Splicecom Maximiser Installation And Reference Manual

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Installation & Reference Manual

Version 3.2

April 2010

Installation and Reference Manual

Document No. 001

Version No. V3.2/0410/6

SpliceCom Ltd

The Hall Business Centre, Berry Lane

Chorleywood, Herts WD3 5EX

Tel: 01923 287700

Website: www.splicecom.com

Installation and Reference Manual

Introduction ......................................................................................................................................................... 1

Platform Overview .............................................................................................................................................. 2

5 Series Hardware ........................................................................................................................................... 2

4 Series Hardware ........................................................................................................................................... 4

maximiser Handsets ....................................................................................................................................... 6

maximiser Software ...................................................................................................................................... 11

System Architecture .......................................................................................................................................... 15

Installing a System............................................................................................................................................. 21

System Operation ............................................................................................................................................. 24

Manager ........................................................................................................................................................ 24

Configuring System Details .......................................................................................................................... 26

Call Status ...................................................................................................................................................... 27

Call Logging .................................................................................................................................................. 27

Configuring the Call Server .......................................................................................................................... 27

Setting the IP Address of the Call Server ..................................................................................................... 28

Using DHCP .................................................................................................................................................. 29

System Time .................................................................................................................................................. 30

Setting up Administrative Access ................................................................................................................. 31

Using Auto Add Phones ............................................................................................................................... 34

Installing Licences......................................................................................................................................... 35

Connecting a Module .................................................................................................................................. 39

Connecting an Analogue Phone ................................................................................................................. 48

Connecting a PCS IP Phone ......................................................................................................................... 51

Installing PCS Software ................................................................................................................................. 58

Connecting Multiple Call Servers ................................................................................................................. 72

Configuring PBX functionality ........................................................................................................................... 77

Configuring an Analogue Extension Port .................................................................................................... 77

Configuring an IP Phone .............................................................................................................................. 79

Working with Users ...................................................................................................................................... 80

Working with Dial Plans .............................................................................................................................. 114

Working with Groups .................................................................................................................................. 122

Routing calls via a Department .................................................................................................................. 128

Setting up the Contacts Database ............................................................................................................. 141

Using a Time Plan ....................................................................................................................................... 148

Conferencing .............................................................................................................................................. 152

Working with Companies ........................................................................................................................... 155

Working with Trunks ................................................................................................................................... 159

Working with DDI Plans .............................................................................................................................. 169

Working with LCR Plans .............................................................................................................................. 175

Using Area Codes ....................................................................................................................................... 177

Configuring Music Channels...................................................................................................................... 177

Using Events ............................................................................................................................................... 181

Using the TAPI Driver .................................................................................................................................. 183

Internal Web Server......................................................................................................................................... 184

Working with Voicemail .................................................................................................................................. 187

Voicemail Ports ........................................................................................................................................... 188

Licensing ..................................................................................................................................................... 189

Enabling Voicemail for a User .................................................................................................................... 190

Contents Installation and Reference Manual v3.2/0410/6 i

Installation and Reference Manual

Installation and Reference Manual v3.2/0410/6

Leaving a Message ..................................................................................................................................... 192

Accessing Voicemail ................................................................................................................................... 193

Remote Access to Voicemail ...................................................................................................................... 202

Setting up Voicemail Email ......................................................................................................................... 207

Setting up SMS Alerts ................................................................................................................................. 209

Using voicemail with a Paging Port ........................................................................................................... 209

Voicemail for a Department ....................................................................................................................... 209

Creating an Auto Attendant ....................................................................................................................... 222

Creating VXML Scripts ................................................................................................................................. 225

Call Recording ............................................................................................................................................. 232

Maintaining and Troubleshooting Voicemail ............................................................................................ 237

Importing and Exporting the Configuration .................................................................................................. 240

Data Routing ................................................................................................................................................... 247

Creating a WAN Link ................................................................................................................................... 247

Setting up IP Routes ................................................................................................................................... 248

Setting up PPP ............................................................................................................................................. 248

Setting up Bandwidth on Demand (BOD) ................................................................................................ 249

Maintenance & Troubleshooting ................................................................................................................... 250

Utilities ........................................................................................................................................................ 250

PHP Utility Pages ......................................................................................................................................... 251

System Access ............................................................................................................................................. 251

Reset Switch ................................................................................................................................................ 253

Powering down a Call Server ..................................................................................................................... 255

Rebooting a module .................................................................................................................................. 255

Backing up the Database ........................................................................................................................... 255

Upgrading the System ................................................................................................................................ 258

Initial Configuration Boot Menu ................................................................................................................ 267

Accessing the log files ................................................................................................................................ 268

Using Setnet ............................................................................................................................................... 269

Using Telnet ................................................................................................................................................ 270

Field Descriptions ............................................................................................................................................ 272

Manage Company ...................................................................................................................................... 272

Users ........................................................................................................................................................... 272

Departments ............................................................................................................................................... 282

Groups ......................................................................................................................................................... 288

Time Plans ................................................................................................................................................... 289

Contacts ...................................................................................................................................................... 290

Meet Me Conference ................................................................................................................................. 292

Dial Plans ..................................................................................................................................................... 293

Companies .................................................................................................................................................. 299

Trunks .......................................................................................................................................................... 300

Trunk Groups .............................................................................................................................................. 304

DDI Plans ..................................................................................................................................................... 304

LCR Plans ..................................................................................................................................................... 305

Voicemail Ports ........................................................................................................................................... 305

Music Channels .......................................................................................................................................... 307

Events .......................................................................................................................................................... 308

VXML Scripts ................................................................................................................................................ 308

Auto Attendant ........................................................................................................................................... 310

Modules ...................................................................................................................................................... 310

Phones ........................................................................................................................................................ 316

Contents ii

Installation and Reference Manual

Unassigned Phones ................................................................................................................................... 319

Unassigned Modules ................................................................................................................................. 319

System ......................................................................................................................................................... 319

Licences ...................................................................................................................................................... 320

Administrators ............................................................................................................................................. 320

Utilities ........................................................................................................................................................ 321

Area Codes .................................................................................................................................................. 322

WAN Links ................................................................................................................................................... 322

Technical Data ................................................................................................................................................. 326

Cable Specifications ................................................................................................................................... 328

Relay Outputs ............................................................................................................................................. 330

Event Inputs ................................................................................................................................................ 331

PCS 580G Technical Details ........................................................................................................................ 332

PCS 570G Technical Details ........................................................................................................................ 332

PCS 570/560 Technical Details ................................................................................................................... 333

PCS 410 Technical Details ........................................................................................................................... 334

PCS 400 Technical Details ........................................................................................................................... 335

PCS 100 Technical Details ........................................................................................................................... 336

PCS 520 Technical Details ........................................................................................................................... 337

PCS 10 Technical Details ............................................................................................................................. 337

PCS 5 Technical Details ............................................................................................................................... 338

Safety Caution ............................................................................................................................................. 338

PCS 505 Technical Details ........................................................................................................................... 339

Voice Compression Card Technical Details ............................................................................................... 339

“Music on Hold” Adapter Technical Details .............................................................................................. 339

“Paging Port” Adapter Technical Details .................................................................................................... 340

Trunk NT Mode Dongles ............................................................................................................................ 341

Glossary ........................................................................................................................................................... 344

Index ................................................................................................................................................................ 349

Contents Installation and Reference Manual v3.2/0410/6 iii

Installation and Reference Manual

Introduction

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The

imiser business telephony system from SpliceCom provides a breakthrough in integrated voice communications. Developed from state of the art technology it delivers real life benefits associated with many traditionally separate components in one single, seamless system, currently supporting 4 to 10,000 IP or analogue extensions. When used in conjunction with SpliceCom’s broad range of Proactive Communication Station (PCS) phones or applications, telephone system with their core business applications, so ensuring that the right information can be delivered to the desktops of the right people at the right time.

Through the use of an innovative architecture,

imiser eliminates the physical and geographical limitations of traditional telephone systems, allowing great savings to be made on administration, management and infrastructure costs, through the unification of resource. This approach allows all businesses, irrespective of size, to benefit from extended communications, and more importantly protects their initial investment by growing with the business as the need for communication scales and becomes ever more demanding.

Embracing open standards wherever they exist, communications as its core. This allows it to be deployed as a simple PBX with sophisticated, yet easy to use call handling facilities. Alternatively,

imiser can be used to deliver voice as just another

application – albeit a very business critical one – in a fully converged IP network. In the latter case,

imiser’s modular components can be distributed throughout a building, spread across a campus or even located in different locations, yet still operate and be managed as a single integrated system. With its ability to integrate seamlessly with multiple applications, independent of platform or location,

imiser delivers the blueprint for business telephony in the 21st Century.

The purpose of this manual is to give a step by step guide on how to install, configure and troubleshoot a

imiser system. The manual starts with a feature guide of each element of a system together with an understanding of the architecture used and finishes with a complete listing of each configuration field and its function together with in-depth technical data.

Please note that where the phrase “Call Server” is used this refers to the 5100 Call Server, the 5108 Call Server, the 4100 Call Server and the 4140 Remote Call Server unless specified.

imiser allows businesses to converge their

imiser utilises a Linux operating system and IP

Introduction Installation and Reference Manual v3.2/0410/6 1

Installation and Reference Manual

Installation and Reference Manual v3.2/0410/6

Platform Overview

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5 Series Hardware

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5100 Call Server

switch

The

imiser system provides a single telephony system for small-to-medium sized companies with support for 4 to 10,000 extensions. It is a purely IP system supporting either traditional analogue or IP Phones.

The 5 series

imiser range consists of the 5100 Call Server, 5300 Phone and 5108 Call Server. Each

module is 1u in height and can be mounted in a 19” rack.

The main control unit of the system, this module provides:

• PBX

• Quad BRI (8 channels) / NT ports (2 ports enabled by

licences)

• 2 x PRI (30 channels) / NT / DPNSS ports (1 channel

on each port enabled, the remaining channels can be enabled by licences)

• 1 x LAN Link port, 10/100 Mbps Full Duplex Ethernet

port for connectivity to existing LAN or dedicated LAN

• 4 x 10/100 Mbps Ethernet ports, integral QoS LAN switch, auto-sensing for MDI/MDIX connectivity and

802.3af Power over Ethernet supported

• 16 x standard analogue (POTS) ports with support for DTMF analogue handsets, PCS 510/505/10/5, fax

machines and modems plus Caller Display support (8 ports enabled by licences)

• IP Router

• H.323 Gatekeeper

• H.323 Gateway

• Hard disk storage for software, LDAP database (supporting 10,000 configuration entries), voicemail,

internal Web Server, Music on Hold and feature licenses

• 1 x power socket for 48 volts PSU supplied as standard

• 2 x Relay inputs for eg burglar alarm

• 2 x External relays for eg door entry system

• 1 x USB port for software upgrade purposes

• Supports H.450.2 (Call Transfer), H.450.4 (Hold), H.450.5 (Park) and H.450.7 (Message Waiting)

• SIP Proxy Server support

• Supports SIP and H.323 trunk services

• Supports NAT Traversal

• Native mode – G.711 64K with echo cancellation

• Can support up to 68 x G.729a Compression channels (requires licences)

• Supports 500 Users (analogue and/or IP)

• Up to 200 Call Servers per system (max 10,000 extensions, analogue and/or IP and max 5,000 trunks)

Platform Overview 2

Installation and Reference Manual

5300 Phone

5108 Call Server

This module provides connection for standard analogue telephones:

• Available as 5315 providing 15 POTS ports or 5330

providing 30 POTS ports

• Support for DTMF analogue handsets, PCS 5/10, fax

machines and modems

• Caller Display support

• On-hook voltage 48V, off-hook current 25 mA, ring

voltage 44V rms

• 4 x 10/100 Mbps Ethernet ports, integral QoS LAN switch, auto-sensing for MDI/MDIX connectivity and

802.3af Power over Ethernet supported

• 1 x LAN Link port, 10/100 Mbps Full Duplex Ethernet port for connectivity to existing LAN or dedicated

LAN switch

• 1 x USB socket, allows wireless LAN connectivity via USB 2.0 WiFi adapter.

• 1 x power socket for 48 volts PSU supplied as standard

• Provided with PCS 60 or PCS 50 application to partner an analogue handset from a Windows or Mac

OSX PC

The 5108 Call Server is designed either for the smaller business or the small remote office as part of a larger network. This module provides:

• PBX

• Dual BRI (4 channels) / NT ports (1 port enabled by licence)

• 4 x standard analogue (POTS) ports with support for DTMF

analogue handsets, PCS 5/10, fax machines and modems plus Caller Display support

• IP Router

• 1 x LAN Link port, 10/100 Mbps Full Duplex Ethernet port for

connectivity to existing LAN or dedicated LAN switch

• 4 x 10/100 Mbps Ethernet ports, integral QoS LAN switch, auto-sensing for MDI/MDIX connectivity and

802.3af Power over Ethernet supported

• H.323 Gatekeeper and H.323 Gateway

• Hard disk storage for software, LDAP database (supporting 10,000 configuration entries), voicemail,

internal Web Server, Music on Hold and feature licenses

• 1 x Relay inputs for eg burglar alarm

• 1 x External relays for eg door entry system

• 1 x power socket for 48 volts PSU supplied as standard

• Supports H.450.2 (Call Transfer), H.450.4 (Hold), H.450.5 (Park) and H.450.7 (Message Waiting)

• SIP Proxy Server support

• Supports SIP and H.323 trunk services

• Supports NAT Traversal

• Native mode – G.711 64K with echo cancellation

• Can support up to 8 x G.729a Compression channels (requires licences)

• Supports 8 Users (or 12 Users with a 5108Plus licence – 4 analogue and 8 IP phone users, or 12 IP

phone users)

Platform Overview Installation and Reference Manual v3.2/0410/6 3

Installation and Reference Manual

Installation and Reference Manual v3.2/0410/6

4 Series Hardware

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4100 Call Server

4200 Trunk

The 4 series

imiser system is made up of 3 components – 4100 Call Server, 4200 Trunk and 4300 Phone. The 4140 Remote Call Server combines the functionality of all three of these modules in a single unit. Each module is 1u in height and can be mounted in a 19” rack.

The main control unit of the system, this module provides:

• PBX

• Quad BRI (8 channels) / NT ports

• Optional single PRI (30 channels) / NT / DPNSS port

• IP Router

• 8 port QoS 10/100 Base-T switch

• Rear 10/100 LAN port

• WAN port V.11 - megastream 2 Mb

• H.323 Gatekeeper

• H.323 Gateway

• Hard disk storage for software (10 GB), LDAP database – supporting 10,000 User entries, voicemail,

Internal Web Server, Music on Hold and feature licenses

• 48 volts PSU (so can purchase commercial UPS)

• 2 x Relay inputs for eg burglars alarms

• 2 x External relays

• Supports H.450.2 (Call Transfer), H.450.4 (Hold), H.450.5 (Park) and H.450.7 (Message Waiting)

• SIP Proxy Server support

• Supports SIP and H.323 trunk services

• Supports NAT Traversal

• LAN PWR port to provide Power over Ethernet (PoE) for the PCS 400/100.

• Native mode – G.711 64K with echo cancellation

• Optional Compression card supporting G.729 8K for WAN links

• Supports 300 Users (analogue and/or IP)

• Supports 4 Trunk modules

• Up to 100 Call Servers and Trunk modules per system (max 5,000 extensions, analogue and/or IP)

This module provides additional ISDN and WAN ports:

• Quad BRI (8 channels) / NT ports

• Optional single PRI (30 channels) / NT / DPNSS port

• WAN port V.11 - megastream 2 MB

• 8 port QoS 10/100 Base-T switch

• Integral H.323 Gateway

• IP Router

• LAN Power port for Power over Ethernet

• Up to 4 Trunk Modules per Call Server

• Up to 100 Trunk Module and Call Servers per system

Platform Overview 4

Installation and Reference Manual

4300 Phone

4140 Remote Call Server

This module provides connection for standard analogue telephones:

• Available as 4315 providing 15 POTS ports or 4330

providing 30 POTS ports

• Support for DTMF analogue handsets, PCS 5/10, fax

machines and modems

• Caller Display support

• On-hook voltage 48V, off-hook current 25 mA, ring

voltage 44V rms

• 10/100 LAN port for connection to network

• PCMCIA slot for wireless LAN connection

• Provided with PCS 60 or PCS 50 application to partner an analogue handset from a Windows or Mac

OSX PC

The 4140 Remote Call Server is designed either for the smaller business or the small remote office as part of a larger network.

• PBX

• 2 x BRI (4 channels) / NT ports

• Optional additional 2 x BRI (4 channels)/NT ports

• Optional PRI (15 channels) / NT / DPNSS port

• 8 x POTS ports

• IP Router

• 1 port QoS 10/100 Base-T switch

• IP WAN port V.11 - megastream 2 Mb (for connection to a Call Server only)

• H.323 Gatekeeper

• H.323 Gateway

• Hard disk storage for software (10 GB), LDAP database, voicemail, Internal Web Server and feature

licenses

• 48 volts PSU (so can purchase commercial UPS)

• Supports H.450.2 (Call Transfer), H.450.4 (Hold), H.450.5 (Park) and H.450.7 (Message Waiting)

• SIP Proxy Server support

• Supports SIP and H.323 trunk services

• Supports NAT Traversal

• Native mode – G.711 64K with echo cancellation

• Optional Compression card supporting G.729 8K for WAN links

• Supports 40 Users (8 analogue plus a further 32 analogue and/or IP)

Platform Overview Installation and Reference Manual v3.2/0410/6 5

Installation and Reference Manual

Installation and Reference Manual v3.2/0410/6

4400 Voice Compression Module

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Handsets

PCS 580G

As an alternative to the 64kbps, G.711 voice supported as standard on maximiser, 8kbps, G.729a based encoding, provides an excellent balance between voice quality and bandwidth efficiency. The 4400 Voice Compression Module provides 16 channels, of 8 kbps, G.729a based compression and can be deployed to provide a higher density of calls between 4100 Call Servers/4140 Remote Call Servers over low speed WAN links.

• 1 x RJ45 port, dual speed, 10/100 Mbps FDX Ethernet

interface with integral LEDs for Link and Data.

• 1 x 48Vdc Power jack.

• Supports G.729a

• Additional 16 channels enabled via a licence, giving

a maximum of 32 compression channels per module

imiser

This IP phone provides:

• Standard telephony functionality

• Graphical colour touch screen LCD interface

• Context Sensitive Screen

• Ability to display real time video and graphical information

during call

• Power - 802.3af Power over Ethernet

• 2 Port 10/100/1000 Mbps FDX LAN switch – supporting

automatic MDI-MDIX crossover, Diffserv Quality of Service,

802.1q VLAN and 802.3af Power over Ethernet

• 2 x USB port – USB 2.0 interface for WiFi and external

keyboard

• Headset connection

• Wide-angle tilting keypad/display panel

• Message waiting/do not disturb “S” LED

• Dual Mode – SIP (configured via the Call Server) & H.323

• Fully integrated with the LDAP system database for viewing and annotating customer records and

notes

• Integral web browser

• Can be wall mounted

• Can be partnered with PCS 60 or PCS 50

Platform Overview 6

Installation and Reference Manual

PCS 570

PCS 570G

PCS 560

This IP phone provides:

• Standard 12 button telephony functionality

• Full colour, backlit, graphics display (240 x 320) with auto-

dimming, context sensitive

• Wide-angle tilting keypad/display panel

• 18 intuitive, multi-functional context sensitive keys

• 10 fixed function keys

• Power - 802.3af Power over Ethernet

• 2 Port 10/100 Mbps FDX LAN switch – supporting

automatic MDI-MDIX crossover, Diffserv Quality of Service,

802.1q VLAN and 802.3af Power over Ethernet

• USB port – USB 2.0 interface for WiFi and external

keyboard

• Headset connector

• Message waiting/do not disturb “S” LED

• Dual Mode – SIP (configured via the Call Server) & H.323

• Fully integrated with the LDAP system database for access to the directories of Users, Departments

and Contacts.

• Can be wall mounted

• Can be partnered with PCS 60 or PCS 50

This IP phone provides the same functionality as the PCS 570 except it has a Gigabit interface providing a 2 Port 10/100/1000 Mbps FDX LAN switch.

This IP phone provides:

• Standard 12 button telephony functionality

• Full colour, backlit, graphics display (240 x 320) with auto-

dimming, context sensitive

• Wide-angle tilting keypad/display panel

• 9 intuitive, multi-functional context sensitive keys

• 10 fixed function keys

• Power - 802.3af Power over Ethernet

• 2 Port 10/100 Mbps FDX LAN switch – supporting automatic

MDI-MDIX crossover, Diffserv Quality of Service, 802.1q VLAN and 802.3af Power over Ethernet

• Headset connector

• Message waiting/do not disturb “S” LED

• Dual Mode – SIP (configured via the Call Server) & H.323

• Fully integrated with the LDAP system database for access to the directories of Users, Departments

and Contacts.

• Can be wall mounted

• Can be partnered with PCS 60 or PCS 50

Platform Overview Installation and Reference Manual v3.2/0410/6 7

Installation and Reference Manual

Installation and Reference Manual v3.2/0410/6

PCS 410

PCS 400

This IP Phone provides:

• Standard telephony functionality

• Graphical touch screen LCD interface

• Context Sensitive Screen

• Ability to display real time video and graphical information

during call

• Power over Ethernet support – can be powered via a LAN

port on the Call Server, via the network, or via the optional dedicated PSU

• 2 Port 10/100 Mbps FDX LAN switch

• USB port for connection of mouse and keyboard

• Internal USB for WiFi

• Headset connection

• Fully integrated with the LDAP system database for viewing and annotating customer records and

notes

• Integral web browser

• Can be wall mounted

• Can be partnered with PCS 60 or PCS 50

This IP Phone provides:

• Standard telephony functionality

• Graphical touch screen LCD interface

• Ability to display real time video and graphical

information during call

• Context Sensitive Screen

• Power over Ethernet support – can be powered via

the network, via the Powered Ethernet PSU or optional dedicated PSU

• Provides LAN port for PC

• 802.11b PCMCIA Card Slot for Wireless LAN

• USB port for connection of mouse and keyboard

• Headset connection

• Fully integrated with LDAP system database for viewing and annotating customer records and notes

• Integral web browser

• Can be wall mounted

• Can be partnered with PCS 60 or PCS 50

Platform Overview 8

Installation and Reference Manual

PCS 100

PCS 520

The PCS 100 is an IP Phone presented as a conventional business telephone. It provides caller ID, a dedicated Busy Lamp Field together with pre programmed buttons for simple and easy access to system features.

• Standard 12 button telephony functionality

• Display of number/name

• 64 x 128 Context Sensitive LCD Display

• 8 Dynamic, Context Sensitive Keys

• Headset connection.

• Powered via the LAN or optional STEPS power

supply

• Fully integrated with system database for viewing

and speed dialling users, departments, contacts and favourites

• 2 Port QOS switch, single cable to desk.

• Can be wall mounted

• Enhanced volume and speech quality of incoming calls with Automatic Gain Control (AGC).

• Can be partnered with PCS 60 or PCS 50

The PCS 520 is an analogue handset providing the following features:

• Standard 12 button telephone keypad

• 3 line LCD display with contrast settings

• Display of Calling Line Number/Name & Called Line Name

• Dual message waiting/DND indicator

• Visual ringing indicator

• 10 pre-programmed keys

• 10 user definable keys

• Internal directory (100 names/numbers)

• Missed call/Clear/Dial/Pause/Recall/Redial/Speaker/Volume

/Mute keys

• Voicemail guide on handset

• Desktop paging

• Auto Answer

• Headset port

• Powered via the line port

• Can be partnered with PCS 60 or PCS 50

Platform Overview Installation and Reference Manual v3.2/0410/6 9

Installation and Reference Manual

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PCS 10

PCS 505

Can be partnered with the PCS 60 or PCS 50

PCS 5

Power over Ethernet PSU

The PCS 10 is an analogue handset providing the following features:

• Standard 12 button telephone keypad

• 3 line LCD display with contrast settings

• Display of Calling Line Number/Name & Called Line Name

• Message Waiting icon

• 10 pre-programmed keys

• 10 user definable keys

• Internal directory (100 names/numbers)

• Missed call/Clear/Dial/Pause/Recall/Redial/

Speaker/Volume/Mute keys

• Headset port

• Powered via a dedicated PSU or with batteries

• Can be partnered with PCS 60 or PCS 50

The PCS 505 is an analogue handset providing the following features:

• Standard 12 button keypad

• Visual call indicator/Message Waiting Lamp

• Headset socket

• Hold/Recall key

• Last number Redial key

• Microphone Mute key

• Headset key

• System Feature Guide

• Voicemail management keys

•

The PCS 5 is an analogue handset providing the following features

• Standard 12 button keypad

• Visual call indicator/Message Waiting Lamp

• Recall/Redial/Mute/Pause keys

• System Feature Guide

• Voicemail management keys

• Can be partnered with the PCS 60 or PCS 50

This is an external power supply for use with the 4100 Call Server and 4200 Trunk Module:

• Providing IP phones, including the PCS 400/100, with power over the LAN when connected to the

integral 10/100 Mbps switch.

Platform Overview 10

Installation and Reference Manual

PCS PSU

Single Terminal Ethernet Power Supply (STEPS)

Voice Compression Card

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Software

PCS 60

• It is connected to LAN PWR port on the back of the 4100 Call Server and/or 4200 Trunk Module

• The LAN PWR port provides power to 2 spare pairs on Ethernet ports

• Up to 8 IP Phones can be powered via the Powered Ethernet PSU.

• Supports the 802.3af Power over Ethernet standard

This is an external power supply providing the PCS 400 IP phones with local power for applications not using power over LAN, eg Wireless LAN connected phones.

This is an External Power over Ethernet PSU for the PCS 580, 570, 560, 410, 400 and100 or other 802.3af compliant devices.

Please note:

• if a PCMCIA card is required in the PCS 400 a standard PCS PSU must be used.

• If used with a PCS 580G or PCS 570G the LAN port (port 1) will not run at 1G.

imiser supports G.711 encoding with echo suppression as standard for transporting voice. Where compressed voice is a requirement, G.729a between Call Servers is supported through optional internal Voice Compression Cards, one fitted at each end. This feature will support up to 8 channels.

Please refer to page 326 for further technical data on these products.

imiser

This PC application provides the follow facilities:

• Standard telephony functionality

• CLI display

• Context sensitive toolbar

• Fully integrated with the system database for

speeding dialling, viewing and annotating customer records and notes

• Ability to display real time video and graphical

information during a call

• Operator Console Mode available with an Operator

Console licence

• Can be run as either an IP soft phone or as a

partner to a PCS 580, 570, 560, 410/400, 100, 10, 5 or an alternative analogue handset.

• Available for MS Windows and Apple Mac OS X v10.4 or above

Please refer to the PCS 60 section from page 58 for further details.

Platform Overview Installation and Reference Manual v3.2/0410/6 11

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PCS 50

PCS Operators Console

Manager

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Voicemail

This PC application provides the same telephony functionality as available on the PCS 410/400. PCS 50 can run as an IP Soft phone or be programmed to partner a User’s PCS 5/10/100/560/570/580 or existing analogue phone from their PC.

Versions for Windows and Mac OS X are currently available.

Please refer to the PCS 50 section from page 69 for further details.

This application combines all of the main PCS 50 functions on a single full screen page for operators and receptionist. Like the PCS 50 from which it is derived it can be run as an IP soft phone or as a Partner to PCS 580, 570, 560, 410/400, PCS 100, PCS 10, PCS 5 or existing analogue phones.

Windows and Mac OS X versions are available.

Please refer to the PCS Operator Console section from page 71 for further details.

This is a web-based application that is used to view and configure the database. It can be run from any web browser (IE v5 or higher, Netscape v6 or higher, Apple Safari) that can connect to the

Please refer to the Manager section from page 24 for further details.

The 5100, 5108, 4100 Call Servers and 4140 Remote Call Server all provide integral voicemail & AutoAttendant facilities.

imiser.

Platform Overview 12

Installation and Reference Manual

Enhanced Speech Processing (ESP)

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• A 5100 Call Server provides up to 16 simultaneous connections, 4 of which are licensed on shipment,

and 1500 hours of message storage

• A 5108 Call Server provides up to 8 simultaneous connections, 2 of which are licensed on shipment,

and 500 hours of message storage

• A 4100 Call Server provides up to 8 simultaneous connections and 30 hours of message storage

• A 4140 Remote Call Server provides up to 4 simultaneous connections and 10 hours of message

storage

This voicemail and Auto-Attendant application can also be run on a standalone Linux PC or Mac PC where more simultaneous connections or greater storage capacity is required.

A MessageBox licence is required for each User and Department requiring a voicemail facility. A Voice Processing Port licence is required for each additional simultaneous connection.

Please refer to the Working with Voicemail section from page 187 for further details.

Enhanced Speech Processing is SpliceCom’s advanced multi-layer auto-attendant and Interactive Voice Response (IVR) system for

imiser. Licensed on a per-port basis ESP utilises standards based VoiceXML to deliver great flexibility and interoperability for voice processing requirements. ESP allows you to answer calls, play files, collect DTMF digits, record files and transfer calls. Features and functionality supported by ESP include;

• Multi-Layer Auto Attendant

• Dial-by Extension

• Voice Forms (DTMF digit and spoken response capture)

• Text-To-Speech (TTS) Conversion

• Reminder/Wake-Up/Alarm Calls

• IMAP Unified Messaging

• Spoken Email Collection & Reply

• Queue Buster

• Call Back When Free

For further information please refer to the Introduction to ESP document available within the Partner area of the SpliceCom website.

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Vision

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SpliceCom Vision is a web based application suite developed from the ground up by SpliceCom to work with

imiser. Utilising the latest AJAX and Web 2.0 technologies, Vision has been designed to deliver business critical information, in an easy to understand manner, wherever and whenever it’s needed. Vision offers four services; Reports (historical), Recording (capture) , Live (real-time) & Mobility (freedom).

For full details on how employ this application suite with your the Vision Installation and Configuration Manual.

imiser – and to only work with

imiser system please refer to

Platform Overview 14

Installation and Reference Manual

System Architecture

Working with the System

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The system operates entirely around a LAN/IP network environment, eliminating any previous geographic constraints imposed upon similar systems. The system is constructed with several distinct and discrete components, all physically independent and operating symbiotically to provide a single seamless system.

All

imiser modules are linked via TCP/IP. Each module is a host on the network with its own IP address. Therefore each module can be connected via standard CAT 5 cabling, leased line, or Wireless LAN networking in the case of the Phone Module and the PCS 400. This allows for multi-site networking with the advantage of only one system to configure & manage – from anywhere. The configuration of the system is stored on the database which resides on each Call Server; this can be accessed via Manager. Manager is a web-based application so can be run from any web browser that has an IP connection to the system.

As the standard for Voice over IP, is supported. This means that the Call Server acts as the H.323 Gateway. A gateway is responsible for translating one transmission type into another, so in the case of the Call Server, for example, it will translate calls received via ISDN so they can travel over an IP network and vice a versa. The Trunk Module will also provide this functionality. The Call Server also acts as the H.323 Gatekeeper and is responsible for controlling the access into and out of the network – phone registration, call routing and call logging etc. Quality of Service, also managed by the Call Server, will ensure that Voice packets have priority over Data packets. This will make sure that a conversation is not broken up or interrupted.

The configuration of the system is stored on the LDAP database held on the hard disk contained within the Call Server. LDAP (Lightweight Directory Access Protocol) is the protocol used to store information on a database that can be searched and queried over a network by any number of users. In the case of the

users to view telephone directories and contact information via a PCS 580, 570/560, PCS 410/410, PCS 100, PCS 60 and PCS 50. As LDAP is the industry standard for accessing databases future applications will include access to third party databases.

imiser system is a purely IP system all calls are carried over IP therefore H.323, the industry

imiser system LDAP allows the configuration to be viewed and amended via Manager and allows

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System Directory

Primary Mode

Secondary Mode

Gatekeeper

Phone Module

Call S erver

Configured with Manager

Cat 5/5e/6

Leased Line

1st Floor

Satellite Site

2nd Floor

Ethernet

via standard Web Browser connected directly to Call Server, or via LAN, IP WAN, VPN etc

The system is powered by a centrally replicated but distributed information directory. In essence this allows components of the system that are on different continents to act and operate in the same way as components only inches away from the central unit.

All systems access this database using LDAP (the internet standard for accessing databases) for both reading and writing to the information.

Call Servers are responsible for holding, distributing and replicating this information. All other modules must have a direct connection via a LAN to a Call Server.

There are two different scenarios that are constructed:

The Call Server that acts as the Primary administration module and holds the master database for the entire system. This Call Server is responsible for replicating this database to other Secondary Call Servers.

Secondary Mode indicates that this Call Server will receive a replica of the database from the Primary Call Server. Any local changes are passed back to the Primary for distribution to all other Secondary Call Servers.

In all of the above cases non-administrative modules, ie Phone and Trunk modules, obtain their configuration from a local Call Server, thus they do not need to be configured directly. This allows live deployment of new modules with minimum complexity.

All Trunks and Phones register with the Gatekeeper in the usual H.323 secure methods. The Gatekeeper is responsible for all Phone Registrations, Call Routing and Call Logging.

System Architecture 16

Installation and Reference Manual

Trunks

Distributed Phone Modules

Computer Telephony

Distributed Voicemail

Although communication peers (eg phones) pass the voice data directly between them, the Gatekeeper will control all of the call routing. The Gatekeeper therefore does not require high performance data capabilities.

The Call Server runs Gatekeeper functionality.

Trunk gateways are registered with the Gatekeeper and act as a resource for the Gatekeeper to route calls to and from the public PSTN networks. Trunks can be on physically separate modules and geographic sites.

Clock synchronisation is performed by a combination of buffering and “Clock Count” signalling between the master system and all slaves.

Call Servers and Trunk Modules run Gateway functionality.

As each Phone Module is LAN connected it can now be distributed geographically within a building or separate buildings and will only require one cable between the module(s) and the administration system. This is particularly useful when wiring different floors of a building; only a small number of vertical risers are needed. Each 100 Mpbs FDX Ethernet connection is capable of handling between 390-780 concurrent calls, assuming no other traffic. Where non-QoS (Quality of Service) Switches are deployed then it is recommended that Voice be separated from normal LAN traffic, each Call Server and Trunk module contains a QoS Switch to allow such segregation. In normal circumstances (simple telephony) it is recommended that the Ethernet cabling be treated as a new generation of flexible voice cabling and independently connected, especially with the use of structured wiring and patch panels.

Because each Phone Module is geographically independent so is the Phone oriented computer telephony. Each Phone Module is responsible for its own handling of a third party TAPI interface.

Additionally the central administrator can have first party telephony ports via TAPI.3.0 and H.323. This is the method used for external voicemail.

Because voicemail utilises H.323, voicemail platforms can be operating system independent and located in any appropriate location. Because of the multi-port registration philosophy voicemail can be spread across many machines increasing the potential capacity of the system. This is especially important when SAPI (speech recognition) is required.

Call Servers are capable of running an Auto attendant and voicemail service as standard. Voicemail can also run on a Linux server and will handle storage limited only by the disk space of the machine (approx

0.5 Mbytes per minute).

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Extendibility

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Manageable Objects

Users

Departments

Groups

Contacts

Meet Me Conferences

Dial Plans

Time Plans

Although the

imiser platform provides primary applications; Proactive Communication Station, voicemail etc, the use of H.323 and LDAP means that independent vendors can hook into the system extending the range of options available. This is especially important in the Call Centre market.

The internal LDAP database contains the following configurable objects.

Each user can make use of the system for communications. These are the primary parameters for all of the telephony facilities that a user may have. They are distinct from the phone parameters since a user may login at any phone. Please refer to the Working with Users section from page 80 for further details.

Departments are used to determine the distribution of calls to a particular group of Users. A caller can dial one number and any available member of a Group can answer the call. A Department is used to determine what will happen to the call out of hours, if the members of the Group are unavailable or busy. Please refer to the Routing calls via a Department section from page 128 for further details.

Groups are collections of users and/or capabilities that can be used to identify groups of users. A distribution group can contain just one user, a thousand users or no users. Alternatively it may have a capability requirement to automatically select users with a particular skill eg French 80%. Groups may contain other groups to form “super groups”. Please refer to the Working with Groups section from page 122 for further details.

The Contacts database provides directory lookup information of, eg customers and suppliers. An incoming call will be alpha tagged if a CLI match is found in the Contacts database and will be displayed on the PCS, or caller display of the phone. Contacts may also be marked with account codes for billing and tracking purposes. Please refer to the Setting up the Contacts Database section from page 141 for further details.

A Meet Me Conference provides the facility to create a conference that can be joined by Users as and when required. Please refer to the Conferencing section from page 151 for further details.

Dial Plans determine how digits dialled on a User’s handset are handled. Please refer to the Working with Dial Plans section from page 114 for further details.

Time Plans provide a positive restriction of when activities may happen. In essence a Time Plan defines when things are allowed to operate by day of the week and time of the day. Please refer to the Using a Time Plan section from page 148 for further details.

System Architecture 18

Installation and Reference Manual

Companies

Trunks

Trunk Groups

DDI Call Plans

LCR Plans

Voicemail Ports

Music Channels

Events

VXML Scripts

Users, Departments, Contacts and Meet Me Conferences can be assigned to different Companies so that in a multi-tenancy system Users are unaware of the usage of the system by other businesses. An Auto Attendant can be assigned to each Company. Please refer to the Working with Companies section from page 155 for further details.

Trunks are the method used for outgoing or incoming communication with the system whether via ISDN, SIP, or H.323. Each Trunk port available on the system can be configured individually. Please refer to the Working with Trunks section from page 159 for further details.

A Trunk group allows the grouping together of multiple Trunks for delivery of calls. This is very useful in geographically organising the flow of calls, and can also be used to differentiate levels of service, by keeping spare capacity for high priority calls. Please refer to the Working with Trunks section from page 161 for further details.

Direct Dial In Call Plans determine how incoming ISDN calls will be routed. Please refer to the Working with DDI Plans section from page 169 for further details.

LCR Plans apply to all calls that will be routed to a trunk via a gateway. It allows the best selection of a trunk and selection of service providers, based upon the number dialled and time of day. It can also be used to control the geographic exit point from the network, eg all calls to London numbers would exit via a London trunk module, but calls to Edinburgh would exit via the Edinburgh trunk module. If a trunk should be busy or fail then alternate translations and trunks would be used. Please refer to the Working with LCR Plans section from page 175 for further details.

A Voicemail Port determines the facilities provided by a Call Server’s voicemail service. Where more than one Call Server is active on a system Voicemail Ports allow different capabilities to be assigned to each Call Server providing a voicemail service. Please refer to the Working with Voicemail section from page 188 for further details

Music Channels determine how music on hold is played when external calls are placed on hold either from an internally stored WAV file or via an external source. Please refer to the Configuring Music Channels section from page 177 for further details.

Events allow web pages to be sent to a PCS 60, PCS 50 or PCS 410/400 to display warning messages, company information etc. A Relay Input port or a Dial Plan entry can activate an Event. Please refer to the Using Events section from page 181 for further details.

VXML Scripts provide the ability to create multi-level auto-attendants incorporating features such as availability based routing, dial by extension, out of hours routing and text-to-speech. Please refer to the Creating VXML Scripts section from page 225 for further details.

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Auto Attendant

Modules

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Phones

Unassigned Phones

Unassigned Modules

System

Licences

Administrators

Utilities

Area Codes

WAN Links

Voicemail provides the ability to create a simple Auto Attendant facility whereby callers can be transferred to a User or Department from a predefined list of options. Please refer to the Creating an Auto Attendant section from page 222 for further details.

Modules determine the components that make up the configured individually. Please refer to the Connecting a Module section from page

imiser system. Each module can be

39 for further details.

Phones are the analogue ports and IP Phones connected to the system. Each phone can be configured individually. Please refer to the Connecting an Analogue Extension section from page 77 and the Connecting a PCS section from page 51 for further information.

An Unassigned Phones is an IP Phone requesting registration to the system. Please refer to the Using Auto Add Phones section from page 34 for further details.

An Unassigned Module is a module requesting registration to the system. Please refer to the Connecting a Module section from page 39 for further details.

This object gives access to system functionality and system setting such a System Name required for licence activation and the system Locale. Please refer to the Configuring System Details section from page 26 for further details.

Licences allow the use of a specific facility on the system or increase the specification of the system. Each licence purchased is exclusive to one system and one function. Please refer to the Installing Licences section from page 35 for further details.

The list of administrators of the system. A default Administrator called “Manager” will be created at startup with read/write access to all objects, and will be given a password of “managerpassword”. Please refer to the Setting Up Administrative Access section from page 31 for further details.

Facilities available for managing, maintaining and troubleshooting the system. Please refer to the Utilities section from page 250 for further details.

Ambiguous telephone numbers to be matched with incoming CLI when a match is not found within the Contacts database. The main purpose of these entries is to match the area code part of the CLI so that caller display will inform the User from which area the call has originated. Please refer to the Using Area Codes section from page 177 for further details.

WAN Links provide the ability to connect remote systems via a 2Mbps Megastream leased line or via ISDN dial up. Please refer to the Data Routing section from page 247 for further details.

System Architecture 20

Installation and Reference Manual

Installing a System

Please note

5100 Call Server – Front and Rear view

Input Triggers

Relay sockets

Power socket

Relay sockets

Integral LAN Switch with 4 x 10/100 Mbps Ethernet ports

16 analogue Phone ports

SpliceCom LED LAN Link

2 x ISDN PRI

4 x ISDN BRI

Disk Drive LED

A 5100 or 5108 Call Server or 4100 Call Server or 4140 Remote Call Server should be the first module to be powered up; all subsequent modules will join this system.

1 Power up the Call Server using the power supply provided. 2 On the 5100 and 5108 Call Servers the Disk Drive LED will flash to indicate a system check, and on

the 4100 and 4140 Call Servers the NET LED will flash to indicate a system check.

3 The Call Server’s IP address will be 192.168.0.1 4 The Call Server will search for a DHCP Server, if not found, the DHCP server functionality will be

enabled. If the Call Server is connected to a network during power up and finds a DHCP Server its internal DHCP facility will be disabled.

5 This Call Server will become the Primary Call Server. 6 The SpliceCom LED will become solid when the unit is ready.

Server exists on the network it will attempt to join this system. Ensure the Call Server is disconnected from the network during power up if this is not required. (Please refer to Connecting Multiple Call Servers from page

: If a Call Server is connected to a network during power up and it detects that another Call

72 for further details.)

USB Port

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5108 Call Server – Front and Rear view

4100 Call Server – Front and Rear view

SpliceCom LED

Power socket

Power over Ethernet PSU

10/100 Mbps

4 ISDN BRI

1 ISDN PRI

Input Triggers

Relay sockets

Integral LAN Switch with 4 x 10/100 Mbps Ethernet ports

LED status lights

4 x analogue Phone ports

Power socket

Input Trigger

Relay socket

2 x ISDN BRI

LAN Link

Please note that the front view diagram is incorrect, the first LED is the LINK LED and the second is the DISK drive LED.

Installing a System 22

LED status lights

Ethernet LAN port

(for future use)

Power socket for

Integral LAN Switch with 8 x 10/100 Mbps Ethernet ports

IP WAN Interface

Installation and Reference Manual

4140 Remote Call Server – Front and Rear view

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8 analogue phone ports

SpliceCom LED

Power socket

10/100 Mbps

4 ISDN BRI

1 ISDN PRI

IP WAN Interface

Input Triggers

Relay sockets

Once the Call Server has been powered up the following should be noted:

• If the Call Server’s DHCP Server functionality has been enabled the Call Server will give out an address

range of 192.168.0.2 to 192.168.0.250, with 192.168.0.250 being the first address given out. (For further information on using DHCP please refer to page 29 for further details.)

• When connecting to the LAN ports on the front of the 5100, 5108 or 4100 Call Server straight or

crossover cables can be used because each port automatically changes to MDI/MDX. When connecting to the LAN port at the back of the 4140 Remote Call Server a crossover cable is required.

• The ISDN Trunk ports available on a Call Server can be used to connect to the exchange or to another

system. Please refer to page 159 for further information.

• The system operates in native mode at G.711 64k with echo cancellation. If the

connected to the company LAN and that LAN is used to carry voice then that LAN must provide Quality of Service via LAN switches.

• The 5100 and 5108 Call Server and the 4140 Remote Call Server provide 16/4/8 analogue extension

ports respectively. When either of these Call Servers is the first to be installed each port will be automatically assigned to a User named Extn2001, Extn2002 etc and given an extension numbers of 2001, 2002 etc. Please refer to page 48 for further information on using an analogue extension port.

• Please note that the first 8 analogue ports on the 5100 Call Server are activated, the remaining 8 ports

will require a POTS licence to activate each port. Please refer to page 35 for further information on Licensing.

LED status lights

Ethernet LAN port

(for future use)

imiser system is

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System Operation

Manager

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It is important to perform an orderly shut down of the system before power down to ensure all changes are copied to the hard disk and so that the hard disk is shut down cleanly

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Running Manager

Working in Manager

The configuration of the Server. Manager is the application that is used to view and configure this database.

Changes to the configuration made via Manager or by a User via their handset are stored in the RAM and copied regularly to the hard disk to be stored permanently.

Manager is a web based application which can be run from any Web browser (Internet Explorer v5 or higher, or Apple Safari v3.2 or higher) that can connect to the

imiser system is controlled and stored on the LDAP database within the Call

- please refer to page 253 for further details.

imiser system.

1 Your PC must have a IP connection to the Call Server 2 Via a web browser, open

ip address of Call Server

http://

3 To view the system’s configuration administration access is required. By default, full access is

available by logging in when requested as follows:

User Name = Manager Password = managerpassword

/manager eg http://192.168.0.1/manager

For security, it is advisable to at least change the password for this Administrator once access has been achieved. Please refer to the Setting Up Administrative Access section from page 31 for further details.

On opening Manager a list of the configurable objects on the database is displayed in orange within the navigation pane on the left hand side (please refer to the Manageable Objects section from page 18 for a description of each object). A single click on one of the objects will display a list of the relevant entries in the database. The User’s list is displayed by default.

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Navigate within a list of entries

item.

Configure an entry

The following options will assist you to navigate and configure the database:

Display the next page of entries

Display the previous page of entries

Display the first page of entries

Create a new entry. The new entry will appear at the bottom of the list and will be called, for example, newDepartment, newUser, newLicence etc. Select this entry to configure the new

To open, view and edit an entry click on the blue link

To save the changes and return to the previous screen

To save the changes and remain in the current entry.

To return to the previous screen. If changes have been made these will not be saved.

To delete the entry currently displayed. A confirmation message will be received.

When a list of entries is displayed the Search box will be available at the top of the navigation pane. Enter the specific entry you wish to find in the Search box and then select Find or press Enter. The entry required will appear at the top of the list. If the Search box is empty, the Find button will take you to the top of the list.

To delete the current entry within the field

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for this particular entry. The Cancel button will return you to the previous screen.

Configuring System Details

An entry may consist of more than one page and the names of these pages are display at the top of the screen. Click on the page name to access that page. Use the Apply button to save a change in one page before moving to another page.

When a field name is displayed in blue the entry for this field can be selected from a list. Click on the field name and select the new entry required. The Cancel button will return you to the previous screen. The Clear button will delete the entry and leave the field empty.

If an entry for a field is orange, clicking on this link will display the configuration form

The System Details form of the configuration is used to enter information to identify the system such as the customer’s name, the supplier’s name and any other text that will help to distinguish the system. If licences have been purchased the text entered must match that provided with the licences. For further information please refer to the Installing Licences section from page 35.

1 In Manager, click on System 2 The System Details form will be displayed. 3 Enter a name for your system in the System Name field. Do not use punctuation in this field. This

field is useful to identify the system for troubleshooting purposes. If you have purchased licences enter your given System Name. (This information will appear on the right hand side of the Manager screen at the top and bottom once this entry has been updated.)

4 The Supplier field is provided purely for information unless licences have been purchased. Enter

the Supplier Name provided with the licences. (This information will appear on the left hand side of the Manager screen at the bottom once this entry has been updated.)

5 The Description field is provided purely for information to help you identify the system. This

information will appear on the left hand side of the Manager screen at the top.

6 The System Locale field will determine the language used for voicemail, ring sequences and Caller

Display.

7 Select Update or Apply when ready.

For information on the use of the Call Logging Password please refer to the Call Logging section from page 27.

For information on the use of the Diagnostic Engineering Password, Web Password and Maintainer Password please refer to the System Access section on page 251.

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Installation and Reference Manual

Call Status

Source

Target

Connected to

Call Logging

Changing the Call Logging Password

Configuring the Call Server

Call Status can be accessed in Manager by selecting Utilities and then Call Status. This application allows all calls on the system to be viewed.

number dialled

displayed if matched to the number dialled

: Name and extension number of the User making the call

: The number dialled by the Source. The User or Contact name will be displayed if matched to the

: The number the Source was actually connected to. The User or Contact name will be

Call information to be used by third party call logging applications can be obtained from the system via Telnet to port 4001.

Please refer to the Call Logging Format document available from Marketing for details on the description of each field.

The Call Logging Password is configurable within Manager and can be used to protect access to the call logging data.

1 In Manager select System 2 In the Call Logging Password field enter the password required. 3 Select Update or Apply when ready.

No logging output will occur without the Call Logging password being received (new line terminated). The password must be transmitted in one block/packet.

The Call Server can be configured within Manager by selecting Modules and then the Call Server. The Module Details form for that Call Server will appear.

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Setting the IP Address of the Call Server

The Call Server’s Name can be changed to help identify the Call Server. The module’s hardware ID (Serial Number) is displayed together with its IP address and the current version of software being run on this Call Server.

The Flash LED button will instruct the Call Server’s SpliceCom LED to flash 20 times. This feature is useful when multiple Call Servers are being used on a system and will identify the physical Call Server from its configuration form. For information on working with multiple Call Servers please refer to page 72.

For further information on upgrading the software on a Call Server please refer to Upgrading a System from page 258. For further information on using the Module Status field please refer to Connecting Mulitple Call Servers on page 72.

By default, a Call Server will have an IP address of 192.168.0.1. To change this address the following configuration can be made:

1 In Manager, select Modules 2 From the Modules list select the Call Server required 3 The Module Details form for that Call Server will be displayed. 4 Within the IP Address field enter the IP Address to be used by the Call Server. 5 Select Update or Apply when ready. 6 Reboot the Call Server (please refer to page 255 for information on how to reboot a Call Server.)

Please note:

• If the Call Server has DHCP enabled (as described below) and this IP address is not on the same

network as the DHCP Start and End Address DHCP will not be operational.

• If the Call Server has DHCP disabled the DHCP Base Address and Base Mask must be configured to

match the network that the Call Server is on.

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Using DHCP

(This is the recommended option if DHCP is being provided

from another vendor)

A list of each module’s IP address can be viewed in Manager by selecting Utilities and then IP Addresses.

During power up a Call Server will search for a DHCP Server, if not found, its internal DHCP server functionality will be enabled and will give out an address range of 192.168.0.1 to 192.168.0.250. If the Call Server finds a DHCP Server the DHCP facility will be disabled.

To change this default configuration the following changes can be made.

1 In Manager, select Modules 2 From the Modules List select the Call Server. 3 The Module Details form for that Call Server will be

displayed.

4 At the top of the form select the Call Server page.

The Call Server Details form will appear.

5 From the DHCP Server Mode list the following

options can be selected:

a Enable – this is the default entry. The Call

Server will act as the DHCP server using the IP Address range specified in the DHCP Start Address and DHCP End Address fields.

Please note: If another DHCP Server is detected on the network and the Call Server is rebooted it will continue to use 192.168.0.1or it’s previously configured IP Address and disable its own DHCP Server which means it will no longer give out IP addresses.

If the DHCP Server mode is re-enabled having previously been disabled, the Call Server will firstly check whether another DHCP Server exists on the network. Only if it does not receive an IP address will it become a DHCP Server.

b Disable – DHCP is disabled; the Call Server will use the IP address entered in the IP Address

field within the General page.

c Client – the Call Server will act as the DHCP client obtaining an IP address from a DHCP Server

on the network.

6 The DHCP Start Address and DHCP End Address fields specify the IP address range for the system

when the DHCP Server Mode is set to Enable. (Please note that these fields are irrelevant if the DHCP Server Mode is set to Disabled.)

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System Time

7 Select Update or Apply when ready. 8 Reboot the Call Server (please refer to page 255 for information on how to reboot a Call Server.)

Please note:

• Using the default configuration 192.168.0.250 will be the first IP address to be given out, followed by

192.168.0.249 etc.

• On initial connection a PCS 580, 570, 560, 410, 400 and 100 will obtain its IP address from a DHCP

server.

• On initial connection a Phone Module will obtain its IP address from a DHCP server.

The System configuration form displays the current date and time used by the system. This setting can be changed as follows:

1 In Manager, select Utilities 2 From the Commands list select Set Time 3 Enter the date and time required 4 Select Update when ready. 5 This information will be sent and updated on all handsets on the system

Please note:

• The system will take into account British Summer Time.

• If multiply Call Servers are being used on one system the above time will be the time used by the Call

Server whose Manager is being viewed. Each Call Server can have a different date and time. This information will be sent and updated on all handsets registered to the specific Call Server.

A Time Zone can be set when the time displayed in, for example, call logging information needs to be identified with its Time Zone. This is useful when, for example, one maximiser system with multiple Call Servers is working across multiple countries or states and the time of a call needs to be identified with its correct Time Zone. (For further information on working with multiple Call Servers please refer to page

72.)

Each Call Server can have a different Time Zone and is configured as follows:

1 Open Manager on the Call Server to be configured 2 Select Utilities 3 Under Commands select Set Time 4 From the Time Zone list box select the option required 5 Select Update when ready. 6 Reboot the Call Server (please refer to page 255 for further information)

Please note: Setnet can be used to set the time zone for a specific Phone Module. For further information please refer to the Using Telnet section from page 270.

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Setting up Administrative Access

Default Administrator

Additional Administrators

Creating a new Administrator

By default administration access to the database via Manager is available by logging in as “Manager” using the password “managerpassword”. This Administrator has full access to the database.

It is recommended that at least the Password be changed to ensure security of the system.

1 In Manager, select Administrators 2 The Administrator List will be displayed 3 Select Manager 4 In the Password field enter the password required 5 Select Update or Apply when ready

Please note: This default Administrator, Manager, gives Authorised Access to new Administrators as described below. It is therefore recommended that Manager is not deleted.

Additional administrators can also be created to enable staff to log on to Manager but have limited access to the configuration. A new administrator can be created as follows.

1 In Manager, select Administrators 2 The Administrator List will be displayed 3 Select the Add button 4 Enter the name required for the new Administrator 5 Enter a description if required to further identify this Administrator 6 Enter the password to be used by this Administrator 7 Tick Show Enabled. This will enable the use of the Show tick boxes below. If not ticked this

Administrator will have full access to the database (dependent on the rights and access allowed).

8 Tick the relevant box beside the section of the database to which this administrator is to have

access eg Show Users, Show Departments etc

9 If you wish this Administrator to be able to create and/or delete entries within the section of the

database selected above tick the relevant box, eg Add/Delete Users, Add/Delete Departments. This option will display the Add button within a list and the Delete button within a specific entry. (Please note that the Show option allows administrators to amend an entry even though Add/Delete is not selected (dependent on the rights and access allowed).)

10 Select Apply

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Assigning rights to the Administrator

Authorising the Administrator

11 If you wish the administrator to be shown a specific page of the database on opening Manager

from the Default Page list box select the page required, eg Contacts.

12 Select Apply

13 Within the Rights section at the bottom of the page select the Add Right button. A new entry will

be displayed providing full rights to the database dependent on the options selected above.

14 Select Update when ready.

15 From the Administrator List select Manager 16 Within the Authorised Access section select the Add Access button 17 Select the Administrator field 18 From the Select Administrator list select the administrator created above. (This entry will provide

full rights to the database dependent on the options selected within the Administrators configuration form.)

19 Select Update or Apply when ready.

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View the configuration logged on as the new Administrator

Creating an Administrator for PCS 60 Operator Console Mode

20 Open Manager via your web browser.

21 In the User Name field enter the name of the new

Administrator and enter this Administrator’s password.

22 The Administrator logged in will only have access to the

sections of the database configured above.

Please note: as the default Administrator, Manager, gives Authorised Access to new Administrators it recommended that Manager is not deleted.

When a User is using PCS 60 in Operator Console Mode the operator can also be given the ability to change a User’s DND and Forwarding settings via their Favourites. To enable this facility the operator will require an Administrator name and password.

1 Follow steps 1 to 7 above to create an new Administrator. 2 Select the Show Users option. This is the minimum option required for this facility to be available.

3 Follow steps 13 to 19 above to enter the Rights and

Access levels for this Administrator.

4 Within the PCS 60 application enter the Administrator

Name and Password in the PCS 60 Preferences dialogue box.

For further information on please refer to the PCS 60 Operator Console Mode section on page 68 and to the PCS 60 User Manual.

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Creating an Administrator for a Company

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Using Auto Add Phones

Turn On/Off Auto Add Phones

Manually register a Phone Module during installation

If the

imiser system is using the Companies feature you may wish an Administrator only to have access to the entries relating to a specific Company. In this case within the required Administrator’s configuration form select the Company field and select the required Company from the Select Company list.

For further information on Companies please refer to page 155.

This feature determines whether phones (analogue or IP) will be able to automatically register with the system.

If turned on (default), each Phone Module and IP Phone will be able to automatically register with the Call Server, each port and phone will be listed in the configuration and a User automatically created.

If turned off, each Phone Module and IP Phone must be manually registered on the system

1 In Manager, select Modules 2 Select the Call Server 3 Select the Call Server page. 4 In the Auto Add Phones check box tick to turn on or untick to turn off this feature.

If the Auto Add Phones feature is turned off a Phone Module must be manually registered during installation as follows:

1 In Manager select Unassigned Modules 2 The Phone Module will be displayed together with its MAC address 3 Select this link 4 The Module Status field will display Available 5 From the Module Status list box select Member 6 Select Update or Apply when ready 7 This unit will now be listed under Modules

The same method is used when installing a Trunk Module.

Please refer to the Connecting a Phone Module section from page 39 for further details.

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Manually register a PCS 580/570/560/410/400/100 during installation

Manually register a PCS 60 IP Phone or PCS 50 IP Phone during installation

Installing Licences

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If the Auto Add Phones feature is turned off an IP phone must be manually registered during installation as follows:

1 In Manager select Unassigned Phones 2 The MAC address of the PCS will identify the unit connected. Click on the unit required 3 From the Status list box select Member 4 Select Update when ready 5 The unit will now appear under Phones

Please refer to the Connecting a PCS section from page 51 for further details.

If the Auto Add Phones feature is turned off a PCS 60/50 IP phone must be manually registered during installation as follows:

1 In Manager select Unassigned Phones 2 The hardware address of the PC will identify the PCS 60 to be connected. Click on the entry

required.

3 From the Status list box select Member 4 Select Update when ready 5 The PCS 60/50 will now appear under Phones, identified by the hardware address of the PC

Please refer to the Connecting a PCS section from page 51 for further details.

Certain functionality of the licence in order to activate the relevant facility.

Each User created on the system requires a licence. There are two types of User licences either an IPUser licence or an IPVirtualUser licence. An IPUser licence will be required to create a User who will be assigned to a physical telephone. By default the which can be used for testing purposes. When a Call Server or Phone Module providing analogue ports is installed this is automatically supplied with the required number of IPUser licences and the corresponding number of analogue phone Users automatically created can utilize the system without any additional configuration. If further Users are to be created when connecting, for example, a PCS 60 (IP Phone), PCS 50 (IP Phone), PCS 580, 570, 560, 410/400 or 100 additional IPUser licences can be purchased. An IPUser licence will also be required to create a hot desking User as this User will be logging on to a physical handset.

An IPVirtualUser licence is required to create a User that requires PBX functionality but does not require a physical telephone. For example, where a company offers virtual office facilities, voicemail and forwarding features are required but the customer will not be using a physical telephone. There is no limit to the number of IPVirtualUser licences that can be run on a Call Server.

An OperatorConsole licence is required for each User who will be using the PCS Operators Console application or the PCS 580/570/560/60 in Operator Console mode. Please note that this is a per-user

imiser system requires the purchase and installation of the appropriate

imiser system is supplied with one IPUser licence

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licence not a concurrent-usage licence. Please refer to the PCS 60 Operator Console Mode section from page 68 for further details.

To provide a User with voicemail facilities a MessageBox licence is required. On purchasing a 5100 Call Server or 4140 Remote Call Server 10 MessageBox licences are provided. A 5108 Call Server is provided with 5 MessageBox licences. A 4100 Call Server is provided with 30 MessageBox licences. Further MessageBox licences can be purchased to increase this number, up to 1,000 per system when using embedded voicemail or up to 100 on a standalone 4140 Remote Call Server and up to 50 on a standalone 5108 Call Server. MessageBox licences are also required to provide a Department with voicemail functionality. Please refer to the Voicemail section on page 187 for further details. (Please note: a list of which Users and Departments have been assigned MessageBox licences can be viewed in Manager by selecting System and then Mailbox Licences. Please refer to page 238 for further details.)

A Voice Processing Port licence will enable an additional port/simultaneous voicemail connection on a 5100 or 5108 Call Server. The relevant number of licences will also be required to run the voicemail application from a standalone Linux PC or server. One licence is necessary for each connection required. Please refer to the Working with Voicemail section from page 187 for further details.

To provide Auto Attendant/IVR functionality beyond the single layer provided as standard on

imiser, an ESPSession licence allows a single channel to be enabled. Enhanced Speech Processing (ESP) is a voicemail function and can run on the Call Server (up to eight licences), Remote Call Server (up to four licences) or a standalone Linux server (up to sixty licences). ESP is a VoiceXML based application and supports Text-to-Speech.

The 5100 Call Server is provided with four BRI ports making 8 ISDN channels available, 4 channels are enabled on installation however to enable an additional channel a Trunk licence is required. A Trunk licence is also required to activate a channel on the PRI ports. The 5108 Call Server provides two BRI ports giving 4 ISDN channels two of which are enabled on installation and Trunk licences are required to enable further channels. The 4100 Call Server and 4200 Trunk module are provided with four BRI ports providing 8 ISDN channels and a PRI port however to activate a PRI channel a Trunk licence is required. The Remote Call Server is provided with four BRI ports making 8 ISDN channels available, 4 channels are active on installation and a Trunk licence is required to activate a further channel. To activate a PRI channel (up to 15 channels) on a Remote Call Server a Trunk licence is also required. To use DPNSS, SIP or H.323 IP Trunk services a Trunk licence is necessary for each concurrent channel required.

The 5100 Call Server is provided with 16 analogue POTS ports of which 8 are activated on installation. To activate the remaining 8 ports a POTS licence is required for each port. Please refer to the Connecting an Analogue Extension section from page 48 for further information.

A 5108Plus licence will increase the number of users supported by a 5108 Call Server to 12. The maximum number of analogue users stays at four however the IP users can increase to 8, or in an IP only environment all 12 users could be IP. Note that the additional users will also need an IPUser Licence.

A VoiceCompression licence allows a further one channel of voice compression to be enabled per Voice Compression Module, giving a maximum of 32 compression channels per module

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the additional users will also need an IPUser Licence.

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Configuring the System Name and Supplier Name

In summary, the licences available on a

imiser system are as follows:

5108Plus

Increases the number of users supported by a 5108 Call Server to 12. The maximum number of analogue users stays at four however the IP users can increase to 8, or in an IP only environment all 12 users could be IP. Note that

ESPSession Enables a single Enhanced Speech Processing channel on the on-board or off-

board voicemail application to facilitate the use of the Extended Attendant, VXML Scripts and Ring Back when Free. One licence per channel.

IPUser Enables the ability to create a User that is assigned to a physical extension.

One licence per User.

IPVirtualUser Enables the ability for a User, which is not assigned to a physical extension, to

be active on the system. One licence per virtual User.

MessageBox Enables voicemail functionality for a User or Department. One licence per User

or Department.

OCS

Enables the use of the interface, written by the Spider Group, between the

imiser and Microsoft Office Communications Server. Full instructions will

be given on the purchase of this service.

OperatorConsole Enables the use of the PCS Operators Console application or

PCS 580/570/560/60 Operator Console mode. One licence per User.

POTS Enables the use of an analogue extension port on the 5100 Call Server. One

per port.

TotalControl For use with the Total Control service. Full instructions will be given on the

purchase of this service.

Trunk Enables the use of an additional BRI or PRI channel, or utilize SIP, H.323 or

DPNSS. One licence per channel.

VisionReport

VisionRecording

For use with Vision. Please refer to Vision Installation and Configuration Manual for further details.

Vision Live

VisionMobility

VoiceCompression Enables a compression channel on a Voice Compression module. One licence

per channel.

Voice Processing Port Enables an additional voicemail connection on a 5100 and 5108 Call Server and

facilitate the use of the external voicemail application. One licence per port.

Licences are issued against a company name and the supplier’s name therefore the System Name and Supplier Name provided with the licences must be entered accurately in the System configuration form. If this information is not entered correctly the licences will not activate.

Please refer to the Configuring System Details section from page 26 for further details.

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Enter the Licences on the system

View the current licence status

Each licence is provided with the type of licence, the quantity of licences purchased and a licence key. This information must be correctly entered; otherwise the licences will not be operational.

1 In Manager, select Licences 2 The Licences List will appear. 3 Click on the Add button 4 In the Key field enter the alpha-numeric code

provided

5 In the Qty field enter the number of licences

purchased

6 In the Licence field enter the type of licence

purchased, eg IPUser, MessageBox

7 Select Update or Apply when ready

If a licence has not been entered correctly and not authenticated error messages will appear in the Warnings screen which can be accessed as follows:

1 In Manager, click on Utilities 2 Under the Diagnostics list select Warnings 3 Any error messages relating to licences will appear in this screen. No error messages indicate that

the licences have been accepted and authenticated correctly.

Please note that licences can be downloaded directly from SpliceCom by using the Download Licences option. You will require an internet connection and you will need to know your Forums registered email address and password. Please refer to Support for further assistance.

A list of the licences installed on the system together with how many of these licences have been used is available by opening:

http://<ip address of call server>/manager/licences.php

eg http://192.168.0.1/manager/licences.php

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Connecting a Module

Connecting a Phone Module

The Phone module enables connection of analogue handsets, including PCS 5/10s, to the system. All configuration information is obtained from the Call Server and is stored in RAM.

1 Connect the Phone module to the system via a LAN port on the front of the Phone module. This

can be either via the network or directly to a LAN port on the Call Server. (A straight LAN cable should be used when connecting the Phone Module to third party LAN switches.)

2 Power up the Phone Module. 3 The module will request an IP address from a DHCP server. It will make 4 attempts and then back

off for a period before trying again. The back off period will extend for each failed attempt.

4 Having obtained an IP address the module broadcasts to find a Call Server to provide its

configuration and become active.

5 The SpliceCom LED will become solid when the unit is ready. 6 If the Auto Add Phones feature is turned on the Phone Module will automatically register with the

system

If the Auto Add Phones feature is turned off the Phone Module must be manually registered as follows:

a In Manager select Unassigned Modules b The Phone Module will be displayed together with its MAC address c Select this link d The Module Status field will display Available e From the Module Status list box select Member f Select Update or Apply when ready g This unit will now be listed under Modules

7 Each analogue handset previously connected will ring once to confirm communication with the

system

8 Connect each analogue handset to a port on the Phone module via a master dongle or RJ45

connection if supplied. For further details on connecting an analogue handset please refer to page 48.

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5330 Front View

5315 Front View

5315/30 Rear View

4300 Front and Rear View

Power

PCMCIA

15 or 30 analogue Phone Sockets

LAN port

30 analogue Phone Sockets

Integral LAN Switch with 4 x 10/100 Mbps Ethernet ports

LAN Link

15 analogue

Phone Sockets

USB Socket

Integral LAN Switch with 4 x 10/100 Mbps Ethernet ports

Power socket

USB Socket

LAN Link

Socket

Card Slot

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Configuring a Phone Module

The Phone Module controls:

• Ringing of the telephones at the requested cadence

• Presentation of the CLI

• Recognition of off-hook state

• Recognition of dialed DTMF tones

• Presentation of DTMF for fax servers

• The left hand LED on a port displays when a phone is ringing

• The right hand LED on a port displays when a phone is in use

Please note:

• Anti static precautions are advised when handling, inserting or removing PCMCIA cards with respect to

the 4300 Phone modules.

• Calls placed on hold will be automatically taken off hold after 10 minutes

• On a reboot of the Phone Module each analogue handset will ring once to confirm communication

with system and display the number of new voicemail messages waiting for the User assigned to that phone (if voicemail enabled).

1 In Manager select Modules 2 The Phone Module will be displayed together with its MAC address 3 Select this link to view and configure this module. 4 For example, you may wish to edit the Name and Description of this module to give an identity and

location for future reference.

5 Select Update or Apply when ready.

The Call Server field indicates to which Call Server the Phone Module is registered. The Call Server is responsible for replicating relevant configuration information to the Phone Module. In turn the Phone Module will store in RAM configuration information relevant to its functionality. The Phone Module will also pass back configuration changes, eg User configuration, to the Call Server.

The Flash LED button will instruct the Phone Module’s SpliceCom LED to flash. This feature is useful when multiple Phone Modules are being used on a system and will identify the physical Phone Module from its corresponding configuration form.

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Changing the Password

Extension Numbering

2017, 2018 etc.

The Product Version field will display which version of software is currently running on the Phone Module. For information on upgrading this software please refer to page 260.

Each port on the Phone Module can now be configured independently, see page 77 for further details.

When accessing a Phone Module via Telnet, when upgrading for example, the module’s password is entered on connection. By default this is set to 7388. However this password can be changed as follows:

1 In Manager select Modules 2 Select the Phone Module required 3 Select the Phone Module page 4 In the Password field enter the password required 5 Select Update or Apply when ready. 6 Reboot the Phone Module

Once a Phone Module has registered with the system each port on the Phone Module, eg Port01, Port02 etc, will be automatically assigned to a User eg Extn2017, Extn2018 etc, and given an extension number, eg

Extension numbering starts with the next available number, therefore if, for example, the existing extension number range on a system is 2001-2016, and a 5130 Phone Module is installed the Users automatically created will be given an extension number within the range of 2017-2046. Each User can be amended to change this default configuration if required, see page 80 for further details.

To change the extension number range automatically used when a Phone Module registers with a system, eg 6001-6030, the following steps can be taken:

1 Before the Phone Module is connected to the system create a new User with an extension number

one before the start of the range required, eg 6000

2 Connect the Phone Module as described on page 39. 3 The new Users created will be given an extension number within the new extension number

range, eg 6001-6030.

4 The User created in step 1 can be deleted if no longer required.

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Setting the IP Address

Connecting a Trunk Module

When connecting a Phone Module to the system for the first time a DHCP server is required to give the Phone Module an IP address. This address can be viewed within the Phone Module’s configuration form as described on page 40. However once connected the Phone Module can be given a static address as follows:

1 Within the Command prompt enter

telnet <ip address of Phone Module> eg telnet 192.168.0.250

then press Enter

diag

2 You will be prompted for a Username, enter 3 You will be prompted for the password for this Phone Module. By default this is

, then press Enter

7388

. (The

Password is set within the relevant Phone Module’s configuration form.) Then press Enter.

4 Wait for a # to appear to confirm a connection has been made to the Phone Module

setnet

5 Enter

then press Enter

6 The current configuration will be displayed. You will be asked if you wish to change any of this

configuration.

7 Enter

then press Enter

8 You will be asked how the IP address will be assigned.

9 Delete the

and enter s then press Enter

10 Follow the onscreen prompts entering the IP address you wish to give the Phone Module, eg

192.168.0.50, the netmask you wish to use, eg 255.255.255.0 , the IP address of the Gateway (Default Router), eg 192.168.0.1 and the IP address of the TFTP server which will be the IP address of the Call Server to provide software to the Phone Module during an upgrade, eg 192.168.0.1.

11 At the following prompts make any changes you require otherwise press Enter to leave the option

unchanged.

12 You will be asked if you wish to save the changes. Enter 13 Telnet will display that the changes are being saved and 14 Enter

poweroff

then press Enter and the Phone Module will reboot

then press Enter

Done

will appear once finished.

Please note:

• to return to using DHCP follow the same instructions above. Enter

when asked how the IP address

will be assigned.

• at any prompt you may enter a

• for further information on working with Telnet please refer to page 270.

• a list of each module’s IP address can be viewed in Manager by selecting Utilities and then IP

for help on that option or q to quit.

Addresses.

• Setnet can be used to set the Time Zone to be used by a specific Phone Module

A Trunk Module can be connected to a system to increase the number of ports required, for example, to provide additional ISDN ports, an additional WAN port etc. A Trunk licence will be required to activate a PRI channel if required. Please refer to page 35 for further details.

1 Connect the Trunk Module to the system via one of LAN ports on the front of the Trunk Module.

This can be either via the network or directly to a LAN port on the Call Server.

2 Power up the Trunk Module.

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Configuring a Trunk Module

Power socket

Power socket for

Power over Ethernet PSU

4 ISDN BRI

1 ISDN PRI

Input Triggers

Relay sockets

IP WAN interface

Integral LAN Switch with 8 x 10/100 Mbps Ethernet ports

LED status lights

3 The module will request an IP address from a DHCP server. It will make 4 attempts and then back

off for a period before trying again. The back off period will extend for each failed attempt.

4 Having obtained an IP address the module broadcasts to find a Call Server to provide its

configuration and become active.

5 The SpliceCom LED will become solid when the unit is ready. 6 In Manager select Unassigned Modules 7 The Trunk Module will be displayed together with its MAC address 8 Select this link

9 The Module Status field will display “Available” 10 From the Module Status list box select “Member” 11 Select Update or Apply when ready 12 This unit will now be listed under Modules

1 In Manager select Modules

2 The Trunk Module (SpliceGateway) will be displayed together with its MAC address

3 Select this link to view and configure this module.

4 For example, you may wish to edit the Name and Description of this module to give an identity and

location for future reference.

5 Select Update or Apply when ready

The Call Server field indicates to which Call Server the Trunk Module is registered. The Call Server is responsible for replicating relevant configuration information to the Trunk Module. In turn the Trunk Module will store in RAM configuration information relevant to its functionality.

The Flash LED button will instruct the Trunk Module’s SpliceCom LED to flash. This feature is useful when multiple Trunk Modules are being used on a system and will identify the physical Trunk Module from its corresponding configuration form.

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Setting the IP Address of the Trunk Module

Connecting a Voice Compression Module

LAN Port

The Product Version field will display which version of software is currently running on the Trunk Module. For information on upgrading this software please refer to page 260.

When connecting a Trunk Module to the system for the first time a DHCP server is required to give the Trunk Module an IP address. This address can be viewed within the Trunk Module’s configuration form as described above. However once connected the Trunk Module can be given a static address as follows:

1 In Manager select Modules

2 Select the Trunk Module (SpliceGateway)

3 Within the IP Address field enter the IP Address to be used by the Trunk Module.

4 Select Update or Apply when ready

5 Reboot the Trunk Module (please refer to page 255 for information on how to reboot a Trunk

module.)

A list of each module’s IP address can be viewed in Manager by selecting Utilities and then IP Addresses.

Each port on the Trunk Module can now be configured independently, please refer to Working with Trunks on page 159 and Creating a WAN Link on page 247 for further details:

The 4400 Voice Compression Module provides 16 channels, of 8kbps, G.729a based compression and can be deployed to provide a higher density of calls between Call Servers over low speed WAN links. Additional channels of voice compression can be enabled via a licence – giving a maximum of 32 compression channels per module (please refer to the Licensing section on page 35 for further information). By using a standard LAN for interconnectivity between the Voice Compression Module and its associated Call Server, there is no limit to the number of compression channels that can be practically deployed. This allows multiple 4400 Voice Compression Modules to be used with single or multiple Call Servers, with the compression channels being dynamically allocated as, or when, required.

1 Connect the VCM to the system via the LAN port on the front of the VCM Module. This can be

either via the network or directly to a LAN port on the Call Server.

2 Power up the VCM.

3 The module will request an IP address from a DHCP server. It will make 4 attempts and then back

off for a period before trying again. The back off period will extend for each failed attempt.

4 Having obtained an IP address the module broadcasts to find a Call Server to provide its

configuration and become active.

5 The SpliceCom LED will become solid when the unit is ready.

6 In Manager, select Unassigned Modules

7 The VCM will be displayed together with its MAC address

8 Select this link

9 The Module Status field will display “Available” 10 From the Module Status list box select “Member” 11 Select Update or Apply when ready 12 This unit will now be listed under Modules

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Configuring a Voice Compression Module

Setting the IP Address

Power

socket

For information on installing a Voice Compression card please refer to page 47.

A Voice Compression Module or Card will required at both ends to provide compression for calls between two Call Servers.

1 In Manager select Modules

2 The VCM (SpliceCompressor) will be displayed together with its MAC address

3 Select this link to view and configure this module.

4 For example, you may wish to edit the Name and Description of this module to give an identity and

location for future reference.

5 Select Update or Apply when ready

The Call Server field indicates to which Call Server the VCM is registered. The Call Server is responsible for replicating relevant configuration information to the VCM. In turn the VCM will store in RAM configuration information relevant to its functionality.

The Flash LED button will instruct the VCM’s SpliceCom LED to flash. This feature is useful when multiple VCMs are being used on a system and will identify the physical VCM from its corresponding configuration form.

The Product Version field will display which version of software is currently running on the VCM. For information on upgrading this software please refer to page 260.

When connecting a VCM to the system for the first time a DHCP server is required to give the VCM an IP address. This address can be viewed within the VCM’s configuration form as described above. However once connected the Voice Compression Module can be given a static address as follows:

1 In Manager select Modules

2 Select the VCM (SpliceCompressor)

3 In the IP Address field enter the IP Address to be used by the VCM.

4 Select Update or Apply when ready.

5 Reboot the VCM (please refer to page 255 for information on how to reboot a VCM.)

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Creating an Intramodule Trunk

Installing a Voice Compression Card

A list of each module’s IP address can be viewed in Manager by selecting Utilities and then IP Addresses.

In order to use the compression provided by a Voice Compression Module or Card (see below) the following should be configured. For calls between two Call Servers a Voice Compression Module or Voice Compression Card is required at both ends together with the following configuration.

1 In Manager select Modules

2 Select the Call Server to which the VCM is connected or in which the card has been installed.

3 Select the Intra Module Trunks page

4 Click on Add Intra Module Trunk

5 In the Name field enter a name to identify this

trunk

6 From the Compression Type list box select

G729A8K

7 In the Capacity field enter the number of

connections to be allowed on this link

8 Select Apply when ready

9 Select the Add Intra Module Member button and

the list of Modules will appear

10 Select the Call Server at the other end 11 Repeat steps 11 to 12 until all the required Call Servers that have a VCM connected or containing a

Voice Compression Card have been entered.

12 Select Update or Apply when ready.

If calls are to be made to a Call Server that does not contain a Voice Compression Card or does not have a Voice Compression Module connected it is recommended that the this Intramodule Trunk is also created.

If fax calls are to be made between two Call Servers where compression is supported a Dial Plan entry similar to the this example should be created within each fax machine’s Dial Plan. The DialTransparent Action will allow calls to by-pass an Intramodule Trunk. A call made via this Dial Plan entry does not consume a compression channel. (For further information on Dial Plans please refer to page 114.)

A Voice Compression Card supports G.729a to compress up to 8 voice calls between two Call Servers. A card must be fitted at both ends as follows:

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4100 Call Server

4140 Remote Call Server

Connecting an Analogue Phone

1 Remove the lid

2 Place nylon standoffs supplied in mounting holes located by silk screen logos MTG14 and MTG15.

3 Fit VCC card to connector J12 and standoffs - take care to correctly align the pins.

4 Replace the lid.

1 Remove the lid

2 Place nylon standoffs supplied in mounting holes located by silk screen logos MTG9 and MTG10.

3 Fit VCC card to connector J11 and standoffs - take care to correctly align the pins.

4 Replace the lid.

Analogue extension ports are available on 5100 Call Server, 4140 Remote Call Server and 5300/4300 Phone Modules providing 16/8/15/30 ports respectively.

When a 5100 or 5108 Call Server, 4140 Remote Call Server or Phone Module is connected to the system 4/8/15/16/30 Users are created and named, for example, Extn2001, Extn2002 etc. Each User is assigned to an extension port, for example, Port 1, Port 2 etc, and given an extension number within the next available extension range, for example, 2001, 2002 etc. This default configuration can be changed as detailed within the Working with Users section from page 80.

To change the extension number range automatically used when a 5100 Call Server, 4140 Remote Call Server or Phone Module registers with a system, eg to 6001-6030, the following steps can be taken:

1 Before the Call Server/Phone Module is connected to the system create a new User with an

extension number one before the start of the range required, eg 6000

2 Connect the Call Server/Phone Module as described from page 39.

3 The new Users created will be given an extension number within the new extension number

range, eg 6001-6030.

4 The User created in step 1 can be deleted if no longer required.

To change the extension numbers for multiple Users after they have been created please refer to page 84.

A PCS 520, PCS 505, PCS 10, PCS 5 or an alternative analogue telephone can be connected to an analogue port via a master dongle or via a RJ45 connection if supplied.

Each port is provided with two LEDs to display the following activity on the port:

Left hand LED = displays when phone ringing Right hand LED = displays when phone in use

When a 5100 or 4140 Call Server or Phone Module is rebooted each analogue handset will ring once to confirm communication with the system and will display the number of new voicemail messages waiting for the User assigned to that phone (if voicemail enabled).

Calls placed on hold will be automatically taken off hold after 10 minutes.

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Analogue Ports on a 5100 Call Server

Increasing the volume on an analogue port

Phone Module

The first eight analogue ports on the 5100 Call Server are activated on installation. However ports 9-16 require a POTS licence to enable each port. Once the required number of POTS licences have been installed as described in the Installing Licences section from page 35 the following configuration is required:

1 In Manager select Phones

2 Select the relevant 5100 Call Server

3 Select the relevant analogue port – port09-port16

4 Remove the tick from the Unlicensed Port field

5 Select Update or Apply when ready.

For further information on configuring an analogue port please refer to page 77.

Each User assigned to ports 9-16 will also required configuration as follows:

1 In Manager select Phones

2 Select the relevant 5100 Call Server

3 Select the relevant analogue port – port09-port16

4 Select the entry displayed in the User field eg Extn2009

5 The User configuration form will be displayed

6 Remove the tick from the Ex Directory field (unless required)

7 Select Apply

8 Select the Licences page

9 Remove the tick from the Disable (lack of licence) field 10 Select Update or Apply when ready.

For further information on User configuration please refer to the Working with Users section from page 80.

Setnet can be used to increase the volume given out on each analogue port on a Phone Module as follows:

1 Within the Command prompt enter

telnet <ip address of Phone Module> eg telnet 192.168.0.250

then press Enter

diag

2 You will be prompted for a Username, enter

3 You will be prompted for the password for this Phone Module. By default this is

Password is set within the relevant Phone Module’s configuration form.) Then press Enter.

4 Wait for a hash (#) to appear to confirm a connection has been made to the Phone Module

setnet

5 Enter

then press Enter

6 The current configuration will be displayed. You will be asked if you wish to change any of this

configuration.

7 Enter

then press Enter

, then press Enter

7388

. (The

System Operation Installation and Reference Manual v3.2/0410/6 49

Installation and Reference Manual

module.

Call Server

Connecting a PCS 520

Caller Display Type

Default Dial Plan entries

Connecting a PCS 10

Caller Display Type

8 Press Enter until the line “Audio Level”

appears

9 Type in either 6 (recommended) or 12

depending on the volume level required.

10 Press Enter 11 You will be prompted to save this new

configuration. Enter

12 The configuration will be saved. 13 Enter

poweroff

to reboot the phone

To increase the volume output for the analogue ports on a Call Server a file called pots_level must be created within the /SpliceCom directory as follows:

1 Telnet on to the Call Server as described in the Root Access to a Call Server section from page 252

2 Enter

vi /SpliceCom/pots_level

3 Press i to go to Insert mode

4 Type in either

(recommended) or 12

5 Press Esc to exit Insert mode

6 Enter :wq to quit the file and save the changes.

7 Reboot the Call Server

When connecting a PCS 520 to an analogue port the following should be considered:

The Default Caller Display Type for an analogue port will be suitable for use with the PCS 520 however to ensure that CLI information is displayed correctly and to allow the handset is to support Paging and Auto Answer the System Phone Caller Display Type should be used. For details on how to configure an Analogue Extension port please refer to page 77.

The PCS 520 is supplied with 10 pre-programmed buttons which dial specific dial plan entries. Therefore the User assigned to the PCS 520 must be configured with a Dial Plan containing these default dial plan entries. For details on working with Dial Plans please refer to page 114 and, within this section, details on using the default dial plan entries from page 121.

When connecting a PCS 10 to an analogue port the following should be considered:

The Default Caller Display Type for an analogue port uses the Caller Display Type required for use with the PCS 10 however if the handset is to support the Message Waiting light the PCS 10 Lamp Caller Display Type should be used. For details on how to configure an Analogue Extension port please refer to page

77.

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Default Dial Plan entries

Connecting a PCS 505 or PCS 5

Caller Display Type

Default Dial Plan entries

Connecting a PCS IP Phone

Power over Ethernet considerations

The PCS 10 is supplied with 10 pre-programmed buttons which dial specific dial plan entries. Therefore the User assigned to the PCS 10 must be configured with a Dial Plan containing these default dial plan entries. For details on working with Dial Plans please refer to page 114 and, within this section, details on using the default dial plan entries from page 121.

When connecting a PCS 505 or PCS 5 to an analogue port the following should be considered:

The Default Caller Display Type for an analogue port uses the Caller Display Type required for use with the PCS 505 or PCS 5 however if the handset is to support the Message Waiting light the Lamp Caller Display Type should be used. For details on how to configure an Analogue Extension port please refer to page 77.

The PCS 505 and PCS 5 are supplied with a System Feature Guide displayed on the handset detailing the use of specific dial plan entries. Therefore if the User is going to utilize these dial plan entries the User must be configured with a Dial Plan containing these entries. For details on working with Dial Plans please refer to page 114 and, within this section, details on using the default dial plan entries from page 121.

Power to the PCS 580G, PCS 570G/570, PCS 560, PCS 410/400 and PCS 100 is provided by Power over Ethernet (PoE). This can be supplied by a third party PoE switch or by the following methods:

• PCS 580/570/560 – via one of the PoE LAN ports on a Call Server or Phone Module, or via a Single

Terminal Ethernet Power Supply (STEPS)

• PCS 410/100 - via one of the PoE LAN ports on a Call server or Phone Module, or via a Single Terminal

Ethernet Power Supply (STEPS)

• PCS 400 - via one of the PoE LAN ports on a Call server or Phone Module, or via a Single Terminal

Ethernet Power Supply (STEPS). The PCS 400 can also obtain power via a PCS PSU when it is not practical to use PoE. Please note that if a PCS 400 is connected to a PoE LAN port on a 5100 or 5108 Call Server the PCMCIA card cannot be used.

Please note that if a STEPS is used with a PCS 580G or PCS 570G the LAN port (port 1) will not run at 1G.

PoE LAN ports are provided as follows:

5100 Call Server, 5108 Call Server and 5300 Phone Module – provides 4 x PoE LAN ports supporting the following combinations:

1 x PCS 580G or 1 x 410/400 or up to 2 x PCS 570G/570 or up to 4 x PCS 100/PCS 560 or up to 1 x PCS 570 and 2 x PCS 100/PCS 560

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PCS 560/570/580

4100 Call Server and 4200 Trunk Module – PoE is supplied to the LAN ports via the Power over Ethernet PSU connected to the LAN PWR port on the back of these modules. This will give power to the 2 spare pairs on each of the 8 Ethernet ports on front of these Call Servers supporting up to eight IP phones (PCS 580, 570, 560, 410/400, 100) in any combination.

(Please note that one PoE port cannot be used to provide PoE to ports on a hub or switch connected to this port.)

Power to a PCS 560/570/580 is supplied by Power over Ethernet (PoE). This can be provided either by a third party PoE switch or directly via a PoE LAN port available on a Call Server or 5300 Phone Module.

1 Connect the PCS 560/570/580 (via its LAN port numbered 1) to the system via the network or

directly to a LAN port on the Call Server or 5300 Phone Module

2 The PCS 560/570/580 is a DHCP client and must obtain its IP address from a DHCP server.

3 An IPUser licence is required and must be installed before registration (See page 35 for further

information.)

4 If the Auto Add Phones feature is turned on the Call Server will automatically register the PCS.

(Please refer to page 34 for further details.)

If the Auto Add Phones feature is turned off:

a In Manager select Unassigned Phones b The MAC address of the PCS will identify the unit connected. Click on the unit required c From the Status list box select Member d Select Update when ready e The unit will now appear under Phones

5 Once the system has registered the PCS a User will be automatically created, configured with the

next available extension number.

6 The User’s Initial Phone field displays the name of the PCS that is assigned to this User.

Please note that the phone’s LAN port numbered 2 can be used to connect the User’s PC, for example.

Each PCS 560/570/580 connected can be configured independently, see page 79 for further details.

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Setting the Password

Viewing the IP address

Setting the IP Address

When using Telnet to, for example, upgrade the phone or change its IP Address. The phone’s password is entered to authorise these changes. By default this is set to 7388 but this password can be changed as follows:

1 In Manager, from the Navigation pane, select Phones

2 Select the module that the phone is connected to, eg Call Server

3 The MAC address of the unit will identify the specific phone required

4 Select the phone required

5 In the Password field enter the alpha-numeric characters required

6 Select Update or Apply to save the changes.

To determine the IP address of a PCS 570/560 select the Favourites button , then Settings and then Phone Information (alternatively press the key at the bottom right hand side and then select Phone Information). The IP address will be displayed at the top of the screen.

To determine the IP address of the PCS 580 press the Settings icon five times. The IP address will be displayed at the top of the Phone Information screen.

A list of the IP addresses for each PCS 570/560/580 can be viewed in Manager by selecting Utilities and then IP Addresses.

When connecting a PCS 560/570/580 to the system for the first time a DHCP server is required to give the PCS an IP address. This address can be viewed as described above. However once connected the PCS 560/570/580 can be given a static address as follows:

1 Within the Command prompt enter

telnet <ip address of PCS 570/560> eg telnet 192.168.0.250

diag

2 You will be prompted for a Username, enter

3 You will be prompted for the password for this phone. By default this is

Setting the Password section from page 53 for further details.)

4 Wait for a # to appear to confirm a connection has been made to the PCS

setnet

5 Enter

7388

. (Please refer to the

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Connect to the Call Server remotely

the remote phones and send this to the system.

6 The current configuration will be displayed. You will be asked if you wish to change any of this

configuration.

7 Enter

8 You will be asked how the IP address will be assigned.

9 Delete the

and enter s

10 Follow the onscreen prompts entering the IP address you wish to give the PCS, eg 192.168.0.50,

the netmask you wish to use, eg 255.255.255.0, the IP address of the Gateway (Default Router), eg

192.168.0.1 and the IP address of the TFTP server which will be the IP address of the Call Server to provide software to the PCS during an upgrade, eg 192.168.0.1.

11 At any other prompt make the changes you require otherwise press Enter to leave the option

unchanged.

12 You will be asked if you wish to save the changes. Enter

13 Telnet will display that the changes are being saved. 14 When finished enter

poweroff

and the PCS will reboot

Please note:

• If a PCS 560/570/580 has been given an IP address already being used on the network a warning

message will be displayed on the screen.

• To return to using DHCP follow the same instructions above. Enter

when asked how the IP address

will be assigned.

• At any prompt you may enter a

• For further information on working with Telnet please refer to page 270.

• Setnet can be used to set the Time Zone to be used by a specific IP phone

for help on that option or q to quit.

If the PCS 580/570/560 is to be installed at a remote location, eg via a VPN connection, the phone should be configured with the IP address of the Call Server it is to connect to as follows.

1 Follow steps 1-7 as described above to telnet on to the phone and access Setnet.

2 Press Enter until the line “Startup arguments” appears.

3 Type in

–gk <ip address of call server>

4 Continue to press Enter until you are asked if you wish to save the changes. Enter

5 Telnet will display that the changes have been saved.

6 When finished enter

poweroff

and the PCS will reboot.

If the Call Server is to collect BLF data from this

–blf

remote phone enter

after the IP address of the call server, eg –gk 192.168.0.1 –blf. The Call Server will then multicast this BLF information to all PCSs on the system.

If there are more than one remote IP Phone connected to the system one of these phones can be configured to collect the BLF data from all

eg –gk 192.168.0.1, and press Enter

This can be achieved by entering would then just be configured with the IP address of the call server, eg –gk 192.168.0.1.

It is recommended that no more than 5-6 remote IP phones should be configured to use this facility in order to keep the overhead on the Call Server to a minimum.

System Operation 54 Installation and Reference Manual v3.2/0410/6

–relay

after the IP address, eg –gk 192.168.0.1 –relay. The other phones

Installation and Reference Manual

Operator Console Mode

dials on a very active system.

PCS 410/400/100

Installing a PCS 410/400/100

To remove this entry repeat steps 1 and 2 above. Delete the entry within Startup arguments and press Enter. Continue with steps 4, 5 and 6 above.

When a PCS 580/570/560 is being used by a receptionist or switchboard operator the Operator Console Mode can be enabled to provide the operator with additional data to assist with the efficient handling of calls. The additional data is displayed within the User’s Favourites screen and can display a User’s forwarding settings, DND settings, when the User is logged out, when the User is on a call and who they are talking to, when a User is listening to voicemail, the number of new voicemail messages awaiting a User/Department and the number of missed calls received by a User/Department. Please refer to the relevant user manual for further details.

The Operator Console Mode is configured as follows:

1 Install an Operator Console licence as described from page 35. 2 Open the User configuration form for the User who wishes to use Operator Console Mode. 3 Select the Licences page 4 From the Operator Console Licence list box select

either:

a Yes - to apply the licence to this User and

enable Operator Console Mode, or

b PCS Partner only – this option can be used

when a PCS 60 is partnering a PCS 5xx IP phone. By selecting this option the Operator Console Mode information will only be displayed on the PCS 60. This will prevent the IP phone from running out of memory when the User has configured many speed

5 Select Update or Apply when ready

Power to a PCS 410/400/100 is supplied by Power over Ethernet (PoE). This can be provided either by a third party PoE switch or directly via a LAN port available on a Call Server or 5330 Phone Module. Alternatively, the individual STEPS Power over Ethernet PSU can be used.

The PCS 400 can also be powered by a standalone PCS PSU connected directly to the back of the phone. When using these individual power supplies, it is recommended that when powering on/off the PCS 400 that this is done via the power block not via the connection to the phone. Please note: if a PCMCIA card is required in the PCS 400 a PCS PSU must be used.

1 Connect the PCS to the system via the network or directly to a LAN port on the Call Server or 5300

Phone Module

2 The PCS is a DHCP client and must obtain its IP address from a DHCP server. 3 An IPUser licence is required and must be installed before registration (See page 35 for further

information.)

4 If the Auto Add Phones feature is turned on the Call Server will automatically register the PCS.

(Please refer to page 34 for further details.)

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Setting the Password

If the Auto Add Phones feature is turned off:

5 Once the system has registered the PCS a User will be automatically created, configured with the

next available extension number.

6 The User’s Initial Phone field displays the name of the PCS that is assigned to this User.

The above procedures also apply to non-SpliceCom IP Phones. A list of recommended IP Phones is available from our Marketing department.

Each IP Phone connected can be configured independently, see page 79 for further details.

Please note that a PCS 410/400 has no storage capabilities and therefore any downloads performed by the User will not be stored permanently.

Please refer to the PCS 400/410 and 50 User Manual and the PCS 100 User Manual for detailed descriptions of how to use these IP phones.

1 In Manager, from the Navigation pane, select Phones 2 Select the module that the phone is connected to, eg Call Server 3 The MAC address of the unit will identify the specific phone required 4 Select the phone required 5 In the Password field enter the alpha-numeric characters required 6 Select Update or Apply to save the changes.

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Viewing the IP address

Setting the IP Address

To determine the IP address of a PCS 410/400, on the handset press the More Options icon twice

and then the Information icon.

The IP address of a PCS 100 is displayed during power up of the phone. Alternatively, after power up press the Down arrow once (or twice if Favourites are displayed) on the handset, the IP address will be displayed on the screen. Press the down arrow or up arrow to return to the previous screen.

A list of the IP Addresses for each PCS 410/400/100 can be viewed in Manager by selecting Utilities and then IP Addresses.

When connecting a PCS 410/400/100 to the system for the first time a DHCP server is required to give the PCS an IP address. This address can be viewed as described above. However once connected the PCS 410/400/100 can be given a static address as follows:

1 Within the Command prompt enter

telnet <ip address of PCS 410/400/100> eg telnet 192.168.0.250

diag

2 You will be prompted for a Username, enter 3 You will be prompted for the password for this IP phone. By default this is

the Setting the Password section from page 56 for further details.)

4 Wait for a # to appear to confirm a connection has been made to the PCS

setnet

5 Enter 6 The current configuration will be displayed. You will be asked if you wish to change any of this

configuration.

7 Enter 8 You will be asked how the IP address will be assigned.

9 Delete the

and enter s

10 Follow the onscreen prompts entering the IP address you wish to give the PCS, eg 192.168.0.50,

the netmask you wish to use, eg 255.255.255.0, the IP address of the Gateway (Default Router), eg

192.168.0.1 and the IP address of the TFTP server which will be the IP address of the Call Server to provide software to the PCS during an upgrade, eg 192.168.0.1.

11 At any other prompt make the changes you require otherwise press Enter to leave the option

unchanged.

12 You will be asked if you wish to save the changes. Enter 13 Telnet will display that the changes are being saved. 14 When finished enter

poweroff

and the PCS will reboot

Please note:

• If a PCS 410, 400 or 100 has been given an IP address already being used on the network a warning

message will be displayed on the screen.

7388

. (Please refer to

System Operation Installation and Reference Manual v3.2/0410/6 57

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Connect to the Call Server remotely

Installing PCS Software

PCS 60

• To return to using DHCP follow the same instructions above. Enter

when asked how the IP address

will be assigned.

• At any prompt you may enter a

• For further information on working with Telnet please refer to page 270.

• Setnet can be used to set the Time Zone to be used by a specific IP phone

for help on that option or q to quit.

If the PCS 410/400/100 is to be installed at a remote location, eg via a VPN connection, the phone should be configured with the IP address of the Call Server it is to connect to as follows.

1 Follow steps 1-7 as described above to telnet on to the phone and access Setnet. 2 Press Enter until the line “Startup arguments” appears. 3 Type in

–gk <ip address of call server>

4 Continue to press Enter until you are asked if you wish to save the changes. Enter 5 Telnet will display that the changes have been saved. 6 When finished enter

poweroff

and the PCS will reboot.

If the Call Server is to collect BLF data from this

–blf

remote phone enter

after the IP address of the call server, eg –gk 192.168.0.1 –blf. The Call Server will then multicast this BLF information to all PCSs on the system.

If there are more than one remote IP Phone connected to the system one of these phones can be configured to collect the BLF data from all the remote phones and send this to the system. This can be achieved by entering

–relay

would then just be configured with the IP address of the call server, eg –gk 192.168.0.1.

It is recommended that no more than 5-6 remote IP phones should be configured to use this facility in order to keep the overhead on the Call Server to a minimum.

To remove this entry repeat steps 1 and 2 above. Delete the entry within Startup arguments and press Enter. Continue with steps 4, 5 and 6 above.

eg –gk 192.168.0.1, and press Enter

after the IP address, eg –gk 192.168.0.1 –relay. The other phones

The PCS 60 is a PC application that provides standard telephony functionality, CLI display, Operator Console mode and gives Users the ability to set features such as forwarding and DND via an easy to use dialogue box. It can be run on MS Windows and Apple Mac OS X version 10.4 or above. It can be installed as an IP soft phone on a multi-media PC or programmed to partner a User’s telephone handset (PCS 580, 570, 560, 410/400, 100, 10, 5 or an alternative analogue handset).

Please refer to the PCS 60 User Manual for detailed descriptions on how to use this software.

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Installing the PCS 60 software on a Windows PC

PCS 60 as an IP Soft Phone on a Windows PC

Before starting:

Running the PCS 60:

1 The set up file will be supplied in a compressed folder called, for example, PCS60.3.2.31.msi.zip.

Open this folder.

2 Run (double click) the set up file called, for example, PCS60.3.2.31.msi 3 At the Welcome screen select Next to continue the

installation

4 By default the software will be loaded into C:\Program

Files\SpliceCom\PCS60. Change this location if required.

5 Where relevant you will also have the option to load the

software for all Users of the PC or just for the currently logged in User.

6 Select Next when ready 7 At the Confirm Installation screen select Next. 8 The software will be installed 9 At the Installation Complete screen select Close

The PCS 60 software can be run as an IP soft phone where the PC becomes the User’s telephone and the voice path will be via the PC. Therefore the PC will require a headset, microphone, sound card etc installed.

1 The PC must have a network connection to a Call Server. 2 An IPUser licence must be previously installed and be available. (For further details please refer to

page 35.)

1 Run the PCS60.exe file either from the Installation folder, eg C:\Program Files\SpliceCom\PCS60, or

create a short cut to this file in the required location and double click on the short cut.

2 The PCS 60 application will open. 3 The “Searching for Server” message will be displayed as the registration process on the Call Server

is started.

If the “Searching for Server” message remains and the PCS 60 does not continue to the next step

check that the PC can communicate with the Call Server or refer to the Connect to the Call Server Remotely section below.

4 The “No Licences Available” message will be displayed while the phone is registered as described

in step 5.

5 If the Auto Add Phones feature is turned on the Call Server will automatically register the PCS 60.

(Please refer to page 34 for further details.)

If the Auto Add Phones feature is turned off:

a In Manager select Unassigned Phones b The hardware address of the PC will identify the PCS 60 to be connected. Click on the entry

required. (If the PCS 60 is not listed check the installation of the IPUser licence as detailed from page 35.)

c From the Status list box select Member d Select Update when ready

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Connect to the Call Server remotely

again.

e The PCS 60 will now appear under Phones, identified by the hardware address of the PC

6 The “Security Access Denial” message will be displayed while an IPUser licence is assigned. 7 A User account will be automatically created configured with the next available extension number.

This User will be assigned to the PCS 60, the “Security Access Denial” message will disappear and the PCS 60 will be ready for use. The name of the assigned User will be displayed in the Status Bar at the bottom left hand corner of the PCS 60.

The User’s Initial Phone field displays the PCS 60 that the User is assigned to, identified by the hardware address of the PC.

If the PCS 60 is to be run remotely over, a different subnet, a VPN, for example, the application must be configured with the IP address of the gatekeeper, in other words the Call Server, before the registration process can take place. This can be done in two ways:

1 Using PCS 60 Preferences

a From the File menu select Preferences b In the IP Address field enter the IP address of the

Call Server, eg 192.168.0.1

c Select OK (Window only d Close down the PCS 60 and run the software

2 Using the PCS 60’s short cut properties

a Open the Properties of the PCS 60’s short cut

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Remote BLF

PCS 60 on a Windows PC as a partner to a User’s handset

max

Before starting:

b In the Target field add

-gk <ip address of Call Server>, eg –gk 192.168.0.1

so that the Target path reads as per the following example:

“C:\Program Files\SpliceCom\PCS60\PCS60.exe” –gk 192.168.0.1

c Select OK. d Double click on the short cut to run the PCS 60.

Please note that entries made in the short cut Properties will override the entries made in the Preferences dialog box.

If the call server is to collect BLF information from this remote PCS 60 IP Phone the Remote BLF option should be enabled as follows.

1 From the File Menu select Preferences 2 Tick Remote BLF 3 Select OK 4 Close down the PCS 60 and run the software again.

It is recommended that no more than 5-6 remote IP phones should be configured to use this facility in order to keep the overhead on the Call Server to a minimum.

The PCS 60 can be run as partner to any PCS or analogue handset being used on the (PCS 520/505/10/5, alternative analogue handset, PCS 100, PCS 410/400, PCS 560, PCS 570 and PCS 580) giving the user the additional functionality provided by the PCS 60.

In order for the PCS 60 to partner a telephone the PCS 60 must connect to the port that the analogue handset is plugged into or to the relevant IP phone, therefore you will require the following information:

1 The IP address of the Phone Module or Call Server where the analogue port is located. This can be

viewed within Manager. Select Modules and then select the relevant module, the IP address is displayed, eg 192.168.0.249. or The IP address of the PCS 560 or 570. This can be viewed by selecting the Favourites button, then Settings and then Phone Information, eg 192.168.0.226 or The IP address of the PCS 580. This can be viewed within the Phone Information screen, which can be accessed by pressing the Settings icon 5 times. or The IP address of a PCS 410/400. This can be viewed by selecting the Information icon from the Home Page toolbar, eg 192.168.0.221 or

imiser system

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Configure a Phone’s Partner Login Code

Run the PCS 60 as a partner

Partner Login Code configured above.

max

The IP address of a PCS 100. This can be viewed by pressing the Down arrow once (or twice if Favourites are displayed) on the handset; the IP address being used by the handset will be displayed, eg 192.168.0.245. The IP address is also displayed during power up.

2 The reference number of the port:-

a Each analogue port on a Phone module or Call Server is given a reference number starting at

5001. Therefore port 1 is referenced as 5001, port 2 as 5002 etc.

b The port reference number for a PCS 580/570/560/410/400/100 will always be 5001.

3 The User’s Partner Login Code. This can be configured as follows:

1 In Manager select Users 2 Select the User who wishes to use the PCS 60 as a partner to their handset 3 Click on the entry within the Initial Phone field eg Phone Module 1.port01 4 You will be taken to the port assigned to this User 5 In the Partner Login Code field enter a 1-8 digit access code, eg 1234

6 Select Update or Apply when ready.

1 Run the PCS60.exe file either from the Installation folder, eg C:\Program Files\SpliceCom\PCS60, or

create a short cut to this file in the required location and double click on the short cut. (Please note that if you do not wish the PCS 60 to be loaded initially as an IP soft phone disconnect the PC from the network or turn off Auto Add Phones.)

2 The PCS 60 application will open. 3 From the File menu select Preferences 4 In the IP Address field enter the IP address of the Call

Server or Phone Module or IP Phone as described above, eg 192.168.0.249

5 In the Port field enter the port number that the

analogue phone is connected to, as described above, or 5001 for an IP phone.

6 In the Partner Login Code field enter the phone’s 7 Select OK when ready

8 Close down the PCS 60. 9 At this point your PC will need a network connection to the

imiser system.

10 Run the PCS 60 again.

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Run more than one version of PCS 60 on the same MS Windows PC

11 Once a connection has been made the name of User assigned to the port that the PCS 60 is

partnering is displayed in the Status Bar at the bottom left hand corner.

12 Lift the handset on the telephone to confirm that the connection between the PCS 60 and the

phone is set up correctly. A call entry will be displayed in Call Status.

Alternatively, the IP address, port number and partner login code can be entered in the Properties of a short cut to the PCS 60 software. This method should be used when you wish to run more than one version of the PCS 60 on the same PC as described below on page 63.

(Please note that any entry made in the short cut’s Properties will override entries in Preferences.)

1 Create a short cut to PCS60.exe. 2 Open the Properties of the short cut and in the Target

field add the following:

-pots <ip address of Phone module>:<port no>,<partner login code> or

-pots <ip address of Call Server>:<port no>,<partner login code> or

-pots <ip address of PCS 580/570/560/410/400/100>:<5001>,<partner login code>

so that the Target path reads as per the following example:

“C:\Program Files\SpliceCom\PCS60.exe” -pots 192.168.0.249:5010,1234

3 Select OK when ready 4 Double click on the short cut to run the PCS 60 application

When a member of staff is responsible for monitoring or using multiple handsets for example, multiple PCS 60 windows can be run on the same PC by selecting the Multiple Instances options within Preferences.

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Installing the PCS 60 software on a Apple Mac OS X PC

PCS 60 as an IP Soft Phone on an Apple Mac PC

Before starting:

Running the PCS 60:

1 From the File menu select Preferences 2 Tick the Multiple Instances field 3 Select OK.

Create a short cut to the PCS 60 software for each PCS 60 window required. Configure the short cut properties for each connection required.

The PCS 60 software will run on Mac OS X version 10.4 or above.

1 The set up file will be supplied in a compressed folder called, for example,

PCS60_MAC.3.2.67.pkg.zip. Open this folder.

2 Run (double click) the set up file called, for example, PCS60_MAC.3.2.67.pkg 3 At the Welcome screen select Continue to start the installation 4 At the Select a Destination screen select the Volume where you want the software loaded 5 By default the software will be loaded into the Applications folder. Change this location if required. 6 Select Continue when ready 7 At the Standard Install screen select Install. (If you wish at this point to change the Install Location

select Change Install Location.)

8 You will be prompted for your PC log in password, enter this in the Password field. 9 The software will be installed

10 At the Installation Completed screen select Close

1 The PC must have a network connection to a Call Server. 2 An IPUser licence must be previously installed and be available. (For further details please refer to

page 35.)

1 Run the PCS60 software either from Applications or create an Alias to this file in the required

location and double click on the Alias.

2 The PCS 60 application will open. 3 The “Searching for Server” message will be displayed as the registration process on the Call Server

is started.

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If the “Searching for Server” message remains and the PCS 60 does not continue to the next step check that the PC can communicate with the Call Server or refer to the Connect to the Call Server Remotely section below.

4 The “No Licenses Available” message will be displayed while the phone is registered as described

in step 5.

5 If the Auto Add Phones feature is turned on the Call Server will automatically register the PCS 60.

(Please refer to page 34 for further details.)

If the Auto Add Phones feature is turned off:

a In Manager select Unassigned Phones b The hardware address of the PC will identify the PCS 60 to be connected. Click on the entry

required. (If the PCS 60 is not listed check the installation of the IPUser licence as detailed from page 35.)

c From the Status list box select Member d Select Update when ready e The PCS 60 will now appear under Phones, identified by the hardware address of the PC

6 The “Security Access Denial” message will be displayed while an IPUser licence is assigned 7 A User account will be automatically created configured with the next available extension number.

The User’s Initial Phone field displays the PCS 60 that the User is assigned to, identified by the hardware address of the PC.

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Connect to the Call Server remotely

Remote BLF

PCS 60 on an Apple Mac PC as a partner to a User’s handset

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Before starting:

1 From the PCS60 menu select Preferences 2 In the IP Address field enter the IP address of the Call Server, eg 192.168.0.1 3 Tab to the next field

4 Close the window 5 Close down the PCS 60 and run the software again.

If the call server is to collect BLF information from this remote PCS 60 IP Phone the Remote BLF option should be enabled as follows.

1 From the PCS 60 Menu select Preferences 2 Tick Remote BLF 3 Close down the PCS 60 and run the software again.

It is recommended that no more than 5-6 remote IP phones should be configured to use this facility in order to keep the overhead on the Call Server to a minimum.

1 The IP address of the Phone Module or Call Server where the analogue port is located. This can be

imiser system

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Configure a Phone’s Partner Login Code

Run the PCS 60 as a partner

The IP address of a PCS 100. This can be viewed by pressing the Down arrow once (or twice if Favourites are displayed) on the handset, the IP address being used by the handset will be displayed, eg 192.168.0.245. The IP address is also displayed during power up.

2 The reference number of the port:-

a Each analogue port on a Phone module or Call Server is given a reference number starting at

5001. Therefore port 1 is referenced as 5001, port 2 as 5002 etc.

b The port reference number for a PCS 580/570/560/410/400/100 will always be 5001.

3 The User’s Partner Login Code. This can be configured as follows:

6 Select Update or Apply when ready.

1 Run the PCS60 software either from Applications or create an Alias to this file in the required

location and double click on the Alias. (Please note that if you do not wish the PCS 60 to be loaded initially as an IP soft phone disconnect the PC from the network or turn off Auto Add Phones.)

2 The PCS 60 application will open. 3 From the PCS60 menu select Preferences 4 In the IP Address field enter the IP address

of the Call Server or Phone Module or IP Phone as described above, eg

192.168.0.249

5 In the Port field enter the port number that the analogue phone is connected to, as described

above, or 5001 for an IP phone.

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Run more than one version of PCS 60 on the same Apple Mac PC

Operator Console Mode

6 In the Partner Login Code field enter the phone’s Partner Login Code configured above.Tab to the

next field

7 Close the window when ready 8 Close down the PCS 60. 9 At this point your PC will need a network connection to the

imiser system.

10 Run the PCS 60 again. 11 Once a connection has been made the name of User assigned to the port that the PCS 60 is

partnering is displayed in the Status Bar at the bottom left hand corner.

12 Lift the handset on the telephone to confirm that the connection between the PCS 60 and the

phone is set up correctly. A call entry will be displayed in Call Status.

When a member of staff is responsible for monitoring or using multiple handsets for example, multiple PCS 60 windows can be run on the same PC by making a copy of the software and entering the required settings in the Preferences form of each copy of the PCS 60 software. The Preference settings are saved with each copy of the software.

When a PCS 60 is being used by a receptionist or switchboard operator the Operator Console Mode can be enabled to provide the operator with additional data to assist with the efficient handling of calls. The additional data is displayed within the User’s Favourites screen and can display a User’s forwarding settings, DND settings, when the User is logged out, when the User is on a call and who they are talking to, when a User is listening to voicemail, the number of new voicemail messages awaiting a User/Department and the number of missed calls received by a User/Department.

The operator can also be given an additional window to display further Speed Dials when there is insufficient room in the Favourites pane.

For further details please refer to the PCS 60 User Manual.

The Operator Console Mode is configured as follows:

1 Install an Operator Console licence as described from page 35. 2 Open the User configuration form for the User who wishes to use Operator Console Mode. 3 Select the Licences page

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Administrator Access

PCS 50

Installing the PCS 50 software on a Windows PC

PCS 50 as an IP Soft Phone on a Windows PC

4 From the Operator Console Licence list box select

either:

a Yes - to apply the licence to this User and

enable Operator Console Mode, or

b PCS Partner only – this option can be used

5 Select Update or Apply when ready

Further configuration of the Operator Console Mode can be done within the Preferences dialogue box, please refer to the PCS 60 User Manual for further details.

A User of a PCS 60 can also be given the ability to change a User’s DND and Forwarding settings via their Favourites. To enable this facility the operator will require an Administrator name and password. Please refer to page 31 for information on how to create an Administrator. Please refer to the PCS 60 User Manual for information on how to configure this facility.

The PCS 50 application provides the same telephony functionality as available on the PCS 410/400. This application can run as an IP soft phone or be programmed to partner a User’s handset from their PC. The PCS 50 application can be installed on a Windows PC or Mac OS X.

Please refer to the PCS 400/410 and 50 User Manual for detailed descriptions of how to use this software.

1 Run the PCS 50 set-up file called, for example, PCS50.2.2.652.setup.exe 2 At the Setup screen select Yes to continue with the installation. 3 At the Welcome screen select Next to continue 4 Select where on the PC you wish the software to be loaded. By default, this will be C:\Program

Files/PCS50. Select Next when ready.

5 Enter the name that you wish to use for the Start Menu folder that will contain the PCS 50 short cut.

By default, this will be PCS50. Select Next when ready.

6 Select whether you wish a short cut to be created on your desktop. By default, this option will be

ticked. Select Next when ready.

7 The Ready to Install screen will confirm your previous choices. Select Back to make any changes.

Select Install when ready.

8 At the Completing the PCS Setup Wizard screen select Finish.

The PCS 50 software can be run as an IP soft phone where the PC becomes the User’s telephone and the voice path will be via the PC. Therefore the PC will require a headset, microphone, sound card etc installed.

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Before starting:

Running the PCS 50:

PCS 50 as a partner to a User’s handset on a Windows PC

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Installing the PCS 50 software on a Mac OS X

1 The PC must have a network connection to a Call Server. 2 An IPUser licence must be previously installed and be available. (For further details please refer to

page 35.)

1 Run the PCS.exe file either from the Installation folder, eg C:\Program Files\SpliceCom\PCS50, or via

the short cut created during the installation

2 The PCS 50 application will open. 3 The PCS 50 application will use the same method as the PCS 60 to register with the system as

described from step 3 on page 59.

Please note that the PCS 50 obtains the time from the DHCP Server.

The PCS 50 can be run to partner any PCS or analogue handset being used on the

imiser system (PCS 520/505/10/5, alternative analogue handset, PCS 100, PCS 410/400 and PCS 580/570/560) giving the user the additional functionality provided by the PCS 50.

The PCS 50 software uses the same method as the PCS 60 to partner an analogue handset as described from page 61.

Please note that the PCS 50 software is stored as PCS.exe in the Installation folder, eg C:\Program Files\SpliceCom\PCS50, or can be access via the short cut created during the installation

1 The set up file will be supplied in a compressed folder called, for example, PCS50.2.2.667.pkg.zip.

Open this folder.

2 Run (double click) the set up file called PCS50 3 At the Welcome screen select Continue 4 At the Select a Destination screen select the Volume where you want the software to be loaded 5 By default the software will be loaded into the Applications folder. Change this location if required.

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PCS 50 as IP Soft phone on a Mac OS X

PCS 50 as a Partner to a User’s handset on a Mac OS X

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PCS Operators Console

6 Select Continue when ready 7 At the Standard Install screen select Install. (If you wish at this point to change the Install Location

select Change Install Location.)

8 You will be prompted for your PC log in password, enter this in the Password field. 9 The software will be installed

10 At the Installation Completed screen select Close

The PCS 50 will use the same method as the PCS 60 to register with the system as detailed from page 64.

The PCS 50 can be run to partner any PCS or analogue handset being used on the

imiser system (PCS 520/505/10/5, alternative analogue handset, PCS 100, PCS 410/400, PCS 580/570/560) giving the user the additional functionality provided by the PCS 50.

The PCS 50 will use the same method as the PCS 60 to partner a User’s handset as detailed from page 66.

The PCS Operators Console application provides the most frequently used features and facilities of the PCS 50 on a single page, full screen display. This application has been specifically developed for Operators & Receptionists and can run as an IP soft phone or be programmed to partner a User’s handset from their PC. The PCS Operators Console application can be installed on PCs or laptops running the Microsoft Windows or Apple Mac OS X operating systems. Please refer to the PCS Operators Console manual for details on working with this application.

Configuration of the PCS Operators Console is identical to that for the PCS 50:

• To install the PCS Operators Console on a Windows PC please refer to page 69.

• To configure the PCS Operators Console as an IP soft phone on a Windows PC please refer to page 69.

• To configure the PCS Operators Console as a Phone Partner on a Windows PC please refer to page 70.

• To install the PCS Operators Console on a Apple Mac OS X platform please refer to page 70.

• To configure the PCS Operators Console as an IP soft phone on an Apple Mac OS X platform please

refer to page 71.

• To configure the PCS Operators Console as a Phone Partner on an Apple Mac OS X platform please

refer to page 71.

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Connecting Multiple Call Servers

Linking two Call Servers across a LAN

Please note that a valid Operator Console Licence must be purchased for each User wanting to use this application. Once the appropriate licences have been entered on the system, enable the Operator Console Licence tick-box for each User who will be using the application. This is located within the Licences page of the User’s configuration form.

Please refer to the Installing Licences section from page 35 for further information.

Two or more Call Servers can be connected to create one system where one will act as the Primary Call Server and all Call Servers connected to the Primary Call Server become Secondary Call Servers. This can be via the LAN or via a WAN link.

On power up, by default, each Call Server will use 192.168.0.1 as its IP address and act as a DHCP Server. If this feature is not required because, for example, 192.168.0.1 is already being used and another DHCP Server exists on the network do not connect the new Call Server to the network until this default configuration has been changed as described on page 28.

1 Ensure you are running the same version of software at both ends. 2 On the Primary Call Server:

a In Manager select Modules and select the Call Server from the list b You may wish to give the Call Server a unique name to help identify this Call Server eg Primary

Call Server

c Select the Call Server page and note the Administration Mode field is set to Primary

3 On the Secondary Call Server:

a Power up the Secondary Call Server b This Call Server must have a unique IP address and be on the same network as the Primary Call

Server. Therefore, if necessary, change the IP address of the Call Server as described on page

28. It is recommended that this address be outside the address range given out by the DHCP Server, eg 192.168.0.252

c You may wish to the rename the Call Server to help identify this Call Server, eg Secondary Call

Server.

d Power down your Call Server, as described on page 255. e Connect the Call Server to the Primary Call Server either via the network or directly to a LAN

port on the Primary Call Server.

f Power up the Call Server g Open Manager on the Secondary Call Server, eg http://192.16.0.252/manager h Select Utilities i Under the Commands list select Ask Master j Within the Enter IP Address field enter the IP Address of the Primary Call Server, eg 192.168.0.1 k Click on the Set Master button

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Linking two Call Servers across a WAN link

4 On the Primary Call Server:

a In Manager select Unassigned Modules b Select the new Call Server, eg Secondary Call Server c From the Module Status list box select Member d Select Update or Apply when ready e Select Modules, select the Secondary Call Server, select the Call Server page and note that the

Administration Mode is set to Secondary.

The Secondary Call Server will now be a part of the system controlled by the Primary Call Server. The Primary Call Server will replicate the relevant configuration information to the Secondary Call Server. This means that the configuration of the system can be administered from either Call Server.

1 Ensure you are running the same version of software at both ends. 2 On the Primary Call Server:

a This Call Server must be on different network to Secondary Call Server. Therefore, if necessary,

change the IP address of the Call Server as described on page 28, eg 194.168.0.1. If this Call Server is to act as the DHCP Server change the DHCP properties as described on page 29, eg

b Start Address = 194.168.0.2 c End Address = 194.168.0.250 d Base Address = 194.168.0.0 e In Manager, select Modules and select the Call Server from the list f You may wish to give the Call Server a unique name to help identify this Call Server eg Primary

Call Server

g Select the Call Server page and note the Administration Mode field is set to Primary

3 On the Secondary Call Server:

a Power up the Secondary Call Server b This Call Server must be on different network to Primary Call Server. Therefore, if necessary,

change the IP address of the Call Server as described on page 28, eg 192.168.1.1. If this Call Server is to act as the DHCP Server change the DHCP properties as described on page 29, eg

Start Address = 192.168.1.2 End Address = 192.168.1.250 Base Address = 192.168.1.0

c In Manager, select Modules and select the Call Server from the list. You may wish to the

rename the Call Server to help identify this Call Server, eg Secondary Call Server.

d In Manager select WAN Links and select the Secondary Call Server e Select the default WANLink. (For further details please refer to page 247.) f Select the IPRoutes page g Remove all the existing routes and add an entry to route back to the Primary Call Server, eg

IP Address = 194.168.0.0 IP Mask = 255.255.255.0 Metric = 1

h Power down your Call Server, as described on page 255. i Connect the Call Server to the Primary Call Server via your WAN link. j Power up the Call Server. k The WAN LEDs on both Call Servers will be solid if a connection has been made. You may

wish to ping the Primary Call Server to verify that you can communicate with the Primary, eg ping 194.168.0.1

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Connecting a Phone or Trunk Module to the Secondary Call Server

Connecting a PCS to the Secondary Call Server

Connecting an existing Primary Call Server to another Primary Call Server

l In Manager select Utilities then Commands m Under the Commands list select Ask Master n Within the Enter IP Address field enter the IP Address of the Primary Call Server, eg 192.168.0.1 o Click on the Set Master button

4 On the Primary Call Server:

a In Manager select Unassigned Modules. b Select the new Call Server, eg Secondary Call Server c From the Module Status list box select Member. d Select Update or Apply when ready e Select Modules. f Select the Secondary Call Server, select the Call Server page and note that the Administration

Mode is set to Secondary.

5 On the Secondary Call Server

a In Manager select the WAN Links. b Select the Secondary Call Server and select WANLink c Select the IPRoutes page and note that the default routes have reappeared. d Repeat step 3(g) above. e You may wish to ping the Primary Call Server to verify that you can communicate with the

Primary, eg ping 194.168.0.1

A Phone or Trunk module can be connected to the Secondary Call Server as described on page 39 and page 43.

The Call Server field within the Phone or Trunk Module’s configuration form will display the name of the Call Server that the module it is connected to, ie the name of the Secondary Call Server.

When connecting a Phone Module to a Secondary Call Server the additional Users created are all listed under the Users list and have taken the next available extension numbers. Their Initial Phone field displays which port on which Phone Module they are using and their Current Home field, which determines which Call Server is providing their voicemail service, displays the name of the Secondary Call Server.

A PCS 580/570/560, PCS410/400, PCS 100, PCS 60 IP phone or PCS 50 IP phone can be connected to the Secondary Call Server as described in the Connecting a PCS section from page 48.

The Initial Phone field within the User’s configuration form will display which Call Server the PCS is connected to, ie the name of the Secondary Call Server. Please refer to page 81 for further details.

Select Phones and then the relevant Call Server to display which PCSs are connected to this Call Server.

Where a Call Server has been previously working as a standalone system and is now to be connected to another system the same method described above is used however the following should be noted:

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A Global System

Primary Call Server

Secondary Call Server

• Any configuration (other than the IP address and Call Server Name) previously stored on the

Secondary Call Server will be lost as this Call Server becomes a part of the Primary Call Server’s system and takes its configuration from the Primary Call Server.

• Any Phone and/or Trunk modules and/or PCS IP phone should be disconnected from the Secondary

Call Server before it is connected to the Primary Call Server. They can be reconnected again as described above.

After the above procedures have taken place each Call Server, Phone Module, Trunk Module, PCS 580/ 570/560, 410/400, 100, PCS 60 IP phone and PCS 50 IP phone is part of the one system regardless of their location. The Primary Call Server will replicate the relevant configuration information to each Secondary Call Server.

Each Secondary Call Server will be responsible for passing on relevant configuration information to Phone and Trunk Modules, IP Phones etc that are directly connected to this Call Server. Any configuration information received from these Phone Modules, Trunk Modules or IP Phones etc will be passed back to the Primary Call Server by the Secondary Call Server for storage in the master database.

If the Primary Call Server fails all Secondary Call Servers will continue to operate and devices that were registered to the Primary Call Server then re-register to a Secondary Call Server (provided that they are on the same LAN). This will be the first Secondary Call Server to respond.

If a device, eg Phone Module or PCS IP phone, is connected to the system via the network the Primary Call Server will always respond the quickest to a registration request, followed by the Secondary Call Server with the smallest number of devices attached and so on. This will ensure that the load is shared among all Secondary Call Servers. Therefore, to ensure that any module or IP Phone connected to the Secondary Call Server does not register with the Primary Call Server before the linking process is complete power up the unit after the Secondary Call Server has connected to the Primary Call Server.

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User Status

System Time

To verify user registration across multiple Call Servers from Manager select Utilities and then User Status. This facility will display which Users are registered to which Call Server. All instructions are detailed within this page.

Each Call Server within one system can have a different date, time and Time Zone. Please refer to page 30 for further information.

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Configuring PBX functionality

Configuring an Analogue Extension Port

Caller Display Type

Physical analogue extension ports on a Phone module or Call Server can be configured as follows:

1 In Manager select Phones 2 Select the relevant Phone Module or Call Server 3 Select the port you wish to configure

The port is automatically assigned a name which cannot be changed. This name is used by a User’s Initial Phone field to indicate the port to which they are assigned. The User field displays the name of the User assigned to this port. Click on the name within the User field to view this User. Use the Cancel button to return to the port.

You may wish to enter a description to assist with identifying the port eg its physical location etc.

For a list of which ports have been assigned to which Users select Utilities and then Sorted Lists. This feature will also highlight a record where a port has been assigned to more than one User in error.

The following options can also be configured:

None – Call Display information will not be sent to the extension and the extension will not ring once after the relevant Phone Module has been rebooted Default – PCS 10 (suitable for most analogue handsets) Alarm – for use with alarm systems, DTMF is not sent out but allows the alarm system to send DTMF tones back out. Bell – USA BT – UK BTNoMessages – for use with InterQuartz Discovery analogue phones DTMF – Europe

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Caller Display Length

Offhook Working

Page Port

None

Paging

MusicOnHold

Remote PSTN

Fax Server Support

Fax – for use with a fax machine connected to this port, provides a strong/robust ring tone improving the fax machine’s ability to answer a call FaxServer – for Fax Server support, please refer to page 182 for further details. Lamp – for use with a PCS 5 to support the message waiting lamp Interquartz – for use with InterQuartz Gemini analogue phones Pay Phone – for use with a pay phone PCS 10 – for use with the PCS 10 PCS 10 Lamp – for use with a PCS 10 to support the message waiting lamp System Phone – for use with the PCS 520, ensures CLI information is displayed correctly and enables paging and auto answer functionality Trunk – Sends a disconnect signal to the end device.

By default the Caller Display Length is set to 0, which means 20 characters will be sent to the handset. Change this length when more or less characters are required. Set to 16 for the PCS 10.

The information sent will display the Caller information and then the Destination information, eg John Barrow – Support. The destination uses the first word of, eg the User name, and remaining characters are used for the Caller’s information.

Tick this box if a headset is used in conjunction with the phone connected to this port. This feature allows the use of a headset without picking up the receiver.

This field is used to determine the functionality of this analogue port. Select one of the following three options:

– this port will be used as a standard extension with an analogue phone.

– this port will be used by an external paging source, for example, a tannoy or PA system.

A Paging Port Adapter is required to connect the powered speakers (please refer to page

340 for further details). This port can be paged via a Dial Plan entry, an example of which is available on page 120. Alternatively, voicemail can be used play a message over this port as described on page 209.

refer to page

– this port will be used to provide Music On Hold from an external source. Please

177 for further details.

For information on the Remote PSTN and Remote Idle fields please refer to the Setting Up Remote PSTN

Operation section from page 111

1 Configure the Caller Display Type on the analogue extension port connected to the Fax Server to

“FaxServer”.

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Configuring an IP Phone

2 Create a new Group and add the User assigned to the above port. (Please refer to page 122 for

information on creating a Group.)

3 Create a new Department and in the Distribution Group add the Group created above. (Please

refer to page 128 for information on creating a Department.)

4 Create a DDI Call Plan entry routed to this Department. (Please refer to page 169 for information

on creating a DDI Call Plan.)

1 In Manager select Phones 2 Select the module that the PCS 60 IP Phone, PCS 50 IP Phone, PCS 100, 400, 410, 560, 570, 580 or IP

phone is connected to, eg Call Server

3 The MAC Address of the unit will identify the specific phone or PC 4 Select this entry if configuration is required.

You may wish to change the Name and/or the Description to assist with identifying the phone eg its physical location etc. This name is used by a User’s Initial Phone field to indicate the IP phone to which they are assigned. The User field displays the name of the User assigned to this port. Click on the name within the User field to view this User. Use the Cancel button to return to the IP Phone.

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Model

Screen Saver Mode

Screen Saver Params field

Bright always

The screen saver will not be used.

Working with Users

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For a list of which IP Phones have been assigned to which Users select Utilities and then Sorted Lists. This feature will also highlight a record where an IP Phone has been assigned to more than one User in error.

The following options can also be configured. Please note that the PCS 410/400 must be rebooted to activate any change of configuration.

This field should be set to ‘Colour640-480’ for a PCS 410/400; otherwise this field should remain in default (Unspecified).

This feature can be used with the PCS 580, 570, 560, 410 and 400 and is used to enable the appearance of a screen saver. This can either be after the handset has been idle for a specified amount of time or during a specified time. In all cases the screen will return to normal by making the handset active, eg on a PCS 570 by picking up the handset, on a PCS 580 by touching the screen.

1 In the Screen Saver Mode field select one of the following options:

Blank on Idle The screen saver will appear after the handset has been idle for the

number of seconds indicated in the Screen Saver Params field.

Blank at Night

The Screen Saver will appear between the times entered in the

2 In the Screen Save Params field enter the parameters required as

Night End Time

Time

eg 900,0,0 or 0,1730,0800 as follows:

Idle Timeout (Secs),Night Start

a If the Screen Save Mode field is set to Blank on Idle enter the number of seconds that the

handset is to remain idle before the screen saver will be appear, eg 900,0,0

b If the Screen Save Mode field is set to Blank at Night enter the time in 24 hour clock that the

screen saver will appear followed by the time in 24 hour clock that the screen saver will be removed eg 0,1730,0800

3 Select Update or Apply when ready

The Users option within Manager displays each user of the assigned to a physical analogue extension or IP Phone, or “floating” users who are not assigned to a specific location and will log onto a phone as and when required, or virtual users who do not need to use a physical phone but require the use of a feature of the system. PBX functionality is assigned to Users rather than to extension ports so that users of the system can log on to any phone and their User settings will follow them.

Please refer to page 245 for details on how to export a User’s list for management purposes.

imiser system. These can be Users

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Creating a new User

Configuring a User

When a 5100 or 5108 Call Server, 4140 Remote Call Server or Phone Module is connected to the system 4, 8, 15, 16 or 30 Users are automatically created and named, for example, Extn2017, Extn 2018 etc. Each User is assigned to an extension port, for example, Port01, Port02 etc, and given an extension number within the next available extension range, for example, 2017, 2018 etc.

When a PCS 60 (IP Phone), PCS 50 (IP Phone), PCS 100, PCS 410/400 or PCS 570/560 is installed a new User is automatically created and named, for example, Extn2031 and given the next available extension number, for example, 2031.

New Users can also be created manually, as follows, when a User is required but does not need to be assigned to a physical phone.

1 In Manager select Users 2 Select the Add button 3 In the Number of Users to Create field enter the number of Users you wish to create. 4 In the Starting Number field enter the extension number you wish to give the first User and any

remaining Users will be given the next number within the range. For example, if you wish to create 10 Users with an extension range of 2061-2070 enter 2061 in this field.

5 Un-tick the Virtual User field unless you wish these Users to be virtual Users and the relevant

licences have been installed (please refer to page 96 for further details on virtual Users).

6 Select Add Users 7 You will be returned to the Users List and the new Users created will be listed at the top and

named, for example, Extn2061, Extn2062 etc.

8 Each User can be selected and configured as describe below.

1 In Manager select Users 2 Click on the relevant User name to view the configuration of each specific User.

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Each User is assigned a Name which is used to identify them on the system, for example, when calls to and from this User are made the name will appear in Caller Display. It is recommended that this be changed to a unique, relevant name. The name should consist of alphanumerical characters only, not contain any punctuation and begin with a letter.

Each User is identified on the database by a GUID (Global Unique Identifier). This unique identifier can be displayed by selecting Utilities and then Voicemail GUIDs.

The Description field is a text field which can be used to further assist with identification of the User in Manager and via the Users directory accessed on the PCS 580/570/560/410/400/60/50, eg Personnel Director, Security, Reception etc

The Telephone Number field determines the internal number of the User’s telephone.

The Initial Phone field indicates the physical analogue port or IP phone that is assigned to this User. Leave this field blank if the User is to log on to an available phone (please refer to the Hot Desking section on page 93 for further information).

The User’s Locale specifies the language to be displayed on their IP phones. If left blank the System Locale will be used.

The Company field determines the Company to which this User is assigned. Please refer to the Working with Companies section from page 155 for further information.

The Current Home field determines which Call Server is providing the User with a voicemail service. Please refer to the Working with Voicemail section from page 187 for further information.

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Duplicate User Names and Extension Numbers

The User Class list box determines the configuration options available to a user of a PCS 60, PCS 50, PCS 580, PCS 570/560, 410/400 or PCS 100.

• The Restricted option disables the Users ability to change settings such as Forwarding, Call Waiting,

Do Not Disturb etc. On a PCS 60 and PCS 50 the Speed Dials and Ring Tunes tabs are the only tabs that are displayed within the User Configuration/Preferences form. On the PCS 60 the Quick Settings pane is not operational. On a PCS 410/400 the Speed Dials, Ring Tunes and Calibration tabs are the only tabs that are displayed within the User Configuration form. On the PCS 580 only the Group Membership and Speed Dial screens are available when selecting Settings. On the PCS 570/560 only the Logout, Favourites, Groups and the Phone Information options are available within the Settings screen. On the PCS 100 the Forwarding and DND options are removed. This option will also disable manual call recording.

• The Locked option will disable the use of the User Configuration/Preferences form completely on the

PCS 60 and PCS 50, and removes the Forwarding and DND options on the PCS 100. On the PCS 60 the Quick Settings pane is not operational. On the PCS 580 the Settings icon is disabled. On the PCS 570/560 the Setting options is removed from the Favourite screen. This option will also disable manual call recording.

• The HotelRoom option is for future use. However, if selected this will be the same as selecting

Locked.

The User Display Pref field determines the size of font to be displayed on the User’s PCS 410/400, if applicable.

The Contact Mode field determines the editing rights to a Contact entry via their PCS. For further information on Contacts please refer to page 141.

The Edit Mode field enables or disables the User’s ability to enter their personal numbers or change their Description field via their PCS 570, PCS 560, PCS 410/400, PCS 60 or PCS 50. Please refer to the Setting the User’s Details section from page 86 for further details.

The Ex Directory option, if ticked, will ensure that this User does not appear in the User Directory on a PCS 410, 400, 100 or 50. This is useful when creating a User that will not be dialled by users of the system.

The Disable System LNR tick box determines whether a User’s Last Number Redial list is stored on the call server or locally. By default, this data is stored on the call server. If this option is ticked the data is stored locally on the User’s phone but will be lost when their phone is rebooted.

It is essential that duplicate User Names and Extension Numbers are not used on a system. To ensure that duplicates have not been entered select Utilities and then Sorted Lists. This program will highlight entries where an identical name or extension number exists. It is also useful for quickly viewing which physical analogue port or PCS a User is assigned to and ensuring that the port or PCS has not been assigned to more than one User.

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Changing the extension numbers for multiple Users

Setting User functionality for multiple Users

Where multiple Call Servers are being run on one system the User Status facility will verify to which Call Server Users are registered. To view this facility select Utilities and then User Status.

When the extension number for more than one User is to be changed the Users (Renumber) feature can be used as follows:

1 In Manager, select Users (Renumber) 2 In the New Number Base field enter the first number of the new range to be used, for example, if

you wish to renumber the extensions from 6001 to 6050 enter 6001.

3 In the Target Users field enter the extension numbers to be changed. Enter either a comma

between each extension number or use a hyphen to indicate a range of extensions. For example: 2001,2003,2005,2009,2015,2020 etc or 2001-2050

4 Select Renumber 5 The extension numbers for the selected Users will be changed, for example, 2001 will become

6001, and the next extension will become 6002 and so on.

You may wish to turn on a particular feature or multiple features for more than one User, rather than turn on these feature in each User’s configuration form you can perform this task quickly via the Users (Bulk) feature in Manager.

1 In Manager, select Users (Bulk) 2 Select the feature(s) you wish to configure 3 Select the Continue button at the bottom of the

page

4 Select the option required for the relevant field from

the list box or tick box provided, or when a field name is displayed in blue the entry for this field can be selected from a list. Click on the field name and select the entry required.

5 In the Target Users field enter the extension

numbers for the Users to be changed. Enter either a comma between each extension number or use a hyphen to indicate a range of extensions. For example: 2001,2003,2005,2009,2015,2020 etc or 2001-2050

6 Select Update when ready. The Target Users will be updated with the changes selected.

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Music on Hold

Voicemail

User’s Dial Plan

Displaying a Web Page during a call

For information on assigning music on hold to a specific User please refer to page 179.

For information on voicemail functionality for a User please refer to page 187.

A Dial Plan determines how digits dialled on the User’s handset are processed. Therefore a Dial Plan must be assigned to a User to allow external dialling. A Dial Plan can also provide access to PBX functionality via short codes, use of Speed Dials, and be used to allow or bar calls to specific numbers.

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page 4 Select Dial Plan 5 From the list of Dial Plans displayed select the entry required 6 The selected Dial Plan will be displayed in the Dial Plan field 7 Select Update or Apply when ready.

For further information on working with Dial Plans please refer to page 114.

A web page can be displayed on a PCS 60, PCS 50 and PCS 410/400 involved in a call to or from a specific User. For example, a member of the Accounts department may require access to accounts records or a member of the Sales team may need access to sales information when dealing with calls to and/or from customers. This web page could be stored on the internal web server (please refer to page 184 for further details), the customer’s intranet or accessed externally if an Internet connection is available.

1 In Manager select Users 2 From the User list select the User required. 3 In the Auto URL field enter the path to the web page to be displayed, eg

www.splicecom.com/accounts or http://192.168.0.1/sales.php

4 From the Auto URL Mode list box select one of the following:

a NoAction – the web page specified in the Auto URL field can be viewed via the Information

icon on a PCS involved in a call to or from this User

b On Ring – the web page specified in the Auto URL field is automatically displayed on a PCS

involved in a call to or from this User while the call is ringing

c On Connect – the web page specified in the Auto URL field is automatically displayed on a

PCS involved in a call to or from this User once the call is answered

5 From the Auto URL Direction list box select one of the following:

a Bothway – the web page will be displayed on both PCSs involved in an internal call to or from

this User. The User’s PCS will also display the web page when an external call is either made or received by the User.

b In – the web page will only be displayed on the PCS receiving the internal call either to or

from this User. The User’s PCS will also display the web page when an external call is received by the User.

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Displaying a Directory

Setting the User’s Details

not

c Out – the web page will only be displayed on the PCS making the internal call either to or

from this User. The User’s PCS will also display the web page when an external call is made by the User.

6 Select Update or Apply when ready

A list of the Users or Departments on the system or the User’s speed dials can be displayed on a PCS 50 or PCS 410/400 when a call is made to and/or received by the User to enable the call to be processed efficiently. (For further information on a User’s speed dials please refer to page 102.)

1 In Manager select Users 2 From the User list select the User required. 3 From the Auto URL Mode list box select one of the following:

a Favourites – the user’s Favourites (Speed Dials) are automatically displayed on a PCS involved

in a call to or from this User once the call is answered

b Users – the Users Directory is automatically displayed on a PCS involved in a call to or from this

User once the call is answered

c Departments – the Department Directory is automatically displayed on a PCS involved in a call

to or from this User once the call is answered

4 From the AutoURL Direction list box select one of the following:

a Bothway – the directory will be displayed on both PCSs involved in an internal call to or from

this User. The User’s PCS will also display the directory when an external call is either made or received by the User.

b Incoming – the directory will only be displayed on the PCS receiving the internal call either to

or from this User. The User’s PCS will also display the directory when an external call is received by the User.

c Outgoing – the directory will only be displayed on the PCS making the internal call either to or

from this User. The User’s PCS will also display the directory when an external call is made by the User.

5 Select Update or Apply when ready.

Please note: If the User is making a call to or receiving a call for a Department the Auto URL and Auto URL Mode settings for the Department take priority. If the User is making a call to or receiving a call from a Contact the Auto URL and Auto URL Mode settings for the Contact take priority.

A User’s Home and Mobile telephone numbers entered via the Details page, allow other Users to speed dial these numbers from a PCS 580, 570, 560, 410/400, 100, 60 and 50. Spare1 and Spare2 Telephone Number fields can be used to enter two additional numbers if required. (Please note: If the Ex Directory option is ticked this User will not appear in the User Directory available via a PCS 580, 570, 560, 410/400, 100, 60 and 50.)

These numbers will also be matched against incoming CLI and display the User’s name in Caller Display. It is therefore very important that extension numbers are also use these numbers as explained in the Voicemail section from page

entered into these fields. Voicemail can

187.

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Using Follow Me

Setting Follow Me

Users can also add and edit their own personal numbers via their PCS 580, 570, PCS 560, PCS 410/400, PCS 60 or PCS 50. This facility is controlled by the Edit Mode field under the General Page. By default this is set to All which gives the User full rights to edit his Number fields. However this right can be disabled by selecting No Personal Numbers from the list box.

The No Personal Numbers option will also disable the User’s ability to edit his Description field via a PCS 410/400, PCS 60 or PCS 50.

The Home Page field determines which web page will be displayed as default on the User’s PCS 410/400, PCS 60 or PCS 50. This could be a page stored on the internal web server (please refer to the Internal Web Server section from page 184 for further details), the customer’s intranet or accessed externally if an Internet connection is available.

The Follow Me feature will allow a User to take calls, for example, on their mobile or at home, when they are out of the office or at another extension if, for example, the User is spending the day in another part of the building or in another office.

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page 4 From the Follow Me list box select one of the following options:

a None – calls will not be forwarded b Personal – all calls to the User’s extension number and DDI will be forwarded. c Dual Personal - all calls to the User’s extension number and DDI will be forwarded, and the

User’s extension will ring. (Calls to the User’s Department(s) will ring on the User’s extension.)

d All - all calls to the User’s extension number and DDI will be forwarded plus any calls to a

Department to which the User is a member.

e Dual All - all calls to the User’s extension number and DDI will be forwarded plus any calls to a

Department to which the User is a member, and the User’s extension will ring.

f Personal External - all external calls to the User’s extension number and DDI will be forwarded.

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Turn off Follow Me

g Dual Personal External - all external calls to the User’s extension number and DDI will be

forwarded, and the User’s extension will ring. (Calls to the User’s Department(s) will ring on the User’s extension.)

h All External - all external calls to the User’s extension number and DDI will be forwarded plus

any calls to a Department to which the User is a member.

i Dual All External - all external calls to the User’s extension number and DDI will be forwarded

plus any calls to a Department to which the User is a member, and the User’s extension will ring.

j Personal Internal - all internal calls to the User’s extension number and DDI will be forwarded. k Dual Personal Internal - all internal calls to the User’s extension number and DDI will be

forwarded, and the User’s extension will ring. (Calls to the User’s Department(s) will ring on the User’s extension.)

l All Internal - all internal calls to the User’s extension number and DDI will be forwarded plus

any calls to a Department to which the User is a member.

m Dual All Internal - all internal calls to the User’s extension number and DDI will be forwarded

plus any calls to a Department to which the User is a member, and the User’s extension will ring.

5 In the Follow Me To field enter the number to which calls should be routed. This can be an

internal extension or an external number eg a mobile number. To forward calls to a Department precede the extension number with a colon eg :8000

6 Select Update or Apply when ready.

Calls to this extension will now follow the User to the number configured and the following will take effect: -

• If the Follow Me To number also has Follow Me set the call will be forwarded on.

• If the Follow Me To number is not answered within the original User’s No Answer Time the call will be

routed to the original User’s voicemail, no further forwarding occurs.

• If the Follow Me To number is not answered within the original User’s No Answer Time and the

original user has Forward on No Answer set the call will be forwarded to this number.

• If the Follow Me To number is busy the call will be routed to the original User’s voicemail, no further

forwarding occurs.

• If the Follow Me To number is busy and the original User has Forward on Busy set this will be ignored,

the call will be routed to the original User’s voicemail, no further forwarding occurs.

• If the Follow Me To number is a Department extension number and a call is not answered and is

routed to voicemail this will be the Department’s voicemail not the original User’s voicemail

• If an external Follow Me To number has a voicemail service, eg a mobile, and this voicemail service

answers the call no further forwarding will occur.

• A broken dial tone on the User’s handset may indicate that Follow Me has been set.

• Dual Personal and Dual All options are useful for staff that regularly move away from their desks, for

example, IT support staff. A call can be answered either at their desk or on, for example, their mobile, whichever is the easiest at the time.

1 In Manager select Users 2 Select the User you wish to configure

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Using Forwarding

Setting Forward on Busy

3 Select the Telephony page 4 From the Follow Me list box select None 5 The Follow Me To number can remain for future use 6 Select Update or Apply when ready

Please note: The above Follow Me features can be set by the User via their PCS 60, 50, 100, 410/400 or 580/570/560, or via a Dial Plan entry. If a User sets a Follow Me option but does not enter a Follow Me To number the setting will be ignored.

The Forwarding feature allows Users to ensure calls are answered when either they are not at their desks or they are busy on another call.

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page 4 From the Forward On Busy list box select one of the following:

a Internal&External - all calls to the User’s extension number and DDI will be passed to the

Forward On Busy To number

b Internal –internal calls to the User’s extension number and DDI will be passed to the Forward

On Busy To number. External calls will be passed to voicemail, if enabled, or will receive the busy tone.

c External – external calls to the User’s extension number and DDI will be passed to the Forward

On Busy To number. Internal calls will be passed to voicemail, if enabled, or will receive the busy tone.

5 In the Forward on Busy To field enter the number to which calls should be routed when this User is

busy. This can be an internal extension or an external number eg a mobile number. To forward calls to a Department precede the extension number with a colon eg :8000

6 To activate this feature tick Forward on Busy 7 Select Update or Apply when ready

Calls to this extension will now be forwarded when the User is busy and the following will take effect: -

• If the Forward on Busy number has Follow Me set the call will be forwarded on.

• If the Forward on Busy number is not answered within the original User’s No Answer Time the call will

be routed to the original User’s voicemail, no further forwarding occurs.

• If the Forward on Busy number is busy and Forward on Busy is set for that extension the call will be

forwarded on.

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Turn off Forward on Busy

Setting Forward on No Answer

• If the Forward on Busy number is busy and Forward on Busy is not set for that extension the call will

be routed to the original User’s voicemail.

• If the Forward on Busy number is a Department extension number and is not answered and is routed

to voicemail this will be the Department’s voicemail not the original User’s voicemail

• If an external Forward on Busy number has a voicemail service, eg a mobile, and this voicemail service

answers the call no further forwarding will occur.

• A broken dial tone on the User’s handset may indicate that Forwarding has been set.

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page 4 From the Forward on Busy list box select None 5 The Forward on Busy To number can remain for future use 6 Select Update or Apply when ready

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page 4 From the Forward On No Answer list box select one of the following:

a Internal&External - calls to the User’s extension number and DDI will be passed to the Forward

On No Answer To number if a call is not answered within the User’s No Answer Time.

b Internal – internal calls to the User’s extension number and DDI will be passed to the Forward

On No Answer To number if a call is not answered within the User’s No Answer Time. External calls will be passed to voicemail, if enabled, if a call is not answered within the User’s No Answer Time.

c External – external calls to the User’s extension number and DDI will be passed to the Forward

On No Answer To number if a call is not answered within the User’s No Answer Time. Internal calls will be passed to voicemail, if enabled, if a call is not answered within the User’s No Answer Time.

5 In the Forward on No Answer To field enter the number to which calls should be routed when this

extension is not answered. This can be an internal extension or an external number eg a mobile number. To forward calls to a Department precede the extension number with a colon eg :8000

6 Select Update or Apply when ready.

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Turn off Forward on No Answer

Setting the No Answer Time

Using Call Waiting

The amount of time a User’s extension will ring before it is considered not answered is determined by the User’s No Answer Time field as described below on page 91.

Calls to this extension will now be forwarded after the User’s No Answer Time and the following will take effect: -

• If the Forward on No Answer number has Follow Me set the call will be forwarded on.

• If the Forward on No Answer number is also not answered within the original User’s No Answer Time

the call will be routed to the original User’s voicemail, no further forwarding occurs.

• If the Forward on No Answer number is busy and Forward on Busy is set for that extension the call will

be forwarded on.

• If the Forward on No Answer number is busy and Forward on Busy is not set for that extension the call

will be routed to the original User’s voicemail.

• If the Forward On No Answer number is a Department extension number and is not answered and is

routed to voicemail this will be the Department’s voicemail not the original User’s voicemail

• If an external Forward number has a voicemail service, eg a mobile, and this voicemail service answers

the call no further forwarding will occur.

• A broken dial tone on the User’s handset may indicate that Forwarding has been set.

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page 4 From the Forward on No Answer list box select None 5 The Forward on No Answer To number can remain for future use 6 Select Update or Apply when ready

Please note: The above Forwarding features can be set by the User via their PCS 60, 50, 100, 410/400, or via a Dial Plan entry. If a User sets a Forwarding option but does not enter a Forwarding number the setting will be ignored.

The No Answer Time is the amount of time the User will be given to answer their extension before the call is routed to voicemail or to their Forward on No Answer number, if configured. It is important to determine the amount of time that will be suitable to the User. By default, this is set to 20 seconds.

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page. 4 In the No Answer Time field enter the number of seconds required 5 Select Update or Apply when ready

Call Waiting will allow the User to receive a second call and to toggle between the two calls. When this feature is enabled the User will hear an intermittent beep in the headset when a second call is received.

A PCS 60, 50, 410/400 or 580/570/560 will also indicate when a call is waiting and will enable the User to toggle between the two calls.

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Configuring Call Waiting

A User may prefer not to hear the intermittent beep; this can be achieved by selecting the NoBeep option. If a User of a PCS 60 or PCS 50 as a partner to their handset would prefer not to hear the intermittent beep in the headset this can be achieved by selecting the PhoneTools option. The intermittent beep can still be heard via the PCS 60 or PCS 50.

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page 4 From the Call Waiting list box select one of the following options:

a Disable (default) – the Call Waiting feature is not set for this User. b Enable – the User can receive a second call and toggle between the two calls. The User will

be informed of a second call by an intermittent beep in the headset and via their PCS 60, 50, 410/400 or 580/570/560.

c Phone Tools – if the User is running a PCS 60 or PCS 50 as a partner to their handset this

option will enable Call Waiting and the intermittent beep will not be heard in the headset but can be heard via the PCS 60 or PCS 50. If the User is not using a PCS 60 or PCS 50 as a partner this option is the same as setting the Enable option.

d NoBeep – enables Call Waiting but disables the intermittent beep.

5 Select Update or Apply when ready

Users can also set this feature via their PCS 60, 50 or 410/400 (please refer to the relevant User manual for further details) or via a Dial Plan entry similar to the following examples:

The Translate To fields are empty as the Dial Plan entry will only enable or disable Call Waiting for the User on whose handset this short code is dialled. On dialling these short codes a recorded message will be played confirming that Call Waiting has been enabled or disabled.

For further information on working with Dial Plans please refer to page 114.

Call Waiting to an analogue handset can be handled as follows:

1 First call is in progress and the User is informed of a call waiting by the intermittent beep 2 Use the Hold facility to place the first call on hold 3 The second call will be automatically presented 4 Use the Hold facility to end the second call and return to the first caller.

If the User replaces the handset instead of following Step 4 the two calls will be connected together.

(Please refer to the relevant User Manual for details on handling a call waiting via a PCS 60, 50, 100, 410/400 or 580/570/560.)

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Call Barring

Using Hot Desking

max

Creating a Hot Desking User

6 Select Add Users

Users can be barred from making calls to specific numbers or for all external calls.

To bar calls to all external numbers:

1 In Manager select Users 2 Select the User you wish to configure 3 Select the Telephony page 4 Tick Outgoing Call Bar 5 Select Update or Apply when ready

Please note: When this feature is set the User cannot make a call to any external number including emergency services. Therefore it is important that their Dial Plan contains an entry using the Dial Emergency action to allow the User to dial emergency services. Please refer to the Working with Dial Plans section from page 114 for further details.

To bar calls to specific numbers please refer to the Working with Dial Plans section from page 114 for further details.

Hot Desking allows Users who may not require a permanent physical extension the ability to use all the facilities provided by the PBX, eg voicemail, forwarding etc, and the ability to log on to a physical phone when in the office, for example. An IPUser licence is required to create a new User on the (please refer to the Licensing section on page

35 for further information).

imiser

Hot Desking Users are given a unique extension number and a Login Code to enable them to log on to a physical phone, which could be an analogue handset or a PCS, as and when required. A Login Idle Time can be used to specify how long an extension is idle before that User is automatically logged off.

1 In Manager select Users 2 Select the Add button

3 In the Number of Users to Create field enter 4 In the Starting Number field enter the extension

number you wish to give the User, eg 2040.

5 Untick the Virtual User field. 7 You will be returned to the Users List and the new User will be displayed at the top and will be

called, eg Extn2040.

8 Select the User and this User can now be configured as required, for example, you may wish to

change the User’s name and enter a description to further identify this User if required.

9 Select Apply to save any changes made.

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number and Login Code.

extension no*login code#

10 Select the Telephony page 11 In the Login Access Code field enter a number (max 8 digits) that will be the User’s passcode. 12 If required, in the Login Idle Time field, enter the number of seconds this User will be idle before

being automatic logged off. (By default this is set to 0 meaning that the User will not be automatically logged off and will need to manually log off.)

13 Select Update or Apply when ready

The User’s Login Access Code field can contain an asterisk (*). This means any number can be entered during the log in process.

The User will now be able to log on to :-

1 An analogue handset via a Dial Plan entry, similar to the following examples:

The Translate to field contains the User’s extension

In this example the User must enter

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Specifications and Main Features

Frequently Asked Questions

User Manual

Contents

Introduction

Platform Overview

5 Series Hardware

5100 Call Server

5300 Phone

5108 Call Server

4 Series Hardware

4100 Call Server

4200 Trunk

4300 Phone

4140 Remote Call Server

4400 Voice Compression Module

PCS 580G

PCS 570

PCS 570G

PCS 560

PCS 410

PCS 400

PCS 100

PCS 520

PCS 10

PCS 505

PCS 5

Power over Ethernet PSU

PCS PSU

Single Terminal Ethernet Power Supply (STEPS)

Voice Compression Card

PCS 60

PCS 50

PCS Operators Console

Manager

Voicemail

Enhanced Speech Processing (ESP)

Vision

System Architecture

Working with the System

System Directory

Primary Mode

Secondary Mode

Gatekeeper

Trunks

Distributed Phone Modules

Computer Telephony

Distributed Voicemail

Extendibility

Manageable Objects

Users

Departments

Groups

Contacts

Meet Me Conferences

Dial Plans

Time Plans

Companies

Trunks

Trunk Groups

DDI Call Plans

LCR Plans

Voicemail Ports

Music Channels

Events

VXML Scripts

Auto Attendant

Modules

Phones

Unassigned Phones

Unassigned Modules

System

Licences

Administrators

Utilities

Area Codes

WAN Links

Installing a System

5100 Call Server – Front and Rear view