Voice mail only systems are
shipped with all software preloaded.
Unified Messaging - provides full voice messaging capabilities plus an unlimited seat
license for the View Mail for Outlook (VMO) Unified Messaging client software. Third
Party Fax Integration and Text-to-Speech are available as optional components. Unified
Messaging systems are shipped without software pre-loaded. All software must be
loaded on-site in order to effectively join the customer’s existing Windows network.
The AD-120 offers both analog and digital station port integration.
PBX Integration – All AD-120 systems include the NEAX Digital and Serial MCI switch
integrations. The following components provide integration support for Cisco Call
Manager (Version. 2.4 or above) and SMDI Serial systems. T1 integration is not
currently supported with Dual IP/PBX Support. Enhanced NEAX OAI Integration must
be ordered for each AD-120 in the cluster or whenever Live Record is desired in an
analog port configuration.
System Architecture
Microsoft® Exchange® 2003 Server ships with AD-120 S/W Port Licenses that include
Exchange. Associates may elect to purchase Exchange through NEC or to provide their own
copies.
Microsoft SQL® 2000 Server is the underlying database for AD-120. SQL is provided either in
MSDE or Server Standard 2000 Runtime Edition license versions. MSDE is provided with S/W
Port Licenses from 4 to 28 ports. Port Licenses above 28 ports require the purchase of SQL
Server Standard 2000, Processor License. As a “Processor License,” SQL must be purchased
in quantities equivalent to the number of CPUs in the server. Therefore, a Dell PE2800 with dual
processor’s, requires two licenses. NEC Authorized Associates may elect to provide their own
copies if desired.

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Voice Cards
Voice boards must be purchased for all NEAXMail AD-120 orders.
Note: Systems 4 to 20 ports require a minimum of 2GB RAM, 24 ports to 32 require a minimum of 3GB RAM,
above 32 ports require 4GB RAM. Systems above 48 ports require dual processors. Systems with 32 ports or
higher must also purchase the proper quantity of SQL Standard Server 2000 Runtime Edition.
Description
4 Port - D/42-NE2 Digital Voice Board
4 Port - D/4 PCI Analog Voice Board
4 Port – D/41 JCT-LS Analog Voice Board
8 Port – D/82 JCT-U Digital Voice Board
12 Port - D/120JCT-LSU Analog Voice Board
24 Port – D/240PCI-T1R2 T1 Voice Board

Level II Server Software License with Exchange 2003
A NEAXMail AD-120 Software License includes:
NEAXMail AD-120 Version 2.8 software
SQL MSDE version (4 – 28 ports only)
Exchange 2003 Server (Voicemail Runtime edition)
Designated Voice Port license
Unified messaging seats or Voice mail seats (240 per 4 ports)
System key & documentation
Microsoft Exchange 2003 Server (Voice Mail Run-time edition) satisfies the requirement for Exchange
2003 Client Access Licenses (CALs) for all ‘voice mail only’ users on the system. CALs are still required
for all “unified messaging” subscribers and may be purchased from NEC or provided through a third
party.
NEC Authorized Associates may elect to provide their own server platform and third party software (i.e.,
Microsoft Windows 2003, PCAnywhere, Exchange 2003, and SQL 2000). The NEC Associate assumes
all responsibility for providing a Microsoft certified Windows 2003 server. NEC’s warranty excludes any
Associate supplied computer platform or software used in conjunction with the NEAXMail AD-120
system.
Unified Messaging license and Voice Mail only license provide 240 seat licenses per 4 ports.
Voice boards must be purchased through NEC. AD120 Port License are ordered as specified in the
chart below.
Voice Mail only Software Licenses
AD120 VM Parent, Exchange 2003, 240 Subscribers, per 4 Ports
AD120 VM Parent, No Exchange, 240 Subscribers, per 4 Ports
Unified Messaging Software Licenses
AD120 UM Parent, Exchange 2003, 240 Subscribers, per 4 Ports
AD120 UM Parent, No Exchange, 240 Subscribers, per 4 Ports
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Third Party Software
Description
AD-120 Third-Party S/W (Win2003 + PCAnywhere)
SQL Server Standard 2000 Runtime Edition (per CPU)
If you wish to purchase the Windows 2003 operating system software package for a new system order
from NEC, you MUST also purchase a Level I or II platform.
Windows 2003 is installed on AD-120 voice mail only platforms. If you elect to purchase only the AD120 software license and supply your own platform, you must provide Windows software.
Microsoft SQL Server Standard 2000 Runtime Edition is required for AD-120 configurations of 32-ports
or greater. One copy of SQL Server Standard 2000 Runtime Edition is required for each processor in
the AD-120 platform. For example, a 32-port AD-120 on a Dell Power Edge 2800 (with Dual CPUs)
would require a quantity of two licenses.

Client Access Licenses (CAL)
Available CALs
AD120, Exchange 5.5 CAL (per seat)
AD120, Exchange 2000 CAL (per seat)
AD120, Exchange 2003 CAL (per seat)
Exchange Server 2003 (Voicemail Runtime Edition) included with the AD-120 server licenses (as
designated above) will exempt all voice mail only users from CAL requirements. CALs are still required
for all unified messaging subscribers and may be purchased from NEC or provided from a third party.
If AD-120 Software License is purchased without Exchange Server 2003 (Voicemail Runtime Edition),
Exchange Server 2003 (Standard Edition) is required to run AD-120. Exchange Server 2003 (Standard
Edition) requires CALs for all users on the AD-120 system, whether they are voice mail only or unified
messaging subscribers. CALs maybe purchased from NEC or provided from a third party.

Platform Components
The NEAXMail AD-120 provides separate platform components (i.e., individual AD-120 Platforms,
Software Licenses, Voice Boards and Third Party Software), allowing NEC Associates a greater
flexibility in meeting their customer’s requirements. NEC Associates may choose to order “software &
voice boards only” or a combination of software, voice boards and a Level I or II platform. A complete
AD-120 system would include a Level I or II Platform, AD-120 Server Software License, voice boards
and third party software ordered together.
If a Unified Messaging Software License is ordered with a Level I or II platform, the system will
ship without pre-loading of software. This is because installation of the AD-120 into an existing
network requires the operating system software (and Exchange) to be loaded while connected
to that Local Area Network (LAN).
RAID hard disk drive redundancy (Level II) may only be added at the time of purchase and is
not field-upgradeable. The optional RAID package provides four (4) additional hard drives for a
total of six (6) hard drives in the Level II server and doubles the internal storage capacity.
Rack Mount Kits allow the Level II Dell Servers to be mounted in a standard 19” rack.
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Description
AD120, Level I, Dell GX280
Dell Level II PE2800 Level II Tower
Dell Level II, 6 Drive RAID Array, (PE2800 only)
Option Dell Level II Rack Mount Kit (PE 2800 only)
Secondary CPU (Dell Level II PE2800, Xeon 2.8GHz)
CPU Option (Dell Level II PE2800, Xeon 3.6GHz)
1GB Memory addition, (Dell GX280 only)
1GB Memory addition, (Dell Level II PE2800 only)
2GB Memory addition, (Dell Level II PE2800 only)
17 Inch Color Monitor
External USB Modem

Optional Software Feature Packages
ActiveNet, AMIS, VPIM and Third Party Fax Server Integration options may only be installed by an NEC
Certified NEAXMail AD-120 engineer or an AD-120 technician with the assistance of a qualified MCSE.
Description
ActiveNet for Unity/AD-120 (per 1 node)
AD120, AMIS Analog Networking (per server)
AD120, VPIM Digital Networking (per server)

E-Mail Reader (Text-to-Speech)
The AD-120 offers a Text-to-Speech (TTS) Reader. The TTS read text in English only. TTS is required
for Outlook Calendar Access and is not available on Voice Mail Only systems.
Capacities: Minimum of 2, Maximum of 20 (RealSpeak) simultaneous sessions per system.
RealSpeak Note: 1 to 3 sessions require a total of 2GB RAM, 4 sessions require a total of 3GB RAM, Above 5
sessions require 4GB RAM. Above 3 sessions requires a Level ll platform. Above 8 sessions require Dual Xeon
2.8 GHz processor option.

Description
RealSpeak Text To Speech Reader, 1 Session

Third Party Fax Server Integration
The following fax servers are supported: Omtool Fax Sr. v.3, Interstar Technologies LightningFax v.5.5
or higher, Fenestrae FAXination v.4 or higher, RightFax v.6 or higher, Optus FACSys v.4.5 or higher,
FAXCOM for Exchange version 6.19 or higher and Esker FaxGate v.7 or higher.

PBX Integration Software
All AD-120 systems include the NEAX Digital and Serial MCI switch integrations. The following
components provide integration support for Cisco Call Manager (ver. 2.4 or above) and SMDI Serial
systems. T1 integration is not currently supported with Dual IP/PBX Support. Enhanced NEAX OAI
Integration must be ordered for each AD-120 in the cluster or whenever Live Record is desired in an
analog port configuration. Load Balancing is required when you want to cluster 2 AD120’s and have
them automatically load balance the calls.
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Description
AD120 Dual Integration (IP + PBX)
AD120 SMDI Serial Integration
AD120 Enhanced NEAX OAI Integration
AD120 Load Balancing

Platform Specifications
NEAXMail AD-120 Specifications
Platform Level
Dell Level I
Dell Level II (PE2800)
CPU Type/Speed
Pentium 4 2.8Ghz
Xeon 2.8Ghz
Dual CPU’s
N/A
Optional (2.8 or 3.6Hz)
MB RAM Installed
2GB
2GB
PCI Slots
3 (all 5-Volt)
5 (one 5-Volt &
Four 3.3-Volt)
Chassis Style
Mini-Tower
Tower
Rack-Mount
Not Available
Optional
Maximum Voice Port
16 Analog, 16 Digital
72 Analog or 64Digital
Capacity*
Hours of Storage
Approx. 530
Approx. 1200 or 2400
with RAID
Hard Disk Drive Size
Dual 40GB SATA
Dual 36GB SCSI
RAID Array Drive
N/A
Optional
Size(s)
6 x 36GB
Tape Backup
Optional SCSI
Optional
Expansion Chassis
N/A
Optional (PCI)
Redundant Power
N/A
Standard
Supplies

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NEAXMail AD-64
The NEAXMail AD-64 voice/unified messaging system offers a powerful centralized message
solution to meet the communications needs of your business. You can manage your
communications over the telephone or from your personal computer (PC) using the Microsoft®
Outlook®, Lotus® Notes®, and GroupWise® Inbox. The NEAXMail AD-64 offers advanced
unified messaging, voice mail and automated attendant functionality, helping your organization
become more responsive than ever. Through its suite of unified messaging and desktop call
management applications, NEAXMail AD-64 provides you with one of the most complete
unified communications solution available.
These unified communication applications integrate the NEAXMail AD-64 with your Local Area
Network (LAN) to give you control over live telephone traffic and messages – voice, fax* and email – all from your desktop PC. The ViewMail® for Microsoft Messaging module, for example,
works with Microsoft Exchange® to allow you to access your messages from one familiar
inbox. ViewMail® for Lotus Notes and ViewMail for GroupWise are also offered. In addition, the
ViewCall® Plus module enables you to handle your live telephone traffic from your desktop PC
and also works with the applications you use every day such as your Outlook contact list and
your Access® database.
The NEAXMail AD-64 provides a foundation for future growth and customization. Built upon
the Windows 2003 Server platform, the NEAXMail AD-64 leverages the power, reliability and
scalability of the Microsoft Windows® 2003 operating system. By using the new Windowsbased system administration console, the NEAXMail AD-64 can be administered from any
Windows PC on your LAN. This new console is intuitively designed and is accessible via an IP
connection, making system administration easier and more efficient. The Windows 2003
Server also provides many of the disk maintenance utilities you already know, eliminating the
need to learn new software interfaces.
The NEAXMail AD-64 is available in 4 to 96-port configurations with analog or digital
integration to your NEAX® PBX. The NEAXMail AD-64 may be purchased on either a Level I
or Level II platform. The specifications for these platforms are listed below; however system
specifications are subject to change without prior written notice.

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Level I/Level II Platform Specifications
Level I/Level II Specifications
Platform Level
CPU Type/Speed
Dual CPUs
RAM Installed (max)
PCI Slots (voltage)
Chassis Style
Rack-Mount
Cabinet Color
Max. Voice Port Capacity

Note

Storage Hours
Hard Disk Drive Size
RAID Array Drive
Tape Backup
Redundant Power Supplies
Max. Wattage

Level I (Dell GX280)
Pentium 4 2.8GHz
N/A
512GB (4GB)
3 slots total – all are PCI (5Volt)
Mini-Tower

Level II (Dell PE2800)
Xeon 2.8GHz
N/A
512GB (6GB)
5 slots total - 4 PCI-X
(3.3Volt) and 1 PCI (5Volt)
Tower

Not Available
Black
48 Analog or Digital

40GB SATA

Optional
Black
96 Analog or 64Digital
Approx. 3200 or 6400
w/RAID
36GB SCSI

Approx. 1650
N/A

3 x 36GB

Optional SCSI

Optional

N/A
250 Watts

Standard

Heat dissipation

853 BTU/hr

Chassis Height

16.7 inches
(42.5 cm)
7.1 inches
(18.1 cm)
17.6 inches
(44.7 cm)
28 lbs
(12.7 kg)
50° to 95°F
10° to 35°C
-40° to 65°C
-40° to 149°F
20% to 80% (noncondensing)

Chassis Width
Chassis Depth
Weight
Operating Temperature
Storage Temperature
Relative humidity

730 Watts
3100 BTU/hr
17.5 inches
(44.5 cm)
9.125 inches
(23.0 cm)
24.54 inches
(62.3 cm)
90 lb (40.9 kg), max.
configuration
50° to 95°F
10° to 35°C
-40° to 65°C
-40° to 149°F
20% to 80% (noncondensing)

Note: may require Expansion Chassis

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Additional Hardware Components
Expansion Chassis
Available in a 7-slot PCI form factor, the Expansion Chassis provide additional card slots for voice or
fax cards.
Note: The Expansion Chassis can be added to any NEC-provided platform. NEC Associate assumes all
responsibility of PCI Expansion Chassis compatibility with all other “Associate-provided” hardware. Up to two
expansion chassis may be connected to a Level II (or equivalent) server platform.

ActiveFax Board
The VFX/41JCT-LS fax card provides 4 fax ports on a single PCI Card. ActiveFax software licenses
are not included and are sold separately.
Tape Backup
The tape backup option includes a Tape drive, 1 DAT tape, software, cleaning cartridge and
documentation.
Note: Level I systems require the addition of a SCSI adapter card, which requires 1 open PCI card slot.

UPS (Uninterruptible Power Supply)
In the event of a power failure, the power supply will instantly switch to its internal battery bank. This
keeps the NEAXMail AD-64 in continuous operation and allows the user to work and save valuable
data before the platform shuts down.
Note: NEC strongly recommends the use of a UPS for all NEAXMail AD-64 systems.

Optional Software
Operating System and Utilities Software
The Third Party software includes:
Windows 2003 Server software (Runtime Edition)
PCAnywhere 11.5 or higher.

Notes:
Windows 2003 Server software (Voicemail Runtime Edition) and PCAnywhere may be purchased from
NEC or through a third party provider. If you choose to purchase the Third Party Software from NEC you
MUST also purchase a PC/Server platform or in conjunction with a NEAXMail AD-64 Upgrade order.
If you choose to provide your own Windows 2003 server and PCAnywhere software, it is the Associates
responsibility to install Windows 2003 server software, PCAnywhere and voice messaging software.
Windows 2003 Server (Voicemail Runtime Edition) may only be used to run the NEAXMail AD-64-related
applications on the NEAXMail AD-64 platform.

Optional ActiveFax Software
Maximum ActiveFax Software capacity is 8 in a single system. Fax card(s) (NEC#0161975) are not
included and are sold separately.

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Unified Messaging
A NEAXMail AD-64 Port Software License includes 25 seats of ViewMail/ViewMail for Microsoft
Messaging/ViewMail for Lotus Notes/ViewMail for GroupWise and 25 seats of ViewCall Plus.
Additional seats of these applications may be purchased in 5, 50 or unlimited bundles. The Unlimited
package provides up to 1,250 simultaneous sessions of ViewMail/VMM/VML and ViewCall Plus.
ViewCall Plus requires either TAPI or TSAPI services for enhanced mode options. Please consult your
NEC Sales Engineer with CTI related questions.

Feature Descriptions
Analog Voice Card support offers greater flexibility in the overall NEAXMail AD-64 configuration.
Analog cards allow for a maximum port capacity of 96 ports and allow previous NEC Repartee® OS/2
customers to upgrade to the NEAXMail AD-64’s Windows® 2003 operating system.
Announcement of number and length of messages NEAXMail AD-64 announces how long
messages last; for example, “You have three new messages totaling three minutes, twenty seconds.
Would you like to hear them?”
Archiving The message archive feature allows subscribers to save messages longer than the preset
system parameter.
Automated Speech Recognition (ASR) provides subscribers with the ability to navigate their mailbox
options using speech commands. By speaking simple commands into any telephone, you can navigate
your voice mailbox and change personal options.
Caller ID information is accessible via the telephone. When a user checks a message that has Caller
ID information associated with it, the number will appear on their Dterm's LCD display. Caller ID can
also be retrieved through the soft key menu “CID” or through ASR by saying "Caller ID." This feature
requires the Digital Dterm voice card(s) (i.e., D/42-NE2, D/42JCT-U or D/82JCT-U).
Call Return is a feature that allows a subscriber to callback the sender of a voice message based upon
the Caller ID captured during that message. This feature requires the Digital Dterm voice card(s) (i.e.,
D/42-NE2, D/42JCT-U or D/82JCT-U).
Date and time stamp. At the end of every message, NEAXMail AD-64 indicates how long ago a
message was sent, providing subscribers with accurate information about their messaging activity.
Greetings Each subscriber mailbox can have up to three greetings: standard, busy, or alternate.
Greetings can be rerecorded or switched by the subscriber via their Dterm set, touchtone phone or
through the Mailbox Manager application. If you do not record greetings in your own voice, the system
plays greetings that include your recorded name or your extension.
Standard greeting - plays when your extension is unanswered. A typical standard greeting is: “Hello, this
is Pat Green. I am not at my desk right now. Please leave a message.”
Busy greeting - plays when your extension is busy. A typical busy greeting is: “Hello, this is Pat Green. I
am on the phone now. Please leave a message.” For callers to hear your busy greeting it must activated.
Alternate greeting - plays for special occasions, such as a vacation. A typical alternate greeting is: “Hello,
this is Pat Green. I am out of the office today and will return tomorrow.” When your alternate greeting is
active, the system plays it instead of any of your other greetings.

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Guests A guest is a person who is hosted by a particular subscriber and given a “guest mailbox.”
Guest privileges allow a person to communicate with their host subscriber through voice messages.
Hacker Prevention Features are as follows:
Mandate a default password on each new box so they are not left “unprotected”.
Allow system administrators to change the default security code via the administration console.
Force new subscribers to change the default security code upon first-time enrollment.
Hospitality is an optional special guest messaging service, which offers room guests a shortened
system conversation for ease of use. This package includes 5 guest languages and PMS Integration.
Last in-First out / First In-First out (LIFO/FIFO). System administrator can set LIFO/FIFO for entire
categories of messages (e.g., all messages from other subscribers would be played back in either LIFO
or FIFO order; message playback no longer grouped by subscriber name).
Message order allows users to hear messages in the order most convenient.
Playback order allows the system administrator to set playback order separately for new messages and
old messages for maximum flexibility and convenience.

Live Monitor is a new digital feature that lets a subscriber listen to a caller leaving a message in their
voice mailbox. The subscriber can also pick up the call during the recording to connect. Live monitor is
initiated by pressing a line key on a Dterm® Series (III or higher) phone. This feature requires the Digital
Dterm voice card(s) (i.e., D/42-NE2, D/42JCT-U or D/82JCT-U).
Message editing by both inside and outside callers With message editing, callers can edit all or part
of a message they’ve just recorded—listen to the message, add to it, delete it, re-record it, etc.
Message-return receipts and receipt summaries. Return receipts give subscribers detailed
information about each message he or she sends, including exactly when the recipient opened the
message. For a group message, return receipts tell when everyone in the group opened the message
(for dispatch groups, it identifies the person who heard the entire message and when). Receipt
summaries quickly tell the user whether or not a particular subscriber opened any or all of the
messages sent and whether he or she left the sender any messages.
Message waiting notification The NEAX PBX’s message waiting indicator is activated when a
subscriber has new messages. This may be the extension’s Message Waiting Lamp or Stutter dial
tone.
Multiple subscriber interfaces. Individual subscribers on the same system have a choice of using the
easy 1 for Yes, 2 for No interface or the Menu options interface.
1 for Yes, 2 for No interface. Using the “1 for Yes and 2 for No” option, users access all functions by
responding to simple yes-or-no questions.
Menu options. NEAXMail AD-64 offers a traditional voice mail menu interface, as well as the yes-and-no
interface. Menu options make it easy for experienced users to access all functions quickly.
Quick keys. Yes-and-no interface users can utilize the 4 through 7 keys to jump to specific options or
features.

Outside caller group messaging Outside callers can leave a message for a group of subscribers
instead of a single subscriber. Messages can be left to either a public or private group and will follow
the delivery method assigned to the group.
Personal Mailbox Each subscriber within an organization can have a private voice mailbox. The
NEAXMail AD-64’s internal database provides up to 65,000 voice mailboxes.
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Private Distribution Lists. Subscribers may have an unlimited number of private distribution lists with
an unlimited number of members per list. Only subscribers may send a message to their private lists.
Private Distribution Lists are configured with one of two delivery methods:
Broadcast Delivery. All members of the list hear the message when they listen to their messages
Dispatch Delivery. Only the first list member to listen to a message receives it. The message is then
automatically deleted from all remaining list members’ mailboxes.

Receipt sent for messages saved as new. Subscribers requesting message receipts will receive a
receipt as soon as the recipient of a message listens to any part of the message, even if the recipient
saves the message as new. Once the recipient has opened a message, the sender cannot cancel the
message or change its special delivery options.
Redirect to Multiple Subscribers with one Introduction (RSI). This feature gives users the ability to
redirect messages to numerous users without having to record an individual introduction for each
recipient.
Rewind, pause, and fast forward. While listening to a message, users press “7” to move backward,
“8” to pause, or “9” to skip forward in the message. When recording a message, users can press “8” to
pause.
Special delivery options Users can mark messages with one or more of the following special delivery
options: urgent, private, future, and return receipt requested. Subscribers can also change or add to the
special delivery settings of a message after it’s sent.
Future. The sender can mark the voice message for delivery at a later date and time up to one year later.
Private. The recipient of a “private” message cannot redirect the message to anyone else.
Return receipt requested. The sender of the message is told when the recipient hears the message.
Subscribers can request return receipts for messages sent to individual subscribers or to groups.
Urgent. Messages marked with “urgent” will be heard first, before regular messages. Urgent messages
from other subscribers and guests are played before urgent messages from outside callers. The system
also tells subscribers how many new messages are marked urgent.

Speed control during message playback. Subscribers can slow down or increase the speed of each
message during playback using the telephone keypad.
Subscriber-controlled groups Subscribers can create, name, add, and delete subscriber groups from
their telephone.
Subscriber-controlled message delivery Messages can be delivered to any telephone—home, work
extension, pager, or mobile telephone. Each subscriber can have up to four separate telephone
numbers and dial-out schedules. The message delivery options can be sequential for each telephone
number and programmed by the subscriber. A subscriber can use any touchtone telephone to turn
message delivery on or off, or change the delivery times.
Subscriber language The system manager can set which language each subscriber hears.
Subscribers will hear prompts in that language when they call the system to check voice mail.
(Additional languages can be purchased separately.)
Subscriber self-enrollment Subscribers personalize their voice mailbox— set up their directory listing,
record personal greetings, and set up security codes—through an easy-to-use, touchtone telephone
conversation. Because self-enrollment is so simple, subscribers can begin using NEAXMail AD-64
immediately.

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Directory listing. Subscribers control whether they want to be listed in the automated attendant directory
(see automated attendant features). Subscribers record and spell their name for the directory.
Personal greetings. Subscribers record their individual personal greetings. Callers hear the greeting
before they are given an opportunity to leave a message. Subscribers can change the content of their
greeting from any touchtone telephone or from their desktop using Mailbox Manager at any time to reflect
changes in their schedule.
Personal ID. This number, which is usually based on the subscriber’s extension number, identifies
subscribers to the voice mail system when they are calling to access their mailbox. The NEAXMail AD-64
allows personal IDs of varying lengths - from one to ten digits.
Personal security code. In addition to a personal ID, each subscriber can also set a security code to
safeguard his or her mailbox. The code, which can be any number of digits up to ten, can be changed at
any time from any touchtone telephone or by using their Mailbox Manager from their PC.

Subscriber-to-subscriber messaging Subscribers can dial NEAXMail AD-64 directly to leave a
message for a fellow subscriber without ringing that extension. In this way, co-workers can exchange
routine information and ask questions without disturbing one another. The co-worker can also be dialed
directly and, if unavailable, the call is automatically forwarded to the co-worker’s voice mail with the
caller being identified automatically.
Address messages to multiple subscribers. Subscribers can send a message to many other
subscribers at once without creating a group for them.
Immediate reply. NEAXMail AD-64 allows subscribers to reply to a message from another subscriber
immediately after hearing it without using touch tones.
Message cancellation. If the recipient has not listened to a message yet, the subscriber sending the
message can delete it and record a new message.
Multiple message action. Subscribers can reply to or redirect one message several times. For example,
subscribers can listen to part of a message, interrupt it and reply, then start listening again and forward a
copy of the message to another subscriber.
Message redirection. NEAXMail AD-64 allows subscribers to forward a copy of a message to another
subscriber easily. Subscribers can also record an introduction to the forwarded message.
Multiple replies to one message. Subscribers can respond to old or new messages as many times as
necessary.

Text-To-Speech for Exchange®, GroupWise and Lotus Notes (TTS) TTS supports US English, UK
English, Australian English, or Dutch however, only one language can be loaded on the system. TTS is
supported with all voice board models and has a maximum of 32 TTS sessions.
ViewMail® for Lotus® Notes® (VML) integration is similar to the current ViewMail for Microsoft®
Messaging (VMM) application and gives subscribers PC access to their voice messages from their
Notes Inbox. 25 Seats of VML are included with any new NEAXMail AD-64 version 2.3 S/W Port
License or 2.3 S/W Version Upgrade.
ViewMail® for GroupWise® (VMG) integration is similar to the current ViewMail for Microsoft®
Messaging (VMM) application and gives subscribers PC access to their voice messages from their
GroupWise Inbox. 25 Seats of VMG are included with any new NEAXMail AD-64 version 2.3 S/W Port
License or 2.3 S/W Version Upgrade Voice sampling rate. The voice-sampling rate (24Kbps @ 6KHz)
provides a high standard of voice messaging.
Voice sampling rate, The voice-sampling rate (24Kbps @ 6KHz) provides a high standard of voice
messaging.
Volume control Subscribers can raise or lower the volume of a message by pressing the “5” key
during message playback.

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Automated Attendant
NEAXMail AD-64 acts as an electronic receptionist, answering and routing incoming calls automatically.
Callers hear an opening greeting that gives instructions, information, and options. Using the automated
attendant, callers can reach a subscriber directly by dialing the extension number during the opening
greeting. The attendant can answer several calls at the same time. It can place callers on hold, inform
the caller how many calls are holding ahead of him or her and update this information periodically. It
can screen calls, announce the name of the caller, and wait for confirmation from the subscriber before
putting the call through. The automated attendant can also be set to answer a specific set of numbers
so subscribers and outside callers who know the dedicated number can connect directly to the right
person, and the receptionist does not have to waste time transferring calls.
Answer call on first ring NEAXMail AD-64 can be programmed to answer calls on the first ring so incoming
calls are handled as quickly as possible.
Caller interviewing NEAXMail AD-64 can interview callers with a series of simple questions (up to 20)
through the use of an interview box.
Call screening. NEAXMail AD-64 has the ability to screen calls that are transferred to a subscriber. The
subscriber can decide whether or not to take a call on the basis of caller identity. When using this feature, the
subscriber hears the caller’s name and is asked to press “1” to accept the call or “2” to transfer the call to voice
mail. Subscribers can turn call screening on and off by telephone.
Daylight-saving time schedule NEAXMail AD-64 can be programmed to set the clock forward or backward
to automatically adjust for daylight-saving time.
Directory listing Callers can enter the first three letters of a subscriber’s name (first or last, depending on how
the system is configured) and the system routes the call appropriately. Callers are told the extension number
for future reference. If the extension is not answered or is busy, NEAXMail AD-64 automatically routes the
caller to the subscriber’s voice mailbox.
Holiday schedule Holiday dates may be programmed into NEAXMail AD-64 each year so the system
automatically runs in night mode on holidays.
Multilingual capability The system can support up to nine different language prompt sets. One system can
handle several different languages at once. Language choice can be per port or per user. In addition, callers
can choose a language from a language selection box. US English is the default language for all systems,
unless otherwise specified.

American English-US
Argentinean Spanish-AR
Australian English-AU
Cantonese Chinese-CC
Danish-DK
Dutch-NL

Available Languages
French Canadian-FC
German-DE
Hebrew-HE
Italian-IT
Latin Am. Spanish-LA
Madrid Spanish-ED

Mexican Spanish-ES
New Zealand English-NZ
Portuguese-PT
Swedish-SE
United Kingdom English-UK

Numeric directory assistance NEAXMail AD-64 provides directory assistance for outside callers without
letters on their telephone keypad. It can be set up to guide the outside caller through directory assistance
using numbers.
One-key dialing this feature allows a caller to press a single touchtone digit instead of a series of touch tones
to route their calls. For example, you can route calls quickly to directory assistance, subscriber extensions,
transaction boxes, voice-detect boxes, language-select boxes or interview boxes.

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Opening greeting NEAXMail AD-64 plays a pre-recorded greeting to callers when it answers a call. The
system manager usually records the opening greeting. The greeting typically includes the organization’s name;
how to reach an extension, operator, and directory; how to send a fax, etc. During the greeting, callers can
enter extension numbers, access the directory, or hold for assistance.
Operator assistance Outside callers can press “0” for assistance at any time.
Personal secretary The personal secretary feature enables individual users to have calls transferred to a
specified extension when temporarily away from their desk or out of the office. In this way, important calls are
handled by a knowledgeable person in the same department rather than by one main operator.
Primary or secondary answering NEAXMail AD-64’s automated attendant can be programmed to answer all
incoming calls or calls that are not answered by an operator after a pre-set number of rings. Also, the
NEAXMail AD-64 can answer all or selected incoming lines.
Public Group (Distribution) Lists. All subscribers on the system can send messages to open groups. A
subscriber can be restricted from using open groups. Only subscribers can be members of an open group.
Schedules To accommodate organizations with operating hours that vary from day to day, NEAXMail AD-64
can use up to three different schedules.
Subscriber-controlled call holding. Call holding gives a caller the option to hold for a busy extension to
become available. The system can place callers on hold, inform callers of their holding queue position, and
update this information periodically. Subscribers can turn call holding on and off by telephone. In addition,
certain subscribers can be restricted from being able to turn call holding on and off.
Subscriber-controlled directory listing Individual subscribers can control whether they want to be listed in
the directory (by telephone).
Transfer to attendant. Callers with touchtone telephones can press “0” to reach a live operator at any time.
Callers using rotary telephones can be transferred to the operator when no touch tones are entered.
Voice detect This feature allows outside callers using rotary telephones to access the voice mail system by
speaking, instead of by pressing touch tones. Callers make choices or respond to prompts by saying “yes” or
remaining silent for “no,” rather than pressing touchtone keys on one-key dialing menus.
Voice detect call holding. Voice detect call holding allows callers without touch tones to hold by saying “yes”
instead of pressing “1.”

Audiotext
Many offices and organizations want to provide around-the-clock information to clients. NEAXMail AD64 audiotext features allow an organization to make this information available to callers at all times. The
voice messaging system can offer callers menu trees and messages that can vary depending upon the
telephone number called or the time of day. Callers use touch tones to select an item or subject that
they want to hear. Even multilingual services can be provided. The audiotext recording can be of any
length, and there is no limit to the number of audiotext boxes available with the system.

Mailbox Manager
This application brings many voice mail features to your Microsoft Windows-based desktop PC. The
graphical user interface (GUI) gives subscribers the ability to set up and control personal mailbox
settings in addition to the Telephone User Interface (TUI). With this new tool, subscribers may modify
their greetings, security code, notification methods, groups, conversation preferences, and much more.

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Record/Edit From a PC or a telephone, subscribers can record their directory name and record and edit their
personal greetings, which the NEAXMail AD- 64 plays to incoming callers.
Administer private distribution list (groups) - subscribers oversee their private distribution lists, adding or
deleting recipients as necessary.
Change directory listing status - individual subscribers can control whether or not they want to be listed in
the directory.
Change telephone password - subscribers can change their telephone password from within the Mailbox
Manager.
Change transfer options - subscribers can turn call transfer on or off, and change the extension or telephone
number to which incoming calls are transferred. When call transfer is turned off, incoming calls are sent
directly to the subscriber’s mailbox. This feature is particularly useful for employees who have offices in
multiple facilities, frequently work at home, or work as part of a team that requires them to spend a significant
amount of time in a colleague’s office. By using Mailbox Manager to change call transfer options, callers and
subscribers are connected quickly to provide improved customer service and efficient personal administration
of call traffic.
Set conversation options - subscribers can specify whether or not they want the default Yes/No or the
alternative Menu Mode conversation, whether they want to hear the date and time stamp before or after a
message, or if they want the voice mail system to greet them by name when they call in to retrieve their
messages.
Set message delivery options - subscribers can establish rules governing message delivery options for the
type and urgency of message or frequency of delivery. For example, a subscriber can set the message
delivery options so that when he or she receives urgent voice messages on Wednesday, the NEAXMail AD-64
will call the subscriber’s home or cellular telephone to deliver them.
Set call screening and holding options - subscribers can set call screening and holding options to handle
incoming calls with ease. Two or more of these options may be combined. The following call screening and hold
options are available.
Announce before connecting caller - The system plays a beep before transferring the call.
Screen Name - The system asks the caller’s name before ringing the extension. The system plays “Call
from ” before transferring the call.
Ask me if I want to take the call - The system says “Enter 1 to take the call, or 2 and I’ll take a
message” and then waits for a response from you before transferring the call.
Remember screen name - The system asks and plays the caller’s name as above, and if the caller
leaves a message, will append the caller’s name to the beginning of the message.
Tell me who the call is for The system says, “Call for ” before transferring the call.

Integration features - NEAXMail AD-64 is capable of delivering the following major features. However,
limitations of the NEAX platform, Telco service or network configuration, may make one or more of
these features unavailable.
Call forward to personal greeting. This feature makes it easy for callers to leave messages for
subscribers who are busy or away from their desks. With this feature, incoming calls routed to an
unanswered or busy extension are automatically forwarded directly to a subscriber’s voice mailbox where
the caller can leave a personal message.
Message waiting indication. NEAXMail AD-64 will light a message waiting lamp, activate a display, or
provide a stutter dial tone on the subscriber’s extension when new messages have been received.
Easy message access. With this feature, subscribers simply press one button on their telephone set to
retrieve new messages from their voice mailboxes without entering a personal ID number. To prevent
unauthorized message access, subscribers may also require a security code to be entered before
message playback.
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Live Record. The feature allows the subscriber to record their telephone conversation and save them as
voice messages in their mailbox. Live Record is available on Dterm sets with a properly configured
feature button. A programmable “recording beep tone” may be configured where required by law.
Constant Message Count. Allows Dterm display set users to see messaging information on their LCD.
The message count is updated to provide old, new and urgent message counts when users are
connected to their voice mailbox.
Soft Key Mailbox Navigation. When connected to an IVS2 or IPS system, mailbox subscribers with
Dterm Series E and Dterm IP sets have the option of using the four soft key buttons to manage
messages and navigate their mailbox menus. Using these visual menu prompts allows the subscriber to
quickly select and access various mailbox features.
ANI/Caller ID Capture and Display. When using Enhanced MCI integration Automatic Number
Identification (ANI) or Caller ID (CLID) integration displays the telephone number of the outside caller on
a desktop PC (using ViewMail, ViewMail for Microsoft Messaging or ViewCall Plus) or the Dterm set’s
display during message playback.
Immediate disconnect. NEAXMail AD-64 immediately disconnects when callers hang up, so ports free
up immediately.
Line resource management controls out-dialing. This gives priority to incoming call traffic when
performing out-dialing operations.

Integration methods – the NEAXMail AD-64 integrates with the NEAX PBX system in one of three
ways. The integration method dictates how the NEAX PBX provides call information when forwarding a
call to a subscriber’s personal greeting and how it receives message-waiting indication from NEAXMail
AD-64. When integrated, both the NEAX PBX and the NEAXMail AD-64 system share information on
the origin and destination of calls. It also notifies subscribers when new messages are pending.
Direct Digital Integration – is established using the Intel®/Dialogic® D/42- NE2 voice board(s) installed in
the NEAXMail AD-64. Each board emulates up to four Dterm sets and transmits digital switch information
between the PBX and the NEAXMail AD-64. The Dialogic board connects to the PBX using one Dterm
station port for each NEAXMail AD-64 voice port. The D/42-NE2 board takes a "snapshot" of the feature set
display information and identifies the display information for caller and calling party information. This
integration also makes the Live Record, Constant Message Count and SoftKey Navigation features
possible.
Message Center Integration (MCI) - is a serial data link between the NEAX PBX and NEAXMail AD-64.
The voice messaging system and the switch share information about each call by passing data over this
serial data link. Message waiting lamp status is sent from the voice messaging system to the telephone
system, saving port usage to handle incoming calls. MCI is used in multi-switch networks whenever
Common Channel Interoffice Signaling (CCIS) networking is used.
Simultaneous MCI & Digital Integration – is a combination of the previously mentioned integrations and
is used whenever multiple NEAX PBXs are networked together via CCIS. With this method the local PBX
subscribers may continue to use the Live Record, Constant Message Count, Soft Keys and ANI Capture &
Display features. All other integration is performed through the MCI link.
Single Point of Entry (SPE) AIMWorX® Voice Mail SPE is an optional product that lets subscribers
manage the most commonly used voice mail functions from their computers. This product provides singlepoint-of-entry data management between information in the AIMWorX database and the NEAXMail AD-64
voice mail system. You can assign mailboxes, set preferences, and other administrative duties from the
Users screen in AIMWorX Manager and then download it to the voice mail system.

Maintenance
The NEAXMail AD-64 is designed to be as easy to maintain as possible. All system hardware requires
minimal maintenance. Software maintenance, including backup and restore, is designed to be virtually
foolproof.
System Administration Console By using the Windows-based system administration console, NEAXMail AD64 can be administered from any Windows based PC on your LAN. This new console is intuitively designed and
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Chapter 19 Voice Messaging Systems
is accessible via an IP connection, making system administrators more efficient when managing the NEAXMail
AD-64 system.
Status Monitor The Status Monitor utility monitors the voice messaging system as it answers and routes calls
and it dials phone numbers to deliver messages or turn on message waiting indicators. The Status Monitor
utility displays any error messages for ports. The Status Monitor utility can also be used to reset ports if they
become unable to answer or place calls.
Disk full warning When the number of minutes of message storage left on the system is equal to or less than a
value specified by the system manager, the system asks subscribers to delete unnecessary messages.
On-line reports NEAXMail AD-64 can provide detailed information regarding an organization’s incoming
telephone calls. The information, generated in reports, can be used to help make decisions about the
organization, such as staffing levels, productivity, and telephone equipment needs. Reports can be viewed on
the screen or printed on a printer connected to the voice processing system. Reports can also be copied to files
and imported into many word processing, database, and spreadsheet programs. There are five types of on-line
reports: usage reports, directory reports, busy ports report, call log, and error log.
Usage reports These reports give an indication of how much the voice messaging system is being used over
time. These reports can be run for an individual guest, subscriber, extension number, or system ID. Usage of
the entire voice messaging system can also be tracked.
The Usage Bar Graph Report shows the percentage of each hour that a person or box was using the
voice mail system. The bar graph report for the entire system shows the percentage of each hour that the
voice mail system’s ports were in use. This percentage equals the number of minutes the ports were busy
divided by the number of minutes they could have been busy. The number of minutes they could have
been busy equals 60 minutes multiplied by the number of ports on the system.
The Usage Table Report may be run for an individual guest, subscriber, and extension number ID or
system ID and reports the total number of calls by system port and total duration of calls in minutes. A
usage table for the entire system shows the total number of calls answered by each system port for each
hour of the day and the duration of calls in total number of minutes. The report also includes grand totals
for day, night, and entire 24-hour periods.
Directory reports Directory reports show the structure of the system, its subscribers, system IDs and message
groups. It is possible to run the report for everyone enrolled in the system, for all extension IDs or for message
groups. Directory reports include a subscriber report, extension list and group reports.
Subscriber Reports Subscriber reports list each subscriber and guest enrolled in the system. The report
includes each personal ID, the number of new messages waiting, the total number of new and old messages,
the date the person last called the voice mail system, the number of days the system saves the subscriber’s old
and archived messages, and the access code.
Extension List Reports Extension list reports show every subscriber on the system with the corresponding
extension number ID and the system IDs for any transaction boxes and interview boxes the subscriber owns.
This report does not include guests. For each subscriber, the report lists whether call transfer is currently on or
off, the telephone number that calls will be transferred to, the call transfer type, the number of times the
extension rings, call transfer options, and whether call holding is on or off.
Group Reports Group reports provide information about message groups in the system. There are three
different group reports: List of Message Groups (lists groups), Group Membership (lists members of groups),
and Groups Including Person (lists all the message groups of which a subscriber or guest is a member).
Busy port report The busy ports report shows the total number of times all or selected ports were busy. It also
shows the average percentage of time the ports were busy during a specific time period. The busy port report
can be run for all ports on the system or for a selected group of ports.

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Chapter 19 Voice Messaging Systems

Call log The call log provides a record of every call NEAXMail AD-64 answers, dials, or transfers. The call log
can be run for the entire system or an individual subscriber.
Error log The error log lists system errors. The report lists each error code, the port on which the error
occurred, and the date and time the error occurred. This report can be used to diagnose and solve system
problems. The system manager can look up the error code on line and follow the steps to correct the error.
Remote maintenance With NEAXMail AD-64’s remote maintenance feature, support and problem solving can
be provided without an on-site visit.

Visual Messaging
Visual Messaging is a suite of optional unified messaging and call management modules that can
enhance your NEAXMail AD-64 voice messaging system by linking your telephone and your Local Area
Network (LAN). Using an intuitive Microsoft Windows interface, the Visual Messaging modules bring
calls and messages to your desktop PC and make managing them as easy as clicking a mouse.
Instead of listening to calls and messages one-by-one on the telephone, you can visually manage
multiple calls or messages on screen and work more efficiently and effectively than ever before. As
your organization grows, you can add seats of the different modules. Each module—ViewMail,
ViewMail for Microsoft Messaging, ViewMail for Lotus Notes, ViewMail for GroupWise and ViewCall
Plus addresses a specific need and can be purchased to customize your NEAXMail AD-64 as your
organization’s requirements change.
ViewMail
ViewMail organizes messages on screen, making them much easier to manage than on the telephone.
ViewMail displays a list of messages that you can quickly scan for important information. You can see who
sent the message, the date and time it was sent, the length, and a subject if one was included. When used
with Automatic Number Identification (ANI) and Caller ID, or when entered manually through NEAXMail AD64 call screening feature, the name and number of the person who sent the message is also included. You
can sort messages in any order and see details about them at a glance. Intuitive buttons and icons make
ViewMail easy to use. You can play and review selected messages, send a message to another subscriber,
send a copy of a message to another subscriber, save a message as a WAV file to a desired location and
delete messages from your mailbox—all with a click of a mouse. ViewMail can be used on Windows 98, NT
3.51, NT 4.0, 2000, and XP. Here’s a quick review of some of these features:
•
•
•
•
•
•
•
•

Archiving messages
Call return
Copying or pasting messages
Future message delivery
Message cancel
Message status
Multimedia playback and record
Password controlled access

•
•
•
•
•
•
•

Push button message control
Return receipt
Save as WAV files
Sorting messages
VCR-style playback controls
Visual and audio message notification
Volume control

ViewMail for Microsoft Messaging/Lotus Notes/GroupWise
ViewMail for Microsoft Messaging/Lotus Notes/GroupWise gives you the flexibility to send and receive
different types of messages interchangeably. With ViewMail for Microsoft Messaging/Lotus Notes/GroupWise,
all types of messages—voice, fax*, and e-mail—are collected in your Inbox and made available from a single
application ViewMail for Microsoft Messaging/Lotus Notes/GroupWise has all the easy-to-use features of you
Inbox, including drop down menus, toolbar buttons, context sensitive help, and tool tips.
With ViewMail for Microsoft Messaging/Lotus Notes/GroupWise, you use intuitive VCR-style buttons for
playing messages. You can use the mouse to click on the Rewind, Pause, or Fast Forward buttons to move
quickly through messages. A position slider shows where you are in the message, and lets you move
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Chapter 19 Voice Messaging Systems
immediately to any part of the message. Using the mouse you can redirect a message to any subscriber,
including groups of subscribers. You can also redirect voice over the Internet. ViewMail will save voice
messages as WAV files when you attach them to an e-mail message.
ViewMail for Microsoft Messaging/Lotus Notes/GroupWise can be used on Windows 98, NT 4.0, 2000, and
XP using Outlook 98, 2000, and 2002. ViewMail for Microsoft Messaging/Lotus Notes/GroupWise gives you
some of the same features as the basic ViewMail, including:
• Save as WAV files
• Message status
• Multimedia playback and record
• Password controlled access
• Push-button message control
• Return receipt
• Rules-based message handling
(with MS Exchange Server)

• Sorting messages
• VCR-style playback controls
• Visual and audio message notification
• Volume control
• Sending messages to the Internet
• Archiving messages
• Copying and pasting messages to
other applications

ViewCall Plus
ViewCall Plus gives you total control over all your incoming, outgoing, external, and internal telephone traffic
by managing calls on a PC via a Windows graphical user interface. ViewCall Plus provides live telephone call
management on your PC in the same way that unified messaging enables you to manage voice mail and
manage data about each caller with the applications—such as Microsoft Office—that you use everyday.
With ViewCall Plus, you simply click a mouse to take calls, send calls to a voice mail box, transfer calls to
other extensions, ask callers to hold or hear the caller’s name. You can do all this from the desktop PC
without picking up the telephone which means you can handle several callers at once without interrupting
conversations. Three completely integrated windows let you control every aspect of your telephone traffic.
The Telephone Control window shows the flow of calls to and from your extension and allows you to interact
with callers without interrupting conversations. The Call Log window records all call activity. You can sort or
print the information in any order, and you can utilize the information to return calls or add contacts to your
database by dragging it to those areas with the mouse. The Contact List manages data about each caller
including multiple telephone numbers and action items and provides you with on-screen information stored in
your personal database before connecting with a caller.
For NEAX PBXs enabled with TAPI services (i.e., TAPI Link or PC Telephony Adaptors), ViewCall Plus also
offers an Enhanced mode that adds powerful new features such as the ability to drag and drop dialing
information. To make a call, simply drag the name from the contact list or Call Log to the telephone window
and ViewCall Plus will dial the correct number.
ViewCall Plus includes PhoneBASIC® which allows programmers familiar with Microsoft’s Visual Basic to
customize ViewCall Plus in almost unlimited ways. ViewCall Plus can be configured to access and interact
with databases, work with spreadsheets, direct calls according to a set of pre-established rules, handle
predictive dialing functions and more. For example, ViewCall Plus can make calls from within an application
such as Microsoft Access or Outlook. Here are just a few of the features available with ViewCall Plus:
• Built-in Personal Information Manager (PIM)
• Integrates with more than a dozen popular PIM’s
• Call log
• Call screening
• Call sorting
• Call status
• Conference calling (Enhanced Mode)
• Multiple call handling
• Drag-and-drop between windows
(in Enhanced mode)

• Push-button call management
• Request hold
• Route call
• Sort messages in any order and print
• TAPI/TSAPI telephone support
• Three integrated windows
• Visual Basic scripting
• Caller identification
(when supported by the telephone system)

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Chapter 19 Voice Messaging Systems

NEAXMail IM-16 LX
An enhanced integrated messaging solution; the NEAXMail® IM-16LX is the next generation of
internal card-based, voicemail/unified messaging system. It delivers scalable port and
message storage capacities with all the features you’ve come to expect from the most
advanced NEC voice messaging products. Designed specifically for the NEAX® 2000 IPS, the
internal card set also offers reliability, space and cost-saving advantages.
Up to 490 hours of messages can be stored on the NEAXMail IM-16LX. The integrated
messaging solution scales from 4 to 16 ports and expands from 2 to 4 fax ports, making it an
ideal messaging choice for the NEAX 2000 business communication platform.
The integrated design of the NEAXMail IM-16LX takes up less space than a standalone
messaging solution. It can also save you money. Because the NEAXMail IM-16LX fits inside
the NEAX 2000, you don’t have to buy line cards normally required for a standalone system. It
also uses less energy because it’s powered by the NEAX 2000. In the event of a power failure,
it will operate off the same battery backup of the NEAX 2000, adding another measure of
reliability to your communication system.
Unified Messaging and Call Management
A feature-rich business communications solution, the NEAXMail IM-16LX simplifies message
management with convenient access to voice, fax and email messages from any networked PC.
Unified Messaging yields fewer misunderstood or lost messages and faster message returns. The
result: Employees can prioritize their messages and streamline their workload.
With Voice Messaging, recorded messages can be left and responded to 24x7. Users benefit from
many messaging options and the ability to set up their mailboxes to suit individual needs. From a
multimedia PC, subscribers can even play and record messages, eliminating the need to pick up a
telephone.
Access email messages by phone with the text-to-speech option. Enjoy 24-hour, two-way access to
messages without the need for a laptop or modem connection.
Communicating by fax is faster, easier and totally confidential with NEAXMail IM-16LX. Inbound and
outbound documents are password protected and can be stored electronically until previewed and
printed from any networked desktop PC.

Full Digital Integration
Through its fully customized digital integration, the NEAXMail IM-16LX combines important business
communication functions into one comprehensive voice processing system -- a system that can help
you increase productivity and improve service dramatically.

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Chapter 19 Voice Messaging Systems

Digital integration enables feature options such as Soft key Navigation, Constant Message Count, Live
Record, Live Monitor and Mailbox Manager. Soft key Navigation allows you to navigate your Dterm®
Series E, I or Dterm IP phone, as well as your NEAXMail IM-16LX voice mail menu options using the
LCD screen. Constant Message Count gives you an accurate count of your new or old messages on
the LCD screen.
With Live Record, subscribers can record calls at any point during the call and save the recording or
forward it to other subscribers. Accurate messages can be passed on to others and phone
conversations can be archived for future reference. Setting your mailbox feature options is quick and
efficient using the NEAXMail IM-16LX’s Web-based Mailbox Manager Application. Subscribers can
modify their own mailbox greetings, security codes, notifications, groups, conversation preferences and
other settings through the intuitive, graphical user interface accessed with their Microsoft® Internet
Explorer browser.

Hospitality Feature Package
NEAXMail IM-16LX offers a version specifically designed for the special needs of the hospitality
industry and integrates with over 60 Property Management Systems (PMS).
Personalized guest messaging is available for every room and guests can choose from an extensive list
of foreign language prompts. The guest directory allows callers to contact the hotel guests or leave
messages without going through an operator. The hospitality package’s wake-up feature allows guests
to add, delete, change and confirm a wake-up call at any time during their stay. Guest information lines
answer routine questions 24 hours a day.
The hospitality package also improves internal communication. The entire hotel staff can be alerted to a
message simultaneously. Employees working different schedules can leave messages for each other.
Housekeeping, maintenance and room service can be alerted via pager or cell phone while mobile.

Enjoy the Benefits of Integrated Messaging
For simplified message management, take advantage of the NEAXMail IM-16LX’s numerous features,
convenient access to messages and customizable user-friendly settings. Easy-to-implement, easy-tomaintain and easy-to-install, the NEAXMail IM-16LX is engineered to be a cost-effective and efficient
integrated messaging solution for your NEAX 2000.

Basic Features
Live Record
Live Monitor
Soft Keys Functionality
Web-Based Mailbox Manager
Browser-based Administration Console
ANI Capture and Display
Constant Message Count
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Chapter 19 Voice Messaging Systems

Urgent Message Count
Date and Time Stamp
Single Digit Dialing
Subscriber Self Enrollment
Special Delivery Options (Future, Urgent, Return Receipt)
Speed Keys for single button access to functions
Multiple Personalized Greetings
Message Editing
Message Rewind, Pause and Fast Forward
Message Waiting Lamp Notification
Out dial Notification to Pager
Cascade Notification
Cut-through-paging
Message Volume Control
Subscriber Controlled Groups
Message Cancel and Redirect
Trunk Mapping for specific trunks to be routed to alternate greetings
Extension Remapping
Supervised and Unsupervised Transfer support
Transfer to Attendant
User Changeable Voice Prompts
Reports – Subscriber and System
Port Independence allows each port to have its own set of parameters to answer calls
Menu-mode Option
Alpha and Numeric Directory
Guest Mailboxes
Holiday Schedule
Variable Length Security Codes and Passwords
Caller Interviewing
Call Screening
Call Accept or Reject
Remote Maintenance
AIMWorX and MA4000 Integration support
Fax Detect, Routing and Notification

Additional Features
Visual Messaging and Call Control Modules: ViewMail, ViewMail for Microsoft Messaging,
ViewMail for Lotus Notes, ViewMail for Novell GroupWise, ViewCall Plus, ViewFax.
Hospitality Feature Package (includes 5 Guest Languages and PMS Integration)
Active Fax: Fax Mail
Text-To-Speech Email Reader
Multiple Languages

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Chapter 20 System Documentation
NEC offers a full complement of documents for the UNIVERGE NEAX 2000 IPS product line.
Technical documentation is available on Compact Disk (CD ROM) or on the WEB through
NTAC On-Line (http://www.ilibrary.com/cng/docs.nsf).
This section lists all documents
included on the Compact Disk (CD ROM).

UNIVERGE NEAX 2000 IPS Documentation List
UNIVERGE NEAX 2000 IPS Technical Manuals
Call Center WorX Business System Manual
Maintenance Manual
Call Center WorX MIS Admin Manual
MATWorX Installation Guide
Call Center WorX MIS Installation Manual
NEAXMail IM 16 LX System Manual
Call Center WorX MIS Reports Manual
OAI System Manual
CCIS System Manual
Office Data Programming
Command Manual
Q-SIG System Manual
Configuration Guide
Q-SIG System Manual (PRT)
Data Interface System Manual
Remote PIM System Manual (Digital Remote)
DM Hardware Installation Guide
Retrofit System Guide
DRS Installation and Configuration Guide
SMDR/MCI/PMS Interface Specification
DTR-1 Dterm Seriesi/ITR-2 DtermIP Install Guide
SNMP Implementation
Feature Programming Manual
System Manual
General Description
Upgrade Guide
INASET Installation Guide
WCS System Manual (PCS)
In-Skin Router Installation Guide
WCS System Manual (PCS) for Series 3300 & up
Installation Procedure Manual
ISDN System Manual
ITR-4D-3 DtermIP Installation Guide
UNIVERGE NEAX 2000 IPS Features and Specification
Business Hotel Features and Specifications
ISDN/Q-SIG Features and Specifications
CCIS Features and Specifications
WCS Features and Specifications
CCWX ACD Features and Specifications
UNIVERGE NEAX 2000 IPS User Manuals
Dterm Assistant IPS User Guide
MATWorX User Guide
Dterm Series E Agent Console User Guide
MATWorX Studio User Guide
Dterm Series E Supervisor Console User Guide
Power Patch Panel User Guide
Dterm Series E Telephone User Guide
Request for Proposal Guide
Dterm Series I/DtermIP User Guide
SN716 Desk Console User Guide
INASET for IPS User Guide

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