DESCRIPTION
1551-ASB 501 04 Uen H2

System ASB 501 04

Contents

1	Introduction
2	System structure
2.1	System overview
2.2	Hardware structure
2.3	Principles of the control system
2.4	Software structure
3	Telephony system, ACS
3.1	General
3.2	Traffic capacity
3.3	Numbering
4	Service system, SES
4.1	General
4.2	Characteristics
4.3	Control System
4.4	Switch system
4.5	I/O Subsystem, IOS
4.6	Bus system and device configurations
4.7	System security
5	LIM features
6	Exchange features
7	Extension features
8	Telephone features
8.1	Digital system telephone features
8.2	Cordless telephone features
8.3	IP telephone features
8.4	ISDN telephone features
8.5	Analogue telephone features
9	ISDN extension
9.1	General
9.2	ETSI ISDN S 0
9.3	National ISDN S 0
9.4	ISDN / Datacom (TAU2680) Interworking
10	PABX operator features
10.1	PABX operator console
10.2	The consoles with the symbolic display unit (symbolic/OPI-II consoles)
10.3	OPI 3214 with alphanumerical display
10.4	OPI 3213 with alphanumerical display
10.5	OWS/PC-based operator console
10.6	Features
11	Add on features
11.1	16 kbit switching and multiplexing
11.2	Access Agent
11.3	Account Code
11.4	Agent Call Account Buffer, ACAB
11.5	APNSS
11.6	Automatic Call Distribution, ACD
11.7	Automatic Network Call Distribution, ANCD
11.8	Authorization code for extension
11.9	Call Information Logging, CIL
11.10	CAS extension, EL7
11.11	Choice of Language
11.12	CIL over Ethernet
11.13	CIL buffer on HDU
11.14	CSTA (Computer Supported Telecommunication Applications)
11.15	Data traffic
11.16	Dial by Name
11.17	Digital Line Repeater, DLR-3
11.18	DNIS for ACD
11.19	DSS1 Network side
11.20	Dual access extension interface
11.21	Duplicated control system
11.22	Dynamic Route Allocation, DRA
11.23	Electronic Mail
11.24	Enhanced CAS interface for mobility
11.25	Estimated waiting time announcement for ACD
11.26	FAX III recognition
11.27	Fixed remote extension
11.28	Free seating
11.29	GICI over Ethernet
11.30	Hospitality
11.31	Integrated Voice Mail
11.32	Interception Service
11.33	Inverter
11.34	IP extension
11.35	Least Cost Routing, LCR
11.36	Least Cost Routing, Time of Day, LCR-ToD
11.37	MD Disk Tool
11.38	Mobile extension
11.39	Mobility
11.40	Modem Group
11.41	Music on Hold, MoH
11.42	Name and Number Log
11.43	Name Identity
11.44	Network services
11.45	Operator Instrument for operator with impaired vision
11.46	Original A-number
11.47	Paging
11.48	Personal Number
11.49	Recorded Voice Announcement
11.50	Repeated Individual Diversion, RID
11.51	Short Message Service, SMS
11.52	Signalling converter
11.53	Simplified interception
11.54	SNMP Agent
11.55	Static Semi-Permanent Connection, SSPC
11.56	TAU-D
11.57	Traffic recording
11.58	Voice Compression
11.59	Voice Mail
11.60	Voice System Interface
12	Private network features
12.1	Overview of features
12.2	System features
12.3	User features
13	Data communication
13.1	General
13.2	Traffic types
13.3	Features
13.4	IBM 3270 communications
14	Operational security and reliability
14.1	General
14.2	Duplication of common functions
14.3	Load sharing
14.4	Backup hard disk unit
14.5	Duplicated control system
14.6	Duplicated Group Switch
15	Operation and maintenance
15.1	Maintenance functions in the ASB 501 04
15.2	Passwords and authority classes
15.3	I/O log
15.4	Supervision
15.5	Fault effect limitation
15.6	Fault localization
15.7	Exchange level software program changes
15.8	Board level software program changes
15.9	Configuration of time slots
15.10	Input and output functions
15.11	Alarms and alarm handling
15.12	Key for marking service personnel present
15.13	Alarm interface board (ALU2)
15.14	Traffic recording
15.15	ISDN trunk protocol data
16	Overview of hardware
16.1	General
16.2	Line Interface Module, LIM
16.3	Boards in the LIM
16.4	Group Switch Module, GSM
16.5	Boards in the Group Switch
16.6	Test boards
16.7	Power supply systems
16.8	Main Distribution Frame, MDF
16.9	PABX operator consoles
16.10	Telephones
16.11	Other interworking units
17	Power supply
17.1	Voltage
17.2	Battery and charging
17.3	Distribution system requirements
17.4	Logic circuits power supply
18	Mechanical design
18.1	Mechanical packaging structure
18.2	Mechanical structure of the stacks
18.3	Main Distribution Frame, MDF
18.4	Main Distribution Module, MDM
19	Installation
19.1	General
19.2	Loading of programs
19.3	Programming of exchange data
19.4	Testing
20	Environmental conditions
20.1	General
20.2	Climatic and mechanical environment
20.3	Electromagnetic compatibility (EMC) and protection
20.4	Environmental and ecological properties of ASB 501 04
21	Transmission
22	Line adaptations
22.1	Extension interfaces
22.2	IP extension interface
22.3	Cordless extension interface
22.4	PABX operator interface
22.5	Data extension interfaces
22.6	Modem gateway interface
22.7	ISDN S 0 interfaces
22.8	External line interfaces
23	Tone and ring signals

1   Introduction

The ASB 501 04 system description is divided into three documents:

x = edition, 1 = R1, 2 = R2 etc.
n = application system

This document describes the features and functions applicable for source system
ASB 501 04 R8.

For acronyms, abbreviations and glossary, see description for ASB 501 04, ACRONYMS, ABBREVIATIONS AND GLOSSARY.

The ASB 501 04 is an SPC-PABX (Stored Program Controlled Private Automatic Branch Exchange) employing digital switching and transmission techniques. Characteristic of the ASB 501 04 is a modularity and flexibility that makes it possible, within the framework of the basic system, to meet varying requirements regarding size, function contents, software security, traffic capacity, physical placing using geographically dispersed units, etc.

The ASB 501 04 functions are licensed to the customer with a Right to Use. In addition, the
ASB 501 04 licences are technically protected.

In short, a licence file stored on the ASB 501 04 hard disk unit enables only those Sales Objects ordered, which means the customer can only use purchased Sales Objects. The licence file is created in the Licence Control System.

Each ASB 501 04 site is given an identity code. The licence file is tied to this specific identity code and does not work on any other ASB 501 04 installation. When a system is upgraded to ASB 501 04 R8 or a new system is installed, unlimited number of ports and all functionality can be used during a test period to find out the number of ports and the functionality that are required.

The ASB 501 04 is built up using two superior types of units, namely Line Interface Modules, LIMs, and a Group Switch, GS.

The LIM is a microprocessor controlled unit that can be equipped with any combination of line circuits and other telephony devices. Each LIM has its own control system and switch, and can function as an autonomous PABX or as an integrated part in a larger system. The capacity of a LIM in normal traffic conditions is approximately 20-640 extensions. See section Overview of hardware . To build larger systems, the LIMs are interconnected via a Group Switch and 32 channel PCM (Pulse Code Modulation) lines for traffic and control procedures. Two LIMs can be interconnected directly, whereas the Group Switch is required in larger systems, see Figure 1-1 .

The Group Switch is a modular expandible digital switch whose task is to transmit PCM voice, data and control signals between LIMs. The Group Switch has no control equipment itself, but is fully controlled by the connected LIMs.

The program structure used in the ASB 501 04, i.e. function-related modules in program units, makes the PABX operationally reliable and easy to administer.

The expression Intelligent Private Network is used in the documentation to describe a private network with two or more ASB 501 04 exchanges interconnected via the Analogue/Digital Private Network Signalling System, APNSS/DPNSS, or ISDN (Integrated Services Digital Network) external lines which are capable of sending and receiving standard or proprietary messages.

The fact that the ASB 501 04 is an SPC-PABX allows it to offer numerous features to extensions and PABX operators. Preparations are made in the basic PABX for those features that also require additional hardware as well as extra software.

It is possible to connect ASB 501 04 Digital System Telephones, DECT cordless telephones and IP extensions through a TCP/IP (Transmission Control Protocol/Internet Protocol) network. DTMF (Dual Tone Multi Frequency) keyset (pushbutton) telephones complying with ITU-T (International Telecommunication Union - Telecommunications) recommendations, and rotary dial telephones can also be connected to the PABX in any combination and used concurrently.

The ASB 501 04 can be housed in different types of stacks. One modular stackable floor standing package can contain:

• Line Interface Modules, LIMs (Processor/Switch Modules, PSMs and InterFace Modules, IFMs)
• All Contained Module, ACM
• Group Switch Module, GSM
• PoWer Module with internal battery, PWM
• Power Backup Module, PBM
• Power Distribution Module, PDM
• Optional Applications Module, OAM
• Main Distribution Module, MDM
• Hospitality Server Module, HSM

A stackable module with its contents can also be wall mounted.

The printed board assemblies (hereafter = boards) are placed in the magazines. The wiring between the boards in a magazine takes place principally via the backplane of the magazine. The backplanes are of the printed circuit board type. The wiring between magazines and to the Main Distribution Frame, MDF, always uses plug-ended cables, that are connected to the front of the relevant boards. In certain cases front connection is also used for the wiring of boards in the same magazine.

The exchange can also be a building block in a private network using digital or analogue signalling systems between the nodes, see Figure 1-2 , see Figure 1-3 and see Figure 1-4 for various configurations.

2   System structure

2.1   System overview

2.1.1   Functional structure

The functional structure of the ASB 501 04 forms a hierarchy with six levels, see Figure 2-1 . The system documentation is also related to this structure.

The ASB 501 04 is divided into two systems: the Advanced Communication System, ACS, which contains telephony and features, and the Service System, SES, which contains processor, switch and monitoring functions. In their turn, these two systems are divided into a number of subsystems that contain the main functions of the system, see Figure 2-2 .

Each subsystem contains a number of function blocks. These blocks constitute natural design objects that are built up from program units, boards or both. Analogue and digital extension lines, DECT cordless telephones, PABX operator lines and different types of external lines are thus represented by their own individual function blocks. These include the terminal boards corresponding to the line type as well as a number of program units. Number Analysis and Abbreviated Dialling are examples of function blocks which only contain program units.

The functional modularity considerably simplifies the administration of the system in respect of adaptation, installation and operation and maintenance. The ASB 501 04 thereby meets the demands of a modern PABX in that, for example:

2.2   Hardware structure

The LIM is the unit in the ASB 501 04 to which extension lines, PABX operator lines and external lines are connected, see Figure 2-3 . A LIM can function as an autonomous PABX or as an integrated unit in a larger system. The basic hardware, which can be duplicated, consists of the control system and time switch. In addition, it is possible to equip the LIM with an arbitrary mixture of analogue extension lines, digital extension lines and analogue external lines, digital external lines as well as devices for tone sending, tone reception and conference calls. A Group Switch and an I/O (Input/Output) terminal may also be installed.

Control system

The control system comprises a processor board, LIM Processor Unit, LPU, with internal memory, expandable to 64 Mbyte. Each LPU contains a processor system built around two commercial microprocessors. A 32 bit processor is used as the LIM's main processor while a 16 bit processor is used as the communication processor whose task is to administer the direct communications with the control circuits of the switch and with telephony devices.

Each device board contains a microprocessor, i.e. a device processor, that controls the detection and operational functions on the board and also administers communications with the communication processor. Use of device processors has facilitated the introduction of a standardized interface with the backplane of the magazines.

LIM Switch

The switch is non-blocking with 1024 time slots (multiple positions) and consists of one basic board, LIM Switch Unit, LSU, and up to four distribution boards, Distribution Switch Units, DSUs.

Line devices

Analogue line boards, Extension Line Units, ELU-A, are used to connect conventional telephones for decadic signalling or DTMF signalling, and Analogue Trunk Line Units are used to connect the traditional type of exchange lines and lines to other PABXes. Analogue/digital conversion takes place on the boards through single channel coders. The design of the line circuits and external line circuits is influenced by signal systems and other market requirements. One Analogue Extension Line Unit normally contains sixteen line circuits and one Analogue Trunk Line Unit contains from two to eight external line circuits.

Digital line boards, Extension Line Unit, ELU-D, are used to connect PABX operator consoles or digital telephones via normal 2-wire lines with the aid of a special technique, burst signalling, whereby the information packet is sent alternately to and from the console or telephone. One Digital Extension Line Unit normally contains sixteen digital line circuits. Digital external lines with channel-associated or common channel signalling can be used for connection to the public exchange or for interworking PABXes.

Cordless extension line boards, ELU31, are used to connect Ericsson Radio Base stations (DECT). Up to 8 Base Stations can be connected to one board. The Base station interface uses burst signalling to communicate with the base station. Functions for the synchronization of several line boards in a ring are provided on all boards.

The IP extension line boards, ELU32, are used as network interfaces towards an IP network in order to facilitate voice over IP traffic in the ASB 501 04. Only networks based on 10 Mbit/s Ethernet interfaces can be connected directly to the IP extension line boards and only H.323-compliant terminals can be connected to these TCP/IP networks.

The group switch terminal devices, GJUs, serve as terminals for 30/32 channel PCM lines and each device occupies one board. Each LIM equipped with GJU is normally equipped with two GJU boards for connection to the Group Switch, GS. This board for GS connection, GJU-L, has a design that permits the PCM line to be connected to two parallel group switches for security reason.

The Network Interface Unit board, NIU, is an I/O interface which consists of one board and permits the connection of hard disks or PCs for program and data backup, as well as three V.24 terminals for operation and maintenance purposes, and one Ethernet interface, either 10BaseT or 10Base2. One NIU is always included in new deliveries of the ASB 501 04. More NIUs can be connected to the system, if needed.

Telephony devices

A Tone and Multiparty Unit, TMU, is used for digital generation and reception of tones as well as providing multiparty (Conference) functionality to the LIM. The TMU consists of one board and provides totally 32 inputs (receivers) and 32 outputs (senders).

One Conference can contain a maximum of 8 members.

The allocation of the different functions to the inputs/outputs is flexible, but normally with the following limitations:

One to five TMUs are normally assigned to a LIM, depending on the number of analogue or digital extensions in the LIM.

Group Switch, GS

The Group Switch is expandable from 1 to 8 Group Switch Modules, GSMs, each of which is housed as a separate module in a LIM stack in smaller exchanges and in separate stacks in larger exchanges.

One Group Switch Module can comprise 31 PCM lines and a fully expanded non-duplicated Group Switch comprises 8 Group Switch Modules.

The Group Switch can be duplicated and a fully expanded Group Switch comprises 2 x 8 Group Switch Modules.

2.3   Principles of the control system

The control functions in the ASB 501 04 have been designed such that the LIMs are as autonomous as possible, while at the same time working so that the system functions as one coordinated PABX when seen from the outside. This means that the LIM must not constitute a limit for extension numbering facilities, operation and maintenance facilities, accessibility to external line circuits and other telephony devices, etc. The required control characteristics have been achieved by the dispersal of the software functions in the distributed control system that the LIM processors together form. The possibility of rapid, reliable signal transmission offered by the signal channels of the PCM lines is a basic prerequisite for the interworking of LIM processors.

The software in the ASB 501 04 has been designed in accordance with the following principles:

The following examples show how these general principles have been considered when implementing some familiar functions in the system.

For obvious reasons each LIM is equipped with its own set of program units that administer signalling, supervision of connections etc., for the connected telephony devices.

Number Analysis

Number Analysis is a function to which all LIMs frequently require access. The data volume is moderate and independent of the system size. This function is placed in each LIM in accordance with the principles 2, 3 and 4, and without seriously disregarding principle 5.

Directory Number Translation

The translation of directory numbers into line numbers is a function whose data volume grows as the number of extensions grows. If each LIM were to have a complete set of data for translation of all directory numbers in the system, the data volume would grow very large in bigger PABXes. The function has therefore been divided into a central part and a regional part.

The regional part exists in each LIM and contains a complete set of data for all extensions connected to the LIM.

The central part exists identically in all LIMs in the system, but contains only information specifying to which LIM each directory number in the system belongs.

Thus the regional parts in the system's existing LIMs are not affected by the addition of more LIMs to the system. However, the data volume of the central part does grow. Due to the small amount of data on each directory number contained in the central part, it only grows moderately as the system grows.

Other examples of functions that, to a greater or less degree, have been designed as common functions are:

• selection of outgoing external lines.
• queue to PABX operators.
• memory loading.
• man-machine communication.

2.4   Software structure

The software is divided into regional and central units.

As the characteristics of the system are such that each LIM can function as an individual unit irrespective of other LIMs, each LIM is furnished with all the software needed for its internal function. This software constitutes the regional software.

For certain inter-LIM and general system functions central software is used. A central function is placed in one LIM in the system. The system contains several central functions. For a multi-LIM system these are distributed among several LIMs.

A program unit, central or regional, is divided into a program part and a data part. The data part can only be managed from the program part of the program unit.

All interworking between program units, within LIMs and between LIMs, is achieved using formal messages, i.e. program signals . The signal channels of the PCM lines are used to send program signals between program units in different LIMs.

A program signal contains the addressee and the data. Both the sending and receiving program units know the arrangement of the data fields in the program signal. Thus, only the variable values (and not references to variables) are conveyed in the program signal. In this manner the program signals become the most important medium for defining the interface between different program units. Each program unit is encapsulated and has its own data and program code. Writing outside its own program unit is prevented by the hardware.

A program unit is subordinate to a function block which can contain several program units. The system contains numerous function blocks.

By confining the implementation of a function to one function block, the block can be managed irrespective of other function blocks in the system. One function block can contain several functions.

The great majority of the programs are written in PLEX-M, which is a high-level language developed for real time applications and to support the defined principles for unit division and signalling. There are also a few programs written in either C++ or C.

3   Telephony system, ACS

3.1   General

The telephony and data communication parts of the ASB 501 04 are implemented in the Advanced Communication System, ACS, which contains all requisite functions for the establishment of voice and data connections between extensions connected in the same LIM or different LIMs, and also between extensions and public network subscribers.

Within the ACS the functions have been split with the intention of solving the traffic functions in a general manner, irrespective of the signal system on the line from which the facility is called. Consequently the ACS is divided into signal system functions and traffic functions. This division of functions is reflected in the seven subsystems:

The implementation of products can consist of both hardware and software. The hardware consists of extension line circuits, external line circuits, tone senders, tone receivers, Conference equipment, etc.

3.2   Traffic capacity

Due to the modular structure of the system and the flexible equipment configuration it is possible to vary the traffic capacity extensively. The internal LIM switch and the Group Switch are non-blocking. The traffic capacity is therefore mainly defined by the number of external lines and PCM lines between the LIM and the Group Switch. Up to 256 external lines can be connected to each LIM and the LIMs can be connected to the Group Switch via up to eight 32-channel PCM lines. However, all 256 external lines cannot use the same signalling system.

3.3   Numbering

The ASB 501 04 offers a flexible numbering system. This means that the available number series, 0-99 999, can be assigned in an arbitrary manner and, furthermore, directory numbers are not associated to permanent multiple positions in the switch.

Call and answer number codes are defined by programming from the I/O terminal.

Permitted number length

Service codes are the numbers used for procedures. Also rotary dial telephones (decadic pulsing) can use these procedures.

The following procedures are applicable with decadic pulsing:

4     Service system, SES

4.1   General

The Service System, SES, has been designed to facilitate its adaptation to different communication systems.

The service system contains all the requisite basic equipment for the operation of a system such as the Advanced Communication System, ACS:

The software in the SES is composed of:

4.2   Characteristics

The Service System has a modular design which provides the communication system with the basic hardware, expandable in stages in accordance with the following, see Figure 4-1 :

4.3   Control System

4.3.1   Single system

A basic characteristic of the ASB 501 04 is that each LIM can function as an autonomous unit or as a single PABX. The requirement for autonomy in every LIM has resulted in each LIM possessing its own control system. LIMs that are interworked via PCM lines and the Group Switch are one system. This requires information transmission between LIMs to an extent which in other PABX systems only occurs within a processor. This has been achieved via the Group Switch which has a high inter-LIM signalling capacity.

The inter-LIM signalling speed is 64 kbit/s.

The control system comprises one LIM Processor Unit, LPU, with a 64 Mbyte internal memory. The LPU is based on two commercial microprocessors. One processor functions as the LIM's main processor (32-bit) while the other works as the communication processor (16-bit), whose task is to administer the direct communications with the control circuits of the switch and with the telephony devices.

Figure 4-2 shows the hardware organization of the control system and the connection to the communication processor that extends signals to and from the devices.

4.3.2   Duplicated system

The LIM control system can be duplicated. This means that, in addition to the duplication of processor boards, the LIM Switch boards are also duplicated.

Inter-control system communication is achieved by means of an internal signal channel via which information about the status of the control system is exchanged.

Figure 4-2 shows the interworking with other control system parts in a duplicated control system.

For more information, see  14 Operational security and reliability .

4.4     Switch system

4.4.1   General

The function of the switch system is to provide data and speech connections, signalling paths, and synchronization between the LIMs in the ASB 501 04.

The switch network in the ASB 501 04 consists of LIM Switches, PCM lines and a Group Switch. Both the LIM Switch and the Group Switch are digital, single stage time switches.

The LIM Switch is used:

• to connect time slots between devices both within the LIM and towards channels on the PCM line.
• to distribute signals between the operating system and devices within the LIM.
• to keep the LIM synchronized with the rest of the system.

The PCM lines are used to transfer data, speech and inter LIM signals between the Group Switch and the LIM, or between LIM Switches. It also carries synchronization information.

The Group Switch supplies data and speech paths, and conveys program signals between the LIMs. The Group Switch also provides synchronization.

4.4.2   LIM switch

The switch is non-blocking and consists of one LIM Switch Unit, LSU, and four Distribution Switch Units, DSUs, for a 1024 x 1024 time slot switch. It contains the speech and control memories for the time switch, and a microprocessor that controls the internal functions of the switch. It maintains contact with the communication processor.

The LIM Switch Unit, LSU, is also equipped with a clock unit that has hold-over capabilities. Hold-over means that the external synchronization frequency can be memorized, and synchronization maintained even if the external synchronization is lost for a short while. The LSU/DSU take synchronization from one selected device board.

Each of the Distribution Switch Units, DSUs, serves 256 time slots and performs serial/parallel conversion of the PCM signals to and from the device boards.

Internally the LIM Switch consists of a control memory and a speech memory. The control memory stores the connection information while the speech memory stores speech samples.

The LIM Switch also has the facility to attenuate or amplify the sample that is switched through. Sixteen alternative levels are accessible in the LSU/DSU for A-law coding.

-6, -3, 0, +3, +6, +9, +12, +15 dB are predefined, where + means attenuation, and - means amplification. The rest may be customer defined to meet specific requirements. There are also defined levels for mu-law coding.

4.4.3   PCM line

The LIMs are connected to each other in a two LIM system or to a Group Switch in a multi LIM system via 30/32 channel PCM lines.

Time slot channels T1-T15 and T17-T31 are used for voice and data, T16 for control signals and T0 for synchronization information.

IDNX (Integrated Digital Network Exchange) capabilities on PCM lines between LIMs and LIM-GS

The IDNX implements the ability to direct traffic, depending on the originator, to a suitable time slot on the PCM line. A time slot can be dedicated to any combination of the following traffic types: voice, data, buffer transfer, loading and Call Information Logging.

Voice but not data may be compressed. Data must be transferred in time slots assigned to be left uncompressed. Voice could be transferred on compressed and uncompressed channels. Thus, some channels must be used for voice only while others could share voice and data.

4.4.4   Group Switch

The task of the Group Switch is to switch voice, data and control signals between the LIMs. It also has functions to maintain system synchronization. The LIMs are connected to a Group Switch via 30/32 channel PCM lines.

The Group Switch is expandible from 1 to 8 Group Switch Modules, GSMs. Each module is housed in a separate magazine and can comprise up to 31 PCM lines. A fully expanded, non-duplicated Group Switch comprises two stacks.

The Group Switch synchronization function is situated on the Group Switch Clock Unit, GCU. The clock unit has hold-over capabilities which means that the external synchronization frequency can be memorized, and synchronization maintained even if the external synchronization is lost for a short while. In a Group Switch with more than one Group Switch Module, the GCUs are connected via a bus. The bus is used for synchronization and communication between the clock units.

A Group Switch is non-blocking and comprises one or a number of Time Switch Modules, TSMs, with 1024 ports. The modules are arranged in a matrix. See figure 4-3. The time switch operates in accordance with the same principles as the LIM Switch. However, no attenuation or amplification takes place. The PCM lines from the LIMs are connected via terminal boards, i.e. Group Junctor Units.

The Time Switch Module, TSM, has a capacity of 31 PCM lines, i.e. 2 x 2 TSMs are required for 62 PCM lines, etc. A Group Switch can be expanded to a maximum of 8 x 8 TSMs and can handle 248 PCM lines.

A Group Switch is controlled from the connected LIMs. The Group Junctor Unit - Group switch side, GJUG, contains a microprocessor that handles the communications with the connected LIM via time slot T16. The processors on the GJU boards convey control information internally within the Group Switch.

The establishment of a connection between two LIMs starts with the LIM processors informing each other of which time slots have been selected on the respective PCM line. The Group Junctor Unit processors are then instructed to establish the connections via the relevant time switch modules. A connection through a Group Switch is by its nature one way and it is therefore necessary to establish two paths through the switch in order to obtain a two-way connection. See figure 4-3. In the illustration the path from LIM A to LIM B proceeds via Time Switch Module TSM 1/2, while the path from LIM B to LIM A proceeds via Time Switch Module TSM 2/1.

4.4.5   Inter LIM signalling

Inter LIM signals are signals sent from one LIM to another LIM via the signal channel (time slot 16) on a PCM line which is located in a device board position.

In establishing the signalling paths in a system, the following principles apply in order:

1. There shall be only one path between any two LIMs.
2. Signal transaction (both sending and receiving) between two LIMs shall take place on the same path.
3. A maximum of two links can be used for signalling.
4. In choosing a specific link, a link with long signalling capability will be considered as the first priority.
5. Should one link be chosen in one magazine, another link in a different magazine will be preferred for the second link selection.

For instance, in a 124-LIM system where the I/O LIM has four PCM lines and the rest have two PCM lines per LIM, two PCM lines will be used in each LIM. The lines with long signalling capacity will be chosen in the I/O LIM, if the I/O LIM is equipped that way. Applying the principle in another example, a two-LIM system will only use one PCM line for inter LIM signalling. In a three-LIM system, the maximum of two PCM lines per LIM will be used for inter LIM signalling.

The above principle also applies for manual blocking, faulty line handling and recovery. If a PCM line is manually blocked or any line in service becomes faulty, another line will be used for inter LIM signalling to allow a maximum of two PCM lines per LIM. When a faulty PCM line is recovered, the PCM line can be automatically put in-service if the LIM had not been utilizing two PCM lines.

The inter LIM signal packet size varies depending on the hardware of the path.

4.5   I/O Subsystem, IOS

4.5.1   General

A PABX requires a number of aids for external administration of programs and data. For example, by using commands it is possible to correct program errors and revise exchange data from a terminal. Data can be stored on a hard disk.

In the ASB 501 04 these functions have been implemented within function blocks in the I/O Subsystem.

The I/O Subsystem consists of hardware and software. The hardware permits the connection of different types of I/O devices.

All communications with the PABX take place via an I/O terminal. A hard disk unit, HDU, is used for memory backup, see Figure 4-4 . The primary task of the I/O Subsystem is to distribute the commands, programs and data to appropriate parts of the system.

4.5.3   Functions

The following functions are implemented in the various function blocks belonging to the I/O Subsystem:

Terminal administration

The terminal is a PC with start/stop protocol and V.24/V.28 connection.

Hard disk unit administration

The hard disk unit must be equipped with a SCSI (Small Computer Standard Interface) in accordance with the relevant interface specification. Similarly, the PC used for initial loading and for safety backup purposes must be equipped with the same interface.

Command administration

The command language constitutes a subset of ITU-T's MML (Man Machine Language).

Printout administration

Loading functions

Dumping functions

File administration

Buffer administration

4.5.4   Initial loading

The initial loading is made from a hard disk unit. The information comprises:

This disposition of the software across LIMs and the generation of exchange data can be made directly in the PABX by command, or with the aid of a command file or floppy disk prepared in advance.

4.5.5   System backup

The security of software stored in the RAM (random access memory) is provided by a backup unit, i.e. there is a hard disk unit in the PABX. The software is dumped to this after the initial loading of the PABX. A local backup of the LIM's data exists in the LIM's RAM memory in order to decrease the time needed to reload data.

If serious errors occur during program execution certain parts or all of the dumped code can be reloaded. On Reload the relevant software is read into the PABX from the backup unit.

4.5.6   Safety backup

A safety copy of the backup may be copied to a PC or a hard disk unit.

4.5.7   Dump of alterations

Dumps of the data and programs which may have been altered can take place at predefined time intervals, or by commands at any time.

4.5.8   I/O LIM

I/O devices can be connected to all LIMs in the system provided that the I/O interface boards and requisite control programs exist in each LIM.

Each system has a specific LIM that is defined as the I/O LIM . This LIM contains all necessary
I/O functions such as I/O checks, buffer administration, load/reload functions, file administration for the hard disk unit, and control programs for the hard disk unit and printer terminals.

I/O terminals

An I/O terminal is a PC or other type of visual display unit.

An I/O terminal can be connected locally to one of the V.24/V.28 interfaces on the Network Interface Unit, NIU.

For remote connection via the NIU the following options exist:

• Non-network connection via a modem connected to a V.24/V.28 interface.
• Network connection by using Telnet over the PPP (Point-to-Point Protocol) via a modem connected to a V.24/V.28 interface.
• Network connection by using Telnet via the Ethernet port.

Six terminals can be logged on concurrently, but the entire PABX permits a maximum of 20 connected terminals.

For connections via V.24/V.28 the terminal cable can be a maximum of 15 metres (50 feet).

The transmission rate over V.24/V.28 ports can vary from 300 bit/s to 9600 bit/s.

4.5.9   Communications language

The command and printout language is a subset of ITU-T's MML (Man Machine Language). For a definition consult ITU-T recommendations Z.311 to Z.341.

A complete command consists of a command code part and a parameter part. The command code part is a five letter mnemonic code that specifies the action to be taken. The parameter part consists of one or more parameter names in which each parameter name is followed by the relevant parameter value.

Generally, printouts are initiated by command, i.e. they are not spontaneous. Printout after command initiation can take place on the stated terminal immediately or may be delayed. Alarms may be printed spontaneously as they occur.

4.5.10   Monitoring communications

To prevent operational disturbance and to guarantee the administration of I/O devices, built-in checks exist in the interworking software. For example, all commands and associated parameters are checked to verify that they exist and are authorized for use in the requested combination.

The parameter formats and values are then checked to see that they are within the defined limits.

The characters and control characters from I/O terminals that are not used in the I/O Subsystem are ignored and an error message is issued.

The built-in checks, together with the detailed operational instructions, eliminate operational disturbance and allow the maintenance operator to handle the system in a logical, simple manner throughout.

4.5.11   Authorization classes

Each individual command can be freely assigned one of eight authorization classes, which are assigned an alterable password with a maximum of 16 characters (digits and letters).

Depending on the class to which a command belongs, it can be changed to another authorization class as required by circumstances.

4.5.12   User accounts

64 user accounts may be initiated with individual passwords, authority levels and expiration dates. Passwords and user names can each be 4-20 characters long. Each I/O port can be assigned an individual authority level.

4.5.13   Hard disk unit

The hard disk unit, HDU, is equipped with SCSI (Small Computer System Interface) and has a transmission rate between the I/O board and the hard disk unit of the order of 1 Mbit/s.

The HDU is used primarily for system backup and is placed in the same magazine as the I/O board.

4.5.14   Hard disk unit backup on PC

A PC may be used either to create a safety backup of a hard disk or to update a hard disk unit, HDU, with new software.

The PC is connected to the system's HDU via a SCSI (Small Computer System Interface) board in the PC. The contents of the PC's hard disk are copied from the system's HDU to create a safety backup. The contents of the PC's hard disk are copied to the system's HDU to recover a backup or upgrade the system. The system is then loaded/reloaded from the system's HDU.

4.5.15   Command functions

The ASB 501 04 contains commands for the following functions
(sorted by the command names):

4.6   Bus system and device configurations

Two types of bus system exist for the interchange of information between the various units in a LIM:

The control system bus is the interface between the LIM processor, the LIM Processor Unit, LPU, and the test boards, i.e. the Test Unit Mother Board, TEU-M, and the Test Unit Daughter Board, TEU-D.

The device bus is the interface between the device boards and the LIM Switch.

The address bus is 32 bits wide, which provides an addressable area of 4 Gbytes. The data bus is 36 bits wide.

The device bus is a 128 kbit/s or 2 Mbit/s serial channel used for the two way transmission of control information between devices and the LPU. The bus consists of two wires, one for each transmission direction.

The interface between the device position and the wiring in the LIM magazine's backplane is general. Each device position can normally serve up to 16 individuals (time slots in the LIM Switch).

The distribution of the switch time slots to the device boards in a LIM is shown in the schematic drawing, figure 4-5. The majority of a LIM's device board positions can be used for all board types requiring up to 16 time slots, i.e. foremost line circuits and external lines.

Device boards requiring 32 time slots must be placed on board positions ending with a 0 such as 00, 10, 20 and excluding the rest in the group. Device boards requiring 16 time slots must be placed on board positions ending with a 0 or a 2 such as 20, 22, 30, 32 excluding the odd slots. Device boards requiring 8 time slots may be placed on every position, the only restrictions being the physical space occupied by the system and the power dissipation requirements. Board position 00 in each magazine is intended for digital external line board(s) and/or Group Switch connection(s).

A duplicated control system has a duplicated control system bus. The device bus is common for both control systems, see Figure 4-5 .

4.7   System security

see  14 Operational security and reliability .

5   LIM features

Relevant details are entirely dependent on the PABX revision and are to be found in the revision associated description, see description SOURCE SYSTEM RELEASE ASB 501 04 R8.

6     Exchange features

Relevant details are entirely dependent on the PABX revision and are to be found in the revision associated description, see description SOURCE SYSTEM RELEASE ASB 501 04 R8.

System features

Features that are not directly user controlled or user dependent.

7     Extension features

Rules for repeated diversion

On a call to directory number B the call is diverted either to C or directory number D as defined below:

8   Telephone features

8.1   Digital system telephone features

Generally speaking an extension equipped with a Digital System Telephone can utilize the PABX features in an easier way than extensions equipped with analogue telephones. As the system telephones are equipped with pre-programmed keys for the most used features and programmable keys for other features (the most advanced also so called soft-keys and display), the features can be used without dialling procedures. For a full list of the available features, see description SOURCE SYSTEM RELEASE ASB 501 04 R8.

The Digital System Telephone is connected via a 2-wire telephone cable to a digital line board. Transmission between the telephone and the PABX uses burst signalling , in which speech and signalling between the telephone and the PABX have their own, mutually independent channels.

A Digital System Telephone can also be used with a Dual Access extension interface. This means that a Digital System Telephone and an ISDN S ₀ Terminal may be connected to the same 2-wire line via an adapter. The telephone then shares the directory number with an S ₀ interface, see  11.20 Dual access extension interface .

All the permanent and programmable functions of the Digital System Telephone are loaded in special function blocks in the LIM. This permits great flexibility and the stage-by-stage development and enhancement of the functions of the Digital System Telephone.

Five families of the Digital System Telephone are currently supported, DBA, DBC 199, DBC 500, DBC 600 and DBC 200. Only DBC 600, DBC 200 and DBVA 575X can be connected to a Dual Access extension. DBVA 575X is unique for the Swedish market.

For detailed information about buttons (keys), LEDs, programming, features, etc. See the appropriate Directions for Use.

A CD-ROM based program for PC, Designation Card Manager (DCM), is available to design, store and print localized or customized key designation cards for the DBC 200 telephones. It is delivered together with a number of pre-cut papers to fit the different telephone types and the extra key panel. Pre-cut papers can be ordered separately.

There are a number of features that are available to many Digital System Telephone models, but not to ordinary telephones. These features are for example:

Display features, such as:

• Message Diversion Information.
• Advice of Charge.
• Name.
• Calling/Connected Line Identity.
• Time and Date Display.
• Soft-keys.
• Call Progress Information .

and other features, such as:

• Multi Directory Diversion.
• Multiple represented Directory Number, MDN.
• Automatic Call Distribution functions for agent and supervisor .
• Additional Directory Number, ADN.
• Choice of Language.
• Dial by function key.

8.2   Cordless telephone features

Cordless telephones are available with and without Short Message Service, SMS.

For more information, see  11.39 Mobility and see  11.51 Short Message Service, SMS .

8.3   IP telephone features

The Ericsson IP telephone, DBC 413 01, is used to access the PABX features via the IP network. For a complete list with the available PABX features that can be accessed from the IP telephone, see the extra facility description for IP EXTENSION.

The IP telephone supports the following features when connected towards ASB 501 04:

• Basic functionality according to H.323v2 standard
• Inherent Free Seating (not the extra facility Free Seating)
• Wireless Application Protocol (WAP) is used between the ASB 501 04 and the telephone to implement the features. The WAP protocol is used on top of H.323 to access the PABX features.

Features, terminal:

• Support of G.711, G.723.1 and G.729a codecs.
• New features and corrections via software updates from a web server (the telephone is automatically updated with the software from the web server at the start of the telephone).
• IP address assignment automatically from a DHCP server or manually entered.
• Hands-free
• Loud-speaking
• Alphanumeric display, 3 rows x 40 characters
• 4 menu buttons (Soft-keys) connected to the display
• 10 programmable buttons
• Volume control for handset, loadspeaker, ringing signals
• Choice of different languages

Also see  11.34 IP extension .

8.4   ISDN telephone features

see  9 ISDN extension .

8.5   Analogue telephone features

Analogue telephones with DTMF signalling can utilize all the PABX features that can be executed by dialling procedures that consist of digits, star (*), hash (#) and R.

Analogue telephones with rotary dialling can utilize all the PABX features that can be executed by dialling procedures that consist of digits and R (if this key exists).

Telephones with a message lamp can be used. The key is switched on to indicate a new message.

Analogue telephones can use their built-in functions that are specific for each type of telephone, such as programmable keys for abbreviated dialling, handsfree etc.

9     ISDN extension

9.1   General

The integrated S ₀ interface supports the appropriate ETSI and National ISDN standards for basic call control and services for voice and data. The ISDN S ₀ terminal supports connection over the S reference point. The interface can be in a configuration for both point-to-point mode of operation and for point-to-multipoint.

It should be noted that the term ISDN S ₀ terminal is not necessarily a telephone. It is used as a common term for all sorts of equipment connected to the user side of the interface. It includes terminal equipment with an S ₀ interface or a terminal adapter with a connected non-ISDN terminal. An ISDN S ₀ terminal can thus be e.g. a telefax GP4, PC with ISDN board, PC with ISDN board and a telephone, terminal adapter with a handset, video telephone, Local Area Network, LAN, connection or an ISDN telephone.

An ISDN S ₀ terminal has access or partly access to most of the ASB 501 04 specific extension features using Keypad, i.e. a string of IRA (International Reference Alphabet) characters. The busy state related services such as Callback, Call Waiting, Diversion on Busy, Individual Call Pick-up, PABX group membership, Call Announcement, Extending and Intrusion are however not supported.

System related features available for other extensions are also available for an ISDN extension.

No proprietary ASB 501 04 extension features using keypad procedures are supported for ISDN terminals connected to the National ISDN S ₀ interface. System related features and features independent of keypad (command initiated) are however available for such terminals. National ISDN S ₀ terminals can also become involved in a number of services, but not request them.

9.2     ETSI ISDN S ₀

The integrated S ₀ interface supports the appropriate ETSI standards for basic call control for voice and data.

To control the supplementary services, the generic keypad protocol specified by ETSI is used. The Generic Functional protocol is only used for the control of Hold and Terminal Portability supplementary services.

The ISDN S ₀ terminal can interwork with Analogue, Digital and Data extensions (TAU2680) and the PABX operator.

The ISDN S ₀ terminal also interworks with the public ISDN network over the T reference point and within private networks over the Q reference point.

Some of the ASB 501 04 specific services listed below are only partly supported for an ISDN extension.

9.2.1   Basic services

Interworking and networking.
Bearer Services.
Teleservices.
Display Handling.
Generic Keypad Protocol.
Generic Functional Protocol.

9.2.2   ETSI standardized supplementary services

Calling Line Identification, Presentation, CLIP.
Calling Line Identification, Restriction, CLIR.
Connected Line Identification, Presentation, COLP.
Connected Line Identification, Restriction, COLR.
Hold and Retrieve.
Multiple Subscriber Number, MSN.
Subaddressing.
Terminal Portability, TP.
User-to-User Signalling, UUS.

Calling Line Identification

Calling Line Identification consists of two parts, i.e. presentation and restriction. The basic functions to support this supplementary service are accessible from the ISDN terminal interface, but a parameter set in the ASB 501 04 can override the information sent from the terminal.

Connected Line Identification

Connected Line Identification consists of two parts, i.e. presentation and restriction. The basic functions to support this supplementary service are accessible from the ISDN terminal interface, but a parameter set in the ASB 501 04 can override the information sent from the terminal.

Hold and Retrieve

This supplementary service allows a user to interrupt on an existing call and then subsequently, if desired, to re-establish the communications.

Multiple Subscriber Number

This supplementary service is defined as the ability to apply a set of different directory numbers (up to 32) to an access. Each directory number has its own Class of Service. This service is valid for both point-to-point and point-to-multipoint configuration.

Subaddressing

This supplementary service allows a user to send and receive extra address information in the form of a subaddress related to the called/calling/connected party number. This information is transferred transparently through the ASB 501 04.

Terminal Portability

This supplementary service allows a user to move a terminal from one telephone wall socket to another within Basic Rate Access, BRA, during the active state of a call. It also allows a user to move a call from one terminal to another terminal within BRA during the active state of a call.

User-to-User Signalling

This supplementary service allows a user to send and receive a limited amount of information to and from another user. Only services 1 (call set-up) and 3 (active phase) are supported. Service 2 (alerting phase) is not supported.

9.2.3   ISDN supplementary services not according to standard

Advice of Charge (call cost display).
Name Identification (name display).
Malicious call Identification.
Negotiation.

9.2.4   ASB 501 04 specific extension services

9.2.5   ASB 501 04 specific services not or only partly supported

Alarm Extension (an ISDN terminal cannot be initiated as an alarm extension).
Automatic Callback.
Call Announcing.
Call Waiting Indication, Call Offer (OK when third party).
Diversion on Busy and No Answer (when calling an ISDN).
Extending on Busy.
Individual Call Pick-up.
Intrusion (OK when third party).
Member in Group Hunting.
Rerouting on Busy and No Answer (when calling an ISDN).
Recorded Voice Announcement.

9.3     National ISDN S ₀

The integrated S ₀ interface supports the appropriate BELLCORE National ISDN-1 and National ISDN-2 standards for basic call control for voice and data. The interface can be in a configuration for point-to-multipoint mode of operation.

The Number Identification services are supported according to present standards. No other supplementary services are supported.

The ISDN terminal can interwork with Analogue, Digital and Datacom (TAU2680) extensions and the PABX operator.

The ISDN terminal also interworks with public ISDN networks over the T reference point and within private networks over the Q reference point.

This extension can use some ASB 501 04 services that are valid for all types of extensions (command initiated or system features). It can also be included in services as the invoked party.

9.3.1   Basic services

Interworking and networking.
Bearer services.
Teleservices.

9.3.2   National standardized supplementary services

Multiple Subscriber Number.
Calling Line Identification, Presentation, CLIP.
Calling Line Identification, Restriction, CLIR.
Connected Line Identification, Presentation, COLP.
Connected Line Identification, Restriction, COLR.
Subaddressing.

9.4   ISDN / Datacom (TAU2680) Interworking

Compatibility verification.
Negotiation.
Speed matching.

10     PABX operator features

10.1   PABX operator console

The ASB 501 04 can be provided with six types of PABX operator consoles:

• earlier type of symbolic display console.
• current type of symbolic display console (OPI-II) which can be connected to a Braille unit.
• earlier type of alphanumeric display console.
• OPI 3213, a DBC 213 based PABX operator console (only for sustaining).
• OPI 3214, a DBC 214 based PABX operator console.
• Operator Work Station (OWS), a PC-based operator console.

All types of PABX operator consoles can operate concurrently in the same system.

The console is connected to the LIM via a single pair of wires and does not require any external power connections.

10.2   The consoles with the symbolic display unit (symbolic/OPI-II consoles)

10.2.1   PABX operator display

The visual display unit consists of five individual displays, each of which is made up of a liquid crystal display capable of presenting symbols and numerical information.

Programmable tone ringer character (OPI-II console ONLY).

The display unit is divided into the following fields:

10.2.2   PABX operator keyboard

The PABX operator keyboard is equipped with 21 function keys and a 12 digit keypad (including the digits 0 to 9 and the characters * and #).

All keys are monostable. There is also a volume control for the headset/handset. No call processing functions require simultaneous depression of several keys.

For detailed information about buttons (keys), programming, symbols etc. See the appropriate Directions for Use.

10.3   OPI 3214 with alphanumerical display

10.3.1   OPI 3214 operator console

The visual display consists of five display rows. Each of these is capable of presenting up to 40 alphanumerical characters. The display rows are used according to predefined character layouts and supports Name Identity and Choice of Language

10.3.2   OPI 3214 operator keyboard

The OPI 3214 keyboard is identical to the normal DBC 214 digital system telephone, except for that four buttons have been removed to give more room for the speech and clear buttons.

For detailed information about buttons etc, see the direction for use for OPI 3214.

10.4   OPI 3213 with alphanumerical display

10.4.1   OPI 3213 operator console

The visual display consists of three individual display rows, each of which is capable of presenting up to 40 alphanumerical characters. The display rows are used according to predefined character layouts and are similar to the ones for the alphanumeric PABX operator console.

10.4.2   OPI 3213 operator keyboard

OPI 3213 operator keyboard is identical to the normal DBC 213 Digital System Telephone.

For detailed information about buttons (keys), soft-keys etc., see the directions for use for OPI 3203 AND 3213.

10.5   OWS/PC-based operator console

This console provides an alternative solution to traditional operator consoles and has additional directory assistance functionalities. It is a Windows NT application which runs on a Windows NT compatible PC. The connection to the exchange is via a COM port on the PC and a TAU-D. The TAU-D allows the PC to communicate with the exchange through the Operator Interface. At the same time, it provides the speech path for the operator through a Digital Telephone Set.

The directory functionalities depend on the directory data maintained by the Directory Manager application.

10.6   Features

11     Add on features

For detailed information about functions, capacities, etc. of the features listed below, see the appropriate Extra Facility Description and the Source System Revision Associated Description.

11.1   16 kbit switching and multiplexing

See also the Extra facility description for 16KBIT SWITCHING AND MULTIPLEXING.

16 kbit/s switching and multiplexing is a feature that can be used in conjunction with Voice Compression. The feature makes it possible to multiplex sub-rate data in a Voice Compression board and then switch the sub-rate data through the exchange. It also allows compressed speech calls to be switched through transit exchanges in compressed format in order to avoid poor speech quality in transit calls. Voice Compression by itself only provides node-to-node compression which may result in poor speech quality if the call has to pass several transit nodes.

It is also possible to use transit switching of 16 kbit/s without compression/decompression.

Four 16 kbit/s channels may be multiplexed into one outgoing 64 kbit/s channel.

With the optional feature 16 kbit Switching and Multiplexing, the end-to-end compression facility can also be used in a network without decompression and compression in the transit nodes. This feature also provides data which can be switched in a 16 kbit/s format.

11.2     Access Agent

See also the Extra facility description for ACCESS AGENT.

The Access Agent consists of hardware and software components. The Access Agent hardware can be mounted in the LIM as a standard circuit board and use power from the LIM. The board has an Ethernet interface. The Access Agent software includes a real time operating system with file system, security control and communication protocols.

The Access Agent can provide general TCP/IP functionality, such as V.24 to Telnet, File Transfer Protocol, Telnet and Point-to-Point Protocol. The SNMP Agent and/or Agent Call Account Buffer applications can be run on top of the Access Agent.

11.3     Account Code

See also the Extra facility description for ACCOUNT CODE.

The primary purposes of the usage of Account Code with verification are:

• To provide the extension with the ability to charge a call to an Account Code, which may represent a particular project, department or client, instead of charging the calling extension's number.
• To provide a means of preventing unauthorized usage by forcing the extension to enter an Account Code before dialling an external number.

An extension dials a procedure and the Account Code prior to the destination number.

11.4     Agent Call Account Buffer, ACAB

See also the Extra facility description for AGENT CALL ACCOUNT BUFFER, ACAB.

The optional Agent Call Account Buffer application is a buffer that can be set up for Call Information Logging, CIL, data which can be stored in compressed or uncompressed format, and further transferred via the File Transfer Protocol, FTP. The Agent Call Account Buffer requires the Access Agent.

11.5   APNSS

See also the Extra facility description for APNSS.

The Analogue Private Network Signalling System is a system for signalling between exchanges in a private network. It is a common channel signalling system that facilitates transmission and use of facilities in networks.

It is primarily used when facilities via the network are needed but where traffic between two exchanges is so low that subscription to a 2048 kbit/s digital interface is not economically satisfactory, or where only an analogue network is available.

11.6     Automatic Call Distribution, ACD

See also the Extra facility description for AUTOMATIC CALL DISTRIBUTION.

11.6.1   General

ACD is an automated solution used to distribute a large quantity of incoming calls to a service controlled by the number dialled by the calling party. Each service is connected to an ACD group consisting of one or more agents who handle these calls. In this way it is possible to handle a large number of incoming calls without the need for PABX operators to route the calls.

The agents are assigned as members of, and can answer calls from, one or more ACD groups. The selection of an agent can be based on selection priority and type of selection.

11.6.2   Facilities

The following facilities are available:

• Agent Log-On/Off
• Temporarily Unavailable
• Agent Unavailable Indication
• Clerical Time
• Free Member Selection
• Delay Call Selection
• No Answer Handling
• Dynamic Queue length
• False B-answer
• Overflowed Calls
• ACD Group Follow-me
• ACD Group Do Not Disturb
• Supervisor Intrusion
• Help Line
• Group Name Display
• Call Qualification
• Member/Queue Display
• ACD Back-up groups

11.7     Automatic Network Call Distribution, ANCD

See also the Extra facility description for AUTOMATIC NETWORK CALL DISTRIBUTION.

11.7.1   General

ANCD provides the ability to distribute calls intelligently to ACD groups located in the same or different nodes of an ASB 501 04 ISDN network. ANCD is a powerful complement to the basic ACD feature.

Incoming calls may be distributed to different ACD groups based on the status of the ACD groups that are handling the required services. The distribution functionality can be used to provide the best answering capability at any moment, and it can also be used to redistribute calls from one ACD group to another in overflow situations.

11.7.2   Facilities

Distribution of Calls

The distribution of calls from an ANCD group to a satellite group, i.e. an ACD group or another ANCD group, is based on the current status of the satellite group. This is to enable distribution of a call to the satellite group which is best suited to answer the call.

Satellite Group Selection

There are four principles for selecting a satellite group with free agents:

• Initiation order

  Searching for a satellite group with free member in the ANCD group is performed in the order in which the satellite groups were initiated into the ANCD group.

• Rotating order

  Satellite group with free members is selected by rotation.

• Highest number of free agents

  The satellite group which has the highest number of free agents is always selected.

• Load sharing

  The selection of a satellite group with free members is based on percentage load sharing.

A distributed call is not allowed to be overflowed to superior ANCD group.

Overflow of Calls

New calls received in a satellite group where there is no possible selection (no free agent and no free queue position), or where the overflow time would be exceeded, are overflowed to the superior ANCD group. If there is no available superior ANCD group the call is diverted to an individual divertee position for the satellite group, if defined. If no individual divertee position is defined the call is rerouted if possible, otherwise it is rejected.

An overflowed call to a superior ANCD group is treated as a new call and distribution takes place.

Supervisor Control of Close/Open Traffic Distribution to a Satellite ACD Group

Supervisor Position is any digital extension with ACD supervisor Class of Service. It allows the supervisor to open and close traffic distribution from an ANCD group to a satellite ACD group located in the same node as the supervisor position. A supervisor can open and close traffic distribution to any satellite ACD group in the node independent of which ANCD network the ACD group belongs.

11.8     Authorization code for extension

See also the Extra facility description for AUTHORIZATION CODE FOR EXTENSION.

There are two types of authorization code available to control or limit access to an extension. Both types of code can be used from an analogue extension, a digital extension, a cordless extension, an IP terminal or an ISDN terminal.

Common Authorization Code

This is a code that does not have to be affiliated to any directory number in the system, but it can be limited to one instrument. The code cannot be changed by the user. Two different functions are provided:

Individual Authorization Code

This is a code that is always affiliated to a directory number in the system. In addition to those types of extension already mentioned, it can also be used from a PABX operator console for locking and unlocking purposes. In addition to the functions available to common codes, individual codes also have the following function:

11.9     Call Information Logging, CIL

See also the Extra facility description for CALL INFORMATION LOGGING.

11.9.1   General

The PABX can provide data for all types of calls and mobility events. Data records for the calls and mobility events are generated in the PABX and, after the end of the call or occurance of mobility event, the required data are output via a special channel to a peripheral unit. Abandoned incoming trunk call to the Hunt Group, Automatic Call Distribution group, extension, or PABX operator due to no answer, are also registered.

Possible peripheral units are:

Call Information Logging is also the base for the CIL over Ethernet feature and the CIL buffer on HDU feature.

11.9.2   Logging of data

The following data are logged in internal data records:

11.9.3   Output of data

The types of call or which types of mobility event output can be defined by command.

The call output criteria are:

The mobility output criteria are:

Call data are output via the Network Interface Unit, NIU, at a speed between 300-19200 bit/s
(ITU-T V.24/V.28 or V.24/V.10). The output can be requested in four different fixed formats and it is possible to select from a number of different interface character sets.
For output of call data via Ethernet, see see  11.12 CIL over Ethernet and see  11.13 CIL buffer on HDU .

A flexible format may be defined where the individual fields in a call record can be selected by command. In case of mobility logging only flexible formats can be used.

11.10   CAS extension, EL7

See also the Extra facility description for CAS EXTENSION, EL7.

11.10.1   General

The Channel Associated Signalling, CAS, extension interface provides a digital connection to external equipment and offers them, through PCM lines, the functionality of analogue extensions.

The interface supports 1.5 and 2 Mbit/s digital systems. Each 2 Mbit/s interface handles 30 extensions and each 1.5 Mbit/s interface up to 23 extensions. Both, 2 Mbit and 1.5 Mbit, can be handled simultaneously in the same ASB 501 04 system.

Different signalling systems are also handled simultaneously thus allowing connections of a variety of external equipment to an ASB 501 04 system, e.g. cordless switch, digital multiplexers, voice mail, etc.

11.10.2   Function

The functionality of the analogue extension is only limited by the connected external equipment, e.g. if the external equipment does not support Hook-flash or Message Waiting.

Besides the analogue extension functionality, the CAS extension offers the ability to switch data from 64 kbit clear-channel data-interfaces connected to the external equipment. This is applicable, for example where a multiplexer, MUX, is used as a simple remote unit and permits connections from multiplexed data extensions to other ASB 501 04 data interfaces such as ISDN external lines and Digital Private Network Signalling System, DPNSS, external lines, but not to internal data extension.

Synchronization can be received via the external equipment clock source. This is only advisable when the ASB 501 04 is not linked to digital exchanges in an external network, and when the external equipment clock stability is better than the ASB 501 04 internal clock.

11.11     Choice of Language

See also the Extra facility description for CHOICE OF LANGUAGE.

11.11.1   General

The purpose of the Choice of Language feature is to allow every user of a Digital System Telephone with a display, an Ericsson IP telephone or an OPI 3214 to select the language in which the text messages appear.

The telephone user can select one of ten available languages by dialling a procedure from the telephone. The system administrator can specify the language for any digital telephone or IP terminal from the ten available languages or set it as the exchange language.

The PABX operator using the OPI 3214 can select one of the five available languages (English, French, German, Spanish and Italian) by dialling a procedure from the operator console.

11.11.2   Functions

This feature allows the text strings in the display in the Digital System Telephone's display, Ericsson IP telephone's display or OPI 3214 console's display on a per-user basis to be selected.

Any user can change the language for the extension or OPI 3214 console by dialling a procedure from that particular terminal.

There is a separate set of eight special characters per language. The exchange administrator can define and view the special characters in any of the languages.

11.12     CIL over Ethernet

See also the Extra facility description for CIL OVER ETHERNET.

Via this feature, Call Information Logging data can be output through the Ethernet port of the Network Interface Unit, NIU, using the TCP/IP (Transport Control Protocol/Internet Protocol) to the Call Information Logging server on the network. This feature must be used in conjunction with the Call Information Logging feature see  11.9 Call Information Logging, CIL .

Through the Ethernet, Call Information Logging can communicate with its server using the high Ethernet bandwidth as compared to the low bandwidth of V.24 based serial communication. This feature can only use the TCP/IP.

The advantage of using the Ethernet connection is that the higher bandwidth enables the data to be collected remotely by users from different locations.

11.13     CIL buffer on HDU

See also the Extra facility description for CIL BUFFER ON HDU.

With this feature, call information logging output data can be stored on the HDU(s) for processing later. The data can be grouped into daily, weekly or monthly files. This feature must be used in conjunction with the CIL feature see  11.9 Call Information Logging, CIL .

11.14   CSTA (Computer Supported Telecommunication Applications)

See also the Extra facility description for COMPUTER SUPPORTED TELECOMMUNICATIONS APPLICATIONS (CSTA), CS.

CSTA is an application protocol which enables a computer domain to communicate with a telephony domain. It supports applications or services normally provided by one domain to be available to the other domain which usually cannot support such an application without major enhancement or redesign.

The main type of application for this feature is the Call Centre, where agents handling incoming calls can view screen updates synchronized with the telephone call. When a call arrives at an agent position, a message is sent from the exchange to the computer to inform it of the event. The message contains information such as which agent received the call, who is calling and which number was dialled. The computer typically uses this information to perform a data base search and then update the agent's screen with the information retrieved.

Other types of application are outbound Call Centres such as telemarketing and debt collection.

The following traffic cases can be handled:

• Directory Number Verification Service.
  
• Monitoring of:
  • Analogue extensions.
  • Channel Associated Signalling, CAS, extensions.
  • Digital extensions (Own Directory Number/Additional Directory Number).
  • Automatic Call Distribution groups.
  • Automatic Network Call Distribution groups.
  • Analogue telephone set.
  • DECT cordless telephones.
    
• Telephony services for digital extensions (Own Directory Number/Additional Directory Number) e.g.:
  • Make call.
  • Press programmable key.
  • Press fixed function key.
  • Transfer.
  • Conference.
  • Call deflection.
  • Single step transfer.
    
• Telephony service for cordless extensions e.g.:
  • Make call.
  • Transfer.
  • Conference.
  • Call deflection.
  • Single step transfer.
    

11.15   Data traffic

see  13 Data communication .

11.16     Dial by Name

See also the Extra facility description for DIAL BY NAME.

The Dial by Name function allows a user to initiate a call by entering the other party's name, or just the beginning of it, via a standard keypad with letter designations and soft-keys.

The function provides a directory of names with their associated telephone numbers. Numbers in the directory can be internal or external.

The Dial by Name function is only available for digital telephones of DBC 203/213 type via a standard keypad with letter designations and soft-keys.

There is one interface for the digital telephone user and another interface for the system administrator:

• The digital telephone user interface allows searching in the name list by use of the keypad and soft-keys. Two of the soft-keys are used to scroll up or down in the list and one soft-key is used to execute the call when the requested name is found.
• Via the I/O terminal, the system administrator interface allows the addition, editing, printing and removal of names in the list.

11.17   Digital Line Repeater, DLR-3

See also the Extra facility description for DIGITAL LINE REPEATER, DLR-3.

DLR-3 is a Digital Line Repeater for use in an Ericsson PABX to provide a digital interface for users at long distances. To this digital interface it is possible to connect telephones, data modems etc. that fulfil the proprietary Ericsson Digital Adapter for Subscriber Loops protocol.

DLR-3 consists of two identical stand alone boxes, unit 1 connected to the exchange and unit 2 connected to a Digital Telephone System.

Unit 2 has an external power supply.

11.18     DNIS for ACD

See also the Extra facility description for DNIS FOR ACD.

Dialled Number Information Service for Automatic Call Distribution.

This service which provides the ability for the Automatic Call Distribution groups to identify different customers based on the number the customer is dialling.

When the call is presented to the agent, i.e. when it is ringing on the digital telephone and after answer, the DNIS number and name are displayed on the telephone set together with the calling party number and name. This is applicable if the DNIS name, and the calling party number and name are available for presentation on the digital telephone's display.

Each DNIS number is stored together with its affiliated name and service group number. The service group number can be either an Automatic Call Distribution group or Automatic Network Call Distribution group. An incoming DNIS call fetches its stored name and the call is sent to its specified service group number. When the call is presented to an agent the DNIS number and name are presented on the digital telephone's display.

11.19   DSS1 Network side

See also the Extra facility description for DSS1 NETWORK SIDE.

Primarily for Branch office applications, the ASB 501 04 can act as the network side of the Digital Signalling System 1 protocol, for example to connect external products such as Inverse Multiplexers, Video Conference units, Routers, LAN gateways, and Data Conference units, primarily designed for use in public networks. The following services are supported:

• Basic Call (with bearer capabilities Speech, 3.1 kHz Audio, 7 kHz Audio, and 64 kbit/s unrestricted digital information).
• Calling Line Identity (presentation/restriction).
• Connected Line Identity (presentation/restriction).
• Direct Dialling In.
• Multiple Subscriber Number.
• Subaddressing.
• User to User Signalling.
• Advice Of Charge (during/end).

11.20     Dual access extension interface

See also the Extra facility description for DUAL ACCESS EXTENSION INTERFACE.

11.20.1   General

A Dual access (ISDN-Digital Telephone System) extension provides the following functionality to a desktop via the Ericsson proprietary Digital Adapter for Subscriber Loops (2B+D) interface:

• Digital telephone functionality.
• Personal Screen Call in a PC with or without voice capabilities and signalling in multimedia protocol via V.24 interface.
• The same ISDN services as available via ISDN S ₀ Terminal Interface.

11.20.2   Function

A dual access (ISDN-Digital Telephone System) extension has a unique Own Directory Number for the whole interface and is handled with standard commands.

When calling to a dual access extension, the calls are distributed depending on the bearer capability and high layer compatibility of the call:

• Speech calls are sent to the Digital Telephone System.
• Data calls are sent to the ISDN interface (e.g. video telephony, telefax group 4 or digital information).
• Audio 3.1 kHz calls are distributed to the Digital Telephone System or ISDN depending on an extension parameter.

Additional Directory Numbers can be affiliated to both the Digital Telephone System and the ISDN interface.

Besides the facilities provided by digital telephones, this type of extension offers a Redirection facility which permits all incoming calls to the Digital Telephone System to be forwarded to the ISDN interface. This is controlled by a dedicated key in the Digital Telephone System.

Additionally it is also possible to switch a call between the Digital Telephone System and an active Personal Screen Call application (Ericsson proprietary) running on a PC connected at the ISDN interface.

11.21   Duplicated control system

see  4 Service system, SES .

11.22     Dynamic Route Allocation, DRA

See also the Extra facility description for DYNAMIC ROUTE ALLOCATION.

The Dynamic Route Allocation provides very flexible route allocation, via digital public or private networks. Logically the dynamic route is identical to a tie line or Virtual Private Network route supporting all the extension and operator services otherwise available via an ISDN tie line. Integrated Voice Compression and Traffic recording are also supported. B-channels are set-up depending on the traffic situation and the dynamic bandwidth allocation. 64 kbit/s clear channels are offered automatically for facsimile transmission.

Dynamic Route Allocation provides switched interconnections over public ISDN networks or digital Channel Associated Signalling networks with full networking functionality without placing requirements on or impacting intervening networks. There is no requirement for user to user capabilities in the network and it is well suited for the integration of small sites into the corporate network. It enables ASB 501 04 networking to encompass small (branch) nodes, or nodes with low intrasite traffic, where the leasing of lines is not an option due to the cost. It also offers potentially large cost reductions for any operator of a private network.

11.23   Electronic Mail

See also the Extra facility description for ELECTRONIC MAIL.

An Electronic Mail System is used for storing messages in the users' mailboxes and for sending them to and from these mailboxes. Interworking between the mail system and the PABX enables waiting messages to be announced, i.e. messages that have not yet been read or printed out.

When a message has been registered in a mail system the message is signalled to the ASB 501 04 which notifies the relevant extension. The signalling is performed via the Network Interface Unit, NIU.

After a message has been presented to the receiving party, or cancelled by other means, the message system informs the sending party and the notification ceases unless other messages to the extension are waiting to be announced.

Apart from the mail system itself, the following hardware is required for connection to
the ASB 501 04:

• Network Interface Unit, NIU (or Information Computer Unit, ICU).
• Cables.
• Modems for data transmission if the cable length exceeds 15 metres (50 feet).

11.24   Enhanced CAS interface for mobility

See also the Extra facility description for ENHANCED CAS INTERFACE FOR MOBILITY.

11.24.1   General

The Enhanced Channel Associated Signalling interface for mobility for freestanding Radio exchanges provides a digital connection to the radio exchange that has the ability to transfer calling, connected and diverted numbers, as well as Message Waiting Indications to the radio exchange. Also, Unavailable cordless state and alerting signal are reported from the radio exchange to the ASB 501 04. All this information is sent and received via Channel Associated Signalling, CAS.

11.24.2   Function

11.25   Estimated waiting time announcement for ACD

See also the Extra facility description for ESTIMATED WAITING TIME ANNOUNCEMENT FOR ACD.

The Estimated Waiting Time announcement for Automatic Call Distribution feature works in conjunction with the Recorded Voice Announcement feature.

When a call enters the Automatic Call Distribution group queue, different messages can be provided instead of a standard group announcement. The selection of the messages is based on the estimation of the time the calling party is likely to wait before the call is answered. Appropriate messages can be assigned to various wait time ranges by commands. These commands can also be used to select whether to play a standard group queue announcement, or the wait time messages for a particular Automatic Call Distribution group.

If the system is unable to provide an appropriate announcement for the estimated waiting time, the standard group queue announcement is provided.

11.26   FAX III recognition

See also the Extra facility description for FAX III RECOGNITION.

FAX III recognition is a feature which makes it possible to transmit FAX group 3 traffic over lines with Voice Compression. The use of FAX III recognition is controlled with FAX licences handled by the Licence Server.

The number of simultaneously ongoing FAX calls in a system are controlled by the Licence Server. The total number of installed FAX licences can be further divided into several pools (one pool in each LIM) by command.

The usage of FAX licences in combination with the Voice Compression board makes it possible to transmit FAX group 3 calls over external lines with Voice Compression.

The V.27t, V.29 and V.17 FAX group III protocols are supported.

11.27   Fixed remote extension

See also the Extra facility description for FIXED REMOTE EXTENSION.

The fixed remote extension feature makes it possible to have public terminals with hotline or delayed hotline functionality as extensions in ASB 501 04. The feature is carried out as CAS extensions in the exchange.

When the public terminal lifts the handset a call is established to a predefined number in the PABX. After validation of the received public A-number, a dial tone will be sent to the public terminal.

The public terminal will then get fully access to the functionality and features as for a CAS extension. Also internal parties which make calls to the fixed remote extension retain full functionality, such as call back, camp on from operator, etc.

11.28     Free seating

See also the Extra facility description for FREE SEATING.

Free seating is a service that allows the user to log on to any available telephone (DTS, CXN, ATS), and get the user's personal categories, calls and messages.

When initiating the virtual generic extension for a Personal number, which must be done for the Free seating feature, a licence for Free seating is seized.

The user logs on/off by using procedures. Log off can also be done remotely from a PABX operator and DISA. An individual authorization code must be used. When logged on, the user's set of function keys will be used, but only certain functions will follow the user. Some functions are barred, e.g. Internal group hunting, ACD and ADN/MDN. A special dial tone is used to indicate that the user is logged on.

11.29   GICI over Ethernet

See also the Extra facility description for GICI OVER ETHERNET.

This feature, General Information Computer Interface over Ethernet, enables applications such as Voice Mail, Electronic Mail and Interception Service to communicate with their respective application computers through the Ethernet port of a Network Interface Unit, NIU, using the TCP/IP (Transport Control Protocol/Internet Protocol). The Ethernet interface can co-exist with the V.24 based GICI and the Information Computer Unit, ICU. All three types of interface may exist in the system at the same time. More than one application of the same type, for example three Voice Mail systems, may be set up to use any combination of the three types of interface.

The advantage of using Ethernet is that it provides higher bandwidth thereby improving performance.

The feature requires no additional hardware other than the NIU, for example.

11.30   Hospitality

See also the Extra facility description for HOSPITALITY APPLICATION.

The Hospitality Application offers functionality especially aimed for the Hospitality industry, i.e. Hotels, Hospitals, Cruise ships, Exhibition centres and Convention centres etc. Functionality is provided within the following areas: guest check-in, guest rooms and Service quarters.

For guest check-in the following functionality is provided:

• Information about the guest can be entered. The information could be language, VIP-status, credit card information, room number etc.
• The guest can choose if name presentation will be allowed or restricted.

For guest rooms the following functionality is provided:

• The rooms can be either vacant or occupied.
• The rooms can be equipped with various types of extensions.
• Some features are barred to prevent improper use.

For Service quarters the following functionality is provided:

• The name of the guest will always be displayed at the Service Quarter Telephone even if the guest has name presentation restriction.
• The information that is entered when a guest checks in will be displayed at the Service Quarter Telephone.

11.31     Integrated Voice Mail

See also the Extra facility description for INTEGRATED VOICE MAIL. There exists also a free standing version see  11.59 Voice Mail .

Integrated Voice Mail allows users to extend the telephone services with an easy-to-use answering service. The extensions are provided with an answering service, that is integrated with the system and that can indicate received messages with the help of the Message Waiting Service. In this way, the users are alerted with a signal/indication that a message is pending. The user then accesses the Integrated Voice Mail system by one key stroke, and is instructed via voice prompts to access the messages.

Users can change their personal greetings and manage the messages in the mailbox from the office telephone or from other locations. It is possible to assign more than one Integrated Voice Mail system per user in order to provide flexibility in the configuration.

The system is designed to be modular and affordable from a one LIM system to maximum system size. As the system is provided in a two Printed Circuit Board configuration built into the ASB 501 04, it uses the internal power and is not effected by external power loss. The configuration of the system is carried out via I/O commands.

A Voice Mail system is identified by the Voice System Directory number which is a normal extension number used to address the user's mailbox. The user calls this number to listen to messages, or to change personal greetings, etc. The number is also used as the diversion position when the user wants incoming calls to be answered by the Integrated Voice Mail system.

11.31.1   Localisation of voice prompts, LVP

The Localisation of Voice Prompts application, LVP, is a Windows based PC-application. LVP provides the user with the possibility to record their own set of Integrated Voice Prompts in any language. The recorded voice prompts will, at recording, be stored on the PC hard disk, and can then be transferred to the hard disk of the Integrated Voice Mail in the ASB 501 04.

11.32     Interception Service

See also the Extra facility description for INTERCEPTION SERVICE.

11.32.1   General

Message Diversion is activated by an extension procedure containing an interception message. The interception message is sent to the connected interception computer. All new calls to an extension which has activated interception diversion are diverted directly to an answering position programmed as the Message Diversion position. The purpose is to provide answering position personnel with a better means of giving callers meaningful interception messages

A prerequisite for using Interception Services is that an external interception computer is connected to the PABX via the general interface (V.24/V.28) for information systems or Network Interface Unit (V.24/V.28) or Ethernet.

11.32.2   Function

An individual or common diversion position must be chosen as the Message Diversion position which is also used for Direct Diversion. An extension, internal group hunting group number, a line pick-up group (Common Bell group), an individual PABX operator or a group of PABX operators may be used as the Message Diversion position.

Message Diversion is activated and cancelled either from the interception terminal or by a procedure from the extension.

When an extension has several types of diversion activated at the same time, they have the following priority order:

The relative priority of 1 and 2 may be reversed.

Message Diversion can also be activated/cancelled for another extension from a Follow-me position.

The procedure for booking Message Diversion includes an interception message which is sent direct to the interception computer.

11.33   Inverter

See also the description for DC/AC INVERTER FAMILY.

The inverter converts -48 V DC to 230 V AC. When mains power disappears, the inverter distributes 230 V AC to e.g. one I/O terminal.

11.34     IP extension

See also the Extra facility description for IP EXTENSION.

The IP extension is a facility in the ASB 501 04 that allows to merge the two networks that usually exist in an office (networks for voice and data communication) into one and the same network; the IP network. It allows employees to work remotely and also employees at remote small offices to connect their terminals to the system. The ASB 501 04 will grant access and provide services for these extensions.

It is important to point out that the IP extension makes use of the IP network (a LAN typically) to transmit the voice. This IP network uses TCP/IP as the underlying protocol and therefore the voice is converted into packets in order to be transmitted via the IP network, and then unpacked in the other end. The use of the LAN to transmit voice may reduce transmission costs, and allows the transmission of different media (voice, video, data) over the IP network.

The IP terminals connected to the IP network, can be either IP telephones or IP PC-clients. The IP terminals, that use a proprietary protocol to communicate with the ASB 501 04 and thus have the possibility to request some services and show call progress and service information in a display, are referred as Ericsson IP telephone.

11.35     Least Cost Routing, LCR

See also the Extra facility description for LEAST COST ROUTING.

11.35.1   General

Least Cost Routing allows the system to select the most economical route for an outgoing public call. Least Cost Routing numbers are programmed from the I/O terminal.

11.35.2   Function

Least Cost Routing performs analysis of the dialled number including the Least Cost Routing access code and attempts to route the call over the most economic available route at any time based on the following:

• external line availability.
• user's Routing Class of Service.
• user's Trunk Call Discrimination Class of Service.

The routing capabilities are applicable to both voice and data traffic.

11.36   Least Cost Routing, Time of Day, LCR-ToD

This is an additional feature to Least Cost Routing.

See also the Extra facility description for LEAST COST ROUTING, TIME OF DAY.

Least Cost Routing, Time of Day is a function that allows the system to make the selection of the most economical route for an outgoing call dependent on the time of day and the day within the week.

This means that it is always possible to select the most economical route even if the cost relations between the different routes vary with the time of day and day within the week.

11.37   MD Disk Tool

MD Disk Tool is a Windows application, available via the service toolbox.

11.38   Mobile extension

See also the Extra facility description for MOBILE EXTENSION.

The mobile extension feature makes it possible to have public terminals as extensions in ASB 501 04. The feature is carried out as CAS extensions in the exchange.

Any public terminal which calls a predefined number in the PABX by dialling either manually, using the phone book or the calling card service and after executed validation or by entering a valid PIN-code will receive dial tone from the PABX.

The public terminal will then get fully access to the functionality and features as for a CAS extension. Also internal parties which make calls to the mobile extension retain full functionality, such as call back, camp on from operator, etc.

It is possible to define additional public terminals which can alternate as answering position by dialling a procedure.

11.39     Mobility

See also the Extra facility description for MOBILITY.

11.39.1   General

The Integrated DECT interface provides connection of a DECT equivalent Radio Fixed Part i.e. a base station, directly to the ASB 501 04. The interface between the Radio Fixed Part and the
ASB 501 04 is an Ericsson proprietary interface.

The physical terminal, the Portable Part (cordless telephone), is carried around and has radio connection using a DECT air-interface to one of the Radio Fixed Parts. The air-interface supports the Generic Access Profile, GAP, and the Cordless Terminal Mobility Access Profile, CAP, standards (subsets of the DECT standard).

The ASB 501 04 provides the Portable Part with the ability to have messages displayed, such as whether to have a feature activated, the state of the call, calling/connected name, calling/connected number, Message Waiting Indication, etc. If and how the information is displayed on the terminal depends on its capabilities.

11.39.2   Function

Basic traffic

The Integrated DECT interface supports incoming and outgoing voice calls.

Before a cordless extension can make or receive a call, the location of the extension has to be known by the ASB 501 04.

Portables that are out of radio range, switched off, or not located, are marked as Not Available in the ASB 501 04.

Location procedures

The location registration procedure makes the location of the extension known to the ASB 501 04, and makes it possible for Portable Parts to move between different Radio Fixed Parts in the same ASB 501 04 system and still be reached by the ASB 501 04.

The Detach function is initiated when the Portable Part is switched off in order to mark it as Not Available in the ASB 501 04.

Handover

Handover is the process of switching a call in speech from one physical channel to another physical channel.

In the ASB 501 04 environment calls can be handed over between Radio Fixed Parts that are located in the same LIM or in different LIMs within the same ASB 501 04 exchange.

Security procedures

The security procedures consist of the Access-rights, Authentication and Ciphering procedures.

The Access Rights procedure supplies the Portable Part with parameters needed for access to the ASB 501 04. This procedure is the on-air-subscription and is executed before the location registration procedure. It has to be done before any calls can be made to or from the Portable Part.

The Authentication procedure verifies the identity of the Portable Part and/or the Radio Fixed Part. This procedure is always invoked in connection with the Location registration procedure.

The voice and signalling channels over the air-interface are encrypted (ciphering) to prevent other radio equipment from listening to the conversation or from tapping data.

Features

The integrated cordless extensions have access to most of the services and features that are offered to other extension types in the ASB 501 04. The services are supported according to DECT standards. In addition the portables can have implemented terminal specific features.

The ASB 501 04 provides the portable with display messages such as feature activated, the state of the call, calling/connected name, calling/connected number, message waiting indication, date and time etc. If and how the information is displayed on the terminal depends on its capabilities.

Short Message Service, SMS, is supported point-to-point.

11.40   Modem Group

see  13 Data communication .

11.41   Music on Hold, MoH

See also the Extra facility description for MUSIC-ON-HOLD.

Music on Hold is a facility for relaying sound such as music or marketing information, for example, to a parked or queued subscriber or extension. The sound information is transmitted from a tape recorder or a radio set.

Music on Hold/Wait is not available to PABX operators, IP terminals or for calls queued to PABX operator.

Two different boards may be used for Music on Hold:

• a multi-purpose Tone and Multipart Unit, TMU, for auxiliary devices with two analogue inputs for the connection of sound equipment.
• a special Tone Sender Unit, TSU, equipped with three analogue inputs for the connection of sound equipment.

Each sound equipment unit is connected to the Main Distribution Frame from where a cable runs to the Tone and Multipart Unit in all LIMs in the PABX. The following hardware is required for the installation of Music on Hold:

• TMU board.
• cables.

If three different Music on Hold messages are to be used in the same LIM, two or more TMU boards are required.

11.42     Name and Number Log

See also the Extra facility description for NAME AND NUMBER LOG.

Name and Number Log is a software feature for logging unanswered calls. By use of the soft-keys on a DBC 203/213 digital telephone, the user can display, dial and delete the stored numbers.
A programmable key with a LED is used by this feature to inform the user that a new unanswered call has been logged.

11.43     Name Identity

See also the Extra facility description for NAME IDENTITY.

The Name Identity function is used to associate an easily recognisable name to various individual items that are normally identified in the system by a number which may not be generally known.

The function is also useful when applied to incoming and outgoing external calls where the external number exceeds the length of the field reserved for it in the display.

The calling/connected Name Identity function allows the assignment of a name to the following individuals:

• Analogue extensions (primary and secondary).
• CAS extensions (primary and secondary).
• Cordless extensions.
• Digital extensions (Own Directory Number and Additional Directory Number).
• Digital POTS extensions (primary and secondary).
• DNIS numbers.
• Dual access extensions.
• Groups (PBX, ACD, ANCD and Common Bell).
• Individual PABX operators.
• Ericsson IP telephone
• ISDN extensions.

The function also allows the name associated with an individual to be conveyed through the system, together with its number, so that both the number and name are displayed together. The Name Identity of the parties involved in a call is transferred. For external calls this is via an ISDN private or public network.

11.44   Network services

See also the Extra facility description for NETWORK SERVICES.

The Private Network Routing function has to be loaded in order to use intelligent network services in a private network. By itself it does not offer any intelligent network services, but it opens up the possibility for using them.

11.45   Operator Instrument for operator with impaired vision

See also the Extra facility description for EQUIPMENT FOR PABX OPERATORS WITH IMPAIRED VISION.

Ericsson provides a PABX operator console which permits operators with impaired vision to use the ASB 501 04. The console consists of a standard operator instrument, OPI-II, with supplementary functions for operators with impaired vision and can therefore also be used by operators with normal vision.

A User Guide in Braille is available.

11.46     Original A-number

See also the Extra facility description for ORIGINAL A-NUMBER.

The Original A-number feature allows the diverted-to party (the C-party) to see the calling party's (the A-party's) number in the following cases:

- the B-party has external follow me to the C-party activated.

- the C-party is an answer position in the B-party's active list for personal number.

11.47     Paging

See also the Extra facility description for PAGING.

11.47.1   General

The connection of peripheral equipment for Paging to the ASB 501 04 facilitates access by extensions, PABX operator consoles or external lines to persons not within hearing distance of their own extension telephone's ring signals but in the vicinity of a telephone.

Paging via telephone can be assigned to arbitrary telephones and individual PABX operator positions as well as a number of special directory numbers for Paging only. It is also possible to assign a secondary directory number to an extension user's own Paging code.

A paged person, i.e. a pagee, is made aware of the Paging by means of either a portable receiver (staff locator) or via common permanent visual and/or acoustic Paging units. Paging via telephone is usually answered from the nearest telephone by means of a special procedure whereby the pager and pagee receive speech connection.

In particular the use of individual Paging receivers can also be used for message transmission to the pagee. Usually the caller's directory number is sent to the display on the Paging receiver.

The answer procedure normally includes the pagee's directory number to permit call contact to be addressed to the correct calling party.

11.47.2   Function

Individual Paging

Each pagee has a Paging receiver of which there are three main types:

• Only the call and (possibly) type of call are indicated.
• The receiver also contains functions for reception, registration and indication of messages.
• The receiver also permits direct speech contact between pager and pagee.

For the PABX use of Individual Paging means that:

• Many Paging jobs can be ongoing concurrently in the same Paging sector.
• Several signals may need to be sent to and from the peripheral Paging equipment.
• Receiver type may need to be programmed per Paging individual.

Common Paging

Common Paging takes place via permanent visual displays placed in prominent positions, or via common loudspeakers that cover the entire Paging sector. For visual Paging the calls are normally answered by telephone while the pager waits (meet me Paging). For acoustic loudspeaker Paging the pager can request the pagee to call him and then go On-hook without waiting for an answer.

11.47.3   Connection to peripheral equipment

Serial connection PG1

For this connection the same hardware unit as for tie lines with E&M signalling is used. Each board possesses three connection channels. The signalling adheres to ESPA standard.

Parallel connection PG2

For this connection a special hardware unit, PGU2, with one connection channel per board is used. The paging code is signalled to the peripheral equipment via relay contacts, either decadically (3 x 10 + 1 contact) or BCD-coded (4 x 4 contacts).

Loop connection PG3

For this connection the same hardware unit as for external line connection to the PSTN is used.

11.48     Personal Number

See also the Extra facility description for PERSONAL NUMBER.

11.48.1   General

A Personal Number is an extension directory number with a terminal of any type assigned, which has the Repeated Deflection service available. The aim of Personal Number is to provide system users (voice extensions) with different possible answering positions for the incoming calls.

11.48.2   Functions

The Personal Number service has the following characteristics:

• Every Personal Number can have up to 5 different lists available, but only one of them can be active.
• Each list can be set up with up to 10 different answering positions that are selected depending on the user location (e.g. at the office, at home).
• The user can activate and deactivate the service or change the active list via a procedure.
• When the Personal Number service is activated, incoming calls to the Personal Number are deflected to the positions in the active list, until the call is answered or is stopped for any reason.
• When the Personal Number service is not activated, all incoming calls to the Personal Number are distributed to the assigned terminal as in a normal call.

11.49   Recorded Voice Announcement

See also the Extra facility description for RECORDED VOICE ANNOUNCEMENT.

11.49.1   General

There are two kinds of announcement machines used for Recorded Voice Announcement feature.

11.49.2   Functions

The RVA feature allows announcements to be supplied to incoming calls to a PABX operator (operator announcement). It also allows announcements to be supplied to a calling party, with the exception of PABX operator calls, data calls and calls from IP terminals, for the following call cases:

When there is no available RVA machine in an announcement group, the call is still processed but with no voice announcement being supplied to the calling party.

11.50     Repeated Individual Diversion, RID

See also the Extra facility description for REPEATED INDIVIDUAL DIVERSION.

11.50.1   General

The aim of the Repeated Individual Diversion is to provide users, i.e. voice extensions, with different possible answering positions for the incoming calls. Each individual extension directory number having a terminal assigned, can have RID applied.

This feature is standard for cordless and IP terminals.

11.50.2   Functions

The service has the following characteristics:

• Every user can have one list available which can be active or inactive.
• The list can be set up with up to 10 different answering positions that can be selected depending on the user location (e.g. at the office, at home).
• The user can activate or deactivate the service via a procedure.
• When the service is activated, incoming calls to the user are deflected to the positions in the list until the call is answered or stopped for any reason.
• When the service is not activated, all incoming calls to the user are distributed to the assigned terminal as in a normal call.

11.51     Short Message Service, SMS

See also the Extra facility description for SHORT MESSAGE SERVICE.

11.51.1   General

The DECT-SMS service makes it possible to send and/or retrieve SMS messages (paging) on the ASB 501 04 integrated DECT terminals to/from an external service centre (SMS-SC).
The ASB 501 04 gives a fully transparent information channel. It is the functionality of the SMS-SC and DECT-terminals that sets the functionality for the end user.