Feature Manual

Business Communication Server

Model No. KX-NSX1000

KX-NSX2000

Thank you for purchasing this Panasonic product.

Please read this manual carefully before using this product and save this manual for future use.
In particular, be sure to read "1.1.1 For Your Safety, page 18" before using this product.

KX-NSX series: PNMPR Software File Version 004.00000 or later

Manuals and supporting information are provided on the Panasonic Web site at:
https://panasonic.net/cns/pcc/support/pbx/
Introduction

Introduction
About this Feature Manual
This Feature Manual is designed to serve as an overall feature reference for the Panasonic Business
Communication Server.
It explains what this Business Communication Server can do, and how to obtain the most out of its many
features and facilities.
In this manual
• The PBX functions provided by Business Communication Server are described here just "PBX" in this
manual.

The Structure of this Manual

This manual contains the following sections:
Section 1, For Your Safety
Provides details about safety precautions for preventing personal injury and/or damage to property.
Section 2, Call Handling Features
Provides details about the call handling features.
Section 3, Unified Messaging System
Provides details about the features of the Unified Messaging system.
Section 4, Network Features
Provides details about public and private networks you can connect the PBX to.
Section 5, System Configuration and Administration Features
Provides details about the system configuration and administration features.
Section 6, Appendix
Provides tables listing capacity of system resources, exclusive features for each PBX model, tone and
ring tone tables, and the revision history of this Feature Manual.

Compatible Telephones and Devices

A list of telephone and device types that are compatible with this PBX can be found in the Installation
Manual. However, if you expand the capabilities of this PBX by connecting a KX-TDA, KX-TDE series PBX,
using Multi-Connection function, the following types of telephones, devices, and trunks can also be used:
• Digital Proprietary Telephones (DPT)
• Analogue Proprietary Telephones (APT)
• Direct Station Selection Consoles (DSS Consoles)
• T1 off-premise extensions (T1-OPX)
• T1/E1 trunks
• DID trunks
• E&M trunks (KX-TDE/KX-TDA series only)
• CLCOT trunks (KX-TDA100D only)
• ELCOT trunks (KX-TDE600/KX-TDA600 only)
These types of telephones, devices, and trunks are listed in the manual for various features. However, be
aware that they are only applicable if a stacking connection is established. For details about stacking, see
"5.3.1 Stacking Connection".

Functional Limitation
KX-NSX Series treats the KX-UT series phone as a general-purpose SIP phone.

2 Feature Manual
 Introduction

Depending on the NSX server’s software version, some features may not function. For details about which
versions support these features, consult your dealer.
• Operation of the KX-NTV series depends on the NSX server software file version and the firmware
version of the KX-NTV series. You can confirm the compatibility on the Panasonic Web site:
https://panasonic.net/cns/pcc/support/pbx/
Video phone calls to KX-HDV430 SIP phones can be established.
Video phone calls between KX-NTV series devices and KX-HDV series SIP phones can be established.
Video phone calls are also supported when an incoming call arrives at an incoming call distribution group to
which the KX-HDV series belongs and then the KX-HDV series responds to the call. This is assuming a
case in which multiple operators are using the KX-HDV series.
However, video phone calls have several limitations as listed below:
• The encryption (SRTP) of the video phone call’s media stream will not function
• Once a call is established, there is no guarantee that the PBX functionality provided by the phone
operating system will work
• We cannot guarantee that third party SIP phones which can receive video phone calls will work with video
phone calls from the KX-HDV series/KX-NTV series

References Found in the Feature Manual

Installation Manual References
The required installation instruction titles described in the Installation Manual are noted for your
reference.
PC Programming Manual References
The PC Programming titles and parameters described in the PC Programming Manual are noted for
your reference.
Feature Manual References
The related feature titles described in this Feature Manual are noted for your reference.
Operating Manual References
The operation required to implement the feature described in the Operating Manual is noted for your
reference.
Abbreviations
There are many abbreviations used in this manual (e.g., "PT", for proprietary telephone). Please refer to
the list in the next section for the meaning of each abbreviation.

About the other manuals

Along with this Feature Manual, the following manuals are available to help you install, and use this NSX
server:
Installation Manual
Provides instructions for installing the hardware and maintenance of the NSX server.
PC Programming Manual
Provides step-by-step instructions for performing system programming using a PC.
Operating Manual
Provides operating instructions for end users using IP-PTs, SIP phones, SLTs, PSs, or DSS Consoles.

Other Information
Trademarks
• Microsoft, Active Directory, and Outlook are either registered trademarks or trademarks of Microsoft
Corporation in the United States and/or other countries.

Feature Manual 3
Introduction

• The Bluetooth® word mark and logos are registered trademarks owned by the Bluetooth SIG, Inc., and
any use of such marks by Panasonic Corporation is under licence.
• All other trademarks identified herein are the property of their respective owners.

Note
• The contents of this manual apply to NSX servers with a certain software version, as indicated on the
cover of this manual. To confirm the software version of your NSX server, refer to "How do I confirm
the software version of the NSX server or installed cards?" in 2.3 Frequently Asked Questions (FAQ)
of the PC Programming Manual.
• Some optional hardware, software, and features are not available in some countries/areas, or for
some PBX models. Please consult your certified Panasonic dealer for more information.
• Product specifications are subject to change without notice. In some cases, additional information,
including updates to this and other manuals, is included in the Maintenance Console’s Information
before programming. Install the latest version of Maintenance Console to view this information.
• Throughout this manual, PT displays and other displays are shown in English. Other languages may
be available, depending on the country or area.
• In this manual, the suffix of each model number (e.g., KX-NSX1000BX) is omitted unless necessary.
• Since KX-NT630 and KX-NT680 are equivalent to the KX-NT500 series, read the KX-NT500 series as
the KX-NT600 series in this manual.

4 Feature Manual
 List of Abbreviations

List of Abbreviations
A CONP
Connected Name Identification Presentation
AA
Automated Attendant CONR
Connected Name Identification Restriction
ACD
Automatic Call Distribution COS
Class of Service
ANI
Automatic Number Identification CPC
Calling Party Control
AOC
Advice of Charge CS
Cell Station
APT
Analogue Proprietary Telephone CT
Call Transfer—by ISDN
ARS
Automatic Route Selection CTI
Computer Telephony Integration

B
D
BGM
Background Music DDI
Direct Dialling In
BRI
Basic Rate Interface DHCP
Dynamic Host Configuration Protocol

C DID
Direct Inward Dialling
CCBS
Completion of Calls to Busy Subscriber DIL
Direct In Line
CDPG
Call Distribution Port Group DISA
Direct Inward System Access
CF
Call Forwarding—by ISDN DND
Do Not Disturb
CLI
Calling Line Identification DPT
Digital Proprietary Telephone
CLIP
Calling Line Identification Presentation DSS
Direct Station Selection
CLIR
Calling Line Identification Restriction DTMF
Dual Tone Multi-Frequency
CNIP
Calling Name Identification Presentation
E
CNIR
Calling Name Identification Restriction EFA
External Feature Access
COLP
Connected Line Identification Presentation
F
COLR
Connected Line Identification Restriction FWD
Call Forwarding

Feature Manual 5
List of Abbreviations

G P-P
Point-to-Point
G-CO
Group-CO P-SIP
Panasonic SIP Phones
- KX-HDV series/KX-TGP600
I - KX-NTV150 (Communication IP camera)
ICD - KX-NTV160 (Video door phone)
Incoming Call Distribution - KX-UCMA (Mobile Softphone)

ICMP P2P
Internet Control Message Protocol Peer-to-Peer

IP-PT PDN
IP Proprietary Telephone Primary Directory Number

IRNA PIN
Intercept Routing—No Answer Personal Identification Number

ISDN PING
Integrated Services Digital Network Packet Internet Groper
PRI
L Primary Rate Interface

L-CO PS
Loop-CO Portable Station

LCS PT
Live Call Screening Proprietary Telephone

LED
Light Emitting Diode S
S-CO
M Single-CO

MCID SDN
Malicious Call Identification Secondary Directory Number

MSN SIP
Multiple Subscriber Number Session Initiation Protocol
SLT
N Single Line Telephone

NTP SMDR
Network Time Protocol Station Message Detail Recording
SNMP
O Simple Network Management Protocol

OGM
Outgoing Message T
OHCA TAFAS
Off-hook Call Announcement Trunk Answer from Any Station
TEI
P Terminal Endpoint Identifier

P-MP TRG
Point-to-multipoint Trunk Group

6 Feature Manual
 List of Abbreviations

TRS/Barring
Toll Restriction/Call Barring

U
UCD
Uniform Call Distribution
UM
Unified Messaging
UPS
Uninterruptible Power Supply

V
VM
Voice Mail
VoIP
Voice over Internet Protocol
VPN
Virtual Private Network

X
XDP
EXtra Device Port

Feature Manual 7
List of Abbreviations

8 Feature Manual
 Table of Contents

Table of Contents
1 For Your Safety ..................................................................................... 17
1.1 For Your Safety .............................................................................................................. 18
1.1.1 For Your Safety ............................................................................................................. 18
2 Call Handling Features ......................................................................... 21
2.1 Incoming Call Features ................................................................................................. 22
2.1.1 Incoming Trunk Call Features ...................................................................................... 22
2.1.1.1 Incoming Trunk Call Features—SUMMARY .............................................................. 22
2.1.1.2 Direct In Line (DIL) ..................................................................................................... 25
2.1.1.3 Direct Inward Dialling (DID)/Direct Dialling In (DDI) .................................................. 26
2.1.1.4 Multiple Subscriber Number (MSN) Ringing Service ................................................. 29
2.1.1.5 Calling Line Identification (CLI) Distribution ............................................................... 32
2.1.1.6 Intercept Routing ....................................................................................................... 33
2.1.1.7 Intercept Routing—No Destination ............................................................................ 36
2.1.2 Internal Call Features ................................................................................................... 37
2.1.2.1 Internal Call Features—SUMMARY .......................................................................... 37
2.1.2.2 Internal Call Block ...................................................................................................... 38
2.1.3 Incoming Call Indication Features ................................................................................ 39
2.1.3.1 Incoming Call Indication Features—SUMMARY ....................................................... 39
2.1.3.2 Ring Tone Pattern Selection ...................................................................................... 40
2.1.3.3 Call Waiting ................................................................................................................ 41
2.2 Receiving Group Features ............................................................................................ 43
2.2.1 Idle Extension Hunting .................................................................................................. 43
2.2.2 Incoming Call Distribution Group Features ................................................................... 44
2.2.2.1 Incoming Call Distribution Group Features—SUMMARY .......................................... 44
2.2.2.2 Group Call Distribution ............................................................................................... 47
2.2.2.3 Queuing Feature ........................................................................................................ 50
2.2.2.4 Distribution Order ....................................................................................................... 53
2.2.2.5 VIP Call ...................................................................................................................... 54
2.2.2.6 Overflow Feature ....................................................................................................... 56
2.2.2.7 Log-in/Log-out ............................................................................................................ 58
2.2.2.8 Supervisory Feature .................................................................................................. 60
2.2.2.9 Supervisory Feature (ACD) ....................................................................................... 62
2.3 Call Forwarding (FWD)/Do Not Disturb (DND) Features ............................................ 68
2.3.1 Call Forwarding (FWD)/Do Not Disturb (DND)—SUMMARY ....................................... 68
2.3.2 Call Forwarding (FWD) ................................................................................................. 68
2.3.3 Do Not Disturb (DND) ................................................................................................... 73
2.3.4 FWD/DND Button, Group FWD Button ......................................................................... 74
2.4 Answering Features ...................................................................................................... 77
2.4.1 Answering Features—SUMMARY ................................................................................ 77
2.4.2 Line Preference—Incoming .......................................................................................... 77
2.4.3 Call Pickup .................................................................................................................... 78
2.4.4 Hands-free Answerback ............................................................................................... 82
2.5 Making Call Features .................................................................................................... 84
2.5.1 Predialling ..................................................................................................................... 84
2.5.2 Automatic Extension Release ....................................................................................... 84
2.5.3 Intercom Call ................................................................................................................ 84
2.5.4 Trunk Call Features ...................................................................................................... 86
2.5.4.1 Trunk Call Features—SUMMARY ............................................................................. 86
2.5.4.2 Emergency Call ......................................................................................................... 86
2.5.4.3 Account Code Entry ................................................................................................... 87
2.5.4.4 Dial Type Selection .................................................................................................... 88

Feature Manual 9
Table of Contents

2.5.4.5 Reverse Circuit .......................................................................................................... 88

2.5.4.6 Trunk Busy Out .......................................................................................................... 89
2.5.4.7 Pause Insertion .......................................................................................................... 90
2.5.4.8 Host PBX Access Code (Access Code to the Telephone Company from a Host PBX)
.................................................................................................................................... 90
2.5.4.9 Special Carrier Access Code ..................................................................................... 92
2.5.5 Seizing a Line Features ................................................................................................ 92
2.5.5.1 Seizing a Line Features—SUMMARY ....................................................................... 92
2.5.5.2 Line Preference—Outgoing ....................................................................................... 93
2.5.5.3 Trunk Access ............................................................................................................. 94
2.6 Memory Dialling Features ............................................................................................. 96
2.6.1 Memory Dialling Features—SUMMARY ....................................................................... 96
2.6.2 One-touch Dialling ........................................................................................................ 97
2.6.3 Last Number Redial ...................................................................................................... 98
2.6.4 Speed Dialling—Personal/System .............................................................................. 100
2.6.5 Speed Dialling—External (LDAP) ............................................................................... 102
2.6.6 Quick Dialling .............................................................................................................. 103
2.6.7 Hot Line ...................................................................................................................... 104
2.7 Toll Restriction (TRS)/Call Barring (Barring) Features ............................................ 106
2.7.1 Toll Restriction (TRS)/Call Barring (Barring) ............................................................... 106
2.7.2 Budget Management .................................................................................................. 113
2.7.3 Extension Dial Lock .................................................................................................... 113
2.7.4 Dial Tone Transfer ....................................................................................................... 114
2.7.5 Walking COS .............................................................................................................. 115
2.7.6 Verification Code Entry ............................................................................................... 116
2.8 Automatic Route Selection (ARS) Features .............................................................. 118
2.8.1 Automatic Route Selection (ARS) ............................................................................... 118
2.9 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Features . 124
2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension .... 124
2.10 Busy Line/Busy Party Features ................................................................................. 130
2.10.1 Automatic Callback Busy (Camp-on) .......................................................................... 130
2.10.2 Executive Busy Override ............................................................................................ 130
2.10.3 Call Monitor ................................................................................................................ 132
2.10.4 Second Call Notification to Busy Extension ................................................................ 133
2.10.4.1 Second Call Notification to Busy Extension—SUMMARY ....................................... 133
2.10.4.2 Call Waiting Tone ..................................................................................................... 134
2.10.4.3 Off-hook Call Announcement (OHCA) ..................................................................... 135
2.10.4.4 Whisper OHCA ........................................................................................................ 136
2.11 Conversation Features ............................................................................................... 138
2.11.1 Hands-free Operation ................................................................................................. 138
2.11.2 Off-hook Monitor ......................................................................................................... 138
2.11.3 Mute ............................................................................................................................ 138
2.11.4 Headset Operation ..................................................................................................... 139
2.11.5 Data Line Security ...................................................................................................... 139
2.11.6 Flash/Recall/Terminate ............................................................................................... 140
2.11.7 External Feature Access (EFA) .................................................................................. 141
2.11.8 Trunk Call Limitation ................................................................................................... 142
2.11.9 Calling Party Control (CPC) Signal Detection ............................................................ 145
2.12 Transferring Features ................................................................................................. 147
2.12.1 Call Transfer ............................................................................................................... 147
2.12.2 SIP Refer Transfer ...................................................................................................... 150
2.13 Holding Features ......................................................................................................... 152
2.13.1 Call Hold ..................................................................................................................... 152
2.13.2 Call Park ..................................................................................................................... 154
2.13.3 Call Splitting ................................................................................................................ 155

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 Table of Contents

2.13.4 Music on Hold ............................................................................................................. 155

2.14 Conference Features ................................................................................................... 158
2.14.1 Conference Features—SUMMARY ............................................................................ 158
2.14.2 Conference ................................................................................................................. 158
2.14.3 Privacy Release .......................................................................................................... 160
2.15 Conference Group Call Features ............................................................................... 161
2.15.1 Conference Group Call ............................................................................................... 161
2.16 Direct Inward System Access (DISA) Features ........................................................ 165
2.16.1 Direct Inward System Access (DISA) ......................................................................... 165
2.16.2 Automatic Fax Transfer .............................................................................................. 176
2.17 Paging Features .......................................................................................................... 178
2.17.1 Paging ........................................................................................................................ 178
2.17.2 Trunk Answer From Any Station (TAFAS) .................................................................. 180
2.18 External Device Features ............................................................................................ 181
2.18.1 Doorphone Call ........................................................................................................... 181
2.18.2 Door Open .................................................................................................................. 182
2.18.3 External Sensor .......................................................................................................... 182
2.18.4 External Relay Control ................................................................................................ 184
2.18.5 Communication IP Camera/Video Door Phone .......................................................... 185
2.19 Caller ID Features ........................................................................................................ 187
2.19.1 Caller ID ...................................................................................................................... 187
2.19.2 Incoming Call Log ....................................................................................................... 191
2.20 Message Features ....................................................................................................... 195
2.20.1 Message Waiting ........................................................................................................ 195
2.20.2 Absent Message ......................................................................................................... 196
2.21 Proprietary Telephone (PT) Hardware Features ....................................................... 198
2.21.1 Fixed Buttons .............................................................................................................. 198
2.21.2 Flexible Buttons .......................................................................................................... 200
2.21.3 LED Indication ............................................................................................................ 203
2.21.4 Display Information ..................................................................................................... 205
2.22 Administrative Information Features ......................................................................... 207
2.22.1 Record Log Features .................................................................................................. 207
2.22.1.1 Station Message Detail Recording (SMDR) ............................................................ 207
2.22.1.2 Syslog Record Management ................................................................................... 215
2.22.2 Printing Message ........................................................................................................ 215
2.22.3 Call Charge Services .................................................................................................. 216
2.23 Hospitality Features .................................................................................................... 219
2.23.1 Hospitality Features—SUMMARY .............................................................................. 219
2.23.2 Room Status Control .................................................................................................. 219
2.23.3 Call Billing for Guest Room ........................................................................................ 221
2.23.4 Built-in FOS Interface ................................................................................................. 224
2.24 Extension Controlling Features ................................................................................. 225
2.24.1 Extension Personal Identification Number (PIN) ........................................................ 225
2.24.2 Extension Feature Clear ............................................................................................. 226
2.24.3 Timed Reminder ......................................................................................................... 227
2.24.4 Smart Desk Features .................................................................................................. 228
2.24.4.1 Service-in Feature ................................................................................................... 230
2.24.4.2 Service-out Feature ................................................................................................. 231
2.24.4.3 Automatic Service-out Feature ................................................................................ 232
2.25 Audible Tone Features ................................................................................................ 235
2.25.1 Dial Tone ..................................................................................................................... 235
2.25.2 Confirmation Tone ...................................................................................................... 236
2.26 Computer Telephony Integration (CTI) Features ...................................................... 238
2.26.1 Computer Telephony Integration (CTI) ....................................................................... 238
2.26.2 CA (Communication Assistant) Operator Console ..................................................... 239

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Table of Contents

2.27 Cellular Phone Features ............................................................................................. 240

2.27.1 Cellular Phone Features—SUMMARY ....................................................................... 240
2.28 Miscellaneous Features .............................................................................................. 241
2.28.1 Background Music (BGM) .......................................................................................... 241
2.28.2 Outgoing Message (OGM) ......................................................................................... 242
2.29 User Container ............................................................................................................. 244
2.29.1 User Container ........................................................................................................... 245
2.29.1.1 User Container Configuration .................................................................................. 245
2.29.1.2 Individual User Information Management of the Construction and Maintenance
Personnel ................................................................................................................. 248
2.29.1.3 Editing User Settings ............................................................................................... 251
2.29.1.4 Security (Password) ................................................................................................. 252
2.29.1.5 Smart Desk Feature ................................................................................................. 252
2.29.1.6 Linking with LDAP User Information ........................................................................ 252
2.29.2 Call Control Feature ................................................................................................... 253
2.29.2.1 Main Device/Device Common ................................................................................. 253
2.29.2.2 Sub Device .............................................................................................................. 254
2.29.2.3 DSS Console ........................................................................................................... 256
2.29.2.4 Wireless Device ....................................................................................................... 256
2.29.2.5 Public (Mobile Phone) Device .................................................................................. 257
2.29.2.6 UM (Mailbox) ........................................................................................................... 262
2.29.2.7 Various Logs and Reports ....................................................................................... 263
2.29.2.8 CTI ........................................................................................................................... 264
3 Unified Messaging System ................................................................ 265
3.1 Unified Messaging System Administration .............................................................. 266
3.1.1 Unified Messaging System Overview ......................................................................... 266
3.1.2 System Administration ................................................................................................ 268
3.1.2.1 Custom Service Builder ........................................................................................... 268
3.1.2.2 Reserved ................................................................................................................. 269
3.1.2.3 Password Administration ......................................................................................... 269
3.1.2.4 System Backup/Restore .......................................................................................... 269
3.1.2.5 System Reports ....................................................................................................... 270
3.1.2.6 System Security ....................................................................................................... 270
3.1.3 Fax Server .................................................................................................................. 270
3.2 System and Subscriber Features .............................................................................. 273
3.2.1 System Features ........................................................................................................ 273
3.2.1.1 Alternate Extension Group ....................................................................................... 273
3.2.1.2 Auto Forwarding ...................................................................................................... 273
3.2.1.3 Automated Attendant (AA) ....................................................................................... 273
3.2.1.4 Automatic Two-way Recording for Manager ............................................................ 274
3.2.1.5 Broadcasting Messages .......................................................................................... 275
3.2.1.6 Call Services ............................................................................................................ 275
3.2.1.7 Call Transfer to Outside ........................................................................................... 276
3.2.1.8 Caller ID Call Routing .............................................................................................. 276
3.2.1.9 Caller ID Screening .................................................................................................. 277
3.2.1.10 Caller Name Announcement .................................................................................... 277
3.2.1.11 Class of Service (COS) ............................................................................................ 278
3.2.1.12 Company Greeting ................................................................................................... 278
3.2.1.13 Company Name ....................................................................................................... 278
3.2.1.14 Covering Extension .................................................................................................. 279
3.2.1.15 Custom Service ....................................................................................................... 279
3.2.1.16 Dialling by Name ...................................................................................................... 280
3.2.1.17 Emergency Greeting ................................................................................................ 280
3.2.1.18 Extension Group ...................................................................................................... 280

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 Table of Contents

3.2.1.19 Fax Service .............................................................................................................. 281

3.2.1.20 Hold ......................................................................................................................... 281
3.2.1.21 Holiday Service ........................................................................................................ 281
3.2.1.22 Hospitality Mode ...................................................................................................... 282
3.2.1.23 Intercept Routing to a Mailbox ................................................................................. 282
3.2.1.24 Intercom Paging ....................................................................................................... 282
3.2.1.25 Interview Service ..................................................................................................... 283
3.2.1.26 List All Names .......................................................................................................... 284
3.2.1.27 Logical Extension (All Calls Transfer to Mailbox) .................................................... 284
3.2.1.28 Message Reception Mode ....................................................................................... 284
3.2.1.29 Message Waiting Notification—E-mail Device ......................................................... 285
3.2.1.30 Message Waiting Notification—Lamp ...................................................................... 286
3.2.1.31 Message Waiting Notification—Telephone Device .................................................. 287
3.2.1.32 Multilingual Service .................................................................................................. 287
3.2.1.33 No DTMF Input Operation ....................................................................................... 288
3.2.1.34 On Hold Announcement Menu ................................................................................ 288
3.2.1.35 Operator Service ...................................................................................................... 288
3.2.1.36 PIN Call Routing ...................................................................................................... 289
3.2.1.37 Play System Prompt After Personal Greeting .......................................................... 289
3.2.1.38 Port Service ............................................................................................................. 290
3.2.1.39 Remote Time Service Set ........................................................................................ 290
3.2.1.40 Service Group .......................................................................................................... 290
3.2.1.41 Simplified Tutorial .................................................................................................... 291
3.2.1.42 System Prompts ...................................................................................................... 291
3.2.1.43 Transfer Recall to a Mailbox .................................................................................... 291
3.2.1.44 Transfer to Mailbox .................................................................................................. 292
3.2.1.45 Trunk Service (Universal Port) ................................................................................. 292
3.2.1.46 Voice Mail Service ................................................................................................... 292
3.2.2 Subscriber Features ................................................................................................... 293
3.2.2.1 Auto Receipt ............................................................................................................ 293
3.2.2.2 Automatic Fax Delivery ............................................................................................ 294
3.2.2.3 Automatic Login ....................................................................................................... 294
3.2.2.4 Autoplay New Message ........................................................................................... 295
3.2.2.5 Bookmark ................................................................................................................. 295
3.2.2.6 Call-through Service ................................................................................................ 296
3.2.2.7 Call Transfer Scenario ............................................................................................. 296
3.2.2.8 Call Transfer Status ................................................................................................. 296
3.2.2.9 Callback Number Entry ............................................................................................ 297
3.2.2.10 Caller ID Callback .................................................................................................... 297
3.2.2.11 Delete Message Confirmation ................................................................................. 297
3.2.2.12 Direct Service Access .............................................................................................. 297
3.2.2.13 External Message Delivery Service ......................................................................... 298
3.2.2.14 Fax Cover Page ....................................................................................................... 298
3.2.2.15 Fax Driver ................................................................................................................ 299
3.2.2.16 Forwarding to a Mailbox .......................................................................................... 300
3.2.2.17 Group Distribution Lists ........................................................................................... 300
3.2.2.18 Incomplete Call Handling Service ............................................................................ 300
3.2.2.19 Live Call Screening (LCS) ....................................................................................... 301
3.2.2.20 Mailbox .................................................................................................................... 302
3.2.2.21 Mailbox Capacity Warning ....................................................................................... 302
3.2.2.22 Manager Service Switching ..................................................................................... 302
3.2.2.23 Message Transfer .................................................................................................... 303
3.2.2.24 Personal Custom Service ........................................................................................ 303
3.2.2.25 Personal Greetings .................................................................................................. 304
3.2.2.26 Private Message ...................................................................................................... 305

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Table of Contents

3.2.2.27 Recover Message .................................................................................................... 305

3.2.2.28 Remote Absent Message ........................................................................................ 305
3.2.2.29 Remote Call Forwarding Set .................................................................................... 305
3.2.2.30 Send to Fax Machine ............................................................................................... 306
3.2.2.31 Subscriber Tutorial ................................................................................................... 307
3.2.2.32 Timed Reminder Setting .......................................................................................... 307
3.2.2.33 Toll Saver ................................................................................................................. 308
3.2.2.34 Two-way Record/Two-way Transfer ........................................................................ 308
3.2.2.35 Urgent Message ...................................................................................................... 310
3.2.2.36 Voice Mail (VM) Transfer Button .............................................................................. 310
3.2.2.37 Web Programming ................................................................................................... 310
3.3 E-mail Client Integration Features ............................................................................. 312
3.3.1 IMAP Integration ......................................................................................................... 312
4 Network Features ............................................................................... 313
4.1 Public Network Features ............................................................................................ 314
4.1.1 SIP (Session Initiation Protocol) Trunk ....................................................................... 314
4.1.1.1 SIP Server Failover .................................................................................................. 315
4.1.2 Integrated Services Digital Network (ISDN) Service Features ................................... 316
4.1.2.1 Integrated Services Digital Network (ISDN)—SUMMARY ....................................... 316
4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/COLP) ........................... 320
4.1.2.3 Advice of Charge (AOC) .......................................................................................... 322
4.1.2.4 Call Forwarding (CF)—by ISDN (P-MP) .................................................................. 323
4.1.2.5 Call Forwarding (CF)—by ISDN (P-P) ..................................................................... 324
4.1.2.6 Call Hold (HOLD)—by ISDN .................................................................................... 325
4.1.2.7 Call Transfer (CT)—by ISDN ................................................................................... 326
4.1.2.8 Three-party Conference (3PTY)—by ISDN ............................................................. 326
4.1.2.9 Malicious Call Identification (MCID) ......................................................................... 327
4.1.2.10 Completion of Calls to Busy Subscriber (CCBS) ..................................................... 327
4.1.2.11 ISDN Service Access by Keypad Protocol .............................................................. 328
4.2 Multi-connection Networking ..................................................................................... 329
4.2.1 Multi-connection Networking Overview ...................................................................... 329
4.2.2 Network Type Comparison ......................................................................................... 337
4.2.3 Multi-connection Networking Survivability .................................................................. 338
4.2.3.1 Simplified Isolated Mode .......................................................................................... 338
4.3 Private Network Features ........................................................................................... 343
4.3.1 TIE Line Service ......................................................................................................... 343
4.3.1.1 Making a TIE Line Call ............................................................................................. 343
4.3.1.2 TIE Line and Trunk Connection ............................................................................... 345
4.3.1.3 TIE Line Programming ............................................................................................. 356
4.3.1.4 Common Extension Numbering for 2 PBXs ............................................................. 364
4.3.2 Voice over Internet Protocol (VoIP) Network .............................................................. 364
4.3.2.1 Gateway Groups ...................................................................................................... 368
4.3.2.2 Common Extension Numbering for Multiple PBXs .................................................. 368
4.3.2.3 Call Distribution Port Group ..................................................................................... 369
4.3.3 ISDN Virtual Private Network (ISDN-VPN) ................................................................. 371
4.3.4 QSIG Standard Features ............................................................................................ 372
4.3.4.1 QSIG Standard Features—SUMMARY ................................................................... 372
4.3.4.2 Calling/Connected Line Identification Presentation (CLIP/COLP) and Calling/
Connected Name Identification Presentation (CNIP/CONP)—by QSIG .................. 374
4.3.4.3 Call Forwarding (CF)—by QSIG .............................................................................. 375
4.3.4.4 Call Transfer (CT)—by QSIG ................................................................................... 376
4.3.4.5 Completion of Calls to Busy Subscriber (CCBS)—by QSIG .................................... 378
5 System Configuration and Administration Features ...................... 379

14 Feature Manual
 Table of Contents

5.1 System Configuration—System ................................................................................. 380

5.1.1 Class of Service (COS) .............................................................................................. 380
5.1.2 Group .......................................................................................................................... 381
5.1.3 Tenant Service ............................................................................................................ 385
5.1.4 Time Service ............................................................................................................... 388
5.1.5 Operator Features ...................................................................................................... 392
5.1.6 Manager Features ...................................................................................................... 393
5.2 System Configuration—Extensions .......................................................................... 395
5.2.1 IP Proprietary Telephone (IP-PT) ............................................................................... 395
5.2.2 SIP (Session Initiation Protocol) Extension ................................................................ 396
5.2.2.1 Simple Remote Connection ..................................................................................... 398
5.2.2.2 Panasonic SIP Phones (P-SIP) ............................................................................... 400
5.2.3 Peer-to-Peer (P2P) Connection .................................................................................. 401
5.2.4 Portable Station (PS) Features ................................................................................... 403
5.2.4.1 Portable Station (PS) Connection ............................................................................ 403
5.2.4.2 PS Ring Group ......................................................................................................... 404
5.2.4.3 PS Directory ............................................................................................................. 407
5.2.4.4 PS Feature Buttons ................................................................................................. 408
5.2.5 ISDN Extension Features ........................................................................................... 408
5.2.5.1 ISDN Extension ....................................................................................................... 408
5.2.6 IP-CS (KX-NS0154) .................................................................................................... 410
5.2.7 Internal Call Encryption ............................................................................................... 411
5.3 Legacy Device Connection ......................................................................................... 413
5.3.1 Stacking Connection ................................................................................................... 413
5.3.2 Trunk Adaptor Connection .......................................................................................... 413
5.4 E-mail Notification Features ....................................................................................... 415
5.4.1 E-mail Notification for Extension Users ...................................................................... 415
5.4.2 E-mail Notification of System-level Events ................................................................. 415
5.4.3 E-mail Notification of Sensor Alarm ............................................................................ 416
5.4.4 E-mail Notification for Manager .................................................................................. 417
5.5 System Data Control ................................................................................................... 418
5.5.1 PC Programming ........................................................................................................ 418
5.5.2 PT Programming ........................................................................................................ 419
5.5.3 DSP Resource Usage ................................................................................................ 420
5.5.3.1 DSP Resource Reservation ..................................................................................... 423
5.5.3.2 DSP Resource Advisor ............................................................................................ 424
5.5.3.3 Resource Depletion Indication Monitoring ............................................................... 426
5.5.4 Automatic Setup ......................................................................................................... 428
5.5.5 Dynamic Host Configuration Protocol (DHCP) Server ............................................... 429
5.5.6 Flexible Numbering/Fixed Numbering ........................................................................ 429
5.5.7 Floating Extension ...................................................................................................... 435
5.5.8 Software Upgrading .................................................................................................... 436
5.6 Fault Recovery/Diagnostics ....................................................................................... 439
5.6.1 Reserved .................................................................................................................... 439
5.6.2 Power Failure Restart ................................................................................................. 439
5.6.3 Local Alarm Information .............................................................................................. 439
5.6.4 Simple Network Management Protocol (SNMP) System Monitor .............................. 440
5.6.5 Dynamic Host Configuration Protocol (DHCP) Assignment ....................................... 442
5.6.6 PING Confirmation ..................................................................................................... 442
5.6.7 System Monitoring ...................................................................................................... 442
5.6.8 Congestion Control ..................................................................................................... 443
5.6.9 Hot Standby ................................................................................................................ 444
6 Appendix ............................................................................................. 449
6.1 Capacity of System Resources .................................................................................. 450

Feature Manual 15
Table of Contents

6.2 Tones/Ring Tones ........................................................................................................ 455

6.2.1 Tones/Ring Tones ....................................................................................................... 455
6.3 Features that Require Activation Keys ..................................................................... 457
6.4 Supported Management Information Base (MIB) Table ........................................... 459
6.5 Revision History .......................................................................................................... 467
6.5.1 PNMPR Software File Version 002.00xxx .................................................................. 467
6.5.2 PNMPR Software File Version 002.01xxx .................................................................. 468
6.5.3 PNMPR Software File Version 002.03xxx .................................................................. 468
6.5.4 PNMPR Software File Version 002.10xxx .................................................................. 469
6.5.5 PNMPR Software File Version 003.00xxx .................................................................. 469
6.5.6 PNMPR Software File Version 004.00xxx .................................................................. 471

16 Feature Manual
 Section 1
For Your Safety

Feature Manual 17
1.1 For Your Safety

1.1 For Your Safety

1.1.1 For Your Safety
Description
To prevent personal injury and/or damage to property, be sure to observe the following safety precautions.

The following symbols classify and describe the level of hazard and injury caused when this unit is
operated or handled improperly.

This notice means that misuse could result in

CAUTION injury or damage to property.

The following types of symbols are used to classify and describe the type of instructions to be
observed.

This symbol is used to alert users to a specific operating procedure that must be followed in
order to operate the unit safely.

CAUTION

• The software contained in the TRS/Barring and ARS features to allow user access to the network must be
upgraded to recognise newly established network area codes and exchange codes as they are placed
into service. Failure to upgrade the on-premise PBXs or peripheral equipment to recognise the new codes
as they are established will restrict the customer and users of the PBX from gaining access to the network
and to these codes.
KEEP THE SOFTWARE UP TO DATE WITH THE LATEST DATA.
• There is a risk that fraudulent telephone calls will be made in the following cases:
– A third party discovers a personal identification number (PIN) (verification code PIN or extension PIN)
of the PBX.
– Using the Trunk-to-Trunk Call feature of DISA.
The cost of such calls will be billed to the owner/renter of the PBX. To protect the PBX from this kind of
fraudulent use, we strongly recommend:
a. Keeping PINs secret.
b. Selecting complex, random PINs that cannot be easily guessed.
c. Changing PINs regularly.
• To the Administrator or Installer regarding account passwords
1. Please provide all system passwords to the customer.
2. To avoid unauthorised access and possible abuse of the NSX server, keep the passwords secret, and
inform the customer of the importance of the passwords, and the possible dangers if they become
known to others.

18 Feature Manual
 1.1.1 For Your Safety

3. The NSX server has no passwords set initially. For security, select an installer password as soon as
the NSX server system is installed at the site.
4. Change the passwords periodically.
5. It is strongly recommended that passwords of 10 numbers or characters be used for maximum
protection against unauthorised access.

Feature Manual 19
1.1.1 For Your Safety

20 Feature Manual
 Section 2
Call Handling Features

Feature Manual 21
2.1 Incoming Call Features

2.1 Incoming Call Features

2.1.1 Incoming Trunk Call Features
2.1.1.1 Incoming Trunk Call Features—SUMMARY
Description
Incoming calls via a trunk (public line) are distributed to their destinations according to one of several
distribution methods.

1. Available Networking Type for Each Card Type

Each trunk port of an optional trunk card or the mother board can be assigned a networking type:
Public, Private, or VPN (Virtual Private Network).

Networking Type
Channel/ Public
Card Type Virtual Private
Protocol Type (DIL/DID/DDI/ Private (TIE)
*1
*2

MSN) Network (VPN)

Mother Board H.323

ü ü*
(V-IPGW)
Mother Board SIP
ü*
(V-SIPGW)
ELCOT/CLCOT/ — ü*
LCOT
BRI/PRI CO ü* ü
Extension
QSIG-Master ü*
QSIG-Slave ü*
DID — ü*
T1 LCOT ü*
GCOT ü*
DID ü*
TIE (E & M) ü ü*
OPX (EXTN.)
E1 DR2 ü* ü
E & M-C ü ü*
E & M-P ü ü*
E&M — ü ü*

ü*: Enable (default); ü: Enable

*1 → 4.3.1 TIE Line Service
*2 → 4.3.3 ISDN Virtual Private Network (ISDN-VPN)

22 Feature Manual
 2.1.1 Incoming Trunk Call Features

2. Distribution Method
One of the following methods can be assigned to each trunk port:
Method Description & Reference
Direct In Line (DIL) Directs a call to a preprogrammed single destination (e.g., the
operator).

→ 2.1.1.2 Direct In Line (DIL)

Direct Inward Dialling (DID) Directs a call with a DID number from a DID line to a
preprogrammed destination.
DID is also known as Direct Dialling In (DDI).

→ 2.1.1.3 Direct Inward Dialling (DID)/Direct Dialling In (DDI)

Multiple Subscriber Directs a call with an MSN from an ISDN line to a preprogrammed
Number (MSN) Ringing destination.
Service
→ 2.1.1.4 Multiple Subscriber Number (MSN) Ringing Service

3. Destination Change with the Caller’s Identification Number

The Calling Line Identification (CLI) Distribution feature works in conjunction with the DIL/DID/DDI/MSN
features.
Feature Description & Reference
Calling Line Identification Directs a call to a CLI destination if the caller’s identification
(CLI) Distribution number has been assigned in the Caller ID Table.

→ 2.1.1.5 Calling Line Identification (CLI) Distribution

4. Available Distribution Feature for Each Optional Trunk Card Type

If Network Type of each optional trunk card is Public, the following Distribution Features are available:

Trunk Card Feature

Channel Type
Type DIL DID/DDI MSN
Mother Board — ü ü*
(V-IPGW)
Mother Board — ü ü*
(V-SIPGW)
ELCOT/CLCOT/ — ü*
LCOT
BRI CO ü ü* ü
PRI CO ü ü*
DID — ü ü*
T1 LCOT ü*
GCOT ü*
DID ü ü*
TIE (E & M) ü* ü

Feature Manual 23
2.1.1 Incoming Trunk Call Features

Trunk Card Feature

Channel Type
Type DIL DID/DDI MSN
E1 DR2 ü ü*
E & M-C ü* ü
E & M-P ü* ü
E&M — ü*

ü*: Enable (default); ü: Enable

5. Available Destinations
Destination Availability
Wired Extension (PT/SLT/SIP Extension/ISDN Extension/T1-OPX) ü
PS ü
Incoming Call Distribution Group ü
PS Ring Group ü
UM Group ü
Fax Unit ü
External Pager (TAFAS) ü
DISA ü
Idle Line Access no. + Phone no.
Trunk Group Access no. + Trunk Group no. + Phone no.
Other PBX Extension (TIE with no PBX Code) ü
Other PBX Extension (TIE with PBX Code)

6. Intercept Routing
After setting distribution, it may also be necessary to set the following features.
Feature Description & Reference
Intercept Routing No Answer (IRNA) If a called party does not answer a call within a
preprogrammed time period (Intercept time), it is
redirected to the preprogrammed destination.

→ 2.1.1.6 Intercept Routing

Busy/DND If a called party is busy or in DND mode, the call is
redirected to the preprogrammed destination.

→ 2.1.1.6 Intercept Routing

No Destination If a destination is not assigned, the call is redirected to
the operator.

→ 2.1.1.7 Intercept Routing—No Destination

PC Programming Manual References

18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings—DIL—Trunk Property

24 Feature Manual
 2.1.1 Incoming Trunk Call Features

2.1.1.2 Direct In Line (DIL)

Description
Provides automatic direction of an incoming trunk call to a preprogrammed destination. Each trunk has a
destination for each time mode (day/lunch/break/night).

[Method Flowchart]

A trunk call is received.

Does the call have its CLI* No

information and is CLI mode enabled
for the trunk and the time mode?

Yes

CLI works.

Yes Is the CLI destination

assigned?

No

Is the DIL destination of No

the time mode assigned?

Yes

The call is routed to the The call is routed to the The call is routed to the
CLI destination. DIL destination. operator (Intercept Routing
—No Destination).

*: Calling Line Identification (CLI) Distribution:

If the CLI routing is enabled and the caller's identification number is assigned in the Caller ID
Table, the call will not be routed to the DIL destination, but routed to the CLI destination.

[Programming Example of DIL Table]

The table can be programmed for each trunk.
CLI Destination*1
Trunk No.
Day Lunch ... Day Lunch ...
01 Enable Disable ... 101 100 ...
02 Enable Disable ... 102 100 ...
: : : : : : :

*1 → 18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings—DIL—DIL Destination—Day, Lunch, Break,
Night

Feature Manual 25
2.1.1 Incoming Trunk Call Features

Note
The following settings can also be specified in the DIL table:
• Tenant number: determines the time mode (day/lunch/break/night) for the corresponding trunk.
• UM service group number: determines the service group to use when a call is handled by the Unified
Messaging system.

Explanation:
If a trunk call is received from trunk 01;
In Day mode: CLI is enabled. Route to CLI destination.
In Lunch mode: CLI is disabled. Route to DIL destination, extension 100.

PC Programming Manual References

18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings—DIL
→ DIL Destination—Day, Lunch, Break, Night
→ Tenant Number
→ UM Service Group No.

Feature Manual References

2.1.1.5 Calling Line Identification (CLI) Distribution
3.2.1.40 Service Group
5.1.3 Tenant Service
5.1.4 Time Service
6.1 Capacity of System Resources

2.1.1.3 Direct Inward Dialling (DID)/Direct Dialling In (DDI)

Description
Provides automatic direction of an incoming call with a DID/DDI number to a preprogrammed destination.
Each DID/DDI number has a destination for each time mode (day/lunch/break/night).
Incoming calls with DID/DDI numbers that match extension numbers at this PBX will be sent to the
corresponding extension. Incoming calls with DID/DDI numbers that match extensions at other PBXs or
trunk access numbers will be sent to the corresponding TIE line or trunk.

26 Feature Manual
 2.1.1 Incoming Trunk Call Features

[Method Flowchart]

A trunk call is received.

Is the DID/DDI number found in No

the DID/DDI table?

Yes Does the DID/DDI Yes

number match an
extension number?
Does the call have its CLI*
information and is CLI mode
enabled for the time mode? No No
The call is routed
to the extension.
Yes

CLI works. Does the DID/DDI

number match an extension Yes
number at another PBX or
Yes Trunk Access no.?
Is the CLI destination assigned?
No
No

Is the DID/DDI destination No The call is routed

for the time mode assigned? to the TIE line or
trunk.
Yes
The call is routed to the
The call is routed to the The call is routed to the
operator (Intercept
CLI destination. DID/DDI destination.
Routing—No Destination).

*: Calling Line Identification (CLI) Distribution:

If the CLI routing is enabled and the caller's identification number is assigned in the Caller ID
Table, the call will not be routed to the DID/DDI destination, but routed to the CLI destination.

[Programming Example of DID/DDI Table]

DDI can be programmed as DID.
CLI*3 Destination*4
Location No.*1 Name*2
Day Lunch ... Day Lunch ...
0001 123-4567 John Enable Disable ... 105 100 ...
White
0002 123-2468 Tom Enable Disable ... 102 100 ...
Smith
0003 123-456 A Enable Disable ... 101 101 ...
company
: : : : : : : : :

*1 → 18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—DDI / DID Number

Feature Manual 27
2.1.1 Incoming Trunk Call Features

*2 → 18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—DDI/DID Name

*3 → 18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—CLI Ring for DDI/DID—Day, Lunch, Break, Night
*4 → 18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—DDI / DID Destination—Day, Lunch, Break, Night

Note
The following settings can also be specified in the DID/DDI table:
• Tenant number: determines the time mode (day/lunch/break/night) for the corresponding trunk.
• UM service group number: determines the service group to use when a call is handled by the Unified
Messaging system.

Explanation:
If the DID/DDI number is "123-4567":
1. Checks the number in the table.
→ Matches the number in location 0001.
2. Checks the time mode.
In Day mode: CLI is enabled. Route to CLI destination.
In Lunch mode: CLI is disabled. Route to DID/DDI destination, extension 100.

Conditions
• To use this feature, DID/DDI service must be assigned as the distribution method for a trunk port.
• DID/DDI Number Modification
It is possible to modify a received DID/DDI number, which may be convenient when programming the
DID/DDI table. The modification method (removed number of digits/added number) can be programmed
on a trunk port basis.

[Modification Example]

Removed number of digits: 6 Modified DID/DDI number: 876543 21 = 1021

Added number: 10
Received DID/DDI number: 87654321
1) Remove the 2) Add "10".
first 6 digits.

• Inter-digit Time
When the Inter-digit time expires, the PBX stops receiving the DID/DDI number and starts to check the
DID/DDI table. (Refer to the [Programming Example of DID/DDI Table] above).
Even if the Inter-digit time does not expire, the PBX stops receiving the DID/DDI number when the
received number is found in the DID/DDI table. The PBX then routes the call to the corresponding
destination. If the received number matches several entries in the table, the call is directed to the
destination of the first matching entry.

[Example]
If a call is received in Lunch mode;
Received Number Destination Explanation
123-4567 Extn. 100 The PBX finds the match in location 0001 in the table
after receiving "7". So the call is routed to extension
100.
123-456 Extn. 101 The Inter-digit time expired after receiving "6". The
PBX finds the match in location 0003 in the table. So
the call is routed to extension 101.

28 Feature Manual
 2.1.1 Incoming Trunk Call Features

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—Incoming Call Inter-digit
Timer—DDI / DID
18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings—DDI / DID / TIE / MSN
→ Distribution Method
→ DDI/DID/TIE/MSN—Remove Digit
→ DDI/DID/TIE/MSN—Additional Dial
18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table

Feature Manual References

2.1.1.5 Calling Line Identification (CLI) Distribution
3.2.1.40 Service Group
5.1.3 Tenant Service
5.1.4 Time Service
6.1 Capacity of System Resources

2.1.1.4 Multiple Subscriber Number (MSN) Ringing Service

Description
Provides automatic direction of an incoming ISDN-BRI (Basic Rate Interface) line call with an MSN to a
preprogrammed destination. One ISDN-BRI port can support a maximum of 10 MSNs. Each MSN has a
destination for each time mode (day/lunch/break/night).
Point-to-multipoint must be selected for the ISDN configuration.

Feature Manual 29
2.1.1 Incoming Trunk Call Features

[Method Flowchart]

A trunk call is received.

Are any MSNs assigned No

in the MSN table?

The call is routed to the

Yes operator (Intercept
Routing—No Destination).

Is the MSN found in the No

MSN table?

Yes The call is ignored.

Does the call have its CLI* No

information and is CLI mode
enabled for the time mode?

Yes
CLI works.

Yes Is the CLI destination

assigned?

No

Is the MSN destination for No

the time mode assigned?

Yes

The call is routed to the The call is routed to the The call is routed to the
CLI destination. MSN destination. operator (Intercept
Routing—No Destination).
*: Calling Line Identification (CLI) Distribution:
If the CLI routing is enabled and the caller's identification number is assigned in the Caller ID
Table, the call will not be routed to the MSN destination, but routed to the CLI destination.

[Programming Example of MSN Table for ISDN BRI Port 1]

A table can be programmed for each ISDN-BRI port. Each BRI port has 10 MSN locations.
→ 18.4 PBX Configuration—[10-4] CO & Incoming Call—MSN Table
CLI Destination
Location No. Name
Day Lunch ... Day Lunch ...
123- A
01 Enable Disable ... 101 100 ...
4567 Company
123- C
02 Enable Disable ... 102 100 ...
2468 Company
: : : : : : : : :

30 Feature Manual
 2.1.1 Incoming Trunk Call Features

CLI Destination
Location No. Name
Day Lunch ... Day Lunch ...
10 : : : : : : : :

Note
The following settings can also be specified in the MSN table:
• Tenant number: determines the time mode (day/lunch/break/night) for the corresponding trunk.
• UM service group number: determines the service group to use when a call is handled by the Unified
Messaging system.

Explanation:
If the MSN "123-4567" is received from BRI port 1:
1. Checks the number in the table.
→ Matches the number in location 01.
2. Checks the time mode.
In Day mode: CLI is enabled. Route to CLI destination.
In Lunch mode: CLI is disabled. Route to MSN destination, extension 100.

Conditions
• To use this feature, the MSN service must be assigned as the distribution method for a trunk port.
• MSN Modification
It is possible to modify a received MSN to make it shorter, which may be convenient when programming
the MSN table. The modification method (removed number of digits/added number) can be programmed
on a trunk port basis.

[Modification Example]

Removed number of digits: 6 Modified MSN: 876543 21 = 1021

Added number: 10
Received MSN: 87654321
1) Remove the first 2) Add "10".
6 digits.

• When using point-to-multipoint configuration with a BRI, do not connect another ISDN terminal device in
parallel with the PBX. As only two channels can be used at one time with the BRI, the other ISDN
terminal device may monopolise both channels.

PC Programming Manual References

18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings—DDI / DID / TIE / MSN
→ Distribution Method
→ DDI/DID/TIE/MSN—Remove Digit
→ DDI/DID/TIE/MSN—Additional Dial
18.4 PBX Configuration—[10-4] CO & Incoming Call—MSN Table

Feature Manual References

2.1.1.5 Calling Line Identification (CLI) Distribution
3.2.1.40 Service Group
5.1.3 Tenant Service
5.1.4 Time Service
6.1 Capacity of System Resources

Feature Manual 31
2.1.1 Incoming Trunk Call Features

2.1.1.5 Calling Line Identification (CLI) Distribution

Description
Directs an incoming trunk call to a preprogrammed destination when the caller’s identification number (e.g.,
Caller ID) matches the number in the System Speed Dialling Table that is used as the Caller ID Table. Each
Caller ID number (telephone number for each System Speed Dialling number) can have its own destination.
CLI Feature Description & Reference
Caller ID Caller’s number is sent from an analogue trunk.

→ 2.19.1 Caller ID
Calling Line Identification Caller’s number is sent from an ISDN line.
Presentation (CLIP)
→ 4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/
COLP)
Automatic Number Caller’s number is sent from an E1 or T1 line (stacking connection
Identification (ANI) only).

→ 5.3.1 Stacking Connection

CLI always works in conjunction with the following call distribution methods:
a. DIL
b. DID/DDI
c. MSN Ringing Service
Each trunk (for DIL) and the DID/DDI/MSN number can enable or disable the CLI feature for each time
mode (day/lunch/break/night) (→ 5.1.4 Time Service).
When the call has Caller ID information and the CLI is enabled for the time mode, the call will be handled by
the CLI method.
[Programming Example of System Speed Dialling Table for CLI]
Location
System Speed
(System Speed Telephone No.*2 CLI Destination*3
Dialling Name*1
Dialling No.)
000 ABC Company 901234567890 200
001 : : :
: : : :

*1 → 14.1 PBX Configuration—[6-1] Feature—System Speed Dial—Name

*2 → 14.1 PBX Configuration—[6-1] Feature—System Speed Dial—CO Line Access Number + Telephone Number
*3 → 14.1 PBX Configuration—[6-1] Feature—System Speed Dial—CLI Destination
Explanation:
If the caller’s number is "0123-456-7890" (The Trunk Access number is disregarded):
1. Checks the number in the table.
→ Matches the number in location 000.
2. The call is routed to the CLI destination, extension 200.

Conditions
• Automatic Caller ID Number Modification

32 Feature Manual
 2.1.1 Incoming Trunk Call Features

The Caller ID number is used after modification by the Automatic Caller ID Number Modification.
(→ 2.19.1 Caller ID)

PC Programming Manual References

18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings—CLI for DIL—CLI Ring for
DIL—Day, Lunch, Break, Night
18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—CLI Ring for DDI/DID—Day, Lunch,
Break, Night
18.4 PBX Configuration—[10-4] CO & Incoming Call—MSN Table—CLI Ring for MSN—Day, Lunch, Break,
Night
14.1 PBX Configuration—[6-1] Feature—System Speed Dial
→ Name
→ CO Line Access Number + Telephone Number
→ CLI Destination

Feature Manual References

2.1.1.2 Direct In Line (DIL)
2.1.1.3 Direct Inward Dialling (DID)/Direct Dialling In (DDI)
2.1.1.4 Multiple Subscriber Number (MSN) Ringing Service
2.6.4 Speed Dialling—Personal/System

2.1.1.6 Intercept Routing

Description
Provides automatic redirection of incoming trunk and intercom calls. There are three types of Intercept
Routing as follows:
Feature Description
Intercept Routing—No Answer If a called party does not answer a call within a preprogrammed time
(IRNA) period (IRNA Timer), the call is redirected to the preprogrammed
destination.
Intercept Routing—Busy If a called party is already handling a call, new calls are handled as
follows:
– The call is redirected to the preprogrammed Intercept Routing—
Busy destination.
– If an Intercept Routing—Busy destination is not enabled, the caller
will hear a busy tone. However, if the call is made through an
ELCOT/CLCOT/LCOT or T1 (LCOT/GCOT) card, the caller hears
a ringback tone.
Intercept Routing—DND If a called party is in DND mode, the call is redirected to the
preprogrammed destination.

Intercept destinations can be assigned to extension ports.

→ 8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Intercept Destination—
When called party does not answer—Day, Lunch, Break, Night

Feature Manual 33
2.1.1 Incoming Trunk Call Features

When the original destination is: The Available Intercept Destination is:
• Wired Extension (PT/SLT/SIP Extension/T1- The destination assigned to the original extension.
OPX)
8.2.1 Users—Advanced Extension Settings—
• PS
Advanced Extension Settings
• Fax Unit
→ Intercept Destination—When called party does not
answer—Day, Lunch, Break, Night
→ Intercept Destination—When Called Party is Busy
• ICD Group The ICD Group Overflow destination assigned to the
group. (→ 2.2.2.6 Overflow Feature)

→ 11.5.1 PBX Configuration—[3-5-1] Group—

Incoming Call Distribution Group—Group Settings—
Overflow No Answer—Time out & Manual Queue
Redirection—Destination-Day, Lunch, Night
• UM Group The destination assigned to the first extension of the
UM group.
• DISA*1 If all DISA ports are busy when a call is made using
DISA, one of the following can be selected through
system programming:
• Disable: Busy tone is sent to the caller. When
using an analogue trunk, a ringback tone is sent.
• Operator: The call will be redirected to the
operator.
• AA-0, AA-9: The call will be redirected to the
destination assigned to that AA number.
→ 13.3.1 PBX Configuration—[5-3-1] Optional
Device—Voice Message—DISA System—Option 1
—DISA Intercept—Intercept when all DISA ports are
busy
• PS Ring Group Not assignable (Intercept Routing is not available.)
• External Pager (TAFAS)
• Other PBX Extension (TIE with no PBX Code)
*1 Intercept Routing for DISA will redirect a call only if all of the Direct Inward System Access (DISA) ports are busy. Once the call
reaches the destination extension by using the DISA feature, the Intercept Routing feature of the extension is used.
Different intercept destinations can be programmed for each time mode (day/lunch/break/night).

[Available Intercept Destinations]

Intercept Destination Availability
Wired Extension (PT/SLT/SIP Extension/ISDN Extension/T1-OPX) ü
PS ü
Incoming Call Distribution Group ü
PS Ring Group ü
UM Group ü

34 Feature Manual
 2.1.1 Incoming Trunk Call Features

Intercept Destination Availability

Fax Unit
External Pager (TAFAS) ü
DISA ü
Idle Line Access no. + Phone no. ü
Trunk Group Access no. + Trunk Group no. + Phone no. ü
Other PBX Extension (TIE with no PBX Code) ü
Other PBX Extension (TIE with PBX Code) ü

Intercept for calls to an outside destination

When an intercom, trunk, or DISA call is received by an extension and forwarded to an outside destination
by FWD—All Calls or FWD—Busy, the Intercept Routing feature can be used, if the outside destination is
busy or does not answer. For example, when a call is forwarded to a cellular phone by FWD—All Calls, and
the cellular phone is busy, the call will be routed to Voice Mail in the PBX.
This feature is also available when a call is forwarded by FWD—No Answer under the following conditions:
– The forwarding destination is in a private network and is busy or does not answer.
– The forwarding destination is in a public network and does not answer.
However, this feature is not available when a call is forwarded by FWD—No Answer to a destination in a
public network that is busy.

Conditions
• Intercept Routing—DND on/off
Intercept Routing—DND can be enabled or disabled system programming.
If disabled, one of the following is activated depending on the type of line that a call arrives through:
a. ELCOT/CLCOT/LCOT or T1 (LCOT/GCOT) Card: The incoming trunk call will ring at the original
destination while the caller hears a ringback tone.
b. Other Trunk Cards: A busy tone will be sent to the caller.
• If the intercept destination cannot receive the call:
a. Intercept Routing—No Answer: Intercept timer will restart at the original destination, until the call
is answered.
b. Intercept Routing—Busy/DND: The call will be sent back to the original destination when the call
arrives through the ELCOT/CLCOT/LCOT or T1 (LCOT/GCOT) card. When the call arrives through
other trunk cards the caller will hear a busy tone.
• Idle Extension Hunting
If an extension is a member of an idle extension hunting group, calls to that extension will not be
redirected by Intercept Routing—Busy/DND. If the extension is busy or in DND mode, calls to that
extension will be redirected to the next extension in the idle extension hunting group.
• Intercept Routing for intercom calls can be enabled or disabled on a system basis.
→ 18.5 PBX Configuration—[10-5] CO & Incoming Call—Miscellaneous—Intercept—Intercept Routing for
Extension Call
• IRNA Timer
The IRNA timer can be set on a system basis and an extension basis for each time mode (day, lunch,
break, night).
→ 8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Intercept No Answer
Time—Intercept No Answer Time—Day, Lunch, Break, Night

Feature Manual 35
2.1.1 Incoming Trunk Call Features

• The Intercept Routing destination for each time mode will not apply for Intercept Routing—Busy. When
the original destination is busy, the call is redirected to the Intercept Routing—Busy destination assigned
through system programming. If no destination is assigned the caller will hear a busy tone.
• The time modes that are selected for trunk calls arriving at extensions and UM groups are decided on a
tenant basis.
• Intercept for calls to an outside destination
– ELCOT/CLCOT/LCOT trunks without reverse circuit detection (→ 2.5.4.5 Reverse Circuit) and
channels of a T1 trunk set to LCOT type do not support this feature.
– This feature may not be available depending on the specifications of the telephone network.
– This feature is not available when the original call was made from a SIP extension.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters
→ Dial / IRNA / Recall / Tone—Intercept Routing No Answer (IRNA)—Day, Lunch, Break, Night
→ DISA / Door / Reminder / U. Conf—DISA—Intercept Timer—Day, Lunch, Break, Night
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Overflow No
Answer—Time out & Manual Queue Redirection—Destination-Day, Lunch, Break, Night
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings
→ Intercept Destination
→ Intercept No Answer Time
13.3.1 PBX Configuration—[5-3-1] Optional Device—Voice Message—DISA System—Option 1—DISA
Intercept—Intercept when all DISA ports are busy
18.5 PBX Configuration—[10-5] CO & Incoming Call—Miscellaneous
→ Intercept—Intercept Routing - DND (Destination sets DND.)
→ Intercept—Routing to Operator - No Destination (Destination is not programmed.)
→ Intercept—Intercept Routing for Extension Call

Feature Manual References

2.3 Call Forwarding (FWD)/Do Not Disturb (DND) Features

2.1.1.7 Intercept Routing—No Destination

Description
Provides automatic redirection of incoming trunk calls that do not have a destination assigned. The intercept
destination is an operator (tenant/PBX).

Conditions
• Intercept Routing—No Destination on/off
The Intercept Routing—No Destination feature can be enabled or disabled through system programming.
If disabled, a reorder tone will be sent to the caller. However, the Intercept Routing—No Destination
feature always functions for calls through the ELCOT/CLCOT/LCOT, or T1 (LCOT/GCOT) card even
when disabled.
• If an operator (tenant/PBX) is not assigned:
The extension that is connected to the lowest-numbered port and is ready to receive calls will be the
intercept destination.
• Intercept Routing—No Destination also applies to calls from doorphones.

PC Programming Manual References

10.2 PBX Configuration—[2-2] System—Operator & BGM—PBX Operator—Day, Lunch, Break, Night

36 Feature Manual
 2.1.2 Internal Call Features

18.5 PBX Configuration—[10-5] CO & Incoming Call—Miscellaneous—Intercept—Routing to Operator - No

Destination (Destination is not programmed.)

Feature Manual References

5.1.5 Operator Features

2.1.2 Internal Call Features

2.1.2.1 Internal Call Features—SUMMARY
Description
The following types of internal calls are available:
Feature Description & Reference
Intercom Call A call from one extension to another.

→ 2.5.3 Intercom Call

Doorphone Call When a call from a doorphone reaches its destination, the recipient
can talk to the visitor.

→ 2.18.1 Doorphone Call

[Available Destination]
The destinations of doorphone calls can be assigned for each time mode (day/lunch/break/night)
(→ 5.1.4 Time Service) on a doorphone port basis.
Calling from
Destination
Extension Doorphone
Wired Extension (PT/SLT/SIP Extension/ISDN Extension/T1- ü ü
OPX)
PS ü ü
Incoming Call Distribution Group ü ü
PS Ring Group ü ü
UM Group ü ü
Fax Unit ü
External Pager (TAFAS) ü ü
DISA
Idle Line Access no. + Phone no. ü ü
Trunk Group Access no. + Trunk Group no. + Phone no. ü ü
Other PBX Extension (TIE with no PBX Code) ü ü
Other PBX Extension (TIE with PBX Code) ü ü

ü: Available

Feature Manual 37
2.1.2 Internal Call Features

PC Programming Manual References

13.1 PBX Configuration—[5-1] Optional Device—Doorphone—Destination—Day, Lunch, Break, Night

2.1.2.2 Internal Call Block

Description
Internal calls can be restricted on a COS basis. This is done by specifying which COS destinations are
blocked for each COS.

[Programming Example]
Called Party
Caller
COS 1 COS 2 COS 3 ...
COS 1
COS 2 ü ü ü
COS 3 ü ü ü
: : : : :

ü: Block

Explanation:
a. COS 1 extensions can make calls to all extensions.
b. COS 2 extensions can make calls to COS 1 destinations only. (COS 2 extensions cannot make calls to
COS 2 destinations.)
c. COS 3 extensions can make calls to COS 3 destinations only.

COS 1

Extn. 100 Extn. 101

COS 2 COS 3

Extn. 102 Extn. 103 Extn. 104 Extn. 105 Extn. 106

Conditions
• Restricted extension numbers cannot be used as the parameter of a feature setting (e.g., FWD).
• All extensions can make an Operator Call (→ 5.1.5 Operator Features) regardless of Internal Call Block.
• This feature can also restrict calling a doorphone from an extension on the basis of the COSs assigned to
the extension and doorphone port. (→ 2.18.1 Doorphone Call)

38 Feature Manual
 2.1.3 Incoming Call Indication Features

PC Programming Manual References

10.7.3 PBX Configuration—[2-7-3] System—Class of Service—Internal Call Block—COS Number of the
Extension Which Receive the Call from Other Extension 1–512
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Main—COS
8.2 Users—Advanced Extension Settings—COS
13.1 PBX Configuration—[5-1] Optional Device—Doorphone—COS
13.4 PBX Configuration—[5-4] Optional Device—External Relay—COS Number

Feature Manual References

5.1.1 Class of Service (COS)

2.1.3 Incoming Call Indication Features

2.1.3.1 Incoming Call Indication Features—SUMMARY
Description
Incoming calls are indicated by various methods as follows:
Type Feature Description & Reference
Ring Tone Ring Tone Pattern A telephone rings when receiving a call. The ring
Selection tone patterns can be changed for each incoming
call type.

→ 2.1.3.2 Ring Tone Pattern Selection

Voice-calling Alternate Receiving— A PT user can select to receive intercom calls by
Ring/Voice ring tone or by voice, through personal
programming.

→ 2.5.3 Intercom Call

LED LED Indication The light shows line conditions with a variety of
(Light Emitting Diode) light patterns.

→ 2.21.3 LED Indication

Display (Caller’s Display Information The display shows the caller’s information.
Information)
→ 2.21.4 Display Information
External Pager Trunk Answer from The external pager sends a ring tone when
Any Station (TAFAS) receiving a call.

→ 2.17.2 Trunk Answer From Any Station

(TAFAS)
Tone/Voice during a Call Waiting A busy extension hears a tone, or voice from the
Conversation handset/built-in speaker indicating that another
incoming call is waiting.

→ 2.1.3.3 Call Waiting

Feature Manual 39
2.1.3 Incoming Call Indication Features

2.1.3.2 Ring Tone Pattern Selection

Description
It is possible to select the type of ring tone pattern that arrives at an extension for each type of incoming call,
etc.

[Ring Tone Patterns]

1280 ms*

Single

Double

Triple

S-Double

* The duration of a ring tone may vary by country/area.

[Ring Tone Pattern Table]

The ring tone pattern table is categorised into three parts, each containing a specified number of pattern
plans.
The ring tone pattern table is categorised as follows:

• Incoming Trunk Calls: each pattern plan can assign a ring tone pattern for each trunk group.
→ 10.8.1 PBX Configuration—[2-8-1] System—Ring Tone Patterns—Call from CO
• Incoming Doorphone Calls: each pattern plan can assign a ring tone pattern for each doorphone.
→ 10.8.2 PBX Configuration—[2-8-2] System—Ring Tone Patterns—Call from DOORPHONE
• Others: each pattern plan can assign a ring tone pattern for incoming intercom calls as well as ring tones
assigned to certain features (e.g., timed reminder).
→ 10.8.3 PBX Configuration—[2-8-3] System—Ring Tone Patterns—Call from Others
The ring tone patterns that arrive at an extension are determined by the pattern plan that is assigned to that
extension through system programming.

[Programming Example of Ring Tone Pattern Table]

Intercom Trunk Call/ Doorphone

Table Hold Recall Call Timed Call External
Call/Hold LCS
No. Reminder Back Sensor
Recall TRG1 ... Port 1 ...
1 Double Single Single
2 Single Double Double
: : : : : : : : : :

Conditions
• "PT Ring Off Setting" can be enabled or disabled through system programming. If disabled, PT users
cannot turn incoming call ringing off for their extension.

40 Feature Manual
 2.1.3 Incoming Call Indication Features

• For the S-CO, G-CO, L-CO, ICD Group, INTERCOM, PDN and SDN buttons, one of 30 ring tones can be
assigned through personal programming.

PC Programming Manual References

10.8.1 PBX Configuration—[2-8-1] System—Ring Tone Patterns—Call from CO—Ring Tone Pattern Plan
1-8
10.8.2 PBX Configuration—[2-8-2] System—Ring Tone Patterns—Call from DOORPHONE—Ring Tone
Pattern Plan 1–8
10.8.3 PBX Configuration—[2-8-3] System—Ring Tone Patterns—Call from Others—Extension—Ring Tone
Pattern Plan 1–8
10.9 PBX Configuration—[2-9] System—System Options—Option 1—PT Operation—PT Ring Off Setting
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings
→ Option 1—Ring Pattern Table
→ Option 6—ICM Tone
8.3.1 Users—Flexible Button—Wired Extension—Optional Parameter (Ringing Tone Type Number) (for
Loop CO, Single CO, Group CO, ICD Group, SDN)
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 1—Ring Pattern Table

Feature Manual References

6.1 Capacity of System Resources

Operating Manual References

3.1.3 Customising the Buttons—To specify the ringing tones for each CO, ICD Group, PDN, SDN, or
INTERCOM button (DPT/IP-PT only)

2.1.3.3 Call Waiting

Description
Used to inform a busy extension that another incoming call is waiting. The busy extension user can answer
the second call by disconnecting the current call or placing it on hold. This feature is also known as Busy
Station Signalling (BSS).
The following notification method can be assigned for each extension depending on the call waiting and the
telephone type:
a. Call Waiting Tone: Tone from the handset or built-in speaker
b. OHCA: Voice from the built-in speaker
c. Whisper OHCA: Voice from the handset
d. Off: No notification.
Notification Method
Call Type
DPT IP-PT Other Telephone
Intercom Call Call Waiting tone/OHCA/ Call Waiting tone/ Call Waiting tone/Off
Whisper OHCA/Off Whisper OHCA/Off
Trunk Call*1 Call Waiting tone/Off

*1 Including a doorphone call, call via an incoming call distribution group, and a trunk call transferred from another extension.

Conditions
• Automatic Call Waiting

Feature Manual 41
2.1.3 Incoming Call Indication Features

Through system programming, it is possible to select whether a call waiting tone is automatically sent to
the extension when receiving trunk calls, doorphone calls, external sensor calls and hold-recall calls.
Through system programming, it is also possible to select whether extensions will receive Automatic Call
Waiting from intercom calls.
• Call Waiting for an extension in a UM group is not available.
• Data Line Security
Setting Data Line Security cancels the Call Waiting setting. (→ 2.11.5 Data Line Security)
• Call Waiting Tone
A PT user can hear different Call Waiting tones for trunk call and intercom call if "Tone 2" has been
selected through personal programming (Call Waiting Tone Type Selection). If "Tone 1" has been
selected, the same Call Waiting tone will be heard for both trunk call and intercom call.
All Call Waiting tone patterns have a default (→ 6.2.1 Tones/Ring Tones).
• Caller Information
With the Call Waiting tone, the caller’s information flashes on the display for five seconds, followed by a
10-second pause, then flashes again for five seconds.
• Call Waiting from the Telephone Company
Besides the Call Waiting service within the PBX, the Call Waiting tone offered by an analogue line from
the telephone company informs the extension user of another incoming trunk call that is waiting. He can
answer the second call by disconnecting the current call or placing it on hold using EFA
(→ 2.11.7 External Feature Access (EFA)). For details, consult your telephone company.
Call Waiting Caller ID (Visual Caller ID):
When using the call waiting tone supplied by the telephone company over analogue lines, the waiting
caller’s telephone number can be received. The number will flash on the display for five seconds, followed
by a 10-second pause, then flash again for five seconds.
Note that the received caller information may not be displayed on telephones or wireless phones
connected to SLT ports.

PC Programming Manual References

9.23 PBX Configuration—[1-1] Configuration—Slot—Card Property - LCO type—Caller ID—Caller ID
Signalling
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Manual Call Waiting for Extension Call
→ Automatic Call Waiting
10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature
→ BSS / OHCA / Whisper OHCA / DND Override
→ BSS / OHCA / Whisper OHCA / DND Override-2
10.9 PBX Configuration—[2-9] System—System Options—Option 5—Call Waiting—Automatic Call Waiting
for Extension Call
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 4—Call Waiting Tone
Type

Feature Manual References

2.10.4 Second Call Notification to Busy Extension
2.19.2 Incoming Call Log
2.22.1.1 Station Message Detail Recording (SMDR)

Operating Manual References

1.4.4 Answering Call Waiting
1.9.3 Receiving Call Waiting (Call Waiting/Off-hook Call Announcement [OHCA]/Whisper OHCA)
3.1.2 Settings on the Programming Mode

42 Feature Manual
 2.2 Receiving Group Features

2.2 Receiving Group Features

2.2.1 Idle Extension Hunting
Description
If a called extension is busy or in DND mode, Idle Extension Hunting redirects the incoming call to an idle
member of the same idle extension hunting group, which can be programmed through system programming.
Idle extensions are automatically searched according to a preprogrammed hunting type. This feature is also
known as Station Hunting.
Type Description
Circular Hunting An idle extension is searched for in the order specified in the idle
extension hunting group in a circular way.
Incoming call

Busy
Extn. Extn. Extn.
Extn.
Assigned order

Terminated Hunting An idle extension is searched for in the order specified in the idle
extension hunting group until reaching the last assigned extension.
Incoming call

Busy
Extn. Extn. Extn.
Extn.
Assigned order

Conditions
• Idle Extension Hunting applies to:
Intercom, trunk, and doorphone calls to a single destination.
• An extension user can belong to only one idle extension hunting group.
• If all the searched extensions are busy:
The PBX redirects the call to an overflow destination which can be assigned for each idle extension
hunting group and each time mode (day/lunch/break/night) (→ 5.1.4 Time Service).

[Available Destination]
Destination Availability
Wired Extension (PT/SLT/SIP Extension/ISDN Extension/T1-OPX) ü
PS ü
Incoming Call Distribution Group ü
PS Ring Group ü
UM Group ü
Fax Unit

Feature Manual 43
2.2.2 Incoming Call Distribution Group Features

Destination Availability
External Pager (TAFAS) ü
DISA ü
Idle Line Access no. + Phone no. ü
Trunk Group Access no. + Trunk Group no. + Phone no. ü
Other PBX Extension (TIE with no PBX Code) ü
Other PBX Extension (TIE with PBX Code) ü

• FWD/DND Mode
While searching for an idle extension within an idle extension hunting group, any extension that has set
FWD—All Calls or DND feature will be skipped, and the call will go to the next extension in the group.

PC Programming Manual References

11.6 PBX Configuration—[3-6] Group—Extension Hunting Group
11.6.1 PBX Configuration—[3-6] Group—Extension Hunting Group—Member List

Feature Manual References

2.3 Call Forwarding (FWD)/Do Not Disturb (DND) Features
6.1 Capacity of System Resources

2.2.2 Incoming Call Distribution Group Features

2.2.2.1 Incoming Call Distribution Group Features—SUMMARY
Description
An incoming call distribution group is a group of extensions programmed through system programming.
→ 11.5.1.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Member List

An incoming call distribution group receives calls directed to the group. Each incoming call distribution group
has a floating extension number (default: 6 + two-digit group number [up to group 64]).
Incoming calls directed to an incoming call distribution group are distributed to the member extensions in the
group using a distribution method. When a preprogrammed number of extensions in the group are busy, the
incoming calls can wait in a queue.
Each incoming call distribution group and member extensions can be programmed as desired to handle
incoming calls. Calls to the group can be monitored by an extension assigned as a supervisor.
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings
→ Main
→ Overflow Queuing Busy
→ Overflow No Answer
→ Miscellaneous

44 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

Programming Item Example for Incoming Call Distribution Group 1 with

Diagram
A through F in the table are described in the following diagram.
A B C D E F
Group Floating Group Distribution Max. No. of Queuing Hurry-up Overflow Overflow Destination Tenant
Busy Call
No. Extn. No.*1 Name Method Capacity Level Time Day ... ... Night No.*2
Extensions
1 601 Sales Ring 3 5 3 60 100 ... ... 100 1
2 602 Engineering UCD Max. 11 8 90 200 ... ... 200 5
3
:

*1 The number of digits for Floating Extn. No depends on the value specified for Numbering Plan in Easy Setup.
→ 2.1.4 Easy Setup Wizard.
*2 The tenant number is required to determine the time mode (day/lunch/break/night) (→ 5.1.4 Time Service) and the music source
(for Music on Hold) for each group.

Calls arriving at incoming call

distribution group 1.

9 F Overflow Feature
a) Sends a busy tone (Busy on Busy), or
8 b) Redirects to the overflow destination.
7
6
D Queuing Feature
Five calls are 5
waiting in a queue. 4 E Manual Queue Redirection *1
The longest waiting call in a queue
B Group Call Distribution 3 can be redirected to the overflow
Calls are distributed by the destination by pressing the Hurry-up
2 button. The button shows the Hurry-
assigned method.
(Only three extensions 1 up status.
[agents] can answer the
call for C Busy on Busy.)

Supervisor Extension*2
Extn. Extn. Extn. Extn. Extn. Extn. Extn.
100 101 102 103 104 105 105

Monitors or controls the A Incoming Call Log-in Log-out *3

incoming call distribution Distribution Group 1
(Floating extension no.: 601;
group status.
Name: Sales)

*1 → 2.2.2.3 Queuing Feature

*2 → 2.2.2.8 Supervisory Feature
*3 → 2.2.2.7 Log-in/Log-out

1. Group Call Distribution [→ 2.2.2.2 Group Call Distribution]

Incoming calls are distributed using one of the following methods:
Distribution Method Description
Uniform Call Distribution Calls are distributed evenly to a different extension each time
(UCD) a call is received.

Feature Manual 45
2.2.2 Incoming Call Distribution Group Features

Distribution Method Description

Priority Hunting An idle extension is searched for in the specified order,
always starting from the same location.
Ring All extensions in the incoming call distribution group ring
simultaneously.

2. Queuing Feature [→ 2.2.2.3 Queuing Feature]

If a preprogrammed numbers of extensions in an incoming call distribution group are busy, a
preprogrammed number of additional calls can wait in a queue.
While calls are waiting in the queue, an outgoing message (OGM) or Music on Hold can be sent to the
waiting callers.
3. VIP Call [→ 2.2.2.5 VIP Call]
It is possible to assign a priority to incoming call distribution groups so that incoming calls can be
received in priority order.
4. Overflow Feature [→ 2.2.2.6 Overflow Feature]
A call is redirected to a preprogrammed destination when it cannot be answered or queued (Intercept
Routing—Overflow in an Incoming Call Distribution Group). It is also possible to send a busy tone
(Busy on Busy) or disconnect the line.
5. Incoming Call Distribution Group Controlling Feature
Feature Description & Reference
Log-in/Log-out Member extensions can join the group to
handle calls (Log-in) or leave the group for a
break (Log-out).
They can leave the group temporarily when
they are away from their desks, to prevent calls
being sent to their extensions.

→ 2.2.2.7 Log-in/Log-out
Supervisory Feature Incoming Call Queue The supervisor extension can monitor various
Monitor information about the incoming calls for each
incoming call distribution group on his display.

→ 2.2.2.8 Supervisory Feature

Log-in/Log-out Monitor: The supervisor extension can monitor
Monitor and Remote the log-in/log-out status of the group members.
Control Remote Control: The supervisor extension can
change the status of the members.

→ 2.2.2.8 Supervisory Feature

Conditions
• One extension can belong to multiple incoming call distribution groups.
• ICD Group button
An Incoming Call Distribution (ICD) Group button can be assigned on a flexible button for each incoming
call distribution group. It receives the incoming calls to the group.
One extension can have more than one ICD Group button of the same or different incoming call
distribution groups (Multiple ICD Group). If all ICD Group buttons in the same incoming call distribution
group are occupied, the next incoming call will be held in a queue or will overflow. If the ICD Group button
is not assigned, incoming calls will arrive at the INTERCOM, CO or PDN button.

46 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

The mode of ICD Group buttons can be selected through system programming, as follows:
– Standard Mode (Group DN Button Mode)
An extension can have an ICD Group button for an incoming call distribution group that the extension
does not belong to through system programming. However, the ICD Group button will not receive calls
to that group.
– Enhanced Phantom Button Mode
An extension can join an ICD Group just by creating a button for that group, even if the extension was
not previously registered as a member. When the button is created, the extension will be automatically
registered in the lowest-numbered available member slot for the group. Calls to the group can be
received at the extension with no further programming. If no member slots are available for that group,
the button cannot be created and an alarm tone will be heard.
When creating an ICD Group button in this mode, the user can also specify the delayed ringing
settings.
If an extension user deletes the last ICD Group button at his extension for a certain group, he will also
be deregistered as a member from that group.
• Group FWD
The FWD feature can be assigned on an incoming call distribution group basis.
• COS for Incoming Call Distribution Groups
Each incoming call distribution group is assigned a COS number. Group FWD to an outside party can be
enabled or disabled for each COS. The COS for incoming call distribution groups is also used for the
Internal Call Block feature; when an extension user calls an incoming call distribution group, the PBX
checks the COS of the calling extension against the COS of the incoming call distribution group
(→ 2.1.2.2 Internal Call Block).

PC Programming Manual References

2.1.4 Easy Setup Wizard
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings
→ Main
→ Overflow Queuing Busy
→ Overflow No Answer
→ Miscellaneous
11.5.1.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Member
List
11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous—Options—
ICD Group Key Mode

Feature Manual References

2.3.2 Call Forwarding (FWD)
2.21.2 Flexible Buttons
5.5.7 Floating Extension
6.1 Capacity of System Resources

2.2.2.2 Group Call Distribution

Description
Incoming calls directed to an incoming call distribution group are distributed to the member extensions using
the selected distribution method until a preprogrammed number of extensions (agents) are busy with calls.
When incoming calls exceed the number of available extensions, calls enter a queue (→ 2.2.2.3 Queuing
Feature).
1. Distribution Method
One of the three distribution methods below can be assigned to each incoming call distribution group.

Feature Manual 47
2.2.2 Incoming Call Distribution Group Features

Distribution Method Description

Uniform Call Calls are distributed evenly to a different extension each time a call is
Distribution (UCD) received. Extensions are hunted in a circular way in the
preprogrammed order for the group, starting at the extension after the
extension that received the last call.

Extn. Extn. Extn. Extn.

A B C D

Received Starts searching from

the last call. extn. B. (Skips extn. A.)

The call will be directed to the user that has been idle (the time that a
user has not been on a call on any of the owned devices) for the
longest period of time. This is known as Automatic Call Distribution
(ACD).
Priority Hunting An idle extension is searched for using the preprogrammed order for
the group.
1st Priority 2nd Priority 3rd ....
Extn. Extn. Extn. Extn.
A B C D

Always starts searching from

the first assigned extension.

Ring All extensions in the group ring simultaneously.

Delayed Ringing:
Delayed ringing or no ringing can be programmed for each extension
in the group. The call can be answered by pressing the flashing
button even if no ring or a delayed time is set.

Extn. Extn. Extn. Extn.

A B C D

Rings immediately simultaneously. Delayed Ringing:

Rings after a
specified time delay.

2. Call Waiting for Incoming Call Distribution Group (Group Call Waiting)
When there are no available extensions in an incoming call distribution group, the group members can
receive the Call Waiting tone. To use this feature:
• Select the Group Call Waiting mode through system programming. This determines the distribution
method for waiting calls.
• Member extensions must assign the Call Waiting mode individually, or they will not be notified. (→
2.1.3.3 Call Waiting)

[How the Group Call Waiting Feature Activates]

48 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

Programming Conditions Result

Group Call Group Call Group Call Waiting Capable
Waiting Mode Distribution Method Distribution Method Telephone
Distribution UCD UCD PT/PS with idle
Priority Hunting Priority Hunting ICD Group button
Ring Not available* Any telephone
All UCD/Priority Hunting/ Ring
Ring

* Incoming calls enter the queue immediately. Member extensions do not receive the Call Waiting tone.

[Example]

• Group Call Waiting mode: All

All extensions hear the Call
• Group call distribution method
Waiting tone (Ring).
for idle extensions: UCD

[ICD Group Button for Group Call Waiting]

The way that the Group Call Waiting feature works depends on the Group Call Waiting Distribution
method as follows:
a. Ring: The Group Call Waiting feature activates for all busy member extensions (even when the
extensions do not have ICD Group buttons) simultaneously for only one incoming call—additional
calls will wait in a queue.
b. UCD/Priority Hunting: The Group Call Waiting feature activates on an idle ICD Group button
located on busy member extensions in a certain order. (This order depends on the type: UCD or
Priority Hunting.) Calls will arrive at idle buttons until all ICD Group buttons are occupied—
additional calls will wait in a queue.

Note
In method b), if an extension has one or more ICD Group buttons for an incoming call
distribution group and all the ICD Group buttons on the extension are occupied, the Group
Call Waiting feature for the group will not work at the extension.

Incoming Call Incoming Call

Distribution Distribution
Group 1 3 Group 2
(Floating (Floating 1
extension 2 extension
no.: 601) no.: 602)

ICD Group 1; 601 (Call Waiting)

ICD Group 1; 601 (Call Waiting)
ICD Group 2; 602 (Answering the Call)

3. No Reply Redirection (UCD or Priority Hunting Method)

If a call received at a member extension is not answered within a preprogrammed time period (No
Answer time), the call will be redirected to the next member extension. If there is no idle group
member, the call queues at the target extension until a group member becomes available.

Feature Manual 49
2.2.2 Incoming Call Distribution Group Features

Conditions
• Automatic Call Distribution (ACD) does not work for ISDN extensions or PS Ring Groups.
• FWD/DND Extension
System programming for each incoming call distribution group is required to skip or ring extensions which
have the FWD or DND feature set. If set to ring, the FWD/DND settings are ignored. (→ 2.3 Call
Forwarding (FWD)/Do Not Disturb (DND) Features)
• The Group Call Waiting feature cannot be used with the VIP Call feature (→ 2.2.2.5 VIP Call) and/or
Wrap-up feature (→ 2.2.2.7 Log-in/Log-out). To use the VIP Call feature and/or Wrap-up feature, Call
Waiting mode on each extension should be off.

PC Programming Manual References

11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Main—Line Hunting Order
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings
→ Distribution Method
→ Call Waiting Distribution
→ Extension No Answer Redirection Time
→ Maximum No. of Busy Extension
11.5.1.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Member
List—Delayed Ring
11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous
→ Options—Wrap-up Timer based on
→ Options—Longest Idle Distribution

2.2.2.3 Queuing Feature

Description
When a preprogrammed number of extensions in an incoming call distribution group are busy, additional
incoming calls can wait in a queue. The number of calls which can wait in the queue is programmable.
While calls are waiting in the queue, the calls are handled by the Queuing Time Table, which can be
assigned for each time mode (day/lunch/break/night) (→ 5.1.4 Time Service). This PBX supports a specified
number of Queuing Time Tables which each have a specified number of sequences (specific commands
which are performed when a caller enters a queue). The following commands can be assigned to each
sequence when making a Queuing Time Table:

[Command Table]
Command Description Condition
OGM xxx An outgoing message is sent to the After the OGM, Music on Hold will be
caller. "xxx" applies to the OGM sent and the next event in the sequence
number. will be activated.
Wait xx s Puts the caller in the waiting queue If an OGM has not been sent to the
for b (01-16) × 5 seconds. caller, the caller hears a ringback tone.
"xx" applies to the number of If an OGM has been sent to the caller,
seconds to wait (05-80). the caller hears Music on Hold.
Sequence xx Redirects to sequence xx. "xx" None
applies to the sequence number.
Overflow Redirects to the overflow destination. None
Disconnect Disconnects the line. None

50 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

Command Description Condition

Queue No. Announces the number of calls in Activation key required:
the waiting queue ahead of the Call Centre Feature Enhancement
caller. (KX-NSXF002 or KX-NSXF021)
Queue No. and Time Announces the number of calls in Activation key required:
the waiting queue ahead of the caller Call Centre Feature Enhancement
and the estimated waiting time. (KX-NSXF002 or KX-NSXF021)
None Redirects to the next sequence. If assigned as sequence 01, the
(No command) Queuing Time Table will not be
activated.

[Programming Example of Queuing Time Table]

Queuing Time Table Sequence*1

No. Sequence 01 Sequence 02 Sequence 03 Sequence 04 ...
01 OGM 001 Wait 30 s OGM 003 Overflow
02*2 OGM 002 Wait 30 s OGM 004 None
03
: : : : : :

*1 → 11.5.2 PBX Configuration—[3-5-2] Group—Incoming Call Distribution Group—Queuing Time Table—Queuing Sequence—
Sequence 01–16
*2 If a call has not reached a destination by the time the final sequence is completed, the call will be disconnected.

Explanation for Queuing Time Table 01:

Queuing Time Table 01

Sequence 01 Sequence 02 Sequence 03 Sequence 04 Overflow
The call
OGM 001 is sent. Music on OGM 003 is sent. Redirects to destination
queues.
Hold is sent the overflow answers.
Thank you for for 30 We are sorry to destination.
calling Panasonic. seconds. keep you holding.
The department The department
you are calling is is still busy. We
busy. Please are transferring
hold the line. We you to the
will answer your operator.
call shortly.

The call is connected to the member

extension as soon as the extension
becomes available.

Exit from Queuing

If DISA Automated Attendant settings (→ 2.16.1 Direct Inward System Access (DISA)—DISA Built-in
Automated Attendant Number (DISA AA Service)) are assigned to an OGM used in the Queuing Time Table,
the caller can dial a digit while listening to the OGM and be redirected to the destination assigned to that

Feature Manual 51
2.2.2 Incoming Call Distribution Group Features

digit. For example, programming the VM group floating extension number as DISA AA number 1 for an
OGM allows a caller to be redirected to voice mail:
(OGM) "Thank you for calling. The department you are calling is busy. Please hold, or press 1 to leave a
voice message."
If the caller presses 1 while the OGM is playing, she will be redirected to voice mail where she can leave a
voice message.

Conditions
• If the call is transferred to the incoming call distribution group and is handled by the Queuing
Time Table:
Transfer Recall will not occur even if the Transfer Recall time expires.
• Manual Queue Redirection
It is possible to redirect the longest waiting call in a queue to the overflow destination by pressing the
Hurry-up button. (If the call is already ringing at an extension, it will not be redirected.) This feature is also
known as Hurry-up Transfer.
• Hurry-up Button
A flexible button can be customised as the Hurry-up button. The number of calls queuing before Manual
Queue Redirection may be performed is programmable. The button shows the current status as follows:
Light Pattern Calls in the Waiting Queue
Off No queued call
Red on At or under the assigned number for Hurry-up
Rapid red flashing Over the assigned number for Hurry-up

The editing of the guidance used in the Queue announcement feature/Backup/Restore feature can only
be used when the optional Storage Memory Card is mounted.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Incoming Call Queue
Monitor
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings
→ Overflow Queuing Busy
→ Overflow No Answer
→ Queuing Time Table
→ Miscellaneous—Extension No Answer Redirection Time
→ Miscellaneous—Maximum No. of Busy Extension
11.7.1 PBX Configuration—[3-7-1] Group—UM Group—System Settings—Call Waiting on UM Group
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Extension Number (for Hurry-up)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Extension Number (for Hurry-up)

Feature Manual References

2.2.2.6 Overflow Feature
2.28.2 Outgoing Message (OGM)
2.21.2 Flexible Buttons
6.1 Capacity of System Resources

52 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

Operating Manual References

1.10.3 Forwarding a Waiting Call (Manual Queue Redirection)

2.2.2.4 Distribution Order

Description
The distribution method of queued calls in an incoming call distribution group to an idle extension that
belongs to several groups can be selected through system programming.
The following modes are available.
Mode Description
Circular The extension receives the first available call that was received
to the incoming call distribution group which comes after the
incoming call distribution group that sent the previous call.
FIFO (First In First Out) The extension receives the first available call that was received
from all the incoming call distribution groups to which it belongs.

[Example]
- Circular

Calls have been distributed by DIL/DID/DDI/CLI.

(The number in the circle is the queuing order.)

Incoming Call Incoming Call Incoming Call

Distribution 6 Distribution 5 Distribution 4
Group 1 1 Group 2 3 Group 3 2

Distribution order: 1 3 2 6 5 4

Explanation:
Calls are distributed according to the order of the incoming call distribution groups. After call 1 in group 1,
the next call received was call 2 in group 3. However, call 3 in group 2 is given priority with this method.
- FIFO

Feature Manual 53
2.2.2 Incoming Call Distribution Group Features

Calls have been distributed by DIL/DID/DDI/CLI.

(The number in the circle is the queuing order.)

Incoming Call Incoming Call Incoming Call

Distribution 6 Distribution 5 Distribution 4
Group 1 1 Group 2 3 Group 3 2

Distribution order: 1 2 3 4 5 6

Explanation:
Calls are distributed according to the order they were received irrespective of the group distribution order.

PC Programming Manual References

11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous
→ Options—ICD Group Distribution order

2.2.2.5 VIP Call

Description
It is possible to assign a priority to incoming call distribution groups. If an extension belongs to multiple
groups and the extension becomes idle, queuing calls in the groups will be distributed to the extension in
priority order.
Each incoming call distribution group can enable or disable the VIP Call mode. When the VIP Call mode is
enabled for a group, that group is given priority over other groups irrespective of the group distribution order.
When multiple groups enable the VIP Call mode, the incoming call distribution group with the lowest
numbered group has the highest priority. After all calls queued in a group have been distributed, the calls
queued in the next lowest-numbered group will be distributed.
When some groups have the VIP Call mode enabled and others do not, after all the calls queued in the
enabled groups have been distributed, the calls queued in the other groups will be distributed according to
the set distribution order.

[Example]
In the call centre, incoming call distribution groups 1 and 3 enable the VIP Call mode, while incoming call
distribution groups 2 and 4 disable the VIP Call mode.

54 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

- Circular

Calls have been distributed by DIL/DID/DDI/CLI.

(The number in the circle is the queuing order.)

Incoming Call Incoming Call Incoming Call Incoming Call

Distribution 8 Distribution 7 Distribution 6 Distribution 5
Group 1 1 Group 2 3 Group 3 4 Group 4 2

Distribution order: 1 8 4 6 3 2 7 5

Explanation:
Calls queued in groups 1 and 3, with VIP Call mode enabled, are given priority according to their group.
After call 1 in group 1, several other calls are received. However, call 8 in group 1 is distributed next.
Once all the calls in groups 1 and 3 have been distributed, the calls in groups 2 and 4 are distributed evenly
from the lowest numbered group.

- FIFO

Calls have been distributed by DIL/DID/DDI/CLI.

(The number in the circle is the queuing order.)

Incoming Call Incoming Call Incoming Call Incoming Call

Distribution 8 Distribution 7 Distribution 6 Distribution 5
Group 1 1 Group 2 3 Group 3 4 Group 4 2

Distribution order: 1 8 4 6 2 3 5 7

Explanation:
Since groups that have VIP Call mode enabled do not follow the regular distribution order, the calls in those
groups are distributed in the order of 1, 8, 4, 6.
Once all the calls in groups 1 and 3 have been distributed, the calls in groups 2 and 4 are distributed in the
order they were received.

Feature Manual 55
2.2.2 Incoming Call Distribution Group Features

PC Programming Manual References

11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Miscellaneous—VIP Call Mode

Feature Manual References

2.2.2.4 Distribution Order

2.2.2.6 Overflow Feature

Description
When waiting calls exceed the waiting queue capacity, they may be redirected to a preprogrammed
destination or a busy tone may be sent to the callers by the following features:
1. Intercept Routing—Overflow in an Incoming Call Distribution Group
2. Busy on Busy

1. Intercept Routing—Overflow in an Incoming Call Distribution Group

Intercept Routing—Overflow in an Incoming Call Distribution Group works in one of following
conditions:
a. There is no space in the waiting queue.
b. The Queuing Time Table is not assigned and there are no extensions logged in.
c. The Queuing Time Table is assigned, but there are no extensions logged in and the "Overflow
immediately when All Logout" setting is enabled.
d. An Overflow command is assigned to the Queuing Time Table.
e. The Overflow time expires.
f. Manual Queue Redirection is performed.

[Available Destination]
The overflow destinations can be assigned for each incoming call distribution group and each time
mode (day/lunch/break/night) (→ 5.1.4 Time Service). The destination can be assigned as follows,
depending on the above conditions.
• For a), b), and c):
→ 11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Overflow Queuing Busy—Queuing Busy—Destination-Day, Lunch, Break, Night
• For d), e), and f):
→ 11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Overflow No Answer—Time out & Manual Queue Redirection—Destination-Day, Lunch, Break,
Night
Destination Availability
Wired Extension (PT/SLT/SIP Extension/ISDN Extension/T1-OPX) ü
PS ü
Incoming Call Distribution Group ü
PS Ring Group ü
UM Group ü
Fax Unit

56 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

Destination Availability
External Pager (TAFAS) ü
DISA ü
Idle Line Access no. + Phone no. ü
Trunk Group Access no. + Trunk Group no. + Phone no. ü
Other PBX Extension (TIE with no PBX Code) ü
Other PBX Extension (TIE with PBX Code) ü

2. Busy on Busy
The Busy on Busy feature works when the destination for the Intercept Routing—Overflow in an
Incoming Call Distribution Group feature is not assigned in one of the following conditions:
a. There is no space in the Waiting queue.
b. The Queuing Time Table is not assigned and there are no extensions logged-in.

[Example of a)]
There are five assistants in a shop. When the answering agent number is "2", and the queuing call
number is "0":
If two of the assistants are talking on the phone, the next caller will hear a busy tone to prevent the
caller from thinking that there is no one in the shop or that the shop is closed.

Conditions
[Intercept Routing—Overflow in an Incoming Call Distribution Group]
• If the Overflow time expires, and the overflow destination is unavailable:
a. If the trunk call arrives through the ELCOT/CLCOT/LCOT or T1 (LCOT/GCOT) card:
(1) If the call was once in a queue and an outgoing message (OGM) was sent to it, or the call
reached an incoming call distribution group by using the DISA feature (→ 2.16.1 Direct Inward
System Access (DISA)): The line is disconnected.
(2) In all other cases: Redirection is ignored and the Overflow timer activates again.
b. If the call arrives through another card: Redirection is ignored and the Overflow timer activates
again.

[Busy on Busy]
• If a trunk call arrives through the ELCOT/CLCOT/LCOT or T1 (LCOT/GCOT) card, a busy tone will not be
sent to the caller.

PC Programming Manual References

11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings
→ Overflow Queuing Busy
→ Overflow No Answer
11.5.2 PBX Configuration—[3-5-2] Group—Incoming Call Distribution Group—Queuing Time Table
11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous—Options—
Overflow immediately when All Logout

Feature Manual References

2.2.2.3 Queuing Feature

Feature Manual 57
2.2.2 Incoming Call Distribution Group Features

2.2.2.7 Log-in/Log-out
Description
Members of an incoming call distribution group can join (Log-in) or leave (Log-out) the group manually.
They can leave the group temporarily when they are away from their desks, to prevent calls being sent to
their extensions. They can return to the group when they are ready to answer calls.
Wrap-up:
While logged in, a member extension can have a preprogrammed time period automatically for refusing
calls after completing the previous call (Wrap-up time). While the Wrap-up timer is active, calls to all
incoming call distribution groups to which the extension belongs will skip the extension so that the extension
user can perform necessary tasks such as reporting on the previous call.
The extension will receive calls when the extension status changes from a status where it cannot receive
calls (Busy, Log-out, Wrap-up, or Not Ready) to a status where it can receive calls (Log-in or Ready).
Wrap-up mode can also be activated manually (Not Ready) by pressing the Wrap-up button.

[Log-in/Log-out and Wrap-up Status Example]

<When the incoming call distribution group is in Priority Hunting distribution method>

Incoming call Ready

Log-in Waiting for a call
Press the
Wrap-up button.

Extn. Extn. Extn. Extn. Extn. Extn. Not Ready

Answering a call The Wrap-up Making a report/
101 102 103 104 105 106 time expires. temporary break
Ready Ready Not Ready Wrap-up Ready
After
completing Press the
the call Wrap-up button.
Wrap-up
Extn. Making a report
102
Log-out

Conditions
• It is programmable whether the last remaining logged-in extension can log out.
• Log-in/Log-out Button
A flexible button can be customised as the Log-in/Log-out button with the following parameters:
Light Pattern
Parameter Usage
Red on Off
No parameter Used with an ICD Group — —
button, or with the floating
extension number of an
incoming call distribution
group, or with (All).
Floating extension number of Used to log in to or out of the Log-out Log-in Status
a specified incoming call specified incoming call Status
distribution group distribution group.

58 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

Light Pattern
Parameter Usage
Red on Off
(All) Used to log in to or out of all After Log-out After Log-in
incoming call distribution Operation Operation
groups to which the extension
user belongs.

• If an ICD Group button is assigned, it also shows the log-in/log-out status of the corresponding group.
The light pattern is the same as the Log-in/Log-out button that includes the group number.
• Wrap-up Timer
– Two wrap-up timers can be programmed, an ICD Group member wrap-up timer and an extension
wrap-up timer. System programming selects which timer is used. When the ICD Group member wrap-
up timer is selected, the timer is only activated after calls to the extension through an ICD Group. When
the extension wrap-up timer is selected, the timer is activated after all calls to or from the device which
belong the user container including a retrieved call on hold.
– Only calls from ICD Groups cannot be received during the wrap-up time. Other calls are received as
normal.
– The wrap-up timer does not work for ISDN extensions or PS Ring Groups.
• Wrap-up Button
A flexible button can be customised as the Wrap-up button. It shows the current status as follows:
Light pattern Status
Slow red flashing Wrap-up
Red on Not Ready
Off Ready (Wrap-up mode cancel)

• Automatic Log-out
A member extension may be logged out automatically, if the Unanswered time expires a preprogrammed
number of times consecutively. The number of consecutive unanswered calls can be assigned for each
incoming call distribution group. If the extension is a member of more than one incoming call distribution
group, the unanswered number is counted across all corresponding incoming call distribution groups. It is
possible to return to log-in mode manually.
The Automatic Log-out feature does not work for extensions in an incoming call distribution group using
the Ring distribution method (→ 2.2.2.2 Group Call Distribution).
• Log-in/Log-out Monitor
The supervisor extension can monitor and control the log-in/log-out status of the incoming call distribution
group members. (→ 2.2.2.8 Supervisory Feature)
• Log-in/Log-out Information on SMDR
Log-in/Log-out information can be printed out on SMDR. (→ 2.22.1.1 Station Message Detail Recording
(SMDR))

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Log-in / Log-out
→ Not Ready (Manual Wrap-up) Mode On / Off
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Miscellaneous
→ No. of Unanswered Calls for Automatic Log-out
→ Last Extension Log-out

Feature Manual 59
2.2.2 Incoming Call Distribution Group Features

11.5.1.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Member

List—Wrap-up Timer
11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous—Options—
Wrap-up Timer based on
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 8—Wrap-up Timer
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Parameter Selection (for Log-in/Log-out)
→ Extension Number (for Log-in/Log-out)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Parameter Selection (for Log-in/Log-out)
→ Extension Number (for Log-in/Log-out)
19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR—Print Information—Log-in / Log-out

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

1.10.1 Leaving an Incoming Call Distribution Group (Log-in/Log-out, Wrap-up)

2.2.2.8 Supervisory Feature

Description
An extension preprogrammed as a supervisor (supervisor extension) can monitor and control each
member’s status within the incoming call distribution group using a 6-line display PT.
Feature Description
Incoming Call Queue Monitor The supervisor extension can monitor the status of an incoming
call distribution group with the display.
Log-in/Log-out Monitor and Monitor: The supervisor extension can monitor the log-in/log-
Remote Control out status of the incoming call distribution group members
through the corresponding DSS button light.
Remote Control: The supervisor extension can change the
status of the members by pressing the corresponding DSS
button.

60 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

[Example]
<Incoming Call Queue Monitor Display>

JAN.31 08:13AM FRI --- Date and time

601:Sales Section --- ICD group’s floating extension number and name
Waiting Calls Now :00006 --- Number of calls waiting in the queue
Max. Waiting Time :05'10 --- Elapsed waiting time of the call queuing the longest

EXIT LOG SPRVS

Since JAN.29 09:10AM --- Monitoring starting date and time
Total Calls :00996 --- Total number of incoming calls
Overflow Calls :00131 --- Total number of overflowed calls
Lost Calls :00039 --- Total number of lost calls
Average Waiting :02'12 --- Average queuing time
EXIT CLEAR

<Log-in/Log-out Monitor/Remote Control Mode with DSS Button light>

JAN.31 08:13AM FRI

601:Sales Section
Waiting Calls Now :00006
Max. Waiting Time :05'10 With

EXIT

Log-in/Log-out Monitor Log-in/Log-out Remote Control

DSS buttons show the status Pressing a DSS button
of the corresponding group changes the extension’s
members. status as follows:
Light pattern Status Status Light pattern
Green on Log-in (Ready) Log-out Red on
Slow Green Flashing Log-in (Not Ready) Log-in (Ready) Green on
Red on Log-out
Off Extension in another ICD
group

Conditions
• Available Extension as a Supervisor Extension
a. One supervisor extension can be assigned for each incoming call distribution group, but it need not
belong to the group.
b. One extension can be the supervisor extension of more than one incoming call distribution group.
• Available Paired DSS Console
This feature is available for the KX-DT390, and KX-DT590.
• Accumulation Value Clear
Accumulation value data (total incoming calls, total overflowed calls, lost calls, average queuing time) can
be cleared manually. The date and time of clearing is saved and is shown on the display (monitoring
starting date and time). When the value exceeds 99999 before clearing, "****" will be shown.
• If a call to an incoming call distribution group is overflowed:
If the display is in idle status, it will change to monitor mode for the corresponding incoming call
distribution group automatically.
If the display is monitoring another incoming call distribution group, it will not change.
• Other Features while in Monitor Mode

Feature Manual 61
2.2.2 Incoming Call Distribution Group Features

The supervisor can use other features on the extension (making calls, pressing the MESSAGE button,
etc.) even while in monitor mode. When each operation is finished, his telephone returns to the queue
monitor display.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Incoming Call Queue
Monitor
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Miscellaneous—Supervisor Extension Number

Operating Manual References

1.10.2 Monitoring and Controlling the Call Status of an Incoming Call Distribution Group (Incoming Call
Distribution Group Monitor)

2.2.2.9 Supervisory Feature (ACD)

Description
By specifying an ACD supervisor for extension users, the ACD supervisor can check and analyse the
operating conditions of an ICD group. This is done by monitoring the current condition of the ICD group, and
by collecting and analysing statistical ACD report information.
Feature Description
ICDG Management–Group The ACD supervisor can monitor the condition of up to 4 ICD
Monitor groups on the same screen by specifying monitoring
conditions. Furthermore, the ACD Supervisor can manage up
to 64 ICD groups, and can monitor up to 16 ICD groups.
ICDG Management–ACD The ACD supervisor can analyse the monitoring result as
Report follows:
• Filter: Filters the monitoring result according to Group,
Agent and Call.
• View Report: Displays the filtered monitoring result in a
format that is easy to analyse (graph, file export, or print).

The screen of a user set as an ACD supervisor is displayed as follows. There are 2 modes – Simple Mode
and Standard Mode – and the displayed information is different depending on the mode.
Group Monitor can be displayed in full screen mode. To enable this feature, match the resolution of your
display to the Full Screen Display (pixels) resolution setting. For details, refer to "8.7.1 Supervisors—ICDG
Management—Group Monitor" in the PC Programming Manual.

[Group Monitor Example]

Standard Mode

62 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

Simple Mode

[ACD Report]
An ACD report can be made with the following items included.

Group
Item Description
Incoming Calls
Total The number of incoming calls received by the target ICD group.
Answered The number of incoming calls answered by the target ICD group.
Lost The number of incoming calls to the target ICD group cancelled by
the caller.
Overflow The number of overflowed incoming calls to the target ICD group.
Talk Time
Total The total talking time of answered calls for the target ICD group.
(HH:MM:SS (D))*
Average The average talking time of answered calls for the target ICD
group. (HH:MM:SS)

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2.2.2 Incoming Call Distribution Group Features

Item Description
Max. The longest talking time of answered calls for the target ICD group.
(HH:MM:SS)
Wait Time
Wait Time (Answered)
Total The total waiting time of answered calls for the target ICD group.
(HH:MM:SS (D))*
Average The average waiting time of answered calls for the target ICD
group. (HH:MM:SS)
Max. The longest waiting time of answered calls for the target ICD group.
(HH:MM:SS)
Wait Time (Lost)
Total The total waiting time of cancelled calls for the target ICD group.
(HH:MM:SS (D))*
Average The average waiting time of cancelled calls for the target ICD
group. (HH:MM:SS)
Max. The longest waiting time of cancelled calls for the target ICD group.
(HH:MM:SS)
Max. Waiting Calls The maximum number of calls waiting in the queue of the target
ICD group.

* "D" indicates the number of days (D=1–366). If the number of days is 0, (D) is not displayed.

Agent
Item Description
Total Answer
Total Answer The number of calls that the target agent answers.
Talk Time
Total The total talking time for the target agent. (HH:MM:SS (D))*
Average The average talking time for the target agent. (HH:MM:SS)
Max. The longest talking time for the target agent. (HH:MM:SS)
Login Time The total login time for the target agent. (HH:MM:SS (D))*
Not-ready Time The total not ready time for the target agent. (HH:MM:SS (D))*
Wrap-up Time The total wrap-up time for the target agent. (HH:MM:SS (D))*

* "D" indicates the number of days (D=1–366). If the number of days is 0, (D) is not displayed.

Call
Item Description
ACD Report - Call Report
Start Date The start date of the call.
Start Time The start time of the call. (HH:MM:SS)

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 2.2.2 Incoming Call Distribution Group Features

Item Description
End Date The end date of the call.
End Time The end time of the call. (HH:MM:SS)
Result The processing result. (Answered/Abandoned/Overflowed/
Unanswered)
ICDG The incoming ICD Group number.
Incoming Agent The extension number of the agent that received the call.
(Displayed only when the ICD group distribution method is set to
Uniform Call Distribution or Priority Hunting.)
Answering Agent The answering member. (Extension Name/Extension Number)
Talk Time The talking time. (HH:MM:SS)
Wait Time The waiting time. (HH:MM:SS)
Trunk The incoming trunk group number.
Caller ID/CLIP The caller’s number.

Reports can be output as a graph, exported as a file or printed out, after filtering results as necessary.
There are 2 ways to export reports.
• Manual export
Export the report manually as a CSV file. The exported file can be saved to a NAS or a PC that can
access Web Maintenance Console.
• Scheduled export
The report is exported automatically as a CSV file at specified times. Up to 10 schedules can be
registered and each one can be enabled or disabled individually. The file is exported according to the
selected report profile filter settings and the exported file can be saved to a USB memory device
connected to the PBX or a NAS. If the file exceeds 10000 lines, it is split into separate files of up to 10000
lines.
Export-related data (date and time, description, export type and setting values, report profile and
completion status) is recorded and can be viewed in the Export History.
For details about ACD report items, refer to the PC Programming Manual.

Feature Manual 65
2.2.2 Incoming Call Distribution Group Features

[Graph Example]

Conditions
CAUTION
A maximum of 2 months (the current month and the previous month) of ACD reports are saved. ACD
reports saved automatically in the PBX that have exceeded the save period are deleted automatically. Any
ACD report data that must be retained should be backed up periodically.
• KX-NSXF002 (Call Centre Feature Enhancement) or KX-NSXF022 (Built-in ACD Report) is required to
use this feature.
• Up to 32 users can be set as an ACD supervisor through system programming.
– A maximum of 12 months (366 days) of ACD reports are saved.
However, if Custom Period is selected in Select Period, Start Date can be set to a date within 1 year
prior to the current date (e.g., if today is 22 December 2016, the oldest allowable date for Start Date is
22 December 2015), and End Date can be set up to the current date.
– Old ACD reports are deleted if they have exceeded the save period, or if the maximum number of
saved ACD reports has been exceeded.
• The number of calls that can have monitoring saved up to 1,200,000 calls.

Feature Manual References

2.23.2 Room Status Control
2.23.3 Call Billing for Guest Room

PC Programming Manual References

8.7 Supervisors—ICDG Management
11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous
11.5.4 PBX Configuration–[3-5-4] Group—Incoming Call Distribution Group—ACD Supervisor

66 Feature Manual
 2.2.2 Incoming Call Distribution Group Features

Operating Manual References

1.12.1 Using the Telephones in a Hotel-type Environment (Hospitality Features)
4.1.3 Supervisory Monitor (ACD) Control

Feature Manual 67
2.3 Call Forwarding (FWD)/Do Not Disturb (DND) Features

2.3 Call Forwarding (FWD)/Do Not Disturb (DND)

Features
2.3.1 Call Forwarding (FWD)/Do Not Disturb (DND)—SUMMARY
Description
When an extension user cannot answer calls (e.g., he is busy, or not at his desk), it is possible to forward or
refuse calls using the following features:
1. Call Forwarding (FWD)
2. Do Not Disturb (DND)

1. FWD
Extensions and incoming call distribution groups can forward their incoming calls to preset destinations.
(→ 2.3.2 Call Forwarding (FWD))
2. DND
Callers to an extension will hear a tone to inform them that the extension user is not available. (→
2.3.3 Do Not Disturb (DND))
3. FWD/DND Button, Group FWD Button
The FWD/DND fixed button, or a customised flexible button, can display the FWD/DND setting status of
the extension. (→ 2.3.4 FWD/DND Button, Group FWD Button)

Conditions
• FWD and DND are set for intercom calls (including doorphone calls), and trunk calls (including a call from
an extension that placed a trunk call on a consultation hold) separately.

2.3.2 Call Forwarding (FWD)

Description
Extensions and incoming call distribution groups can forward their calls to preset destinations. There are
several different types of forwarding, and the circumstances under which the calls are forwarded for each
type differ as follows:
Type Circumstance
All Calls Any time
Follow Me:
When an extension user fails to set this feature before leaving the
desk, this feature can be set from the destination extension.
Busy When the extension user’s line is busy.
No Answer When the extension user does not answer within a
preprogrammed time.
Busy/No Answer When the extension user’s line is busy or the user does not
answer within a preprogrammed time.

Depending on the type of incoming intercom or trunk calls, it is possible to set a different destination for
each.

68 Feature Manual
 2.3.2 Call Forwarding (FWD)

Intercom Calls Extension Forwards to

to Extension Another Extension

Trunk Calls to Forwards to

Extension Outside Party

Available Forwarding Type: All Calls

Busy
No Answer
Busy/No Answer

Incoming Call Distribution Group

Intercom Calls to Forwards to
Incoming Call Another Extension
Distribution Group

Trunk Calls to Forwards to

Incoming Call Outside Party
Distribution Group

Available Forwarding Type: All Calls

[Available Destinations]
Condition for Original Extension/
Destination Availability
Incoming Call Distribution Group
Wired Extension (PT/SLT/SIP Extension/ISDN ü Only available when FWD to
Extension/T1-OPX) extension is allowed through COS
programming.*1
PS ü
Incoming Call Distribution Group ü
PS Ring Group ü –
UM Group ü –
Fax Unit ü –
External Pager (TAFAS) ü –
DISA ü Only available for incoming trunk
calls. Incoming intercom and
doorphone calls cannot be
forwarded to a DISA floating
extension number.
Idle Line Access no. + Phone no. ü Only available when FWD to trunk
is allowed through COS
Trunk Group Access no. + Trunk Group no. + ü
programming.
Phone no.
Other PBX Extension (TIE with no PBX Code) ü –

Feature Manual 69
2.3.2 Call Forwarding (FWD)

Condition for Original Extension/

Destination Availability
Incoming Call Distribution Group
Other PBX Extension (TIE with PBX Code) ü Only available when FWD to trunk
is allowed through COS
programming.

*1 If an extension user is not permitted by COS to call a certain extension (→ 2.1.2.2 Internal Call Block), the FWD feature will not
function if that extension is set as the forwarding destination.

Parallel Ringing When Forwarding to Trunk

When an unanswered call is forwarded to an outside line, such as a cellular phone, the forwarding
extension’s phone will continue ringing until the forwarded call is answered at either phone. This feature can
be enabled for each extension through system programming. Even when the outside destination answers
the call, DSS buttons for the forwarding extension are displayed as busy. The Intercept Routing feature is
available for when parallel ringing is not answered.

Conditions
[General]
• FWD for Trunk Calls/Intercom Calls
The extension user can set the FWD feature for trunk calls, for intercom calls, or for both.
• FWD from Incoming Call Distribution Group (Group FWD)
COS programming determines the incoming call distribution groups that can use this feature.
• FWD to Trunk
COS programming determines the extensions or incoming call distribution groups that can forward calls
externally. The original extension’s TRS/Barring and ARS still apply to the forwarded call.
• Trunk Call Duration
The duration of a trunk call can be restricted by a system timer. Trunk call duration is assigned separately
for calls between an extension user and an outside party, and calls between two outside parties.
If the timer expires, the line will be disconnected. (→ 2.11.8 Trunk Call Limitation)
• Multiple FWD
Calls can be forwarded up to four times. The following forwarding features are counted as Multiple FWD:
– FWD—Busy or Busy/No Answer (when the destination extension is busy), or All Calls
– Idle Extension Hunting—Overflow
– Intercept Routing—Busy/DND/No answer (when the destination extension is busy or in DND or No
answer mode)*1
– Incoming Call Distribution Group—Overflow
*1 Intercept Routing features can be applied to the original destination (refer to 2.1.1.6 Intercept Routing).

Incoming
call 1 2 3 4 5
A B C D E F
Original
destination

In the above illustration, forwarding stops at extension E. However, forwarding can go farther in the
following cases:
– If a destination extension rings, and then the call is redirected to the forward destination by the FWD—
No Answer or Busy/No Answer feature.

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 2.3.2 Call Forwarding (FWD)

– If a call waits in a queue of an incoming call distribution group, and then the call is redirected to the
overflow destination by the Queuing Time Table. (→ 2.2.2.3 Queuing Feature)

In the above cases, the forwarding counter resets to zero, and the call can be forwarded up to four times
again from the destination extension described above.

Incoming
call 1 2 3 1 2
A B C D E F
Original
FWD—No Answer
destination

• Boss & Secretary feature

It is possible to call the original extension from the destination extension regardless of the forward setting.
FWD—All Calls
Incoming
call

Call or
transfer a call
Boss Secretary
(Original) (FWD destination)

• Remote Forward setting

A flexible button customised as a Group FWD button can be used to set or unset the FWD feature for an
extension by specifying its extension number.
To allow this feature, the extension subject to this operation must specify the Remote FWD COS setting.
The authorised extension with the specified Remote FWD COS can then assign a Group FWD button to
perform remote forwarding. The forwarding status of the target extension can be checked via the LED of
the button.

Note
Pressing this button only allows the forwarding feature to be enabled or disabled. The forwarding
destination should be set at the extension subject to this operation. Also, DND cannot be enabled or
disabled.

[Example]
– Setting Group FWD from "Secretary"
extension to "Boss" extension Forwarding
If the "Secretary" (Ext. 102) presses the DSS button enabled
assigned Group FWD button, forwarding (FWD 101)
for the "Boss" (Ext. 101) will be turned on
(the LED will turn red). Secretary Boss
Extn. 102 Extn. 101

– Cancelling Group FWD from "Secretary"

extension to "Boss" extension Forwarding
If the "Secretary" (Ext. 102) presses the DSS button disabled
assigned Group FWD button again, (FWD 101)
forwarding for the "Boss" (Ext. 101) will be
turned off (the LED will turn off). Secretary Boss
Extn. 102 Extn. 101

Feature Manual 71
2.3.2 Call Forwarding (FWD)

• Message Waiting
While calls are forwarded, Message Waiting information is not forwarded. The Message button light turns
on at the originally called extension. (→ 2.20.1 Message Waiting)
• Idle Extension Hunting
Idle Extension Hunting applies to calls forwarded to a busy extension in an idle extension hunting group.
• Second Line LCD Display Information for ICD Group Redirected Call
If an incoming call distribution receives a redirected call, the second line of the PT receiving the call can
be set to display either of the following information, through system programming.
– The extension number and name of the extension or ICD Group that originally received the call.
– The floating extension number and name of the ICD Group currently receiving the call.
[All Calls and Busy]
• If the forward destination is not available to answer a call, this feature is cancelled and the original
destination will ring for the following type of call:
– Doorphone call
– Trunk calls via the ELCOT/CLCOT/LCOT or T1 (LCOT) cards
[No Answer and Busy/No Answer]
• No Answer Time
The number of rings before the call is forwarded is programmable for each extension.
[Follow Me]
• This feature is only available when the original extension has set "Remote Operation by Other
Extension" to "Allow" through COS programming.
[Parallel Ringing When Forwarding to Trunk]
• KX-NSUN001, KX-NSUN010, KX-NSUN050, KX-NSUN100, KX-NSUN500, KX-NSUM001,
KX-NSUM010, KX-NSUM050, KX-NSUM100, KX-NSUM500, KX-NSUA001, KX-NSUA010,
KX-NSUA050, KX-NSUA100 or KX-NSUA500 must be installed to the Activation Key in order to use this
feature. One activation key is required for each extension that will use this feature.
• Even though DSS buttons for the forwarding extension indicate that the extension is busy, it can still
receive calls. If another call is received, FWD—No Answer will operate as normal.
• It is possible to set the forwarding extension to busy through system programming.
• When the forwarding extension is a public device name
– The No Answer Time setting is ignored: All calls are forwarded immediately, even if No Answer is
specified as the forwarding method.
– If the forward destination is unavailable, the forwarding extension is treated as busy. (In this case, FWD
—Busy is ignored even if it is enabled.) However, if the original caller is on an analogue trunk, to which
busy signals/tones cannot be sent, the PBX will continue to try to connect to the forward destination
until a connection is established or the original caller hangs up.
– It is possible to set the forwarding extension public device name to busy through ICD Group
programming.
• If an extension goes on-hook while transferring a call to an extension ringing in parallel with a trunk, the
trunk will stop ringing for a moment, then begin ringing again.
• This feature is not available when the outside destination is an ELCOT/CLCOT/LCOT trunk without
reverse circuit detection (→ 2.5.4.5 Reverse Circuit). It is also not available for channels of a T1 trunk set
to LCOT type.
• This feature may not be available depending on the specifications of the telephone network.

72 Feature Manual
 2.3.3 Do Not Disturb (DND)

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—FWD No Answer Timer
Set
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings
→ CO & SMDR—Call Forward to CO
→ Manager—Group Forward Set
→ Optional Device & Other Extensions—Remote Operation by Other Extension
11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Main
→ CO-CO Duration Time (*60s)
→ Extension-CO Duration Time (*60s)
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Miscellaneous—Status of FWD for Virtual PS
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Group Log /
Group FWD
11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous
→ Options—2nd line LCD display information for ICD Group redirected call
8.1.1 Users—User Container—Add User/Edit User—Telephony Feature
29.1.8 My Portal—Incoming Call Control

Feature Manual References

2.1.1.6 Intercept Routing
2.2.1 Idle Extension Hunting
2.2.2.6 Overflow Feature
5.1.1 Class of Service (COS)

Operating Manual References

1.6.1 Forwarding Calls

2.3.3 Do Not Disturb (DND)

Description
An extension user can make use of the DND feature. If this feature is set, calls will not arrive at the
extension, but will arrive at another extension using the Idle Extension Hunting feature (→ 2.2.1 Idle
Extension Hunting) or the Intercept Routing—Busy/DND feature (→ 2.1.1.6 Intercept Routing). When a
destination cannot be found, the calling extension will hear the DND tone, while the calling outside party will
hear a busy tone.

Conditions
• DND for Trunk Calls/Intercom Calls
The DND feature can be set for trunk calls, for intercom calls, or for both of them by the extension user.
• DSS button in DND Mode
The DSS button light will turn red if the assigned extension has set DND.
• DND Override
An extension in DND mode can be called by other extension users who are allowed to override DND in
their COS.
• Paging DND
It is programmable whether the PBX pages extensions in DND mode through system programming. (→
2.17.1 Paging)
• Intercept Routing—Busy/DND

Feature Manual 73
2.3.4 FWD/DND Button, Group FWD Button

If a call arrives at an extension in DND mode, the call can be redirected to a preprogrammed destination
by the Intercept Routing—Busy/DND feature.
• Idle Extension Hunting
While searching for an idle extension within an idle extension hunting group, any extension that has DND
set will be skipped. The call will go to the next extension in the group, not the Intercept Routing—
Busy/DND destination.
• If (1) a trunk call via the ELCOT/CLCOT/LCOT or T1 (LCOT) card arrives at an extension in DND mode
and (2) the Intercept Routing—Busy/DND destination is not available and (3) there is no available
extension in the idle extension hunting group, then the original extension in DND mode will ring.
• Calls from a doorphone arrive at the extension even when the extension is in DND mode.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Tone Length—
Busy Tone / DND Tone
10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature
→ BSS / OHCA / Whisper OHCA / DND Override
→ BSS / OHCA / Whisper OHCA / DND Override-2
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Assistant—DND Override
8.1.1 Users—User Container—Add User/Edit User—Telephony Feature
29.1.8 My Portal—Incoming Call Control

Feature Manual References

5.1.1 Class of Service (COS)

Operating Manual References

1.2.4 When the Dialled Line is Busy or There is No Answer
1.9.2 Refusing Incoming Calls (Do Not Disturb [DND])

2.3.4 FWD/DND Button, Group FWD Button

Description
The FWD/DND fixed button, or a customised flexible button, can display the FWD/DND setting status of the
extension. Using this button, the FWD status and DND status of the extension can be temporarily set or
cancelled without clearing FWD destination settings.

FWD/DND Button Types

Multiple types of FWD/DND buttons can be customised on an extension.
Type Description
FWD/DND for FWD/DND—Internal Works for incoming intercom calls
Extension
FWD/DND—External Works for incoming trunk calls
FWD/DND—Both Works for all incoming calls

[Button Status]
The FWD/DND button shows the current status as follows:
Light Pattern Status (default)
Red on FWD on

74 Feature Manual
 2.3.4 FWD/DND Button, Group FWD Button

Light Pattern Status (default)

Slow red flashing DND on
Off FWD/DND off

The functions assigned to the "on" and "flashing" patterns can be changed through system programming.

Group FWD Button Types

The FWD feature for the incoming call distribution group can be customised on a flexible button. Multiple
types of Group FWD buttons can be customised on an extension.
Type Description
FWD for Incoming Call Group FWD— Works for incoming intercom calls
Distribution Group Internal
Group FWD— Works for incoming trunk calls
External
Group FWD—Both Works for all incoming calls

[Button Status]
The Group FWD button shows the current status as follows:
Light Pattern Status (default)
Red on FWD on
Off FWD off

FWD/DND Setting by Fixed FWD/DND button

Pressing the fixed FWD/DND button in idle status allows the extension user to set the following items for
FWD/DND:
• FWD/DND for trunk calls
The FWD/DND status for trunk calls to that extension can be switched temporarily without clearing the
FWD destination. During setting, the LED of the button shows the current trunk call FWD/DND status.
The forwarding type and destination for trunk calls can also be set.
• FWD/DND for intercom calls
The FWD/DND status for intercom calls to that extension can be switched temporarily without clearing the
FWD destination. During setting, the LED of the button shows the current intercom call FWD/DND status.
The forwarding type and destination for intercom calls can also be set.
• FWD—No Answer timer
The length of time until unanswered calls are forwarded can be modified. This setting is applied to both
intercom and trunk call forwarding.
• FWD for Public device name
If the extension is the first registered extension in an Incoming Call Distribution Group, the extension user
can set the FWD destination and forwarding status (on/off) for up to 4 public device names preregistered
to the group.
These settings are only available when FWD/DND buttons are set through system programming to
FWD/DND Setting mode.

Feature Manual 75
2.3.4 FWD/DND Button, Group FWD Button

Conditions
• When FWD/DND buttons are set to FWD/DND Cycle Switch mode, pressing the FWD/DND button cycles
the FWD/DND setting.
In this mode, when intercom calls are set to be handled differently from trunk calls (forward destination,
DND on/off):
a. in idle mode, the light patterns of the FWD/DND—Both button (including FWD/DND button [fixed
button]) and the Group FWD—Both button will indicate the setting for either trunk calls or intercom
calls, but not both.
b. the FWD and DND icons on a PS display reflect the settings for trunk calls only.
c. pressing the FWD/DND—Both button (including FWD/DND button [fixed button]) or the Group FWD
—Both button will be ignored.
• When both the FWD and DND features are assigned simultaneously, pressing the button changes the
settings as follows:
FWD DND Off

• A FWD/DND button customised on a flexible button is always in FWD/DND Cycle Switch mode, and the
mode cannot be changed.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ FWD/DND Set / Cancel: Call from CO & Extension
→ FWD/DND Set / Cancel: Call from CO
→ FWD/DND Set / Cancel: Call from Extension
→ Group FWD Set / Cancel: Call from CO & Extension
→ Group FWD Set / Cancel: Call from CO
→ Group FWD Set / Cancel: Call from Extension
10.9 PBX Configuration—[2-9] System—System Options—Option 1
→ PT Fwd / DND—Fwd LED
→ PT Fwd / DND—DND LED
→ PT Fwd / DND—Fwd/DND key mode when Idle
8.1.1 Users—User Container—Add User/Edit User—Telephony Feature
29.1.8 My Portal—Incoming Call Control
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Extension Number (for Group Fwd (Both))
→ Extension Number (for Group Fwd (External))
→ Extension Number (for Group Fwd (Internal))
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Extension Number (for Group Fwd (Both))
→ Extension Number (for Group Fwd (External))
→ Extension Number (for Group Fwd (Internal))

Feature Manual References

2.21.1 Fixed Buttons
2.21.2 Flexible Buttons

Operating Manual References

3.1.2 Settings on the Programming Mode

76 Feature Manual
 2.4 Answering Features

2.4 Answering Features

2.4.1 Answering Features—SUMMARY
Description
An extension user can answer incoming calls by the following methods:
Destination Feature Description & Reference
At the own Line Preference— A user can select the line seized when going off-hook.
extension (PT Incoming
only) → 2.4.2 Line Preference—Incoming
Direct One-touch A user can answer an incoming call simply by pressing
Answering the flashing button.
Hands-free A user can receive a call automatically and establish a
Answerback hands-free conversation.

→ 2.4.4 Hands-free Answerback

At another Call Pickup— A user can pick up a specified extension’s call or a call
extension Directed/Group in a specified call pickup group.

→ 2.4.3 Call Pickup

2.4.2 Line Preference—Incoming

Description
A PT user can select the method used to answer incoming calls from the following three line preferences:
Each of these line preferences can be assigned on each extension through personal programming
(Preferred Line Assignment—Incoming).
Type Description
No Line Select a line by pressing the desired Line Access button to
answer an incoming call after you go off-hook.
PDN Answers a call arriving at a Primary Directory Number (PDN)
button simply by going off-hook. This works even when multiple
calls are received simultaneously.
(→ 2.9.1 Primary Directory Number (PDN)/Secondary Directory
Number (SDN) Extension)
Prime Line Answer a call arriving at a Flexible CO or ICD Group button (on
which the "Prime Line" is assigned) simply by going off-hook.
This works even when multiple calls are received
simultaneously.
Ringing Line (default) Answer the longest ringing call at one’s telephone simply by
going off-hook when multiple calls arrive.

Conditions
[Prime Line]
• The priority of the incoming call is as follows:

Feature Manual 77
2.4.3 Call Pickup

1. The call arriving at a button on which the "Prime Line" is assigned.

2. The call arriving at the INTERCOM button.

PC Programming Manual References

8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 4—Incoming Preferred
Line

Operating Manual References

3.1.2 Settings on the Programming Mode

2.4.3 Call Pickup

Description
An extension user can answer a call ringing at any other extension.
The following types are available:
Type Picking up Call Type
Directed A specified extension’s call.
Group A call within a specified call pickup group.

Caller Information Display before Call Pickup

PT/PS users such as colleagues can confirm the caller information of a call to another extension using the
DSS button before picking up the call.
Pressing the corresponding DSS button will call the extension receiving the call rather than picking up the
call. While hearing the busy tone, the user can check the caller’s information on the LCD. The call can be
picked up by pressing the same DSS button again.

[Example]

Telephone Company

PBX
Outside Party
(01-2345-6789)

Caller’s name
Caller’s number
Extension Status

DSS button
Extn. 101 Extn. 102 (Extn. 101)
(Ringing) (Colleague)

Call Pickup Group Monitor

Through call pickup group and extension settings, users of PTs (with a 3-line LCD or larger) such as
colleagues can view the caller information of a call to another extension which belongs to the same call

78 Feature Manual
 2.4.3 Call Pickup

pickup group. The user can view the caller’s information on the LCD while their PT is idle. The call can be
picked up by pressing the ANSWER button.

Note
While the caller information is displayed on the LCD, going off-hook will not answer the call.

[Example]

Telephone Company

PBX
Outside Party
(01-2345-6789)
Call Pickup Group

Extn. 101
(Ringing)

Caller’s name Caller’s name Caller’s name

Caller’s number Caller’s number Caller’s number
Called Extn. Information Called Extn. Information Called Extn. Information
Extension Number Extension Number Extension Number
Extension Name Extension Name Extension Name
Absent Message Absent Message Absent Message

Extn. 102 Extn. 103 Extn. 104

(Idle) (Idle) (Idle)

Call Pickup Deny:

Preventing other extensions from picking up calls ringing at your extension is also possible.

Conditions
• Call Pickup applies to:
Intercom, trunk, and doorphone calls
• Internal Call Block
An extension that is restricted by COS from calling certain extensions (→ 2.1.2.2 Internal Call Block) also
cannot pick up any calls ringing at those extensions.

[Directed Call Pickup]

• A user can also pick up a call to a specified extension by pressing the corresponding DSS button. This
feature is only available when (1) the user’s extension is allowed to use this feature through COS
programming, (2) DSS buttons for extensions or incoming call distribution (ICD) groups have this feature
enabled through system programming, and (3) the light pattern of DSS buttons for incoming calls to
extensions or ICD groups is set to "On or Flash" through system programming.

Feature Manual 79
2.4.3 Call Pickup

The light pattern of a DSS button for an incoming call to an extension or incoming call distribution group
can be programmed through system programming. Call Pickup is available only when the DSS button is
flashing red.

[Group Call Pickup]

• A specified number of call pickup groups can be created, each of which consist of extension user groups.
One extension user group can belong to several call pickup groups. (→ 5.1.2 Group)

[Example]

Call Pickup Group 1 Call Pickup Group 2 Call Pickup Group 3

Extension Extension Extension Extension

User Group 1 User Group 2 User Group 3 User Group 4

Extn. 100 Extn. 101 Extn. 102 Extn. 103 Extn. 104 Extn. 105 Extn. 106 Extn. 107

[Caller Information Display before Call Pickup]

• This feature is available only under the following conditions:
– The "Caller Information Display before Call Pick-up" setting is enabled.
– Directed Call Pickup is enabled for the extension picking up the call.
– The assigned button is a DSS button.

[Call Pickup Group Monitor]

• This feature is available only under the following conditions:
– The Call Pickup Group Monitor Feature setting is enabled in call pickup group settings.
– The Call Pickup Group Monitoring setting is set to "Display & Tone1–3" or "Display only" in
extension settings.
– The LCD display time is set in Call Pickup Group Monitor—LCD Display Duration (s) in timer
settings.
• When a call arrives at a call pickup group, the following information is displayed on idle PTs.
– Name associated with the caller’s number
– The caller’s number
– Name and extension number of the called extension
– Absent Message of the called extension
• If an extension belongs to several call pickup groups, its user can monitor calls for all call pickup groups.
However, the lowest numbered call pickup group is prioritised.
• Up to 256 extensions can be monitored (refer to Call Pickup Group Monitoring in extension settings).
• The caller information will stop being displayed in the following cases:
– When the PT becomes active from an idle state.
– When a button other than the ANSWER button is pressed. (The display will change according to the
performed operation.)

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 2.4.3 Call Pickup

• For one-numbered extensions, the main extension and sub extension operate separately according to
their individual call pickup group monitoring settings.
• Telephones that support this feature are as follows:
– IP-PTs*1
– DPTs*1
*1 Except models with a single line LCD.

• Extensions that are subject to restrictions through COS (e.g., restrictions on internal or trunk calls) cannot
display monitoring information.
• This feature is not affected by whether a DSS button is set to pick up a call at a specified extension.
• Telephones in the following states cannot use this feature:
– When in wrap-up mode
– When waiting on the Automatic Redial feature
– When displaying CTI information on the LCD
– When receiving a paging call
• Caller information will be displayed while listening to background music.
• The LCD corresponds with the call waiting tones 1, 2, and 3 as follows.

Monitor cycle (15 s)

Tone 1

Tone 2

Tone 3

Monitor display time

(Default: 5 s)

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—Call Pickup Group Monitor—
LCD Display Duration (s)
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Group Call Pickup
→ Directed Call Pickup
→ Call Pickup Deny Set / Cancel
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Assistant—Call Pickup by
DSS
10.9 PBX Configuration—[2-9] System—System Options—Option 4
→ DSS Key—DSS key mode for Incoming Call
→ DSS Key—Call Pick-up by DSS key for Direct Incoming Call
→ DSS Key—Call Pick-up by DSS key for ICD Group Call
→ DSS Key—Caller Information Display before Call Pick-up
11.3 PBX Configuration—[3-3] Group—Call Pickup Group

Feature Manual 81
2.4.4 Hands-free Answerback

11.3.1 PBX Configuration—[3-3-1] Group—Call Pickup Group—Main Setting—Main—Call Pickup Group

Monitor Feature
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 3
→ Call Pickup Deny
→ Call Pickup Group Monitoring

Feature Manual References

5.1.1 Class of Service (COS)
6.1 Capacity of System Resources

Operating Manual References

1.3.3 Answering a Call Ringing at Another Telephone (Call Pickup)

2.4.4 Hands-free Answerback

Description
A PT user with a speakerphone can talk to a caller without lifting the handset. If the user receives a call
while in Hands-free Answerback mode, a hands-free conversation is established using one of the following
methods:
Type Answering Method
Intercom Call Established immediately after a beep tone at the called
extension and the caller hears a confirmation tone.
Trunk Call Established after a specified number of rings, a called
extension hears a beep tone.

Class of Service (COS) with Hands-free Answerback

Hands-free Answerback can be enabled only for specific callers based on the caller’s COS setting. For
example, a nurse could call a room and the called extension will answer automatically so that the nurse can
check on the occupant of the room. However, other intercom calls would ring even if Hands-free
Answerback was set. Walking COS can also be used with this feature.

Conditions
• Hands-free Answerback applies to:
Intercom calls and trunk calls, including calls directed to an incoming call distribution group in UCD or
Priority Hunting distribution method. (→ 2.2.2.2 Group Call Distribution)
• Hands-free Answerback for Trunk Calls
System programming is required to use this feature.
• Hands-free Answerback for Calls From an Extension That Placed a Trunk Call on Consultation
Hold
Calls from an extension that placed a trunk call on Consultation Hold can be treated by this feature as
either intercom calls or trunk calls, depending on system programming. If treated as intercom calls, the
call will be established immediately.
When transferring a call from an analogue trunk, users are strongly recommended to perform a screened
transfer, so that the outside caller is not automatically connected to an extension using Hands-free
Answerback when the extension user is absent.
Extensions that perform unscreened transfers often, such as operators, should have the Class of Service
(COS) with Hands-free Answerback feature disabled. Otherwise, transferred outside calls may be
automatically connected by Hands-free Answerback, even when the transfer destination is absent.

82 Feature Manual
 2.4.4 Hands-free Answerback

• Secret Monitor
The beep tone that the called party hears before answering can be eliminated through system
programming.
• Alternate Receiving/Calling Mode (Ring/Voice) Override
Hands-free Answerback overrides the Alternate Receiving mode preset on the telephone and the
Alternate Calling mode from the caller.
• Hands-free Answerback with Headset
The Hands-free Answerback feature can also be used with a headset.
• The Hands-free Answerback feature cannot be used with incoming calls from a UM/VM extension (the
same as with voice calling).

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Tone Length—
Reorder Tone for PT Hands-free
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Assistant—Automatic
Answer (Caller)
10.9 PBX Configuration—[2-9] System—System Options
→ Option 1—PT Operation—Automatic Answer for Call from CO after
→ Option 3—Confirmation Tone—Tone 2 : Paged / Automatic Answer
→ Option 4—Transfer—Automatic Answer for Transferred Call
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings
→ Option 5—Automatic Answer for CO Call
→ Option 6—Forced Automatic Answer

Feature Manual References

2.5.3 Intercom Call
5.1.1 Class of Service (COS)

Operating Manual References

1.3.2 Answering Hands-free (Hands-free Answerback)

Feature Manual 83
2.5 Making Call Features

2.5 Making Call Features

2.5.1 Predialling
Description
A display PT user can check and correct the entered number before it is dialled, while on-hook. The call will
be initiated after going off-hook.

Conditions
• Storing the Predialled Number in the Personal Speed Dialling
The predialled number can be stored in the Personal Speed Dialling by pressing the AUTO DIAL/STORE
button. (→ 2.6.4 Speed Dialling—Personal/System) In this case, the extension will enter into the personal
programming mode automatically so that a name can be assigned for the stored number.

Operating Manual References

3.1.2 Settings on the Programming Mode

2.5.2 Automatic Extension Release

Description
After going off-hook, if an extension user fails to dial any digits within a preprogrammed time period, the user
will hear a reorder tone. This operation applies to intercom calls only. This feature is also known as
Automatic Station Release.

Conditions
• A PT/PS user hears a reorder tone for a preprogrammed time period, and then the PT/PS returns to idle
status automatically. However, an SLT user hears a reorder tone until he goes on-hook.
• This feature works in one of the following cases:
When making an intercom call
a. If the first digit is not dialled within a preprogrammed time period.
b. After a digit is dialled, if subsequent digits are not dialled within a preprogrammed time period.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone
→ Dial—Extension First Digit
→ Dial—Extension Inter-digit
→ Tone Length—Reorder Tone for PT Handset
→ Tone Length—Reorder Tone for PT Hands-free

2.5.3 Intercom Call

Description
An extension user can call another extension user.

Conditions
• Extension Number/Name Assignment
Extension numbers and names are assigned to all extensions. The assigned number and name are
shown on display PTs during intercom calls.

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 2.5.3 Intercom Call

• DSS Button
It is possible to access another extension with one touch by pressing the corresponding Direct Station
Selection (DSS) button. A flexible button can be customised as a DSS button.
• Call Directory—Extension Dialling
A display PT user can make a call by selecting one of the stored names on the display.
• Limiting the display by tenant—Call Directory
For Call Directory, an extension can reference the data for all tenants or for each tenant the extension is
member of, depending on system programming. In "Each Tenant" mode, Call Directory is displayed on
display PTs as follows:
Only information about extensions that belong to the tenant is displayed.
• Alternate Receiving—Ring/Voice
A PT user can select to receive intercom calls by ring tone or by voice, through personal programming
(Alternate Receiving—Ring/Voice). If a user selects voice-calling, the calling party talks to the user
immediately after a confirmation tone. Denying voice-calling can also be selected.
• Alternate Calling—Ring/Voice
A caller can change the called party’s preset call receiving method (ring tone or voice) temporarily. By
doing so, ring-calling is switched to voice-calling, or vice versa, at the called party. The called party may
deny voice-calling.
• PDN/SDN
It is not possible to temporarily change the called party’s preset call receiving method when making a call
using a Primary Directory Number (PDN) button or Secondary Directory Number (SDN) button (→
2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension).
• Tone after Dialling
After dialling an extension number, a user will hear one of the following:
Type Description
Ringback Tone Indicates the called party is being called.
Confirmation Tone Indicates the called party has set voice-calling.
Busy Tone Indicates the called party is busy.
DND Tone Indicates the called party has set DND.

PC Programming Manual References

10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature—Alternate Calling -
Ring / Voice
8.1.1 Users—User Container—Add User/Edit User
→ Extension Number
→ Display Name
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 3—Intercom Call by
Voice

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

1.2.1 Basic Calling
1.2.6 Alternating the Calling Method (Alternate Calling—Ring/Voice)
1.14.2 Using the Directories
3.1.2 Settings on the Programming Mode

Feature Manual 85
2.5.4 Trunk Call Features

2.5.4 Trunk Call Features

2.5.4.1 Trunk Call Features—SUMMARY
Description
An extension user can use the following features when making a trunk call:
Feature Description & Reference
Emergency Call A user can dial the preprogrammed emergency numbers
regardless of the restrictions imposed on the extension.

→ 2.5.4.2 Emergency Call

Account Code Entry A user can enter an account code to identify outgoing calls for
accounting and billing purposes.

→ 2.5.4.3 Account Code Entry

Pulse to Tone Conversion A user can temporarily switch from Pulse mode to DTMF mode
to access special services.

→ 2.5.4.4 Dial Type Selection

Pause Insertion A user can insert a preprogrammed Pause time into a dialling
number by pressing the PAUSE button, or it is automatically
inserted between the user-dialled code (e.g., Host PBX Access
code or Special Carrier Access code) and the following digits.

→ 2.5.4.7 Pause Insertion

→ 2.5.4.8 Host PBX Access Code (Access Code to the
Telephone Company from a Host PBX)
→ 2.5.4.9 Special Carrier Access Code

2.5.4.2 Emergency Call

Description
An extension user can dial the preprogrammed emergency numbers after seizing a trunk regardless of the
restrictions imposed on the extension.
In addition, Extension can make a preprogrammed emergency numbers without seizing a trunk (USA Only).

Conditions
• A specified number of emergency numbers can be stored (some may have default values).
• Emergency numbers may be called even when:
– in Account Code—Forced mode (→ 2.5.4.3 Account Code Entry)
– in any TRS/Barring levels (→ 2.7.1 Toll Restriction (TRS)/Call Barring (Barring))
– after the preprogrammed call charge limit is reached (→ 2.7.2 Budget Management)
– in Extension Dial Lock (→ 2.7.3 Extension Dial Lock)
• CLIP Number Notification
When dialling an emergency number, the preassigned CLIP number for the extension will be sent as a
location identification number. (→ 4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/
COLP))

86 Feature Manual
 2.5.4 Trunk Call Features

The CLIP number assigned to the extension will be sent regardless of the settings such as CLIR or CLIP
number assigned to an ISDN port to be used. This feature is only available when using a PRI (PRI23) line
with E911-compatible services.

PC Programming Manual References

15.4 PBX Configuration—[7-4] TRS—Emergency Dial

Feature Manual References

6.1 Capacity of System Resources

2.5.4.3 Account Code Entry

Description
An account code is used to identify outgoing trunk calls for accounting and billing purposes. The account
code is appended to the SMDR call record. If, for example, a firm uses an account code for each client, the
firm can determine what calls were made for the client, and can submit a bill to the client according to the
client’s account code on the SMDR call record.
There are two methods of entering account codes as follows:
One of the methods is selected for each extension on a COS basis.
Mode Description
Option A user can enter an account code if needed at any time desired.
Forced A user must always enter an account code before seizing a trunk.

Conditions
• An account code can be stored into Memory Dialling (e.g., One-touch Dialling).
• Account Button
A flexible button can be customised as the Account button. The Account button is used in place of the
feature number for entering an account code. This button is useful because it can be used at any time,
while feature number entry is allowed only when hearing a dial tone before seizing a trunk.
• Extension users can enter an account code at any time during a call, including after the call has been
disconnected and a reorder tone is heard. However, if an account code is entered after there is no longer
a reorder tone, the call will not be stored in the SMDR record.
• If more than one account code is entered, the code entered last is printed out on SMDR.
• Even in Forced mode, emergency numbers can be dialled out without an account code.
(→ 2.5.4.2 Emergency Call)
• PT users can also enter an account code for incoming trunk calls during a conversation.
• Verification Code Entry
To identify who made a trunk call for accounting and billing purposes, a verification code is used. This
code can be used at any extension. (→ 2.7.6 Verification Code Entry)

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Account Code Entry
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—CO & SMDR—Account
Code Mode

Feature Manual References

2.21.2 Flexible Buttons

Feature Manual 87
2.5.4 Trunk Call Features

2.22.1.1 Station Message Detail Recording (SMDR)

5.1.1 Class of Service (COS)

Operating Manual References

1.2.1 Basic Calling

2.5.4.4 Dial Type Selection

Description
The dialling mode (rotary or tone) can be selected for each analogue trunk through system programming
regardless of the originating extension (under contract with the telephone company).
There are the following modes:
Mode Description
DTMF (Dual Tone Multi- The dialling signal from an extension is converted to tone dialling.
Frequency) DTMF signals are transmitted to the trunk.
Pulse Dial (Rotary) The dialling signal from an extension is converted to rotary dialling.
Rotary pulses are transmitted to the trunk.

Conditions
• Pulse to Tone Conversion
It is possible for an extension user to temporarily switch from Pulse mode to DTMF mode so that the user
can access special services such as computer-accessed long distance calling or voice mail services. To
switch to DTMF mode, wait for a preprogrammed time period (Default: five seconds) after the trunk is
connected, or press . This feature works only on trunks set to Pulse mode. DTMF mode cannot be
changed to Pulse mode.
• It is possible to select the pulse rate for a trunk port that has been set to Pulse mode. There are two pulse
rates: Low (10 pps) and High (20 pps).
• It is possible to assign the minimum duration of the DTMF signal sent to a trunk port that has been set to
DTMF mode.

PC Programming Manual References

9.24 PBX Configuration—[1-1] Configuration—Slot—Port Property - LCO Port
→ Dialling Mode
→ DTMF Width
→ Pulse Speed

2.5.4.5 Reverse Circuit

Description
A circuit in the PBX can detect the reverse signal from the telephone company when an extension user tries
to make a trunk call. This detects the start (a called party goes off-hook) and end (the called party goes on-
hook) of an outgoing trunk call. When a trunk call is received, the circuit can also detect the reverse signal
after an outside caller goes on-hook.
If Reverse Circuit Detection is disabled, the total duration of the call is not accurately recognised by the
PBX. The duration of a call can be verified on SMDR using this feature (→ 2.22.1.1 Station Message Detail
Recording (SMDR)).
It is possible to select whether the PBX detects the reverse signal for outgoing trunk calls only, or for both
outgoing and incoming trunk calls, or for no trunk calls (detection disabled) through system programming.

88 Feature Manual
 2.5.4 Trunk Call Features

Conditions
• This feature is not available for the following ELCOT/LCOT cards:
KX-TDA6381 (ELCOT16)
KX-TDA6382 (ELCOT16)
KX-TDA0181AL (LCOT16)
KX-TDA0180AL (LCOT8)

PC Programming Manual References

9.24 PBX Configuration—[1-1] Configuration—Slot—Port Property - LCO Port—Reverse Detection

2.5.4.6 Trunk Busy Out

Description
The PBX can monitor the loop current sent through analogue trunks, preventing users from seizing trunks
where a loop current is not detected. When loop currents are not detected, trunks are set to Busy Out
status, and become unable to make or receive calls. A trunk in Busy Out status cannot be used for making
calls as a TIE line, as part of a trunk group, or with the ARS feature, and cannot receive trunk calls. If a user
tries to seize a trunk set to Busy Out status, the user will hear a reorder tone.
This is useful if some or all trunks are occasionally unavailable because of problems with the external
telecommunications environment.

Conditions
• This feature is not available for the following LCOT cards:
KX-NCP1180NE (LCOT4)
KX-TDA0181NE (LCOT16)
KX-TDA0180NE (LCOT8)
KX-TDA0183NE (LCOT4)
• Loop current detection is performed on active trunks whenever the trunk is seized and/or at fixed
intervals.
• When a trunk is in busy-out status, loop current detection is performed at fixed intervals, returning the
trunk to in-service status once a loop current is detected. An extension assigned as the manager can
manually change the trunk back to in-service status.
• Trunk status changes are recorded in the error log of the PBX.
• Busy Out status is maintained even when the PBX is reset.
• Busy Out status is cleared when:
– a call is successfully received (i.e., a loop current is detected) on that trunk.
– the S-CO button for that trunk is pressed and a loop current is detected.

PC Programming Manual References

9.24 PBX Configuration—[1-1] Configuration—Slot—Port Property - LCO Port—Busy Out Status
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Busy Out Cancel
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Manager—Manager
10.9 PBX Configuration—[2-9] System—System Options—Option 5—Busy Out—Busy Out for Analogue CO

Feature Manual References

5.1.6 Manager Features

Feature Manual 89
2.5.4 Trunk Call Features

Operating Manual References

2.1.6 Allowing Users to Seize an Unavailable Outside Line (Trunk Busy Out)

2.5.4.7 Pause Insertion

Description
Pressing a PAUSE button inserts a preprogrammed Pause time between digits of a user-dialled number
before the number is dialled out, allowing certain numbers separated with a pause to be used to access
certain features (e.g., access codes, seizing idles lines, etc.).
When a pause is needed, pauses must be inserted manually (pressing the PAUSE button) in all cases,
except for the following access codes where a pause is automatically inserted between the user-dialled
access code and the subsequent digits:
a. Host PBX Access code (→ 2.5.4.8 Host PBX Access Code (Access Code to the Telephone Company
from a Host PBX))
b. Special Carrier Access code (→ 2.5.4.9 Special Carrier Access Code)
c. Second Dial Tone Waiting code

Conditions
• The Pause time is programmable for each trunk.
• Pauses can be stored in Memory Dialling.
• When a Second Dial Tone Waiting code is dialled after seizing a trunk, a preprogrammed number of
pauses are inserted after the code.
• ARS
A pause is not automatically inserted between the user-dialled access code and the subsequent digits
when the ARS mode is enabled. (→ 2.8.1 Automatic Route Selection (ARS))

PC Programming Manual References

9.24 PBX Configuration—[1-1] Configuration—Slot—Port Property - LCO Port—Pause Time
10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—During Conversation—Pause
Signal Time
14.4 PBX Configuration—[6-4] Feature—Second Dial Tone

2.5.4.8 Host PBX Access Code (Access Code to the Telephone

Company from a Host PBX)
Description
This PBX can be installed behind an existing PBX (host PBX) by connecting the extension ports of the host
PBX to the trunk ports of this PBX (behind PBX). A Host PBX Access code is required for the behind PBX to
access the telephone company (e.g., to make outside calls) through the host PBX. The Trunk access
number of the host PBX should be stored as a Host PBX Access code on a trunk group of the behind PBX.
A preprogrammed Pause time will be automatically inserted between the user dialled Host PBX Access
code and the subsequent digits (→ 2.5.4.7 Pause Insertion).

90 Feature Manual
 2.5.4 Trunk Call Features

[Example]

Telephone Company

Host PBX
Access Code: 0

Host PBX
Outside Party
(01-23-4567)

Idle Line
Access No.: 9
Extn. 101 Extn. 102

TRG1 Dials "0-01-23-4567".

Host PBX
PBX Access Code
Telephone
No.

Dials "9-0-01-23-4567". Dials "9-101".

Idle Line Telephone No. Idle Line

Access No. Access No.

Host PBX Extn. No.

Access Code of the Host PBX

Note
"0" should be assigned as a Host PBX Access code for trunk group (TRG) 1 of the behind PBX.

Conditions
• TRS/Barring
TRS/Barring checks only the dialled telephone number excluding the Host PBX Access code when
accessing the telephone company through the host PBX. (→ 2.7.1 Toll Restriction (TRS)/Call Barring
(Barring))
• ARS
A pause is not automatically inserted between the user-dialled access code and the subsequent digits
when the ARS mode is enabled. (→ 2.8.1 Automatic Route Selection (ARS))
• SMDR
The dialled number including the Host PBX Access code will be recorded on SMDR only if the modified
number setting is selected in the ARS setting for SMDR.
• When a Host PBX Access code is assigned to a trunk group, calls to extensions of the host PBX are not
recorded on SMDR.
• A Host PBX Access Code can be used to record only long distance calls on SMDR when a trunk port is
connected directly to the telephone company (not a host PBX). This is allowed when the long distance

Feature Manual 91
2.5.5 Seizing a Line Features

code (e.g., "0") is assigned as the Host PBX Access code. All local calls (e.g., calls that do not require a
"0" to be dialled first) are treated as extensions of the telephone company and do not get recorded on
SMDR, because in this case this PBX recognises the telephone company as the host PBX.
Therefore, only long distance calls are recorded on SMDR.

PC Programming Manual References

11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Host PBX Access Code
19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR Options—Option—ARS Dial

Feature Manual References

2.22.1.1 Station Message Detail Recording (SMDR)
6.1 Capacity of System Resources

2.5.4.9 Special Carrier Access Code

Description
If the PBX has access to multiple telephone companies, a Special Carrier Access code assigned through
system programming is required every time a trunk call is made without using ARS.
A preprogrammed Pause time will be automatically inserted between the user-dialled Special Carrier
Access code and the subsequent digits. (→ 2.5.4.7 Pause Insertion)

Conditions
• TRS/Barring
TRS/Barring checks only the dialled telephone number excluding the Special Carrier Access code.
(→ 2.7.1 Toll Restriction (TRS)/Call Barring (Barring))
• ARS
A pause is not automatically inserted between the user-dialled access code and the subsequent digits
when the ARS mode is enabled. (→ 2.8.1 Automatic Route Selection (ARS))
• If this PBX is installed behind an existing host PBX:
A Special Carrier Access code and a Host PBX Access code should be assigned separately: these codes
cannot be assigned together as one code. (→ 2.5.4.8 Host PBX Access Code (Access Code to the
Telephone Company from a Host PBX))

PC Programming Manual References

15.3 PBX Configuration—[7-3] TRS—Special Carrier

Feature Manual References

2.8.1 Automatic Route Selection (ARS)
6.1 Capacity of System Resources

2.5.5 Seizing a Line Features

2.5.5.1 Seizing a Line Features—SUMMARY
Description
An extension user can select the line seized for making calls by the following methods:

92 Feature Manual
 2.5.5 Seizing a Line Features

Feature Description & Reference

Line Preference—Outgoing A user can select the line to be seized when going off-hook.

→ 2.5.5.2 Line Preference—Outgoing

Trunk Access A user can select the Trunk Access method every time he makes a
trunk call.

→ 2.5.5.3 Trunk Access

2.5.5.2 Line Preference—Outgoing

Description
Through personal programming (Preferred Line Assignment-Outgoing), PT users can select the outgoing
line they prefer to originate calls on when going off-hook, from the following line preferences:
Line Preference Description
ICM/PDN When an extension user goes off-hook, an extension line is selected
automatically. If the extension is a PDN extension, the first available
Primary Directory number (PDN) button is selected automatically.
(→ 2.9.1 Primary Directory Number (PDN)/Secondary Directory
Number (SDN) Extension)
Idle Line When an extension user goes off-hook, an idle trunk is selected
automatically from the assigned trunk groups.
No Line When an extension user goes off-hook, no line is selected. The
extension user must select the desired line to make a call.
Prime Line When an extension user goes off-hook, the preset line is selected
automatically. A prime line can be selected from the Line Access
buttons: S-CO, G-CO, L-CO, ICD Group.

Conditions
• Line Preference Override
A user can override the preset Line Preference temporarily by pressing the desired Line Access button or
Memory Dialling button (e.g., One-touch Dialling) before going off-hook.
• To select Idle Line Preference, the trunk groups available to the extension should be programmed on a
COS basis. Also trunk groups available for Idle Line Access should be assigned.

PC Programming Manual References

10.7.2 PBX Configuration—[2-7-2] System—Class of Service—External Call Block
11.1.2 PBX Configuration—[3-1-2] Group—Trunk Group—Local Access Priority
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 4—Outgoing Preferred
Line

Operating Manual References

3.1.2 Settings on the Programming Mode

Feature Manual 93
2.5.5 Seizing a Line Features

2.5.5.3 Trunk Access

Description
The following methods can be used to access a trunk:
Method Description Accessing method
Idle Line Access Selects an idle trunk Dial the Idle Line Access number, or
(Local Access) automatically from the assigned press a L-CO button.
trunk groups.
Trunk Group Access Selects an idle trunk from the Dial the Trunk Group Access
corresponding trunk group. number and a trunk group number,
or press a G-CO button.
S-CO Line Access Selects the desired trunk Dial the S-CO Line Access number
directly. and the trunk number, or press the
S-CO button.

Conditions
• COS programming determines the trunk groups available for making calls.
• Trunk numbers can be referred on a trunk port basis.
• Button Assignment
A flexible button can be customised as a G-CO, L-CO, or S-CO button as follows:
Type Parameter
No parameter (all assigned trunk groups through system
Loop-CO (L-CO)
programming are applied.)
Group-CO (G-CO) A specified trunk group.
Single-CO (S-CO) A specified trunk.

It is possible to assign:
– the same trunk to the S-CO button and to a G-CO button.
– the same trunk group to more than one G-CO button.
– more than one L-CO button.

Dialling the Trunk Access number selects a CO button in the following order: S-CO → G-CO → L-CO
• Direct Trunk Access
Pressing an idle CO button automatically switches on the hands-free operation mode and allows a user to
use On-hook Dialling. The user need not press the SP-PHONE button, MONITOR button, or lift the
handset.
• Group Hunting Order for Idle Line Access
An idle trunk is selected from the trunk groups assigned for Idle Line Access. If multiple trunk groups are
available, the trunk group hunting sequence can be determined through system programming.
• Trunk Hunting Order for Idle Line Access and Trunk Group Access
The trunk hunting sequence in a trunk group (from lowest numbered trunk, from highest numbered trunk
or rotation) can be determined through system programming.
• A company name or customer name can be assigned on a trunk port basis so that the operator or
extension user can view the destination that the external caller is trying to reach before answering. This is
useful, for example, when multiple companies share the same operator.

94 Feature Manual
 2.5.5 Seizing a Line Features

• It is possible to identify the trunk ports that have trunks connected. This prevents extension users from
originating a call to a trunk that is not connected.

PC Programming Manual References

9.22 PBX Configuration—[1-1] Configuration—Slot—SLC2 - Port Property—Connection
9.24 PBX Configuration—[1-1] Configuration—Slot—Port Property - LCO Port—Connection
9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Main—Connection
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Main—Connection
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Idle Line Access (Local Access)
→ Trunk Group Access
→ Single CO Line Access
10.7.2 PBX Configuration—[2-7-2] System—Class of Service—External Call Block
11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Main—Line Hunting Order
11.1.2 PBX Configuration—[3-1-2] Group—Trunk Group—Local Access Priority
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Parameter Selection (for Single CO)
→ Parameter Selection (for Group CO)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Parameter Selection (for Single CO)
→ Parameter Selection (for Group CO)
16.1 PBX Configuration—[8-1] ARS—System Setting—ARS Mode
18.1 PBX Configuration—[10-1] CO & Incoming Call—CO Line Settings—CO Name

Feature Manual References

2.21.2 Flexible Buttons
5.1.1 Class of Service (COS)

Operating Manual References

1.2.1 Basic Calling

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2.6 Memory Dialling Features

2.6 Memory Dialling Features

2.6.1 Memory Dialling Features—SUMMARY
Description
An extension user can store frequently dialled numbers in the PBX extension data and/or the PBX system
data. A stored number is dialled automatically with a simple operation.
1. Features
Feature Storing Method & Reference
One-touch Dialling • Personal Programming
• System Programming
(PC Programming only)

→ 2.6.2 One-touch Dialling

Last Number Redial (Outgoing Call Recently dialled telephone numbers are
Log) automatically stored.

→ 2.6.3 Last Number Redial

Speed Dialling Personal • Personal Programming
• Personal Operation with the Feature Number
• System Programming
(PC Programming only)

→ 2.6.4 Speed Dialling—Personal/System

System System Programming

→ 2.6.4 Speed Dialling—Personal/System

External System Programming

→ 2.6.5 Speed Dialling—External (LDAP)

Quick Dialling System Programming
(PC Programming only)

→ 2.6.6 Quick Dialling

Hot Line • Personal Programming
• Personal Operation with the Feature Number
• System Programming
(PC Programming only)

→ 2.6.7 Hot Line

Incoming Call Log Incoming call information is automatically
stored.

→ 2.19.2 Incoming Call Log

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 2.6.2 One-touch Dialling

2. Valid Input
Display while
Input Description
Entering
0–9/ /# 0–9/ /# Store the digits and #.
PAUSE (Pause) P Store a pause by pressing the PAUSE
button. (→ 2.5.4.7 Pause Insertion)
FLASH/RECALL F Store a flash/recall signal (EFA mode)
(Hooking)
*1
by pressing the FLASH/RECALL button
at the beginning of the number.
(→ 2.11.7 External Feature Access
(EFA))
*1
INTERCOM (Secret) [] Conceal all or part of the number by
pressing the INTERCOM button at the
beginning and at the end of the number
to be concealed. It is programmable
whether the concealed part will appear
on SMDR.
*1
TRANSFER (Transfer) T Store a transfer command by pressing
the TRANSFER button at the beginning
of the number (used only for a One-
touch Dialling). (→ 2.12.1 Call Transfer)
[Example] Storing "T + 305"=
Transferring a call to extension 305.

*1 Available only when in system/personal programming mode

[Secret Dialling Example]

When storing the number "9-123-456-7890" and concealing the telephone number "123-456-7890",

Enter 9 INTERCOM 1 2 3 4 5 6 7 8 9 0 INTERCOM .

Note
• It is possible to store a Memory Dialling feature number at the beginning of the Memory Dialling
numbers.
• It is possible to store several feature numbers in one Memory Dialling location.

Conditions
• Trunk Access by Memory Dialling
A specific Trunk Access number can be stored with the telephone number in Memory Dialling. However, if
Memory Dialling is done after selecting a trunk, the stored Trunk Access number is ignored and the
telephone number is sent using the selected trunk.

2.6.2 One-touch Dialling

Description
A PT user can access a person or feature by pressing a single button. This is activated by storing the
number (e.g., extension number, telephone number, or feature number) in a One-touch Dialling button.

Feature Manual 97
2.6.3 Last Number Redial

Example: One-touch Voice Mail Feature Access

It is possible to assign a One-touch Dialling button for direct access to a Unified Messaging feature (→
3 Unified Messaging System). For example, to record a message to mailbox number 123 directly and the
UM group’s floating extension number is 165, assign "165#6123" to a One-touch Dialling button. When
pressing this button, the mailbox’s greeting message will be heard.

Conditions
• One-touch Dialling Button
A flexible button can be customised as a One-touch Dialling button.
• Full One-touch Dialling
There is no need to go off-hook before pressing the One-touch Dialling button.

PC Programming Manual References

8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 6—Flexible Key
Programming Mode
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Dial (for One-touch)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Dial (for One-touch)

Feature Manual References

2.21.2 Flexible Buttons
6.1 Capacity of System Resources

Operating Manual References

1.2.2 Easy Dialling

2.6.3 Last Number Redial

Description
Every extension automatically saves recently dialled external telephone numbers and extension numbers to
allow the same number to be dialled again easily. Through system programming, the outgoing call log can
be set to log dialled extension numbers.

Automatic Redial:
If Last Number Redial is performed in hands-free mode and the called party is busy, redialling will be
automatically retried a preprogrammed number of times at preprogrammed intervals. The Redial Call No-
answer Ring Duration time is programmable.
This feature is available only on certain PT models which have the SP-PHONE button.
Outgoing Call Log:
Information on outgoing trunk calls and intercom (including TIE) calls is automatically logged up at each
*1
extension. Users of display PTs APT and DECT PS can view details of a preset number of recently dialled
telephone numbers, and easily call the same party again. By selecting Enable in the program settings, the
call logs of calls made to the extensions can be acquired.
*1 DECT terminals can only use this function with models that can distinguish the data in the outward call logs between extensions
outgoing and extensions incoming.

Conditions
[General]

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 2.6.3 Last Number Redial

• If a new number is dialled when the Outgoing Call Log is full and/or Automatic Redial contains a number,
the data of the oldest stored call will be deleted, and the new number will be stored.
• If any dialling operations are performed or an incoming call is answered during Automatic Redial,
Automatic Redial is cancelled.
• Automatic Redial may not be available depending on the busy tone pattern.
• Automatic Redial is not available on SIP extensions.
• Speed Dial External (LDAP) also supports redial, during redial the registered name of the external server
is shown on the LCD screen (Require AK KX-NSXF005).
• Interrupt Redial
When an outside party, seized trunk, or extension number (including TIE connections) is busy, a user can
attempt to redial the number by pressing the REDIAL button without going on-hook. This can be
performed several times without having to go on-hook.
• Outgoing Call Log Display by REDIAL Button
Pressing the REDIAL button on a display PT while on-hook can display the Outgoing Call Log. System
programming is required for this operation.
• If the Outgoing Call Log is used to redial an outside party or an extension number (including TIE
connections) or if a number that is already stored in the Outgoing Call Log is manually redialled again, the
number will be stored in the call log multiple times. However, calls made using the REDIAL button are not
stored in the Outgoing Call Log again.
• It is possible to change the number of records that can be stored at each extension through system
programming.
• To log intercom calls in the outgoing call log, refer to "10.9 PBX Configuration—[2-9] System—System
Options—Option 7—Outgoing Call Log—Extension Call" in the PC Programming Manual.
• Logs for multiple calls to the same destination are combined and displayed with the most recent call log.
• If an extension user makes a call over a TIE connection using the PBX Code method (Access with PBX
Code), the outgoing call log does not display the Access Code on the PT’s display.
• If an extension user uses a DSS key to make a call to another extension, the user can use the redial
feature to call the same extension number.
• If an extension user uses an SDN key to make a call to the corresponding owner extension, the user
cannot use the redial feature to call the owner extension again.
[Personal/System Speed Dialling Name Display]
• When a trunk call is made from the Outgoing Call Log, if a name is registered to the Personal/System
Speed Dialling entry, the name is displayed on the second line of a display PT’s LCD. The name is not
displayed on a phone with a one line LCD.
• The redial name for System Speed Dial and outgoing calls using LDAP is shown up to 20 characters.
• If a name is registered to the Personal/System Speed Dialling entry, when another call is made using the
REDIAL button after a trunk call is made using Personal/System Speed Dialling, the name is displayed on
the second line of a display PT’s LCD.
• If a name is registered to the LDAP server, the name is registered to redial entry.
When another call is made using the REDIAL button after a trunk call is made System Speed Dialling
External, the name is displayed on the second line of a display PT’s LCD.
• When a trunk call is made from the Outgoing Call Log, if a name is not registered to the Personal/System
Speed Dialling entry, the dialled number is displayed on a display PT’s LCD.
• If the number registered in the Personal/System Speed Dialling is changed after the number is recorded
in the Outgoing Call Log, the Personal/System Speed Dialling entry name will not be displayed. Instead,
the name registered in the Outgoing Call Log will be displayed.

Feature Manual 99
2.6.4 Speed Dialling—Personal/System

• If the name registered in the Personal/System Speed Dialling is changed after the name is recorded in
the Outgoing Call Log, the changed name will be displayed on a display PT’s LCD.
• If the mode to save digits dialled after connection is enabled, all digits (including the automatically entered
"P" for pause) dialled up until the end of the call must be registered in the Personal/System Speed
Dialling entry to enable this feature.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone
→ Automatic Redial—Repeat Counter
→ Automatic Redial—Repeat Interval
→ Automatic Redial—Redial Call Ring Duration
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Redial
10.9 PBX Configuration—[2-9] System—System Options—Option 2
→ Redial—Automatic Redial when No Answer (ISDN)
→ Redial—Save Dial After Connection to Redial Memory
→ Redial—Call Log by Redial key
10.9 PBX Configuration—[2-9] System—System Options—Option 7—Outgoing Call Log—Extension Call
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 7—Outgoing Call Log
Memory

Feature Manual References

5.2.1 IP Proprietary Telephone (IP-PT)
6.1 Capacity of System Resources

Operating Manual References

1.2.3 Redial
1.14.1 Using the Call Log

2.6.4 Speed Dialling—Personal/System

Description
An extension user can make calls using abbreviated dialling for frequently dialled numbers which are stored
in the PBX extension data, or the PBX system data.
Personal Speed Dialling is also known as Station Speed Dialling.
Depending on system programming, the System Speed Dial items displayed on a display PT can be limited
to items related to the extension’s tenant.

Conditions
[General]
• Any number (e.g., telephone number, feature number) can be stored in a speed dialling number. A name
can be assigned to each Personal Speed Dialling number through personal programming, and System
Speed Dialling number.
IP Softphones can only be operated in the classical mode.

[Name Display on Trunk Calls]

• When a trunk call is made using Personal/System Speed Dialling, if a name is registered to the Personal/
System Speed Dialling entry, the name is displayed on the second line of a display PT’s LCD. The name
is not displayed on a phone with a one line LCD.
• After a trunk call is made using Personal/System Speed Dialling, if Connected Name Identification
Presentation (CONP) information is returned, the name registered in the CONP information takes

100 Feature Manual

 2.6.4 Speed Dialling—Personal/System

precedence over the Personal/System Speed Dialling name and is displayed on a display PT’s LCD.
(→ 4.3.4.2 Calling/Connected Line Identification Presentation (CLIP/COLP) and Calling/Connected Name
Identification Presentation (CNIP/CONP)—by QSIG)
• If a recall feature or a transfer feature is used, the name registered to the Personal/System Speed Dialling
entry will no longer be displayed on a display PT’s LCD.

[Personal Speed Dialling]

• Display Lock
An extension user can lock the Personal Speed Dialling number display to prevent the numbers from
being viewed at any extension through personal programming (Display Lock). In this case, the Incoming/
Outgoing Call Log displays are also locked, and the voice messages in the user’s mailbox cannot be
played back. An extension personal identification number (PIN) is required to use this feature. (→
2.24.1 Extension Personal Identification Number (PIN))

[System Speed Dialling]

• TRS/Barring Override by System Speed Dialling
It is possible to override the TRS/Barring using the System Speed Dialling. (→ 2.7.1 Toll Restriction
(TRS)/Call Barring (Barring))
• System Speed Dialling Display by AUTO DIAL/STORE Button
Pressing the AUTO DIAL/STORE button on a display PT while on-hook can display the System Speed
Dialling Directory.

[Limiting the display by tenant—System Speed Dial]

For System Speed Dial, an extension can reference the data for System or for each tenant the extension is
member of, depending on system programming. In "Tenant Exclusive" mode, System Speed Dial is
displayed on display PTs as follows:
• Only information about extensions that belong to the tenant as is displayed.
• Changes to System Speed Dial settings only affect each tenant. They do not affect the whole system.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ System Speed Dialling / Personal Speed Dialling
→ Personal Speed Dialling - Programming
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—TRS—TRS Level for System
Speed Dialling
8.4 Users—Speed Dial
14.1 PBX Configuration—[6-1] Feature—System Speed Dial
14.6 PBX Configuration—[6-6] Feature—Tenant—System Speed Dial
14.6 PBX Configuration—[6-6] Feature—Tenant—Extension Directory

Feature Manual References

6.1 Capacity of System Resources

Operating Manual References

1.2.2 Easy Dialling
1.14.1 Using the Call Log
1.14.2 Using the Directories
3.1.2 Settings on the Programming Mode

Feature Manual 101

2.6.5 Speed Dialling—External (LDAP)

2.6.5 Speed Dialling—External (LDAP)

Description
During calls from the proprietary telephone with display (PT), you can use the user’s Active Directory®
(LDAP) as a phone book.
In addition, when receiving a call from an outside line, the caller name can be shown on the telephone by
searching the caller's number in the Active Directory (LDAP).

Conditions
[General]
• This feature can be used with telephones with LCD and Navigator key.
• Activation key for LDAP Connection (KX-NSXF005W) is required to use this feature.
• The system setting for LDAP phone book service availability needs to be enabled.
(→ 28.3.4.2 Network Service—[3-4-2] Client Feature—Directory Service—External Directory—External
Directory Service—Service Availability)
• The search start position of the LDAP database (Base DN) can be configured by each tenant.
• IP Softphones can only be operated in classical mode.

[LDAP phone book service]

• You can make a call to a telephone number obtained from the LDAP database of the Active Directory etc.,
by entering the name you want to search for on the private phone with display.
• The conditions will be configured by the Web-MC in advance as LDAP Database search conditions.
(→2.29.1.6 Linking with LDAP User Information).
• Up to 200 candidates can be displayed in the search results. The number of candidates can be
configured in Program Settings. (Initial value shows 24 candidates)
• Up to 3 LDAP attribute telephone numbers can be acquired from the name of 1 search result. The
acquired LDAP attributes will be configured in the Web-MC in advance. LDAP attributes that can be
selected are as follows.
– cn
– givenname
– homePhone
– mobile
– mail
– company
• With telephones with a single line LCD, select the name displayed on the LCD to make a phone call. The
telephone numbers registered in the LDAP database are dialled with the priority order shown below (if no
telephone number is registered, the telephone number with the next priority will be dialled):
1. Number Attributes1 → 2. Number Attributes2 - 3. Number Attributes3
• When telephone numbers obtained from the LDAP database contain other characters than the numbers
1–9, 0 or the characters # or *, the numbers can still be called but they will be displayed without showing
those characters.
• The redial and call history feature can also be used when making a call using the Active Directory When
redialling, the name searched in the database server is displayed. In addition, searched names are
displayed up to 20 characters and the history is displayed by caller name in Call History.

102 Feature Manual

 2.6.6 Quick Dialling

• When making a call using the LDAP database, you can specify whether or not to apply an outgoing
access code for each attribute that was acquired as a search result. (System setting) The only
Automatically applied access codes applied are [Local outgoing access codes], other access codes are
not supported.
The operation becomes a transmission operation attached with a Caller ID when applying an outgoing
access code. Also in this case, the acquired telephone number is displayed with the outgoing access
code shown in front of the telephone number. (Initial value: attributes 1–3 for telephone numbers are not
applied)
However, if you use this phone book from an outside line capture state, the access number will not
automatically be applied. (Note: In this case, the Caller ID will be displayed but this number will not be
dialled.)

[Incoming Call Name Retrieval Display]

• During an incoming call, the caller number is first searched for in Personal Speed Dial, then in System
Speed Dial, and finally in the LDAP database. Finally, the name of the data consistent with the number of
the caller is displayed.
• In order to perform Name Retrieval Display during a call, the LDAP Name Retrieval and Display
availability System Setting for during a call must be Enabled.
• The LDAP database is queried. If there is no matching data, a name from the network will be displayed; if
there is no name registered to the network, no name will be displayed.
• If any of the following cases are detected when using the LDAP database, a name from the network will
be displayed. If there is no name registered to the network, no name will be displayed.
1. When you cannot connect to the LDAP database
2. When you cannot login to the LDAP database
3. During a data search time-out
• During a call, when the response to the incoming call, and the search results during a call after the
response have been reported before the LDAP database search results are reported to the NSX, the
reported LDAP data will be displayed after being overwritten.
• When a name which matches the LDAP database is displayed, the displayed name is saved in Call
History up to 20 characters and that name will be displayed as an indication of the Call History.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone
28.3.4.3 Network Service—[3-4-3] Client Feature—Directory Service—User Container Sync

Feature Manual References

2.29.1.6 Linking with LDAP User Information

Operating Manual References

1.2.3 Redial

2.6.6 Quick Dialling

Description
An extension user can access an extension or feature by simply dialling a 1–8 digit Quick Dialling number.

Conditions
• Quick Dialling is convenient in cases such as the following:

Feature Manual 103

2.6.7 Hot Line

– Room service calls in a hotel

– Calling another branch via the public network.
• Quick Dialling numbers follow the flexible numbering plan.
(→ 5.5.6 Flexible Numbering/Fixed Numbering)
• The following example shows how Quick Dialling numbers can be stored and utilised:
Location No. Quick Dialling No. Desired Number
Quick Dialling 01 110 9110 (Trunk Call)
Quick Dialling 02 5 3016 (Room Service)
Quick Dialling 03 2011 90123456789 (Another Branch)
: : :

• It is possible to use Quick Dialling to make a trunk call via an extension at another site in a Multi-
connection network when the PBX is in Break mode (→ 5.1.4 Time Service). System programming at
each site is required to enable this feature. This feature is not available when calling extensions at your
own site. For details about the programming for each site, refer to the PC Programming Manual (→
10.6.2 PBX Configuration—[2-6-2] System—Numbering Plan—Quick Dial).

PC Programming Manual References

10.6.2 PBX Configuration—[2-6-2] System—Numbering Plan—Quick Dial

Feature Manual References

4.3.3 ISDN Virtual Private Network (ISDN-VPN)
5.1.4 Time Service
6.1 Capacity of System Resources

Operating Manual References

1.2.2 Easy Dialling

2.6.7 Hot Line

Description
An extension user can set his extension to automatically dial a preprogrammed telephone or extension
number whenever he goes off-hook. This feature is also known as Pickup Dialling.
If the Hot Line feature is set, a dial tone is generated for a specified Waiting time assigned through system
programming when the user goes off-hook. During the Waiting time the user can dial another party,
overriding the Hot Line feature. If no number is dialled, the preprogrammed number will automatically start
being dialled.

Conditions
• Capable Telephone
PT, SLT, T1-OPX, and PS

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Dial—Hot Line
(Pickup Dial) Start
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Hot Line (Pickup Dial)
Program Set / Cancel

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 2.6.7 Hot Line

8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 2

→ Pickup Dial Set
→ Pickup Dial No.

Feature Manual References

6.1 Capacity of System Resources

Operating Manual References

1.2.2 Easy Dialling

Feature Manual 105

2.7 Toll Restriction (TRS)/Call Barring (Barring) Features

2.7 Toll Restriction (TRS)/Call Barring (Barring)

Features
2.7.1 Toll Restriction (TRS)/Call Barring (Barring)
Description
TRS/Barring can prohibit an extension user from making certain trunk calls by COS programming. It is
applied when the user goes off-hook, a trunk is seized and then a dialled number is sent to the trunk.
Each COS is programmed to have a TRS/Barring level for each time mode (day/lunch/break/night).
There are seven levels available. Level 1 is the highest level and level 7 is the lowest. That is, level 1 allows
all trunk calls and level 7 prohibits all trunk calls. Levels 2 through 6 are used to restrict calls by combining
preprogrammed Denied and Exception Code Tables.

Denied Code Tables

An outgoing trunk call made by an extension with a level between 2 and 6 is first checked against the
applicable Denied Code Tables. If the leading digits of the dialled number (not including the Trunk Access
number) are not found in the table, the call is made. There are five Denied Code Tables, one for each of
Levels 2 through 6 respectively.
Complete every table by storing numbers that are to be prohibited. These numbers are defined as denied
codes.

Exception Code Tables

These tables are used to override a programmed denied code. A call denied by the applicable Denied Code
Tables is checked against the applicable Exception Code Tables, and if a match is found, the call is made.
There are five Exception Code Tables, for Levels 2 through 6 respectively.
Complete every table by storing numbers that are exceptions to the denied codes. These numbers are
defined as exception codes.

TRS/Barring Override by System Speed Dialling

If the call is made using System Speed Dialling, the call can override the TRS/Barring. Each COS is
programmed to have a TRS/Barring level for System Speed Dialling. Except External (LDAP)
Once this feature is set, it permits all extension users to make System Speed Dialling calls with the level for
System Speed Dialling. Any extension which sets Extension Dial Lock can also make a call using System
Speed Dialling.
→ 10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—TRS—TRS Level for
System Speed Dialling

TRS/Barring Level
The TRS/Barring level is determined by the telephone codes set in the Denied Code Tables and Exception
Code Tables.
As shown in the table below, the Denied Code Tables for the higher levels are applied to all levels below it,
and the Exception Code Tables for the lower levels are applied to all levels above it.
Denied Code Tables*1 Exception Code Tables*2
Level 1 Not Programmable Not Programmable
Level 2 Table for Level 2 Tables for Levels 2 through 6
Level 3 Tables for Levels 2 and 3 Tables for Levels 3 through 6
Level 4 Tables for Levels 2 through 4 Tables for Levels 4 through 6

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 2.7.1 Toll Restriction (TRS)/Call Barring (Barring)

Denied Code Tables*1 Exception Code Tables*2

Level 5 Tables for Levels 2 through 5 Tables for Levels 5 through 6
Level 6 Tables for Levels 2 through 6 Table for Level 6
Level 7 Not Programmable Not Programmable

*1 → 15.1 PBX Configuration—[7-1] TRS—Denied Code—Level 2—Level 6

*2 → 15.2 PBX Configuration—[7-2] TRS—Exception Code—Level 2—Level 6

[Usage Example]
Using this method, certain outgoing trunk calls (e.g., international/cellular phone/long distance) can be
restricted as in the example below:
Restricted Allowed
Level 1 No restriction
• International Calls • Countries where Clients are
Located
Level 2 • Cellular Phone Calls
(Boss)
• Long Distance Calls
• Local Calls
• International Calls • Boss’s Cellular Phone
Level 3
• Cellular Phone Calls • Long Distance Calls
(Secretary)
• Local Calls
• International Calls • Local Calls
Level 4
• Cellular Phone Calls
(Operator)
• Long Distance Calls
: : :

In this example, a level 1 user can make any trunk calls. A level 2 user can make international calls to the
countries where clients are located, and can also make cellular phone/long distance/local calls. A level 3
user cannot make international/cellular phone calls apart from to the boss’s cellular phone, but can make
long distance/local calls. A level 4 user cannot make any international/cellular phone/long distance calls, but
can make local calls.
To set TRS/Barring as in the example above, it is necessary to programme the Denied Code and Exception
Code Tables as follows:
Denied Code Tables Exception Code Tables
Level 1 Not Programmable Not Programmable
Leading number to deny Leading number for
Level 2 00 00xx
international calls countries to be allowed
Leading number to deny 090xxxxx Boss’s cellular phone
Level 3 090
cellular phone calls xxx number
Leading number to deny long
Level 4 0 – Not required
distance calls
: : :

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2.7.1 Toll Restriction (TRS)/Call Barring (Barring)

[Programming Example: COS Settings]

Level for Time Mode*1 Level for System Speed
COS No.
Day Lunch Break Night Dialling*2

1 1 1 1 6 1
2 2 2 2 6 1
: : : : : :

*1 → 10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—TRS—TRS Level—Day, Lunch, Break, Night
*2 → 15.5 PBX Configuration—[7-5] TRS—Miscellaneous—TRS Override by System Speed Dialling

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 2.7.1 Toll Restriction (TRS)/Call Barring (Barring)

[Flowchart]

An extension user makes

a trunk call.

Is the call made by System Yes

Speed Dialling ?

No
No Is TRS/Barring Override by
System Speed Dialling enabled?

Yes
Checks the Checks the TRS/Barring
TRS/Barring level for level for System Speed
the time mode of the Dialling of the
extension's COS. extension's COS.

Level 7 Level 1
What is the TRS/Barring level?

Levels 2, 3, 4, 5, 6

Is the dialled number found in No

applicable Denied Code Tables?

Yes

Is the dialled number found in Yes

applicable Exception Code Tables?

No

The call is denied. The call is made.

The user hears reorder tone.

TRS/Barring Settings for Each Level

Through system programming, it is possible to select a different method of TRS/Barring. With this method,
each level has its own separate set of denied codes and exception codes, which are only applied to that
level.
→ 15.5 PBX Configuration—[7-5] TRS—Miscellaneous—TRS Table Mode for Level N (N=2_6)
Denied Code Tables*1 Exception Code Tables*2
Level 1 Not Programmable Not Programmable
Level 2 Table for Level 2 Table for Level 2

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2.7.1 Toll Restriction (TRS)/Call Barring (Barring)

Denied Code Tables*1 Exception Code Tables*2

Level 3 Table for Level 3 Table for Level 3
Level 4 Table for Level 4 Table for Level 4
Level 5 Table for Level 5 Table for Level 5
Level 6 Table for Level 6 Table for Level 6
Level 7 Not Programmable Not Programmable

*1 → 15.1 PBX Configuration—[7-1] TRS—Denied Code—Level 2—Level 6

*2 → 15.2 PBX Configuration—[7-2] TRS—Exception Code—Level 2—Level 6

[Usage Example]
Using this method, it is possible to restrict certain outgoing trunk calls (e.g., international/cellular phone/long
distance) on a department basis, as follows:
Restricted Allowed
Level 1 No restriction
• International Calls • Country where Factory is Located
Level 2 • Cellular Phone Calls
(Engineering) • Long Distance Calls
• Local Calls
• Cellular Phone Calls • Company Cellular Phone
Level 3 • International Calls
(Overseas Sales) • Long Distance Calls
• Local Calls
• International Calls • Cities where Clients are Located
Level 4
• Long Distance Calls • Cellular Phone Calls
(Accounting)
• Local Calls
: : :

In this example, a level 1 user can make any trunk calls. A level 2 user can only make international calls to
the country where the factory is located, and can also make cellular phone/long distance/local calls. A level
3 user can only make cellular phone calls to the company cellular phone, and can also make any
international/long distance/local calls. A level 4 user cannot make any international calls or most long
distance calls, but can make long distance calls to cities where clients are located, cellular phone calls and
local calls.
To set TRS/Barring as in the example above, it is necessary to programme the Denied Code and Exception
Code Tables as follows:
Denied Code Tables Exception Code Tables
Level 1 Not Programmable Not Programmable
Leading number to deny Leading number for country
Level 2 00 00xx
international calls to be allowed
Leading number to deny Leading number for cellular
Level 3 090 090xxxx
cellular phone calls phones to be allowed

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 2.7.1 Toll Restriction (TRS)/Call Barring (Barring)

Denied Code Tables Exception Code Tables

03 Long distance numbers for
Leading number to deny
cities to be allowed, and
Level 4 0 both international and long 06
leading number of cellular
distance calls
090 phones
: : :

Conditions
CAUTION
The software contained in the TRS/Barring feature to allow user access to the network must be
upgraded to recognise newly established network area codes and exchange codes as they are placed
into service.
Failure to upgrade the on-premise PBXs or peripheral equipment to recognise the new codes as they
are established will restrict the customer and users of the PBX from gaining access to the network and
to these codes.
KEEP THE SOFTWARE UP TO DATE WITH THE LATEST DATA.
• A COS should be assigned for each extension.
→ 8.2 Users—Advanced Extension Settings—COS
• TRS/Barring checks are applied to the following:
– ARS
– Trunk Access (Idle Line/Trunk Group/S-CO Line)
• It is programmable whether " " or "#" is checked by the TRS/Barring. This is useful in preventing
unauthorised calls which could be possible through certain telephone company exchanges.
→ 15.5 PBX Configuration—[7-5] TRS—Miscellaneous—TRS Check for Dial " * # "
• It is programmable whether TRS/Barring checks the digits dialled after the External Feature Access
during a trunk call. (→ 2.11.7 External Feature Access (EFA))
→ 15.5 PBX Configuration—[7-5] TRS—Miscellaneous—TRS Check after EFA
• Host PBX Access Code/Special Carrier Access Code
TRS/Barring checks for numbers dialled with a Host PBX Access code (→ 2.5.4.8 Host PBX Access
Code (Access Code to the Telephone Company from a Host PBX)) or a Special Carrier Access code
(→ 2.5.4.9 Special Carrier Access Code) in the following cases:
Stored
Type Not stored
Found Not found
Host PBX Access Deletes the code. A The call is made TRS/Barring checks
Code TRS/Barring check is (excepted from TRS/ the whole number.
carried out on the Barring).
following digits.
Special Carrier Deletes the code. A TRS/Barring checks TRS/Barring checks
Access Code TRS/Barring check is the whole number. the whole number.
carried out on the
following digits.

• ARS

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2.7.1 Toll Restriction (TRS)/Call Barring (Barring)

If ARS is applied to a dialled number, TRS/Barring will check the user-dialled number (not the modified
number by ARS). In this case, a Host PBX Access code and/or a Special Carrier Access code will not be
checked.
• Dialling Digit Restriction during Conversation
The dialling of digits can be restricted while engaged on a received trunk call. If the number of dialled
digits exceeds the preprogrammed limitation, the line will be disconnected.
→ 15.5 PBX Configuration—[7-5] TRS—Miscellaneous—Dial Digits Limitation After Answering—Dial
Digits
• It is possible to select through system programming whether the trunk is disconnected when the Inter-digit
time expires without the TRS/Barring check being completed.
→ 15.5 PBX Configuration—[7-5] TRS—Miscellaneous—Mode when Dial Time-out before TRS Check
– If no disconnection is chosen, the TRS/Barring check will also be performed after the Inter-digit time
expires.
– If disconnection is chosen, the line will be disconnected when the trunk Inter-digit time expires. This
also prevents EFA from being used.
This setting applies to all trunks.
• A TRS/Barring level can be changed by some features. The priority of features, when multiple features
are used, is as follows:
1. Dial Tone Transfer (→ 2.7.4 Dial Tone Transfer)
2. Budget Management (→ 2.7.2 Budget Management)
3. TRS/Barring Override by System Speed Dialling
4. Walking COS/Verification Code Entry
(→ 2.7.5 Walking COS, 2.7.6 Verification Code Entry)
5. Extension Dial Lock

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—TRS
→ TRS Level—Day, Lunch, Break, Night
→ TRS Level for System Speed Dialling
8.2 Users—Advanced Extension Settings—COS
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Parameter Selection (for TRS Level Change)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Parameter Selection (for TRS Level Change)
15.1 PBX Configuration—[7-1] TRS—Denied Code
15.2 PBX Configuration—[7-2] TRS—Exception Code
15.3 PBX Configuration—[7-3] TRS—Special Carrier
15.5 PBX Configuration—[7-5] TRS—Miscellaneous

Feature Manual References

2.5.5.3 Trunk Access
2.6.4 Speed Dialling—Personal/System
2.7.3 Extension Dial Lock
2.8.1 Automatic Route Selection (ARS)
5.1.1 Class of Service (COS)
5.1.4 Time Service
6.1 Capacity of System Resources

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 2.7.2 Budget Management

2.7.2 Budget Management

Description
Limits the telephone usage to a preprogrammed budget on an extension basis. If the amount of the call
charge reaches the limit, an extension user cannot make further trunk calls. An extension assigned as the
manager may increase the limit or clear the amount of the call charge.

Conditions
• If the limit is reached, TRS/Barring Level 7 is applied. (→ 2.7.1 Toll Restriction (TRS)/Call Barring
(Barring))
• Budget Management for Verified Call
If an extension user makes a trunk call with a verification code, the call charge will be added to the total
for the verification code (not the extension). (→ 2.7.6 Verification Code Entry) Each verification code can
be assigned a call charge limit.
• Budget Management for Walking COS
If an extension user makes a trunk call from an extension using the Walking COS feature, the call charge
will be added to the extension of the extension user (not the extension that the call was made on). (→
2.7.5 Walking COS)
• Pay tone service or ISDN Advice of Charge (AOC) service is required for this feature.
• It is possible to select whether to disconnect the line (disconnect mode) after a warning tone or only to
send a warning tone when the amount of the call charge reaches the preprogrammed limit during a
conversation.
• When multiple extension users are using the same verification code or the same extension (through the
use of Walking COS) simultaneously, each caller can have access to the total remaining budget of the
extension or verification code.

PC Programming Manual References

8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 3—Charge Limit
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Charge—Charge Options—Action at Charge
Limit
14.3 PBX Configuration—[6-3] Feature—Verification Code—Budget Management

Operating Manual References

4.1.2 Manager Programming

2.7.3 Extension Dial Lock

Description
An extension user can change the TRS/Barring level of the telephone (→ 2.7.1 Toll Restriction (TRS)/Call
Barring (Barring)) so that other users cannot make inappropriate trunk calls. An extension personal
identification number (PIN) is used to unlock the telephone (→ 2.24.1 Extension Personal Identification
Number (PIN)). This feature is also known as Electronic Station Lockout.

Conditions
• This feature also restricts changing the FWD destination. (→ 2.3.2 Call Forwarding (FWD))
• Remote Extension Dial Lock
Overrides Extension Dial Lock. If an extension assigned as the manager sets Remote Extension Dial
Lock on an extension that has already been locked by the extension user, the user cannot unlock it. If a

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2.7.4 Dial Tone Transfer

manager extension unlocks an extension that has been locked by the extension user, the extension will
be unlocked. This feature is also known as Remote Station Lock Control.
• TRS/Barring Level
COS programming determines the TRS/Barring level for Extension Dial Lock.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Extension Dial Lock Set / Cancel
→ Remote Extension Dial Lock Off
→ Remote Extension Dial Lock On
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—TRS—TRS Level on
Extension Lock

Feature Manual References

5.1.1 Class of Service (COS)

Operating Manual References

1.6.3 Preventing Other People from Using Your Telephone (Extension Dial Lock)
2.1.1 Extension Control

2.7.4 Dial Tone Transfer

Description
An extension assigned as the manager can change the TRS/Barring level (→ 2.7.1 Toll Restriction (TRS)/
Call Barring (Barring)) for an extension user temporarily. After that, the extension user can make his call.

[Example]
An extension user can call a manager to release the restriction on outgoing calls (e.g., international calls).

(3) Make a trunk call

(2) Change
TRS/Barring level
Toll Restriction/
Call Barring button
(1) Call
Guest Room Manager
(Trunk call
restricted)

Conditions
• The modified TRS/Barring level only applies to the next one call placed at the user’s extension.
• Toll Restriction/Call Barring Button
A manager extension must store the desired TRS/Barring level in the Toll Restriction/Call Barring button.
A flexible button can be customised as the Toll Restriction/Call Barring button.

PC Programming Manual References

8.3.1 Users—Flexible Button—Wired Extension

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 2.7.5 Walking COS

→ Type
→ Parameter Selection (for TRS Level Change)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Parameter Selection (for TRS Level Change)

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

2.1.3 Restriction Level Control (Dial Tone Transfer)

2.7.5 Walking COS

Description
A user can enter his extension number and extension personal identification number (PIN)
(→ 2.24.1 Extension Personal Identification Number (PIN)) at another extension, to make the following
types of call using his Class of Service, including TRS level, overriding the other extension’s Class of
Service.
• Trunk call
• TIE line call
• Intercom call
• External Relay Control (→ 2.18.4 External Relay Control)
After performing Walking COS, the following features are also available for the specified extension:
• Call Forwarding (FWD)/Do Not Disturb (DND) setting (→ 2.3 Call Forwarding (FWD)/Do Not Disturb
(DND) Features)
• Incoming Call Distribution Group Log-in/Log-out (→ 2.2.2.7 Log-in/Log-out)
• Absent Message setting (→ 2.20.2 Absent Message)
• Extension Dial Lock (→ 2.7.3 Extension Dial Lock)
• Time Service—Changing the Time Mode (day/lunch/break/night) (→ 5.1.4 Time Service)

Conditions
• When a trunk call is made using Walking COS:
– the Class of Service of the specified extension is applied (→ 5.1.1 Class of Service (COS))
– the budget of the specified extension is applied (→ 2.7.2 Budget Management)
– the Itemised Billing code of the specified extension is applied (→ 2.8.1 Automatic Route Selection
(ARS))
– the specified extension number is recorded on SMDR as the call originator, instead of the extension
number of the actual extension used (→ 2.22.1.1 Station Message Detail Recording (SMDR)).
• Walking COS is also available through DISA. (→ 2.16.1 Direct Inward System Access (DISA))
• Extension PIN
An extension personal identification number (PIN) is required to use this feature. (→ 2.24.1 Extension
Personal Identification Number (PIN)) If the wrong PIN is entered three times, the line will be
disconnected.

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2.7.6 Verification Code Entry

• This feature cannot be used for extensions which the extension being operated is prevented from calling
by Internal Call Block. (→ 2.1.2.2 Internal Call Block)

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—User Remote Operation /
Walking COS / Verification Code
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Optional Device & Other
Extensions—Remote Operation by Other Extension
8.1.1 Users—User Container—Add User/Edit User—User Information—User PIN / Mailbox Password ["0-9"]

Feature Manual References

2.16.1 Direct Inward System Access (DISA)
5.1.1 Class of Service (COS)

Operating Manual References

1.2.7 Calling without Restrictions
1.2.9 Setting Your Telephone from Another Extension or through DISA (Remote Setting)

2.7.6 Verification Code Entry

Description
An extension user can enter a verification code when calling from his own or any other extension, to change
the TRS/Barring level (→ 2.7.1 Toll Restriction (TRS)/Call Barring (Barring)) or to identify the call for
accounting and billing purposes. A verification code personal identification number (PIN) is required to use
this feature.

Conditions
CAUTION
There is a risk that fraudulent telephone calls will be made if a third party discovers a personal
identification number (PIN) (verification code PIN or extension PIN) of the PBX.
The cost of such calls will be billed to the owner/renter of the PBX.
To protect the PBX from this kind of fraudulent use, we strongly recommend:
a. Keeping PINs secret.
b. Selecting complex, random PINs that cannot be easily guessed.
c. Changing PINs regularly.
• When a trunk call is made using Verification Code Entry:
– the Class of Service of the specified extension is applied (→ 5.1.1 Class of Service (COS))
– the budget of the specified extension is applied (→ 2.7.2 Budget Management)
– the Itemised Billing code of the specified extension is applied (→ 2.8.1 Automatic Route Selection
(ARS))
– + verification code is recorded on SMDR as the call originator, instead of the extension number of
the actual extension used (→ 2.22.1.1 Station Message Detail Recording (SMDR)).
• Verification Code Entry through DISA
This feature is also available through DISA. (→ 2.16.1 Direct Inward System Access (DISA))
• Verification Code PIN

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 2.7.6 Verification Code Entry

A verification code PIN must be assigned for each verification code through system programming or
through manager programming.
• Verification Code PIN Lock
If the wrong PIN is entered three times, the line will be disconnected. If the wrong PIN is entered a
preprogrammed number of times successively, the PIN for the verification code will be locked. Only an
extension assigned as the manager can unlock it. In this case, the PIN will be unlocked and cleared.
• Budget Management for Verified Calls
A limit can be assigned to the total of all call charges for each verification code.

[Example of Verification Codes and Their Programming]

Itemised Billing
Location Code*1 Name*2 PIN*3 COS*4 Budget*6
Code for ARS*5
0001 1111 Tom Smith 1234 1 2323 5000Euro
0002 2222 John White 987654321 3 4545 3000Euro
0
: : : : : : :

*1 → 14.3 PBX Configuration—[6-3] Feature—Verification Code—Verification Code

*2 → 14.3 PBX Configuration—[6-3] Feature—Verification Code—User Name
*3 → 14.3 PBX Configuration—[6-3] Feature—Verification Code—Verification Code PIN
*4 → 14.3 PBX Configuration—[6-3] Feature—Verification Code—COS Number
*5 → 14.3 PBX Configuration—[6-3] Feature—Verification Code—Itemised Billing Code for ARS
*6 → 14.3 PBX Configuration—[6-3] Feature—Verification Code—Budget Management

PC Programming Manual References

6.6 Tool—Export
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—User Remote Operation /
Walking COS / Verification Code
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings
→ ARS Itemised Code
→ Charge Limit
14.3 PBX Configuration—[6-3] Feature—Verification Code

Feature Manual References

2.7.2 Budget Management
2.8.1 Automatic Route Selection (ARS)
5.1.1 Class of Service (COS)
5.1.6 Manager Features
6.1 Capacity of System Resources

Operating Manual References

1.2.7 Calling without Restrictions
4.1.2 Manager Programming

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2.8 Automatic Route Selection (ARS) Features

2.8 Automatic Route Selection (ARS) Features

2.8.1 Automatic Route Selection (ARS)
Description
ARS automatically selects the carrier available at the time an outgoing trunk call is made according to
preprogrammed settings. The dialled number will be checked and modified to connect the appropriate
carrier.
[Carrier Selection Procedure Flowchart]
The numbers X in the flowchart correspond to the [Programming Procedures] on the following pages.

A trunk call is made.

No (Normal Trunk Access)

Is the ARS mode ( 1 ) enabled?

Sends the telephone

Yes number to the user-
selected carrier.
Is the dialled number found in the Yes
Leading Number Exception Table ( 2 )?

No

Is the dialled number found in the No

Leading Number Table ( 3 )?

Yes
Checks the Routing Plan Table ( 4 )
to determine which carrier to use.

Is the carrier ( 6 ) found in the No

appropriate time block ( 5 )?

Yes

Is there an available No
trunk group ( 11 )?

Because all No
Yes
trunks are busy?
Modifies the dialled number by Yes
removing the digits ( 9 ) and Is normal (default)
Yes Trunk Access
following the modify commands ( 12 ).
allowed?
No
Sends the
Sends the modified number Sends a Sends a telephone number
to the trunk. busy tone. reorder tone. by the Idle Line
Access.

[Programming Procedures]
1. ARS Mode 1 Assignment

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 2.8.1 Automatic Route Selection (ARS)

It is possible to select whether ARS operates when an extension user makes a call using any Idle Line
Access method or when an extension user makes a call using any Trunk Access method. (→
2.5.5.3 Trunk Access)
→ 16.1 PBX Configuration—[8-1] ARS—System Setting—ARS Mode
2. Leading Number Exception Table 2 Assignment
Store the telephone numbers that will avoid using the ARS feature.
→ 16.6 PBX Configuration—[8-6] ARS—Leading Number Exception
2 ARS Leading Number Exception Table
Location Leading No.
No. Exception
001 033555
002 06456
: :

3. Leading Number Table 3 Assignment

Store the area codes and/or telephone numbers as leading number that will be routed by the ARS
feature. In this table, the Routing Plan (refer to "4. Routing Plan Table 4 Assignment") is selected for
each number.
The additional (remain) number of digits must be assigned only when "#", for example, is needed after a
dialled number. The "#" is added after the assigned number of digits of dialled number (excluding a
leading number).
→ 16.2 PBX Configuration—[8-2] ARS—Leading Number—Leading Number
→ 16.2 PBX Configuration—[8-2] ARS—Leading Number—Additional Number of Digits
→ 16.2 PBX Configuration—[8-2] ARS—Leading Number—Routing Plan Number
3 ARS Leading Number Table

Location Leading Additional (Remain) Routing Plan

No. No. No. of Digits Table No.
0001 039 7 1
0002 03 0 4
0003 0444 5 5
: : : :

If a dialled number matches a leading number, the number will be modified according to the
corresponding Routing Plan Table and the modified number will be sent to the trunk when the assigned
additional (remain) number of digits are dialled.
If a dialled number matches multiple leading number entries, the leading number entry with the lowest
numbered location will have priority.

[Example]
Corresponding
Dialled Number Description
Routing Plan Table No.
039-123-4567 1 "039" is found in location 0001 and seven digits
(assigned additional [remain] number of digits in
location 0001) were dialled. The Routing Plan
Table 1 is selected just after the seventh digit.

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2.8.1 Automatic Route Selection (ARS)

Corresponding
Dialled Number Description
Routing Plan Table No.
039-654-321 1 "039" is found in location 0001 and the Inter-digit
time expired before the seventh digit is received.
The Routing Plan Table 1 is selected after the
Inter-digit time expired.

→ 10.3 PBX Configuration—[2-3] System—Timers

& Counters—Dial—Extension Inter-digit
038 4 "03" is found in two locations (locations 0001 and
0002), so the PBX waits for the next digit "8".
"038" is not found in any location, then "03"
(location 0002) is selected. The Routing Plan
Table 4 is selected.

4. Routing Plan Table 4 Assignment

Arrange the time schedule as desired and store the carrier priority.

Time Table 5
As the best carrier may vary with the day of the week and the time of day, four time blocks (Time-A
through D) can be programmed for each day of the week.
→ 16.3.1 PBX Configuration—[8-3] ARS—Routing Plan Time—Time Setting

Carrier Priority 6
Assign the appropriate carrier (refer to "5. Carrier Table 7 Assignment") and their priority in each time
block. The carrier is selected in the entry order (the order in which entries are listed).
→ 16.4 PBX Configuration—[8-4] ARS—Routing Plan Priority
3 ARS Leading Number Table
Location Leading Additional Routing Plan
(Remain)
No. No. No. of Digits Table No. 4 ARS Routing Plan Table
0001 03 8 1 Routing Plan Table 1
: : : : 6 Carrier
5 Time Table
Priority 1 Priority 2 ...
SUN Time-A 9:00 1 (A telecom) 4 (D telecom) ...
Time-B 12:00 1 (A telecom) 2 (B telecom) ...
Time-C 15:00 1 (A telecom) 2 (B telecom) ...
Time-D 21:00 3 (C telecom) 1 (A telecom) ...
: : : : : ...
SAT Time-A 9:00 3 (C telecom) 2 (B telecom) ...
Time-B 12:00 3 (C telecom) 1 (A telecom) ...
Time-C 15:00 3 (C telecom) 1 (A telecom) ...
Time-D 21:00 3 (C telecom) 2 (B telecom) ...

5. Carrier Table 7 Assignment

A specified number of carriers can be programmed. Assign the following items for each Carrier Table:
→ 16.5.1 PBX Configuration—[8-5-1] ARS—Carrier—Main Setting
Carrier Name 8 : Assign the carrier name.
→ 16.5.1 PBX Configuration—[8-5-1] ARS—Carrier—Main Setting—Carrier Name
Removed Number of Digits 9 : Assign the number of digits to remove from the beginning of the user-
dialled number.
→ 16.5.1 PBX Configuration—[8-5-1] ARS—Carrier—Main Setting—Removed Number of Digits
Carrier Access Code 10 : Assign the code to access the carrier.

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 2.8.1 Automatic Route Selection (ARS)

→ 16.5.1 PBX Configuration—[8-5-1] ARS—Carrier—Main Setting—Carrier Access Code

Trunk Group 11 : Assign the trunk groups which connect to each carrier and the priority in which they
are selected.
When using Web Maintenance Console, trunk groups can be assigned to a carrier using an on/off
setting for each trunk group. They can also be given a priority setting (1–4) which decides the order they
are searched when seizing a line. If there are no available lines in the trunk groups set to priority 1–4,
the other trunk groups set to on are searched in the order of smallest number first.

[Example]
Priority Setting Trunk Groups
Carrier Searching Order
1 2 3 4 Set to On

ABC 9 3 1 7 5, 7, 9, 11 9 → 3 → 1 → 7 → 5 → 11
XYZ 12 4 None None 6, 10 12 → 4 → 6 → 10

→ 16.5.2 PBX Configuration—[8-5-2] ARS—Carrier—TRG Priority—TRG Priority

→ 16.5.2 PBX Configuration—[8-5-2] ARS—Carrier—TRG Priority—TRG 01–TRG 128
Modify Command 12 : Assign the commands to modify the dialled number to access the carrier.
→ 16.5.1 PBX Configuration—[8-5-1] ARS—Carrier—Main Setting—Modify Command
CLIP Table No. 16 : Assign the CLIP number for the carrier. CLIP numbers are assigned according to the
CLIP Table No. assigned for the carrier.
→ 16.5.1 PBX Configuration—[8-5-1] ARS—Carrier—Main Setting—CLIP Table No.

[Command Explanation]
Command Description
Number Add the number.
C Add the Carrier Access code.
P Analogue Line: Insert a pause.
ISDN/T1/E1 Line: Insert a pause and change to tone (DTMF)
signal.
A Add the Authorisation code for a tenant ( 13 ).
G Add the Authorisation code for a trunk group ( 14 ).
I Add the Itemised Billing code ( 15 ).
H Add the dialled number after the digits are removed (Home
position).

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2.8.1 Automatic Route Selection (ARS)

[Programming Example]

7 Carrier Table 1 2 Ext. 1001

8 Carrier Name A telecom B telecom CLIP No. CLIP
9 Removed Number of Digits 6 0 1 0123456789
10 Carrier Access Code 0077 0088 2 0234567861
11 Trunk Group 1, 2, 3 1, 2 3 0356894526
12 Modify Command CH#12 CH :
16 CLIP Table No. 2 1 8 0856325889

[Example]
Dialled number: 0123456789
(Trunk Access no. is ignored.)
Modification:
12
012345 6789 0077 6789 #12
H

9 Remove 6 digits. #12 Add the number.

C Add the Carrier Access
code ( 10 ).
16 CLIP: 0234567861

Note
• If the ARS Itemised Code is set to be sent as a CLIP with ARS, the following settings are
prioritised and used as the CLIP.
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 1—ARS
Itemised Code
• CLIP Table No.1 is set automatically according to the following settings.
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—CLIP—CLIP ID
6. Optional Assignment

Authorisation Code for a Tenant 13

An Authorisation code can be assigned for each carrier and each tenant.
→ 16.5.3 PBX Configuration—[8-5-3] ARS—Carrier—Authorisation Code for Tenant

Authorisation Code for a Trunk Group 14

An Authorisation code can be assigned for each trunk group and each carrier.
→ 16.7 PBX Configuration—[8-7] ARS—Authorisation Code for TRG

Itemised Billing Code 15

An Itemised Billing code can be assigned for each extension and for each verification code.
If a call is not made from an extension (e.g., DISA or TIE) and no verification code is used, the Itemised
Billing code assigned in the location 1 of the verification code will be used.
→ 8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 1—ARS
Itemised Code
→ 14.3 PBX Configuration—[6-3] Feature—Verification Code—Itemised Billing Code for ARS

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 2.8.1 Automatic Route Selection (ARS)

Conditions
CAUTION
The software contained in the ARS feature to allow user access to the network must be upgraded to
recognise newly established network area codes and exchange codes as they are placed into service.
Failure to upgrade the on-premise PBXs or peripheral equipment to recognise the new codes as they
are established will restrict the customer and users of the PBX from gaining access to the network and
to these codes.
KEEP THE SOFTWARE UP TO DATE WITH THE LATEST DATA.
• Dialled Number on SMDR
It is possible to choose to print either the user-dialled number or the modified number on SMDR through
system programming. (→ 2.22.1.1 Station Message Detail Recording (SMDR))
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR Options—Option—ARS Dial
• ARS Data Download/Upload
It is possible to download or upload the following ARS data to the PBX using PC programming:
– 2 ARS Leading Number Exception Table
– 3 ARS Leading Number Table
– 4 ARS Routing Plan Table
→ 6.5 Tool—Import
→ 6.6 Tool—Export

This is useful when a carrier has changed the call charge, and the updated data can be used for multiple
customers.
• A TRS/Barring check is done before ARS is applied. (→ 2.7.1 Toll Restriction (TRS)/Call Barring
(Barring))

PC Programming Manual References

6.5 Tool—Import
→ ARS - Leading Digit
→ ARS - Except Code
→ ARS - Routing Plan
6.6 Tool—Export
10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Dial—Extension
Inter-digit
10.9 PBX Configuration—[2-9] System—System Options—Option 3—Dial Tone—Dial Tone for ARS
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 1—ARS Itemised Code
14.3 PBX Configuration—[6-3] Feature—Verification Code—Itemised Billing Code for ARS
16 PBX Configuration—[8] ARS
19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR Options—Option—ARS Dial

Feature Manual References

6.1 Capacity of System Resources

Feature Manual 123

2.9 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Features

2.9 Primary Directory Number (PDN)/Secondary

Directory Number (SDN) Features
2.9.1 Primary Directory Number (PDN)/Secondary Directory
Number (SDN) Extension
Description
Primary Directory Number (PDN) buttons and Secondary Directory Number (SDN) buttons are ideal for use
between bosses and secretaries. When a call (intercom or trunk) arrives at a PDN button on the boss’s
extension, the call will ring and the LED of the SDN button will flash at the secretary’s extension as well,
indicating that an incoming call is arriving at the boss’s extension. In addition, caller information (e.g., Caller
ID) of the incoming call will appear on the secretary’s extension. The secretary can answer the call for the
boss by simply pressing the SDN button. Delayed ringing can be set for a PDN or SDN button.
A secretary can hold a call answered on the SDN button, and the boss can retrieve the held call simply by
pressing the PDN button, like when answering a call with an S-CO button. In addition, a secretary can
transfer calls from an SDN button or other button (e.g., S-CO button) to the boss’s extension with a simple
operation, like when using a DSS button.
An extension can have several SDN buttons, each registered to a different boss’s extension. However, only
one SDN button can be registered for a single boss at each extension. An extension can have up to eight
PDN buttons. PDN buttons can simplify the use of an extension because both intercom and trunk calls can
be made and received at a PDN button.

Making Calls with an SDN Button

When Standard SDN Key mode is assigned to an SDN extension (secretary) through COS programming,
SDN extensions (secretaries) can make calls for PDN extensions (bosses) on the SDN button. For example,
a boss can ask a secretary to make a call and put the call on hold, after which, the boss can retrieve the
held call.
Through COS programming, it is possible to allow an SDN extension to make calls using the COS of the
PDN extension. All other settings that are available when using the Walking COS feature are also applied
(→ 2.7.5 Walking COS).
SDN Direct Dial
An SDN extension can call a PDN extension or transfer a call to a PDN extension using an SDN button.
In this case:
– Only the PDN extension rings (i.e., other SDN extensions do not ring).
– The delayed ringing and DND settings of the PDN extension are ignored.

Depending on the mode selected through COS programming, SDN Direct Dial is performed in one of two
ways, as follows:
– Enhanced DSS Key mode: pressing the SDN button once.
– Standard SDN Key mode: pressing the SDN button twice (a dial tone is heard the first time the SDN
button is pressed).
Calls answered using the SDN button can be transferred to the PDN extension by simply pressing the SDN
button once, regardless of the mode.

LED Indication
The LED patterns and the corresponding status of PDN and SDN buttons are as follows:
Light Pattern PDN Button Status SDN Button Status
Off This extension is idle. The corresponding PDN extension is idle.

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 2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension

Light Pattern PDN Button Status SDN Button Status

Green on The extension is on a call using the The extension is on a call using the SDN
PDN button. button.
Slow green A call is on hold using the PDN button. A call is on hold using the SDN button.
flashing
Moderate • A call on a PDN button is on • A call answered using the SDN button is
green Exclusive Call Hold or consultation on Exclusive Call Hold or consultation
flashing hold. hold.
• The PDN extension is adding a • The SDN extension is adding a member
member to a conference or using the to a conference, or using the line for an
line for an Unattended Conference, Unattended Conference.
on a PDN button.
Rapid green An incoming call is arriving at this Receiving Hold Recall or automatic
flashing extension. callback ringing from a call answered
using the SDN button.
Red on A corresponding SDN extension is: The corresponding PDN extension or
another corresponding SDN extension is:
• on a call.
• holding the line using Exclusive Call • on a call.
Hold or consultation hold. • holding the line using Exclusive Call
Hold or consultation hold.
• adding a member to a conference.
• using the line for an Unattended • adding a member to a conference.
Conference. • using the line for an Unattended
Conference.
• receiving Hold Recall or automatic
callback ringing. • receiving an incoming call directed only
to the PDN extension (e.g., callback
ringing).
Slow red A call is on hold by a corresponding A call is on hold by the corresponding
flashing SDN extension. PDN extension or another corresponding
SDN extension.
Rapid red A call is arriving at an Incoming Call The corresponding PDN extension is
flashing Distribution (ICD) group in Ring receiving an incoming call.
Distribution method that this extension
is a member of.

When multiple calls are on a PDN extension, the LED pattern that appears on the corresponding SDN
buttons is displayed according to the following priority:
Receiving an incoming call → holding a call → on a call → idle
For example, if a PDN extension receives an incoming call while on a call, the LEDs on the corresponding
SDN extensions will show the incoming call.
However, if an SDN extension is handling a call using the SDN button (e.g., on a call, has a call on hold,
etc.), the status of that call will be displayed on the SDN button, regardless of the call status of the PDN
extension.

Example of a Secretary Handling Calls for Multiple Bosses

The following example shows the LED patterns of the PDN and SDN buttons of each extension and how
calls can be handled.

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2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension

1. A Call From 111-1111 Arrives at Ext. 101

Extn. 101 (Boss) Extn. 102 (Boss) Extn. 103 (Secretary)
PDN buttons SDN buttons PDN buttons SDN buttons PDN buttons SDN buttons
(Off) (Off) (Off) (Off) (Off) (Off)
(Extn. 103) (Extn. 103) (Extn. 102)
(Off) (Off) (Off)
(Off) (Rapid Red (Rapid Red
(Extn. 102) (Extn. 101) Flashing) (Extn. 101) Flashing)
(Rapid Green (Off) (Off)
Flashing)

2. The Call From 111-1111 is Answered by Ext. 103

Extn. 101 (Boss) Extn. 102 (Boss) Extn. 103 (Secretary)
PDN buttons SDN buttons PDN buttons SDN buttons PDN buttons SDN buttons
(Off) (Off) (Off) (Off) (Off) (Off)
(Extn. 103) (Extn. 103) (Extn. 102)
(Off) (Off) (Off)
(Off) (Red On) (Green On)
(Extn. 102) (Extn. 101) (Extn. 101)
(Red On) (Off) (Off)

3. The Call From 111-1111 is On Hold by Ext. 103

Extn. 101 (Boss) Extn. 102 (Boss) Extn. 103 (Secretary)
PDN buttons SDN buttons PDN buttons SDN buttons PDN buttons SDN buttons
(Off) (Off) (Off) (Off) (Off) (Off)
(Extn. 103) (Extn. 103) (Extn. 102)
(Off) (Off) (Off)
(Off) (Slow Red (Slow Green
(Extn. 102) (Extn. 101) Flashing) (Extn. 101) Flashing)
(Slow Red (Off) (Off)
Flashing)

4. The Call Held by Ext. 103 is Answered by Ext. 101

Extn. 101 (Boss) Extn. 102 (Boss) Extn. 103 (Secretary)
PDN buttons SDN buttons PDN buttons SDN buttons PDN buttons SDN buttons
(Off) (Off) (Off) (Off) (Off) (Off)
(Extn. 103) (Extn. 103) (Extn. 102)
(Off) (Off) (Off)
(Off) (Red On) (Red On)
(Extn. 102) (Extn. 101) (Extn. 101)
(Green On) (Off) (Off)

5. A Call From 222-2222 Arrives at Ext. 101

Extn. 101 (Boss) Extn. 102 (Boss) Extn. 103 (Secretary)
PDN buttons SDN buttons PDN buttons SDN buttons PDN buttons SDN buttons
(Off) (Off) (Off) (Off) (Off) (Off)
(Extn. 103) (Extn. 103) (Extn. 102)
(Rapid Green (Off) (Off)
Flashing) (Off) (Rapid Red (Rapid Red
(Extn. 102) (Extn. 101) Flashing) (Extn. 101) Flashing)
(Green On) (Off) (Off)

6. A Call From 333-3333 Arrives at Ext. 102

Extn. 101 (Boss) Extn. 102 (Boss) Extn. 103 (Secretary)
PDN buttons SDN buttons PDN buttons SDN buttons PDN buttons SDN buttons
(Off) (Off) (Off) (Off) (Off) (Rapid Red
(Extn. 103) (Extn. 103) (Extn. 102) Flashing)
(Rapid Green (Off) (Off)
Flashing) (Rapid Red (Rapid Red (Rapid Red
(Extn. 102) Flashing) (Extn. 101) Flashing) (Extn. 101) Flashing)
(Green On) (Rapid Green (Off)
Flashing)

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 2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension

7. The Call From 333-3333 is Answered by Ext. 103

Extn. 101 (Boss) Extn. 102 (Boss) Extn. 103 (Secretary)
PDN buttons SDN buttons PDN buttons SDN buttons PDN buttons SDN buttons
(Off) (Off) (Off) (Off) (Off) (Green On)
(Extn. 103) (Extn. 103) (Extn. 102)
(Rapid Green (Off) (Off)
Flashing) (Red On) (Rapid Red (Rapid Red
(Extn. 102) (Extn. 101) Flashing) (Extn. 101) Flashing)
(Green On) (Red On) (Off)

8. The Call From 333-3333 is Transferred by Ext. 103 to Extension 102

Extn. 101 (Boss) Extn. 102 (Boss) Extn. 103 (Secretary)
PDN buttons SDN buttons PDN buttons SDN buttons PDN buttons SDN buttons
(Off) (Off) (Off) (Off) (Off) (Moderate
(Extn. 103) (Extn. 103) (Extn. 102) Green Flashing)
(Rapid Green (Off) (Off)
Flashing) (Red On) (Rapid Red (Rapid Red
(Extn. 102) (Extn. 101) Flashing) (Extn. 101) Flashing)
(Green On) (Rapid Green (Off)
Flashing)

Conditions
[General]
• A flexible button of a PT and a PS can be customised as a PDN or SDN button. A flexible button on a
DSS Console can be customised as an SDN button.
• An extension can have up to eight PDN buttons.
• If none of an extension’s PDN buttons are idle, the extension will not receive incoming calls, including Call
Waiting. Therefore, it is strongly recommended for PDN extensions to have at least three PDN buttons.
• Through COS programming, it is possible to select whether extensions can create SDN buttons on their
own extensions using PT programming.
• Up to eight different extensions can assign SDN buttons corresponding to the same PDN extension.
• When a PDN extension has an idle CO button or ICD Group button, calls will arrive on the following
buttons according to the following priority:
– Incoming intercom calls to an ICD group: ICD Group button → PDN button
– Incoming trunk calls: S-CO button → G-CO button → L-CO button → PDN button
– Incoming trunk calls to an ICD group: ICD Group button → S-CO button → G-CO button → L-CO
button → PDN button
• When multiple calls of the same status (e.g., on hold) are on a PDN extension, the status of the oldest
call will be displayed on the corresponding SDN extensions. For example, if a PDN extension has two
calls ringing, an SDN extension will answer the call that arrived at the PDN extension first, when pressing
the SDN button.
• When a PDN extension is a member of an ICD group in Ring Distribution method, and an incoming call
arrives at the ICD group, the incoming call status will not appear on the LEDs of the corresponding SDN
extensions (→ 2.2.2.1 Incoming Call Distribution Group Features—SUMMARY).
• The Sub device designated by the User Container cannot perform the PDN/SDN key operation.
• When a PDN key is configured for the main device designated by the User Container, the incoming call
for that user can only arrive on the specified main device and public call destinations. The call will not
arrive on the sub devices or wireless devices. In addition, the PDN/SDN key operation of the sub devices
and wireless devices will be ignored.

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2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension

• Only the extension number of the user (=main device) can be specified as additional information
(extension number) when configuring the SDN key settings.
There are cases where devices other than the main device are configured when deleting a user or
changing the device settings of the User Container, however, in such cases the SDN button will not
function.
• No PDN status display
• No incoming call display to the SDN button possible
• Unable to answer by pressing the SDN button
• Unable to call by pressing the SDN button
• If none of an extension’s PDN buttons are idle, DSS buttons of other extensions registered to the PDN
extension will turn on red.
• Ring Tone Pattern
Through system programming, each extension can set ring tone patterns for PDN buttons. Ring tone
patterns can be assigned separately for each SDN button.
• Outgoing Line Preference
When "PDN" is selected as the outgoing line preference, outgoing calls will originate on the first available
PDN button (→ 2.5.5.2 Line Preference—Outgoing).
• Incoming Line Preference
Through system programming, it is possible for only incoming calls arriving at PDN buttons to be
answered simply by going off-hook, by selecting "PDN" as the incoming line preference (→ 2.4.2 Line
Preference—Incoming). This prohibits calls that arrive on non-PDN buttons (e.g., an SDN button) to be
answered when going off-hook.
• OHCA/Whisper OHCA
A PDN extension cannot receive OHCA or Whisper OHCA unless the call is made using a corresponding
SDN button (→ 2.10.4.3 Off-hook Call Announcement (OHCA), → 2.10.4.4 Whisper OHCA).
• Alternate Calling—Ring/Voice
It is not possible to temporarily change the called party’s preset call receiving method (ring tone or voice)
when calling a PDN extension, unless the call is made using a corresponding SDN button
(→ 2.5.3 Intercom Call).
• Through system programming, it is possible to force an extension to become idle (the SP-PHONE button
light will turn off) when a speakerphone call using a PDN/SDN button is put on hold using CTI.

[Delayed Ringing]
• The same delayed ringing setting is applied to all PDN buttons on an extension. Delayed ringing can be
assigned separately for each SDN button.
• Through system programming, it is possible to select whether caller information (such as Caller ID) is
shown immediately on a PS when a call is received while delayed ringing is set.
• Caller information (such as Caller ID) is not shown immediately on a PT when a call is received while
delayed ringing is set.
• SDN buttons can be set to not ring (only flash) for incoming calls. However, this setting is not available for
PDN buttons.
• The forward no answer timer starts when a PDN extension starts ringing.

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—PDN/SDN
10.8.3 PBX Configuration—[2-8-3] System—Ring Tone Patterns—Call from Others—Extension—Ring Tone
Pattern Plan1—8

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 2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension

10.9 PBX Configuration—[2-9] System—System Options—Option 4—System Wireless—SDN Delayed

Ringing with LCD
10.9 PBX Configuration—[2-9] System—System Options—Option 6 (CTI)—CTI Hold—Forced Idle when
Hold by PDN/SDN Key
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 9—PDN Delayed
Ringing
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Parameter Selection(for SDN)
→ Extension Number(for SDN)
→ Optional Parameter (Ringing Tone Type Number)(for Loop CO, Single CO, Group CO, ICD Group,
SDN)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Parameter Selection(for SDN)
→ Extension Number(for SDN)
12.1 PBX Configuration—[4-1] Extension—DSS Console

Feature Manual References

2.2.2.2 Group Call Distribution
2.21.3 LED Indication
5.1.1 Class of Service (COS)
6.1 Capacity of System Resources

Operating Manual References

1.4.2 Holding a Call
1.5.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension
1.2.7 Calling without Restrictions—Using Your Calling Privileges at Another Extension (Remote COS
Access)

Feature Manual 129

2.10 Busy Line/Busy Party Features

2.10 Busy Line/Busy Party Features

2.10.1 Automatic Callback Busy (Camp-on)
Description
If the destination or line is busy when a call is made, an extension user can set the Automatic Callback Busy
feature. The PBX will monitor the status of the destination or trunk and, when it becomes available, will send
a callback ringing to the calling extension to inform the user. After the extension answers the callback
ringing, the previously dialled extension number is automatically redialled, or the trunk is automatically
seized.

Conditions
• If the callback ringing is not answered within 10 seconds, the callback is cancelled.
• If the extension hears a busy tone before dialling the telephone number, only the trunk or trunk group is
reserved. After answering the callback ringing, the extension should dial the telephone number.
• An extension can set only one Automatic Callback Busy. The last setting is effective.
• Multiple extension users can set this feature to one trunk simultaneously.
However, a maximum of four extension users can set this feature to one extension.
Callback ringing will be sent to extensions in the order that the feature was set. In other words, the
extension that set the feature first will receive a callback ringing first.
• This feature cannot be used for calls to the Unified Messaging system, a SIP extension, or an ISDN
extension.
• The automatic call back busy (camp-on) feature will not work while the following features are working on
your phone:
– Unattended conference
– Call hold

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Automatic Callback Busy
Cancel
10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature—Automatic
Callback Busy

Operating Manual References

1.2.4 When the Dialled Line is Busy or There is No Answer

2.10.2 Executive Busy Override

Description
Allows an extension user to interrupt an existing call to establish a three-party conference call.

Executive Busy Override Deny:

It is possible for extension users to prevent their calls from being intercepted by another extension user.

One-touch Executive Busy Override:

Extension users can perform Executive Busy Override by simply pushing the S-CO button of a call in
progress without entering a feature number. This feature can be enabled through system programming.

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 2.10.2 Executive Busy Override

Caller Information before Executive Busy Override:

A PT/PS user, such as a boss, can confirm the caller information of a call to another extension using a DSS
button before performing Executive Busy Override.
After pressing the corresponding DSS button, and while hearing the busy tone, the user can check the
caller’s information on the LCD. Then, the user can interrupt the call (Executive Busy Override), if
necessary.

Conditions
[General]
• COS programming determines the extension users who can use Executive Busy Override and set
Executive Busy Override Deny.
• This feature does not work when the busy extension is in one of the following conditions:
a. Executive Busy Override Deny or Data Line Security (→ 2.11.5 Data Line Security) has been set.
b. While being monitored by another extension (→ 2.10.3 Call Monitor).
c. While receiving OHCA (→ 2.10.4.3 Off-hook Call Announcement (OHCA), → 2.10.4.4 Whisper
OHCA).
d. During a Conference call (→ 2.14 Conference Features).
e. During a doorphone call (→ 2.18.1 Doorphone Call).
f. While Live Call Screening (LCS) or Two-way Record is activated (→ 3.2.2.19 Live Call Screening
(LCS) and 3.2.2.34 Two-way Record/Two-way Transfer).
g. During Consultation Hold.
• This feature is not available for a trunk-to-trunk call via DISA.

[One-touch Executive Busy Override]

• Automatic Callback Busy cannot be used on trunks that have this feature enabled (→ 2.10.1 Automatic
Callback Busy (Camp-on)).

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Executive Override Deny
Set / Cancel
10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature—Executive Busy
Override
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Executive
→ Executive Busy Override
→ Executive Busy Override Deny
10.9 PBX Configuration—[2-9] System—System Options—Option 1—PT Operation—One-touch Busy
Override by SCO key
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 3—Executive Override
Deny

Feature Manual References

5.1.1 Class of Service (COS)

Operating Manual References

1.2.4 When the Dialled Line is Busy or There is No Answer
1.9.7 Preventing Other People from Joining Your Conversation (Executive Busy Override Deny)

Feature Manual 131

2.10.3 Call Monitor

2.10.3 Call Monitor

Description
Allows an extension user to listen to a busy extension user’s existing conversation. The user can hear the
conversation, but the user’s voice is not heard. If desired, interrupting the call to establish a three-party
conference call is available.

Caller Information before Call Monitor:

A PT/PS user, such as a boss, can confirm the caller information of a call to another extension using a DSS
button before performing Call Monitor.
After pressing the corresponding DSS button, and while hearing the busy tone, the user can check the
caller’s information on the LCD. Then, the user can monitor the call, if necessary.

Conditions
• COS programming determines extension users who can use this feature.
• This feature is available only when the busy extension is in a conversation with another extension or
outside party.
• This feature does not work when the busy extension is in one of the following conditions:
a. Executive Busy Override Deny (→ 2.10.2 Executive Busy Override) or Data Line Security
(→ 2.11.5 Data Line Security) has been set.
b. While receiving OHCA (→ 2.10.4.3 Off-hook Call Announcement (OHCA), → 2.10.4.4 Whisper
OHCA).
c. During a Conference call with 4 or more participants (→ 2.14 Conference Features).
d. During a doorphone call (→ 2.18.1 Doorphone Call).
e. While Live Call Screening (LCS) is activated (→ 3.2.2.19 Live Call Screening (LCS)).
f. During Consultation Hold.
g. When using wireless devices linked to the User Container (→ 2.29 User Container).
• This feature stops when the busy extension user presses the following buttons during a conversation
(→ 2.21.1 Fixed Buttons and 2.21.2 Flexible Buttons):
– FLASH/RECALL button
– HOLD button
– TRANSFER button
– CONF (Conference) button
– DSS button
– EFA button
– Two-way Record button
– Two-way Transfer button
– One-touch Two-way Transfer button
– Voice Mail (VM) Transfer button
• Monitoring a Call that is Being Recorded
When a user monitors a call that is being recorded, the user's voice will not be recorded. When an
intercom call is being recorded, call monitoring can be performed on either extension.
Executive Busy Override can also be performed on a call that is being recorded. In this case, the user's
voice will be recorded.

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 2.10.4 Second Call Notification to Busy Extension

When a member of the original call leaves the conference created by Executive Busy Override, recording
will stop and the remaining extensions will operate as follows:
– If Automatic Two-way Recording for Manager has been set, recording will start again according to
those settings.
– If Automatic Two-way Recording for Manager has not been set, one of the parties can press the Two-
way Record button to start a new recording.
A call being recorded between 2 extensions cannot be monitored if the user performing Call Monitoring is
connected via QSIG (TIE).
• Monitoring a Conference
When a user monitors a three-party conference, the voices of all 3 participants can be heard.
Executive Busy Override can also be performed on a three-party conference. In this case, a four-party
conference will be established.
Call monitoring cannot be performed on a conference on hold.
Call monitoring cannot be performed on a three-party conference where the originator of the conference
is on an outside line (e.g., when calling from a cellular phone).
A 3-party conference call cannot be monitored if the user performing Call Monitoring is connected via
QSIG (TIE).

PC Programming Manual References

10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Executive—Call Monitor
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings
→ Option 2—Data Mode
→ Option 3—Executive Override Deny

Feature Manual References

2.14 Conference Features
3.2.1.4 Automatic Two-way Recording for Manager
3.2.2.34 Two-way Record/Two-way Transfer
5.1.1 Class of Service (COS)

Operating Manual References

1.2.4 When the Dialled Line is Busy or There is No Answer
1.15.1 Cellular Phone Features

2.10.4 Second Call Notification to Busy Extension

2.10.4.1 Second Call Notification to Busy Extension—SUMMARY
Description
When attempting to call a busy extension (ringing or having a conversation), an extension user can send a
Call Waiting indication to the busy extension (Call Waiting). The notification receiving method depends on
the called extension’s personal setting and the telephone type:
Notification Receiving Method Description & Reference
Call Waiting Tone Sends the Call Waiting tone to the busy extension.

→ 2.10.4.2 Call Waiting Tone

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2.10.4 Second Call Notification to Busy Extension

Notification Receiving Method Description & Reference

Off-hook Call Announcement Talk with the busy extension using the built-in speaker and
(OHCA) microphone of the called extension while the existing call is
made using the handset.

→ 2.10.4.3 Off-hook Call Announcement (OHCA)

Whisper OHCA Send a spoken message to a busy extension that will be heard
directly by only the called extension user, through the handset,
without interrupting the ongoing conversation.

→ 2.10.4.4 Whisper OHCA

Conditions
• Each extension user can choose to receive Call Waiting tone, OHCA, Whisper OHCA, or none of these.
• OHCA and Whisper OHCA are enabled or disabled by the COS of the calling extension.
• OHCA and Whisper OHCA do not work for some telephone types. In such cases, the Call Waiting tone
will be sent to the called extension.
Calling Called Extension’s Call Waiting Mode
Extension’s
OFF ON
OHCA COS
Mode Cancel Call Waiting Tone OHCA Whisper OHCA
Disable Call Waiting Call Waiting tone Call Waiting tone Call Waiting tone
disabled
Enable Call Waiting Call Waiting tone OHCA (or Call Whisper OHCA
disabled Waiting tone) (or Call Waiting
tone)

• The notification receiving methods (Call Waiting tone, OHCA, and Whisper OHCA) are available only
when the called extension is having a conversation with another party. If the called party is not yet
connected with the other party (e.g., still ringing, on hold, etc.), the calling extension will hear a ringback
tone and will be kept waiting until the called extension becomes available to receive the call waiting
notification.
• If none of these notification receiving methods (Call Waiting tone, OHCA, or Whisper OHCA) are set at
the called party’s extension, the caller will hear a reorder tone.

Feature Manual References

2.1.3.3 Call Waiting

Operating Manual References

1.2.4 When the Dialled Line is Busy or There is No Answer
1.9.3 Receiving Call Waiting (Call Waiting/Off-hook Call Announcement [OHCA]/Whisper OHCA)

2.10.4.2 Call Waiting Tone

Description
When an extension user attempts to call a busy extension (ringing or having a conversation), the Call
Waiting tone can be sent to the called extension to let him know another call is waiting.

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 2.10.4 Second Call Notification to Busy Extension

Conditions
• This feature only works if the called extension has activated Call Waiting. If it is activated, the calling
extension will hear a ringback tone.
• Call Waiting tone can be selected (Tone 1 or Tone 2) through personal programming (Call Waiting Tone
Type Selection).
• When the headset mode is on, you can choose whether the call waiting tone is heard from the speaker
phone of the telephone or the earpiece of the headset. However, this setting is only available for terminals
that support call waiting tone path switching (KX-DT521, KX-DT543, KX-DT546, KX-NT553, and
KX-NT556).

PC Programming Manual References

9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property—Option—C.Waiting with
Headset
9.28 PBX Configuration—[1-1] Configuration—Slot—Legacy-GW1/Legacy-GW2
10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—Caller ID—Visual Caller ID
Display
10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature
→ BSS / OHCA / Whisper OHCA / DND Override
→ BSS / OHCA / Whisper OHCA / DND Override-2
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 4—Call Waiting Tone
Type

Operating Manual References

1.9.3 Receiving Call Waiting (Call Waiting/Off-hook Call Announcement [OHCA]/Whisper OHCA)
3.1.2 Settings on the Programming Mode

2.10.4.3 Off-hook Call Announcement (OHCA)

Description
An extension user can talk with a busy extension through the built-in speaker and microphone of the called
party’s PT. If the existing call is using a handset, a second conversation is made using the speakerphone
and microphone so that the called extension can talk to both parties.

Conditions
• COS programming determines which extensions can use this feature.
• This feature is available when the called extension uses one of the following telephones:
– KX-DT333, KX-DT343, KX-DT346, KX-DT521, KX-DT543, KX-DT546
These telephones can be connected only through a legacy gateway. They cannot be connected directly to
a KX-NSX Series PBX. (→ 5.3.1 Stacking Connection)
• The OHCA feature cannot be used in the following cases:
a. COS or called extension’s telephone type is not available for this feature.
b. The called extension (DPT) is in the Digital XDP connection.
The Call Waiting tone is sent to the called extension. (→ 2.10.4.2 Call Waiting Tone)
When the Called Extension is IP-PT, the device will enter BSS Mode without starting OHCA.
• While an extension is receiving OHCA, if the extension user places the current trunk call on hold or
transfers the current intercom call or trunk call, OHCA will become disabled and the calling extension will
start to hear a ringback tone.

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2.10.4 Second Call Notification to Busy Extension

• While an extension is receiving OHCA, if the extension user places the current intercom call on hold, the
called extension can talk to the calling extension through the handset.

PC Programming Manual References

10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature
→ BSS / OHCA / Whisper OHCA / DND Override
→ BSS / OHCA / Whisper OHCA / DND Override-2
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Assistant—OHCA / Whisper
OHCA

Feature Manual References

5.1.1 Class of Service (COS)

Operating Manual References

1.9.3 Receiving Call Waiting (Call Waiting/Off-hook Call Announcement [OHCA]/Whisper OHCA)

2.10.4.4 Whisper OHCA

Description
An extension user can send a spoken message to a busy extension that will be heard directly by only the
called extension user, through the handset, without interrupting the ongoing conversation. The caller cannot
hear the ongoing conversation or the called extension user’s reply, unless the called extension user puts the
current party on hold and switches to the waiting caller.

Conditions
• COS programming determines which extensions can use this feature.
• This feature is available when the calling and called extension use one of the following telephones:
– KX-DT300 series
– KX-DT500 series
– IP-PT
• If the Whisper OHCA feature cannot be used due to COS or telephone type, the Call Waiting tone will be
sent to the called extension. (→ 2.10.4.2 Call Waiting Tone)
• To receive Whisper OHCA on an IP-PT, the preferred codec must be either G.711 or G.729A. When an
extension user is on a call using the G.722 codec and receives Whisper OHCA, he will hear the Call
Waiting tone instead. (→ 2.10.4.2 Call Waiting Tone)
• If the called extension does not use a KX-DT300 or KX-DT500 series telephone, or an IP-PT, but forces
Whisper OHCA, the announcement may be heard by the other party.
• It is possible to enable Whisper OHCA on any telephone. However, it may not work properly. (e.g., The
voice may be heard by the other party.)
• When a non-IP extension is connected to a non-IP trunk and the extension receives Whisper OHCA,
Whisper OHCA will not function. The extension user will hear the Call Waiting tone instead.
(→ 2.10.4.2 Call Waiting Tone)
• While an extension is receiving Whisper OHCA, if the extension user places the current trunk call on hold
or transfers the current intercom call or trunk call, Whisper OHCA will become disabled and the calling
extension will start to hear a ringback tone.
• While an extension is receiving Whisper OHCA, if the extension user places the current intercom call on
hold, the called extension can talk to the calling extension through the handset.

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 2.10.4 Second Call Notification to Busy Extension

PC Programming Manual References

9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property—Option— IP Codec
Priority
10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature
→ BSS / OHCA / Whisper OHCA / DND Override
→ BSS / OHCA / Whisper OHCA / DND Override-2
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Assistant—OHCA / Whisper
OHCA

Feature Manual References

5.1.1 Class of Service (COS)

Operating Manual References

1.9.3 Receiving Call Waiting (Call Waiting/Off-hook Call Announcement [OHCA]/Whisper OHCA)

Feature Manual 137

2.11 Conversation Features

2.11 Conversation Features

2.11.1 Hands-free Operation
Description
A PT user can talk to another party without lifting the handset. Pressing specific buttons (e.g., REDIAL)
automatically activates hands-free mode.

Conditions
• PTs with the MONITOR Button
PTs with the MONITOR button can only dial in hands-free mode and cannot be used for hands-free
conversations.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Tone Length—
Reorder Tone for PT Hands-free
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 4—LCS Answer Mode

2.11.2 Off-hook Monitor

Description
A PT user can let others listen to the user’s conversation through the built-in speaker, during a conversation
using the handset.

Conditions
• Capable Telephones
– KX-DT300 series
– KX-DT500 series
– KX-NT series
• To enable this feature, system programming is required. If disabled, hands-free conversation is performed
instead.

Operating Manual References

1.4.7 Letting Other People Listen to the Conversation (Off-hook Monitor)

2.11.3 Mute
Description
During a conversation, a PT user can disable the speaker microphone or the handset microphone to consult
privately with others while listening to the other party on the phone through the built-in speaker or the
handset receiver. The user can hear the other party’s voice during Mute, but cannot be heard.

Conditions
• This feature is available with all PTs that have the AUTO ANS/MUTE button.

Operating Manual References

1.4.6 Mute

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 2.11.4 Headset Operation

2.11.4 Headset Operation

Description
This PBX allows the use of headset-compatible PTs. A PT user can talk to another party without lifting the
handset. This feature is also known as Handset/Headset Selection.
For connection and operation, refer to the Operating Instructions for the headset.

Conditions
• Hardware Requirement: An optional headset
• If headset mode is on, pressing the SP-PHONE button activates the headset, not the built-in speaker.
• To set headset mode on a DPT or IP-PT, use personal programming (Headset Operation) or press the
Headset button. To set headset mode on an APT, use the handset/headset selector provided on the set
and/or on the headset.
• Headset Button
A flexible button on a DPT or IP-PT can be customised as a Headset button. It is possible to assign a
Headset button to a flexible button on an APT, but the button will not function.
• Answer/Release Button
A flexible button can be customised as an Answer button or a Release button. Such buttons are useful for
headset operation. It is possible to answer an incoming call by pressing an Answer button. While hearing
the Call Waiting tone during a conversation, pressing an Answer button enables one to answer the
second call by placing the current call on hold. Pressing a Release button enables one to disconnect the
line during or after conversation, or to complete a Call Transfer.
• It is possible to switch from headset mode to hands-free mode or vice versa during a conversation by
pressing the Headset button.
• Headset users cannot use the following features:
– Automatic Redial (→ 2.6.3 Last Number Redial)
– Receiving OHCA (DPT through a stacking connection only [→ 5.3.1 Stacking Connection])
– Receiving Whisper OHCA (→ 2.10.4.4 Whisper OHCA)

PC Programming Manual References

8.3.1 Users—Flexible Button—Wired Extension—Type
9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property—Option—Headset
OFF/ON

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

1.4.8 Using the Headset (Headset Operation)
3.1.2 Settings on the Programming Mode

2.11.5 Data Line Security

Description
Setting Data Line Security at an extension protects communications between the extension and the other
party from being interrupted by signals such as Call Waiting, Hold Recall and Executive Busy Override. An

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2.11.6 Flash/Recall/Terminate

extension that is using a connected data device (e.g., a fax machine) can set this feature to maintain secure
data transmission by preventing tones or interruptions from other extensions during communication.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Data Line Security Set /
Cancel
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 2—Data Mode

Operating Manual References

1.9.9 Protecting Your Line against Notification Tones (Data Line Security)

2.11.6 Flash/Recall/Terminate
Description
The FLASH/RECALL button (Flash/Recall mode or Terminate mode) or Terminate button (Terminate mode)
is used when a PT user disconnects the current call and originates another call without hanging up first. It
performs the same function as going on-hook and then going off-hook.
[Explanation of Each Mode]
Flash/Recall Mode: Disconnects the line. The extension user hears the dial tone from the line used last.
For example, if a trunk call is disconnected, the extension user will hear a new dial tone from the telephone
company.
Terminate Mode: Disconnects the line. The extension user hears the dial tone determined by the Line
Preference—Outgoing setting. (→ 2.5.5.2 Line Preference—Outgoing)

Conditions
• FLASH/RECALL Button Mode
One of the following modes can be selected for each extension through system programming:
– Flash/Recall mode
– Terminate mode
– External Feature Access (EFA) mode. (→ 2.11.7 External Feature Access (EFA))
• Terminate Button
A flexible button can be customised as the Terminate button.
• Disconnect Time (Only for Flash/Recall Mode)
The amount of time between successive accesses to the same trunk is programmable for each trunk port.
• This feature outputs an SMDR call record (→ 2.22.1.1 Station Message Detail Recording (SMDR)),
restarts the call timer, inserts the automatic pause, and checks the TRS/Barring level (→ 2.7.1 Toll
Restriction (TRS)/Call Barring (Barring)) again.
• The Terminate feature will be performed when pressing the FLASH/RECALL button regardless of the
mode that the FLASH/RECALL button has been set to, in the following situations:
– When a call is made using ARS. (→ 2.8.1 Automatic Route Selection (ARS))
– When a trunk call is made with the INTERCOM button.
– When a trunk call is made with an ICD group button.
• For general SIP phones, the function of a FLASH button differs depending on the phone, and its
functionality does not depend on the setting specified in the PBX.

PC Programming Manual References

9.24 PBX Configuration—[1-1] Configuration—Slot—Port Property - LCO Port—Disconnect Time

140 Feature Manual

 2.11.7 External Feature Access (EFA)

8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 7—Flash Mode during

CO Conversation
8.3.1 Users—Flexible Button—Wired Extension—Type
8.3.2 Users—Flexible Button—Portable Station—Type

Feature Manual References

2.21.2 Flexible Buttons

2.11.7 External Feature Access (EFA)

Description
Normally, an extension user can only access features within the PBX. However, when performing External
Feature Access (EFA) the extension user performs features outside of the PBX, such as using the transfer
services of the telephone company or host PBX. When EFA is performed, the PBX sends a flash/recall
signal to the telephone company or the host PBX (→ 2.5.4.8 Host PBX Access Code (Access Code to the
Telephone Company from a Host PBX)).
This feature is only available on trunk calls.
This feature is performed by pressing the EFA button or the FLASH/RECALL button that is set to EFA mode
(→ 2.11.6 Flash/Recall/Terminate).

Conditions
• Flash/Recall Time
The Flash/Recall time can be assigned for each trunk port.
• EFA Button
A flexible button can be customised as the EFA button.
• It is possible to perform this feature by entering the feature number while the current call is placed on
Consultation Hold (e.g., is going to be transferred to an extension of the host PBX).
Consultation Hold: a condition that a party is in, when an extension is calling other parties in order to
perform Call Transfer, Conference, or Call Splitting.
In Consultation Hold, the original call is treated as if it is on hold, allowing the extension to call a third
party all on one line. In Call Hold, the party on hold and the third party are connected to the extension
using separate lines.

PC Programming Manual References

9.24 PBX Configuration—[1-1] Configuration—Slot—Port Property - LCO Port—Flash Time
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—External Feature Access
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 7—Flash Mode during
CO Conversation
8.3.1 Users—Flexible Button—Wired Extension—Type
8.3.2 Users—Flexible Button—Portable Station—Type
15.5 PBX Configuration—[7-5] TRS—Miscellaneous—TRS Check after EFA

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

1.11.4 If a Host PBX is Connected

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2.11.8 Trunk Call Limitation

2.11.8 Trunk Call Limitation

Description
Trunk calls are limited by the following features:
Feature Description
Extension-to-Trunk Call If a call between an extension user and an outside party is
Duration established, the call duration can be restricted by a system timer
selected for each trunk group. Both parties will hear warning tones
at five-second intervals starting 15 seconds before the time limit.*1
When the time limit expires, the line will be disconnected. COS
programming determines whether this feature is enabled or
disabled. Whether this feature applies to outgoing calls only, or to
both outgoing and incoming calls is determined through system
programming.
Trunk-to-Trunk Call (except If a call between two outside parties is established, the call duration
Unattended Conference Call) can be restricted by a system timer selected for each trunk group.
Duration Both parties will hear warning tones at five-second intervals starting
15 seconds before the time limit.*1 When the time limit expires, the
line will be disconnected.
If both parties involved in the trunk-to trunk call were established by
an extension (e.g., an extension makes a trunk call, then transfers
the call to an outside party), the time limit applied to the trunk call
that was made first will be used.
Budget Management When the preprogrammed call charge limit has been reached, an
extension user will hear 3 warning tones at five-second intervals. It
is programmable whether the line is disconnected after the third
tone. After the call has ended, the extension user cannot make
further trunk calls until the charge limit has been increased or
cleared by an extension assigned as a manager (→ 2.7.2 Budget
Management).
Dialling Digit Restriction While engaged in an incoming trunk call, the dialling of digits can
during Conversation be restricted. If the number of dialled digits exceeds the limitation,
the line will be disconnected.
Logical Partitioning Feature Some types of trunk calls can be restricted with the following
Logical Partitioning settings:
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property
—Main—Main—Area ID for logical partition
10.9 PBX Configuration—[2-9] System—System Options—Option 2
—Applying logical partitioning
The following log data can be collected and displayed through PC
Programming. Refer to "7.3.5 Utility—Log—Call Control Log" in the
PC Programming Manual.
1. Log data (condition) will be recorded when you change any of
the Logical Partitioning settings.
2. Log data (status) will be recorded whenever a call is restricted
by the Logical Partitioning feature.

*1 A party connected via an IP trunk or SIP trunk will not hear the warning tone.

142 Feature Manual

 2.11.8 Trunk Call Limitation

Conditions
• During an Unattended Conference Call, the Unattended Conference Recall time is applied.
(→ 2.14.2 Conference)
• When using LCO trunks that do not support Calling Party Control (CPC) signal detection (→
2.11.9 Calling Party Control (CPC) Signal Detection), the Trunk-to-Trunk Call Duration timer should not be
disabled, as automatic end of call detection cannot be performed.
• For SIP Extension users, the line will be disconnected without hearing any warning tones when the trunk
call limitation expires.

[Logical Partitioning]
• When Logical Partitioning is enabled, the following types of calls are restricted.
1. Legacy trunk and IP trunk calls

Legacy Trunk IP Trunk

PBX

Restricted

2. TIE-to-trunk calls

Legacy Trunk/
IP Trunk

PBX
Private line

PBX

Extension
TIE-to-CO: Restricted

3. Trunk-to-TIE calls

Feature Manual 143

2.11.8 Trunk Call Limitation

Legacy Trunk/
IP Trunk

PBX
Private line

PBX

CO-to-TIE: Restricted

4. Conference calls that include legacy trunks and IP trunks

Legacy Trunk IP Trunk

PBX

Conference Restricted

Extension

5. Calls to an outside line (legacy or IP trunk) that cross over a Multi-connection network and the area
ID is different for each PBX.

Legacy Trunk/
IP Trunk

PBX A Multi-connection
(Area ID: 1)
PBX B
(Area ID: 2)

Extension
Outside call over Multi-connection network: Restricted

144 Feature Manual

 2.11.9 Calling Party Control (CPC) Signal Detection

If two PBXs in a Multi-connection network have the same area ID, restrictions are applied as if they
were one PBX (restriction types 1 to 4 above).
→ 9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Main—Area ID for
logical partition

IP Trunk Legacy Trunk

Multi-connection
PBX A PBX B
(Area ID: 1) (Area ID: 1)

IP Trunk to Legacy Trunk: Restricted

• When logical partitioning is applied, calls over a TIE line and a trunk (cases 3 and 4) are restricted. Calls
from an extension over a TIE line, and calls over TIE lines acting as relays are not restricted. However,
QSIG connections using PRI adaptors are restricted.

PC Programming Manual References

7.3.5 Utility—Log—Call Control Log
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Main—Area ID for logical
partition
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—CO & SMDR—Extension-
CO Line Call Duration Limit
10.9 PBX Configuration—[2-9] System—System Options—Option 2—Extension - CO Call Limitation—For
Incoming Call
10.9 PBX Configuration—[2-9] System—System Options—Option 2—Applying logical partitioning
11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Main
→ CO-CO Duration Time (*60s)
→ Extension-CO Duration Time (*60s)
15.5 PBX Configuration—[7-5] TRS—Miscellaneous—Dial Digits Limitation After Answering—Dial Digits

Feature Manual References

5.1.1 Class of Service (COS)

2.11.9 Calling Party Control (CPC) Signal Detection

Description
The Calling Party Control (CPC) signal is an on-hook indication (disconnect signal) sent from the analogue
trunk when the other party hangs up. To maintain efficient utilisation of trunks, the PBX monitors their state
and when CPC signal is detected from a line, disconnects the line and alerts the extension with a reorder
tone.

Conditions
• CPC signal detection is programmable for incoming trunk calls, and for outgoing trunk calls.
• If your telephone company sends other signals similar to CPC, it is recommended not to enable CPC
signal detection on outgoing trunk calls.

Feature Manual 145

2.11.9 Calling Party Control (CPC) Signal Detection

• If a CPC signal is detected during a Conference call (→ 2.14.2 Conference), that line is disconnected, but
the remaining parties stay connected.
• If a CPC signal is detected during a call between a caller using the DISA feature (→ 2.16.1 Direct Inward
System Access (DISA)) and an extension or an outside party, the line is disconnected.

PC Programming Manual References

9.24 PBX Configuration—[1-1] Configuration—Slot—Port Property - LCO Port—CPC Signal Detection Time
—Outgoing, Incoming

146 Feature Manual

 2.12 Transferring Features

2.12 Transferring Features

2.12.1 Call Transfer
Description
An extension user can transfer a call to another extension or an outside party. The following features are
available:
Feature Transferring method
With Announcement Transfer is completed after announcing the destination party.
Transfer is completed without an announcement.
Without Announcement After dialling the destination, while hearing a ringback tone, the
originator can replace the handset.

Call Transfer with Announcement is also known as Call Transfer—Screened.

Call Transfer without Announcement is also known as Call Transfer—Unscreened.

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2.12.1 Call Transfer

Transfer Recall for Call Transfer without Announcement

If the transfer destination does not answer within the preprogrammed Transfer Recall time, the call will be
redirected to the Transfer Recall destination assigned to the extension which transferred the call.
If the transfer destination has a destination set as Intercept Routing—No Answer, the call will be routed to
that destination.

A call is transferred without announcement.

Does the transferrer Yes

have a Transfer Recall
destination assigned?

No

Is the first transfer No

destination an extension?

Yes

Does the first transfer

No
destination have a destination set as
Intercept Routing No Answer?

Yes

Is the intercept destination No No Is the recall destination an

an extension, ICD group, or extension, ICD group, or
UM group? UM group?

Yes Yes

The set extension, ICD group, UM The transferrer is The set extension, ICD group, or
group, memorised as the Transfer memorised as the UM group, is memorised as the
Recall destination. Transfer Recall destination. Transfer Recall destination.

The Transfer Recall timer is started.

[Available destination]
Destination Availability
Wired Extension (PT/SLT/SIP Extension/ISDN Extension/T1-OPX) ü
PS ü
Incoming Call Distribution Group ü
PS Ring Group
UM Group ü*1
Fax Unit
External Pager (TAFAS)
DISA

148 Feature Manual

 2.12.1 Call Transfer

Destination Availability
Idle Line Access no. + Phone no.
Trunk Group Access no. + Trunk Group no. + Phone no.
Other PBX Extension (TIE with no PBX Code)
Other PBX Extension (TIE with PBX Code)

*1 If the transfer destination does not answer, the call is sent to Voice Mail and a message can be recorded in the mailbox of the
transfer destination.

Conditions
• When an extension is transferring a party to another destination, the party will be in consultation hold until
they reach the transfer destination.
Consultation Hold: a condition that a party is in, when an extension is calling other parties in order to
perform Call Transfer, Conference, or Call Splitting.
In Consultation Hold, the original call is treated as if it is on hold, allowing the extension to call a third
party all on one line. In Call Hold, the party on hold and the third party are connected to the extension
using separate lines.
• If Music on Hold is enabled, music can be sent to the held party while the call is transferred. (→
2.13.4 Music on Hold) It is programmable whether a ringback tone or music is sent.
• If the transfer destination extension has set FWD to an outside party, the call will be transferred to the
outside party. (→ 2.3.2 Call Forwarding (FWD))
• COS programming determines the extensions that are able to transfer a call to an outside party. COS can
also prohibit transferring to an extension of another PBX via the TIE line service using the PBX Code
method (Access with PBX Code) (→ 4.3.1 TIE Line Service).
• One-touch Transfer
One-touch Transfer can be performed by pressing a One-touch Dialling button that has been assigned
the TRANSFER command and the telephone number of the transfer destination. This is useful for
transferring calls to an outside destination. (→ 2.6 Memory Dialling Features)
• Automatic Transfer by SDN Button or DSS Button
Pressing an SDN button or DSS button during a conversation with an extension or outside party can
automatically transfer the call to the specified destination (→ 2.9.1 Primary Directory Number (PDN)/
Secondary Directory Number (SDN) Extension). It is possible through system programming to prevent
this feature from operating for extension to extension calls.
• Transfer to Busy Extension using Queuing (Camp-on Transfer)
Through system programming, it is possible to enable the transferring of a call to a busy extension
without needing to send a call waiting notification, based on the transferring party’s COS setting. The
transferred call will be placed in a queue.
This feature is not available for SIP extensions.
• When transferring a call from an analogue trunk, users are strongly recommended to perform a screened
transfer, so that the outside caller is not automatically connected to an extension using Hands-free
Answerback when the extension user is absent.
• Hold Recall is heard at the extension immediately (→ 2.13.1 Call Hold). On other types of extensions,
Hold Recall is heard after the Hold Recall timer expires.
• This PBX supports the Blind transfer feature found on some SIP phones. For details, refer to the phone’s
documentation.

PC Programming Manual References

10.2 PBX Configuration—[2-2] System—Operator & BGM—BGM and Music on Hold—Sound on Transfer

Feature Manual 149

2.12.2 SIP Refer Transfer

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Recall—Transfer

Recall
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—CO & SMDR—Transfer to
CO
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Assistant—Transfer to busy
Extension w/o BSS Operation
10.9 PBX Configuration—[2-9] System—System Options—Option 4—DSS Key—Automatic Transfer for
Extension Call
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 1—Transfer Recall
Destination

Feature Manual References

5.1.1 Class of Service (COS)

Operating Manual References

1.4.1 Transferring a Call (Call Transfer)

2.12.2 SIP Refer Transfer

Description
If enabled through system programming, calls transferred to a SIP trunk will be transferred using the SIP
service provider’s Transfer feature instead of the PBX’s.
The following types of transfer are available:
Attended Transfer Transfer is completed after announcing the transferred party.
From the extension user’s perspective, this is the same as
Call Transfer with Announcement (→ 2.12.1 Call Transfer).
Blind Transfer Transfer is completed immediately after dialling the transfer
destination’s number.
This is similar to Call Transfer without Announcement (→
2.12.1 Call Transfer), except that the transferrer does not
hear even a ringback tone; the transferred call is connected
directly to the destination.

Conditions
[General]
• The availability of this feature depends on the SIP service provider.
• Since the SIP service provider takes control of the transfer, the transferred call cannot be returned to the
PBX for further handling even if the transfer fails.

[Blind Transfer]
• ISDN extensions and SIP extensions cannot use this feature.
• This feature cannot be used when calling through DISA (→ 2.16.1 Direct Inward System Access (DISA)).

Installation Manual References

4.4 Virtual Cards

Operating Manual References

1.4.1 Transferring a Call (Call Transfer)—Transferring to an Outside Party Using the SIP Service

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 2.12.2 SIP Refer Transfer

PC Programming Manual References

9.11 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Port Property—Supplementary Service
→ Blind Transfer(REFER)
→ Attended Transfer(REFER)

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2.13 Holding Features

2.13 Holding Features

2.13.1 Call Hold
Description
An extension user can put a call on hold. The following features are available depending on the result.
Feature Description
Regular Hold Any extension can retrieve a held call.
Exclusive Call Hold Only the extension user who held the call can retrieve it.

The result of the holding operation can be determined through system programming. Pressing the HOLD
button again just after the first time alternates the mode between Regular and Exclusive Call Hold.

Conditions
• Call Hold Limitation
A PT user can hold one intercom call and/or multiple trunk calls at a time. An SLT user can hold either
one intercom call or one trunk call at a time. By using the Call Park feature, PT and SLT users can hold
multiple trunk calls and intercom calls simultaneously. (→ 2.13.2 Call Park)
• Music on Hold
Music, if available, is sent to the held party. (→ 2.13.4 Music on Hold)
• Hold Recall
If a call on hold is not retrieved within a preprogrammed time period, Hold Recall is heard at the extension
which put the call on hold. If the extension is engaged in a call, the Hold Alarm will be heard.
• If an outside party is placed on hold and not retrieved within a preprogrammed time period, the call is
automatically disconnected. This timer starts when Hold Recall activates.
• Automatic Call Hold
A PT user can be programmed holding of the current call when pressing another CO/ICD Group/
INTERCOM/PDN button, through system programming. If this feature is not enabled, the current call will
be disconnected.

[Example]
It is possible to receive a call by pressing the flashing ICD Group button, this puts the current intercom
call (on the INTERCOM button) on hold. To return to the held call, press the INTERCOM button.
• Call Hold Retrieve Deny
If an extension user cannot call certain extensions on a COS basis (→ 2.1.2.2 Internal Call Block), he
cannot retrieve the held call which the extensions made.
• SLT Hold Mode
It is possible to choose how to hold a line and transfer a call with an SLT in the following methods through
system programming:

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 2.13.1 Call Hold

Hold
Transfer to Transfer to
Hold (to be Retrieved from
Trunk Extension
Another Extension)*1
Flashing the Flashing the Flashing the Flashing the
hookswitch hookswitch hookswitch hookswitch
+ + + +
Mode 1
Going on-hook Hold Feature No. Trunk Access No. Extension No.
+
Going on-hook
Flashing the Flashing the Flashing the Flashing the
hookswitch hookswitch hookswitch hookswitch
Mode 2 + + + +
(Default) Hold Feature No. Hold Feature No. Trunk Access No. Extension No.
+ +
Going on-hook Going on-hook
Flashing the Flashing the Flashing the Flashing the
hookswitch hookswitch hookswitch hookswitch
+ + + +
Hold Feature No. Hold Feature No. Hold Feature No. Extension No.
Mode 3
+ + +
Going on-hook Hold Feature No. Trunk Access No.
+
Going on-hook
Flashing the Flashing the Flashing the Flashing the
hookswitch hookswitch hookswitch hookswitch
+ + + +
Hold Feature No. Hold Feature No. Hold Feature No. Hold Feature No.
Mode 4
+ + + +
Going on-hook Hold Feature No. Trunk Access No. Extension No.
+
Going on-hook

*1 These operations must be performed when the held call is intended to be retrieved from another extension using the holding
extension number.
If the following occurs frequently with an SLT, choose "Mode 2", "Mode 3", or "Mode 4":
a. When an SLT user receives a call, reorder tone is heard or nobody answers the call.
b. When an SLT user goes off-hook, reorder tone is heard instead of a dial tone.
If a call is not terminated after going on-hook, the above cases occur. To avoid these problems, choose
"Mode 2", "Mode 3", or "Mode 4". Every call will be terminated unless the Hold feature number is entered
after flashing the hookswitch in Mode 2, Mode 3, and Mode 4.
• Hold Alarm tone pattern has a default. (→ 6.2.1 Tones/Ring Tones)

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone
→ Recall—Hold Recall
→ Recall—Disconnect after Recall
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Call Hold / Call Hold Retrieve
→ Call Hold Retrieve : Specified with a Holding Extension Number
→ Hold Retrieve : Specified with a Held CO Line Number
10.9 PBX Configuration—[2-9] System—System Options

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2.13.2 Call Park

→ Option 1—PT Operation—Automatic Hold by ICM / CO / ICD Group Key

→ Option 1—PT Operation—Hold key mode
→ Option 5—SLT—SLT Hold Mode

Operating Manual References

1.4.2 Holding a Call

2.13.2 Call Park

Description
An extension user can place a call into a common parking zone of the PBX. The Call Park feature can be
used as a transferring feature; this releases the user from the parked call to perform other operations. The
parked call can be retrieved by any extension user.

Conditions
• Automatic Call Park
It is possible to select an idle parking zone automatically.
• Retry
If the specified parking zone is occupied or there is no vacant zone for Automatic Call Park, the originator
will hear a busy tone. Retrying is possible while hearing the busy tone by selecting parking zone or a
vacant zone.
• Call Park Recall
If a parked call is not retrieved within a preprogrammed time period, Call Park Recall will be heard at the
Transfer Recall destination assigned to the extension which parked the call. If the destination is engaged
in a call, the Hold Alarm will be heard.
• If a parked trunk call is not retrieved within a preprogrammed time period (Default: 30 minutes), it is
automatically disconnected.
• Call Park Button
Pressing the Call Park button parks or retrieves a call in a preset parking zone.
A flexible button can be customised as the Call Park button. It shows the current status of the preset
parking zone as follows:
Light pattern Status
Slow red flashing Parked in the preset parking zone
Off No parked call

• Call Park (Automatic Park Zone) Button

Pressing the Call Park (Automatic Park Zone) button parks a call in an idle parking zone automatically. A
flexible button can be customised as the Call Park (Automatic Park Zone) button.
• Call Park Retrieve Deny
If an extension user cannot call certain extensions on a COS basis (→ 2.1.2.2 Internal Call Block), he
cannot retrieve the parked call which the extensions made.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone
→ Recall—Call Park Recall
→ Recall—Disconnect after Recall

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 2.13.3 Call Splitting

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Call Park / Call Park

Retrieve
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Parameter Selection (for Call Park)
→ Optional Parameter (Ringing Tone Type Number) (for Call Park)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Parameter Selection (for Call Park)
→ Optional Parameter (or Ringing Tone Type Number) (for Call Park)

Feature Manual References

2.21.2 Flexible Buttons
6.1 Capacity of System Resources

Operating Manual References

1.4.2 Holding a Call

2.13.3 Call Splitting

Description
During a conversation, an extension user can call another extension while putting the original party on
Consultation Hold. The extension user can then alternate between the two parties and/or connect the
original party with the third party.

Conditions
• Consultation Hold: a condition that a party is in, when an extension is calling other parties in order to
perform Call Transfer, Conference, or Call Splitting.
In Consultation Hold, the original call is treated as if it is on hold, allowing the extension to call a third
party all on one line. In Call Hold, the party on hold and the third party are connected to the extension
using separate lines.
• When the extension user is having a conversation with one party, the other party is in consultation hold.

Operating Manual References

1.4.3 Talking to Two Parties Alternately (Call Splitting)

2.13.4 Music on Hold

Description
Music can be played to a party that has been put on hold. The following audio sources are available:
a. External music source
b. User-supplied audio file
c. Tone

The audio source for Music on Hold is selected from either a BGM number (1 to 8) or the built-in tone. BGM
can be either an external music source or a user-supplied audio file. The following table shows which audio
sources can be assigned to which BGM numbers:

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2.13.4 Music on Hold

[BGM Number and the Music Source]

BGM No. Music Source
1 External music port 1 or User audio data
2 User audio data
3 User audio data
4 User audio data
5 User audio data
6 User audio data
7 User audio data
8 User audio data

For tenant users, each tenant can select one of the BGMs or the tone to use for Music on Hold.
For each site one of the 8 available patterns can be selected as the BGM sound resource. The outgoing
BGM will subject to the settings of the site to which each terminal is connected.

Conditions
[General]
• Hardware Requirement: User-supplied music source (when an external music source is assigned)
• The ESVM card implemented by the Expansion Gateway and the External input terminal of the
Expansion Gateway cannot be selected as a BGM resource.
• Volume Control
It is possible to change the volume of an internal and/or external music source.
• For tenants, the type of call determines which tenant’s music source is used, as follows:
Type Music Source
Incoming Intercom Calls/Outgoing Selected based on the tenant setting to which the
Calls extension user belongs.
Selected based on the tenant setting of the distribution
Incoming Trunk Calls
method (DIL/DID/DDI/MSN).

• DSP Resource Usage

Music on Hold playback requires a certain number of DSP resources. If all DSP resources are in use, this
operation cannot be performed. (→ 5.5.3 DSP Resource Usage)
• Even if an external music source or a user-supplied audio file is selected for Music on Hold, an IP-PT or
SIP extension user who is put on hold by another extension will not hear the specified Music on Hold.
• External music sources can only be used when connected to the Expansion GW.
[User-supplied audio files]
• User-supplied audio files are uploaded via Web Maintenance Console. Audio files must meet the
following specifications:
– Format: WAV
– Size: 80 MB or less
– Length: 8 minutes or less

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 2.13.4 Music on Hold

• Initially, a preinstalled audio file is set as the audio source for BGM 1. Through system programming, this
file can be removed or replaced like any other BGM audio file. However, if the PBX is reinitialised, this
preinstalled audio file is set to BGM 1 again.
• Multi-connection Networking (→ 4.2 Multi-connection Networking)
When uploading a file to the Master unit, the user can select which sites to upload the file to and the BGM
number for each site. When uploading to a Slave unit, the user can select only the BGM number, and the
file is uploaded only to that PBX.

Installation Manual References

4.5 Connection of Peripherals

PC Programming Manual References

5.2 System Control—MOH
10.2 PBX Configuration—[2-2] System—Operator & BGM
10.11.1 PBX Configuration—[2-11-1] System—Audio Gain—Paging/MOH
→ Internal MOH—MOH 1–8 (Music on Hold 1–8)

Feature Manual References

2.28.1 Background Music (BGM)
5.1.3 Tenant Service

Feature Manual 157

2.14 Conference Features

2.14 Conference Features

2.14.1 Conference Features—SUMMARY
Description
A conference call allows a conversation between three or more parties simultaneously. The following
features are available to establish a conference call:
Feature Description & Reference
Conference During a two-party conversation, an extension user can add other
parties to establish a conference call with up to eight parties.

→ 2.14.2 Conference
Executive Busy Override An extension user can interrupt an existing call to establish a three-
party conference call.

→ 2.10.2 Executive Busy Override

Privacy Release During a conversation with an outside party on the S-CO button, a
PT/PS user can allow another extension to join the conversation.

→ 2.14.3 Privacy Release

Conditions
• One conference call supports up to 32 Parties.
• The maximum number of parties that can be engaged in conference calls simultaneously differs
depending on the type of PBX:
– KX-NSX Series: max. 256 parties
– Legacy gateway: max. 96 parties
Parties are counted at the PBX where the conference originated.
• DSP Resource Usage
A conference call requires a certain number of DSP resources. If all DSP resources are in use, this
operation cannot be performed. To ensure a minimum level of performance, DSP resources can be
reserved for conference calls. (→ 5.5.3 DSP Resource Usage)

PC Programming Manual References

9.33 PBX Configuration—[1-5] Configuration—DSP Resource

2.14.2 Conference
Description
An extension user can establish a conference call by adding additional parties to an already existing two-
party conversation. This PBX supports three-party through eight-party conference calls. Conferences with
more than four parties are only possible when a PT or PS user originates the conference.
Unattended Conference:
The conference originator can leave the conference and allow other parties to continue. Establishing an
Unattended Conference allows the originator to return to the conference. Unattended Conferences can only
be established by PT and PS users.

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 2.14.2 Conference

Conditions
• When an extension is establishing a conference call the original party is put on hold.
• CONF (Conference) Button
For a PT/PS which does not have the CONF button, a flexible button can be customised as the
Conference button.
• Unattended Conference Call Duration
The length of time that a conference call can remain unattended is restricted by the following timers:
– Callback Start Timer
– Warning Tone Start Timer
– Disconnect Timer
These timers behave and operate according to the following chain of events:
1. When the unattended conference is established, the Callback Start Timer will begin.
2. When the Callback Start timer expires, the Unattended Conference originator’s extension will start to
receive a callback ringing from the PBX and the Warning Tone Start Timer begins.
3. When the Warning Tone Start Timer expires, the remaining parties of the conference will start to
hear a warning tone, the callback ringing will continue to be heard at the Unattended Conference
originator’s extension, and the Disconnect Timer begins.
4. When the disconnect Timer expires, the conference is disconnected.
If the Unattended Conference originator returns to the conference before the line is disconnected, all
timers are cleared.
• If the originator of a conference with two trunks leaves the conference, the call can become a trunk-to-
trunk call, if enabled through system programming.
– When a trunk-to-trunk call is established, the call will end when the Trunk-to-Trunk Call Duration timer
has elapsed (→ 2.11.8 Trunk Call Limitation). The timer applied is that of the trunk group containing the
trunk being used for the call immediately preceding the conference. The timer cannot be extended.
– If both trunks are analogue trunks, the end of the trunk-to-trunk call may not be detected. For this
reason, if analogue trunks are used, it is not recommended to enable the establishment of trunk-to-
trunk calls after a conference call through system programming.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf
→ Unattended Conference—Recall Start Timer
→ Unattended Conference—Warning Tone Start Timer
→ Unattended Conference—Disconnect Timer
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—CO & SMDR—Transfer to
CO
10.9 PBX Configuration—[2-9] System—System Options—Option 2—CO - CO Call Limitation—After
Conference
10.9 PBX Configuration—[2-9] System—System Options—Option 3
→ Confirmation Tone—Tone 4-1 : Start Conference
→ Confirmation Tone—Tone 4-2 : Finish Conference
→ Echo Cancel—Conference
10.9 PBX Configuration—[2-9] System—System Options—Option 8—Conference Group—Maximum
Number of Speakers During a Conference Group Call
8.3.1 Users—Flexible Button—Wired Extension—Type
8.3.2 Users—Flexible Button—Portable Station—Type

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2.14.3 Privacy Release

Feature Manual References

2.21.2 Flexible Buttons
6.1 Capacity of System Resources

Operating Manual References

1.4.5 Multiple Party Conversation

2.14.3 Privacy Release

Description
By default, all conversations which take place on trunks, extension lines and doorphone lines are protected
by privacy (Automatic Privacy).
Privacy Release allows a PT/PS user to suspend Automatic Privacy for an existing trunk call on the S-CO
button in order to establish a three-party call.
System programming is required to enable or disable this feature.

Conditions
• S-CO Button
A flexible button can be customised as the S-CO button.
• Privacy Release Time
Privacy is released for five seconds to allow the conversation to be joined.
• This feature overrides Data Line Security (→ 2.11.5 Data Line Security) and Executive Busy Override
Deny (→ 2.10.2 Executive Busy Override).

PC Programming Manual References

10.9 PBX Configuration—[2-9] System—System Options—Option 1—PT Operation—Privacy Release by
SCO key
8.3.1 Users—Flexible Button—Wired Extension—Type
8.3.2 Users—Flexible Button—Portable Station—Type

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

1.4.5 Multiple Party Conversation

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 2.15 Conference Group Call Features

2.15 Conference Group Call Features

2.15.1 Conference Group Call
Description
Conference group calling allows you to call a pre-determined group (conference group) of parties
simultaneously. Each party that answers the call joins the conference.
An extension user can call a conference group of up to 31 parties to establish a conference call, for a
maximum of 32 participants. During a conference group call, the caller can restrict the ability of other
members to speak.
The following telephones will answer automatically and play the announcement through the telephone’s
speaker, even if Hands-free Answerback (→ 2.4.4 Hands-free Answerback) is not enabled for the extension:
• PTs
• KX-WT125 (PS)
• KX-TCA175 (PS)
• KX-TCA275 (PS)
• KX-TCA185 (PS)
• KX-TCA285 (PS)
• KX-TCA385 (PS)
• KX-TCA364 (PS)

Broadcast Mode
When Broadcast Mode is enabled through system programming, an extension user can call a conference
group of up to 31 call members to make a voice announcement. Members can listen to the announcement
by answering the call.
During the announcement, the voices of members will not be heard. However, the caller can allow up to 31
specific members to speak, making a conference call. This conversation can be heard by the other
members.

[Push-to-talk feature for PT/PS/SLT users]

PT/PS/SLT users that are members of a Broadcast Mode call can enable their own ability to speak by
pressing any of their dial keys during the Broadcast Mode announcement. This feature can be disabled
through system programming.
Broadcast Mode can be used to broadcast an announcement to multiple PS users. With Automatic Answer
enabled, the PS users will all hear the announcement through their headset or the PS’s hands-free speaker.
Then, any PS user can respond by using push-to-talk to enable their ability to speak, and their reply is heard
by all broadcast members.
For example, a central operator could make a Broadcast Mode call paging PS users working throughout a
building for assistance, and an available PS user can reply. The reply is heard by the other PS users as well.
This makes it easy to assign and coordinate tasks with multiple staff members that are frequently moving
throughout a building.

Conference Group Call Control

During a conference group call, the caller can restrict or allow members’ ability to speak, and can remove
members from the call using the following buttons. These buttons will function irrelevant of the Conference
Group Call mode. Pressing the other buttons during the conversation will be ignored.

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2.15.1 Conference Group Call

Note
The operation of these buttons during a conference group call is different from the operations for the
Conference feature (→ 2.14 Conference Features).
Button Function
DSS Disables or enables the corresponding member’s ability to speak.
CONF (Conference) Establishes a conversation with the current members in the order
assigned in the conference group. Pressing this button again will
add the next available member in the group to the conversation.
TRANSFER Removes the member who joined the conversation last. The
member can still listen to the announcement.
FLASH/RECALL (Flash/Recall Removes the member who joined the conversation last. The
mode) member will be disconnected from the conference group call and
hear a reorder tone.
SP-PHONE Enables a hands-free conversation.

A member extension can inform the caller that he wants to speak or join the conversation by sending a
notification. The caller will hear a notification tone and the requesting extension’s information will be shown
on the display for five seconds.

Conference Groups
48 conference groups can be programmed, and a maximum of 31 members can be assigned to each group.
The available destinations as members of the conference group are as follows:
Destination Availability
Wired Extension (PT/SLT/SIP Extension/ISDN Extension/T1-OPX) ü
PS ü
Incoming Call Distribution Group
PS Ring Group
UM Group
Fax Unit
External Pager (TAFAS)
DISA
Idle Line Access no. + Phone no. ü
Trunk Group Access no. + Trunk Group no. + Phone no. ü
Other PBX Extension (TIE with no PBX Code) ü *1
Other PBX Extension (TIE with PBX Code) ü *1

*1 Only available when the networking type of the trunk is assigned as private.

Join After Time Out

When conference group members do not answer a conference group call within the preprogrammed time
limit, the member’s telephones will stop ringing. However, even after the time limit has expired, members
can join the conference.

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 2.15.1 Conference Group Call

Additionally, extension users not registered in the called conference group can join a conference after it has
started. Outside callers using DISA and TIE line users can also join a conference after accessing their
extension using Walking COS.
An entry code can also be specified to restrict access to the call. The entry code can be set when the caller
initiates the conference group call. If an entry code is to be set, usually the caller will inform call participants
in advance.

[Starting a Conference Group Call using Hands-free Answerback]

Hands-free Answerback can be enabled for conference group member extensions. By initiating a
conference group call with a conference group that includes an extension that has Hands-free Answerback
(→ 2.4.4 Hands-free Answerback) enabled, such as a softphone, the call can be automatically answered
and the conference begins with only one participant (the initiator of the conference group call). Then, up to 6
participants can use Join After Time Out to join the conference. For example, a manager can organise a
meeting where all the members are calling from cellular phones and use Join After Time Out to join the
conference.

Conditions
• Only extensions that are permitted by COS programming can originate conference group calls.
• Conference Group Call Control features are unavailable if an SLT or SIP extension is used to initiate the
call.
• Users of PSs other than the KX-TCA175/KX-TCA275/KX-TCA185/KX-TCA285/KX-TCA385/KX-TCA364
may be able to enable the automatic answering of calls for this feature by changing their PSs’ settings.
For details, refer to the operating instructions of the PS.
• After one conference group member answers the call, the conference or announcement is established.
• If no members answer the call within the preprogrammed time limit, the caller will hear a busy tone.
• The caller will hear a confirmation tone every time a member answers the call.
• When the originating caller of a conference group call goes on hook, the call ends and all participating
members will be disconnected.
• The conference group call will reach a member extension regardless of settings such as Call Forwarding
(except DND).
• If a member extension is busy and has Call Waiting for trunk calls activated when a conference group call
is made, a call waiting tone will be sent to the extension.
• For members who use a KX-TCA175/KX-TCA275/KX-TCA185/KX-TCA285/KX-TCA385/KX-TCA364,
when automatic answer is enabled for the conference group and the extension is busy when a
conference group call is made, the PS will automatically answer the call if the member goes on-hook
while the conference call is still ringing. A PT will ring instead of answering the call.
• The call information of the caller (not members) will be recorded on SMDR.
• A caller cannot make a conference group call with a call on hold.
• Call Pickup is not available for a conference group call. (→ 2.4.3 Call Pickup)
• The conference group call will not reach members when:
– the member extension has set DND for intercom calls.
• When using wireless devices linked to the User Container (→ 2.29 User Container)
• If a conversation has reached the maximum number of participants, the Join After Time Out feature
cannot be used to join the conversation.
• If a member uses push-to-talk to enable the ability to speak during a Broadcast Mode call, the member
cannot disable this ability. They can mute their microphone manually, or the originator of the call can use
conference group call control to disable their ability to speak.

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• Since each PS requires one wireless channel, note your PBX’s wireless capacity when assigning multiple
PSs to a conference group.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—Conference Group Call—
Ring Duration
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Conference Group Call
Operation
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Miscellaneous—Conference
Group Call Operation
10.9 PBX Configuration—[2-9] System—System Options—Option 8—Conference Group—Maximum
Number of Speakers During a Conference Group Call
11.9 PBX Configuration—[3-9] Group—Conference Group
→ Broadcast Mode
→ Ability to Talk
→ Automatic Answer w/o Extension Setting
11.9.1 PBX Configuration—[3-9] Group—Conference Group—Member List

Feature Manual References

2.4.4 Hands-free Answerback
6.1 Capacity of System Resources

Operating Manual References

1.7.3 Making a Conference Group Call

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2.16 Direct Inward System Access (DISA) Features

2.16.1 Direct Inward System Access (DISA)
Description
An outside caller can access specific PBX features as if the caller is an SLT extension user in the PBX,
when the incoming call destination is a DISA floating extension number assigned to each DISA message.
The caller can have direct access to features such as:
• Placing an intercom call to an extension, operator or any floating extensions (e.g., an external pager for
TAFAS).
• Calling an outside party via the PBX.
• Operating some PBX remote features (e.g., FWD)

DISA Intercept Routing—No Dial

If the caller fails to dial any digits within a preprogrammed time period (DISA 1st Dial Time for Intercept)
after hearing the outgoing message (OGM), one of the following can be selected through system
programming:
a. Disable: The call will be terminated.
b. Operator: The call will be redirected to the operator.
c. AA-0, AA-9: The call will be redirected to the destination assigned to that AA number.
→ 10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf—DISA—
No Dial Intercept Timer
→ 13.3.1 PBX Configuration—[5-3-1] Optional Device—Voice Message—DISA System—Option 1—DISA
Intercept—Intercept when No Dial after DISA answers

DISA Built-in Automated Attendant Number (DISA AA Service)

After listening to the outgoing message (OGM), the caller may dial a single digit (DISA AA number). The
destination for each DISA AA number can be assigned for each message. It is also possible to assign other
DISA floating extension numbers as the destination (Multistep DISA AA Service).
If the caller dials a second digit within a preprogrammed time period (DISA 2nd Dial Time for AA), the DISA
AA service is not employed.
→ 13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message—1 Digit AA
Destination (Extension Number)—Dial 0–9
→ 10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf—DISA—
2nd Dial Timer for AA

Outgoing Message (OGM)

When a call arrives on a DISA line, a prerecorded DISA message will greet and guide the caller.
Any extension assigned as the manager can record outgoing messages (OGMs). (→ 2.28.2 Outgoing
Message (OGM))

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[Programming Example]
Outgoing Automated Attendant No.*2
Floating Busy/DND
Message
Extn. Message
(OGM) 0 1 2 3 4 5 6 7 8 9
No.*1 No.*3
No.
1 5801 100 301 200 103 202 101 102 400 104 205 4
2 5802 5
: : : : : : : : : : : : :

*1 → 13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message—Floating Extension Number

The default floating extension number depends on the value specified for Numbering Plan in Easy Setup.
*2 → 13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message—1 Digit AA Destination (Extension
Number)—Dial 0–9
*3 → 13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message—Busy / DND Message No.

DISA Security Mode and Available Features

If the DISA AA service is not employed, the caller may access the PBX features by entering the feature
numbers. To prevent others from accessing the PBX features, it is possible to assign DISA security.
→ 13.3.1 PBX Configuration—[5-3-1] Optional Device—Voice Message—DISA System—Option 1
The features available depend on the preprogrammed DISA security mode as follows:

Intercom TIE Line Call

Security Mode Trunk Call
Call Without PBX Code With PBX Code
All Security
Trunk Security ü ü
No Security ü ü ü ü*1

ü: Available
*1 If trunk call is available, Account Code Entry (→ 2.5.4.3 Account Code Entry) is also available.

Note
DISA AA service and Operator Call (→ 5.1.5 Operator Features) are available for any security mode.
Security Mode Override by Verification Code Entry
If the caller performs Verification Code Entry (→ 2.7.6 Verification Code Entry) while hearing a DISA
message, the security mode can be temporarily changed to No Security mode.
Entry method:
Verification Code Entry feature number + + verification code + verification code PIN
After changing mode, the new mode remains in force for the duration of the call.

DISA Intercept Routing—Busy

If the first destination called by the outside party is busy, the call is redirected as follows:
a. The call is redirected to the Intercept Routing—Busy destination assigned to the first destination.
b. If an Intercept Routing—Busy destination is not assigned to the first destination and a prerecorded
DISA Busy Message is assigned, the caller will hear the DISA Busy Message.
c. If neither an intercept destination nor a DISA Busy Message is assigned, the caller will hear a busy
tone.
→ 8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Intercept Destination—
Intercept Destination—When Called Party is Busy

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DISA Intercept Routing—DND

If the destination called by the outside party is in DND mode and Idle Extension Hunting is not available, one
of the following can be selected through system programming:
a. Busy Tone: The caller will hear a busy tone.
b. Enable: DND will redirect the call to the preprogrammed destination on an extension basis.
c. OGM: An outgoing message (OGM) will be sent to the caller. The message for DND mode can be
assigned for each outgoing message (OGM) which has a DISA floating extension number.
→ 13.3.1 PBX Configuration—[5-3-1] Optional Device—Voice Message—DISA System—Option 1

DISA Intercept Routing—No Answer

If a destination is not available to answer a DISA call within a preprogrammed time period (DISA Intercept
time) after the call is reached, the call will be redirected to the programmed destination by the Intercept
feature.
If the intercept destination is not available to answer the call within a preprogrammed time period (DISA
Disconnect Time after Intercept) after the DISA Intercept time expires, the call will be disconnected.
10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf
→ DISA—Intercept Timer—Day, Lunch, Break, Night
→ DISA—Disconnect Timer after Intercept
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Intercept Destination—
Intercept Destination—When called party does not answer—Day, Lunch, Break, Night

Walking COS Through DISA

If the caller performs Walking COS (extension number and PIN entry) while hearing a DISA message, the
security mode can be temporarily changed to No Security mode (→ 2.7.5 Walking COS). After performing
Walking COS, the following features are available, using the settings of the specified extension:
• Intercom call
• TIE line call
• Trunk call
• Call Forwarding (FWD)/Do Not Disturb (DND) setting (→ 2.3 Call Forwarding (FWD)/Do Not Disturb
(DND) Features)
• Incoming Call Distribution Group Log-in/Log-out (→ 2.2.2.7 Log-in/Log-out)
• Absent Message setting (→ 2.20.2 Absent Message)
• Extension Dial Lock (→ 2.7.3 Extension Dial Lock)
• Time Service Switching Mode (→ 5.1.4 Time Service)

Note
When making a trunk call using Walking COS through DISA, the CLIP number for that call will be that of
the extension seized by Walking COS. (→ 4.1.2.2 Calling/Connected Line Identification Presentation
(CLIP/COLP))
DISA Automatic Walking COS
Registered outside destinations such as cellular phones can be automatically recognised as PBX
extensions when calling through DISA. When the Caller ID of a received trunk call matches an entry in the
System Speed Dialling Table, the calling telephone is given Walking COS authorisation as assigned to the
corresponding CLI destination extension. Therefore, the "CLI destination" setting in the System Speed
Dialling Table is used here to specify the target extension that the calling telephone will be recognised as for
Walking COS.
[Programming Example of DIL Table]

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2.16.1 Direct Inward System Access (DISA)

CLI Destination
Trunk No.
Day Lunch ... Day Lunch ...
5801 5801
01 Disable Disable ... ...
(DISA) (DISA)
: : : : : : :

"CLI" must be set to Disable, to allow incoming calls to be received by DISA.

[Programming Example of System Speed Dialling Table]
Trunk Access +
Location Name CLI Destination
Telephone Number
000 J. Smith 912341115678 200
001 : : :
: : : :

In this example, calls received on trunk 01 are routed to the DISA OGM with floating extension number
5801. If the number of the received call (after modification according to the Caller ID table) is
"12341115678", the call originator is recognised as extension 200, and the Walking COS feature is
automatically activated.
System programming is required to enable this feature.
SMDR
The call information for DISA is recorded as the one of the DISA floating extension numbers.
(→ 2.22.1.1 Station Message Detail Recording (SMDR))

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[Flowchart]

A DISA call from an outside party is received.

No
Is there a port available?

The call is routed to an operator,

(DISA Delayed etc. (DISA Intercept when All
Yes
Answer time expires) DISA Ports are busy)

The PBX answers the call.

(DISA Mute & OGM Start Time

A after Answering expires)

The OGM plays and the PBX starts to

receive the DTMF signalling.

No
Is the first digit dialled?
(DISA First Digit Time
When No Dial expires)
What method is assigned for
DISA Intercept Routing No Dial?

Operator Disable AA-0, AA-9

The call is routed to an The call is redirected to

operator. The call is the destination assigned
disconnected. to AA-0 or AA-9.
Yes

The OGM stops.

C C

No
Is a second digit dialled?
(DISA Second Digit Time for
Automated Attendant expires)

Yes
No Is the first dialled digit assigned a
destination for the DISA AA service?
The PBX receives the dialled
digits and checks the dialled Yes
number.

The call is routed to the destination.

What is the DISA security mode?

Continued on next page

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2.16.1 Direct Inward System Access (DISA)

Continued from previous page

No Security
None Trunk Security All Security

B Is the dialled number No

an extension number or
floating extension number?

Yes

Is the Walking COS/Verification Yes

C Code Entry feature number dialled
for Security Mode Override?

What is the dialled number? No

Is the correct Yes
PIN entered?

Extension No./ Trunk Access No. Feature No.* Other No

Floating + Telephone No. (Absent Message,
D
Extension No. FWD, etc.)
Reorder tone B
The feature is set. D
The dialled number is sent to the trunk.

C Does the caller press No

while hearing the reorder
(DISA Reorder
tone (Call Retry)?
Tone time
Is the extension No Is the extension Yes expires)
Yes
in DND mode? busy?

Yes No A
E
The call is directed to the extension. The call is disconnected.
Is Call Waiting No
The caller hears a ringback tone.
mode on?

Yes
No Does the caller press
while hearing a ringback
tone (Call Retry)? E

Yes

Does the
destination A
answer No
the call? (DISA Intercept
time expires)

Yes The call is routed to the

intercept destination.
(DISA Intercept Routing
The call is established. No Answer)

Does the destination No

answer the call? (DISA Disconnect
Time after
Yes Intercept expires)

Continued on next page The call is established. The call is disconnected. Continued on next page

* Feature numbers are available only when the Walking COS feature is used.

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Continued from previous page Continued from previous page

What method is assigned

for DISA Intercept Is an Intercept Yes
Routing DND? Routing Busy destination
assigned?
The call is routed to
Intercept Routing Busy tone OGM
No the intercept destination.
(Intercept Routing Busy)

No Does the caller

press while hearing a busy Is a DISA Busy No
(Busy Tone / tone (Call Retry)? Message assigned?
DND Tone G
Continuation
time expires) Yes The message for Yes
the DND mode is
sent to the caller.
The call is disconnected.
A The message for
busy is sent to the
The call is routed to caller.
Busy tone
the intercept destination. F
(Intercept Routing DND)

F
No Does the caller
press while hearing a busy
G (Busy Tone / tone (Call Retry)?
DND Tone
Continuation
time expires) Yes

The call is disconnected.

A

Conditions
CAUTION
There is a risk that fraudulent telephone calls will be made using the Trunk-to-Trunk Call feature of
DISA.
The cost of such calls will be billed to the owner/renter of the PBX.
To protect the PBX from this kind of fraudulent use, we strongly recommend:
a. Enabling DISA security (Trunk Security or All Security).
b. Keeping passwords (verification code PINs/extension PINs) secret.
c. Selecting complex, random PINs that cannot be easily guessed.
d. Changing PINs regularly.
• DISA Delayed Answer Time
It is possible to set the Delayed Answer time so that the caller will hear a ringback tone within a
preprogrammed time period first before hearing an outgoing message (OGM).
→ 10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf—DISA
—Delayed Answer Timer
• Call Retry
While hearing a ringback, reorder, or busy tone, retrying the call is possible by pressing " ". System
programming selects whether pressing " " during a trunk-to-trunk conversation returns to the DISA top
menu or sends a DTMF tone.
• DISA Mute Time

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It is possible to set the Mute time until the outgoing message (OGM) plays and the PBX starts to receive
the DTMF signalling after the caller reaches the DISA line.
→ 10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf—DISA
—Mute & OGM Start Timer after answering
• End of Call Detection
If a call through DISA is routed to a trunk, DISA can be used to detect the end of the call. This function
can be disabled through system programming. If disabled, DISA is released when the trunk-to-trunk
connection is made.
The following three types of tone detection can be enabled for each trunk group to disconnect a trunk-to-
trunk call via DISA.
– Silence Detection
→ 11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Tone Detection—DISA
Tone Detection—Silence
– Continuous Signal Detection
→ 11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Tone Detection—DISA
Tone Detection—Continuous
– Cyclic Signal Detection
→ 11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Tone Detection—DISA
Tone Detection—Cyclic
• Trunk-to-Trunk Call Duration Limitation
For a call between two outside parties, even if end of call detection cannot be performed, the call can be
disconnected by a system timer. (→ 2.11.8 Trunk Call Limitation) If the timer expires, the line will be
disconnected unless the originating caller extends the time by sending any DTMF signalling. The caller
can prolong the call duration within the preprogrammed time period and preprogrammed number of times.
10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf
→ DISA—CO-to-CO Call Prolong Counter
→ DISA—CO-to-CO Call Prolong Time
• Automatic DISA Activation
DISA can be set through system programming to automatically activate for the following types of trunk-to-
trunk call, to enable detection of the end of the call.
– When a trunk call is forwarded to another trunk
– When a trunk call is transferred to another trunk
– When a trunk call to an incoming call distribution group is answered by an outside destination member
Before the call is made, the PBX confirms that a DISA port is available. If no DISA ports are available, the
call is not routed to a trunk. For transferred calls or calls to an ICD Group, if the DISA port has become
unavailable when the trunk-to-trunk conversation is actually established, the call is established without
DISA.
When using this feature, the Trunk-to-Trunk Call Limitation timer should be enabled. In addition,
prolonging the call through DTMF signalling is not available.
• DISA Call Transfer from Outside Destination
An outside party such as a cellular phone can transfer a trunk call to an extension (including TIE) or an
outside party by pressing "#" + extension number (including TIE) or an outside party’s number, if DISA is
connected by the Automatic DISA Activation feature. This feature can be enabled or disabled through
system programming. With NSX, this feature can only be used from a Cell Phone (Public device)
assigned to the User Container. (→ 2.29.1 User Container)
It is also possible to establish a Conference call (→ 2.14 Conference Features), perform Call Splitting (→
2.13.3 Call Splitting), and page with a call on hold to transfer the call (→ 2.17.1 Paging).

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– DISA security mode should be set to No Security.

– If the called extension does not answer, is busy, or is in DND mode, the DISA Intercept feature
operates.
– The party on hold can use the Call Retry feature.
– If the destination trunk supports End of Call Detection, a paging call can be made after dialling "#".
– If the transferred call is forwarded to another outside destination, COS settings are ignored.
• If the cellular phone’s number is registered in system speed dialling, its COS will be the COS of the
extension specified as the CLI destination.
• If the cellular phone’s number is not registered in system speed dialling, its COS will be the COS of
the trunk that the transferrer is using.
• Operation is as follows, depending on the "10.7.1 PBX Configuration—[2-7-1] System—Class of
Service—COS Settings—CO & SMDR—Transfer to CO" setting and the availability of End of Call
Detection on the destination trunk:
– Case 1: Transfer to CO is enabled.
Transfer is available.
– Case 2: Transfer to CO is disabled and the destination trunk supports End of Call Detection
A reorder tone is heard. (Consultation hold can be released by pressing "#".)
– Case 3: Transfer to CO is disabled and the destination trunk does not support End of Call
Detection.
Transfer is cancelled, and the conversation returns to the trunk on hold.
– If the call is transferred to an ICD group, the call will wait in a queue until answered, but the Queuing
Time Table will not function.
– When using this feature, do not use the Executive Busy Override feature to interrupt the established
call.
Redial with DTMF " " when receiving a trunk call (before the transfer destination—cellular phone—
answers).
– It is possible to redial by using the DTMF tone " " before the transfer destination trunk (cellular phone)
answers.
– If the transfer destination (cellular phone) is a line that does not support answer notification, outgoing
dial completion is considered as answering.
– If the destination trunk supports End of Call Detection, screened transfer is available. If the transferring
party dials "#" while talking to the called party, the call can be put back on consultation hold.
– If the destination trunk does not support End of Call Detection, the call is transferred unscreened.
• DISA Call Transfer to Outside User
When a call is received through DISA and the receiving extension is set to forward the call to an
extension (including over a TIE connection) or an outside destination, the call can be forwarded
automatically to an outside telephone number if the following conditions are met:
– An outside telephone number (e.g., a cellular phone number) is registered as the forward destination of
the receiving extension.
– The forward destination’s telephone number is registered in the System Speed Dial of the receiving
extension.
– The forward destination’s telephone number is set in the CLI Destination of the receiving extension.
– Automatic Walking COS is enabled for the receiving extension.
After a conversation is established with the forward destination (e.g., a cellular phone number), it is
possible to establish a Conference call, perform Call Splitting, and page with a call on hold to transfer the

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call. This is the same as DISA Call Transfer from Outside Destination. For details, see "DISA Call
Transfer from Outside Destination".

[Example]
a. Outside Caller calls Extn. 101 through DISA.
b. Extn. 101 forwards the call to Cellular Phone-1.
Outside Caller establishes a conversation with Cellular Phone-1.
c. Cellular Phone-1 presses "#" to put the conversation on hold, and then transfers the call to Cellular
Phone-2.
At this point, the CLIP information shown on Cellular Phone-2 may be one of the following:

Case 1: When Extn. 101 forwarded the call, Automatic Walking COS was performed with Cellular
Phone-1’s telephone number.
– Displayed CLIP: Same as if Extn. 101 made a trunk call.

Case 2: When Extn. 101 forwarded the call, Automatic Walking COS was not performed, and Option
4 is set to Enable.
– Displayed CLIP: Outside Caller’s telephone number and name.

Case 3: When Extn. 101 forwarded the call, Automatic Walking COS was not performed and Option
4 is set to Disable.
– Displayed CLIP: The CLIP set for the line that Cellular Phone-1 used to transfer the call.

Cellular Phone-2

Trunk Telephone Company Trunk

Outside Caller
Trunk Trunk
Telephone Transfer to
c. Company Cellular Phone-2
PBX
a.
Trunk Trunk
Cellular Phone-1
b.

Forward to
Cellular Phone-1

Extn. 101

• DISA Reorder Tone Duration

It is possible to set the DISA Reorder Tone Duration time. This specifies the length of time that a reorder
tone will be sent to the caller. When the timer expires, the call will be disconnected. Call Retry is possible
during the DISA Reorder Tone Duration time.
→ 10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf—DISA
—Reorder Tone Duration

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• Call Deny
Extensions can deny DISA calls on a COS basis.
→10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Optional Device &
Other Extensions—Accept the Call from DISA
• Verification Code PIN Lock/Extension PIN Lock
If the wrong PIN is entered three times, the line will be disconnected. If the wrong PIN is entered a
preprogrammed number of times successively, that extension or verification code will become locked, and
even entering the correct PIN will not unlock it. Only an extension assigned as the manager can unlock it.
In this case, the PIN will be unlocked and cleared.
→ 10.12 PBX Configuration—[2-13] System—Security
• DISA Automatic Walking COS
Mobile user or Advanced user can use this feature. (→ 2.29.1 User Container)
• Each outgoing message (OGM) can be assigned a name through system programming for programming
reference.
→ 13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message—Name

PC Programming Manual References

2.1.4 Easy Setup Wizard
10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf
→ DISA—Delayed Answer Timer
→ DISA—Mute & OGM Start Timer after answering
→ DISA—No Dial Intercept Timer
→ DISA—2nd Dial Timer for AA
→ DISA—Intercept Timer—Day, Lunch, Break, Night
→ DISA—Disconnect Timer after Intercept
→ DISA—CO-to-CO Call Prolong Counter
→ DISA—CO-to-CO Call Prolong Time
→ DISA—Progress Tone Continuation Time before Recording Message
→ DISA—Reorder Tone Duration
10.12 PBX Configuration—[2-13] System—Security
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Optional Device & Other
Extensions—Accept the Call from DISA
11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings
→ Tone Detection—DISA Tone Detection—Silence
→ Tone Detection—DISA Tone Detection—Continuous
→ Tone Detection—DISA Tone Detection—Cyclic
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Intercept Destination—
Intercept Destination—When called party does not answer—Day, Lunch, Break, Night
13.3.1 PBX Configuration—[5-3-1] Optional Device—Voice Message—DISA System
13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message

Feature Manual References

2.1.1.6 Intercept Routing
2.1.1.7 Intercept Routing—No Destination
2.3 Call Forwarding (FWD)/Do Not Disturb (DND) Features
5.1.1 Class of Service (COS)
5.5.7 Floating Extension

Operating Manual References

1.2.8 To Access Another Party Directly from Outside (Direct Inward System Access [DISA])

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2.16.2 Automatic Fax Transfer

1.2.9 Setting Your Telephone from Another Extension or through DISA (Remote Setting)

2.16.2 Automatic Fax Transfer

Description
The PBX can distinguish between fax calls and other types of calls arriving on DISA lines, and automatically
transfer fax calls to preprogrammed destinations. When a call arrives on a DISA line, an OGM is played
(→ 2.28.2 Outgoing Message (OGM)). At the same time, the PBX begins fax signal detection. If a fax signal
is detected, the PBX recognises that the call is a fax call, and transfers the call to the fax destination
assigned to that OGM through system programming. This allows a single trunk to be used seamlessly for
both voice and fax calls, with only voice calls arriving at user extensions.

[Available Automatic Fax Transfer Destinations]

Destination Availability
Wired Extension (PT/SLT/ISDN Extension/T1-OPX) ü
PS ü *1
Incoming Call Distribution Group ü
SIP Extension
PS Ring Group
UM Group ü
Fax Unit ü
External Pager (TAFAS)
DISA
Idle Line Access no. + Phone no.
Trunk Group Access no. + Trunk Group no. + Phone no.
Other PBX Extension (TIE with no PBX Code)
Other PBX Extension (TIE with PBX Code)

*1 A PS destination can be used to forward fax calls to a fax machine at another PBX connected by TIE line.
A public device name can be specified as the destination of fax calls. Then, the extension number of the fax machine at the other
PBX can be specified as the FWD—ALL Calls destination for calls to that public device name.

Conditions
• This feature is only effective for calls arriving on DISA lines.
• If a fax signal is not detected before the DISA Intercept Routing—No Dial timer expires, the call is
redirected to the operator extension, and fax detection ends.
• If the fax tone (CNG signal) detection is delayed because of the fax machine type or the state of the line,
the DISA intercept timer may time out and the fax cannot be received. In this case, increasing the DISA
intercept timer by 5 to 10 seconds will help avoid this problem.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf—DISA—
Intercept Timer—Day, Lunch, Break, Night
13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message—Fax Extension

176 Feature Manual

 2.16.2 Automatic Fax Transfer

Feature Manual References

2.16.1 Direct Inward System Access (DISA)
3.1.3 Fax Server

Feature Manual 177

2.17 Paging Features

2.17 Paging Features

2.17.1 Paging
Description
NSX has no External Paging ports, so when you need to connect External music source, you can use
expansion gateway units which has external paging port.
An extension user can make a voice announcement to many destinations simultaneously.
The message is announced over the built-in speakers of PTs and/or external speakers (external pagers)
which belong to the paging group (The Expansion Gateway is required when using an external page). The
PBX can connect to one external pager.
The paged person can answer the page from a nearby telephone.
It is possible to page with a call on hold in order to transfer the call.
Paging Deny:
An extension user can choose not to receive paging announcements.

Paging Group
Each paging group consists of extension user groups and external pagers. One extension user group or
external pager can belong to several paging groups. In a Multi-connection network, external pagers at any
site can be assigned to a paging group.
(→ 5.1.2 Group)

[Example]

Paging Group 01 Paging Group 02 Paging Group 03

Extension Extension Extension

User Group 1 User Group 2 User Group 3

Extn. 100 Extn. 101 Extn. 102 Extn. 103 Extn. 104 Extn. 105 Pager 01 Pager 02

[Programming Example]
*1 *2
Extension User Group No. External Pager
Paging Group No. Pager Pager Pager
001 002 003 ... ...
01 02 03
01 ü ... ...
02 ü ü ... ü ...
03 ü ... ü ü ...
04 ... ...
05 ü ü ü ... ü ü ü ...
: : : : ... : : : ...

ü: Constituent
*1 → 11.4.1 PBX Configuration—[3-4-1] Group—Paging Group—Priority Setting
*2 → 11.4.2 PBX Configuration—[3-4] Group—Paging Group—External Pager

178 Feature Manual

 2.17.1 Paging

Conditions
• Paging announcements cannot be heard at the following types of extensions:
– PSs
– SLTs
– Ringing or busy PTs
– PTs in Paging Deny mode
– PTs in Paging DND mode
– SIP phones
Although paging announcements cannot be heard at these types of extensions, they can answer paging
announcements.
*1 Changing the extension user group of the extension, allows it to receive pages. However, doing so may affect the number of
simultaneous IP extension and IP trunk calls available on the mother board.

• External Pager Priority

External pagers can be used with the following priorities:
TAFAS → Paging → BGM
(→ 2.17.2 Trunk Answer From Any Station (TAFAS), 2.28.1 Background Music (BGM))
• Volume Control
Paging volume from the PTs and the external pagers can be changed through system programming.
• Paging DND
When DND (→ 2.3.3 Do Not Disturb (DND)) is set for incoming calls, it is programmable whether your
extension receives paging through system programming.
• Multi-connection Network (→ 4.2 Multi-connection Networking)
In a Multi-connection network, pagers are numbered sequentially based on the site number and the
pager's port.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Group Paging
→ Group Paging Answer
→ Paging Deny Set / Cancel
10.9 PBX Configuration—[2-9] System—System Options
→ Option 1—PT Fwd / DND—Paging to DND Extension
→ Option 3—Confirmation Tone—Tone 2 : Paged / Automatic Answer
10.11.1 PBX Configuration—[2-11-1] System—Audio Gain—Paging/MOH
→ Paging—EPG 1–6 (External Pager 1–6)
→ Paging—Paging Level from PT Speaker
11.4 PBX Configuration—[3-4] Group—Paging Group
11.4.1 PBX Configuration—[3-4-1] Group—Paging Group—Priority Setting
11.4.2 PBX Configuration—[3-4] Group—Paging Group—External Pager
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 6—Paging Deny

Feature Manual References

6.1 Capacity of System Resources

Operating Manual References

1.7.1 Paging
1.7.2 Answering/Denying a Paging Announcement

Feature Manual 179

2.17.2 Trunk Answer From Any Station (TAFAS)

2.17.2 Trunk Answer From Any Station (TAFAS)

Description
When a call is received at the floating extension number assigned to the external pager, a ring tone is sent
through the pager. Any extension user can then answer the call.

Conditions
• Hardware Requirement: An Expansion Gateway and External Pager need to be prepared because the
NSX has no external page port.
• Floating Extension Number
A floating extension number must be assigned for an external pager (default: 600 or 6000*1). It is possible
to access an external pager by dialling its floating extension number.
• Pager Volume
It is possible to change the volume of an external pager through system programming.
*1 The default floating extension number depends on the value specified for Numbering Plan in Easy Setup.
→ 2.1.4 Easy Setup Wizard

Installation Manual References

4.5 Connection of Peripherals

PC Programming Manual References

2.1.4 Easy Setup Wizard
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—TAFAS Answer
10.11.1 PBX Configuration—[2-11-1] System—Audio Gain—Paging/MOH—Paging—EPG 1–6 (External
Pager 1–6)
13.2 PBX Configuration—[5-2] Optional Device—External Pager

Feature Manual References

5.5.7 Floating Extension

Operating Manual References

1.3.4 Answering a Call via an External Speaker (Trunk Answer From Any Station [TAFAS])

180 Feature Manual

 2.18 External Device Features

2.18 External Device Features

2.18.1 Doorphone Call
Description
It is possible to connect doorphones directly to the PBX. When a visitor presses the call button on a
doorphone, the doorphone calls a preprogrammed destination (extension or outside party). In addition,
extension users can dial the preset number of a doorphone to call that doorphone.

Conditions
• Hardware Requirement:
An optional doorphone and a DOORPHONE card
To connect a doorphone to a legacy gateway (→ 5.3.1 Stacking Connection), refer to the documentation
of the legacy gateway.
• Each doorphone port can only be assigned to one tenant. The Time Table (day/lunch/break/night) of the
tenant applies. (→ 5.1.4 Time Service)
• Call Destination
The incoming doorphone call destination(s) can be assigned for each time mode (day/lunch/break/night)
for each doorphone port. Destinations can be selected. (→ 2.1.2.1 Internal Call Features—SUMMARY)
• COS programming determines the doorphone ports that are able to make an outgoing trunk call.
• Internal Call Block determines which extensions can call a doorphone. (→ 2.1.2.2 Internal Call Block)
• Ring Duration
If an incoming call is not answered within a preprogrammed time period, ringing stops and the call is
cancelled.
• Call Duration
The call duration can be restricted by a system timer. If the timer expires, the call will be disconnected.
• Door Open
While engaged on a doorphone call, the extension user can unlock the door to let the visitor in.
(→ 2.18.2 Door Open)
• A doorphone number can be referenced for each doorphone port.
• Communication IP Camera/Video Door Phone
The doorphone feature can be used from the Communication IP Camera and Video Door Phone. (→
2.18.5 Communication IP Camera/Video Door Phone)

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf
→ Doorphone—Call Ring Duration
→ Doorphone—Call Duration
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—DOORPHONE Call
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—TRS—TRS Level—Day,
Lunch, Break, Night
10.8.2 PBX Configuration—[2-8-2] System—Ring Tone Patterns—Call from DOORPHONE
10.9 PBX Configuration—[2-9] System—System Options—Option 3—Confirmation Tone—Tone 1 : Called
by Voice
13.1 PBX Configuration—[5-1] Optional Device—Doorphone

Operating Manual References

1.11.1 If a Doorphone/Door Opener is Connected

Feature Manual 181

2.18.2 Door Open

2.18.2 Door Open

Description
An extension user can unlock the door for a visitor using his telephone.
The door can be unlocked by extension users who are allowed to unlock the door through COS
programming. However, while engaged on a doorphone call, any extension user can unlock the door to let
the visitor in (→ 2.18.1 Doorphone Call).

Conditions
• Hardware Requirement:
A user-supplied door opener on each door, and a DOORPHONE card.
To connect a door opener to a legacy gateway (→ 5.3.1 Stacking Connection), refer to the documentation
of the legacy gateway.
• The door opener will unlock the door even if a doorphone is not installed.
• Door Open Duration
The door can remain unlocked for a preprogrammed time period.
If the door opener is a type that locks automatically when the door is closed, it is recommended that Door
Open Duration be set to 2 seconds.
• Video Door Phone
The Door Open feature can be used from the Video Door Phone. (→ 2.18.5 Communication IP Camera/
Video Door Phone)

PC Programming Manual References

9.29 PBX Configuration—[1-1] Configuration—Slot—DOORPHONE Card—Card Property—For Output -
Device Type
10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf—
Doorphone—Open Duration
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Door Open
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Optional Device & Other
Extensions—Door Unlock

Feature Manual References

5.1.1 Class of Service (COS)

Operating Manual References

1.11.1 If a Doorphone/Door Opener is Connected

2.18.3 External Sensor

Description
External sensing devices, such as security alarms or smoke detectors, can be connected to the PBX. When
the PBX receives input from a sensor, a call is made to the preset destination, alerting the extension user.
The available destinations of a sensor call are as follows:

[Available Destinations]
Destination Availability
User (Main device) ü
User (Sub device) ü

182 Feature Manual

 2.18.3 External Sensor

Destination Availability
User (Wireless device) ü
User (Public device)
Incoming Call Distribution Group ü
PS Ring Group ü
UM Group
Fax Unit
External Pager (TAFAS) ü
DISA
Idle Line Access no. + Phone no. ü
Trunk Group Access no. + Trunk Group no. + Phone no. ü
Other PBX Extension (TIE with no PBX Code) ü
Other PBX Extension (TIE with PBX Code) ü

When the call is answered, if distinctive dial tones are enabled, dial tone 3 is heard, and continues until the
user goes on-hook. If the sensor call is not answered within a specified time, the call will be cancelled. It is
possible to set a different ring tone pattern for calls received from each external sensor, to distinguish
between them.

Conditions
• Hardware Requirement:
An external sensor and a DOORPHONE card
To connect an external sensor to a legacy gateway (→ 5.3.1 Stacking Connection), refer to the
documentation of the legacy gateway.
• Some devices may be unable to communicate correctly with the PBX. Confirm compatibility with the
manufacturer of a device before installing it.
• After a sensor has been activated, the PBX will ignore any further alerts from the same sensor for the
duration specified by a timer. This timer can be set separately for each sensor.
• As long as the previous sensor call is still being performed, any further alerts from the same sensor are
ignored.
• The assigned sensor name and/or number are shown on the display of PTs and PSs when a sensor call
is received.
• If the destination of a sensor call has set FWD, the sensor call will be redirected to the FWD destination.
However, if the FWD destination is not supported as the destination of a sensor call (e.g., an outside
party), the call will be received at the original destination. (→ 2.3.2 Call Forwarding (FWD))
• The following features cannot be used when a sensor call is received:
– Alternate Receiving—Voice (→ 2.5.3 Intercom Call)
– Hands-free Answerback (→ 2.4.4 Hands-free Answerback)
– Consultation Hold/Call Hold (→ 2.13.1 Call Hold)
– Call Transfer (→ 2.12.1 Call Transfer)
– Executive Busy Override (→ 2.10.2 Executive Busy Override)
• Sensor call information is output on SMDR.

Feature Manual 183

2.18.4 External Relay Control

PC Programming Manual References

9.29 PBX Configuration—[1-1] Configuration—Slot—DOORPHONE Card—Card Property
→ For Sensor - Input Signal Decision Time
→ For Sensor - Input Signal Detection Reopening Time
10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—External Sensor—Ring
Duration
10.8.3 PBX Configuration—[2-8-3] System—Ring Tone Patterns—Call from Others—External Sensor—Ring
Tone Pattern Plan 1–8
13.5 PBX Configuration—[5-5] Optional Device—External Sensor

2.18.4 External Relay Control

Description
By turning external device relays on and off, the PBX can control external devices such as alarms.
When an extension user enters the External Relay Control feature number, the specified relay turns on for a
preprogrammed length of time. When this timer expires, the relay turns off automatically. This gives the PBX
simple control over other equipment, allowing an extension user to, for example, activate an alarm from his
extension.
If the relay cannot be accessed (for example, because use is not permitted by COS, or the port is not in
service), a reorder tone will be heard at the extension.

Conditions
• Hardware Requirement:
An external relay device and a DOORPHONE card.
To connect an external relay to a legacy gateway (→ 5.3.1 Stacking Connection), refer to the
documentation of the legacy gateway.
• The port of the DOORPHONE card to which the relay is connected must be assigned through system
programming as a relay port (not a door opener port).
• Some devices may be unable to communicate correctly with the PBX. Confirm compatibility with the
manufacturer of a device before installing it.
• Each external relay port has a COS assigned. This and the COS of an extension determine the extension
users who can use External Relay Control.
• The length of time that a relay is turned on can be specified separately for each relay through system
programming.
• If the same or another extension tries to access an external relay that has already been switched on, the
timer for that relay is reset.

PC Programming Manual References

9.29 PBX Configuration—[1-1] Configuration—Slot—DOORPHONE Card—Card Property—For Output -
Device Type
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—External Relay Access
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Optional Device & Other
Extensions—External Relay Access
13.4 PBX Configuration—[5-4] Optional Device—External Relay

Feature Manual References

5.1.1 Class of Service (COS)

184 Feature Manual

 2.18.5 Communication IP Camera/Video Door Phone

Operating Manual References

1.11.2 If an External Relay is Connected

2.18.5 Communication IP Camera/Video Door Phone

Description
KX-NTV100 series SIP extensions (e.g., KX-NTV150, KX-NTV160) can be registered to the NSX server as
a Communication IP Camera or Video Door Phone. KX-NTV100 series are SIP extensions with a call button
and a camera, and they can make and receive calls in the same way as other SIP extensions. When the call
button is pressed, the dial sequence set in the Communication IP Camera or Video Door Phone will be sent
to the NSX server, which will then call the appropriate extension.

KX-NTV100 series SIP extensions have the following features:

• KX-NTV150 and KX-NTV160 have the Doorphone Call feature. However, they cannot be linked to legacy
doorphones. (→ 2.18.1 Doorphone Call)
• KX-NTV160 has the Door Open feature. However, it cannot be linked to legacy doorphones. (→
2.18.1 Doorphone Call)
• Upgrading the firmware for KX-NTV series SIP extensions is performed thorough the device. For details,
refer the manual for the device.
• When the door open feature is operated using an IP-PT (KX-NT300, etc.) or a SIP phone that does not
support RFC2833, the telephone must not belong to the same P2P group as that of the door phone.

Conditions
[General]
• KX-NTV150 and KX-NTV160 require a Panasonic device activation key (KX-NSXN001/KX-NSXN010).
• KX-NTV150 and KX-NTV160 require a V-SIPEXT card.
• The maximum number of KX-NTV150 and KX-NTV160 that can be registered is 128.
• KX-NTV150 and KX-NTV160 are registered as an optional IP device rather than a user device.
Optional IP devices are not assigned to a user container, but to an optional IP device container.
• KX-NTV150 and KX-NTV160 are counted as SIP extensions. The total number of SIP extensions,
including KX-NTV150 and KX-NTV160, cannot exceed the maximum number of SIP extensions.
• The dial sequence assigned to the call button of KX-NTV150 and KX-NTV160 can include feature
numbers, outside line access numbers, etc. NSX server does not restrict the dial sequence.
[Camera Control]
• KX-NTV150 and KX-NTV160 support camera control using Outband DTMF signals while talking or
monitoring video. This feature enables users to control the camera using their extension's key pad.
• Camera control is available from the following devices:
– KX-NT500 series IP-PTs
– KX-HDV series SIP phones
– Third party SIP phones that comply with RFC2833
For devices connected to NSX server, this feature is only available for the recommended third party
SIP phones which have a video monitoring feature.

PC Programming Manual References

13.6 PBX Configuration—[5-6] Optional Device—Optional IP Device

Feature Manual 185

2.18.5 Communication IP Camera/Video Door Phone

Operating Manual References

1.11.1 If a Doorphone/Door Opener is Connected

186 Feature Manual

 2.19 Caller ID Features

2.19 Caller ID Features

2.19.1 Caller ID
Description
The PBX receives caller information, such as the caller’s name and telephone number, through the trunk.
This information can then be shown on the displays of PTs, PSs, or SLTs that support FSK-type Caller ID.
The PBX can modify a received number according to preprogrammed tables, so that an extension user can
easily use the received number to call the caller back. For example, if an area code is not required to call
outside destinations in a certain area, but received Caller ID numbers from that area contain an area code, it
is possible to store that area code in a modification table so that it is deleted automatically from received
numbers (Automatic Caller ID Number Modification).
1. Features
Caller ID includes the following features:
Feature Description & Reference
Caller ID Caller’s information which is sent from an analogue trunk.
The following Caller ID signalling types are supported: FSK and
DTMF.
Calling Line Identification Caller’s information which is sent from an ISDN line.
Presentation (CLIP)
→ 4.1.2.2 Calling/Connected Line Identification Presentation
(CLIP/COLP)
Automatic Number Caller’s information which is sent from an E1 or T1 line (stacking
Identification (ANI) connection only).

→ 5.3.1 Stacking Connection

2. Service Features
Features Description & Reference
Calling Line Identification Directs a DIL/DID/DDI/MSN call to a CLI destination if the caller’s
(CLI) Distribution identification (Caller ID/CLIP/ANI) has been assigned to the Caller
ID Table.

→ 2.1.1.5 Calling Line Identification (CLI) Distribution

Incoming Call Log Caller’s information is automatically recorded in the call log of the
called extension. This information is used for confirming the caller,
calling the caller back, and/or storing the number and name in the
Personal Speed Dialling.

→ 2.19.2 Incoming Call Log

3. Number/Name Assignment
Automatic Caller ID Number Modification
This PBX automatically modifies the incoming caller’s number according to preprogrammed tables. The
modified number will be recorded for calling back.
This PBX supports 4 modification tables, each of which can be used for any number of trunk groups.
Each table has 10 locations for local/international calls and one for long distance calls. The PBX checks
the local/international call data first. If a match is not found, the long distance call data is applied.

Feature Manual 187

2.19.1 Caller ID

After the caller’s number is modified by the Length of Digits modification tables or CLIP modification
tables, the PBX checks the leading digits of the modified number for an area code programmed in the
Caller ID modification table assigned to that trunk group. For more information, refer to 11.1.3 PBX
Configuration—[3-1-3] Group—Trunk Group—Caller ID Modification—Leading Digits in PC
Programming Manual.

[Example]
<Table Selection>
Trunk Modification
Group No. Table <Modification Table>
1 1 Modification Table 1
2 3 Area Code Removed No. of Digits Added No.
: : Local/International
012 3 Blank
Call Data 1
Local/International
00 2 001
Call Data 2
: : : :
Local/International
Call Data 10
Long Distance Not
0 0
Call Data programmable

Note
When caller’s information is sent through an ISDN line and the call type is Subscriber, National, or
International, the following modification table is used instead of the above table:

<Modification Table>
Removed No. of Digits Added No.
Subscriber Call Data 0 Blank
National Call Data 0 0
International Call Data 0 00

188 Feature Manual

 2.19.1 Caller ID

<Modification Flowchart>

A trunk call with the caller's

information is received.

Checks the Table Selection.

Table 1

Is the area code found in

Yes: local/international call data
No:
e.g., 00987654321 in the modification table?
e.g., 3344556677
Checks the local/
international call data. Checks the long distance call data.
Matches (Data 2)

Modifies the number as programmed.

Removed number of digits: 2
Added number: 001 Modifies the number as programmed.
Received number: 00987654321 (Removed number of digits: 0,
Added number: 0)
Modified number: 00987654321 = 001987654321

1) Remove the 2) Add "001".

first 2 digits.

Modification is completed:
Modification is completed: 03344556677.
001987654321.

Caller ID Table Assignment

The System Speed Dialling Table is also used as the Caller ID Table.
In each location of the table, the following items can be assigned:
a. Telephone number (Trunk Access number + caller’s telephone number)
b. System Speed Dialling name (caller’s name)
(shown on the display or SMDR)
c. CLI destination
(used for CLI feature)
When a caller’s modified telephone number matches a telephone number (the Trunk Access number is
disregarded.) in the table, the call is sent to the assigned CLI destination.

[Example]
Location (System Speed *1 System Speed Dialling *3

Dialling No.) Telephone No. *2 CLI Destination

Name
000 90123456789 ABC Company 200
001 : : :

Feature Manual 189

2.19.1 Caller ID

Location (System Speed *1 System Speed Dialling *3

Dialling No.) Telephone No. *2 CLI Destination

Name
: : : :

*1 → 14.1 PBX Configuration—[6-1] Feature—System Speed Dial—CO Line Access Number + Telephone Number
*2 → 14.1 PBX Configuration—[6-1] Feature—System Speed Dial—Name
*3 → 14.1 PBX Configuration—[6-1] Feature—System Speed Dial—CLI Destination

Caller’s Name Reference

A name can also be shown on the display or SMDR. The PBX searches for the name in the following
order:
1. Personal Speed Dialling data of the original called extension
2. System Speed Dialling (Caller ID) Table
3. Caller ID name received from the public line (Caller ID Name Reference)
If the name is not found, it will not be displayed.

Conditions
[General]
• Caller ID signalling type can be selected through system programming.
• The Caller ID Name Reference is only available for calls from the public network.

[Caller ID to SLT Port]

• Hardware Requirement:
An LCOT2+SLC2, BRI4+SLC2, PRI30+SLC2, or PRI23+SLC2 card
For details about the hardware requirements for a legacy gateway (→ 5.3.1 Stacking Connection), refer to
the documentation of the legacy gateway.
• This feature complies with ETSI (European Telecommunications Standards Institute)-type FSK and
Bellcore-type FSK.
• For SLC-type cards, you can set the "Ringing Start Mode of Extension Caller ID" setting as follows.
– Mode 1: Enables pre-ringing. Pre-ringing control is performed when a call with caller ID information is
received. Even if pre-ringing is enabled, the FSK signal is sent between the first and second rings.
– Mode 2: Disables pre-ringing (caller ID information is sent after the first ring).
Changes to this setting are effective immediately. For more information, refer to the documentation of
the legacy gateway.
• When the caller’s number is sent to an SLT, a Trunk Access number can be automatically added to the
telephone number through system programming for calling back.
• When the caller’s number exceeds 16 digits, the SLT receives only the first 16 digits, not counting the
preceding Trunk Access number (if it is programmed to be added).
• If a call is transferred to an SLT, the transferring extension’s information will be shown on the SLT. If the
transferring extension goes on-hook before the call is answered, the original caller’s information will be
shown.
• When the Caller ID has information, such as private, out of area, or long distance, the information will be
shown instead of the caller’s number and name.
• Even if the caller’s name is sent, the name may not be shown depending on the type of SLT.

190 Feature Manual

 2.19.2 Incoming Call Log

• Incoming Call Log information is not shown on the SLT.

• Caller ID shows whether the call is an intercom or trunk call by default. This setting can be disabled
through system programming.
• A caller ID name received from the network via a PRI23 card will not be displayed on an SLT. However, if
the received caller ID matches a System Speed Dialling entry, and a name is registered for that entry, it
will be displayed.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous
→ Caller ID—Waiting to receive
→ Caller ID—Visual Caller ID Display
10.9 PBX Configuration—[2-9] System—System Options—Option 4—Private Network—Public Call through
Private Network—Minimum Public Caller ID Digits
10.10 PBX Configuration—[2-10] System—Extension CID Settings
11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Main—Caller ID Modification
Table
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 8
→ Extension Caller ID Sending
→ Incoming Call Wait Timer for Extension Caller ID
14.1 PBX Configuration—[6-1] Feature—System Speed Dial
11.1.3 PBX Configuration—[3-1-3] Group—Trunk Group—Caller ID Modification

Feature Manual References

2.6.4 Speed Dialling—Personal/System
2.22.1.1 Station Message Detail Recording (SMDR)

2.19.2 Incoming Call Log

Description
When an incoming public trunk call with the caller’s information (e.g., Caller ID) is directed to an extension,
the information is automatically recorded in the call log of the called extension. Also, depending on system
programming, incoming intercom calls (including over a TIE connection) are logged in the incoming call log.
This information is shown on the telephone display and is used for confirming the caller, calling back, or
storing the number and name in the Personal Speed Dialling.
The Incoming Call feature can be used with DPT, IP-PT, PS, IP-Softphone and APT. However, for PSs, this
feature is available only with models that distinguish between intercom calls and trunk calls in the incoming/
outgoing call logs.

Feature Manual 191

2.19.2 Incoming Call Log

[Example]

John White --- Caller’s name (20 digits max.)*1

DEC.12 10:00AM MON --- Date and time of a call received
NEW: Not Answered --- Answering Status*2
123456789 --- Caller’s number (16 digits max.)

Call Log buttons

Own extension
Incoming call distribution group

*1: If a call is received from an extension and no name is assigned to the extension,
the incoming call log shows the extension number.
*2: "NEW" is displayed for call records which have not previously been viewed;
"OLD" is displayed for call records which have previously been viewed.

Conditions
• Call Log Button
A flexible button can be customised as the Call Log button for the extension or an incoming call
distribution group. The button light shows the current status as follows:
Light pattern Status
Red on There is unchecked information.
Off All information has been checked.

• If the answering destination is not the original extension (FWD—No Answer, Intercept Routing—
No Answer, Overflow, and Call Pickup):
If a call is forwarded because it is not answered or another extension picks up the call, the information is
logged in the call logs of both the original destination and the answering destination. If a call is forwarded
to several extensions before being answered, the information is logged in the call logs for all the
extensions it was forwarded to. If a call is forwarded to an incoming call distribution group and is not
answered, the information is not logged in the call log for the incoming call distribution group.
• The following types of calls will be recorded as "Not Answered" in the incoming call log of the original
destination:
– Calls received when the extension is in use (the caller hears a busy tone).
– Calls rerouted using the Intercept Routing—Busy, FWD—All Calls, or FWD—Busy features.
If disabled through system programming, these types of calls will not leave a record in the incoming call
log.
It is also possible to specify through system programming if calls answered using Call Pickup are
recorded as "Not Answered" or "Answered" in the incoming call log of the original destination.
• Call Log for PS Calls
If a PS or a CS is in one of the following situations when a call arrives, the information is logged in the call
log for the PS:
The PS Incoming Call Log feature is only provided on models that can distinguish between calls from
outside lines and calls from extensions
a. When the PS is out of range.
b. When the PS is turned off.
c. When the CS is busy.

192 Feature Manual

 2.19.2 Incoming Call Log

• Display Lock
An extension user can lock the Incoming Call Log display to prevent the call information from being
viewed at any extension through personal programming (Display Lock). In this case, the Outgoing Call
Log display and the Personal Speed Dialling number display are also locked. An extension personal
identification number (PIN) is required to use this feature. (→ 2.24.1 Extension Personal Identification
Number (PIN))
• Storing the Call Log Information in Personal Speed Dialling
When storing the number and name into Personal Speed Dialling from the call log information, the Idle
Line Access number or the TIE Line Access number is automatically attached to the telephone number.
• Storing the Call Log Information from an Extension
Depending on system programming, the information about an extension (including over a TIE connection)
logged in the incoming call log can be stored in Personal Speed Dialling.
• Incoming Call Log Memory
The total memory for the Incoming Call Log is determined in the PBX. The maximum number that can be
logged for each extension and incoming call distribution group is also determined through system
programming. If the memory becomes full, the new call record overwrites the oldest one.
• Call Log for Incoming Call Distribution Group Calls
If the original destination of a call is an incoming call distribution group, and the call is not answered, the
information is logged in the call log for the group. If it is answered, the information is logged in the call log
for the answering extension.
• Through system programming, it is possible to select which Incoming Call Logs record call information
when a call to an incoming call distribution group is answered by a member of the group:
– Only the Incoming Call Log of the extension that answered the call.
– Both the Incoming Call Log of the extension that answered the call and that of the incoming call
distribution group.
• E-mail Notification of Missed Calls
Extension users can receive an e-mail notification when they have a missed trunk call.
→ Contact—Email 1–3 in 8.1.1 Users—User Container—Add User/Edit User
→ Email notification in 8.1.1 Users—User Container—Add User/Edit User
• When the Incoming Call Log—Extension / TIE Call setting is enabled, the caller’s information (e.g.,
extension number) is logged in the incoming call log of the extension that answered the call.

PC Programming Manual References

10.9 PBX Configuration—[2-9] System—System Options
→ Option 7—Incoming Call Log—Extension / TIE Call
→ Option 7—Outgoing Call Log—Extension Call
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Miscellaneous—Supervisor Extension Number
11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous—Options—
Call Log to ICD Group when ICD Member Answered
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings
→ Option 5—Incoming Call Display
→ Option 7—Incoming Call Log Memory
8.3.1 Users—Flexible Button—Wired Extension—Type

Feature Manual References

2.6.4 Speed Dialling—Personal/System
2.21.2 Flexible Buttons
5.4.1 E-mail Notification for Extension Users

Feature Manual 193

2.19.2 Incoming Call Log

6.1 Capacity of System Resources

Operating Manual References

1.14.1 Using the Call Log
3.1.2 Settings on the Programming Mode

194 Feature Manual

 2.20 Message Features

2.20 Message Features

2.20.1 Message Waiting
Description
An extension user can notify another extension user that he wishes to talk to the user. The notified
extension user can return the call or listen to the messages recorded by the Unified Messaging system.
When a message is left on a PT, the Message button lights or the Message/Ringer Lamp turns on red, and a
message is shown on the display of a display PT. Pressing the lit Message button while on-hook shows the
caller’s information as shown below:

[Example]

105:Tom Smith --- Extension no. and name of who left the message

Message buttons
Own extension
Incoming call distribution group
Other extension*

*: For example, this button is useful

when the secretary checks the
message for the boss (Boss &
Secretary Feature).

Conditions
• Message Button
Pressing the Message button on the telephone unit makes a call to the built-in UM. While receiving an
incoming call or talking on the phone, it functions as the VM Transfer key. In the system settings, the
original behaviours of the Message button can be selected. When the Message button is set to the
Flexible button, the original behaviours will be performed. The following shows the original behaviours:
A flexible button can be customised as the Message button for the extension, other extensions, or an
incoming call distribution group.
• Distinctive Dial Tone for Message Waiting
If the Distinctive Dial mode is enabled, dial tone 4 will be sent to an extension when a message has been
left on the extension. (→ 2.25.1 Dial Tone)
• It is possible to set Message Waiting while hearing a ringback tone, busy tone, or DND tone.
• Messages are always left on the original destination extension, regardless of that extension’s FWD
settings.
• Both the extension that sent and received a message waiting notification can cancel the left message.
• If the extension that received a notification calls back the extension that sent the notification, and the call
is answered, the notification will be cleared automatically. However, if a voice message has been left in a
mailbox, whether the notification is cleared or not depends on the Unified Messaging settings or the
VPS’s settings.

[Panasonic KX-T7710/ KX-T7716]

• Message Waiting Lamp is supported by 80 to 100 VDC.

Feature Manual 195

2.20.2 Absent Message

• To use this feature, it must be enabled for each extension port through system programming. (→8.2.1
Users—Advanced Extension Settings—Advanced Extension Settings—Option 8—SLT MW Mode)

[Other Panasonic SLTs with a Message feature]

• Programming is not required to use this feature.
• Message Waiting Lamp is supported by the FSK signal. The SLT cards that support Message Waiting
Lamp (FSK) are as follows:
– Pre-installed SLC card (e.g., KX-NS300)
– MCSLC8 (KX-NS5173)
– MCSLC16 (KX-NS5174)
– EMSLC24 (KX-TDA6179)
– MCSLC16 (KX-TDA1176)
– MCSLC24 (KX-TDA1178)

[SIP Phones]
• To use this feature, it must be enabled for each extension port through system programming. (→8.2.1
Users—Advanced Extension Settings—Advanced Extension Settings—Option 8—SLT MW Mode)
• Only Standard type or Unsolicited type general SIP extensions can control message waiting indication
LEDs. To change the message waiting indication method, refer to 9.18 PBX Configuration—[1-1]
Configuration—Slot—V-SIPEXT128—Port Property—Main—MWI Method.

PC Programming Manual References

9.18 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Port Property
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 8—SLT MW Mode
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Message Waiting Set /
Cancel / Call Back
10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature—Message Waiting
Set
10.9 PBX Configuration—[2-9] System—System Options—Option 3—Dial Tone—Distinctive Dial Tone
10.9 PBX Configuration—[2-9] System—System Options—Option 5—SLT—Message Waiting Lamp Pattern
8.3.1 Users—Flexible Button—Wired Extension—Type
8.3.2 Users—Flexible Button—Portable Station—Type

Feature Manual References

2.21.2 Flexible Buttons
3.2 System and Subscriber Features
6.1 Capacity of System Resources

Operating Manual References

1.2.4 When the Dialled Line is Busy or There is No Answer
1.8 Using the Unified Messaging Features

2.20.2 Absent Message

Description
An extension user can set or select a message (e.g., the reason for absence) to be displayed on his
telephone. When a display PT user calls the extension, the message is shown on the caller’s telephone.
The following messages can be programmed as desired:

196 Feature Manual

 2.20.2 Absent Message

Message
Type Message (Example) Description
No.
System 1 Will Return Soon Messages may be edited
message through system programming.
2 Gone Home
They are used for every
3 At Ext %%%% (Extension extension user commonly.
Number)
4 Back at %%:%% (Hour:Minute)
5 Out until %%/%% (Month/
Day)
6 In a Meeting
7
8
Personal 9 A message is programmable at
message each extension through
personal programming
(Personal Absent Message),
which can only be used by that
extension user.

Note
The "%" means a parameter to be entered when assigning a message at an individual extension.
Up to seven "%"s can be stored for each message.

Conditions
• An extension user can select only one message at a time. The selected message is displayed at the
extension while on-hook.
• An extension user who has a Unified Messaging mailbox can also set his absent message from a remote
location by following the voice guidance (→ 3.2.2.28 Remote Absent Message).

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Absent Message Set /
Cancel
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 3—Absent Message
14.5 PBX Configuration—[6-5] Feature—Absent Message

Feature Manual References

6.1 Capacity of System Resources

Operating Manual References

1.6.2 Showing a Message on the Caller’s Telephone Display (Absent Message)
3.1.2 Settings on the Programming Mode

Feature Manual 197

2.21 Proprietary Telephone (PT) Hardware Features

2.21 Proprietary Telephone (PT) Hardware Features

2.21.1 Fixed Buttons
Description
PTs, DSS Consoles, and Add-on Key Modules are provided with the following feature/Line Access buttons:
Depending on your device type, some buttons may not be provided.
As for buttons on PSs, please refer to the Operating Instructions for each PS.

[PT and Add-on Key Module]

Button Usage
Used to adjust the ringer, speaker, handset and
headset volume and the display contrast. Navigator
Key and Jog Dial can also be used to select data from
the Call Directory and the System Feature Access
Menu on the display.

Navigator Key, Jog

Dial, Volume Key

ENTER Used to confirm the selected item.

CANCEL Used to cancel the selected item.

PROGRAM Used to enter and exit the programming mode.

Used to disconnect the current call and make another

call without hanging up (Flash/Recall mode or
Terminate mode) or used to send a flash/recall signal
FLASH/RECALL
to the telephone company or host PBX to access their
features (External Feature Access mode). This button
can also be used as a CANCEL button while on-hook.

HOLD Used to place a call on hold.

SP-PHONE Used for hands-free operation. Also used to switch

(Speakerphone) between handset and hands-free operation.

Used for a hands-free dialling. Also used to monitor

MONITOR
the party’s voice in hands-free mode.

198 Feature Manual

 2.21.1 Fixed Buttons

Button Usage
Used to leave a message waiting indication or call
MESSAGE back the party who left the message waiting
indication.
REDIAL Used to redial the last dialled number.

TRANSFER Used to transfer a call to another party.

Used to make or receive a trunk call or can be

Flexible CO (Trunk) reassigned to a different Trunk Access button
(Default: S-CO) or to another feature button.
INTERCOM Used to make or receive intercom calls.

Used to receive an incoming call in hands-free mode,

AUTO ANS (Auto
or used for microphone or handset mute during a
Answer)/MUTE
conversation. (Dual feature button)
Used to monitor an intercom call automatically (a
VOICE CALL/MUTE hands-free conversation is not possible). Also used for
handset microphone mute during a conversation.
Used for System/Personal Speed Dialling and storing
AUTO DIAL/STORE
programme changes.
CONF (Conference) Used to establish a multiple-party conversation.

FWD/DND Used to perform FWD or DND. (Dual feature button)

Used to insert a pause in a stored number. With an

PAUSE
APT, it is used as the PROGRAM button.
Used to select the item displayed on the bottom line of
Soft
the display.
Used to select the displayed item or to call the
SELECT
displayed number.
SHIFT Used to access the second level of Soft button items.
MODE Used to shift the display to access various features.
Used to switch the page for the Self Labelling feature
NEXT PAGE
(KX-NT366/KX-NT553/KX-NT556/KX-NT560 only).

[DSS Console]
Button Usage
Used to answer an incoming call or place the current
ANSWER
call on hold and answer another call with one touch.
Used to disconnect the line during or after a
RELEASE
conversation or to complete a Call Transfer.
Used to make or receive a trunk call or can be
Flexible CO (Trunk) reassigned to a different trunk or to another feature
button.

Feature Manual 199

2.21.2 Flexible Buttons

Button Usage
Used to access an extension with one touch. Every
Flexible DSS (Direct Station Selection) button is programmed to correspond to an extension.
DSS buttons can also be reassigned to other features.
Used to access a preprogrammed feature with one
PF (Programmable Feature)
touch. (no default)

Conditions
• Certain buttons are equipped with a light to show line or feature status.

2.21.2 Flexible Buttons

Description
You can customise the flexible buttons and/or programmable feature (PF) buttons on PTs, Add-on Key
Modules, and PSs through either system or personal programming. They can then be used to make or
receive intercom or trunk calls or be used as feature buttons, as follows:

[Button Usage]
Button Usage
Single-CO (S-CO) Used to access a specified trunk for making or receiving calls.
Group-CO (G-CO) Used to access an idle trunk in a specified trunk group for making
calls. Incoming calls from trunks in the assigned trunk group arrive
at this button.
Loop-CO (L-CO) Used to access an idle trunk for making calls. Incoming calls from
any trunk arrive at this button.
Direct Station Selection (DSS) Used to access an extension with one touch.
One-touch Dialling Used to access a preprogrammed party or feature with one touch.
Incoming Call Distribution (ICD) Used to access a specified incoming call distribution group for
Group making or receiving calls.
Message Used to leave a message waiting indication or call back the party
who left the message waiting indication.
FWD/DND (External/Internal/ Used to perform the FWD or DND feature for the extension. The
Both) *1 feature is applied to trunk calls, intercom calls, or both.
Group FWD (External/Internal/ Used to perform the FWD feature for a specified incoming call
Both) distribution group. The feature is applied to trunk calls, intercom
calls, or both.
Account Code Entry (Account) Used to enter an account code.
Conference Used to establish a multiparty conversation.
Terminate Used to disconnect the current call and make another call without
hanging up.
External Feature Access (EFA) Used to send a flash/recall signal to the telephone company or
host PBX to access their features.
Charge Reference Used to check the total call charge for your own extension.
Call Park Used to park or retrieve a call in a preset PBX parking zone.

200 Feature Manual

 2.21.2 Flexible Buttons

Button Usage
Call Park (Automatic Park Zone) Used to park a call in an idle PBX parking zone automatically.
Call Log Used to show the incoming call information.
Log-in/Log-out *1 Used to switch between log-in and log-out mode.
Hurry-up Used to redirect the longest waiting call in the queue of an
incoming call distribution group to the overflow destination.
Wrap-up *1 Used to switch the Wrap-up/Not Ready and Ready modes.
System Alarm Used to confirm a PBX error. Also, pressing the System Alarm
button displays the current Multi-Connection network status.
Time Service *1 Used to switch the assigned time modes: day, lunch, break or
night. Also used to check the current time mode status.
Answer Used to answer an incoming call.
Release Used to disconnect the line during or after a conversation, or to
complete a Call Transfer.
Toll Restriction/Call Barring Used to change the TRS/Barring level of other extension users
temporarily.
ISDN Service Used to access an ISDN service.
Calling Line Identification Used to switch between the CLIP and CLIR service.
Restriction (CLIR)*1
Connected Line Identification Used to switch between the COLP and COLR service.
Restriction (COLR) *1
ISDN Hold Used to transfer a call using the telephone company.
Headset Used to turn on/off the headset mode while idle.
Used to switch between hands-free mode and headset modes
during a conversation.
Time Service Switching Mode Used to switch between the Automatic Switching and Manual
(Automatic/Manual)*1 Switching mode.
Two-way Record Used to record a conversation into your own mailbox.
Two-way Transfer Used to record a conversation into the mailbox of a specified
extension.
One-touch Two-way Transfer Used to record a conversation into the mailbox of a specified
extension with one touch.
Live Call Screening (LCS) Used to monitor your own voice mailbox while an incoming caller
is leaving a message and, if desired, intercept the call.
Voice Mail (VM) Transfer Used to transfer a call to the mailbox of a specified extension. Also
used to access the Unified Messaging system (→ 3.2 System and
Subscriber Features).
Check-in Used to switch the status of extensions from Check-out to Check-
in.
Check-out Used to switch the status of extensions from Check-in to Check-
out.

Feature Manual 201

2.21.2 Flexible Buttons

Button Usage
Cleaned-up Used to switch the room status of extensions between Ready and
Not Ready.
CTI Used to access CTI features.
Primary Directory Number Used to make and receive both outside and intercom calls.
(PDN) (→ 2.9.1 Primary Directory Number (PDN)/Secondary Directory
Number (SDN) Extension)
Secondary Directory Number Used to show the current status of another extension, call the
(SDN) extension, and pick up or transfer calls to it.
(→ 2.9.1 Primary Directory Number (PDN)/Secondary Directory
Number (SDN) Extension)

*1 One-touch Feature Setting Buttons: Pressing these buttons while on-hook changes the feature settings. The new mode will be
displayed for a preprogrammed time period.
Self Labelling (KX-NT366/KX-NT553/KX-NT556/KX-NT560 only)
The KX-NT366 IP-PT and KX-NT553/KX-NT556/KX-NT560 IP-PT have an LCD screen next to their flexible
buttons. The label for each button can be set through personal/system programming to reflect the button’s
function. Additionally, the flexible buttons can be organised into multiple "pages". You can toggle between
pages by pressing the NEXT PAGE key, as follows:

Press NEXT PAGE

Note
The appearance of the NEXT PAGE button differs depending on the telephone model.

Conditions
[Self Labelling]
• Up to 12 characters can be assigned to the LCD of each flexible button through personal/system
programming.
→ 8.3.1 Users—Flexible Button—Wired Extension—Label Name (Max. 12 characters)
• Changes the Self Labelling display of the DSS key in conjunction with a change of extension name
displayed in each DSS key.
→ 10.9 PBX Configuration—[2-9] System—System Options—Option 8—Display extension name on key
(KX-NT)
• When an incoming trunk call is answered or a trunk is seized, the corresponding CO button will turn
Green and the LCD display will switch to the page that the corresponding CO button is registered in.
• It is not recommended to assign the System Alarm button when using this feature, because if an alarm
occurs when the System Alarm button is not on the visible page, the alarm will not be noticed.

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—PT Display—PT Last Display
Duration in Idle Mode

202 Feature Manual

 2.21.3 LED Indication

10.9 PBX Configuration—[2-9] System—System Options—Option 8

8.3.1 Users—Flexible Button—Wired Extension
8.3.1.1 Users—Flexible Button—Wired Extension—Flexible button data copy
8.3.2 Users—Flexible Button—Portable Station

Operating Manual References

1.14.4 Self Labelling (KX-NT366/KX-NT553/KX-NT556/KX-NT560 only)
3.1.3 Customising the Buttons

2.21.3 LED Indication

Description
The LED (Light Emitting Diode) of the Message/Ringer Lamp and following buttons (Line Status Buttons and
Corresponding Extension Status Button) show line conditions with a variety of light patterns.
Line Status Buttons: S-CO, G-CO, L-CO, INTERCOM, ICD Group, PDN
Corresponding Extension Status Button: DSS, SDN
1. Light Pattern of the Message/Ringer Lamp
[IP-PT and DPT]
• Incoming call from a trunk: Red flashing
• Incoming call from another extension: Green flashing
• Message(s) present (no incoming call): Red on
• No message(s) present (no incoming call): Off
[APT]
• Incoming call: Red flashing
• Message(s) present (no incoming call): Red on
• No message(s) present (no incoming call): Off
2. Light Pattern of the Line Status Buttons
Line Status Button
Incoming
Call
Light Intercom
Trunk Status Distribution
Pattern Line Status
Group Line
Status
S-CO G-CO L-CO INTERCOM ICD Group
Off Idle
Green on This extension is using the line.
Slow green This extension is holding the line.
flashing
Moderate This extension is holding the line using Exclusive Hold or using the line for an
green Unattended Conference.
flashing
Rapid green Incoming call/ Incoming call
flashing Privacy
Release

Feature Manual 203

2.21.3 LED Indication

Line Status Button

Incoming
Call
Light Intercom
Trunk Status Distribution
Pattern Line Status
Group Line
Status
S-CO G-CO L-CO INTERCOM ICD Group
Red on Incoming call Other — This
for another extensions extension is
extension/ are using all logged out of
Another trunks in the the incoming
extension is trunk group. call
using the line/ distribution
Another group.
extension has
the line on
Exclusive
Hold.
Slow red Another —
flashing extension is
holding the
line.
Rapid red Incoming call to the incoming call distribution —
flashing group in Ring distribution method

For information on the light patterns of PDN and SDN buttons, refer to 2.9.1 Primary Directory Number
(PDN)/Secondary Directory Number (SDN) Extension.
3. Light Pattern of the Corresponding Extension Status Button
Light Pattern Corresponding Extension Status Button (DSS)
Off Idle
Red on *1
Busy/Incoming call /DND for trunk calls
Rapid red flashing Incoming call
*2

*1 Only when Call Pickup by a DSS button is disabled.

*2 Only when Call Pickup by a DSS button is enabled.

4. Flashing Light Patterns

1s

Slow Flashing

Moderate Flashing

Rapid Flashing

204 Feature Manual

 2.21.4 Display Information

Conditions
• The incoming call shows on the buttons in the following priority:
ICD Group → S-CO → G-CO → L-CO → PDN → INTERCOM
• The light pattern of a DSS button for incoming call can be set to "Off" through system programming. In
this case, the DSS button light will not indicate the status of the corresponding extension.

PC Programming Manual References

10.9 PBX Configuration—[2-9] System—System Options
→ Option 1—PT Fwd / DND—Fwd LED
→ Option 1—PT Fwd / DND—DND LED
→ Option 4—DSS Key—DSS key mode for Incoming Call

Feature Manual References

2.2.2.2 Group Call Distribution
2.2.2.7 Log-in/Log-out
2.3 Call Forwarding (FWD)/Do Not Disturb (DND) Features
2.13.1 Call Hold
2.14.2 Conference
2.14.3 Privacy Release

2.21.4 Display Information

Description
A display PT shows the user the following information while making or receiving calls if they are available:
Display Item Display Example Condition
The extension number and name of the 123: Tom Smith –
calling or called extension, or incoming call
distribution group
Status of the called extension 123: Busy –
The number and name of the optional device D02: 1st Door –
The dialled telephone number 1234567890 –
The received call information The first line message
ABC Company can be either (a), (c), or
a. Caller’s name 12345678 (e) at each extension
b. Caller’s number Line 001: Sales through system
→ 102:Mike programming.
c. Trunk number/name Panasonic These can be displayed
d. Original Destination, if the call is in turn by pressing the
forwarded TRANSFER button or
DISP Soft button during a
e. DDI/DID/MSN name call.
Call charge fee during a trunk call. 12.35€ The currency, position of
the currency symbol, and
the decimal point are
programmable.
Call duration during a trunk call. Line 001 11:02'28 –

Feature Manual 205

2.21.4 Display Information

Conditions
• Multilingual Display
Each extension can select its display language through personal programming (Display Language
Selection).
• Display Contrast
It is possible to adjust the display contrast through personal programming (Display Contrast Selection).
This is available only for DPTs and IP-PTs.
• Display Backlight
Some extensions can select whether to turn the display backlight on or off through personal programming
(Display Backlight Selection). For details, refer to the manual for your telephone.
• Characters (name) or digits (number) exceeding the display’s size limitation are not displayed. In this
case, information which have been programmed is hidden, but not altered.

PC Programming Manual References

10.9 PBX Configuration—[2-9] System—System Options—Option 5—PT Feature Access—No. 1–8
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings
→ Option 5—Display Language
→ Option 5—Incoming Call Display
→ Option 5—Automatic LCD Switch when Start Talking
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Charge
→ Charge Options—Digits After Decimal Point
→ Charge Options—Currency (3 characters)
→ Charge Options—Currency Display Position
18.1 PBX Configuration—[10-1] CO & Incoming Call—CO Line Settings—CO Name

Operating Manual References

3.1.2 Settings on the Programming Mode

206 Feature Manual

 2.22 Administrative Information Features

2.22 Administrative Information Features

2.22.1 Record Log Features
2.22.1.1 Station Message Detail Recording (SMDR)
Description
Automatically records detailed information for each extension.
1. SMDR Output Port
The following output methods can be selected through system programming:
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR—SMDR Format—Port
Output Method Description
Telnet compatible terminal emulator SMDR information is sent to a Telnet compatible terminal
emulator via LAN.

2. SMDR Output Data

The following data will be recorded and sent to the SMDR output port:
a. Trunk call information (incoming/outgoing)
b. Intercom call information (outgoing)
c. Log-in/Log-out information
d. PBX error log (→ 5.6.3 Local Alarm Information)
e. Hospitality feature information (→ 2.23.1 Hospitality Features—SUMMARY)
f. Printing Message information (→ 2.22.2 Printing Message)
Memory for SMDR: A specified number of call records can be stored in the PBX. If more calls are
originated or received, the oldest record is overwritten by the newest one.
3. SMDR Format Type and Contents
The following three types of output format can be selected through system programming:

Feature Manual 207

2.22.1 Record Log Features

Pattern A: 80 digits without call charge information

Date Time Ext CO Dial Number Ring Duration ACC Code CD

(8 digits) (7) (5) (2) (25) (4) (8) (10) (3)

01/02/02 10:03AM 1200 01 <I>12345678901234567890 5'15 00:00'00 NA

01/02/02 10:07AM 1200 01 <I> 0'05 00:01'05 9876543210
01/02/02 10:15AM 1200 01 1234567890123456 00:01'05 9876543210 TR
01/02/02 10:30AM *123 01 1234567890123456 00:01'05 9876543210

01/02/02 01:07PM 1234 01 <I>ABC COMPANY12345678 0'05 00:01'05 9876543210

01/02/02 01:07PM 1234 01 <D>CDE9876<I>Q COMPANY 0'05 00:01'05 9876543210
01/02/02 01:07PM 1234 01 ABC COMPANY12345678 00:01'05
01/02/02 01:07PM 1234 01 123.............. 00:01'05
01/02/02 01:07PM 1234 01 123456XX 00:12'05 98765

01/02/02 08:33AM 1234 In the office

01/02/02 01:07PM 1234 LOG IN
01/02/02 03:35PM 1234 LOG OUT
01/02/02 03:45PM 1234 EXT1235
01/02/02 03:50PM 1234 Check in
01/02/02 03:55PM 1234 Check out
01/02/02 04:00PM 1234 Timed Reminder/Start
01/02/02 04:01PM 1234 Timed Reminder/No Answer
01/02/02 04:01PM 1234 Timed Reminder/Answer
01/02/02 04:05PM 1234 <I>S003 RC

(1) (2) (3) (4) (5) (6) (7) (8) (9)

Pattern B: 80 digits with call charge information

Date Time Ext CO Dial Number Duration Cost ACC Code CD

(8 digits) (7) (5) (2) (20) (8) (8+2) (10) (3)

01/02/02 10:03AM 1210 01 <I> NA

01/02/02 10:07AM 2005 01 1234567890123456789 00:00'05 00560.00EU 9876543210

(1) (2) (3) (4) (5) (7) (10) (8) (9)

Pattern C: 120 digits

Date Time Ext CO Dial Number Ring Duration Cost ACC Code CD
(8 digits) (7) (5) (4) (50) (4) (8) (8+3) (10) (3)

01/02/02 10:03AM 1230 0001 123456789012345678901234567890 00:00'05 00560.00EUR 9876543210 TR

01/02/02 10:07AM 1230 0001 <I>ABC COMPANY123456789012345 0'05 00:00'05 9876543210 TR

(1) (2) (3) (4) (5) (6) (7) (10) (8) (9)

[Explanation]
The following table explains the SMDR contents which are based on the numbers in the previous
pattern examples. For the programmable items, refer to the following [Programmable Items].
Number in
Item Description
the Pattern
(1) Date Shows the date of the call.

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Number in
Item Description
the Pattern
(2) Time Shows the end time of a call as Hour/Minute/AM or PM.
(3) Ext Shows the extension number of a user container or optional
(Extension) device, floating extension number, etc., which was engaged in
the call.
Also shows the following codes:
Dxxx: Outgoing trunk call from a doorphone (xxx=doorphone
number) (→ 2.18.1 Doorphone Call)
Txxx: Outgoing trunk call by TIE line service (xxx=trunk group
number)
*xxx: Verified call (xxx=verification code) (→ 2.7.6 Verification
Code Entry)
(4) CO (Trunk) Shows the trunk number used for the call.
For patterns A and B, "00" will be shown for trunk numbers over
hundred.

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Number in
Item Description
the Pattern
(5) Dial Number [Trunk Call]
Outgoing Trunk Call
Shows the dialled telephone number.
Valid digits are as follows:
0 through 9, , #
P: Pause
F: EFA signal
=: A Host PBX Access code (→ 2.5.4.8 Host PBX Access Code
(Access Code to the Telephone Company from a Host PBX))
. (dot): Secret dialling
X: Privacy dial
–: Transferred call
If the transfer destination extension enters some digits, the
entered digits will be added after "–".

Incoming Trunk Call

Shows <I> + the caller’s identification name/number.
It is also possible to show the DDI/DID/MSN call information. In
this case, <D> + DDI/DID/MSN name/number is added before
<I>.
[Outgoing Intercom Call]
Shows the dialled extension number followed by "EXT".
[Log-in/Log-out]
Shows the log-in or log-out status.
[Check-in/Check-out]
Shows the check-in or check-out status. (→ 2.23.2 Room
Status Control)
[Timed Reminder]
Shows the status of a timed reminder, either "Start", "No
Answer", or "Answer". (→ 2.24.3 Timed Reminder)
[Printing Message]
Shows the selected message. (→ 2.22.2 Printing Message)
[Sensor Call]
Shows calls from an external sensor as follows:
<I> S + sensor number. (→ 2.18.3 External Sensor)
(6) Ring Shows the ring duration before answering a call in Minutes/
Seconds.
(7) Duration Shows the duration of the trunk call in Hours/Minutes/Seconds.
(8) Acc Code Shows the account code appended to the call.
(Account (→ 2.5.4.3 Account Code Entry)
Code) Shows the same information as the Sub Code from the Error
Log, since an extension might not have an extension number
while in Service-out mode. (→ 7.1.4 Troubleshooting by Error
Log in the Installation Manual)
Shows "PS" + 4 digits of the PS number if a PS dialled an
emergency number.

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Number in
Item Description
the Pattern
(9) CD (Condition Shows other call information with the following codes:
Code) CL: Collect call
TR: Transfer
FW: FWD to trunk
D0: Call using DISA or TIE line service
NA: Not answered call
RC: Received call
AN: Answered call
VR: Received call with Call Waiting Caller ID (Visual Caller ID)
VA: Answered call with Call Waiting Caller ID (Visual Caller ID)
(10) Cost Shows the call charge.

[Programmable Items]
Item Description
Outgoing trunk call Controls whether the outgoing trunk calls are shown. This
setting is common throughout the PBX. COS programming is
also required.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR—Print Information—Outgoing Call
Incoming trunk call Controls whether the incoming trunk calls are shown.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR—Print Information—Incoming Call
Outgoing intercom call Controls whether the outgoing intercom calls are recorded.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR—Print Information—Intercom Call
Log-in/Log-out status Controls whether the log-in/log-out status is recorded.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR—Print Information—Log-in / Log-out
ARS dial Controls whether the user-dialled number or the modified
number is shown.
The Host PBX Access code ("=" followed by the access code)
can be shown (as supplementary information) only when the
modified number is selected in this setting. (→ 2.8.1 Automatic
Route Selection (ARS))
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR Options—Option—ARS Dial
Caller’s identification Controls whether the caller’s identification number, name,
number and name, or nothing is shown. If "none" is selected,
<I> will not be shown.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR Options—Option—Caller ID Number & Name
DID/DDI number Controls whether the DID/DDI number, name, number and
name, or nothing is shown. If "none" is selected, <D> will not
be shown.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR Options—Option—DDI/DID Number & Name

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Item Description
Secret dialling Controls secret dialling. If enabled, the dialled number will be
shown as dots.
This setting is effective only when the modified number is
selected in ARS dial setting above. If the user-dialled number is
selected in ARS dial setting, the dialled number will be shown
as dots regardless of this setting.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR Options—Option—Secret Dial
Privacy dial Enables or disables privacy dial. If enabled, the last four digits
of the dialled telephone number and any additional digits after
connection will be shown as "X". (e.g., 123-456-XXXX)
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR Options—Option—Privacy Mode
Date order The date order is changeable: month/day/year, day/month/year,
year/month/day, year/day/month.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR—SMDR Format—Date Format
Received call Controls whether the time of receiving an incoming trunk call is
shown.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR Options—Option—Condition Code "RC"
Answered call Controls whether the time of answering an incoming trunk call is
shown.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR Options—Option—Condition Code "AN"
Room status Controls whether room status changes are shown.
→ 14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—
Main—SMDR for External Hotel Application 1—Room Status
Control
Timed Reminder call Controls whether Timed Reminder calls are shown
(→ 2.24.3 Timed Reminder).
→ 14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—
Main—SMDR for External Hotel Application 1—Timed
Reminder (Wake-up Call)
Printing Message Specifies the messages that can be selected from an extension
(→ 2.22.2 Printing Message).
→ 14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—
Main—SMDR for External Hotel Application 2—Printing
Message 1–8
Time format Controls whether time is displayed in 12-hour or 24-hour format.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—
SMDR—SMDR Format—Time Format (12H / 24H)

Conditions
[General]
• SMDR Format
The following SMDR format can be set through system programming in order to match the paper size
being used in the printer:

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a. Page Length: determines the number of lines per page.

→ 19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR—SMDR Format—Page Length
(Number of Lines)
b. Skip Perforation: determines the number of lines to be skipped at the end of every page.
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR—SMDR Format—Blank Footer
Length (Number of Lines)
The page length should be at least four lines longer than the skip perforation length.
Explanation:

Page
Length
Skip
Perforation

Machine
Perforation

• SMDR data is not deleted even if the PBX is reset.

• SMDR data from a user is not deleted even if the user is deleted.
If the user is deleted while some SMDR data has not been output, the SMDR data may be output as
SMDR data for a new user added later.
• If the PBX is reset during a conversation, the call will not be recorded on SMDR.
• When a call is made from an extension to a number in the Emergency Dial Table (→ 2.5.4.2 Emergency
Call), the PBX can be programmed to record the call information on SMDR both immediately after the
number is dialled and after the call ends. (Normally, the PBX records the call information only after the
call ends.)
→ 19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR Options—Emergency Call Notification
• The following calls are regarded as two separate calls for SMDR:
– Calls before and after the flash/recall/EFA signal is manually sent during a conversation
– Trunk-to-trunk calls by Call Transfer, FWD or DISA (recording each as "incoming call" and "outgoing
call")
– Incoming calls to a PDN or SDN extension.
• The PBX waits for a preprogrammed time period between the end of dialling and start of the SMDR timer
for outgoing trunk calls. When the PBX has sent out all dialled digits to the telephone company and this
timer expires, the PBX starts counting the call. A display PT shows the elapsed time of the call. The
starting time and the total duration of the call are recorded on SMDR.
→ 10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Dial—
Analogue CO Call Duration Start
If the reverse signal detection has been set (→ 2.5.4.5 Reverse Circuit), the PBX will start counting the
call after detecting the reverse signal from the telephone company regardless of the above timer.
[Host PBX Access Code]
• The dialled number including the Host PBX Access code will be recorded on SMDR only if the modified
number setting is selected in the ARS setting for SMDR.

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• When a Host PBX Access code is assigned to a trunk group, calls to extensions of the host PBX are not
recorded on SMDR.
• A Host PBX Access Code can be used to record only long distance calls on SMDR when a trunk port is
connected directly to the telephone company (not a host PBX). This is allowed when the long distance
code (e.g., "0") is assigned as the Host PBX Access code. All local calls (e.g., calls that do not require a
"0" to be dialled first) are treated as extensions of the telephone company and do not get recorded on
SMDR, because in this case this PBX recognises the telephone company as the host PBX. Therefore,
only long distance calls are recorded on SMDR.
[Output to a Telnet compatible Terminal Emulator]
• In order to activate a connection to a terminal emulator, the IP address of the mother board, port number,
user ID ("SMDR"), and password must be entered.
• If a terminal emulator user incorrectly enters the user ID or password 3 times consecutively, an alarm will
be sent and connection will not be possible for 10 minutes.
• Through system programming, it is possible to assign the PBX port number and password.
• The terminal emulator application must be running constantly. If the application is terminated, call records
that occur after the termination will be recorded in the PBX’s memory. However, if the number of call
records exceeds the PBX’s capacity, older records will be deleted. Also, when the application restarts or
is reconnected, duplicated call records may be output.
[Using SMDR with applications]
SMDR data can also be monitored by applications. For more information, see your application’s
documentation.

Installation Manual References

4.5 Connection of Peripherals

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Dial—Analogue
CO Call Duration Start
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—CO & SMDR—Outgoing CO
Call Printout (SMDR)
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge
→ Main—SMDR for External Hotel Application 2—Printing Message 1–8
→ Charge—Charge Options—Currency (3 characters)
19.1 PBX Configuration—[11-1] Maintenance—Main
→ SMDR
→ SMDR Options

Feature Manual References

2.1.1.3 Direct Inward Dialling (DID)/Direct Dialling In (DDI)
2.1.1.4 Multiple Subscriber Number (MSN) Ringing Service
2.2.2.7 Log-in/Log-out
2.5.4.8 Host PBX Access Code (Access Code to the Telephone Company from a Host PBX)
2.12.1 Call Transfer
2.16.1 Direct Inward System Access (DISA)
4.3.1 TIE Line Service
5.1.1 Class of Service (COS)
6.1 Capacity of System Resources

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2.22.1.2 Syslog Record Management

Description
By connecting this PBX to a Syslog server over a LAN, it is possible to output local alarm information (major
alarms/minor alarms) to an external PC.

Conditions
• To be able to use this feature, through system programming, it is required to enable this feature and
register the IP address of the Syslog server.

PC Programming Manual References

7.3.2 Utility—Log—Syslog
28.3.2 Network Service—[3-2] Client Feature—SYSLOG

Feature Manual References

5.6.3 Local Alarm Information

2.22.2 Printing Message

Description
An extension user can select a message to be output on SMDR. Up to eight messages can be
preprogrammed in the Printing Message table, and are available to all extensions connected to the PBX. A
message can contain the "%" symbol, which requires a number to be entered in its place when the message
is selected at an extension.
Depending on the content of the preprogrammed messages, this feature can be used to record a variety of
information, which can be output on SMDR to, for example, a connected PC.
[Example]
If message 1 is preprogrammed as "Started work", and message 2 as "Finished work", employees can sign
in by selecting message 1 when starting work, and sign out by selecting message 2 when finishing. A
connected PC can then be used to generate employee work records.

Conditions
• Up to seven "%"s can be stored for each message.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Printing Message
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Main—SMDR for External Hotel Application 2—
Printing Message 1–8

Feature Manual References

2.22.1.1 Station Message Detail Recording (SMDR)
6.1 Capacity of System Resources

Operating Manual References

1.12.1 Using the Telephones in a Hotel-type Environment (Hospitality Features)

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2.22.3 Call Charge Services

2.22.3 Call Charge Services

Description
The PBX receives a call charge signal during or after a conversation with an outside party. The call charge
information is shown on the telephone display and SMDR.

1. Call Charge Signal Services

The type of call charge service that is used by the PBX is decided by the type of signal received from the
telephone company. The type of call charge signal received from the telephone company depends on
the trunk of the outgoing call. The services for each available trunk type are as follows:
Trunk Service
Analogue trunk Pay Tone
*1

ISDN line Advice of Charge (AOC) (→ 4.1.2.3 Advice of Charge (AOC))

E1 line Meter Pulse

*1 The pay tone call service through an analogue trunk is available only on a trunk connected to a legacy gateway.
(→ 5.3.1 Stacking Connection)

2. Call Charge Display

– Up to eight digits including a decimal (e.g., 12345.78)
– The decimal point position (the number of significant decimal digits) for currency is programmable.
– Up to three currency characters are programmable. (e.g., EUR or € for Euro).
– Through PC programming, you can select whether the currency characters or symbol are placed in
front of or behind the call charge. (e.g., € 45.12 or 45.12 €)
3. Margin/Tax Rate Assignment
It is possible to add a margin and a tax to the call charges. The call charge rate per meter indication is
programmable on a trunk group basis.
[Calculation Method]
The margin or tax rate must consist of four digits, two digits before and after the decimal (xx.xx%). The
calculation method used by the PBX varies, depending on whether the telephone company sends the
meter indication or the actual call charge.
a. Call charge with tax and margin in meter indication:
[Meter indication received from the telephone company] [Call Charge Rate] [1 Tax Rate]
[1 – Margin Rate]

b. Call charge with tax and margin in charge:

[Charge received from the telephone company] [1 Tax Rate]
[1 – Margin Rate]

The calculation result is rounded up to the least significant decimal digit.

4. Total Call Charge
– A PT user can show the total call charges on the display.
– The call charge is totalled on an extension, trunk, or verification code basis.
– When a verification code is used, the call is charged on the verification code and not the extension
that the call was made on.

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5. Budget Management
It is possible to limit telephone usage to a preprogrammed budget on each extension or verification
code. For example, an extension in a rented office has a prepaid limit for telephone usage. If the amount
of the call charge reaches the limit, the extension user cannot make further trunk calls. An extension
assigned as the manager may increase the limit or clear the previous call charge (→ 2.7.2 Budget
Management).
6. Call Charge Management
An extension assigned as a manager can perform the following:
a. Clear the call charges for each extension and verification code.
b. Clear the call charges of all extensions and verification codes.
c. View the call charges (Call Charge Reference) for each trunk, extension, or verification code.
d. Set the call charge rate for each trunk group.
e. Print out the total call charges for all extensions and verification codes.
f. Set a budget for each extension and verification code.
[Examples of Call Charge Reference]

******************************************************
* Charge Meter Print Out - Total & All CO *
******************************************************
Total Charge: 00175.95

CO Line
001: 00194.00 002: 00073.00 003: 00161.00 004: 00033.00

*******************************************************
* Charge Meter Print Out - All Extensions *
*******************************************************

*775: 00194.00 *102: 00073.00 *776: 00161.00

104: 00194.00 105: 00073.00 106: 00161.00 107: 00033.00

Note
*: extension or verification code number

Conditions
[General]
• Call Charge Reference by Call Charge Reference Button
A display telephone user can check the total call charge for his own extension using the Call Charge
Reference button. A flexible button can be customised as the Call Charge Reference button.
[Pay Tone Service]
• It is possible to select whether the PBX starts counting the call charge from when the PBX detects the
answer signal from the telephone company.

PC Programming Manual References

11.1.5 PBX Configuration—[3-1-5] Group—Trunk Group—Charge Rate

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2.22.3 Call Charge Services

8.3.1 Users—Flexible Button—Wired Extension—Type

8.3.2 Users—Flexible Button—Portable Station—Type
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Charge
→ Margin & Tax—Margin Rate for "Telephone" (%)
→ Margin & Tax—Tax Rate for "Telephone" (%)
→ Charge Options—Digits After Decimal Point
→ Charge Options—Currency
→ Charge Options—Currency Display Position
→ Charge Options—Action at Charge Limit
→ Charge Options—Meter Start on Answer Detection

Feature Manual References

2.7.6 Verification Code Entry
2.21.2 Flexible Buttons
6.1 Capacity of System Resources

Operating Manual References

1.12.1 Using the Telephones in a Hotel-type Environment (Hospitality Features)
4.1.2 Manager Programming

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2.23 Hospitality Features

2.23.1 Hospitality Features—SUMMARY
Description
This PBX has several features that support its use in a hotel-type environment, where extensions
correspond to guest rooms.
Feature Description & Reference
Room Status Control An extension designated as the hotel operator can set the check-in
status of rooms remotely.

→ 2.23.2 Room Status Control

Call Billing for Guest Room Charges for calls from guest rooms can be logged and output as a
guest bill.

→ 2.23.3 Call Billing for Guest Room

Remote Wake-up Call An extension designated as the hotel operator can set a timed
reminder for a room remotely.

→ 2.24.3 Timed Reminder

SMDR for External Hotel Hospitality feature data, including check-in*1, check-out*1, and timed
Application reminder times, can be output to SMDR for use in a PC-based hotel
application.

→ 2.22.1.1 Station Message Detail Recording (SMDR)

Hospitality Mode for Unified Extensions whose Unified Messaging mailboxes are set to
Messaging hospitality mode can be restricted to using certain features, such as
listening to messages and changing the mailbox owner name.

→ 3.2.1.22 Hospitality Mode

*1 Check-in and check-out information may not be output, depending on system settings. For details, refer to 2.23.2 Room Status
Control.

2.23.2 Room Status Control

Description
A PT with a 6-line display designated as a hotel operator extension can be used to view and set the Check-
in/Check-out/Cleaned-up (Ready or Not Ready) status of guest rooms associated with extensions.
Any wired extension can be used as a room extension without special programming.
Flexible buttons on the hotel operator’s extension can be set as Room Status Control buttons. The 3 types
of Room Status Control buttons are as follows:
• Check-in
Switches the status of selected room extensions from Check-out to Check-in.

Telephone charges are cleared and Remote Extension Dial Lock is turned off, allowing calls to be made
from the extension.

• Check-out

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2.23.2 Room Status Control

Switches the status of selected room extensions from Check-in to Check-out.

Room extension data, such as Timed Reminder or Last Number Redial data, is cleared, and Remote
Extension Dial Lock is turned on, restricting some calls. This can be useful to prevent the room extension
from being used when no guest is checked in.
When checking a room extension out, the operator can enter customer charges such as minibar charges.
A guest bill showing these charges, as well as call charges, can be printed. If necessary, the guest charge
data entered can be edited later, and the bill reprinted.

• Cleaned-up
Switches the status of selected room extensions between Ready and Not Ready.

When a guest checks out of a room, the room status becomes Checked-out and Not Ready. After the
room has been cleaned, the status can be changed to Checked-out and Ready using this button. It is also
possible to change the status back to Checked-out and Not Ready if necessary.

Room Status Control Mode

Pressing a Room Status Control button when the PT is idle allows the hotel operator extension to enter
Room Status Control mode. When in Room Status Control mode, the corresponding Room Status Control
button’s light flashes red. The Room Status Control button that was pressed determines which room status
each room extension can be switched to. For example, if the Check-in button was pressed, the Check-in
button’s light flashes red and the hotel operator can select which room extensions to check-in.
In addition, DSS buttons on the hotel operator’s extension or a paired DSS Console show the room status of
each extension as follows:
Light Pattern Status
Off Checked-out and Ready
Flashing Red Checked-out and Not Ready
Red on Checked-in

[Example Use: Checked-in Mode]

DSS button
Check-in button Room101 Rooms 101 and 103
Check-out button Room102 are currently in
Cleaned-up button Room103 checked-in status.

Room104
Room105

When in Room Status Control mode, the hotel operator’s extension is treated as a busy extension, similar to
when performing PT programming. Callers to that extension will hear a busy tone.
All other operations, including pressing other Room Status Control buttons, will be ignored. In addition, the
lights of fixed and flexible buttons do not show their normal display pattern. In order to perform other
operations, the hotel operator must exit Room Status Control mode.

Conditions
CAUTION
• Messages left on the extension’s Voice Mail (VM) will be cleared at Check-out.
• A maximum of four hotel operators can be assigned.
• Only one of each type of Room Status Control button can be assigned.

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 2.23.3 Call Billing for Guest Room

• Extensions associated with rooms must be one of the following types:

PT, general SIP, SLT, ISDN Extension, T1-OPX
PSs cannot be checked in and out as room extensions.
• It is recommended that the extension number of a room extension is the same or similar to the room
number, for convenience.
• The previous guest’s billing data is only cleared when an extension is set back to Check-in status.
Thus, it is possible to edit guest charge data and reprint the bill at any time until another guest is
checked in to the same room.
• If enabled through system programming, the check-in and check-out information is recorded on
SMDR.

PC Programming Manual References

11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous—Options—
Call Log for Built-in ACD Report
8.3.1 Users—Flexible Button—Wired Extension—Type
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge
→ Main—Hotel Operator—Extension 1–4
→ Bill—Checkout Billing—Billing for Guest

Feature Manual References

2.2.2.9 Supervisory Feature (ACD)
2.6.3 Last Number Redial
2.7.3 Extension Dial Lock
2.22.1.1 Station Message Detail Recording (SMDR)
2.24.3 Timed Reminder

Operating Manual References

1.12.1 Using the Telephones in a Hotel-type Environment (Hospitality Features)
4.1.3 Supervisory Monitor (ACD) Control

2.23.3 Call Billing for Guest Room

Description
Separately from SMDR, it is possible to output a record of calls along with charges (e.g., telephone charges,
minibar, etc.), which can be used in billing a guest.
Charge Items
This feature provides three types of programmable charge items (Charge Item 1, Charge Item 2, and
Charge Item 3) which can be used for billing guests for various services (e.g., telephone charges). Each
charge item can be customised in the following ways:
– A name that appears on the call billing print out.
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Bill
→ Checkout Billing—Bill (SMDR) for "Telephone"
→ Checkout Billing—Bill (SMDR) for "Minibar"
→ Checkout Billing—Bill (SMDR) for "Others"
– A name that appears on the display telephone of the hotel operator.
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Bill
→ Checkout Billing—LCD for "Telephone"
→ Checkout Billing—LCD for "Minibar"
→ Checkout Billing—LCD for "Others"

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2.23.3 Call Billing for Guest Room

– A tax rate.
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Charge
→ Margin & Tax—Tax Rate for "Telephone" (%)
→ Margin & Tax—Tax Rate for "Minibar" (%)
→ Margin & Tax—Tax Rate for "Others" (%)
Charge Item 1 can also be assigned a margin rate, which is useful for charging guests an additional rate for
using the telephone services.
→ 14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Charge—Margin & Tax—Margin Rate for
"Telephone" (%)
It is possible to print out a bill for a guest. The bill will show the following items:
[Example of Call Billing Sheet]

****************************************
(1) * Hotel *
****************************************
(2) Check in : 01.JAN.00 06:31PM
(3) Check out : 03.JAN.00 07:03AM
(4) Room : 202 : Mr. Smith

(5) 01/01/00 06:52PM 202 01 Call amount:0012 01:24'30 FR00084.50 001

02/01/00 06:07PM 202 01 123456789 00:10'12 FR00010.20 1234567890
02/01/00 07:30PM 202 01 012345678901234 00:06'36 FR00006.60 12345
02/01/00 08:45PM 202 01 0011234567890123 00:03'00 FR00003.00 12345

(6) Telephone 104.30 (Tax 10.000% = 9.48)

Minibar 4.00 (Tax 10.000% = 0.36)
Others 0.00 (Tax 15.000% = 0.00)
(7) Total FR 108.30 (Tax Total = 9.84)
(8) Sheet : 002
(9) ======= Hotel PBX =======
Tel: +41 3 12 34 56 78 Fax: +41 3 12 34 56 78
E-Mail: 12345678 hotelpbx.ch

1. A programmable title (e.g., hotel name).

→ 14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Bill—SMDR for External Hotel Application
—Header 1–3
2. The check-in time.
3. The check-out time.
If the guest has already been checked out, the check-out time will be shown. If not, the time that the bill
was printed will be shown.
4. The extension number and name of the user container.
5. A list of all calls made and call charges (using the same format as SMDR output Pattern B [→
2.22.1.1 Station Message Detail Recording (SMDR)]).
6. The total charge for each charge item and tax, including the preprogrammed tax rate.
7. The combined charges of all three charge items, currency of the charge, and tax.

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 2.23.3 Call Billing for Guest Room

8. The sheet number (the number of times that the current guest’s charge data has been printed out and
then cleared).
9. A programmable footer (e.g., the contact information of the hotel).
→ 14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Bill—SMDR for External Hotel Application
—Footer 1–3

It is possible to select the language used on the guest bill.

→ 14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Bill—SMDR for External Hotel Application—
Language for Bill (SMDR)
Walking COS
If guests are given extension personal identification numbers (PINs), it is possible for calls made from other
extensions (e.g., an extension in a hotel restaurant) to be charged to the guest’s room extension by using
the Walking COS feature (→ 2.7.5 Walking COS).
→ 8.1.1 Users—User Container—Add User/Edit User—User Information—User PIN / Mailbox Password
["0-9"]

Conditions
CAUTION
If the Call Log for Built-in ACD Report setting for the Supervisory Monitor (ACD) Control feature is
enabled through system programming, this feature cannot be used. For details, refer to the relevant
chapter in the PC Programming Manual or consult your system administrator.
• If the total number of call records exceeds 90 % of available memory, call records from the extension with
the largest number of records will be automatically printed out, and the records printed out will be
combined in memory into one aggregate record to save space.

PC Programming Manual References

11.5.3 PBX Configuration—[3-5-3] Group—Incoming Call Distribution Group—Miscellaneous—Options—
Call Log for Built-in ACD Report
8.1.1 Users—User Container—Add User/Edit User—User Information—User PIN / Mailbox Password ["0-9"]
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Bill—Checkout Billing—LCD for "Telephone"
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Charge
→ Margin & Tax—Margin Rate for "Telephone" (%)
→ Margin & Tax—Tax Rate for "Telephone" (%)
→ Margin & Tax—Tax Rate for "Minibar" (%)
→ Margin & Tax—Tax Rate for "Others" (%)

Feature Manual References

2.2.2.9 Supervisory Feature (ACD)
2.6.3 Last Number Redial
2.7.3 Extension Dial Lock
2.22.1.1 Station Message Detail Recording (SMDR)
2.24.3 Timed Reminder
6.1 Capacity of System Resources

Operating Manual References

1.12.1 Using the Telephones in a Hotel-type Environment (Hospitality Features)
4.1.3 Supervisory Monitor (ACD) Control

Feature Manual 223

2.23.4 Built-in FOS Interface

2.23.4 Built-in FOS Interface

Description
You can connect a FOS system (Front Office Software) for use in hotels directly to the PBX.

Conditions
• KX-NSXF004 (Activation Key for Multiple CSTA Connection) is required to use this feature.
• The built-in FOS interface supports the following commands:
FOS → PBX
– Link Alive
– Link Start
– Check In/Check Out
– Message Waiting On/Off
– Do Not Disturb Set/Clear
– Wakeup Time Set/Clear
– Change COS
– Room Sharing/Guest Change
– Post Answer
PBX → FOS
– Link Alive
– Link Start
– Link Setup
– Station Message Detail Recording/Call Charge Posting
– Room Status
– Minibar Charge Posting
– Wakeup Confirmation OK/NG
– Message Waiting On/Off
– Do Not Disturb Set/Clear
• When a Change COS command is received from the FOS system, the built-in FOS interface converts the
COS number from the FOS system to an internal COS number as follows:
COS number (FOS) 0 1 2 3
COS number (PBX) 1 2 3 4

• If communication with the FOS system is interrupted, the PBX will record data on SMDR. When
communication is re-established, the PBX will send the recorded data to the FOS system.

PC Programming Manual References

9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Port Number—FOS Interface
Port Number

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 2.24 Extension Controlling Features

2.24 Extension Controlling Features

2.24.1 Extension Personal Identification Number (PIN)
Description
Each extension user can have his own PIN through system programming or personal programming
(Extension PIN [Personal Identification Number]) to set features or access his own telephone remotely.
The following features cannot be used without the PIN:
a. Live Call Screening (LCS)*1 (→ 3.2.2.19 Live Call Screening (LCS))
b. Display Lock (→ 2.6.4 Speed Dialling—Personal/System, 2.19.2 Incoming Call Log)
c. Extension Dial Lock (→ 2.7.3 Extension Dial Lock)
d. Walking COS (→ 2.7.5 Walking COS)
e. Walking COS through DISA (→ 2.16.1 Direct Inward System Access (DISA))
*1 If an extension user has assigned an extension PIN, this feature cannot be used without the PIN.

Conditions
CAUTION
There is a risk that fraudulent telephone calls will be made if a third party discovers a personal
identification number (PIN) (verification code PIN or extension PIN) of the PBX.
The cost of such calls will be billed to the owner/renter of the PBX.
To protect the PBX from this kind of fraudulent use, we strongly recommend:
a. Keeping PINs secret.
b. Selecting complex, random PINs that cannot be easily guessed.
c. Changing PINs regularly.
• Extension PIN Lock
If the wrong PIN is entered three times, the line will be disconnected. If the wrong PIN is entered a
preprogrammed number of times successively, that extension will become locked, and even entering the
correct PIN will not unlock it. Only an extension assigned as the manager can unlock it. In this case, the
PIN will be unlocked and cleared. This feature is also known as Station Password Lock.
• Remote Extension PIN Clear
If an extension user forgets his PIN, a manager can clear the PIN. Then the extension user can assign a
new PIN.
• Extension PIN Display
It is possible to select whether to show the extension PIN on the display through system programming. By
default, it is shown as dots.

PC Programming Manual References

10.12 PBX Configuration—[2-13] System—Security
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Extension PIN Set /
Cancel
10.9 PBX Configuration—[2-9] System—System Options—Option 1—PT LCD—Password / PIN Display
8.1.1 Users—User Container—Add User/Edit User—User Information—User PIN / Mailbox Password ["0-9"]

Operating Manual References

3.1.2 Settings on the Programming Mode

Feature Manual 225

2.24.2 Extension Feature Clear

4.1.2 Manager Programming

2.24.2 Extension Feature Clear

Description
Extension users can clear all the following features set on their own telephone at once. This feature is also
known as Station Programme Clear.
Features After Setting
Absent Message Off
BGM Off
FWD*/DND* Off
Call Pickup Deny Allow
Call Waiting* Disable (In Canada, the default setting is "Enable"
[Call Waiting tone].)
Data Line Security Off
Executive Busy Override Deny Allow
Log-in/Log-out Log-in
Message Waiting All messages left by other extensions will be
cleared.
Paging Deny Allow
Hot Line* Off
Timed Reminder Cleared

Note
The features with "*" can be programmed not to be cancelled by this feature.

Conditions
• Extension Dial Lock (→ 2.7.3 Extension Dial Lock) and the extension personal identification number (PIN)
(→ 2.24.1 Extension Personal Identification Number (PIN)) will not be cleared by this feature.
• For Users in Canada only
If dial tone 2 is heard after Extension Feature Clear:
After performing Extension Feature Clear, Call Waiting will be enabled if "Extension Clear: Call Waiting"
is set to "Clear" through system programming. In this case, dial tone 2 will be heard when going off-hook.
(→ 2.25.1 Dial Tone)

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Extension Feature Clear
10.9 PBX Configuration—[2-9] System—System Options—Option 2
→ Extension Clear—Call Waiting
→ Extension Clear—Fwd/DND
→ Extension Clear—Hot Line (Pick-up Dial)

Operating Manual References

1.9.12 Clearing Features Set at Your Extension (Extension Feature Clear)

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 2.24.3 Timed Reminder

2.24.3 Timed Reminder

Description
User's devices (Main / Sub / Wireless) can be preset to ring at a certain time, to act as a wake-up call or
reminder. This feature can be programmed to activate only once, or daily. If the user answers the alarm call,
a prerecorded voice message will be heard. If a message is not assigned, a special dial tone (dial tone 3)
will be heard.
Timed reminders can be set in one of two ways:
• By the user, from his own device.
• Remotely, by the hotel operator (Remote Wake-up Call)

Conditions
• Be sure that the PBX clock works.
• Only one timed reminder can be set for an extension at a time. Setting a new reminder clears the
previous reminder. If both the extension user and the hotel operator set a timed reminder for the same
extension, the timed reminder that was set most recently is effective.
• When the specified time has come, all the user devices (Main / Sub / Wireless) will ring.
• SMDRs are recorded as many as the number of the devices that rang. At this time, users' extension
numbers are recorded in the respective SMDRs.
• The timed reminder feature will not work while the following features are working on your phone:
– Unattended conference
– Call hold
• When some devices of the same user are installed to two or more PBXs with time differences, each
device will ring at a separate timing according to the local time set on each PBX.

Example：
Ring time
User Device Site Set time Local time Ring time (GMT)
(Local)
User001 Main PBX-A - GMT+0h 9:00 9:00
Sub PBX-B 9:00 GMT+8h 9:00 17:00

• Programmable Time
The Alarm Ringing Duration time, the number of alarm repeat times, and intervals are programmable
through system programming.
• To use the voice message feature:
An extension assigned as the manager can record messages (→ 2.28.2 Outgoing Message (OGM)). A
different message can be assigned for each time mode (day/lunch/break/night) (→ 5.1.4 Time Service).

PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf
→ Timed Reminder—Repeat Counter
→ Timed Reminder—Interval Time
→ Timed Reminder—Alarm Ringing Duration
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Remote Timed Reminder (Remote Wakeup Call)
→ Timed Reminder Set / Cancel

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2.24.4 Smart Desk Features

10.8.3 PBX Configuration—[2-8-3] System—Ring Tone Patterns—Call from Others—Timed Reminder—

Ring Tone Pattern Plan 1–8
10.9 PBX Configuration—[2-9] System—System Options—Option 1—PT LCD—Time Display
13.3.1 PBX Configuration—[5-3-1] Optional Device—Voice Message—DISA System—Option 2—Timed
Reminder Message—Day, Lunch, Break, Night
14.2 PBX Configuration—[6-2] Feature—Hotel & Charge—Main—SMDR for External Hotel Application 1—
Timed Reminder (Wake-up Call)
19.1 PBX Configuration—[11-1] Maintenance—Main—SMDR—Print Information—Timed Reminder (Wake-
up Call)

Feature Manual References

2.23.2 Room Status Control

Operating Manual References

1.9.1 Setting the Alarm (Timed Reminder)
1.12.1 Using the Telephones in a Hotel-type Environment (Hospitality Features)

2.24.4 Smart Desk Features

Description
Allows you to Service-in when you want to use Hot Desking with a telephone that does not have a fixed
extension number.
By performing the Service-in operation, the optional devices within the system can be used as your own
extension telephone terminal. (Optional devices need to be configured in advance.)
[Example]
Usage method:
• Use the same extension number for multiple devices set at different locations. Example) Day time or work
telephone, night time or home telephone
• When the user of the device differs depending on the day or time. Example) Day and night shifts of call
centre operators.
The LCD shows the following when operational and during the Service-out state.
When operational Service-out Operation When serviced out

SEP.29 11:13AM MON Service-out

PROG INFO RING MENU START

Service-in Operation
[Related to the Service-in and Service-out state]
• Indicates whether or not it can be used as a user device.
• The Service-out state is a state in which the devices that the user uses, such as the main devices, sub
devices, and wireless devices, are registered in the User Container.
In addition, the Service-out state refers to a state where the device does not operate as a user device. In
this state, only the Service-in Operation and making emergency calls is allowed.
※ For details on the User Container, see "2.29 User Container".
• When the main device services out, the sub devices and wireless devices configured in the User
Container will also Service-out simultaneously.
The sub devices and wireless devices configured in the User Container will also Service-in when the main
device services in.

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 2.24.4 Smart Desk Features

(Except from when it is registered to another User Container when servicing out).

Available device for User Available device for User

Main Device
Main Device Service-Out Main Device
(by Main Device) means "Service-In"

Sub Device/ Sub Device/ means "Service-Out"

Wireless Device Wireless Device

Available device for User Available device for User

Main Device Main Device Main Device

Service-In

Sub Device/ Sub Device/

Wireless Device Wireless Device

Available device for User Available device for User

Main Device Main Device Main Device

Service-In

Sub Device/ Sub Device/

Wireless Device Wireless Device

Sub/Wireless Device Service-In

Available device for User Available device for User Available device for User

Main Device Main Device

Main Device Service-Out Main Device Service-In Main Device
(by Main Device) (by Main Device)

Sub Device/ Sub Device/ Sub Device/

Wireless Device Wireless Device Wireless Device

Conditions
[Adaptation Extension]

Feature Manual 229

2.24.4 Smart Desk Features

• This feature can be used on the PS/SLT/DPT/IP-PT/IP Softphone and APT extension devices.
• When performing the Service-in Operation on a device in the Service-out state, the device can be used
as a device which uses the extension number of the specified user.
The main devices, sub devices, and wireless devices configured in the User Container can perform the
Service-in Operation.
For details about the User Container settings, see "2.29 User Container".
• Smart Desk can regulate the services of the Service-out Operation through the Class of Service (COS) of
the user.
• The Mailbox set by the user will not be deleted upon servicing in or servicing out.
• When devices in the Service-in state are being used, devices linked to the user cannot be serviced in.
• Devices with a DSS console connected to them cannot use Smart Desk, and although Servicing in is
possible, a device cannot service out once it is linked to the user (i.e., once the device is linked using the
User Container settings).
• SIP devices do not support Smart Desk.
• Depending on the Class of Service (COS) settings, login/logout to an ICD group can be linked to Service-
in/Service-out.
• When a DPT that is connected to a DHLC port has an XDP extension, the XDP port does not move when
Smart Desk is used to move to another port. (This is because the DPT and XDP are independent,
separate extensions.)
• You cannot Service-in or Service-out via DISA/CTI.
• If the Time Service mode is switched or the device is not operated for a given period of time, the user can
be serviced out automatically by the settings for each user. For more information, See 2.24.4.3 Automatic
Service-out Feature.

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Smart Desk

Operating Manual References

1.13.2 Service-in/Service-out (Smart Desk)

2.24.4.1 Service-in Feature

Description
• Shows the Service-in Dial Plan type and summary.
Parameters following the dial
Dial plan Details
plan
Main device Service-in Access Code for Servicing in as a main Extension number + password
device
Sub device Service-in Access Code for Servicing in as a sub Extension number + password
device
PS Service-in Access Code for Servicing in as a Extension number + password
wireless device

• For details about access codes, see 5.5.6 Flexible Numbering/Fixed Numbering.
• Service-in Operation for users that are serviced in on other devices

230 Feature Manual

 2.24.4 Smart Desk Features

If the specified device (main/sub/wireless) of specified user already serviced in when performing the
Service-in Operation, you can newly Service-in the device by performing the Service-in Operation after
servicing out the already serviced in devices.
Any sub device or wireless device serviced in at that time will continue to be serviced in.
Operation for servicing
User Container: A User Container: A
in from Port N to
Main device: 300 User A (main device) Main device: 300

Sub device: 310 Sub device: 310

Wireless device: 320 Wireless device: 320

[Port M] [Port N] [Port M] [Port N]

300 (Service-out) (Service-out) 300

Conditions
• Service-in can only be operated from devices in a Service-out state.
Therefore you need to Service-out the telephone before other users can Service-in on a device that is
already serviced in.
• When disconnecting (breaking the link of the device set in the User Container) the terminal linked to the
user (Service-in state) by servicing it out, the terminal becomes unable to perform any other services than
servicing in and making emergency calls.
• When dialling the Service-in Access Code on a device in the Service-in state, a reorder tone will be
heard.
• Service-in can operate the main devices, sub devices, and wireless devices of the User Container.
• Service-in on a sub device or wireless device is only possible from an operational device which can be
registered as a sub device or wireless device.
• For details about the registrable devices, see "2.29 User Container".
• A Normal User cannot Service-in on a sub device. In addition, when entering the sub device Service-in
Access Code, a reorder tone will be heard.
It is not possible to Service-in on sub devices or wireless devices when the settings for linking the user to
the device have not been configured.
For details about the User Container settings, see "2.29 User Container".

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Smart Desk

Operating Manual References

1.13.2 Service-in/Service-out (Smart Desk)

2.24.4.2 Service-out Feature

Description
• Shows the Service-out Dial Plan type and summary.

Feature Manual 231

2.24.4 Smart Desk Features

Dial plan Details Parameters following the dial plan

Service-out Access Code for servicing out Password

• For details about access codes, see 5.5.7 Flexible Numbering/Fixed Numbering.

Conditions
• In the Service-out state, the language displayed on the terminal will be the same as the language of the
system.
• You can Service-out by entering the Service-out access code or password on a serviced-in device.
• Service-out is not possible when on a call, when on hold, or when receiving a call. In these situations, a
reorder tone will be heard when you try to Service-out.
• When servicing out the main device, the sub devices and wireless devices will automatically be serviced
out as well.
However, unlike with the normal Service-out state, before other user services in, the previous user is
stored before servicing out, and the sub devices and wireless devices will automatically reconnect and
become operational when the main device of that user is serviced in again.
• You can change the setting that determines whether password authentication is required when servicing
out. (Default: password authentication enabled)

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Smart Desk

Operating Manual References

1.13.2 Service-in/Service-out (Smart Desk)

2.24.4.3 Automatic Service-out Feature

Description
If the Time Service mode of the PBX is switched or the device of the user is not operated for a given period
of time, the system automatically services out the user.
For example, this feature services out the user in the following cases:
• The agent in the call centre forgets to perform the service-out operation.
• The time to close the call centre operation comes.

The automatic service-out feature has the following two settings. These settings are independent of each
other:

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 2.24.4 Smart Desk Features

Feature Description
Service-out by the Time When the Time Service mode of the PBX is switched, the user will
Service mode switching automatically be serviced out. For information about the Time Service
mode, refer to 5.1.4 Time Service. The system administrator can set this
setting individually for each user as follows:
• Disable： The automatic service-out feature is disabled (Default)
• Night Mode： Services out the user when the system is switched to
the Night mode
• Lunch Mode: Services out the user when the system is switched to the
Lunch mode
• Break Mode: Services out the user when the system is switched to the
Break mode
Service-out by the no If the device of the user (Main/Sub/Wireless) is operated for a specified
operation timer period of time (Default: 1 hour), the user will automatically be serviced
out. The system administrator can set whether or not to activate this
feature individually for each user.
Once this feature is activated, the system immediately starts counting the
no operation timer.

Conditions
• When the device of the user is in the following state, the user will not be serviced out automatically:
– The device is receiving a call (except for a paging call and a PS ring group call)
– The device is holding a call (including the Consultation Hold) (→ 2.13.1 Call Hold)
– The device is parking a call (→ 2.13.1 Call Hold)
– Automatic Redial (→ 2.6.3 Last Number Redial)
– Unattended Conference (→ 2.14.2 Conference)
• The automatic service-out feature operates forcibly even when password authentication is required for
servicing out.
• The timing when the automatic service-out operates may be delayed depending on the situation of the
system (e.g. when the system is congested).
• When the user is serviced out automatically, this will be recorded in the Call Control Log.

Service-out by the Time Service mode switching

• If the system is switched to the Time Service mode that services out the user automatically while the user
is operating the device, the user will not be serviced out automatically. The user will be serviced out when
the Time Service mode is switched next time (for example, if the Time Service mode that services out the
user automatically is the Break mode, the user will be serviced out when the Time Service mode is
switched to the Break mode again).
• When the system fails to operate the automatic service-out, this will be recorded in the Call Control Log.
• If the user presses the Time Service Button on the device, and the system switches to the Time Service
mode that services out the user automatically, the user will be serviced out immediately.

Service-out by the no operation timer

• If the device of the user is operated as shown below, the no operation timer of the user will be reset:

Feature Manual 233

2.24.4 Smart Desk Features

Device Type Operation

PT • Going off-hook
DECT PS • Making/Answering a call
• PT Programming
SLT • Going off-hook
ISDN terminal • Making/Answering a call
SIP Phone • Making/Answering a call
• Even when the wireless device of the user cannot communicate with the PBX (out of the communication
range, power OFF, etc.), the system continues counting the no operation timer.

PC Programming Manual References

7.3.5 Utility—Log—Call Control Log
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings

Feature Manual References

5.1.4 Time Service

234 Feature Manual

 2.25 Audible Tone Features

2.25 Audible Tone Features

2.25.1 Dial Tone
Description
The following distinctive dial tones inform extensions about features activated on their extensions.
Each dial tone type has two frequencies (e.g., dial tone 1A and dial tone 1B).
Type Description
Tone 1A/1B A normal dial tone is heard when:
1. No features listed for dial tones 2 through 4 has been
set, or
2. ARS is used.
Tone 2A/2B This tone is heard when:
• Any of the features below are set.
– Absent Message
– BGM
– FWD
– Call Pickup Deny
– Call Waiting
– DND
– Extension Dial Lock
– Executive Busy Override Deny
– Hot Line
– Timed Reminder
Tone 3A/3B This tone is heard when:
• A called PS is being searched for.
• Any of the features below are performed.
– Account Code Entry
– Consultation Hold
– Answering a Timed Reminder call with no message
– Answering a sensor call
Tone 4A/4B This tone is heard when new messages have been
recorded for the extension.

Conditions
• Dial Tone Type A/B
It is possible to select dial tone type A or B for dial tones 1 through 4. If "Type A" is selected, all dial tones
1 through 4 will become dial tone type A.
The dial tone type for the ARS feature can be selected separately. If "Type A" is selected for the ARS,
dial tone 1A will be heard. If "Type B" is selected, dial tone 1B will be heard.
• Dial Tone Patterns

Feature Manual 235

2.25.2 Confirmation Tone

All dial tone patterns have a default (→ 6.2.1 Tones/Ring Tones).

PC Programming Manual References

10.9 PBX Configuration—[2-9] System—System Options—Option 3
→ Dial Tone—Distinctive Dial Tone
→ Dial Tone—Dial Tone for Extension
→ Dial Tone—Dial Tone for ARS

2.25.2 Confirmation Tone

Description
At the end of feature operations, the PBX confirms the success of the operation by sending a confirmation
tone to extension users.
Type Description
Tone 1 1. Sent when the setting is accepted.
2. Sent when a call is received in voice-calling mode
(Alternate Receiving—Ring/Voice). The caller’s
voice will be heard after the tone.
Tone 2 1. Sent from an external paging device or an
extension before being paged.
2. Sent when a call is received in Hands-free
Answerback mode.
Tone 3-1 1. Sent before a conversation is established when
using the Paging feature.
2. Sent when a conversation is established with the
extension in the following modes after the call
making operation:
• Hands-free Answerback mode
• Voice-calling mode (Alternate Receiving—Ring/
Voice)
3. Sent when making a call to or from a doorphone.
Tone 3-2 Sent just before a conversation is established when
accessing the following features by the feature
numbers:
• Call Park Retrieve
• Call Pickup
• Hold Retrieve
• Paging Answer
• TAFAS
Tone 4-1 Sent when moving from a two-party call to a three-
party call. (e.g., Executive Busy Override, Conference,
Privacy Release, Two-way Record.)

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 2.25.2 Confirmation Tone

Type Description
Tone 4-2 Sent when moving from a three-party call to a two-
party call. (e.g., Executive Busy Override, Conference,
Privacy Release, Two-way Record.)
Tone 5 Sent when a call is placed on hold (including
Consultation Hold).

Conditions
• Confirmation Tone Patterns
All confirmation tone patterns have a default (→ 6.2.1 Tones/Ring Tones).
• It is possible to eliminate each tone.

PC Programming Manual References

10.9 PBX Configuration—[2-9] System—System Options—Option 3
→ Confirmation Tone—Tone 1 : Called by Voice
→ Confirmation Tone—Tone 2 : Paged / Automatic Answer
→ Confirmation Tone—Tone 3-1 : Start Talking after Making Call / Call from DOORPHONE
→ Confirmation Tone—Tone 3-2 : Start Talking after Answering Call
→ Confirmation Tone—Tone 4-1 : Start Conference
→ Confirmation Tone—Tone 4-2 : Finish Conference
→ Confirmation Tone—Tone 5 : Hold

Feature Manual 237

2.26 Computer Telephony Integration (CTI) Features

2.26 Computer Telephony Integration (CTI)

Features
2.26.1 Computer Telephony Integration (CTI)
Description
The PBX supports a CTI interface using the LAN port of the mother board. The CTI interface allows
extension users to make or receive calls with advanced features:
– Extension users can make calls easily from a phone book in their PC.
– When an extension user receives an incoming call, detailed caller information can be displayed on the PC
automatically.
A PC and CTI server application software are required to use CTI features. The PC running the application
monitors the status of the PBX and controls the PBX via the CTI Server.

PBX
CTI Server

Mother LAN
Board

PC PC

LAN

Conditions
• CTI application software must be installed on the connected PC. In addition, KX-NSXF004 (Activation
Key for CTI interface) is required to use CTI applications. In a Multi-Connection network, the activation
key is required only for the Master unit.
• To use UC Pro with API/Protocol "ECMA CSTA", KX-NSXF004 is not necessary.
• The protocols supported by the KX-NS Series are shown below.
Type API/Protocol
Third Party Call Control • ECMA CSTA Phase 3
• TAPI 2.1

• Only one CTI server can connect to the PBX at a time.

• For details about specific CTI features, refer to the manual for your CTI application software.
• When using a Panasonic TSP, refer to "Before Installing" in the KX-NSX Series TSP Installation Manual.

Installation Manual References

4.5 Connection of Peripherals

238 Feature Manual

 2.26.2 CA (Communication Assistant) Operator Console

PC Programming Manual References

9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Port Number
→CTI Port Number
→Built-in Communication Assistant Server
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Dial Information (CTI)
10.9 PBX Configuration—[2-9] System—System Options—Option 6 (CTI)

2.26.2 CA (Communication Assistant) Operator Console

Description
Panasonic Communication Assistant (CA) is a CTI application that can be used with PT, DECT PS or SLT.
KX-NSX supports the CA Operator Console only.

Conditions
• A maximum of 8 extension users can use the CA Operator Console.
• To use the CA Operator Console, KX-NSA401 (CA Operator Console (user)) is required. One
KX-NSA401 supports one CA Operator Console. If KX-NSXP101 (Ultimate Activation Key for Panasonic
Terminal Users) has already been installed, KX-NSA401 will not be required.
• The CA server is not required for use of the CA Operator Console. (Direct log-in to the NSX server only)
• The following CA features are not supported:
– Other CA modes (Basic-Express / Pro / Supervisor)
– CA Network Feature
– CA Thin Client Server
• The CA Operator Console users can use the first 1000 System Speed Dial numbers.
• The CA Operator Console users can start a conference call with group no.1–8.
• Other conditions are basically the same as the CA Operator Console for the KX-NS series. For more
information, refer to the manual for CA.

PC Programming Manual References

8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 9—Built-in
Communication Assistant
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Port Number—Built-in
Communication Assistant Server
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings

Feature Manual References

2.6.4 Speed Dialling—Personal/System
2.14 Conference Features

Feature Manual 239

2.27 Cellular Phone Features

2.27 Cellular Phone Features

2.27.1 Cellular Phone Features—SUMMARY
Description
This PBX provides features to support the use of cellular phones and other outside destinations with the
PBX. Calls can be forwarded from public device names to outside destinations such as cellular phones, and
then answered as if the user was at an extension within the PBX. Also, when the receiving extension directly
forwards the call to a cellular phone, the cellular phone can use PBX extension features in the same way as
if it called the PBX directly.
The following features can be used with cellular phones and other outside extensions:
Feature Description & Reference
Parallel Ringing When When an unanswered call is forwarded to an outside line, such
Forwarding to Trunk as a cellular phone, the forwarding extension’s phone will
continue ringing until the forwarded call is answered at either
phone.

→ 2.3.2 Call Forwarding (FWD)

DISA Automatic Walking COS Registered cellular phones are automatically recognised as PBX
extensions when calling through DISA.

→ 2.16.1 Direct Inward System Access (DISA)

DISA Call Transfer From A cellular phone user who answers a trunk call forwarded from
Outside Destination the PBX using DISA can transfer that call to an extension
(including over a TIE connection) or to an outside party. It is also
possible to establish a Conference call, perform Call Splitting,
and page with a call on hold to transfer the call.

→ 2.16.1 Direct Inward System Access (DISA)

DISA Call Transfer to outside From the transfer destination (including the outside party), it is
user possible to establish a Conference call, perform Call Splitting,
and page with a call on hold to transfer the call.

→ 2.16.1 Direct Inward System Access (DISA)

Conditions
• KX-NSUM001, KX-NSUM010, KX-NSUM050, KX-NSUM100, KX-NSUM500 (Activation Key for Mobile) or
KX-NSUA001, KX-NSUA010, KX-NSUA050, KX-NSUA100, KX-NSUA500 (Advanced User AK) is
required to use these features. One activation key is required for each extension that will use these
features.
• Call disconnection detection
When LCOT is used for the trunk, the system cannot detect call disconnection. Therefore, the system
disconnects the trunk side after transferring the call (unscreened transfer).

PC Programming Manual References

8.1.1 Users—User Container—Add User/Edit User—Device Settings—Mobile Phone—Quantity of Mobile
devices

240 Feature Manual

 2.28 Miscellaneous Features

2.28 Miscellaneous Features

2.28.1 Background Music (BGM)
Description
A PT user can listen to BGM through the built-in speaker while on-hook and idle. The following audio
sources are available for BGM:
• External Music Source
• Internal Music Source

BGM—External:
BGM can also be broadcast in the office through the external pagers, this can be turned on and off by an
extension assigned as the manager. An Expansion Gateway is required to use this feature.

Conditions
[BGM]
• Hardware requirement: User-supplied Expansion Gateway and music source (when an external music
source is assigned)
• The music through the PT is interrupted when going off-hook.
• Each user can set/cancel BGM, and also select the music source.
• Through system programming, it is possible to specify the maximum number of IP-PTs that can
simultaneously perform the BGM feature. Changing this setting may affect the number of simultaneous IP
extension and IP trunk calls available on the mother board.
For systems connected to an Expansion Gateway, a BGM can be selected from up to 8 sound resources
for each Gateway, and the BGM can be played to the terminals integrated by these terminals.
[BGM—External]
• Hardware requirement: A user-supplied external pager
• External pagers can be used with the following priorities:
TAFAS → Paging → BGM
(→ 2.17.1 Paging, 2.17.2 Trunk Answer From Any Station (TAFAS))

Installation Manual References

4.5 Connection of Peripherals

PC Programming Manual References

9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—VoIP-DSP Options—IP
Extension Count of BGM
10.2 PBX Configuration—[2-2] System—Operator & BGM
→ BGM and Music on Hold—Music Source of BGM/MOH 1
→ BGM and Music on Hold—Music Source of BGM/MOH 2
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ External BGM On / Off
→ BGM Set / Cancel
10.11.1 PBX Configuration—[2-11-1] System—Audio Gain—Paging/MOH
→ External MOH—MOH 1–8 (Music On Hold 1–8)
→ Internal MOH—MOH 1–8 (Music on Hold 1–8)
13.2 PBX Configuration—[5-2] Optional Device—External Pager

Feature Manual 241

2.28.2 Outgoing Message (OGM)

Operating Manual References

1.9.8 Turning on the Background Music (BGM)
2.1.4 Turning on the External Background Music (BGM)

2.28.2 Outgoing Message (OGM)

Description
An extension assigned as the manager (manager extension) can record outgoing messages (OGMs) for the
following features:
Feature Usage & Reference
Direct Inward System Access When a call arrives on a DISA line, the caller will hear a message.
(DISA)
→ 2.16.1 Direct Inward System Access (DISA)
Queuing Feature If assigned in the Queuing Time Table of the incoming call
distribution group, any caller who is waiting in a queue will hear a
message.

→ 2.2.2.3 Queuing Feature

Timed Reminder When answering the Timed Reminder Alarm, the user will hear a
message.

→ 2.24.3 Timed Reminder

Conditions
• Number of Messages
A maximum of 128 messages can be recorded on a PBX. In a Multi-Connection network, 128 messages
can be recorded at each site.
• Multi-Connection Networking (→ 4.2 Multi-connection Networking)
– A floating extension number is assigned to each outgoing message (OGM) (default: 58 + two-digit
*1
OGM number ). Even though messages are stored separately for each site, the message numbers
(OGM 1 to 128) and their corresponding floating extension numbers are shared among all sites.
Therefore, the contents of message 01 (floating extension number 5801) at site A may be different
from the contents of message 01 (floating extension number 5801) at site B.
– Messages at all sites can be recorded, listened to, and deleted using the Master unit. However, in
feature operation a PBX can use only the messages stored locally. When an OGM is sent to an outside
caller, the OGM is provided by the unit to which the caller is connected. Therefore, the caller will hear a
different OGM depending on the unit that provides the message.
*1 The number of digits for Floating Extn. No depends on the value specified for Numbering Plan in Easy Setup.
→ 2.1.4 Easy Setup Wizard

• DSP Resource Usage

Playing back an OGM requires a certain number of DSP resources. If all DSP resources are in use, this
operation cannot be performed. To ensure a minimum level of performance, DSP resources can be
reserved for OGM playback. (→ 5.5.3 DSP Resource Usage)
• There is no limit to the length of an individual message, but the maximum recording time (all messages
combined) is approximately 64 minutes.
• The same message can also be played simultaneously to multiple callers.
• Recording Methods

242 Feature Manual

 2.28.2 Outgoing Message (OGM)

Record voice messages through the extension telephone.

• After recording messages, a manager extension can also play them back for confirmation.
• Progress tone is sent to a manager extension before recording messages during a preprogrammed time
period, or during clearing the prerecorded message stored at the floating extension number of desired
message. The longer one is applied.
• When the manager tries to record a message, he will hear ringback tone if a message channel is in use.
When all message ports become idle, he will hear the progress tone for a preprogrammed time period.
After that, the PBX will automatically proceed into the recording mode.
• Copying messages to and from a PC
Via Web Maintenance Console, messages can be copied to and from a PC. This feature is useful for
copying the same message to multiple sites.
– Messages are downloaded to a PC in WAV (G.711a/µ) format.
– Messages uploaded from a PC must be in one of the following formats: WAV (linear PCM) or WAV (G.
711a/µ).

Installation Manual References

4.3.3 DSP S Card (KX-NS0110), DSP M Card (KX-NS0111), DSP L Card (KX-NS0112)

PC Programming Manual References

2.1.4 Easy Setup Wizard
7.2.5 Utility—File—Message File Transfer PC to PBX
7.2.6 Utility—File—Message File Transfer PBX to PC
10.3 PBX Configuration—[2-3] System—Timers & Counters—DISA / Door / Reminder / U. Conf—DISA—
Progress Tone Continuation Time before Recording Message
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—OGM Record / Clear /
Playback
11.5.2 PBX Configuration—[3-5-2] Group—Incoming Call Distribution Group—Queuing Time Table—
Queuing Sequence—Sequence 01–16
13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message—Floating Extension
Number

Feature Manual References

5.5.7 Floating Extension
6.1 Capacity of System Resources

Operating Manual References

2.1.5 Recording Outgoing Messages (OGM)

Feature Manual 243

2.29 User Container

2.29 User Container

Description
The User Container feature allows you to easily configure the maintenance settings per user and provide
services by linking the user to the device.
By using this feature, you can easily use various features provided by in the PBX.
• Basically, all of the devices integrated into the PBX must belong to either one of the users.

The following is an overview of the User Container.

By creating a system that considers the "User" as the centre of the system, it is easy to configure, operate,
and maintain the system.
Hereinafter, in the NSX series integrating view, the Extension User Group shown by the red frame is a
container for linking the user to the usable devices, and will be referred to as "User Container". The User
Container is provided as a unit provided with the system setting and call control services.

The integration configuration of the User Container is shown below.

Tenant #A Tenant #B

Paging Pickup #A Paging Pickup #B

Call Pickup #A Call Pickup #B Call Pickup #V

Extension User Grp #1 Extension User Grp #2

User A User B User C User D User E

Extension: Extension: Extension: Extension: Extension:
100 101 201 UM Mail-box 102 203

User A User D
User B User E
Wired terminal User C Wireless terminal
Wired terminal
Wired terminal Wired terminal
Wireless terminal Wireless terminal Wireless terminal

KX-NSX Series Integration Configuration (User Container)

Example of the figure

• User A, B, and C are assigned to User Group #1,
the main device of User A is assigned to the device with extension number 100.
User B is assigned to Extension 101 and User C is assigned to extension 201 in the main device.
• Similarly, users D, and E are registered as members of the extension group in User Group #2, User D is
assigned to device extension number 102 and User E is assigned to device extension number 203 and
are signed in to the main device.

244 Feature Manual

 2.29.1 User Container

• Extension User Group＃１ belongs to tenant Group A, Call Pick Up Group #A, #B, and Paging Group #A,
#B, and Extension User Group #2 belongs to tenant group B, Call pick Up Group #V, and Paging Group
#B.
• In addition, all the users A, B, C, D and E have, besides the main device, wired devices to function as sub
devices.
• This way, the User Container treats wired, wireless and public devices as one user, and groups relating to
extension terminals of tenants, paging and call pickups etc. are able to easily setup and use various PBX
services provided by the KX-NSX series SIP phones by specifying this in the user unit.

2.29.1 User Container

2.29.1.1 User Container Configuration
Description
The User Container is a "container" for linking the user to the usable devices.
You can easily monitor the users and devices by using this container.
All wired extensions, wireless extensions, Public (general-purpose cellular) device and the extension
terminals integrated in the PBX, must be integrated into one of the User Containers.
1. User Type
Depending on the type of Activation Key, the available services can be configured for each device
linked to the user. Usable devices vary depending on the Activation Key.

The configurable devices depending on the Activation Key are shown below.
Grade Main device Sub device Wireless device Public device UM Mailbox
Advanced user Usable Usable Usable Usable Usable
Mobile user Usable Unusable*1 Usable Usable Usable
Normal user Usable Unusable Usable Unusable Usable

*1 Only the mobile softphones (KX-UCMA) can be used by mobile users as sub devices.

In addition, the availability of certain usable features can be configured with the Activation Key.
Auto 2way-rec Manager UM E-mail conjunction 2way-
Grade UC
authority Rec
Advanced user Usable Usable Usable
Mobile user Usable Unusable (Note 1) Usable
Normal user Usable Unusable (Note 1) Usable

(Note 1) Normal and Mobile users can be assigned by the Supervisor in the settings, but the Auto
2way-rec manager authority feature will not be operational.
2. Activation Key
The user can be activated in the Web Maintenance Console when the Activation Key for each user
type is correctly installed.
An error message is displayed to the user if the required Activation Key is not installed when activating
a user.
As long as you do not delete the User Container, the Activation Key used by an already activated user
cannot be used for other users.
However, the information of the User Container does not have to be deleted in order to perform a user
type upgrade.

Feature Manual 245

2.29.1 User Container

If the KX-NSXP101 (Ultimate Activation Key for Panasonic Terminal Users) is installed, all User
Containers can be used as Advanced Users.
3. Amount of Users and Device Resources
The amount of User Containers supported by the KX-NSX1000/2000 of the NSX Series varies.
Model Max. amount of integrable users
NSX1000 1000
NSX2000 2000

Multiple devices can be registered to each User Container, however, because there is a limit on how
much devices a system can integrate, not all User Containers can use various kinds of devices
unblocked.
For that reason, the total number of main, sub-, wireless, and public devices needs to be less than the
maximum amount of users. Below examples using NSX2000 are shown.
User Container # Criteria
1 2 3 … 1000/2000
Storage Main Wired Wired Wireless … Wired Check the
device following during
Sub Wired Unregistered … … Wired
configuration
Wireless Unregistered Registered … … Unregistered when adding
users
Public Registered Unregistered Registered … Registered
• The total
amount of
main, sub-,
wireless, and
public devices
is less than
2000
UM Registered Registered Registered … Registered No criteria. (the
number of UM
Mailboxes is
unblocked)

Note
An image displaying the amount of users is shown below.
Because the user can integrate multiple
devices, the amount of registrable users
depends on the amount of remaining
devices.
*The maximum amount of users will not
exceed the maximum amount of
integrated devices.
User User 1 User 2

1 2 3 4 5 6 2000
Amount of
devices ...

246 Feature Manual

 2.29.1 User Container

The AK type according to each grade is shown below.

Name Product Number Grade
Normal User AK (1 user) KX-NSUN001W Normal User
Normal User AK (10 users) KX-NSUN010W Normal User
Normal User AK (50 users) KX-NSUN050W Normal User
Normal User AK (100 users) KX-NSUN100W Normal User
Normal User AK (500 users) KX-NSUN500W Normal User
Mobile User AK (1 user) KX-NSUM001W Mobile User
Mobile User AK (10 users) KX-NSUM010W Mobile User
Mobile User AK (50 users) KX-NSUM050W Mobile User
Mobile User AK (100 users) KX-NSUM100W Mobile User
Mobile User AK (500 users) KX-NSUM500W Mobile User
Advanced User AK (1 user) KX-NSUA001W Advanced User
Advanced User AK (10 users) KX-NSUA010W Advanced User
Advanced User AK (50 users) KX-NSUA050W Advanced User
Advanced User AK (100 users) KX-NSUA100W Advanced User
Advanced User AK (500 users) KX-NSUA500W Advanced User
Ultimate Activation Key for Panasonic Terminal KX-NSXP101 Advanced User

Conditions
• A User Container contains the following types of devices:
a main device, a sub device, a wireless device, public devices, and a UM mailbox
• A maximum of 4 public devices can be used at the same time.
• The user can use PBX service features on all devices by using the extension number of the main device
as a representative number.
This representative number is referred to as the main extension number.
• When the DSS button of the main device is designated to the number other than the user extension
number, it turns on red and doesn't function.
• A sub device cannot be specified to the PS if the PS is designated to a main device.
• The main device, sub device, wireless device, and public devices all have limitations as a usable device.
Those limitations are shown in the table below.
Device type Usable terminals Operating conditions
Main device IP-PT, IP Softphone, SIP The sub devices and/or wireless devices cannot be
terminal, DPT, SLT, DECT PS, used when using the DECT PS as the main device.
T1-OPX, ISDN extension, The SIP terminal cannot perform the Service-out
APT *1 operation.
Sub device IP-PT, IP Softphone, SIP Operates as the equivalent of the One Numbered
terminal, DPT, SLT, T1-OPX Extension of the main device. The SIP terminal cannot
perform the Service-out operation.

Feature Manual 247

2.29.1 User Container

Device type Usable terminals Operating conditions

Wireless DECT PS Operates as the equivalent of the SXDP Extension of
device the main device.
Public device General fixed telephone/ Operates as cellular interlocking terminal of the main
cellular phone device.

*1 The APT can only be integrated to the Expansion Gateway and Legacy Gateway multi-connected to the NSX1000/2000.

• There is no user limit on the amount of integrated users, the total consists out of the total amount of
extension numbers configurable to the devices (main, sub, and wireless combined) + the amount of
public devices (Note 1).
(Note 1) The amount of usable public devices can be configured for each user and the total amount of
configured devices becomes the amount of public devices.
• The criteria for the usable amount of public devices differs depending on the user grade.
Normal User: "0" (unchangeable)
Mobile/Advanced User: "0–4" (initial value)

PC Programming Manual References

8.1 Users—User Container

Operating Manual References

1.13 User Device Management

2.29.1.2 Individual User Information Management of the Construction

and Maintenance Personnel
Description
An operation to add users and edit user information is provided in the Web Maintenance Console. The
required sub/public device according to the user's grade is selected when adding new users, or when editing
the users after you have added them.
The UM mailbox is automatically created, automatically associated with the user and can be used
immediately after adding a new user.
The system will automatically create the appropriate group configuration data for the public device.
Therefore, the group resources of the incoming call group for mobile interlocking are not needed.

1. Add a User
A feature which specifies unused extension numbers and creates multiple users at once or individually
is provided.
The specified extension number is set as the main extension number and a User Container is created
with initial values for the other setting items.
Note that a UM mailbox will automatically be created when adding a new user. For details about the
conditions for creating a mailbox, see "2.29.2.6 UM (Mailbox)".
New user data can be created by making a template of existing user data.
If you select "generate based on the template", the contents of the fields below are copied and
automatically generated.

248 Feature Manual

 2.29.1 User Container

Data attribute Item

User Information User Group*1
Class of Service
WebMC Language
Admin Capability
Type of User
Main Device - Type
Telephony Feature PT Display Language
Flexible Button View/Edit for Wired Extension

*1 When a Sub-Administrator creates a user, the user group is configured as follows.

• If the Sub-Administrator has authority to manage the user group of the user template:
The user group number of the user template is set.

• If the Sub-Administrator does not have authority to manage the user group of the user template:
The lowest numbered user group that the Sub-Administrator is authorised to managed is set.

2. Edit User Information

The system administrator can change the user information of the User Container during system
operation.

Example of editing a user extension number is shown as the following.

[Before Editing]
Port User Container

301 Main Mailbox

301 301
302

303
Main Mailbox

302 302

[Editing]
a. Change user 301 to 302
User 302 already exists so this chance cannot be made.
b. Change user 301 to 303

Feature Manual 249

2.29.1 User Container

301 Main Mailbox

301 303 301 303

302

303

Automatically changes
The registration
in response to the user
number changed
number change

c. Change user 301 to 304 (number 304 does not exist in the port)

301 304 Main Mailbox

301 304 301 304

302

303

Automatically changes in Automatically changes

response to the change in response to the user
in the registered number number change

3. Deleting a User
The association with the sub device, settings used in order to use the public device, and the recording
data in the UM mailbox will all be erased when deleting a user.
The used Activation Key will also be released, allowing you to use it on other users. The information
related to the groups to which the user belongs will be cleared in the same way.
4. Activating a user
Activation is an operation which reflects the creation of a new user or changes in the user settings on
the actual operations.
At this time, the required Activation Key will be consumed.
5. User Container Back-up/Restore
It is not possible to restore or perform a separate backup of the User Container. Setting data of the
Export/Import feature and system data, can be alternated by the backup/restore function of the UM
data.

Conditions
• Users using a device can also be deleted, all calls of the operational state are released, and the device
will be serviced out. (Active calls will be ended when the user gets deleted.)
• When assigning an Activation Key that is currently being used to another user, the user using it must first
be deleted.
• Please be careful of the following things when adding a new user.
– A new user can be created, even if the device of the specified extension number or wireless device
does not exist when adding a user.
– The automatic re-create feature (automatically delete all the settings and regenerate the user with the
specified number) of the user is not supported.
– When you specified an extension number that is already being used while adding a user, an error will
be displayed, a user creation result log will be left, and you will be unable to add a user.

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 2.29.1 User Container

– Added users will not be deleted even if you perform the Service-out operation.
– You cannot register only a sub device or only a wireless device when the main device is not configured.
– The following will be displayed in the Service-out status, disabling you to use any services.
– However, the Forward setting is not affected by the Service-in/Service-out Operation.
Service-out state Service-in state

SEP.29 11:13AM MON

Service-out
PROG INFO RING MENU

Service-in operation by Access Number

• When target user of the configured settings of the User Container used by the System Manager is in use,
the configurations will not be applied.
– A user type upgrade is possible, however, because the target user becomes inactive due to the
upgrade, the re-activation operation needs to be performed afterwards. You cannot downgrade a user
type. A user needs to be deleted once in order to "downgrade".

PC Programming Manual References

8.1 Users—User Container

Operating Manual References

1.13 User Device Management

2.29.1.3 Editing User Settings

Description
The end user can change the various settings of the Access code or the various settings of each used
device in the User Web Settings.

1. User WEB Settings

A window displaying the configurable settings where specific items can be adjusted will be shown when
signed in with your own user account. Items of which you do not possess editing permission can only
be shown for reference.

Conditions
• Settings regarding the relation between the user and the devices cannot be configured.
• Inactivated users cannot login with the user ID.

PC Programming Manual References

8.1 Users—User Container

Operating Manual References

1.13 User Device Management

Feature Manual 251

2.29.1 User Container

2.29.1.4 Security (Password)

Description
After linking the users to the devices by use of the User Container, a password is needed in order to enable
the each user to use the device.

Conditions
• The passwords manageable by the user are the 2 listed below.
1. Mailbox Password
Used as your mailbox password and when servicing in or out
2. User PIN
Used when log in to the User Portal

Note
The mailbox password of each user is the same as the User PIN, which can be configured and edited in the
User Container. (Linked)

PC Programming Manual References

8.1 Users—User Container

Operating Manual References

1.13 User Device Management

2.29.1.5 Smart Desk Feature

Description
By using the service-in operation on the device user wants to use by entering their own extension number,
the user can use their extension as the telephone terminal.
For more information, refer to "2.24.4 Smart Desk Features".

2.29.1.6 Linking with LDAP User Information

Description
A part of the User Container settings can automatically be configured by linking the User Container to the
Active Directory (LDAP) that the user uses.
For more information, refer to "5.10 Configuration of Users—Synchronising User Information with an LDAP
Directory" in the Installation Manual.

Conditions
For the User Container information and the key to link to the LDAP Database (hereafter, referred to as "Link
Key"), "Enable (use "mail" for Microsoft® Active Directory®)" or "Enable (use "uid" for OpenLDAP™)" can be
selected from the Web-MC settings (User Container Synchronisation). (refer to "28.3.4.3 Network Service—
[3-4-3] Client Feature—Directory Service—User Container Sync" in PC Programming Manual).
If you change the (User Container Synchronisation) settings, the User Container and LDAP Database Link
Key will be updated. This setting can be changed during operation; however, as the Link Key is updated, the
synchronisation between the User Container and the LDAP Database will become unavailable.
After updating, in order to synchronise the User Container and the LDAP database again, it is necessary to
delete all users’ Link Keys and re-register them in the LDAP Database.

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PC Programming Manual References

28.3.4.3 Network Service—[3-4-3] Client Feature—Directory Service—User Container Sync

Feature Manual References

2.6.5 Speed Dialling—External (LDAP)

2.29.2 Call Control Feature

2.29.2.1 Main Device/Device Common
Description
By calling the user's main device, it is possible to send an incoming call signal to the main devices, sub
devices, wireless devices and public devices registered in the User Container at the same time.
It is also possible to send an incoming call signal to a sub device, wireless device and/or public device at a
predetermined time delay from the time of an incoming call to the main device depending on the user
settings.

Conditions
The delay time can be configured for each device (you can choose from immediately, 5 seconds, 10
seconds, and 15 seconds). For details about the incoming call conditions for each device, see the section of
the respective device.
When one of the devices registered to the user answers a call incoming on multiple devices, the incoming
signal on the other devices will be stopped and only the device with which was responded will be able to
speak with the outside party.
It is possible to use the rejection of incoming and Call Forward feature (Note 1) by performing the DND/FWD
settings on one of the devices owned by the user.
(Note 1) The regardless of the configured FWD type, everything operates as "FWD ALL". The operation
availability for this operation is can be selected in the System Settings. (Initial value: disabled)
[Status Display]
• The other devices of the same user will not be able to call when one of the user's devices is on a call.
The operation criteria differ between devices on a call and devices being used. (See the table below)
Busy telephone
Operational
Main Sub Wireless Cellular
telephone
Main (Off-hook) - Call interception Call interception Busy tone played
after pressing Dial
after hearing the DT
Sub (Off-hook) Call interception - Call interception Busy tone played
after pressing Dial
after hearing the DT
Wireless (Off- Call interception Call interception - Busy tone played
hook) after pressing Dial
after hearing the DT
Cellular A normal outgoing A normal outgoing A normal outgoing A normal outgoing
(Outgoing calls call is made via call is made via call is made via call is made via DISA
via DISA) DISA instead of DISA instead of DISA instead of instead of making a
making a call from making a call from making a call from call from a public
a public device. a public device. a public device. device.

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2.29.2 Call Control Feature

DSS
• The user status can be displayed by assigning the user's extension code to the DSS.
– When the user's main device is serviced in and the user status is on Idle (DND settings excluded), the
light of the DSS button is off.
The configured user can receive calls by pressing the DSS button when the light is off.
– The DSS button is flashing red during in an incoming call state supporting Call Pickup, and is lit red
during an incoming call state which does not support Call Pickup.
The call can be accepted by pressing the red flashing DSS key.
For information about the incoming call types supporting Call Pickup, see "2.4.3 Call Pickup".
For information about the LED flash patterns, see "2.21.3 LED Indication".
– The DSS button will light up red when the user enters one of the following states.
1. When the user inactive (including when the User Container is not created)
2. When the user's main device is serviced out
3. When one of the devices of the user is busy on a call
4. When a user has DND enabled
5. When in an incoming call state that does not support Call Pickup

PC Programming Manual References

8.1.1 Users—User Container—Add User/Edit User

2.29.2.2 Sub Device

Description
1. Standby
• The display during standby mode shows the main extension number.
2. Outbound call
• Operations performed from a sub device are treated as if performed from the main device.
3. Inbound call
[Parallel ringing]
– Calls to the user also arrive on the sub devices.
– The main extension number/extension name of the user is displayed on the extension display of the
calling party during an extension call to a sub device.
– The paging signal will be delivered to all extensions, the main extension, sub-extensions and PSs.
However, the paging signal cannot be delivered on SIP or SLT type terminals.
– LCS calls operate according to the settings of the main extension or sub extension that uses PT.
However, when the LCS call is set arrive to whichever extension, the main extension will be
prioritised.
– It is possible to send an incoming call signal to a sub device at a predetermined time delay from the
time of an incoming call to the main device.
The delay time can be configured for each device (you can choose from immediately, 5 seconds, 10
seconds, and 15 seconds).
– When the incoming call to the main device is stopped by the receiver modification feature, the
incoming call to the sub devices, wireless devices, and public devices will also be stopped.
– Recalls for the terminal handling such as, pending recalls and callbacks only arrive on the operation
device.

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Transferred recalls arrive on the transfer recall destination extension as configured in the main
extension.
4. Call
[Call interception between user devices]
– It is possible to forward incoming calls to other devices (operation terminal), by having any of the
main/sub-/wireless devices (except general purpose SIP terminals) performing one of the following
operations. (Call interception)
• Handset is off hook during standby.
• Press [SP-Phone] [Monitor] during standby.
– It is possible to intercept calls from general-purpose SIP terminals by using the Access Code
operation.
At this time, the caller will be placed on hold for a short while after which he/she will be connected to
the device that intercepted the call.
The Access Code operation can specify the main extension number of the user by use of the Call
Pickup operation. See "2.4.3 Call Pickup".
[Call waiting]
– Sounds heard during a call (such as call waiting tones and hold alarms), are only sent to the party
that is on the call.
[Setting data between devices]
• Sub devices operate based on the COS of the main device and/or the settings of each terminal. Key
settings (including PF key (One dial) and LCD data of self-labelling), or, specified priority calling/
specified wired calling also follow the device settings in main.

Conditions
• The paging signal cannot be delivered on SIP or SLT type terminals.
• Voice calls can only be made by main extensions that use PT.
• The settings of the main device are obeyed when receiving a call when already on a call. At this time the
incoming call is only displayed on the main device, however, by disconnecting the incoming call, the call
will be displayed on all the devices as configured in the delay settings. However, the call will immediately
arrive on the terminal on a call without applying the delay settings.
• It is not possible to directly call the extension number of the sub-extension with CTI control.
• Calls cannot be intercepted with another device if the calling device is in any of the following states.
– During extension DT heard, extension dialling, outside line DT hearing, outside line dialling
– During consultation hold
– During conference calls (can be used during 2 way recording)
• The settings per device listed below, operate based on the settings of the main extension/sub extension.
– On/Off of the BGM status
– LCS On/Off (when the main extension and sub extension both are on LCS-On "Private Mode" the LCS
incoming calls will be performed prioritising the main extension)
– Auto Answer Switching (switching by using the AUTO key of the PT)

PC Programming Manual References

8.1.1 Users—User Container—Add User/Edit User

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2.29.2 Call Control Feature

2.29.2.3 DSS Console

Description
The DSS console can be used with the sub devices configured in the User Container.

Conditions
The pair extension numbers of the DSS console can only be configured by the number of the main device
(=extension number of the user).
Extensions with DSS cannot Service-out.

PC Programming Manual References

10.9 PBX Configuration—[2-9] System—System Options—Option 4—DSS Key—DSS key mode for
Incoming Call

2.29.2.4 Wireless Device

Description
By calling the user's main device, it is possible to simultaneously send an incoming call signal to the
wireless devices that are registered in the User Container.
1. Outgoing call
• The operations performed from a wireless device are treated as operations performed from a main
device.
The information shown when making a call from a wireless device (extension number/extension
name, caller number/caller name) is in accordance with the user's caller information.
2. Incoming call
• By calling the user's main device, the wireless devices that are registered to the User Container will
also ring.
Voice-based calls are also possible for the wireless device.
• It is possible to send an incoming call signal to a main device at a predetermined time delay from the
time of an incoming call to the main device.
The delay time can be configured for each device (you can choose from immediately, 5 seconds, 10
seconds, and 15 seconds).
3. Call
[Call interception between user devices]
– It is possible to forward incoming calls to other devices (operation terminal), by having another
device performing one of the operations listed below, if any of the main/sub-/wireless devices
(except general purpose SIP terminals) is on a call. (Call interception)
• Handset is off hook during standby.
• Press [SP-Phone][Monitor] during standby.
* The conditions are listed in the section <Call interception between user devices> of "Sub devices".
[Call waiting]
– Sound during a call (call waiting tone / hold alarm, etc.) is carried out only for the person on a call.
– DSS button for public devices
The DSS button light of the extension registered first in an ICD Group will turn red when the
public device:
• is on a trunk call that was received via the ICD Group.

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• is on a trunk call that was made using the Walking COS Through DISA feature. (→
2.16.1 Direct Inward System Access (DISA))

Conditions
• The call doesn't arrive on the extension number of the wireless device.
• If Parallel Connection Call Mode is switched OFF, calls cannot be received on wireless devices.
However, although the incoming call is not displayed, it operates the same as your own extension during
a Non-parallel Connection. (Example) Calls are answered by performing the response operation.

PC Programming Manual References

8.1.1 Users—User Container—Add User/Edit User

2.29.2.5 Public (Mobile Phone) Device

Description
By calling the user's main device, it is possible to simultaneously send an incoming call signal to the public
devices that are registered in the User Container.
• Virtual extension number of the ICD Group, integrated into the other PBX in the closed number/open
number
→ An Incoming Call Distribution (ICD) Group can include extensions at other PBXs in a private network,
including the floating extension number of another ICD Group. This is done by assigning a Public Devices
user as a member of the ICD Group, and then setting the number of a destination at another PBX as the
forwarding destination for that Public Device within the User Container.
This allows multiple ICD groups at remote locations to receive calls together.
In addition to the Ring distribution method, Uniform Call Distribution (UCD) and Priority Hunting can also
be selected. (→ 2.2.2.2 Group Call Distribution)

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2.29.2 Call Control Feature

Telephone Company

ICD Group
PBX-1 Private network PBX-2
(KX-NSX1000/2000) (KX-NSX1000/2000)

ICD Group

User Container PBX-3

Main Device (KX-NSX1000/2000)
•••
Public Device 1 ICD Group
Public Device 2
PBX-4
Public Device 3
(KX-NSX1000/2000)
Public Device 4
ICD Group

PBX-5
(KX-NSX1000/2000)

• PS extension numbers integrated in other PBX's

Example: Image of an incoming call on a cellular phone of another network

Telephone Company

PBX-1 Private network PBX-2

(KX-NSX1000/2000) (KX-NSX1000/2000)

User Container PBX-3

Main Device (KX-NSX1000/2000)
•••
Public Device 1
Public Device 2
PBX-4
Public Device 3
(KX-NSX1000/2000)
Public Device 4

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• A mobile number operating in a mobile network

Example: Image of an incoming call on a cellular phone of another network

Telephone Cellular
Company Company

PBX-2 TIE PBX-1

(KX-NSX1000/2000) (KX-NSX1000/2000)

User Container
Main Device
•••
Public Device 1
Public Device 2
Public Device 3
Public Device 4

Conditions
1. Inbound call
• Calls can be made to a public device (up to 4 devices) registered in the User Container by calling to
the extension number registered to that Container.
The outside line is acquired through the outside line acquisition method registered in the User
Container, and automatically calls the relevant public device.
It is possible to send an incoming call signal to each public device, at a predetermined time delay
from the time of an incoming call to the main device. The delay time can be configured for each
device. (you can choose from immediately, 5 seconds, 10 seconds, 15 seconds, and 20 seconds)
• When delayed a call is received on the public device you can select whether or not to stop the
incoming of the call to the main device, sub device and wireless device.
The settings are enabled if there is even a single public device of which the delay time is set to
anything other than immediate, the main devices, sub devices and wireless devices will stop
receiving calls at the timing when delayed ringing is triggered. (In case multiple public devices have
the delayed call setting configured, the settings will become active at the first ring.)
2. During a Call
[Transfer Call After Answering]
• You can transfer a call from devices such as a cellular device to other extensions when responding
to a call received via the PBX on an outside line.

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2.29.2 Call Control Feature

PSTN Mobile network

Transfers to extension
101 by entering
DTMF DTMF "#101"
#101

101 102

This feature can only be used from a mobile device (public) linked to the User Container.
• The call is treated the same as when "Hooking" from an SLT extension of an SLT extension call by
the "#" sent to the PBX outside line from the cellular device during a call.
• The outside line of the other party can temporarily be put on hold, and depending on the dial
operation after put on hold an extension can be called or the call can be transferred to an extension.
The outside line performing the temporary hold action will hear the 1A dial tone, the side being put
on temporary hold will hear the hold tone just like when a normal extension would be put on hold.
(the tone will stop playing when a number is dialled)
• When the "outside line - propriety of the transfer of calls originating from an outside line" setting of
the System Unit is set at "Disable", the "# of DTMF" will be disabled without starting temporary hold.
Outside line (Enabled as the "DTMF" of call delay) - Only when using an outside line <Optional
Device – Voice Message – DISA System – CO_CO With DISA>
• Call restriction, and the connection regulations are applied to the transfer destination, following the
COS information configured by the user of the cellular device.
All the extension types are suitable as a destination extension.
[Call interception]
• Intercepting calls on a public device with other devices, and from other devices with a public device
is not possible.
For the operation criteria, see 2.29.2.2 Sub Device—Description—4. Call—[Setting data between
devices].
[Call Waiting]
• Sounds heard during a call (such as call waiting tones and hold alarms), are not sent to wireless
public devices.
3. Calls from a cellular phone
[Password input abridgment of a cellular phone (DISA)]
It is possible to use DISA from the telephone number of a public device registered to the User
Container without having to input the extension + password.
Depending on the caller ID notified at call initiation, the call will be treated as if the "Walking COS
Access Code + main number of the User Container + Pin" was automatically entered after DISA's
response, when DISA responds to the call with a telephone number matching that of the public device
registered in the User Container.
It is also possible to use DISA from the telephone number registered to System Speed Dial without
having to input the extension + password.
Depending on the caller ID notified at call initiation, the call will be treated as if the "Walking COS
Access Code + extension number of the CLI caller destination extension + Pin" was automatically

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entered after DISA's response, when it is registered in "CLI caller destination extension" when DISA
responds to the call with a telephone number matching the one registered in System Speed Dial (And
caller ID table). (Applies for all for NS Series)
The "CLI caller destination extension" includes, "CLI caller destination extension" settings and
"Automatic password authentication extension" settings.
Speed dial number Telephone number Name Call extension
000 90344445555 Suzuki 105
001 90923334444 Yamada
002 80100112013334444 Tom 201

The telephone numbers of public devices registered to the User Container are prioritised over
telephone numbers registered to System Speed Dial.
• After the extension number + password is automatically entered, the recorded OGM will be played
instead of the dial tone.
The same goes for when the extension number + password are not automatically entered at timings
such as when, the OGM message is playing or when manual DTMF dial is enabled.
• Automatic input will not be performed, when the settings for the propriety of using DISA without
entering the expansion number + password is set on "Disabled".
Optional Device – DISA – System – Remote Walking COS through DISA without PIN (commonly
used for the NS Series)
• An Advanced user licence or a Mobile user licence is required to use this feature (password
abridgment from DISA). The same goes for when the Availability settings of this feature are set to
"Disabled" at time of function limitations.
• When calling again by using the "*" dial key, the message (OGM replay) will be heard without having
to enter the Walking COS Access Code + extension number.
• The same goes for when SMDR, CTI, billing, and call restriction are entered manually.
• "Password automatic authentication extension" will be treated as if automatic input was available
(walking COS enabled for the configured extension), even when it does not have a PIN setting
(including when it is locked).
• The extension registered first in an ICD Group can program the Forward settings for trunk calls to up
to 4 public devices through PT programming. Using this feature, an extension user can assign his
cellular phone to ring with his PT, so that he can easily receive trunk calls even when not at his desk.
4. Call Log
• The Public (Mobile phone) device does not make incoming/outgoing call logs.
5. Feature Number
• When an incoming call is received at a public device where an OGM is being played, only the
following features are available.
For details about feature numbers, refer to "5.5.6 Flexible Numbering/Fixed Numbering".
– Operator Call
– Idle Line Access (Local Access)
– Trunk Group Access
– TIE Line Access
– Speed Dialling-System
– Group Paging
– S-CO Line Access

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2.29.2 Call Control Feature

– Account Code
– External Relay On
– FWD/DND set/cancel – Both
– FWD/DND set/cancel – External
– FWD/DND set/cancel – Internal
– FWD/DND No Answer Timer Set
– Group FWD/DND Set / Cancel – Both
– Group FWD/DND Set / Cancel – External
– Group FWD/DND Set / Cancel – Internal
– Login / Logout
– Absent Message Set / Cancel
– Station Lock Set / Cancel
– Time Service Switch
– Station Password Set
– Quick Dial
• When an incoming call is received at a public device, only the following features are available after
the device user presses "#".
For details about feature numbers, refer to "5.5.6 Flexible Numbering/Fixed Numbering".
– Idle Line Access (Local Access)
– Trunk Group Access
– TIE Line Access
– S-CO Line Access

PC Programming Manual References

8.1.1 Users—User Container—Add User/Edit User

2.29.2.6 UM (Mailbox)
Description
• By assigning a main extension when adding a user, an UM mailbox with the same the extension as the
user number will be created automatically, and it will automatically be associated with the user.
Even when you change the user extension number, the recorded files of linked mailbox are saved and the
mailbox will continue to be linked to the user (the number of the mailbox will change as well).
If you delete the user or delete the user's extension number, all recorded information etc. will be cleared.
• The mailbox cannot be generated from the UM settings (Mailbox settings menu). The mailbox can only be
generated as an effect of a user being generated.
Also, the below listed mailboxes will be made separately from the user mailbox.
ICDG Mailbox At the initial start-up the ICDG mailbox itself automatically creates mailboxes
respective to the number of ICDG.
If the ICDG extension number is changed, the number of the respective mailboxes
will also be updated.
Manager Mailbox The mailbox for the system manager of the tenant unit and the mailbox for the
message manager will be automatically created during initial start-up.

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Automatic Call Generation by the setting screen of the Automatic Call Recording Mailbox (the
Recording same for NS)
Mailbox

• If a new e-mail has been registered in the UM, all the MSW lamps of the main device, sub device, and
wireless device will light up.

Conditions
• The below listed information is automatically reflected to the mailbox registered to the user by the settings
configured in the User Container.
If you change the settings in the User Container, the settings of the mailbox will also be updated
accordingly.
On the other hand, the settings of the mailbox registered to the user cannot be adjusted in the settings
menu of that mailbox.
– Mailbox Password (input User PIN)
– First Name, Last Name
– Voice Mail Notification, Email address
– COS
• You can intercept calls from other devices when recording a call, however, it will be saved as another
message after intercepting.
[Auto 2 way REC]
• The main device, sub device and wireless device of the users supporting automatic call recording are call
recording compatible.

Feature Manual References

3.1 Unified Messaging System Administration

2.29.2.7 Various Logs and Reports

Description
• The various logs of the sub devices, wireless devices, and public devices are stored as if from the main
device.

Conditions
• User SMDR and billing (Hotel Charge) information is recorded as data of the operations of each device
which are recorded as operations of the main device.
When calls are restricted due to the total billings at the main device, the locked state is also applied to the
other devices, restricting their calls as well.
• Regarding the user's outgoing call log, incoming call log and message waiting log (hereafter referred to
as MSW log), the operations of each device and operations regarding the sub extensions are recorded in
every log of the user's main device.
The sub device and wireless device can refer to operate/delete the contents from the main extension of
every log by use of the confirmation operation.
However, for UT terminals and general-purpose SIP terminals/SIP Softphones, in order to operate and
record incoming and outgoing call logs when the terminals are set as sub extensions, see operations in
accordance with the settings from the sub device to the main device/main extension log.
• The information on all the devices of the User Container is recorded as information on the main device in
the ACD report/call monitor.

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2.29.2 Call Control Feature

2.29.2.8 CTI
Description
Operation from any of the devices owned by the user (main/sub/wireless/public) will be notification
controlled for CTI by the extension number of the main device of the user.

Conditions
• Extensions supporting CTI control are; IP-PT, IP Softphone, SIP terminal, DPT, SLT, and DECT PS.
• When receiving a response with CTI client for the user call and for Makecalls originating from the CTI
client, you can switch between which user-owned device receives a response and which device makes a
Makecall with the following methods.
1. Select in the User Portal menu
CTI control device selection: main (initial value)/wireless/sub/public 1/public 2/public 3/public 4

* Only devices in the Service-in state can be selected, but KX-NSX Series cannot be selected as a
public device.
When a selected devices services out, the control device is switch back to main (initial value).
2. Access code operation of the device
Devices you want to have support CTI control can be configured with the Access Code.
This Access Code operation allows you to conjunct with the settings from, 1. and automatically
switch between the two.
[Operation]
Dial the "CTI control support registration Access Code" when hearing the dial tone of the extension.
→ Can hear the confirmation tone
• The CTI feature in the SIP terminal is partially limited. Consult your dealer for more information.

Feature Manual References

2.26.1 Computer Telephony Integration (CTI)

Installation Manual References

3.1.3 Using CTI Applications

264 Feature Manual

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Unified Messaging System

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3.1 Unified Messaging System Administration

3.1 Unified Messaging System Administration

3.1.1 Unified Messaging System Overview
Description
The KX-NSX Series PBX has a built-in messaging system that provides voice mail and fax services to its
subscribers. The Unified Messaging system can also provide voice guidance to outside callers, either
directing them to their desired destination or to the mailbox of a subscriber, where they can leave a voice
message.
Users
The user owns his/her own mailbox.
The following three types of users exist in the Unified Messaging system:
• Subscriber Max 1608/KX-NSX1000, 2608/KX-NSX2000
A subscriber is an extension user who has a mailbox assigned to his extension. Subscribers can play
back messages saved in their mailboxes, leave messages for other subscribers, record a message that is
then sent to multiple parties (including outside parties), record greeting messages, and more.
• Message Manager 1 Manger for each tenant
The message manager is in charge of the general delivery mailbox, as well as performing some setup,
such as recording prompts and changing notification settings. Through system programming, subscribers
can also be assigned Message Manager privileges.
• System Manager 1 Manger for each tenant
The system manager is in charge of the Unified Messaging system as a whole, and can access many
settings. The system manager’s duties include setting up mailboxes, assigning COS settings, and
changing the service mode.
UM Ports and the UM Group
A port acts as a pathway into the Unified Messaging system, so when a call is directed to the Unified
Messaging system, it requires one available UM port. The total number of available port is 128 for a PBX.
The UM ports of a PBX belong to the PBX’s UM group. This group has a floating extension number, which
can be the destination for incoming calls, redirected calls, transferred calls, etc. When incoming calls are
received at the floating extension number of the UM group, calls will hunt starting at the lowest UM port
number. Once an available port is found, the service assigned to the port, trunk, etc., determines how the
call is handled (→ 3.2.1.40 Service Group). Services include features such as Voice Mail service
(→ 3.2.1.46 Voice Mail Service), which allows a caller to leave a voice message at a subscriber’s mailbox,
and Automated Attendant (→ 3.2.1.3 Automated Attendant (AA)), which directs the caller to a subscriber’s
extension.
Incoming call

× (Busy) × (Busy) × (Busy)

UM UM UM UM UM
Port 1 Port 2 Port 3 Port 4 Port 5

UM Group

Fax Server
• The Fax server is provided as a standard feature on the NSX. Received Faxes will automatically be
saved to the mail box as a Fax message. Fax messages stored in the mailbox can be managed, operated

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the same way as voice messages and Fax messages that are stored in the mailbox can be managed and
operated in the same way as voice messages, and can be viewed on a PC by via the IMAP client, output
to the specified Fax machine, and be sent with an e-mail as an attachment. A FAX Card, as well as an
Expansion Gateway which can mount the FAX Card is needed in order to receive a FAX. (1 channel/1
card)

Time Management
• When an Expansion Gateway is installed at a site with a different Time Zone, services such as Time
Mode can be applied as below:
1. Having the calling party (Originator) follow the Gateway settings of the recipient.
2. “Follow the site time set by each tenant?"
Either of the above can be chosen.
• The time stamp of the messages recorded in the mailbox is set according to the time settings of the
Gateway connected to the main device of that user's mailbox.
• The time stamp on the messages in a mailbox not associated with any user is set according to,
– Auto 2way-rec mailbox: the site time of the main extension of users with the ability to record calls.
– ICDG mailbox: the site time of the call line (outside line/extension)
– Administrator's mailbox: the site time of the tenant operator (DAY-Mode) of the administrator's mailbox.
If the tenant operator (DAY-Mode) is not set, the time will be set according to the time of site number
'0'.

Conditions
When the system is configuring the Multi-connection on a NSX series model, the UM service is monitored in
the Master Unit.
• As the Unified Messaging system is part of the same system as the PBX, the Unified Messaging system’s
data coordinates with PBX settings. For details, refer to "5.10 Configuration of Users" in the Installation
Manual.
• Each port is assigned an extension number.
→ 9.7 PBX Configuration—[1-1] Configuration—Slot—UM Property—UM Port Property—Main—
Extension Number
• It is programmable whether the calls queue when all ports in the group are busy through system
programming. If queuing is disabled through system programming, the call will be redirected to the
intercept destination of the first member extension of the UM group.
• UM function needs Storage Memory Card to record message such as KX-NSX2135 (200 Hours),
KX-NSX2136 (400 Hours), KX-NSX2137 (800 Hours) and KX-NSX2138 (1600 Hours)
• DSP Resource Usage
Connecting to the Unified Messaging system (including using features such as Two-way Recording)
requires a certain number of DSP resources. If all DSP resources are in use, this operation cannot be
performed. To ensure a minimum level of performance, DSP resources can be reserved for Unified
Messaging operations. (→ 5.5.3 DSP Resource Usage)

Notice
Reserving resources for Two-way Recording (→ 3.2.1.4 Automatic Two-way Recording for Manager,
→ 3.2.2.34 Two-way Record/Two-way Transfer) reserves the necessary number of UM ports
exclusively for Two-way Recording. For example, if 2 UM ports are available and you reserve
resources for 2 Two-way Recording sessions, both UM ports will be reserved for Two-way Recording,
and the Unified Messaging system will not be available for other uses.

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3.1.2 System Administration

Two-way Recording Incoming Call

UM UM
Port 1 Port 2

Reserved (Two-way) Reserved (Two-way)

UM Group

To provide access to the Unified Messaging system in this case, either the number of UM ports must
be increased or the number of resources reserved for Two-way Recording must be lowered.
• When a trunk call is made from the UM port, TRS/Barring Level 7 is applied (→ 2.7.1 Toll Restriction
(TRS)/Call Barring (Barring)). However, when a trunk call is made from the UM port using a feature
number for System Speed Dial, this call will not be restricted.

Installation Manual References

4.3.2 Expansion Storage Memory Card (KX-NSX2135, KX-NSX2136, KX-NSX2137, KX-NSX2138)
5.10 Configuration of Users

PC Programming Manual References

9.7 PBX Configuration—[1-1] Configuration—Slot—UM Property—UM Port Property
10.5 PBX Configuration—[2-5] System—Holiday Table
10.9 PBX Configuration—[2-9] System—System Options—Option 9
11.7 PBX Configuration—[3-7] Group—UM Group

Operating Manual References

1.8 Using the Unified Messaging Features

3.1.2 System Administration

Description
System administration (programming, diagnosis, system prompt administration, etc.) can be performed by
the System Administrator using Web Maintenance Console. For more information, refer to the PC
Programming Manual.

3.1.2.1 Custom Service Builder

Description
Allows the System Administrator to create Custom Services visually, using Web Maintenance Console. Each
Custom Service and its functions can be edited and arranged using this feature.

PC Programming Manual References

23.3 UM Configuration—[4-3] Service Settings—Custom Service

Feature Manual References

3.2.1.15 Custom Service

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3.1.2.2 Reserved
3.1.2.3 Password Administration
Description
Allows the System Administrator or System Manager to clear a subscriber password (so that a new one can
be assigned).

Operating Manual References

Manager Operation
2.2.1 System Manager Features—Setting Up Mailboxes

3.1.2.4 System Backup/Restore

Description
Allows the System Administrator to back up or restore the following data as individual files: System Prompts,
Mailbox Prompts, and Mailbox Messages. Data can be backed up in 2 ways:
• Manual Backup: The specified data is backed up manually.
Backup data can be saved to the following destinations:
– An external USB memory device
– NAS
– The local PC where you are running Web Maintenance Console
• Scheduled Backup: The specified data is backed up automatically at the specified times. Backup data
can be saved to an external USB memory device or a NAS. When scheduled backup is enabled, the
following parameters can be set:
– Set whether to back up messages received only within a specified period.
– Set whether to back up messages only when the remaining capacity of the mailbox reaches a certain
level.
– Set whether to delete messages from the mailbox after backup.
– Set whether all messages or just old messages are backed up.
A record of backup information (Description, Completion Status, Date & Time, and Total Elapsed Time) can
be viewed in the backup history.

Conditions
• KX-NSXU003 (Activation Key for Message Backup) is required to use the scheduled backup feature. One
activation key is required for each site where this feature will be used.
• If data is being backed up to the local PC, individual messages that are larger than 100 MB (about 3.5
hours long) cannot be backed up. However, even if some messages cannot be backed up, all other data
will be backed up normally.
• If data is saved to a USB memory device, be sure to unmount the device before removing it from the PBX
to avoid data loss or corruption.
→ 4.1.5 Status—Equipment Status—USB

PC Programming Manual References

6.8 Tool—UM Data Backup
6.9 Tool—UM Data Restore

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3.1.3 Fax Server

3.1.2.5 System Reports

Description
There are several System Reports available to the System Administrator to monitor operating status. The
System Administrator can print or export the System Reports. Certain reports can be printed in tabular form
or graph form.
The following reports can be generated:
• Mailbox Information Report
• Call Account Report (Group calls by UM Extn)*1*2
• Call Account Report (Group calls by Mailbox)*1*2
• UM Extn Usage Report*1*2
• Memory Usage Report*1*2
• Mailbox Usage Report*1*2
• Fax Transfer Report*2
• Call Handling Statistics Report*2
• Custom Service Report*2
• Message Status Report
• Subscriber Setup Report
• Security Information Report
• Hourly Statistics Report*2
• Fax Server Report
*1 This report can be printed in tabular form or graph form.
*2 This report can be set to be deleted automatically. This report is stored expansion Storage Memory Card, not default installed
Storage Memory Card.

PC Programming Manual References

7.5 Utility—Report

3.1.2.6 System Security

Description
Protects the Unified Messaging system from unauthorised programming and/or use. By default, the System
Manager and Message Manager cannot access the system from their telephones until the System
Administrator enables the relevant settings and sets a password. The System Administrator can also
choose to set a default password for subscribers’ mailboxes. If this setting is enabled, a default password is
automatically assigned when mailboxes are created.

PC Programming Manual References

26.1 UM Configuration—[7] System Security

3.1.3 Fax Server

Description
If an optional fax interface card is installed on the expansion Gateway, the PBX can receive, distribute, and
send faxes. Depending on the Expansion Gateway configuration, faxes from up to 31 channels can be

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integrated. Received faxes can be saved in mailboxes and then forwarded, printed, downloaded, etc. Also,
subscribers, the Message Manager, and the System Manager can send faxes. Each Fax board should be
assigned virtual extension number, then assign pilot number (FAX Main number) for these FAX number.
This mainly manages the FAX device connected to several sites and assigns the vacant FAX devices.
The Unified Messaging system can be configured to receive faxes in the following ways:
Method Explanation
Trunk/port service Setting the incoming call service of a service group to fax service
and then assigning the service group to a port allows faxes to be
received at that port.
The mailbox to which faxes are forwarded must also be specified.

→ 22.1 UM Configuration—[3-1] UM Extension / Trunk Service—

Service Group—Day, Night, Lunch, and Break Mode -
Incoming Call Service
Caller ID/PIN routing Setting the transfer destination for caller ID routing or PIN routing
to fax service allows faxes to be received from the specified
phone number.
The mailbox to forward faxes to must also be specified.

→23.1 UM Configuration—[4-1] Service Settings—Caller ID / PIN

Call Routing—Caller-ID Mode
→ 23.1 UM Configuration—[4-1] Service Settings—Caller ID / PIN
Call Routing—PIN Mode
DIL/DDI/DID/MSN routing Routing a call to the Unified Messaging system via
DIL/DDI/DID/MSN and assigning that entry a UM service group
(→ 3.2.1.40 Service Group) whose service type is Fax allows
faxes to be received through that line.

→ 18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL

Table & Port Settings—DIL—UM Service Group No.
→ 18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI /
DID Table—UM Service Group No.
→ 18.4 PBX Configuration—[10-4] CO & Incoming Call—MSN
Table—MSN—UM Service Group No.
Custom service Setting the service type for a custom service to fax service allows
faxes to be received through that custom service.
The mailbox to forward faxes to must also be specified.

→23.3 UM Configuration—[4-3] Service Settings—Custom

Service
Top menu of a call service If a fax tone is detected at the top menu of a call service, the fax
is received and then saved in a specified mailbox.

→ 24.4 UM Configuration—[5-4] System Parameters—

Parameters—Fax Management

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3.1.3 Fax Server

Method Explanation
During playback of personal If a fax tone is detected while a subscriber’s personal message is
message, or during message being played or while a message is being recorded, the fax can
recording be received at that mailbox.
If the sender leaves a voice message before sending the fax, the
voice message is attached to the fax as a voice comment.
DISA If a DISA line detects a fax tone and forwards the call to the UM
group’s floating extension number (→ 2.16.2 Automatic Fax
Transfer), the call is forwarded to a specified mailbox.

→ 24.4 UM Configuration—[5-4] System Parameters—

Parameters—Fax Management

Conditions
• Hardware requirement:
FAX Card (KX-NS0106) with KX-NS1000 as Expansion Gateway
• COS programming determines which subscribers/mailboxes can send and receive faxes.
• A header is added to each page, except the cover page*1, of all sent faxes. Through system programming,
it can be specified whether the sender’s information (system fax number) or the recipient’s information
(name or fax number) is included in the header.
→ 24.4 UM Configuration—[5-4] System Parameters—Parameters—Fax Management—Page
Retransmission
• Fax activity is recorded in the fax report. (→ 3.1.2.5 System Reports)
• Simplified Isolated Mode
• Even if a Fax Card is integrated into the Expansion Gateway, the fax feature cannot be used during
Simplified Isolated Mode
• Error handling
Through system programming, it can be specified whether a fax is retransmitted if an error occurs while
sending a fax.
→ 24.4 UM Configuration—[5-4] System Parameters—Parameters—Fax Management—Page
Retransmission
*1 If the fax is sent from a computer (→ 3.2.2.15 Fax Driver), the header is printed on the cover page as well.

PC Programming Manual References

5.3 System Control—FAX Card
9.4 PBX Configuration—[1-1] Configuration—Slot—System Property—Fax
9.5.2 PBX Configuration—[1-1] Configuration—Slot—Site Property—FAX Card
20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Fax Options
21.1 UM Configuration—[2] Class of Service—General—Fax Option
24.4 UM Configuration—[5-4] System Parameters—Parameters—Fax Management

Feature Manual References

3.2.1.31 Message Waiting Notification—Telephone Device
3.2.2.2 Automatic Fax Delivery
3.2.2.15 Fax Driver
3.2.2.30 Send to Fax Machine

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3.2 System and Subscriber Features

3.2.1 System Features
Description
System management can be performed by the system manager and the message manager using an
extension telephone, or by the System Administrator using Web Maintenance Console.
An example of items which are programmed by the system manager include:
• Creating, editing, deleting, and resetting mailboxes
• Class of Service (COS) settings
• Changing the company greeting
An example of items which are programmed by the message manager include:
• General Delivery Mailbox maintenance
• Message notification settings
• Recording/deleting messages (system prompts, company greetings, custom service menus, system caller
names, etc.)

3.2.1.1 Alternate Extension Group

Description
A group of extensions that require a different call transfer sequence than other extensions and are therefore
placed into a separate group.

PC Programming Manual References

23.2 UM Configuration—[4-2] Service Settings—Parameters—Automated Attendant—Alternate Extension

3.2.1.2 Auto Forwarding

Description
Moves or copies unplayed messages from one mailbox to another after a specified period of time has
passed. A message can be forwarded up to 9 times, and forwarding stops at the 9th designated mailbox.
Note that you cannot auto forward messages to the Mailbox Group, and messages marked as "private"
cannot be forwarded. Also, a message is never forwarded to the original sender of the message.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—External MSG Delivery/Auto FWD/
Personal Custom Serv

3.2.1.3 Automated Attendant (AA)

Description
Allows you to direct incoming calls without the use of an operator. Callers can be redirected to the desired
extension in one of the following ways:
a. Dial extension numbers directly
b. Spell the name of the desired party using the dial keys on their telephones (→ 3.2.1.16 Dialling by
Name)

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3.2.1 System Features

c. Listen to all subscriber names and select the desired extension (→ 3.2.1.26 List All Names)
Automated Attendant (AA) answers incoming calls and redirects them to the desired extension based on
numbers dialled by callers.
When calls from extensions are transferred to other subscribers, "Transferring you to (name)." can be heard
by callers before the calls are transferred. This feature is not available when the name of called party is not
recorded.
The service can be programmed for day, night, lunch, and break time modes, and is available for both Port
and Trunk Services.
Callers will reach Automated Attendant service when:
a. The Incoming Call Service of a trunk or port is set to "Automated Attendant Service" (→ 22.1 UM
Configuration—[3-1] UM Extension / Trunk Service—Service Group—Day, Night, Lunch, and Break
Mode - Incoming Call Service).
b. The call service of a Holiday is set to "Automated Attendant Service"
→ 10.5 PBX Configuration—[2-5] System—Holiday Table—UM Parameter—Service
c. They press [#8] (Automated Attendant Service Access Command) during a call.
d. A Custom Service or Personal Custom Service option is set to transfer callers to Automated Attendant
service.
→ 20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—External MSG Delivery/Auto FWD/
Personal Custom Serv
→ 23.3 UM Configuration—[4-3] Service Settings—Custom Service
e. A subscriber transfers the caller to Automated Attendant service.

PC Programming Manual References

23.2 UM Configuration—[4-2] Service Settings—Parameters

3.2.1.4 Automatic Two-way Recording for Manager

Description
Calls to and from specified extensions can be automatically recorded to a mailbox. Extensions are assigned
to a supervisor, who can listen to the recorded messages through Web Maintenance Console. For each
targeted extension, the following types of calls can be recorded:
• Intercom (internal) calls
• Trunk (external) calls
• Incoming ICD group calls only (i.e., when recording trunk calls, limit recording to incoming ICD group calls
only)
Recorded conversations are treated as new messages in the dedicated mailbox.
The period within which calls are recorded can be set through system programming.
A manager can receive an e-mail notification when Automatic Two-way Recording is used. For details, see
"5.4.4 E-mail Notification for Manager".

Note
• You should inform the other party that the conversation will be recorded.
• Irrespective of the Default Password for New Mailboxes setting, the Automatic Two-way Recording
mailbox does not have a default password. It is recommended to set a password.

Conditions
• KX-NSXU004 (Activation Key for Two-way Recording Control) is required to use this feature.
• For the user level to manage this feature, refer to 2.29.1.1 User Container Configuration.

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• Mailboxes specified as the recording destination become dedicated Automatic Two-way Recording
mailboxes. It is not possible to record messages to these mailboxes through other means, and
subscribers cannot log in to them.
• Forwarded calls and calls retrieved from being on hold will also be automatically recorded. However,
conference calls will not be recorded.
• Recording will automatically stop when the mailbox reaches its capacity. Delete older messages in order
to use this feature again.
• If the enabled extension has a Two-way Record button assigned to it, the button will flash during
recording. However, that button cannot be used to cancel this feature.
• An extension whose conversation is being recorded cannot be the target of the following features:
– Executive Busy Override (→ 2.10.2 Executive Busy Override)
– Whisper OHCA (→ 2.10.4.4 Whisper OHCA)
• Automatic recording will not be performed when both the target extension and the other party are ISDN
extensions.
• Conversations recorded with this feature are backed up via System Backup/Restore (→ 3.1.2.4 System
Backup/Restore).

PC Programming Manual References

7.10 Utility—Automatic Two-way Recording
8.6 Supervisors—Automatic Two-way Recording
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—UM—Two-way Recording

Feature Manual References

5.4.4 E-mail Notification for Manager

3.2.1.5 Broadcasting Messages

Description
Allows the System Manager to deliver the same message to the mailboxes of all subscribers simultaneously.
Broadcast Messages have priority over other regular or urgent messages during playback, but otherwise are
treated like regular messages. They are not treated as "Urgent" messages; if "Only Urgent Messages" is
selected as the Notification Type in the Notification Schedule, the Message Waiting Notification is not
activated when a Broadcast Message is received.

Conditions
• This feature is only available for the System Manager.

Operating Manual References

Manager Operation
2.2.1 System Manager Features—Broadcasting Messages

3.2.1.6 Call Services

Description
Include a series of both incoming and outgoing call services.
Incoming Call Services: Automated Attendant service, Voice Mail service, Interview service, Custom
Service.
Outgoing Call Services: Message Waiting Notification and External Message Delivery.

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PC Programming Manual References

22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group
→ Day, Night, Lunch, and Break Mode - Incoming Call Service
→ Day, Night, Lunch, and Break Mode - Incoming Call Service Parameter
→ Day, Night, Lunch, and Break Mode - Incoming Call Service Prompt

3.2.1.7 Call Transfer to Outside

Description
Enables the Unified Messaging system to transfer a call to a trunk from the following services:
• Custom Service
• Call Transfer Service
• Personal Custom Service
• Caller ID Callback
• Call-through Service

Conditions
• This feature cannot be used when COS programming does not allow incoming trunk calls to be
transferred to a trunk.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Call Transfer—Call Transfer to Outside
24.4 UM Configuration—[5-4] System Parameters—Parameters—Transfer to Outside

Operating Manual References

Subscriber Operation
1.8.5 Transferring Calls—Assigning and Cancelling Telephone Numbers for Call Transfer to Outside Line

3.2.1.8 Caller ID Call Routing

Description
Allows the System Administrator to store a maximum of 200 telephone numbers and assign a specific
destination (extension, mailbox, Mailbox Group, or Custom Service) to each telephone number for each time
mode (day, night, lunch, and break). When Caller ID information is received that matches one of the stored
telephone numbers, the call is automatically directed to its destination. Calls that are "Private" (when the
caller’s number is not received), "Out of Area" (when the caller is calling from an area that does not support
Caller ID), and "Long Distance" (when the caller made a long distance call) can also be directed to a specific
extension, mailbox or Custom Service.

Conditions
• The Company Greetings will not be played for callers when calls are routed by this feature.

PC Programming Manual References

23.1 UM Configuration—[4-1] Service Settings—Caller ID / PIN Call Routing

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3.2.1.9 Caller ID Screening

Description
Allows the Unified Messaging system to announce the name of the caller when transferring a call to an
extension from a preprogrammed caller (e.g., "You have a call from [name of caller]."). Caller names must
be recorded beforehand. Caller names can be recorded by each subscriber (Caller Name Announcement—
Personal) and for the entire system (Caller Name Announcement—System).

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Call Transfer—Caller ID Screen

Feature Manual References

3.2.1.10 Caller Name Announcement

Operating Manual References

Manager Operation
2.2.1 System Manager Features—Setting Class of Service (COS) Parameters

3.2.1.10 Caller Name Announcement

Description
Allows you to store telephone numbers and record a caller name for each telephone number. The caller
name is announced when playing a message in their mailbox from one of the preprogrammed callers, when
the system directs a call to the subscriber from one of the preprogrammed callers (Caller ID Screening), and
when the system pages the subscriber by intercom (Intercom Paging).
There are 2 types of Caller Name Announcement.
• Caller Name Announcement—Personal
Allows subscribers to store a maximum of 30 telephone numbers using a telephone.
• Caller Name Announcement—System
Allows the System Administrator to store a maximum of 200 telephone numbers using a PC.

Conditions
• If the same telephone number is programmed for both the system and personal caller name
announcement, the personal caller name will be announced.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Mailbox—Number of CIDs for Caller Name Announcement
(Selection)
24.3 UM Configuration—[5-3] System Parameters—System Caller Name Announcement

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Personal Caller Name Announcement
Manager Operation
2.2.1 System Manager Features—Setting Class of Service (COS) Parameters
2.2.2 Message Manager Features—Recording Messages—To record system caller names

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3.2.1.11 Class of Service (COS)

Description
Each mailbox is assigned a Class of Service (COS) that determines the set of services that are available to
its subscriber. 514 COS classes are supported by the system.
Mailboxes can be assigned to their own or to the same COS as needed. COS No. 513 and 514 are
assigned by default to the Message Manager and to the System Manager, respectively. No other mailboxes
can be assigned to COS No. 513 and 514.

Conditions
• The System Administrator (using a PC) and the System Manager (using a telephone) can change user's
COS parameters assignments.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Mailbox Parameters
21 UM Configuration—[2] Class of Service

3.2.1.12 Company Greeting

Description
Is a prerecorded message designed to greet all incoming callers and provide relevant information. A
maximum of 32 Company Greetings can be recorded, and a Company Greeting can be assigned for each
time mode (day, night, lunch, and break) and holiday for each Service Group. The start time of morning
greeting, afternoon greeting, and evening greeting can be set.

Conditions
• The System Manager can change the Company Greeting setting remotely by simply calling the Unified
Messaging system.
• The System Administrator can assign specific greetings for holidays.

PC Programming Manual References

22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group—Day, Night, Lunch, and Break
Mode - Company Greeting No. (Selection)
10.5 PBX Configuration—[2-5] System—Holiday Table—UM Parameter—Company Greeting No.
24.4 UM Configuration—[5-4] System Parameters—Parameters—Daily Hours Setting

Operating Manual References

Subscriber Operation
1.8.4 Sending Messages—Receiving External Delivery Messages
Manager Operation
2.2.1 System Manager Features—Changing the Company Greeting and Incoming Call Service Setting
2.2.2 Message Manager Features—Recording Messages

3.2.1.13 Company Name

Description
Is used by External Message Delivery Service when the intended receiver enters the password incorrectly 3
times. The Unified Messaging system announces the Company Name so that the receiver realises what
company placed the call to him or her.

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Operating Manual References

Manager Operation
2.2.2 Message Manager Features—Recording Messages

3.2.1.14 Covering Extension

Description
Forwards calls to a second extension when the first extension’s subscriber is not available to take the call.
The caller can also access the Covering Extension by pressing [0] while a Personal Greeting is being
played, or while leaving a message.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Covering Extension

Operating Manual References

Subscriber Operation
1.8.5 Transferring Calls—Assigning Your Covering Extension

3.2.1.15 Custom Service

Description
The following Custom Service types are available:
• Menu & Transfer
Allow callers to perform specific functions by pressing dial buttons on their telephones while listening to
voice guidance (Custom Service Messages). Callers can be guided to an extension, mailbox, outside
destinations (including mobile phones), operator, fax machine, other Custom Services etc., without the
assistance of an operator.
• Date Control
Allow a different operation to be assigned for up to 5 time periods defined by date. The caller makes no
selection and no menu is announced.
• Time Control
Allow a different operation to be assigned for up to 5 time periods defined by time of day. The caller
makes no selection and no menu is announced.
• Day Control
Allow a different operation to be assigned for up to 7 days defined by day of the week. The caller makes
no selection and no menu is announced.
• Password
Require that callers enter a password. Each password is assigned an operation. If a password is entered
correctly, the caller is handled by the password’s preprogrammed operation.
Custom Service Messages ("Press 1 for Sales, press 2 for Service…", etc.) can be recorded by the System
Administrator or the Message Manager, and can be recorded in multiple languages if needed. In total, a
maximum of 200 Custom Services can be created.
Calls can be handled by Custom Service by:
• Setting the Incoming Call Service of a trunk or port to "Custom Service Menu" and specifying a Custom
Service number.
→ 22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group—Day, Night, Lunch, and
Break Mode - Incoming Call Service
• Setting the call service of a Holiday to "Custom Service Menu" and specifying a Custom Service number.

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3.2.1 System Features

→ 10.5 PBX Configuration—[2-5] System—Holiday Table—UM Parameter—Service

• Using Custom Service or Personal Custom Service. A Custom Service can be assigned to one of the
available options provided by another Custom Service or Personal Custom Service. After pressing the
appropriate dial key, the caller is sent to assign Custom Service.
Custom Services can be created and edited using the Custom Service Builder utility of Web Maintenance
Console.

PC Programming Manual References

7.5.4 Utility—Report—UM View Reports
23.3 UM Configuration—[4-3] Service Settings—Custom Service

Operating Manual References

Manager Operation
2.2.2 Message Manager Features—Recording Messages

3.2.1.16 Dialling by Name

Description
Allows the caller to be connected to the desired subscriber’s mailbox or extension by searching for the
subscriber by name. Using the dialling keys, the caller can enter the first few letters of the subscriber’s first
and/or last name. The Unified Messaging system searches for possible matches and offers to connect the
caller with the subscriber. Subscriber names are included or excluded from the Directory Listing according to
their Class of Service (COS).

Conditions
• The System Administrator can change the Name Entry Mode (first, last, or full name entry) in Service
Settings.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—First Name
20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Last Name
21.1 UM Configuration—[2] Class of Service—Directory Listing

3.2.1.17 Emergency Greeting

Description
An emergency greeting can be recorded as one of company greetings. The emergency greeting can be
used in times such as when the company must be closed due to bad weather.

Operating Manual References

Manager Operation
2.2.2 Message Manager Features—Recording an Emergency Greeting

3.2.1.18 Extension Group

Description
An Extension Group is a group of extensions that share a common mailbox. Extension Groups are created
by the System Administrator. Each group has an Extension Group number. There are 20 group lists
available, and each group list can have a maximum of 100 members.

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PC Programming Manual References

24.2 UM Configuration—[5-2] System Parameters—Extension Group

3.2.1.19 Fax Service

Description
Allows faxes to be received and stored in a specified mailbox.
Calls reach Fax service when:
a. The Incoming Call Service of a trunk group or port is set to "Fax Service".
→ 22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group—Day, Night, Lunch,
and Break Mode - Incoming Call Service
b. The Caller ID Call Transfer setting of Caller ID/PIN routing is set to "Fax Service".
→ 23.1 UM Configuration—[4-1] Service Settings—Caller ID / PIN Call Routing
c. The Incoming Call Service of a service group assigned to a DIL/DDI/DID/MSN entry is set to "Fax
Service".
→ 18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings—DIL—UM Service
Group No.
→ 18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—UM Service Group No.
→ 18.4 PBX Configuration—[10-4] CO & Incoming Call—MSN Table—MSN—UM Service Group No.
d. The service type of a Custom Service is set to "Fax Service".
→ 23.3 UM Configuration—[4-3] Service Settings—Custom Service

Conditions
• An interview mailbox cannot be specified as the mailbox for storing faxes.

PC Programming Manual References

22 UM Configuration—[3] UM Extension / Trunk Service

3.2.1.20 Hold
Description
Provides the caller with the option of temporarily going on hold when the called extension is busy. The
Unified Messaging system automatically recalls the extension after a specified period of time. When several
callers are holding for the same extension, callers are connected to the extension in the order in which they
originally called.

PC Programming Manual References

23.2 UM Configuration—[4-2] Service Settings—Parameters—Automated Attendant—Call Hold Mode

3.2.1.21 Holiday Service

Description
Allows the system to override the normal settings assigned for the appropriate trunk service or port service
by playing a special holiday greeting. (e.g., "Happy New Year! Our office is closed today. If you wish to
record a message, please press 1 now".) A maximum of 24 holidays can be programmed. A holiday can be
a specific day or range of days.

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Conditions
• Holidays cannot overlap with each other.
• The holidays stored in the Unified Messaging holiday table are managed separately from those in the
holiday table used for Time Service (→ 5.1.4 Time Service). However, holidays specified in the Time
Service holiday table can be copied to the Unified Messaging holiday table.

PC Programming Manual References

10.5 PBX Configuration—[2-5] System—Holiday Table—UM Parameter

3.2.1.22 Hospitality Mode

Description
Allows a subscriber to access certain subscriber services in a dedicated, 'hospitality' mode. Subscribers in
hospitality mode can listen to messages, change the password, change personal greetings, and change the
owner’s name, if enabled to do so through system programming. Also, it is possible to set whether the
password, personal greeting and owner name settings are deleted upon check-out.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Hospitality Mode

Feature Manual References

2.23 Hospitality Features

3.2.1.23 Intercept Routing to a Mailbox

Description
If the Intercept Routing destination is set to the floating extension number of the UM group, and the
"Intercept to Mailbox for Call to Extension" setting is enabled through system programming, intercepted
trunk calls will be redirected to the extension’s mailbox. Therefore the caller can leave a message in the
mailbox of the extension without knowing the mailbox number.

PC Programming Manual References

11.7.1 PBX Configuration—[3-7-1] Group—UM Group—System Settings—Intercept to Mailbox for Call to
Extension

Feature Manual References

2.1.1.6 Intercept Routing

3.2.1.24 Intercom Paging

Description
Allows callers to page subscribers when the subscribers have set Intercom Paging for Call Transfer or when
the subscribers have set Intercom Paging for Incomplete Call Handling. The caller is briefly placed on hold
while the system announces the page and until the subscriber answers the page. The subscriber can
answer the page from any extension using the paging answer feature number. If Caller ID information is
received and the caller’s name has been recorded for the Caller Name Announcement feature, the name
will be announced at the end of the page.
To utilise this feature, the following settings are necessary:
1. Assign an Intercom Paging Group to the desired Class of Service.

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→ 21.1 UM Configuration—[2] Class of Service—Call Transfer—Intercom Paging Group

2. Set "No Answer Time for Intercom Paging" to the desired setting.
→ 24.4 UM Configuration—[5-4] System Parameters—Parameters—Intercom Paging Parameters—No
Answer Time for Intercom Paging (1-30 s)

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Call Transfer—Intercom Paging Group
24.4 UM Configuration—[5-4] System Parameters—Parameters—Intercom Paging Parameters

Feature Manual References

2.17.1 Paging

3.2.1.25 Interview Service

Description
Allows the Unified Messaging system to "interview" a caller by playing a series of prerecorded questions and
recording the caller’s responses.
As the callers answer questions, they are recorded into an interview mailbox.
• After a caller records the answers, the interview mailbox lights a message waiting lamp on the interview
mailbox owner’s extension.
• When retrieving the messages, the subscriber only hears the answers to the questions.
In order to use Interview service, at least one interview mailbox must be created in the system.
→ 20.1 UM Configuration—[1] Mailbox Settings
Callers will reach an interview mailbox when:
a. The Incoming Call Service of a trunk or port is set to "Interview Mailbox".
→ 22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group—Day, Night, Lunch,
and Break Mode - Incoming Call Service
b. The call service of a Holiday is set to "Interview Mailbox".
→ 10.5 PBX Configuration—[2-5] System—Holiday Table—UM Parameter—Service
c. A Custom Service or Personal Custom Service option is set to transfer callers to an interview mailbox.
→ 23.3 UM Configuration—[4-3] Service Settings—Custom Service
→ 20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—External MSG Delivery/Auto FWD/
Personal Custom Serv—Personal Custom Service
d. A subscriber has configured his or her mailbox so that unanswered calls are directed to the mailbox’s
interview mailbox via Incomplete Call Handling, and has set Message Reception Mode to "Interview
Mode".
→ 1.8.6 Other Features—Interview Mailbox—Message Reception Mode
e. A subscriber with a logical extension has set Message Reception Mode to "Interview Mode".
→ 1.8.6 Other Features—Interview Mailbox—Message Reception Mode
f. A subscriber transfers the caller to an interview mailbox.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Interview Mailbox

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Interview Mailbox

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Manager Operation
2.2.1 System Manager Features—Setting Up Mailboxes

3.2.1.26 List All Names

Description
Allows callers to listen to a list of all subscribers’ names and extension numbers in Automated Attendant
service or Custom Service.

Conditions
• This feature is only available when:
a. The subscriber’s name has been recorded.
b. The subscriber’s extension number is set.
c. The "Directory Listing" parameter for the subscriber’s Class of Service is set to "Yes".
d. The "List All Names" is set to "Enable". (→ 23.2 UM Configuration—[4-2] Service Settings—
Parameters—Automated Attendant—List All Names)
• To return to the previous menu in Automated Attendant service or Custom Service, press .

Feature Manual References

3.2.1.3 Automated Attendant (AA)
3.2.1.15 Custom Service

3.2.1.27 Logical Extension (All Calls Transfer to Mailbox)

Description
Is an extension that always receives calls directly into its mailbox. This feature is used by subscribers that
are often unavailable or that do not have a telephone.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—All Calls Transfer to Mailbox

Operating Manual References

Manager Operation
2.2.1 System Manager Features—Setting Up Mailboxes

3.2.1.28 Message Reception Mode

Description
Determines whether incoming calls are directed to the subscriber’s regular mailbox or interview mailbox.
This mode is effective for Incomplete Call Handling Service (when the subscriber’s line is busy, when the
subscriber cannot take the call, or when the subscriber has enabled Call Blocking), and when the transfer
destination of calls is set to a Logical Extension.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—All Calls Transfer to Mailbox

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Feature Manual References

3.2.1.25 Interview Service
3.2.1.27 Logical Extension (All Calls Transfer to Mailbox)

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Interview Mailbox—Message Reception Mode

3.2.1.29 Message Waiting Notification—E-mail Device

Description
Enables subscribers (including the message manager) to be notified by e-mail when they have new
messages (both voice and fax). The notification will contain the message sender’s information, the length of
the message, the number of messages (new/old), and a callback number (if programmed). Subscribers can
choose to have the voice message and/or fax data attached to the notification and can also choose to have
the message deleted after it has been sent. Up to 3 devices can be configured for receiving notifications.

Note
To receive notifications about missed calls, extension users should specify e-mail addresses in their user
settings. (→ 5.4.1 E-mail Notification for Extension Users)

Conditions
• KX-NSUN001, KX-NSUN010, KX-NSUN050, KX-NSUN100, KX-NSUN500, KX-NSUM001,
KX-NSUM010, KX-NSUM050, KX-NSUM100, KX-NSUM500, KX-NSUA001, KX-NSUA010,
KX-NSUA050, KX-NSUA100 or KX-NSUA500 (Activation Key for User Container) is required for this
feature.
• The System Administrator can enable or disable this feature for each mailbox, and can customise the
time frame during which notifications are sent. For example, if the System Administrator sets the time
frame for Monday to Friday between 9 AM and 5 PM, notifications will be sent only during those hours.
• When sending long voice message notifications, note the following, and confirm the settings.
– If the voice message is longer than the time specified for "Maximum Message Length", the surplus
parts of the message may be discarded when sending the notification.
→ 24.4 UM Configuration—[5-4] System Parameters—Parameters—E-mail Option—Maximum
Message Length (Selection)
– Depending on the settings of the sender and recipient, voice messages may not be sent or received
properly.
– When the Unified Messaging system is programmed to delete messages after they are sent, the
message will be deleted even if there is an error and the notification cannot be sent.
• If there are any errors when the system tries to send notifications, error messages will be sent to the
preset address.
→ 24.4 UM Configuration—[5-4] System Parameters—Parameters—E-mail Option—Mail Address (Up to
128 ASCII characters)

PC Programming Manual References

7.5.4 Utility—Report—UM View Reports
20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Notification Parameters
→ E-mail/Text Message Device—Device No. 1, 2, 3—User Name
→ E-mail/Text Message Device—Device No. 1, 2, 3—E-mail Address
→ E-mail/Text Message Device—Device No. 1, 2, 3—Notification Type

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→ E-mail/Text Message Device—Device No. 1, 2, 3—Only Urgent Messages

→ E-mail/Text Message Device—Device No. 1, 2, 3—Title Order
→ E-mail/Text Message Device—Device No. 1, 2, 3—Title String
→ E-mail/Text Message Device—Device No. 1, 2, 3—Callback Number
→ E-mail/Text Message Device—Device No. 1, 2, 3—Send Wait Time [0-120 min]
→ E-mail/Text Message Device—Device No. 1, 2, 3—Attach Voice File
→ E-mail/Text Message Device—Device No. 1, 2, 3—Attach Fax File
→ E-mail/Text Message Device—Device No. 1, 2, 3—Use Mode
21.1 UM Configuration—[2] Class of Service—General
→ E-mail Option
→ Fax Option
24.4 UM Configuration—[5-4] System Parameters—Parameters—E-mail Option

Feature Manual References

3.1.3 Fax Server
5.4.1 E-mail Notification for Extension Users

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Message Waiting Notification
→ Device Notification:
→ To turn Device Notification On/Off

3.2.1.30 Message Waiting Notification—Lamp

Description
Automatically lights the message waiting lamp on the subscriber’s telephone when subscribers have new
messages. When a proprietary telephone with display is used, the number of unplayed messages will be
displayed when the message waiting lamp is lit. When the Message button light turns on, pressing the
button allows the subscriber to play back the messages stored in his mailbox without dialling a mailbox
number.

Conditions
• In order to display the number of unplayed messages on the display:
– A compatible Panasonic Proprietary Telephone with 6-line display must be used.
– The subscriber must have his or her own extension.
– The Message Manager’s extension number must be assigned as Operator 1 in Day Mode.
• Extensions assigned as operators can be called by dialling [0], however, when setting this feature the
extension number (not "0") must be specified.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Notification Parameters—Message
Waiting Lamp

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Message Waiting Notification
Manager Operation
2.2.2 Message Manager Features—Setting Up Message Waiting Notification

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3.2.1.31 Message Waiting Notification—Telephone Device

Description
Automatically calls a preprogrammed telephone number when a subscriber has new messages (both voice
and fax). New messages will be automatically played back when the subscriber answers the call.
Notifications can be scheduled. The System Administrator can program 2 different time frames (for example,
9 AM to 12 PM and 8 PM to 10 PM) independently for each day of the week. Message waiting notifications
will be sent on the selected days during the programmed time frames. Notifications can also be sent
"continuously", meaning a notification will be sent whenever a new message is received, 24 hours a day.

Conditions
• Notifications can be sent for either all messages or for urgent messages only, depending on whether the
"Only Urgent Messages" setting is enabled. When it is enabled, notifications are sent (according to the
notification schedule, if programmed) only when urgent messages have been left in the subscriber’s
mailbox.
• A maximum of 3 devices can be programmed by the System Administrator or subscriber for use with this
feature. When the Unified Messaging system calls a device and the call is not answered, the system will
try to call the device again. The number of retries and the delay time between retries can be programmed
by the System Administrator. The lowest numbered device is called first. If the first device cannot be
called, the second (then third) device will be called.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Notification Parameters
→ Telephone Device
→ Telephone Device—Device No. 1, 2, 3—Notification Type
→ Telephone Device—Device No. 1, 2, 3—Only Urgent Messages
→ Telephone Device—Device No. 1, 2, 3—Dial Number [0-9 * # T X , ;]
→ Telephone Device—Device No. 1, 2, 3—No. of Retries
→ Telephone Device—Device No. 1, 2, 3—Busy Delay Time (min)
→ Telephone Device—Device No. 1, 2, 3—No Answer Delay Time (min)
→ Telephone Device—Device No. 1, 2, 3—Use Mode
→ Telephone Device—Device Notification Timer—Device Start Delay Time (0-120 min)
→ Telephone Device—Device Notification Timer—Device Interval Time between Device 1, 2, 3 and
Next Device
24.4 UM Configuration—[5-4] System Parameters—Parameters—Dialling Parameters/MSW Notification

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Message Waiting Notification
Manager Operation
2.2.2 Message Manager Features—Setting Up Message Waiting Notification

3.2.1.32 Multilingual Service

Description
Allows a maximum of 8 languages to be used for system prompts. The System Administrator can select 5
languages for the multilingual selection menu, and assign a key for each language in the menu. Callers can
select the desired language by pressing the assigned key. The language used for system prompts heard by
subscribers is determined by Class of Service (COS).

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PC Programming Manual References

24.4 UM Configuration—[5-4] System Parameters—Parameters—Prompt Setting

Operating Manual References

Manager Operation
2.2.2 Message Manager Features—Recording Messages

3.2.1.33 No DTMF Input Operation

Description
Provides guidance to callers when several seconds pass without anything being entered by the caller.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—External MSG Delivery/Auto FWD/
Personal Custom Serv—No DTMF Input Operation

3.2.1.34 On Hold Announcement Menu

Description
Allows callers (other than the first caller) in a queue to listen to prerecorded announcements or music. The
On Hold Announcement Menu can be recorded by the System Administrator with a telephone while using
Web Maintenance Console or by importing WAV files, and also by the Message Manager using a telephone.
When the On Hold Announcement Menu is recorded, this feature is automatically enabled.

PC Programming Manual References

7.9 Utility—UM – System Prompts Customisation

Operating Manual References

Manager Operation
2.2.2 Message Manager Features—Recording Messages

3.2.1.35 Operator Service

Description
Allows callers to be connected to a live operator. The operator can then transfer the caller to the appropriate
party or service as needed. A maximum of 3 operators can be specified to receive calls in the day, night,
lunch, and break modes, and each operator can be assigned a mailbox for message-taking.

Note
• Operator 1 in day mode is automatically designated as the Message Manager and is assigned the
extension number "0" or "9". This operator’s mailbox is the General Delivery Mailbox.
• Operators 2 and 3 can be assigned to a floating number that is assigned to a ring group (Incoming
Call Distribution Group).
Calls can be directed to an operator when:
a. A caller does not or cannot send any DTMF signals to the Unified Messaging system (i.e., the caller
does not dial any digits).
b. A caller dials "0" when the call is being handled by Automated Attendant service.

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c. A Custom Service or Personal Custom Service option is set to transfer callers to Automated Attendant
service. After pressing the appropriate dial key, the caller is directed to an operator.
→ 23.3 UM Configuration—[4-3] Service Settings—Custom Service
→ 20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—External MSG Delivery/Auto FWD/
Personal Custom Serv—Personal Custom Service
Operator Service can be structured as a cascade so that if Operator 1 cannot take the call, it goes to
Operator 2. If that fails, the call goes to Operator 3. If that fails, the caller can record a message. At each
stage, there are other options for busy cases and no-answer cases.
• Busy Coverage Mode
Determines how calls to an operator will be handled when the line is busy. The Busy Coverage options
are: Hold, No Answer Coverage, Call Waiting, and Disconnect Message.
• No Answer Coverage Mode
Determines how calls will be handled when an operator does not answer within the time specified for
"Operator No Answer Time". The No Answer Coverage options are: Caller Select, Leave Message,
Disconnect Message, and Next Operator.

PC Programming Manual References

23.2 UM Configuration—[4-2] Service Settings—Parameters—Automated Attendant
→ Operator Service—Operator’s Extension
→ Operator Service—Busy Coverage Mode
→ Operator Service—No Answer Coverage Mode

Operating Manual References

Manager Operation
2.2.2 Message Manager Features—Customising the Message Manager’s Mailbox

3.2.1.36 PIN Call Routing

Description
Allows the System Administrator to store a maximum of 200 PIN (Personal Identification Number) (max. 20
digits) numbers and assign a destination (extension, mailbox, mailbox group, or Custom Service) to each
PIN for each time mode (day, night, lunch, and break). In a Custom Service, callers can be required to enter
a PIN number in order to be directed to a destination.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Mailbox Parameters—Call Transfer
Anytime in Incomplete Handling Menu
23.1 UM Configuration—[4-1] Service Settings—Caller ID / PIN Call Routing

3.2.1.37 Play System Prompt After Personal Greeting

Description
Allows the Guidance for Recording message to be played for the caller after the Personal Greeting. The
Guidance for Recording message instructs the caller how to terminate the call, access more features, and
rerecord the message.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Mailbox—Play System Prompt after Personal Greeting

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3.2.1 System Features

Operating Manual References

Manager Operation
2.2.1 System Manager Features—Setting Class of Service (COS) Parameters

3.2.1.38 Port Service

Description
Allows assignment of call services to each port. The incoming call service determines which service is used
when answering incoming calls. These services include: Voice Mail, Automated Attendant, Interview,
Custom Service, Fax, and Transfer to Mailbox. Custom Service is the most flexible of all the services
because it allows access to the other services by pressing one key.

Note
For caller convenience, we recommend programming all ports to use Custom Service as the Incoming
Call Service (→ 3.2.1.15 Custom Service).

PC Programming Manual References

22 UM Configuration—[3] UM Extension / Trunk Service

Feature Manual References

3.2.1.3 Automated Attendant (AA)
3.2.1.15 Custom Service
3.2.1.19 Fax Service
3.2.1.25 Interview Service
3.2.1.44 Transfer to Mailbox
3.2.1.46 Voice Mail Service

3.2.1.39 Remote Time Service Set

Description
Allows the System Manager to programme his or her extension from a remote location in order to change
the PBX’s time service mode (day/night/lunch/break).

Operating Manual References

Manager Operation
2.2.1 System Manager Features—Remote Time Service Mode Setting

3.2.1.40 Service Group

Description
Is a group of parameters that determine how incoming calls will be handled. 384 per Tenant group Service
Groups can be configured; one Service Group can be assigned to each port and each trunk group.

PC Programming Manual References

10.4 PBX Configuration—[2-4] System—Week Table
22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group

Feature Manual References

5.1.4 Time Service

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Operating Manual References

Manager Operation
2.1.2 Time Service Mode Control

3.2.1.41 Simplified Tutorial

Description
If enabled in a subscriber’s COS settings, the tutorial that guides the subscriber through setting up his or her
mailbox is simpler and takes less time to complete than the standard tutorial.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Tutorial

3.2.1.42 System Prompts

Description
Are announcements that instruct a caller. A maximum of 8 languages can be stored as system prompts. The
name of each language can be stored as well. When the System Administrator selects a language for the
system settings, the desired language can be selected using this name label data. "Primary" can be
specified by selecting one of the stored languages, and become the default setting unless another language
is selected in the system setting.

Note
• System prompts can be changed or turned on/off, system prompts for each language can be imported
as WAV files, or re-recorded using a telephone.
• In order to leave more time for recording, the System Administrator is also able to delete specific
system prompts or one of the installed languages used for system prompts.

PC Programming Manual References

7.9 Utility—UM – System Prompts Customisation
21.1 UM Configuration—[2] Class of Service—Prompt Mode
24.4 UM Configuration—[5-4] System Parameters—Parameters—Prompt Setting

Operating Manual References

Manager Operation
2.2.1 System Manager Features—Setting Class of Service (COS) Parameters
2.2.2 Message Manager Features—Recording Messages

3.2.1.43 Transfer Recall to a Mailbox

Description
If a call is transferred to an extension via the Automated Attendant (AA) service and the call is not answered
within a preprogrammed Transfer Recall time, the call is redirected to the mailbox of the transfer destination
extension. Therefore the caller can leave a message in the mailbox of the extension without knowing the
mailbox number. The "Transfer Recall to Mailbox" setting must be enabled through system programming
to use this feature.

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PC Programming Manual References

10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Recall—Transfer
Recall
11.7.1 PBX Configuration—[3-7-1] Group—UM Group—System Settings—Transfer Recall to Mailbox

Feature Manual References

3.2.1.3 Automated Attendant (AA)

3.2.1.44 Transfer to Mailbox

Description
Forwards the call to a specified mailbox.
Calls reach the Transfer to Mailbox Service when the Incoming Call Service of a trunk group or port is set to
"Transfer to Mailbox".

PC Programming Manual References

22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group—Day, Night, Lunch, and Break
Mode - Incoming Call Service

3.2.1.45 Trunk Service (Universal Port)

Description
Allows call handling features to behave differently depending on the service group that the calls are received
on.
A service can be assigned to each service group, including: Voice Mail, Automated Attendant, Interview,
Custom Service, Fax, and Transfer to Mailbox. Custom Service is the most flexible of all the services
because it allows access to the other services by pressing one key.

Note
For caller convenience, we recommend programming all trunk groups to use Custom Service as the
Incoming Call Service (→ 3.2.1.15 Custom Service).

PC Programming Manual References

22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group

Feature Manual References

3.2.1.3 Automated Attendant (AA)
3.2.1.15 Custom Service
3.2.1.19 Fax Service
3.2.1.25 Interview Service
3.2.1.44 Transfer to Mailbox
3.2.1.46 Voice Mail Service

3.2.1.46 Voice Mail Service

Description
Is a message recording service that allows callers to leave messages for subscribers. They can dial mailbox
numbers directly or "spell" the name of the desired party using the dial keys on their telephones
(→ 3.2.1.16 Dialling by Name). Once the caller has dialled an appropriate number, the Unified Messaging
system transfers the caller to the party.

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Callers will reach Voice Mail service when:

a. The Incoming Call Service of a trunk group or port is set to "Voice Mail Service".
→ 22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group—Day, Night, Lunch,
and Break Mode - Incoming Call Service
b. The call service of a Holiday is set to "Voice Mail Service".
→ 10.5 PBX Configuration—[2-5] System—Holiday Table—UM Parameter—Service
c. They press [#6] (Voice Mail Service Access Command) during a call.
d. A Custom Service or Personal Custom Service option is set to transfer callers to Voice Mail service.
→ 20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—External MSG Delivery/Auto FWD/
Personal Custom Serv—Personal Custom Service
→ 23.3 UM Configuration—[4-3] Service Settings—Custom Service
e. A subscriber transfers the caller to Voice Mail service.

PC Programming Manual References

22 UM Configuration—[3] UM Extension / Trunk Service

3.2.2 Subscriber Features

Description
PBX users who are assigned a mailbox in the Unified Messaging system are called subscribers.
Subscribers can customise their mailboxes in the following ways:
• Set a password
• Record names
• Record Personal Greetings
• Set Personal Group Distribution Lists
• Set a Personal Custom Service
• Set a Call Transfer Status
• Set Covering Extensions
• Set a Message Reception Mode
• Set an Incomplete Call Handling Status
• Set a Message Waiting Notification
• Record questions for an interview mailbox

3.2.2.1 Auto Receipt

Description
Allows a subscriber or the System Manager to receive a message to confirm the reception of sent
messages. When the sent messages are received by a subscriber, the sender will receive a message
announcing "(mailbox name/number) has received your message". Subscribers need to request the Auto
Receipt when they send messages.

Conditions
• This feature is not available when the System Manager sends a Broadcasting Message.

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• The reception of External Delivery Messages can be confirmed using this feature as well. Subscribers
can request an Auto Receipt when sending a message to a single recipient or to all members of an
External Delivery Message List, and receive an Auto Receipt for each member in the list.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Mailbox—Auto Receipt

3.2.2.2 Automatic Fax Delivery

Description
Mailboxes can be set so that they automatically forward received faxes to a specified fax machine.
Status management
After the fax has been forwarded, the message is automatically marked as "old" by default. However, this
behaviour can be changed through system or personal programming so that the message remains marked
as "new" or is deleted.

Conditions
• If a fax message has an attached voice comment (i.e., the sender left a voice message before sending
the fax), the message remains marked as new, regardless of the setting for changing the message’s
status after forwarding.
• If the fax machine is busy, it will be retried a preprogrammed number of times. If the fax still cannot be
sent, a non-delivery notification message is sent to the subscriber’s mailbox.
• If an error occurs while sending the fax, whether to retry can be set through system programming. Also,
whether to send the fax again from the start or to send it from the page where the error occurred is also
programmable.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Fax Options
→ Automatic Delivery Status
→ Automatic Delivery Status Default Fax Number [0-9 * # T , ; ]
→ Automatic Delivery Status Printed Fax Message
24.4 UM Configuration—[5-4] System Parameters—Parameters—Fax Management
→ Fax Properties—Page Retransmission
→ Fax Properties—Page Retransmission—Retransmission Start Page

3.2.2.3 Automatic Login

Description
Allows subscribers and managers to log in to their mailbox directly without entering the mailbox number. A
subscriber/manager can log in to his or her mailbox directly by:
• dialling a Unified Messaging extension number directly from his or her extension.
• calling the Unified Messaging system from a telephone number that is assigned to log him or her into the
mailbox.
• dialling (from an outside telephone) the DID number assigned to log him or her into the mailbox.
• calling the Unified Messaging system so that the call is received on the trunk group assigned to log him or
her into the mailbox.
It is also possible for subscribers and managers to log in to their mailboxes without entering a password.

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Notice
When disabling the password requirement, ensure that an unauthorised third-party is not allowed access
to your extension.

Conditions
• Access from outside telephones will be enabled automatically after a Caller ID number, DID number, or
trunk group number is assigned. A trunk group number or DID number can be assigned by the System
Administrator only.
• When this feature is activated, "Toll Saver" is also available.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Remote Call/Automatic Login/Direct
Service
→ Auto Login Extension
→ Auto Login Extension Password Entry Requirement
→ Auto Login Caller ID 1, 2
→ Auto Login Caller ID Password Entry Requirement
→ Auto Login DDI/DID
→ Auto Login TRG No.
→ Auto Login DDI/DID, TRG No. Password Entry Requirement
→ Auto Login Toll Saver

Feature Manual References

3.2.2.33 Toll Saver

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Assigning Your Telephone Numbers for Remote Automatic Log-in and Toll Saver

3.2.2.4 Autoplay New Message

Description
It is possible to play new messages automatically when a subscriber or the System Manager/Message
Manager logs into his or her mailbox. There is no need to press [1] to receive the new messages.

Conditions
• If there is more than one new message in the mailbox, it can be set whether or not messages will be
played continuously without system prompts.
→ 21.1 UM Configuration—[2] Class of Service—Mailbox—Play New Messages Sequentially

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Mailbox—Autoplay New Message

3.2.2.5 Bookmark
Description
Enables a subscriber to set one bookmark per message while pausing a message. After setting a
bookmark, a subscriber can listen to the message from that bookmark by pressing the specified key while
pausing the message or after the message was played.

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Operating Manual References

Subscriber Operation
1.8.3 Message Playback and Related Features—Bookmarks (Voice Messages only)

3.2.2.6 Call-through Service

Description
Allows subscribers to make outside calls by accessing the subscriber service menu (from an outside
telephone) and dialling an outside destination. This creates a trunk-to-trunk call.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—General
24.4 UM Configuration—[5-4] System Parameters—Parameters—Transfer to Outside—Outside Transfer
Sequence—Call Transfer to Outside Sequence (Up to 16 digits / [0-9 * # D F R T , ; N])

Feature Manual References

2.16.1 Direct Inward System Access (DISA)

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Call-through Service

3.2.2.7 Call Transfer Scenario

Description
By combining the settings for Call Transfer and Incomplete Call Handling, you can create a call handling
"scenario" for your extension. Up to 20 scenarios can be created, and a scenario can be assigned to each
absent message.

PC Programming Manual References

29.1.10 My Portal—Voicemail—Advanced Call Transfer Setting
29.1.10 My Portal—Voicemail—Advanced Settings—Scenario Setting

Feature Manual References

2.20.2 Absent Message
3.2.2.8 Call Transfer Status
3.2.2.18 Incomplete Call Handling Service

3.2.2.8 Call Transfer Status

Description
Allows subscribers to specify how the Unified Messaging system will handle calls to their individual
extensions. Call Transfer Status options include: Call Blocking, Call Screening, Intercom Paging, Transfer to
Mailbox, Transfer to Specified Telephone Number, and Custom Service.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Mailbox Parameters

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Feature Manual References

3.2.1.24 Intercom Paging

3.2.2.9 Callback Number Entry

Description
Enables the caller to leave a callback number in several different ways depending upon which option is
programmed into the Unified Messaging system.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Notification Parameters
24.4 UM Configuration—[5-4] System Parameters—Parameters—Dialling Parameters/MSW Notification

Operating Manual References

Manager Operation
2.2.2 Message Manager Features—Setting Up Message Waiting Notification

3.2.2.10 Caller ID Callback

Description
Enables a subscriber to call back a message sender with the Caller ID number attached to the message.
Please consult the seller where you purchased this PBX to use this feature.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Mailbox—Caller ID Callback
24.4 UM Configuration—[5-4] System Parameters—Parameters—Transfer to Outside—Trunk Group (1–
128)—Caller ID Callback

Feature Manual References

3.2.1.7 Call Transfer to Outside

3.2.2.11 Delete Message Confirmation

Description
It is possible to request confirmation from the subscriber before erasing a message left in the mailbox. When
messages are deleted, they will be retained until the time specified in System Maintenance Start Time on
the following day.

PC Programming Manual References

7.11 Utility—UM - System Maintenance—System Maintenance Start Time
21.1 UM Configuration—[2] Class of Service—Mailbox—Delete Message Confirmation

3.2.2.12 Direct Service Access

Description
Allows a subscriber to access a feature directly by dialling a Unified Messaging extension number. The
following features can be accessed directly:
• Record No Answer Greeting

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• Record Busy Greeting

• Record After Hours Greeting
• Record Temporary Greeting
• Change Day Main Menu*1
• Change Night Main Menu*1
• Change Emergency Greeting*1
*1 Access to this feature is available only to the Message Manager and to subscribers with Message Manager privileges.

Conditions
• This feature is only available when the Automatic Login feature is enabled.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Remote Call/Automatic Login/Direct
Service

3.2.2.13 External Message Delivery Service

Description
Allows a subscriber to send a message to several subscribers and non-subscribers (including outside
parties) at a specified time (or immediately). This feature also allows the receiver to reply to the message
without having to specify the mailbox number.
Further, an External Message Delivery List allows a subscriber to send a message to multiple parties with a
single operation. One subscriber can maintain up to 2 lists with a maximum of 8 entries in each.
Also, it is possible to require the receiver to enter a 4-digit password to receive the sender’s message. If the
receiver enters the password incorrectly 3 times, the Unified Messaging system plays: the Company Name
(if it has been recorded), the Company’s Telephone Number (if registered), and the sender’s extension (if
both the Company’s Telephone Number and the Extension of the Owner have been registered). With this
information, the receiver can track down the message even if he or she does not remember the password.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Mailbox Parameters
20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—External MSG Delivery/Auto FWD/
Personal Custom Serv
→ External Message Delivery Active
→ External Message Delivery Prompt Mode
24.4 UM Configuration—[5-4] System Parameters—Parameters—External Message Delivery

Operating Manual References

Subscriber Operation
1.8.4 Sending Messages
→ External Message Delivery
→ External Message Delivery Lists
→ Receiving External Delivery Messages

3.2.2.14 Fax Cover Page

Description
If enabled through system programming, a cover page will be added to a fax before the fax is sent. A cover
page is added when a fax is sent with the following features:

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• Automatic Delivery (→ 3.2.2.2 Automatic Fax Delivery)

• Fax Resend*1 (→ 3.2.2.30 Send to Fax Machine)
• Sending a Fax from a Computer (→ 3.2.2.15 Fax Driver)
*1 A cover page is not added if the subscriber is calling from a fax machine and is sending the fax to that machine.

The cover page contains the following information:

• Date and time the fax was sent
• Subscriber’s name
→ 20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Mailbox Parameters—First Name/
Last Name
• System name
→ 24.4 UM Configuration—[5-4] System Parameters—Parameters—Fax Management—Fax Properties
—System Name (Up to 25 characters)
• System fax number
→ 24.4 UM Configuration—[5-4] System Parameters—Parameters—Fax Management—Fax Properties
—System Fax Number (Max. 20 digits)
• Number of pages in the fax (not including the cover sheet)

The language of the cover sheet can be specified through system programming.
→ 20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Mailbox Parameters—First Name/Last
Name

Conditions
• When sending a fax from a computer, the cover page’s contents can be customised. For details, refer to
the fax driver’s documentation.

PC Programming Manual References

24.4 UM Configuration—[5-4] System Parameters—Parameters—Fax Management—Fax Properties—
Cover Page

3.2.2.15 Fax Driver

Description
Subscribers who install the fax printer driver can send faxes from their computer using the standard "print"
function of an application.
A link to where users can download the fax printer driver is provided in Web Maintenance Console.
→"Unified Message"—"Fax Driver Tool" in 8.1.1 Users—User Container—Add User/Edit User

Conditions
• For details about the fax driver, refer to the driver’s documentation.
• The Fax Card is mounted to the Expansion Gateway.
• The FAX Unit can only be connected to the Expansion Gateway.
• TIF files of up to 999 pages can be sent.

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Sending a Fax from your PC

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3.2.2.16 Forwarding to a Mailbox

Description
A subscriber can set calls to be forwarded to the floating extension number of the UM group. In this case,
the call is forwarded directly to the extension’s mailbox. Therefore the caller can leave a message without
knowing the mailbox number.

Feature Manual References

2.3.2 Call Forwarding (FWD)

3.2.2.17 Group Distribution Lists

Description
There are 2 types of Group Distribution Lists:
• Group Distribution List—Personal
Allows a subscriber to simultaneously send a message to several mailboxes. These lists can be created
or edited by the System Administrator and by the subscriber. Each subscriber can maintain a maximum of
4 lists with up to 40 entries in each list.
• Group Distribution List—System
Also called "Mailbox Groups".
Allows subscribers and callers to send messages to all mailboxes that belong to the list at once. These
lists are created by the System Administrator. There are 20 group lists available, and each group list can
have a maximum of 200 members.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Personal Distribution List
24.1 UM Configuration—[5-1] System Parameters—Mailbox Group

Operating Manual References

Subscriber Operation
1.8.4 Sending Messages—Personal Group Distribution Lists
Manager Operation
2.2.1 System Manager Features—Broadcasting Messages

3.2.2.18 Incomplete Call Handling Service

Description
Allows the subscriber to offer callers several service options when the extension is busy or there is no
answer. Options available for Incomplete Call Handling for Busy or No Answer are:
• Leaving a Message
• Transfer to Covering Extension
• Returning to the Automated Attendant Top Menu
• Intercom Paging
• Custom Service
• Calling Operators

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PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Mailbox Parameters
→ Incomplete Call Handling for No Answer
→ Incomplete Call Handling for Busy

Feature Manual References

3.2.1.24 Intercom Paging

Operating Manual References

Subscriber Operation
1.8.5 Transferring Calls—Incomplete Call Handling

3.2.2.19 Live Call Screening (LCS)

Description
Allows the subscriber to monitor incoming calls as messages are being recorded. The subscriber has the
option of answering calls while monitoring or allowing the message to be recorded without interruption.
There are 2 modes in this feature: Hands-free and Private. Hands-free mode allows the subscriber to hear
the caller through the telephone’s speaker and answer the call by lifting the handset. Private mode alerts
with a tone and requires the telephone handset to be lifted before the message can be monitored. The
subscriber can talk to the caller by pressing the LCS button on his or her proprietary telephone (if
programmed).

Conditions
• This feature is not available for ISDN extensions and SIP extensions. For more information about
telephone types that support this feature, refer to the Operating Manual.
• LCS Button
A flexible button can be customised as the LCS button.
• Extension Personal Identification Number (PIN)
To prevent unauthorised monitoring, it is recommended the LCS user assign an extension PIN. This PIN
will be required when setting LCS (→ 2.24.1 Extension Personal Identification Number (PIN)). If the user
forgets the PIN, it can be cleared by an extension assigned as the manager.
• Each extension can be programmed to either end recording or continue recording the conversation after
the call is intercepted, through personal programming (LCS Mode Set [After Answering]).

PC Programming Manual References

8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 4
21.1 UM Configuration—[2] Class of Service—Mailbox—Message Cancel for Live Call Screening

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Live Call Screening (LCS)
Manager Operation
2.2.1 System Manager Features—Setting Class of Service (COS) Parameters
Available telephone type

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5.1.1 List of Operations by Telephone Type

3.2.2.20 Mailbox
Description
Is a place where all messages left for a subscriber are stored. Several mailbox options exist: Subscriber
mailbox, Interview Mailbox, System Manager’s mailbox, and Message Manager’s mailbox (General Delivery
Mailbox).

PC Programming Manual References

7.5.4 Utility—Report—UM View Reports
20 UM Configuration—[1] Mailbox Settings

Operating Manual References

Subscriber Operation
1.8.2 Logging in to and Configuring Your Mailbox
Manager Operation
2.2.1 System Manager Features—Setting Up Mailboxes

3.2.2.21 Mailbox Capacity Warning

Description
Allows the Unified Messaging system to alert subscribers when recording time for their mailboxes is running
low. The warning announcement will be heard at the beginning of Subscriber’s Service.

Conditions
• If the setting is longer than the Mailbox Capacity Maximum Message Time, this feature is not available.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—Mailbox
→ Mailbox Capacity Warning (Selection)
→ Mailbox Capacity Maximum Message Time (Limited) (min)

3.2.2.22 Manager Service Switching

Description
Allows a subscriber to access the System Manager’s mailbox and Message Manager’s mailbox. Subscribers
that have this feature enabled through Class of Service (COS) programming can switch mailboxes simply by
pressing [#] twice from the subscriber service menu, in the following order:
Subscriber service Message Manager service System Manager service

Conditions
• If another user is already using the manager service, the subscriber will hear an error tone and cannot
access that service.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—General—System Manager Authority

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21.1 UM Configuration—[2] Class of Service—General—Message Manager Authority

3.2.2.23 Message Transfer

Description
Allows the subscriber to transfer messages to other mailboxes after listening to them. The subscriber can
also add a personal comment at the beginning of the message before transferring it. One or more individual
mailboxes can be specified for message transfer. Messages can also be transferred using either the System
or Personal Group Distribution Lists.

Operating Manual References

Subscriber Operation
1.8.3 Message Playback and Related Features—Transferring Messages
Manager Operation
2.2.2 Message Manager Features—Managing the General Delivery Mailbox

3.2.2.24 Personal Custom Service

Description
Allows a subscriber to use the following Custom Services in his or her Personal Greeting (Personal Custom
Service):
• Transfer to Mailbox
• Transfer to Extension
• Transfer to Voice Mail Service
• Transfer to Automated Attendant Service
• Custom Service
• Transfer to Operator
• Transfer to Outside
• Page the Party
• Repeat Greeting
A caller can select the desired service from those above while listening to the Personal Greeting or
recording a message.
You can also choose what happens if a caller does not dial anything after the Personal Greeting (No DTMF
Input Operation) as follows:
• Message Recording: The caller will be guided to leave a message.
• Disconnect (All Day): The line will be disconnected all day.
• Disconnect (Only After Hours): The line will be disconnected only after hours.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—External MSG Delivery/Auto FWD/
Personal Custom Serv—Personal Custom Service

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Personal Custom Service

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3.2.2.25 Personal Greetings

Description
Are the greetings heard when a caller reaches a subscriber’s mailbox. Subscribers can record 6 types of
personal greetings. The maximum recording time for each personal greeting is determined by the
subscriber’s COS. Personal greetings are used in the following priority:
1. Temporary Personal Greeting
Used to inform callers of the subscriber’s absence (e.g., "I’m out of the office today…"). The Temporary
Personal Greeting overrides other Personal Greetings; it is always played regardless of the time mode,
whether or not the line is busy, etc. This feature is automatically activated when the subscriber records a
Temporary Personal Greeting. He will then be asked whether it should be turned off the next time he
logs in to his mailbox. When the Temporary Personal Greeting is turned off, it will be deleted
automatically and other Personal Greetings will be played as normal.
2. Busy Signal Greeting
Played whenever the subscriber’s extension is busy.
3. Personal Greeting for Caller ID
Allows subscribers to record a maximum of 4 personal greetings that are played for calls received from
preprogrammed telephone numbers. Each greeting can be assigned to a maximum of 8 telephone
numbers.
4. Absent Message Greeting
Allows subscribers to assign a personal greeting to each of the PBX’s absent messages.
5. After Hours Greeting
Played when the system is in night mode.
6. No Answer Greeting
Played during business hours (day, lunch, and break modes) when:
– the caller is connected to the subscriber’s extension but the call is not answered.
– the caller logs in to the mailbox directly.
– the Busy Signal greeting or the After Hours greeting has not been recorded.

PC Programming Manual References

21 UM Configuration—[2] Class of Service
21.1 UM Configuration—[2] Class of Service—Mailbox—Personal Greeting for Caller ID

Feature Manual References

2.20.2 Absent Message

Operating Manual References

Subscriber Operation
1.8.2 Logging in to and Configuring Your Mailbox
→ Changing or Deleting Your Personal Greeting Messages
→ Personal Greetings for Caller ID
Manager Operation
2.2.1 System Manager Features—Setting Class of Service (COS) Parameters

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3.2.2.26 Private Message

Description
Allows a subscriber and the System Manager to mark a message as "Private" when sending it to other
subscribers, the Message Manager and/or the System Manager. A private message cannot be transferred.
It is also possible to mark messages recorded in the Voice Mail service as "Private".

Operating Manual References

Subscriber Operation
1.8.4 Sending Messages—Message Delivery
Manager Operation
2.2.1 System Manager Features—Broadcasting Messages

3.2.2.27 Recover Message

Description
Enables a subscriber to recover deleted messages. After deleting messages, they will remain in the Deleted
Message Box until the time specified in System Maintenance Start Time on the following day. Until then,
the subscriber can retrieve the deleted messages from the Deleted Message Box and move them to the Old
Message Box. Message retention time is reset when a message is recovered.

PC Programming Manual References

7.11 Utility—UM - System Maintenance—System Maintenance Start Time

Operating Manual References

Subscriber Operation
1.8.3 Message Playback and Related Features—Recovering Deleted Messages

3.2.2.28 Remote Absent Message

Description
Allows a subscriber to change her absent message from a remote location by following the voice guidance.

Feature Manual References

2.20.2 Absent Message

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Setting Absent Message Remotely

3.2.2.29 Remote Call Forwarding Set

Description
Allows subscribers and the Message Manager to program their extensions from a remote location in order to
forward calls to another extension or to an outside telephone.
There are 6 forwarding settings available:
• FWD All: Forwards all incoming calls to the desired extension number.
• FWD Busy: Forwards all incoming calls to the desired extension number when the line is busy.

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• FWD No Answer: Forwards all incoming calls to the desired extension number when there is no answer.
• FWD Busy or No Answer: Forwards all incoming calls to the desired extension number when the line is
busy or there is no answer.
• FWD to CO: Forwards all incoming calls to Telephone number 1 or 2 (programmed in the Mailbox
Setting), or to any other number.
• FWD Cancel: Cancels the forwarding setting.

Conditions
• In order to use the FWD to CO option, the ability to forward calls to trunks must be enabled through
system programming.
• Extensions assigned as operators can be called by dialling [0], however, when setting this feature the
extension number (not "0") must be specified.

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings
20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Remote Call/Automatic Login/Direct
Service
21.1 UM Configuration—[2] Class of Service

Operating Manual References

Subscriber Operation
1.8.5 Transferring Calls—Remote Call Forwarding
Manager Operation
2.2.1 System Manager Features—Setting Class of Service (COS) Parameters
2.2.2 Message Manager Features—Remote Call Forwarding Set

3.2.2.30 Send to Fax Machine

Description
Through the voice mail guidance, a subscriber can send received faxes to another destination. This feature
can be used to send received faxes to a fax machine to print them out.
The subscriber can send faxes to the following destinations:
• A preprogrammed fax machine
• A fax machine specified by entering its number via the telephone
• The fax machine the subscriber is calling from (if calling from a fax machine)
The subscriber can select which faxes are sent, as follows:
• The current fax being "listened" to
• All unprinted faxes
• All faxes

Conditions
• If the forward destination is busy, it will be retried a preprogrammed number of times. If the fax still cannot
be sent, a non-delivery notification message is sent to the subscriber’s mailbox.
• If an error occurs while sending the fax, whether to retry can be set through system programming. Also,
whether to send the fax again from the start or to send it from the page where the error occurred is also
programmable.

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PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Fax Options—Fax Number Confirmation

3.2.2.31 Subscriber Tutorial

Description
Provides voice guidance to subscribers when they log in to their mailboxes for the first time. By following the
guidance, subscribers can configure the following items:
• Password
• Owner’s Name
• Personal Greetings (No Answer Greeting, Busy Signal Greeting, After Hours Greeting)

Note
If the simplified tutorial was selected through system programming, Busy Signal Greeting and After
Hours Greeting cannot be recorded.

PC Programming Manual References

21.1 UM Configuration—[2] Class of Service—General

Operating Manual References

Subscriber Operation
1.8.2 Logging in to and Configuring Your Mailbox—Subscriber Tutorial (Easy Mailbox Configuration)
Manager Operation
2.2.1 System Manager Features—Setting Class of Service (COS) Parameters

3.2.2.32 Timed Reminder Setting

Description
Allows a subscriber to set a Timed Reminder by following system prompts provided by the Unified
Messaging system. Timed Reminder is a feature that is similar to an alarm clock; the telephone will ring at
the set time (once or daily) as set by the subscriber or message manager.
A special dial tone or prerecorded message will be heard by the subscriber when going off-hook to answer
the Timed Reminder.

Conditions
• The time format (12 or 24 hour) used when setting the Timed Reminder is determined by the setting of
"Position of "AM/PM" in Time Stamp".
→ 24.4 UM Configuration—[5-4] System Parameters—Parameters—Prompt Setting—System Guidance
—Select Language—Position of "AM/PM" in Time Stamp
• A subscriber must have his or her own extension in order to use this feature.
• Extensions assigned as operators can be called by dialling [0], however, when setting this feature the
extension number (not "0") must be specified.

PC Programming Manual References

24.4 UM Configuration—[5-4] System Parameters—Parameters

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Feature Manual References

2.24.3 Timed Reminder

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Setting the Alarm (Timed Reminder)
Manager Operation
2.2.2 Message Manager Features—Setting the Timed Reminder

3.2.2.33 Toll Saver

Description
Allows a subscriber to check his or her mailbox from preprogrammed Caller ID numbers, the DID number or
the telephone number specified to the preprogrammed trunk group number without incurring telephone
charges.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Remote Call/Automatic Login/Direct
Service
→ Auto Login Extension
→ Auto Login Extension Password Entry Requirement
→ Auto Login Caller ID 1, 2
→ Auto Login Caller ID Password Entry Requirement
→ Auto Login DDI/DID
→ Auto Login TRG No.
→ Auto Login DDI/DID, TRG No. Password Entry Requirement
→ Auto Login Toll Saver
23.2 UM Configuration—[4-2] Service Settings—Parameters—Toll Saver
→ Delayed Answer Time for New Message (5-60 s)
→ Delayed Answer Time for No New Message (5-60 s)

Feature Manual References

3.2.2.3 Automatic Login

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Assigning Your Telephone Numbers for Remote Automatic Log-in and Toll Saver

3.2.2.34 Two-way Record/Two-way Transfer

Description
The following manual recording features are available:
• Two-way Record
Allows a subscriber to record the conversation that he or she is having with a caller. The conversation is
saved in the subscriber’s mailbox as either an old or new message, depending on the subscriber’s COS.
• Two-way Transfer
Allows a subscriber to record the conversation that he or she is having with a caller. The conversation is
saved in another subscriber’s mailbox as a new message.
• One-touch Two-way Transfer

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Allows a subscriber to record the conversation that he or she is having with a caller, with a one-touch
operation. The conversation is saved in another subscriber’s mailbox as a new message.
Unlimited Message Length
Allows subscribers to record for an unlimited length of time when recording two-way conversations into their
own or another subscriber’s mailbox (Two-way Record or Two-way Transfer). The maximum recording time
for other messages will automatically be set to 60 minutes.

Note
You should inform the other party that the conversation will be recorded before beginning to record any
telephone conversation.

Conditions
• KX-NSUN001, KX-NSUN010, KX-NSUN050, KX-NSUN100, KX-NSUN500 (Normal User AK) or
KX-NSUM001, KX-NSUM010, KX-NSUM050, KX-NSUM100, KX-NSUM500 (Mobile User AK) or
KX-NSUA001, KX-NSUA010, KX-NSUA050, KX-NSUA100, KX-NSUA500 for Two-Way recording is
required to use this feature. One activation key is required for each extension that will use this feature.
Also, the Two-way Recording setting for each extension must be set to Enable.
• Two-way Record/Two-way Transfer Button
A flexible button can be customised as the Two-way Record or the Two-way Transfer button. An
extension number can be assigned to the Two-way Transfer button so that it can be used as a one-touch
record button for the mailbox of the specified extension (One-touch Two-way Transfer Button).
• When all of the Unified Messaging ports are busy:
a. Pressing the Two-way Record button sends a warning tone.
b. Pressing the Two-way Transfer button followed by an extension number sends a warning tone.
• To allow unlimited recording time, the "Message Length" parameter of the subscriber’s Class of Service
(COS) must be set to "Unlimited".
→ 21.1 UM Configuration—[2] Class of Service—Mailbox—Message Length (Selection)

PC Programming Manual References

8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—UM—Two-way Recording
8.3.1 Users—Flexible Button—Wired Extension
→ Extension Number (for Two-way Record)
→ Extension Number (for Two-way Transfer)
→ Extension Number (for Voice Mail Transfer)
8.3.2 Users—Flexible Button—Portable Station
→ Extension Number (for Two-way Record)
→ Extension Number (for Two-way Transfer)
→ Extension Number (for Voice Mail Transfer)
21.1 UM Configuration—[2] Class of Service—Mailbox—Two-way Recorded Message Save Mode

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

Subscriber Operation
1.8.6 Other Features—Recording Your Conversation
3.1.3 Customising the Buttons

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3.2.2 Subscriber Features

3.2.2.35 Urgent Message

Description
Allows subscribers, callers, and the System Manager to specify a message as "Urgent". When a subscriber
listens to messages, urgent messages will be given priority over other messages in a mailbox (i.e., they will
be played first) if the "First Playback Urgent Messages" parameter of the subscriber’s Class of Service
(COS) is set to "Yes". If "Only Urgent Messages" is set as the Notification Type for the subscriber’s mailbox,
Message Waiting Notification will be activated only when an urgent message is received.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Notification Parameters
→ Telephone Device—Device No. 1, 2, 3—Only Urgent Messages
→ E-mail/Text Message Device—Device No. 1, 2, 3—Only Urgent Messages
21.1 UM Configuration—[2] Class of Service—Mailbox—First Playback Urgent Message

3.2.2.36 Voice Mail (VM) Transfer Button

Description
A subscriber can transfer a call to a mailbox of an extension by pressing the VM Transfer button and
entering the number of the extension. The transferred caller can then leave a message in the mailbox. The
VM Transfer button can also be used by extension users at other times, as follows:
a. By pressing the VM Transfer button when the extension is idle, the extension’s mailbox is called and
messages can be listened to. This feature can be used even if the Message Waiting lamp is not on.
b. Pressing the VM Transfer button while a call is incoming will redirect the call to the called extension’s
mailbox. This is useful when the called extension user does not want to answer the call.
c. If you call an extension, and the other party does not answer, pressing the VM Transfer button will
transfer your call to the called extension’s mailbox so that you can leave a message. This feature can
also be used when the called extension is busy or set to DND.

Conditions
• A flexible button can be customised as the VM Transfer button with the floating extension number of the
UM group as the parameter.
• If you are to use a PS linked to the User Container, the PS’s VM Transfer button cannot be used to
redirect an incoming call to the called extension’s mailbox.

PC Programming Manual References

8.3.1 Users—Flexible Button—Wired Extension—Extension Number (for Voice Mail Transfer)
8.3.2 Users—Flexible Button—Portable Station—Extension Number (for Voice Mail Transfer)

Feature Manual References

2.21.2 Flexible Buttons

3.2.2.37 Web Programming

Description
Subscribers can access and change various settings via Web Maintenance Console.
Users
Users logged in to a "User (User)" account can access the following items:

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• Mailbox settings
Subscribers can configure settings such as the name associated with their mailbox, the mailbox
password, and notification parameters.
• Voice prompts
Subscribers can play, record, and delete the following types of voice prompts for their mailbox:
– Mailbox owner name
– Personal greetings
– Personal caller ID name
– Personal distribution list voice label
– Interview mailbox questions
Administrators
Users logged in to a "User (Administrator)" account can access the following items in addition to the settings
available to "User (User)" accounts:
• Voice prompts
– Custom service
– Mailbox group name
– System caller name
• Reports
Administrators can view and clear the various types of reports.
→ 7.5.4 Utility—Report—UM View Reports

Conditions
• Subscribers must have a user ID and password to log in to Web Maintenance Console.
• The System Manager and the Message Manager do not necessarily have administrator privileges.

PC Programming Manual References

20 UM Configuration—[1] Mailbox Settings
22.1 UM Configuration—[3-1] UM Extension / Trunk Service—Service Group—Day, Night, Lunch, and Break
Mode - Incoming Call Service Prompt
24.3 UM Configuration—[5-3] System Parameters—System Caller Name Announcement
24.4 UM Configuration—[5-4] System Parameters—Parameters—Prompt Setting

Feature Manual References

5.5.1 PC Programming

Operating Manual References

3.2 System Programming Using Web Maintenance Console

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3.3 E-mail Client Integration Features

3.3 E-mail Client Integration Features

3.3.1 IMAP Integration
Description
By configuring an IMAP account, subscribers can access the contents of their mailboxes through an e-mail
client. All that is necessary is an e-mail client that supports IMAP4.
With IMAP integration, users can do the following:
• Play back voice messages
• View received fax messages
• Save voice and fax message data to their PCs
• Delete voice and fax messages

Conditions
• KX-NSUN001, KX-NSUN010, KX-NSUN050, KX-NSUN100, KX-NSUN500 or KX-NSUM001,
KX-NSUM010, KX-NSUM050, KX-NSUM100, KX-NSUM500 or KX-NSUA001, KX-NSUA010,
KX-NSUA050, KX-NSUA100, KX-NSUA500 for IMAP Integration is required.
• New voice messages and fax messages can be sent as attachments to e-mails to a separate e-mail
account (→ 3.2.1.29 Message Waiting Notification—E-mail Device).
• A tool for users to help manage IMAP connections is available. A link to where users can download the
tool is provided in Web Maintenance Console.
→ "Unified Message"—"Unified Messaging Plug in" in 8.1.1 Users—User Container—Add User/Edit User
• Up to 72 sessions can be used simultaneously with IMAP.

PC Programming Manual References

20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Mailbox Parameters—Mailbox Password
(Message Client)
21.1 UM Configuration—[2] Class of Service—General
→ Fax Option
→ Desktop Messaging

Operating Manual References

Subscriber Operation
1.8.6 Other Features—IMAP

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Network Features

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4.1 Public Network Features

4.1 Public Network Features

4.1.1 SIP (Session Initiation Protocol) Trunk
Description
Through a V-SIPGW card, the PBX can connect to an Internet telephony service provided by an ITSP
(Internet Telephony Service Provider).
An ITSP provides its telephony service partly through the conventional telephone network (e.g., ISDN and
Mobile), which is fee-based. An ISP (Internet Service Provider), which can also act as a SIP provider, does
not provide the telephone connection itself. However, providing its users with Internet access, an ISP
provides voice communication on the Internet for free. In this way, with VoIP technology based on the SIP
protocol, the cost of voice communication can be much cheaper than conventional telephone networks.

WAN (Wide Area Network)

Internet

ITSP

Local
Telephone
ISP

LAN Router
(Local Area Network)
Switching
Hub

PC IP-PT
V-SIPGW16

PBX

Conditions
• A subscription with an ISP is required for an Internet connection.
• A subscription with an ITSP is required for a telephone connection. The ISP and ITSP may be part of the
same company.

Installation Manual References

4.4 Virtual Cards
6.3 SIP Trunks

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PC Programming Manual References

9.9 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Shelf Property
9.11 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Port Property

4.1.1.1 SIP Server Failover

Description
When communicating with a SIP server (e.g., when the V-SIPGW card is in INS status, when sending
periodic REGISTER requests), if the SIP server does not reply after sending the REGISTER/INVITE
request, the REGISTER/INVITE request is sent again using one of the following methods.
• If a failover SIP server is registered, the REGISTER/INVITE request is sent to that server.
• If a failover SIP server is not registered, a REGISTER/INVITE request is immediately sent to the
destination obtained from a DNS resolution of the registrar server name.

Conditions
• The failover destination is decided based on the following order.
1. The failover destination IP address of the SIP server/Registrar server
2. DNS SRV record / A record round robin
• If the above failover operation fails, communication is attempted over a public trunk rather than a SIP
trunk.
Automatic Rerouting of SIP Trunk Calls to Public Trunks
When a SIP trunk call cannot be completed successfully, the PBX can automatically attempt to make the call
using a public trunk instead. This provides a backup method of making calls in cases when IP network
transmission cannot be completed successfully. For more information, refer to "4.3.2 Voice over Internet
Protocol (VoIP) Network—Automatic Rerouting of VoIP Calls to Public Trunks".

[Example]

Telephone Company Telephone Company

(area code: 098) (area code: 012)

012-345-1011
PBX-1 PBX-2

Extn.1000 V-SIPGW IP Network V-SIPGW Extn.1011

(012-345-1011)
Dials
"012-345-1011".

Installation Manual References

4.4 Virtual Cards
6.3 SIP Trunks

PC Programming Manual References

9.11 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Port Property—Main—SIP Server IP
Address for Failover

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4.1.2 Integrated Services Digital Network (ISDN) Service Features

9.11 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Port Property—Register—Registrar

Server IP Address for Failover
9.11 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Port Property—Option—Failover
Timer(REGISTER)

4.1.2 Integrated Services Digital Network (ISDN) Service

Features
4.1.2.1 Integrated Services Digital Network (ISDN)—SUMMARY
Description
ISDN is a digital switching and transmission network. ISDN transmits voice, data, and image in digital
format. ISDN lines, if available, can be connected to public line (trunk), private line (QSIG), or ISDN terminal
devices (extension).
1. ISDN Interface and Configuration
Interface Type Description Port Mode Configuration
*3
Type
Basic Rate Interface Provides two 64 kbps B Trunk, Point-to-Point/
(BRI) channels for communication and Extension
*1
Point-to-multipoint
one 16 kbps D channel for
signalling (2B + D). QSIG (Master, Point-to-Point
*1
Slave)
Primary Rate Interface Provides thirty or twenty-three Trunk, QSIG Point-to-Point
(PRI) 64 kbps B channels for (Master, Slave),
communication and one 64 kbps Extension
*2

D channel for signalling (30B +

D/23B + D).

*1 → 9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Main—Port Type

*2 → 9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Main—Port Type
*3 For BRI only:
→ 9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Network—Access Mode

Note
Point-to-Point (P-P):
One ISDN terminal device can be connected to one ISDN port.
Point-to-multipoint (P-MP):
A maximum of eight ISDN terminal devices can be connected to one ISDN port.

2. ISDN Supplementary Service Table for Public Network

Service Description & Reference
Direct Dialling In (DDI) Directs a call with a DDI number to a preprogrammed destination.
(P-P only)

→ 2.1.1.3 Direct Inward Dialling (DID)/Direct Dialling In (DDI)

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Service Description & Reference

Multiple Subscriber Number One ISDN port can support a maximum of 10 MSNs for routing
(MSN) incoming calls. (P-MP only)

→ 2.1.1.4 Multiple Subscriber Number (MSN) Ringing Service

Calling Line Identification Sends the caller’s telephone number to the network when making a
Presentation (CLIP) call. The called party can see the number on his telephone display
before answering the call.

→ 4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/

COLP)
Connected Line Sends the telephone number of the answered party to the network
Identification Presentation when answering a call. The caller can see the number on his
(COLP) telephone display when the line is connected.

→ 4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/

COLP)
Calling Line Identification Prevents the caller’s CLI being presented to the called party by the
Restriction (CLIR) caller.

→ 4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/

COLP)
Connected Line Prevents COLP being sent by the answered party.
Identification Restriction
(COLR) → 4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/
COLP)
Subaddressing (SUB) You may add digits after the telephone number. These digits will be
passed to ISDN terminal device.
User-to-user Signalling Type Transmits a specified volume of the data through the D channel
1 (UUS-1) between the ISDN terminal devices of different PBXs using UUS
types 1.
Advice of Charge (AOC) The PBX can receive the call charge information on ISDN lines from
the telephone company.

→ 4.1.2.3 Advice of Charge (AOC)

Call Forwarding (CF)—by Forwards an incoming call to another outside party using the ISDN
ISDN service of the telephone company.

→ 4.1.2.4 Call Forwarding (CF)—by ISDN (P-MP)

→ 4.1.2.5 Call Forwarding (CF)—by ISDN (P-P)
Call Hold (HOLD)—by ISDN Puts one ISDN call on hold.

→ 4.1.2.6 Call Hold (HOLD)—by ISDN

Call Transfer (CT)—by ISDN Transfers an ISDN call to an outside party. Call Transfer with
Announcement and Call Transfer without Announcement are
available.

→ 4.1.2.7 Call Transfer (CT)—by ISDN

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4.1.2 Integrated Services Digital Network (ISDN) Service Features

Service Description & Reference

Three-party Conference Establishes a three-party conference call using the ISDN service of
(3PTY)—by ISDN the telephone company.

→ 4.1.2.8 Three-party Conference (3PTY)—by ISDN

Malicious Call Identification An extension user can ask the telephone company to trace a
(MCID) malicious caller. Information on the malicious call will be received
later on.

→ 4.1.2.9 Malicious Call Identification (MCID)

Completion of Calls to Busy If a call is made to an outside party and the party is busy, an
Subscriber (CCBS) extension can receive callback ringing when the called party
becomes free.

→ 4.1.2.10 Completion of Calls to Busy Subscriber (CCBS)

3. ISDN Centrex Service

An extension user can have access to the features of the ISDN Centrex Service of the telephone
company (e.g., Call Transfer). This is used by putting the current ISDN call on hold by sending a flash/
recall signal. This feature is enabled or disabled for each ISDN port.
→ 9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—ISDN CO—ISDN
Centrex
→ 9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—CO Setting—ISDN
Centrex
4. Private Networking Connection (QSIG)
Private networking with QSIG is possible using an ISDN line. The QSIG mode, Master or Slave, can be
enabled on an ISDN port basis.
5. ISDN Extension (→ 5.2.5.1 ISDN Extension)
An ISDN (BRI/PRI) port can be used for extension connection. While the extension connection is
enabled, ISDN terminal devices (e.g., ISDN telephone, G4 fax machine, PC) or a behind PBX can be
connected to the port.
When the ISDN port is in P-P configuration, one ISDN terminal device can be connected to the port.
When the ISDN port is in P-MP configuration, a maximum of eight ISDN terminal devices can be
connected to the port. However, only a maximum of two devices can be used simultaneously.

Conditions
• Overlap/En bloc
For each ISDN port, either Overlap or En bloc can be selected as the dialling method for which the PBX
sends telephone numbers to the telephone company. The selected dialling method must be offered by the
telephone company. When "Overlap" is selected, the PBX sends each dialled digit individually.
When "En bloc" is selected, the PBX sends all of the dialled digits at once.
→ 9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—ISDN CO—ISDN
Outgoing Call Type
→ 9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—CO Setting—ISDN
Outgoing Call Type

In En bloc mode, the PBX recognises that the user is finished dialling when:
– the # key is pressed (programmable).
→ 10.9 PBX Configuration—[2-9] System—System Options—Option 2—ISDN en Bloc Dial—[#] as
End of Dial for en Bloc mode
– the dialled number is a preprogrammed telephone number.

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 4.1.2 Integrated Services Digital Network (ISDN) Service Features

→ 11.1.4 PBX Configuration—[3-1-4] Group—Trunk Group—Dialling Plan

– the inter-digit timer expires.
→ 10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Dial—
Extension Inter-digit
• Some supplementary services are provided by the key protocol (→ 4.1.2.11 ISDN Service Access by
Keypad Protocol).
• Extension Number
An extension number can be assigned to each ISDN port.
→ 8.1.1 Users—User Container—Add User/Edit User—User Information—Extension Number
• Network Type (BRI only)
Each BRI port can be set to the requirements of each country/area.
→ 9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Network—Network
Type
• L1 Active Mode and L2 Data Link Mode (BRI only)
The active mode of Layer 1 (L1) and the data link mode of Layer 2 (L2) can be selected individually
(Permanent or Call by Call) on an ISDN port basis.
9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Network
→ L1 Mode
→ L2 Mode
• TEI Assignment Mode (BRI only)
The Terminal Endpoint Identifier (TEI) assignment mode can be selected on an ISDN port basis
→ 9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Network—TEI Mode
• Bearer Mode
The bearer mode can be assigned on an extension basis.
→ 8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 7—ISDN Bearer
• Automatic ISDN Configuration
ISDN (BRI) port configuration can be set automatically through system programming.
→ 6.2 Tool—BRI Automatic Configuration
The following items will be set by making and receiving a call using the subscriber number assigned for
each ISDN (BRI) port:
– L1 Active Mode
– L2 Data Link Mode
– Access Mode (Point-to-Point/Point-to-Multipoint)
– TEI Assignment Mode (Fix 00–63/Automatic)

PC Programming Manual References

6.2 Tool—BRI Automatic Configuration
9.25 PBX Configuration—[1-1] Configuration—Slot—Card Property - BRI type/PRI type
9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port
10.3 PBX Configuration—[2-3] System—Timers & Counters—Dial / IRNA / Recall / Tone—Dial—Extension
Inter-digit
10.9 PBX Configuration—[2-9] System—System Options—Option 2—ISDN en Bloc Dial—[#] as End of Dial
for en Bloc mode
8.1.1 Users—User Container—Add User/Edit User—User Information—Extension Number
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 7—ISDN Bearer
11.1.4.1 PBX Configuration—[3-1-4] Group—Trunk Group—Dialling Plan—Auto Assign

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4.1.2 Integrated Services Digital Network (ISDN) Service Features

Feature Manual References

4.3.4 QSIG Standard Features

4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/

COLP)
Description
Calling Line Identification Presentation (CLIP):
The PBX can send a preprogrammed telephone number to the network when an extension user makes a
call. The called party can see the number on his telephone display before answering the call.
Connected Line Identification Presentation (COLP):
The PBX sends a preprogrammed telephone number to the network when the extension user answers an
incoming call. The caller can see the number of the answering party on his telephone display when the call
is answered.

[CLIP Example]

1) Dials 2) "12345678"
"87654321". is displayed.
PBX

ISDN

Caller Called party

(CLIP/COLP No.: 12345678) (CLIP/COLP No.: 87654321)

[COLP Example]

1) Dials Called party

"111222333". (CLIP/COLP No.:
111222333)
PBX

ISDN FWD, IRNA, etc.

Caller Answering party

3) "111222444" (CLIP/COLP No.:
is displayed. 111222444)

2) Answers the call.

CLIP/COLP Number:
The telephone numbers sent to the network for CLIP/COLP can be assigned as follows:
• CLIP/COLP number for each ISDN port (subscriber’s number).
• CLIP/COLP number for each extension.
• CLIP/COLP number for each incoming call distribution group.
Each extension can select either the CLIP/COLP number for the ISDN port or the extension to be used. The
CLIP/COLP number for the incoming call distribution group is used when making a call by pressing the ICD
Group button or receiving a call which arrives at the ICD Group button.

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Calling/Connected Line Identification Restriction (CLIR/COLR):

It is possible for each extension to restrict the sending of its telephone number to the network by pressing
the CLIR button, COLR button, or entering the feature number.

Conditions
• The availability of this feature is dependent on the contract with the telephone company.
• CLIP/COLP features comply with the following European Telecommunication Standard (ETS)
specifications:
– ETS 300 092 Calling Line Identification Presentation (CLIP) supplementary service.
– ETS 300 097 Connected Line Identification Presentation (COLP) supplementary service.
• CLIR/COLR features comply with the following European Telecommunication Standard (ETS)
specifications:
– ETS 300 093 Calling Line Identification Restriction (CLIR) supplementary service.
– ETS 300 098 Connected Line Identification Restriction (COLR) supplementary service.
• The CLIP/COLP number for the connected ISDN port can be used for the ISDN terminal devices which
cannot be assigned their own CLIP/COLP number, such as a doorphone.
• COLP/CLIR/COLR Assignment for Each Port
Each service can be enabled or disabled on each ISDN port of the PBX.
• CLIR Button and COLR Button
It is possible to switch between CLIP and CLIR by pressing the CLIR button, and COLP and COLR by
pressing the COLR button. A flexible button can be customised as the CLIR or COLR button.
• The CLIP/COLP number must match the telephone number provided by the telephone company.
Otherwise it will be ignored or replaced by another number.
• When using a private network, the extension number assigned for each extension through system
programming is sent for CLIP/COLP. (→ 4.3.4.2 Calling/Connected Line Identification Presentation (CLIP/
COLP) and Calling/Connected Name Identification Presentation (CNIP/CONP)—by QSIG)
• When forwarding calls to a public trunk, system programming selects whether the CLIP number of the
calling party or of the forwarding extension is sent to the forward destination.
However, if the call is transferred to another PBX via a private network from a UM, the CLIP number of
the calling party is always sent, regardless of system programming.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port
→ ISDN CO—Subscriber Number
→ Supplementary Service—COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT,
CCBS, AOC-D, AOC-E, 3PTY
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port
→ CO Setting—Subscriber Number
→ Supplementary Service—COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT,
CCBS, AOC-D, AOC-E, E911, 3PTY
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ COLR Set / Cancel
→ CLIR Set / Cancel
→ Switch CLIP of CO Line / Extension
10.9 PBX Configuration—[2-9] System—System Options—Option 4
→ Send CLIP of CO Caller—when call is transferred to CO (CLIP of Held Party)
→ Send CLIP of CO Caller—when call is forwarded to CO
→ Send CLIP of Extension Caller—when call is forwarded to CO

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4.1.2 Integrated Services Digital Network (ISDN) Service Features

11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Main—CLIP

on ICD Group Button
8.1.1 Users—User Container—Add User/Edit User—User Information—Extension Number
8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings
→ CLIP—CLIP ID
→ CLIP—CLIP on Extension/CO
→ CLIP—CLIR
→ CLIP—COLR
8.3.1 Users—Flexible Button—Wired Extension—Flexible Button—Type
8.3.2 Users—Flexible Button—Portable Station—Flexible Button—Type

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

1.9.4 Displaying Your Telephone Number on the Called Party and Caller’s Telephone (Calling/Connected
Line Identification Presentation [CLIP/COLP])
1.9.5 Preventing Your Telephone Number Being Displayed on the Caller’s Telephone (Connected Line
Identification Restriction [COLR])
1.9.6 Preventing Your Number Being Displayed on the Called Party’s Telephone (Calling Line Identification
Restriction [CLIR])

4.1.2.3 Advice of Charge (AOC)

Description
The PBX can receive the call charge information on ISDN lines from the telephone company.
There are the following types:
Type Description
Advice of Charge During AOC is received during the call and when the call is completed.
Call (AOC-D)
Advice of Charge AOC is received when the call is completed.
At End of Call (AOC-E)

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 182 Advice of Charge (AOC) supplementary service Digital Subscriber Signalling System No.
One (DSS1) protocol.
• A DPT user can see the call charge information on the display during the call.
• Budget Management
If the amount of call charge reaches the preprogrammed limit, an extension user cannot make further
calls. (→ 2.7.2 Budget Management)
• AOC for ISDN extension
An ISDN extension also receives AOC.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, 3PTY
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911, 3PTY

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4.1.2.4 Call Forwarding (CF)—by ISDN (P-MP)

Description
An extension user can forward the incoming ISDN call to another outside party using the ISDN service of
the telephone company, instead of the PBX feature, when the call is received through an ISDN line.
Extension users can set the forward destination to the network on an MSN basis. The setting must have
been done beforehand to use this feature. There are the following types:
Type Description
Call Forwarding— Incoming calls are forwarded unconditionally.
Unconditional (CFU)
Call Forwarding—Busy An incoming call is forwarded when the destination in the PBX is
(CFB) busy.
Call Forwarding—No Reply An incoming call is forwarded when the destination in the PBX
(CFNR) does not answer within a preprogrammed time period.

When the CFB or CFNR has been set, the network determines to forward the incoming call to the
preprogrammed destination after the call reached the PBX once. When the CFU has been set, the network
forwards the call directly to the preprogrammed destination.

[Example]

<FWD> <Call Forwarding (CF) by ISDN (P-MP)>

ISDN
ISDN MSN:123-4567
CFU
Destination: 01-23-4567

MSN: 123-4567
987-6543

PBX PBX

Outside Caller Outside Party Outside Caller Outside Party

(01-23-4567) (01-23-4567)
Dials "123-4567". Dials "123-4567".

Extn. 1011 Extn. 1011

(MSN: 123-4567
Forward Destination
of Trunk Calls: 01-23-4567)

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 207 Diversion supplementary service.
• The availability of this feature is dependent on the contract with the telephone company.

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4.1.2 Integrated Services Digital Network (ISDN) Service Features

• The feature requires the MSN service. (→ 2.1.1.4 Multiple Subscriber Number (MSN) Ringing Service)
• COS programming determines the extensions that are able to use this feature.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—ISDN-FWD (MSN) Set /
Cancel / Confirm
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—CO & SMDR—CF (MSN)

Feature Manual References

2.3.2 Call Forwarding (FWD)
5.1.1 Class of Service (COS)

Operating Manual References

1.6.1 Forwarding Calls

4.1.2.5 Call Forwarding (CF)—by ISDN (P-P)

Description
An extension user can forward the incoming ISDN call to another outside party using the ISDN service of
the telephone company, instead of the PBX feature, when the call is received through an ISDN line.
The network directly forwards the call to the destination which the extension user has set in the PBX as the
forward destination of trunk calls; the network is instructed by the PBX. This feature is available only when
the call is received through an ISDN port which supports this feature.
Call Forwarding—Unconditional (CFU), Call Forwarding—Busy (CFB), and Call Forwarding—No Reply
(CFNR) are applied to this feature.

[Example]
<FWD> <Call Forwarding (CF) by ISDN (P-P)>

ISDN ISDN

CF Request
to 01-23-4567
PBX
PBX

Outside Caller Outside Party Outside Caller Outside Party

(01-23-4567) (01-23-4567)
Dials "01-45-6789". Dials "01-45-6789".

Extn. 1011 Extn. 1011

(DDI No.: 01-45-6789 (DDI No.: 01-45-6789
Forward Destination Forward Destination
of Trunk Calls: 01-23-4567) of Trunk Calls: 01-23-4567)

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 4.1.2 Integrated Services Digital Network (ISDN) Service Features

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 207 Diversion supplementary service.
• The availability of this feature is dependent on the contract with the telephone company.
• This feature can be enabled or disabled on each ISDN port of the PBX.
• This feature is available when the same trunk group is used for the incoming call and the forwarded call.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, 3PTY
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911, 3PTY
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—ISDN-FWD (MSN) Set /
Cancel / Confirm
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—CO & SMDR—CF (MSN)

Feature Manual References

2.3.1 Call Forwarding (FWD)/Do Not Disturb (DND)—SUMMARY
2.3.2 Call Forwarding (FWD)

Operating Manual References

1.6.1 Forwarding Calls

4.1.2.6 Call Hold (HOLD)—by ISDN

Description
An ISDN call can be put on hold using the ISDN service of the telephone company, instead of the PBX
feature. This can be a part of a Call Transfer (CT)—by ISDN (→ 4.1.2.7 Call Transfer (CT)—by ISDN) and
Three-party Conference (3PTY)—by ISDN (→ 4.1.2.8 Three-party Conference (3PTY)—by ISDN). This
feature allows an ISDN call to be held, and a call to be made to another outside party using only one
communication channel of ISDN. A PT user can easily use this feature by pressing the ISDN Hold button.

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 141 Call Hold (HOLD) supplementary service.
• ISDN Hold Button
A flexible button can be customised as the ISDN Hold button.
• The availability of this feature is dependent on the contract with the telephone company.
• The TRS/Barring feature is applied when making a call after activating this feature. (→ 2.7.1 Toll
Restriction (TRS)/Call Barring (Barring))
• ARS cannot be applied to the call dialled after activating this feature. (→ 2.8.1 Automatic Route Selection
(ARS))
• It is impossible to seize any other trunk during this feature.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—ISDN Hold

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4.1.2 Integrated Services Digital Network (ISDN) Service Features

8.3.1 Users—Flexible Button—Wired Extension—Type

8.3.2 Users—Flexible Button—Portable Station—Type

Feature Manual References

2.21.2 Flexible Buttons

4.1.2.7 Call Transfer (CT)—by ISDN

Description
An ISDN call can be transferred to an outside party using the ISDN service of the telephone company,
instead of the PBX feature, without occupying a second ISDN line.

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 369 Explicit Call Transfer (ECT) supplementary service.
• The availability of this feature is dependent on the contract with the telephone company.
• This feature can be enabled or disabled on an ISDN port basis.
• If an ISDN port is in P-P configuration, this feature can be used only when the network supports the
"explicit linkage" option.
• Call Transfer with Announcement and Call Transfer without Announcement is possible. (→ 2.12.1 Call
Transfer)
• The call charges after completing this feature will not be recorded by the PBX.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, 3PTY
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911, 3PTY

Operating Manual References

1.4.1 Transferring a Call (Call Transfer)

4.1.2.8 Three-party Conference (3PTY)—by ISDN

Description
During a conversation using an ISDN line, an extension user can add another party and establish a three-
party conference call using the ISDN service of the telephone company, instead of the PBX feature.

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 188 Three-Party (3PTY) supplementary service.
• The availability of this feature depends on the contract with the telephone company.
• This feature can be enabled or disabled on an ISDN port basis.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, 3PTY

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9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service—

COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911, 3PTY

Operating Manual References

1.4.5 Multiple Party Conversation

4.1.2.9 Malicious Call Identification (MCID)

Description
An extension user can ask the telephone company to trace a malicious caller during a call or while hearing
reorder tone after the caller hangs up. Information on the malicious call will be received later on.

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 130 Malicious Call Identification (MCID) supplementary service.

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—MCID

Operating Manual References

1.3.6 Identifying Malicious Calling Parties (Malicious Call Identification [MCID])

4.1.2.10 Completion of Calls to Busy Subscriber (CCBS)

Description
If the called party is busy and the call has been made using an ISDN line, an extension user can set to
receive callback ringing when the called party becomes free. When the user answers the callback ringing,
that party’s number is automatically dialled.

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 359 Completion of Calls to Busy Subscriber (CCBS) supplementary service.
• This feature is available under the following conditions:
a. The caller’s PBX is capable of using CCBS and the service is provided by the network.
b. The called party’s PBX is capable of accepting CCBS.
• To receive and send CCBS, receiving and sending CCBS must be enabled individually on an ISDN port
basis through system programming.
• An extension user can set only one CCBS. The last setting is effective.
• The CCBS setting is cancelled if there is no callback ringing within 60 minutes or callback ringing is not
answered within 10 seconds.
• After using the CCBS feature, using Last Number Redial will not retrieve the number dialled by CCBS. (→
2.6.3 Last Number Redial)
• An extension user that has set the CCBS feature cannot receive callback ringing while the extension is
holding a call.

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4.1.2 Integrated Services Digital Network (ISDN) Service Features

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service
→ Supplementary Service
→ CCBS Option
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service
→ COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911,
3PTY
→ CCBS Type
→ CCBS Delete Digits

Operating Manual References

1.2.4 When the Dialled Line is Busy or There is No Answer

4.1.2.11 ISDN Service Access by Keypad Protocol

Description
ISDN provides some supplementary services by key protocol, and they may require a service access code
to be dialled.

Conditions
• This feature complies with the following European Telecommunication Standard (ETS) specification:
– ETS 300 122 Generic keypad protocol for the support of supplementary service (ISDN Service
Access).
• ISDN Service Button
A flexible button can be customised as an ISDN Service button. A service access code can also be
assigned on this button for a quick operation.
• This feature is not available to an SLT.

PC Programming Manual References

8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Dial (for ISDN Service)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Dial (for ISDN Service)

Feature Manual References

2.21.2 Flexible Buttons
6.1 Capacity of System Resources

Operating Manual References

1.2.5 Accessing the ISDN Service (ISDN Service Access)

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 4.2 Multi-connection Networking

4.2 Multi-connection Networking

4.2.1 Multi-connection Networking Overview
Description
You can create a private IP network in the Multi connection Network.
By integrating an existing PBX with the NSX Server Unit, many ways to expand the system will become
available.
The KX-NSX series will be referred to as "NSX Server Unit" and the KX-NS series will be referred to as
"Expansion Gateway".
The NSX Server Unit and Expansion Gateway are referred to as "sites".

Note
In this manual, the NSX Expansion Box is treated the same as the KX-NS1000. For details about the
NSX Expansion Box, refer to the NSX Expansion Box manual.

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4.2.1 Multi-connection Networking Overview

An example of a Multi-connection Network configuration is shown below.

You can perform maintenance on all

the sites by logging on to the NSX Server.

LAN

Site #2

IP Networking

Expansion GW
Site #1

Site #3

NSX Server
Expansion GW

Stacking

Site #4
:
:

Expansion GW

A connection seamless service

is provided to the terminal of
Stacking
each site. Equivalent to when
connected to one PBX.
(One Look)

[KX-NS300/KX-NS500/KX-NS1000 Expansion Gateway connection]

Connect the Legacy gateway/terminal and integrate the existing PBX as the Expansion Gateway.

Features of Multi-connection Networking

• Simple setup and expansion

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When setting up a PBX initially, the first step is to specify whether it is a NSX Server Unit or an Expansion
Gateway. If a PBX is specified as a Slave unit, it automatically locates the NSX Server Unit if they are on
the same LAN.
→ 2.1.4 Easy Setup Wizard
• After the initial settings have been configured, the only additional step is to register the Expansion
Gateway to the NSX Server Unit using a simple wizard. Note that an Expansion Gateway will not function
until it is registered.
→ 3.1.1 Home Screen—Add Site Wizard
• Common extension numbering
All extensions are managed from the NSX Server Unit and behave as if they were all connected to one
PBX.
This feature is similar to common extension numbering over a TIE connection.
→ 4.3.2.2 Common Extension Numbering for Multiple PBXs, but it does not require any system
programming.
• Trunk access
Extension users can access trunks at remote PBXs. Through system programming, outgoing calls can be
routed so as to reduce long-distance call charges.
• Simplified programming
All PBXs in a Multi connection network can be programmed by logging in to the NSX Server Unit via Web
Maintenance Console. Both the global settings (settings that apply to all PBXs) and local settings
(settings that apply only to a single PBX) can be programmed without logging in to each PBX individually.
You can also log in to a Slave unit and program that PBX’s local settings.
Restricting trunk access through COS settings
Trunk lines in a Multi-connection network are available to extensions at any site. While this feature may be
desirable in some cases (for example allowing extension users to access a local trunk from a distant
location to reduce call charges), COS settings can be used to reserve access to specified trunks for
extensions at a certain site. The following example shows how to allow only extensions 101, 102, and 103
access to trunks 1 and 2.
1. In the trunk line settings, assign trunks 1 and 2 to a unique trunk group. (Trunk group 3, in this
example.)
→ 18.1 PBX Configuration—[10-1] CO & Incoming Call—CO Line Settings

Programming example
Trunk Trunk Name Trunk Group Number
1 Trunk 1 3
2 Trunk 2 3
3 Trunk 3 1
: : :

2. In the COS settings, assign extensions 101, 102, and 103 to a unique COS. (COS 2, in this example.)
→ 8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings

Programming example
Extension Number Extension Name COS
101 Andrews 2
102 Barton 2
103 Cantor 2

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4.2.1 Multi-connection Networking Overview

Extension Number Extension Name COS

104 Dixon 1
: : :

3. In the COS settings, ensure that access to trunk group 3, as assigned in step 1, is not blocked for the
selected COS (COS 2, as assigned in step 2), but is blocked for all other COS.
→ 10.7.2 PBX Configuration—[2-7-2] System—Class of Service—External Call Block

Programming example
Outgoing Trunk Group
COS Number
1 2 3 4
1 x
2
3 x
4 x
: x

×：Blocked
Similarly, you can create a System Speed Dialling directory for a site by grouping all extensions at that
site as a tenant.
→ (5.1.3 Tenant Service)
Networks that span multiple time zones
The time zone is set independently for each PBX in a Multi-connection network. The PBX can acquire the
current time either automatically from the NSX Server Unit, an NTP (Network Time Protocol) server or caller
ID information over ISDN, or the time can be set manually.
Certain PBX features use the local PBX’s time information during operation, and other features use the NSX
Server Unit's time information. The following table lists which features use which time information:
Feature Operational permissions per site
Incoming Call Log Time, TAM Log Time Supported
ARS route plan Supported
Time Mode Supported
Timed reminder Supported
Holiday table Supported
Week table Not supported
Summer table Supported
Room status Not supported
Print message information Not supported
Billing information Not supported
Remote timed reminder Not supported
Other hotel related features Not supported
Call information Not supported

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 4.2.1 Multi-connection Networking Overview

Feature Operational permissions per site

Failure information Not supported
Compiled billing information Not supported
Invoice information for the built-in hotel function Not supported
External hotel application information Not supported
Login/Logout information Not supported
Wireless handover information Not supported
Master password Supported
Timed update Supported
Periodic diagnosis Supported
Time notification to LPR Supported
Waiting time display Supported
Display format (12H/24H etc.) Not supported
Waiting time display Supported
Memo alarm feature Supported
Activation Key expiration date Not supported
Error log Supported
SNMP, syslog Supported
Time stamp of various files (such as DCSYS) Supported
CTI application (1st party) Follow the computer time
CTI application (3rd party) Follow the computer time
Time notification to voice mail Supported
Read out loud of the voice mail mailbox recording time Follow the voice mail time
Other logs for debugging -

1. NSX server/Expansion Gateway

1-1 The NS series PBX with the NSX software installed in the Expansion Gateway is used as the
Expansion Gateway.
Orange light is installed by using the Web Maintenance Console of each NS system.
The NS system that installed the NSX Expansion Gateway software configures the settings for the
NSX server during the initial setup.
1-2 Start-up Mode of the Expansion Gateway
The NS series with NSX software installed will always start up as the Expansion Gateway.
1-3 Status Management of the Expansion Gateway
The LEDs of each Expansion Gateway (LED names are KX-NS1000: Master LED, KX-NS300/
KX-NS500/KX-NS700: PBX MODE LED) show the connection status to the NSX and the operational
status of the Expansion Gateway. It is not possible to identify the network attributes from the lighting
patterns of the system LED (System Status LED).
The status when connecting to the NSX Server is indicated by a flashing light and the status when
stable is indicated by a normal light. Additional information is represented by the 3 colours Green,
Orange and Red.

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4.2.1 Multi-connection Networking Overview

Attributes/status of the Network Status-LED

Remarks
Expansion GW lighting types
Normal operation Orange light Flashes Orange while connecting to the NSX
Server.
(*1)
Abnormal status Flashing RED light When no SD card is mounted to the NS300.
(240 times/minute) Transitions to normal operation after mounting a
SD card.
Waiting on the NSX RED light Does not differ from the Maintenance status.
Server to recover
ISOLATE status Flashing RED light Is connected to the NSX Server, however, a job
(60 times/minute) change is possible in the ISOLATE status if it
flashes RED
Maintenance status RED light Does not differ from the error recovery Master

*1 The Easy Setup Wizard will show an error when you use a KX-NS300 without a SD card mounted.

If the SD card gets dismounted during operation the "slave abnormality" error log will be output.
If detected that there is no SD card mounted to the KX-NS300 while restarting when Multi-connection is
operational, "Isolate" will be displayed and the error log will be migrated to Isolated when the Isolated
Mode settings are active or, "Slave (abnormality)" will be displayed and it will not be migrated to
Isolated when Isolated Mode is not enabled. (In both cases the "Slave abnormality" error log will be
output.)

Separately, in Web programming, information on the NSX server and Expansion Gateway (sites ID,
site names, IP address, etc.) can be displayed, and the model name of the Expansion Gateway can be
identified.

The identification information for each site will be displayed as below.

• Site ID (1– maximum number of integrated Expansion Gateways)
• Site name (example: branch office, 32 characters)
• Device status (such as OUS / INS)
• Expansion Gateway model name (such as KX-NS1000NE)
• NSX Server Expansion Gateway software version
• Software destination ※shown within the () of the unit name. (Example：KX-NS1000NE (SP))

Besides the above mentioned information, information such as separate stacking configurations and status
can also be displayed.
1-4 Program files of the NSX Server Unit/Expansion Gateway NSX Server/Expansion Gateway
Operated in the individual program files of each type of NSX Server or Expansion Gateway.

Conditions
[General]
• KX-NSXF007 (Activation Key for Expansion GW (NS)) is required for each PBX in the network to use this
feature.
• The types of PBXs that can be assigned as NSX Server Unit and Expansion Gateways are as follows:
– NSX Server Unit: KX-NSX1000/2000
– Expansion Gateway KX-NS1000, KX-NS700, KX-NS500, KX-NS300

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 4.2.1 Multi-connection Networking Overview

CAUTION
When changing from product version NS software to Expansion Gateway software, the existing
product version NS software has to be version 004.22000 or higher.
• A maximum of 31 Expansion Gateways can be registered to the NSX Server Unit.
• Tenants (→ 5.1.3 Tenant Service) and ICD groups (→ 2.2.2 Incoming Call Distribution Group Features)
can contain extensions from multiple sites, and therefore multiple time zones. For time-sensitive
operations, such as Time Service (→ 5.1.4 Time Service), system programming determines which PBX’s
clock is used for determining the current time.
→ 14.6 PBX Configuration—[6-6] Feature—Tenant
• Due to hardware constraints, the following resources are not shared between sites:
– Echo canceller
– OGM
– BGM
– VoIP resources
– PS handover
– Network time synchronisation signal
• It is recommended that Multi-connection networks be located in one country/area. Operation cannot be
guaranteed for Multi-connection networks that span multiple countries/areas.
• For information about which types of PBXs are available to use as Slave units in your country/area,
consult your dealer.
[Using KX-NS300/KX-NS500/KX-NS700/KX-NS1000 as Expansion Gateways]
• All units must have a DSP card installed.
• KX-NS300 PBXs must have an SD card installed.
• The Fax Server feature is supported only by KX-NS1000 PBXs (→ 3.1.3 Fax Server)
Therefore, the UM groups of non-KX-NS1000 PBXs should not be specified as the destination for faxes.
• The Activation Key needed for the Expansion Gateway is installed via the NSX server.
• The following features normally available on KX-NS300, KX-NS500, KX-NS700, and KX-NS1000 PBXs
are not available on a Multi-connection network:
– SVM (Built-in Simplified Voice Message)
– PT system programming
– Remote maintenance via ISDN or analogue modem
• The NS series cannot operate as a NSX Server Unit.
• The KX-TDA and the KX-TDE series can connect to NS series models as an Legacy GW, however, the
maximum amount of Legacy GW that can be connected is 31 gateways.
When using KX-NS700/KX-NS500/KX-NS300 as an Expansion Gateway, the number of integrated
expansion units (KX-NS720/KX-NS520/KX-NS320) is also added to the total.
→ 5.3 Legacy Device Connection
• The Expansion Gateway itself is not equipped with the UM and SVN features.
• The activation key needed for the Expansion Gateway is installed via the NSX Server.
• Expansion Gateways with an Analogue Remote Card can use remote maintenance through the CLI
command in the Simplified Isolated Mode only.
• The Legacy Gateway connection location for the Multi-connection Network is shown in the chart below.

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4.2.1 Multi-connection Networking Overview

Site 1 NSX Server Unit:

PBX
(KX-NSX1000/2000)

Private IP Network

Site 2 Expansion GW Site 3 Expansion GW

PBX PBX
(KX-NS1000) (KX-NS300/NS500/NS700)
Stack-M Exp-M

Exp-S1
Stack-S Stack-S
Exp-S2
Legacy Legacy
Gateway Gateway Exp-S3

[DSP card]
• KX-NSX1000/2000 can mount up to 3 mountable DSP cards (DSP-S/DSP-M/DSP-L).
• When implementing Multi-connection with KX-NS300, KX-NS500, KX-NS700 or KX-NS1000 as the site,
the resources use the DSP of those sites. Also, since the DSP is required when making a call between
locations with Legacy terminals, the DSP is essential during the configuration of Multi-connection.
(KX-NS300/KX-NS500/KX-NS700 is not installed in the first implementation of DSP)
• A DSP Card for NS series can be used with the KX-NSX1000/2000 models.

Installation Manual References

2.1.1 System Configurations
6.4.1 Multi-connection Network
6.4.3 Information about Expansion Gateways

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 4.2.2 Network Type Comparison

PC Programming Manual References

10.1.1 PBX Configuration—[2-1-1] System—Date & Time—Date & Time Setting
10.1.2 PBX Configuration—[2-1-2] System—Date & Time—SNTP / Daylight Saving

Feature Manual References

5.5.1 PC Programming

4.2.2 Network Type Comparison

Description
This section summarizes the main differences between the two methods for connecting PBXs over an IP
network: Multi-connection networking and H.323 (VoIP) QSIG networking. It also explains which situations
each type of network is suited for.

Overview of characteristics
Item Multi-connection Network H.323 QSIG Network
Connection type Private IP network Private IP network
Compatible PBX type ・KX-NS series • KX-NS series
• KX-NCP series
• KX-TDE series
• KX-TDA series
PBX relationship NSX server Unit – Expansion Gateway Independent
Programming and Unified: Settings made on the NSX Server Independent: Each PBX in the
configuration Units are automatically propagated to network must be programmed
Expansion Gateways Individually.
Maximum number of 32 (1 NSX Server Unit, 31 Expansion 512
nodes Gateways)
Minimum system Each one KX-NSXF006 (Activation Key for One V-IPGW card per PBX*1
requirements Expansion GW (Master)), KX-NSXF007
(Activation Key for Expansion GW(NS)) per
PBX

*1 For non-KX-NS1000 PBXs, requirements may differ.

When to choose a Multi-connection network

If the network will consist only of the NSX Server Unit and a private IP network is feasible, then a Multi-
connection network is probably the better option. A considerable amount of setup is done automatically by
the PBXs, so setup and maintenance is much simpler. All units can be programmed from one location. This
simple setup and programming means that the PBX network can be expanded with considerably less effort.
Extensions are managed so they behave as if they were all connected to one PBX, which eliminates the
need to manage information such as PBX access codes.
However, resource sharing also requires a lot of bandwidth, since the NSX Server Unit must constantly
monitor and manage the status of the Slave units and connected devices. Therefore, if bandwidth is a
concern, an H.323 QSIG network may be preferable.

When to choose an H.323 QSIG network

If the network will include non-KX-NS series PBXs (e.g., KX-TDE600), then an H.323 QSIG network is
necessary.

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4.2.3 Multi-connection Networking Survivability

An H.323 QSIG network is preferable if strict resource separation between sites is necessary. Although it is
possible to reserve certain resources for certain extension users in a Multi-connection network, the default is
to share resources. On the other hand, in a QSIG network, resources are not available to extension users of
other PBXs without explicit programming.
Programming and configuring an H.323 QSIG network is much more complex than a Multi-connection
network. It is also possible to connect a Multi-connection network to other PBXs via QSIG. In an H.323
QSIG network, the Multi-connection network appears as one PBX. The NSX Server Unit represents the
Multi-connection network.

Installation Manual References

6.4.1 Multi-connection Network
6.4.4 H.323 QSIG Network
6.4.5 Working with Multiple PBX Networks

4.2.3 Multi-connection Networking Survivability

4.2.3.1 Simplified Isolated Mode
The Survival Gateway feature (Simplified Isolated Mode) is a feature that activates when the Expansion
Gateway (hereafter referred to as EXP-GW) enters an isolated state due to an operation halt of the NSX or
due to a network error between the sites.
This feature is a limited service for actions such as sending and receiving calls and is operational for 30
days after transitioning to Simplified Isolated Mode.

The operation of Simplified Isolated Mode is as follows.

NSX NSX

Normal Mode Simplified Isolated Mode

(Normal Operation)

Expansion GW Expansion GW

[Example]
During an NSX operation halt or a network error between the sites, the communication means of each site
are secured.

Description
1. Compatible EXP-GW
Below, the EXP-GW models applicable to the Survival Gateway feature (Simplified Isolated Mode) are
shown.
Gateway type Applicable Remarks
1 KX-NS1000 ü Including when connecting to the Legacy GW
2 KX-NS700 ü Including when connecting to an Expansion Unit
3 KX-NS500 ü Including when connecting to an Expansion Unit

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 4.2.3 Multi-connection Networking Survivability

Gateway type Applicable Remarks

4 KX-NS300 ü Including when connecting to an Expansion Unit

2. Simplified Isolated Mode Settings

To use Simplified Isolation Mode, the operational settings of Simplified Isolation Mode need to be
configured beforehand. The during normal operation (Normal Mode) of the EXP-GW, it operates on
system data of the NSX, while it operates on system data specific for the used mode during Simplified
Isolated Mode.
Accordingly, the settings for Simplified Isolation Mode need to be configured individually.
The Simplified Isolation Mode becomes the setup (some functions are not available) for the NS series,
and is not affected by the User Container configuration.
See → 2.29 User Container

The individual setting of Simplified Isolated Mode can be carried out by any of the following.
a. Initial Settings
When using the Easy setup wizard of the EXP-GW, the system data files for Simplified Isolation
Mode are automatically affected by the IP address and network settings of the NSX.
b. Change Settings
The Web settings of the EXP-GW can also be changed during Simplified Isolation Mode.
The import of system data files through file transfer is possible.
c. Settings of the Offline Settings Tool
Settings for Simplified Isolated Mode can be created and adjusted by use of the NS series Offline
Tool. The created DxSYS files can be imported into the EXP-GW through the file transfer feature.

Note
Software file version 4.2 or 7.0 of the system data file (DxSYS) for KX-NS series PBXs can be
used in an EXP-GW operating in Simplified Isolated Mode.

[Regarding System Data Files (DxSYS files)]

The system data files for Simplified Isolation Mode are different from the system data files (DxSYS) for
the EXP-GW (Normal Mode). The file name of the system data files for Simplified Isolation Mode are
the same as those of the NS series as shown in the table below.
Simplified Isolation Mode system
GW type Expansion GW system data file name
data file name
KX-NS1000 DCSYS_EXPGW DCSYS
KX-NS700 DFSYS_EXPGW DFSYS
KX-NS500/300 DFSYS_EXPGW DFSYS

3. Activation Key
An Activation Key is not required to use Simplified Isolated Mode.
During Simplified Isolation Mode the following Activation Keys will become licence-free. However, the
licence will only be valid for a period of 30 days.
Activation keys that have been installed and pre-installed during normal operation cannot be activated
again. Therefore the Activation Keys cannot be used during Simplified Isolation Mode.
Listed below are features that do not require a licence.
Model number Features Remarks
KX-NSM099 IP terminal capacity expansion (up to Only applicable for KX-NS1000
640)

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4.2.3 Multi-connection Networking Survivability

Model number Features Remarks

KX-NSM102/108/116 IP outside line integration channel
expansion
KX-NSM201/205/210/220 IP Softphone, IP proprietary terminal
integration
KX-NSM501/505/510/520 IP proprietary terminal integration
KX-NSM701/705/710/720 General-purpose SIP terminal
integration
KX-NSE201/205/210/220 IP-CS channel expansion
KX-NSF990 VoIP/UM feature (only for KX-NS300) Only applicable for KX-NS300
Note: Only the VoIP feature
can be used. The UM feature
cannot be activated.
KX-NSF991 Expansion Unit connection Only applicable for KX-NS700
KX-NSN101 Built-in router (SIP outside line Only applicable for KX-NS1000
integration use)

4. Simplified Isolated Mode Operation Overview

• The EXP-GW enters Simply Isolated Mode when the events described in 4-2-2 are triggered.
At that time, it becomes operational by itself by using the system data files configured for Simplified
Isolated Mode.
• Simplified Isolation Mode operation is limited to 30 days. Turning back your system time will not
reduce the number of operated days.
• To transition to Simplified Isolation Mode, Simplified Isolation Mode operation settings must be
enabled in the NSX settings. (Initial value: operation disabled)
• The EXP-GW transitions to Simplified Isolated Mode in the following cases.
a. When the communication with the NSX is interrupted for a certain period during the EXP-GW
start-up
b. When the communication with the NSX is interrupted for a certain period when operating as an
EXP-GW within Multi-connection Networking
c. When transitioned from the NSX Web Console to Simplified Isolated Mode
• You can return to Normal Operation Mode by logging on to the NSX Web Maintenance Console and
performing the release operation after restoring the communication to the NSX.
You can specify the desired activation time and date when setting up the release. These time and
date settings are cleared when the system restarts.
• During Simplified Isolated Mode, when the communication with the NSX is confirmed, the release
operation of Simplified Isolation Mode is performed in order to transition to Normal Operation Mode
(Initial value: Disabled)
• An EXP-GW in Simplified Isolated Mode can periodically be returned to Normal Operation Mode at
previously specified times.
Note that Simplified Isolation Mode will be released and the 30 day period will be reset when
changing to Normal Operation Mode.
• When Simplified Isolation Mode is used for longer than 30 days, all services except for accessing
the Web Settings, connecting to the NSX, and Recovery Monitoring are stopped.

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<Limitation of Simplified Isolated Mode operation>

The new features of NSX series such as LDAP directory service are not supported for the EXP-GW.
In addition, the following features are not available in the Simplified Isolated Mode.
– Built-in UM (including Auto 2-way recording, 2-way recording, FAX)
– CTI Interface (3rd party CSTA)
– Mobile Softphone (KX-UCMA)
– Built-in ACD monitor/report
– Remote maintenance by using the KMS server and Panasonic cloud
System log data such as SMDR, Call Log etc. are not merged with the NSX call server when the
Simplified Isolated Mode is back to Normal Mode. The customised flexible button on the PT
shows the mode status.
• If it entered Simplified Isolation Mode by providing the Isolated Mode Notification feature, the
following notification means can notify the state to the user or installer.
1. Mode indication according to the Networking Status LED
During Simplified Isolated Mode, the Networking Status LED (LED names are KX-NS1000:
Master LED, KX-NS300/KX-NS500/KX-NS700: PBX MODE LED) reports the state of the EXP-
GW to the user.
Modes of the expansion GW Network-Status LED lighting types Remarks
Normal operation status Orange light
Simplified Isolated Mode RED light

2. Mode indication according to the Alarm Key

When transitioning to Simplified Isolated Mode, the Alarm key that was set to FF key of the
IP-PT or the DPT beforehand lights up red, and notifies the operation of Simplified Isolated
Mode to the user.
When the user presses the Alarm key, the Operation Mode will be shown on the telephone's
LCD according to the following table.
LCD when pressing
Number System operation mode LED display Remarks
down the Alarm Key
1 Normal operation OFF Normal Mode
2 Simplified Isolated Mode R-ON Isolated Mode
3 Other than 1 and 2 OFF Nothing displayed

Note 1: This is an example, for details see the LCD language data specification.
Note 2: The recover time of the LED of the alarm key is the same as that of conventional Alarm
key.

Note that the relation to the Local Alarm feature (local Alarm) is as follows.
This item is only compatible with the Alarm Key Settings and extensions configured with the
local alarm feature. You can display the operation status by pressing the Alarm Key during
standby.

3. Error Log and E-mail notification

When transitioning to Simplified Isolated Mode or when releasing Simplified Isolated Mode, it is
possible to store the error information as an error log.
At that time the details can be notified by E-mail.

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Conditions
• When transitioning between Normal ⇔ Simplified Isolation Mode, accumulated data such as SMDR data
and Call Log data will not be transferred.
• Firmware updates of various cards installed to the GW or Legacy GW can also be performed from the
Web Console during Simplified Isolation Mode.
• User management (adding, deleting, configuration) is disabled.
However, INSTALLER Level IDs and Passwords can be configured.
• Each device must be set up in order to make and receive calls.
• You are only able to view the status of the Activation Key. Installing or deleting an Activation Key is not
possible.
• A Sub-Administrator account cannot log in to EXP-GW.

PC Programming Manual References

1.2.3.1 Multi-connection Networking Survivability

Feature Manual References

6.3 Features that Require Activation Keys

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4.3 Private Network Features

4.3.1 TIE Line Service
Description
A TIE line is a privately leased communication line between two or more PBXs, which provides cost effective
communications between company members at different locations.
Interface
The following interfaces can be used to establish a private network:
Interface Network Type
VoIP (H.323) Internet Protocol (IP)
BRI/PRI (QSIG) Digital (ISDN 2B+D/30B+D/23B+D)
E&M Analogue
T1 (TIE [E & M]) Digital (64 kbps × 24 channels)
E1 (E & M) Digital (64 kbps × 30 channels)

The available interfaces depend on the cards installed in the PBX.

→ 9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Network Numbering Plan
—Trunk Property
→ 9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Network Numbering Plan
—Trunk Property

Conditions
• A TIE line connection can be established through a Trunk Adaptor using a PRI line (QSIG).
• For details about establishing a TIE line connection over an E & M, T1, or E1 interface, refer to the
documentation of the legacy gateway through which the connection will be made.

Feature Manual References

2.1.1.6 Intercept Routing
2.3.2 Call Forwarding (FWD)
2.7.1 Toll Restriction (TRS)/Call Barring (Barring)
2.8.1 Automatic Route Selection (ARS)
2.12.1 Call Transfer
2.16.1 Direct Inward System Access (DISA)
5.5.6 Flexible Numbering/Fixed Numbering
6.1 Capacity of System Resources

Operating Manual References

1.2.1 Basic Calling

4.3.1.1 Making a TIE Line Call

Description
One of the following two methods can be used to make a TIE line call.
<Extension Number Method (Access without PBX Code)>
Dial the [Extension Number] only.

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[Example]

PBX-1 PBX-2 PBX-3

TIE Line TIE Line

Interface Interface Interface Interface

Extn.1011 Extn.1012 Extn. 2011 Extn. 3011

Dials "3011". Dials "2011".

Explanation:
To use this method, it is necessary to change the first one or two digits of extension numbers of either PBX
(e.g., 10XX for PBX-1, 20XX for PBX-2) to allow calls to be routed properly.
Case 1:
Extension 1012 of PBX-1 dials extension number "2011".
→ Extension 1012 of PBX-1 is connected to extension 2011 of PBX-2.
Case 2:
Extension 1011 of PBX-1 dials extension number "3011".
→ Extension 1011 of PBX-1 is connected to extension 3011 of PBX-3.

<PBX Code Method (Access with PBX Code)>

Dial the [TIE Line Access Number] + [PBX Code] + [Extension Number].
→ 10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features— TIE Line Access
→ 17.1 PBX Configuration—[9-1] Private Network—TIE Table—Own PBX Code

[Example]
PBX-1 PBX-2 PBX-3
PBX Code 951 PBX Code 952 PBX Code 953

TIE Line TIE Line

Interface Interface Interface Interface

Extn.1011 Extn.1012 Extn. 1011 Extn. 1011

Dials "7-953-1011". Dials "7-952-1011".

[PBX code]

[TIE line
access no.]

[Extn. no.]

Explanation:
To use this method, it is necessary to know each PBX code in order to identify the location of an extension.
Case 1:

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Extension 1012 of PBX-1 dials TIE line access number "7", PBX code "952", and extension number "1011".
→ Extension 1012 of PBX-1 is connected to extension 1011 of PBX-2.
Case 2:
Extension 1011 of PBX-1 dials TIE line access number "7", PBX code "953", and extension number "1011".
→ Extension 1011 of PBX-1 is connected to extension 1011 of PBX-3.

4.3.1.2 TIE Line and Trunk Connection

Description
To connect the TIE line with the trunk, the following patterns are available:
1. Trunk-to-TIE Access
2. TIE-to-Trunk Access
3. Trunk-to-TIE-to-Trunk Access
Trunk-to-TIE Access
It is possible to assign an extension of another PBX as the destination of incoming trunk calls to the own
PBX.
It is also possible to forward calls using a Public device name. Using this method, trunk calls received at
PBX-1 are forwarded directly to the extension at PBX-2, even when using the PBX Code method.
a. Incoming Trunk Call Destination Assignment

[Example]

Telephone Company

Trunk
TIE Line Network
PBX-1
PBX-2
Trunk DID No: 4567
Destination: 2011
TIE Line
Interface Interface

Outside Caller

Dials "123-4567". Extn. 2011

Extn. 1011

Explanation:
An outside caller dials "123-4567". The call is sent to extension "2011" of PBX-2 through the TIE line
according to the assignment of the DID call destination of PBX-1. (→ 2.1.1.3 Direct Inward Dialling
(DID)/Direct Dialling In (DDI))
→ 18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—DDI / DID
b. FWD/Call Transfer/Intercept Routing to the TIE Line

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[Example]

Telephone Company

Trunk TIE Line Network

PBX Code: 951 PBX Code: 952
Trunk PBX-1 PBX-2
TIE Line
Interface Interface

Forwarded/Transferred
/Intercepted to 7-952-2011

Outside Caller
Extn. 1011 Extn. 2011
Dials "123-4567".

Explanation:
An outside caller dials "123-4567". The call reaches the destination (extension 1011 of PBX-1), and the
call is forwarded, transferred, or intercepted to extension "2011" of PBX-2 through the TIE line.
TIE-to-Trunk Access
The PBX sends TIE line calls to the trunks of another PBX through the TIE lines.
a. Trunk Call through Other PBXs

[Example]

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<Extension Number Method (Access without PBX Code)>

Telephone Company

Trunk 211-4567 Trunk

TIE Line Network

PBX-1 PBX-2
9-211-4567

Interface
TIE Line Interface
TRG 2

Outside Party
Extn. 1011 Extn. 2011 (211-4567)

Dials "802-9-211-4567".

Explanation:
1. Extension 1011 of PBX-1 dials the Trunk Group Access number of PBX-1 "8", trunk group number
"02" (TRG2), Idle Line Access number of PBX-2 "9", and telephone number "211-4567".
→ 10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Trunk Group
Access
2. PBX-1 sends the call to PBX-2 through the trunk group (TRG) 2 (TIE line).
3. PBX-2 sends the call to the outside party "211-4567".

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4.3.1 TIE Line Service

<PBX Code Method (Access with PBX Code)>

Telephone Company

Trunk 211-4567 Trunk

TIE Line Network

PBX-1 PBX-2
PBX Code 951 PBX Code 952
952-9-211-4567

Interface
TIE Line Interface
TRG 2

Outside Party
Extn. 1011 Extn. 1011 (211-4567)

Dials "7-952-9-211-4567" or
"802-952-9-211-4567".

Explanation:
1. Extension 1011 of PBX-1 dials the TIE line access number "7", PBX code "952", Idle Line Access
number of PBX-2 "9", and telephone number "211-4567"; or dials the Trunk Group Access
number of PBX-1 "8", trunk group number "02" (TRG2), PBX code "952", Idle Line Access
number of PBX-2 "9", and telephone number "211-4567".
2. The call is connected to the outside party "211-4567" through PBX-2 which has PBX code "952".
Trunk Call through Other PBXs—by the ARS feature

[Example]

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 4.3.1 TIE Line Service

<Extension Number Method (Access without PBX Code) using ARS>

Telephone Company

Trunk 211-4567 Trunk

TIE Line Network

PBX-1 PBX-2
9-211-4567

Interface
TIE Line Interface
TRG 2

Outside Party
Extn. 1011 Extn. 2011 (211-4567)

Dials "9-211-4567".

Explanation:
1. Extension 1011 of PBX-1 dials the Idle Line Access number of PBX-1 "9" and telephone number
"211-4567".
2. PBX-1 modifies the call (adds the Idle Line Access number of PBX-2 "9") and sends the call to
PBX-2 through the TIE line (trunk group [TRG] 2) according to the ARS programming of PBX-1.
3. PBX-2 sends the call to the outside party "211-4567".

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4.3.1 TIE Line Service

<PBX Code Method (Access with PBX Code) using ARS>

Telephone Company

Trunk 211-4567 Trunk

TIE Line Network

PBX-1 PBX-2
PBX Code 951 PBX Code 952
952-9-211-4567

Interface
TIE Line Interface
TRG 2

Outside Party
Extn. 1011 Extn. 1011 (211-4567)

Dials "9-211-4567".

Explanation:
1. Extension 1011 of PBX-1 dials the Idle Line Access number of PBX-1 "9" and telephone number
"211-4567".
2. PBX-1 modifies the call (adds "952" and the Idle Line Access number of PBX-2 "9") and sends the
call to PBX-2 which has PBX code "952" through the TIE line (trunk group [TRG] 2) according to
the ARS programming of PBX-1.
3. PBX-2 sends the call to the outside party "211-4567".

b. Blocking trunk calls made through another PBX and how to override it:
Whether an incoming TIE line call can make a trunk call through this PBX (i.e., PBX-2), depends on the
COS that is assigned to the trunk group of this PBX, that the incoming TIE line is connected to. If the
COS of the trunk group is unable to make outgoing calls by the Toll Restriction/Barring feature or
External Call Block feature, trunk calls made through this PBX will be prohibited.
To override this prohibition, an extension of PBX-1 must enter a verification code assigned to PBX-2 to
change the COS temporarily. It is also possible to override the prohibition by specifying an extension at
PBX-2 with the Walking COS feature, to temporarily switch to that extension’s COS.
→ 10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—TRS—TRS Level—
Day, Lunch, Break, Night
→ 10.7.2 PBX Configuration—[2-7-2] System—Class of Service—External Call Block
→ 11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Main—COS
→ 14.3 PBX Configuration—[6-3] Feature—Verification Code

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[Programming Example of PBX-2]

Trunk Group No. COS No.

1 3
2 2
3 2
: :

TRG of Incoming Outgoing Call

Call TRG 1 TRG 2 TRG 3 …

COS 1
COS 2
COS 3
: : : : :

: Block

[Example]
<Extension Number Method (Access Without PBX Code)>

Telephone Company

Trunk 211-4567 Trunk

TRG 3
TIE Line Network of PBX-2

PBX-1 PBX-2
9-211-4567
TRG 1 (COS 3)
TIE Line of PBX-2
Interface Interface
TRG 2
of PBX-1
verification code entry feature no.
+ + verification code + verification code
PIN + 9-211-4567

Outside Party
(211-4567)
Extn. 1011 Extn. 1012 Extn. 2001

Dials "8-02-9-211-4567". Dials "8-02+verification code entry feature

no. + + verification code + verification code
PIN + 9-211-4567".

Explanation:

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Case 1:
1. Extension 1011 of PBX-1 dials the Trunk Group Access number of PBX-1 "8", TIE line trunk group
number (TRG 2), Idle Line Access number of PBX-2 "9", and the telephone number "211-4567".
2. The call is not connected to the outside party through PBX-2 because the COS of TRG 1 (COS 3)
is blocked from accessing TRG 3 of PBX-2.
Case 2:
1. Extension 1012 of PBX-1 dials the Trunk Group Access number of PBX-1 "8", TIE line trunk group
(TRG2), verification code entry feature number, , verification code, verification code personal
identification number (PIN), Idle Line Access number of PBX-2 "9", and the telephone number
"211-4567".
2. If the specified verification code applies COS 2 of PBX-2, the call is connected to the outside party
through PBX-2, because COS 2 is not blocked from accessing TRG 3 of PBX-2.

<PBX Code Method (Access with PBX Code)>

Telephone Company

Trunk 211-4567 Trunk

TRG 3
TIE Line Network of PBX-2

PBX-1 PBX-2
PBX Code 951 PBX Code 952
952-9-211-4567
TRG 1 (COS 3)
TIE Line of PBX-2
Interface Interface
TRG 2
of PBX-1
952+verification code entry feature no.
+ + verification code + verification code
PIN + 9-211-4567

Outside Party
(211-4567)
Extn. 1011 Extn. 1012 Extn. 1001

Dials "7-952-9- Dials "7-952+verification code entry feature

211-4567". no. + + verification code + verification code
PIN + 9-211-4567".

Explanation:
Case 1:
1. Extension 1011 of PBX-1 dials the TIE line access number "7", PBX code "952", Idle Line Access
number of PBX-2 "9", and the telephone number "211-4567".
2. The call is not connected to the outside party through PBX-2 because the COS of TRG 1 (COS 3)
is blocked from accessing TRG 3 of PBX-2.
Case 2:

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1. Extension 1012 of PBX-1 dials the TIE line access number "7", PBX code "952", verification code
entry feature number, , verification code, verification code personal identification number (PIN),
Idle Line Access number of PBX-2 "9", and the telephone number "211-4567".
2. If the specified verification code applies COS 2 of PBX-2, the call is connected to the outside party
through PBX-2, because COS 2 is not blocked from accessing TRG 3 of PBX-2.

c. Override using an Itemised Billing Code for ARS

By assigning an Itemised Billing Code for ARS to PBX-1, an extension’s verification code can be sent
to PBX-2 automatically, without the extension having to dial the verification code.
→ 8.2.1 Users—Advanced Extension Settings—Advanced Extension Settings—Option 1—ARS
Itemised Code
→ 16.5 PBX Configuration—[8-5] ARS—Carrier

[Example]
<Extension Number Method (Access without PBX Code)>

[Programming Example of PBX-1]

Extn. No. Itemised Billing Code

1012 11112222
1013 33334444

Carrier Name TIE Line Telephone Company

Trunk Group 2
Carrier Access Code 7
Removed No. of Digits
Trunk 211-4567 Trunk
0
Modify Command CI9H
TRG 3
of PBX-2
TIE Line Network

PBX-1 PBX-2 Outside Party

(211-4567)
TRG 1 (COS 3)
TIE Line of PBX-2
Interface Interface
TRG 2
of PBX-1 [Programming Example
verification code entry feature no. of PBX-2]
+ + verification code + verification code Flexible Numbering
PIN+9-211-4567
TIE: 7 (Default) None
Verification code:
*47 (Default) 7
Extn. 1011 Extn. 1012 Extn. 2001 Veri. Code Veri. PIN COS
1111 2222 2
Dials "9-211-4567".
3333 4444 2

[SMDR Output Example]

Date Time Ext CO Dial Number Ring Duration
06/04/18 03:21PM 1001 03 2114444 00:01'23
06/04/18 04:32PM *1111 04 2114567 00:23'45

Explanation:

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1. Extension 1012 of PBX-1 dials the Idle Line Access number of PBX-1 "9", and the telephone
number "211-4567".
2. PBX-1 modifies the call (adds the verification code entry feature number [you must change the
feature number from its default value so that it not start with an asterisk (*)], verification code and
verification code PIN, and the Idle Line Access number of PBX-2 "9") and sends the call to PBX-2
through the TIE line (trunk group [TRG] 2) according to the ARS programming of PBX-1.

<PBX Code Method (Access with PBX Code)>

[Programming Example of PBX-1]

Extn. No. Itemised Billing Code

1012 11112222
1013 33334444

Carrier Name TIE Line Telephone Company

Trunk Group 2
Carrier Access Code 952 47
Removed No. of Digits Trunk 211-4567 Trunk
0
Modify Command CI9H
TRG 3
TIE Line Network of PBX-2

PBX-1 PBX-2
PBX Code 951 PBX Code 952 Outside Party
(211-4567)
TRG 1 (COS 3)
TIE Line of PBX-2
Interface Interface
TRG 2
of PBX-1 [Programming Example
952+verification code entry feature no. of PBX-2]
+ + verification code + verification code
PIN+9-211-4567
Veri. Code Veri. PIN COS
1111 2222 2
3333 4444 2

Extn. 1011 Extn. 1012 Extn. 1001

Dials "9-211-4567".

[SMDR Output Example]

Date Time Ext CO Dial Number Ring Duration
06/04/18 03:21PM 1001 03 2114444 00:01'23
06/04/18 04:32PM *1111 04 2114567 00:23'45

Explanation:
1. Extension 1012 of PBX-1 dials the Idle Line Access number of PBX-1 "9", and telephone number
"211-4567".
2. PBX-1 modifies the call (adds "952", the verification code entry feature number, verification code
and verification code PIN, and the Idle Line Access number of PBX-2 "9") and sends the call to
PBX-2 which has PBX code "952" through the TIE line (trunk group [TRG] 2) according to the
ARS programming of PBX-1.

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d. FWD/Call Transfer/Intercept Routing to the Trunk

[Example]

Telephone Company

Forwarded/Transferred/
Trunk Intercepted to 211-4567 Trunk
TIE Line Network
PBX-1 PBX-2
PBX Code 951 PBX Code 952
952-1011

TIE Line
Interface Interface

Outside Party
Extn. 1011 Extn. 1011 (211-4567)

Dials "7-952-1011".

Explanation:
1. Extension 1011 of PBX-1 dials the TIE line access number "7", PBX code "952", and extension
number "1011".
2. The call reaches the destination (extension 1011 of PBX-2) through the TIE line, and the call is
forwarded, transferred or intercepted to the outside party "211-4567" through the trunk.
Trunk-to-TIE-to-Trunk Access
An outside caller can be connected to an outside party through the TIE line by using the DISA feature.

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4.3.1 TIE Line Service

[Example]

Telephone Company Telephone Company

(area code: 09) (area code: 01)

Trunk Trunk Trunk 23-4567 Trunk

TIE Line Network

PBX-1 PBX-2
PBX-Code 951 PBX-Code 952
952-9-01-23-4567

DISA Interface TIE Line Interface

TRG 2

Outside Caller Outside Party

(23-4567)
Dials "(DISA phone
number)-9-01-23- Extn. 1011 Extn. 1011
4567".

Explanation:
1. The outside caller dials the "DISA phone number of PBX-1", Idle Line Access number of PBX-1 "9",
and telephone number "01-23-4567".
2. PBX-1 modifies the call (adds "952" and the Idle Line Access number of PBX-2 "9") and sends the call
to PBX-2 which has PBX code "952" through the TIE line (trunk group [TRG] 2) according to the ARS
programming of PBX-1.
3. PBX-2 sends the modified call to the outside party "23-4567" according to its ARS programming.

4.3.1.3 TIE Line Programming

Description
To Make a TIE Line Call
The TIE Line Routing and Modification Table is referenced by the PBX to identify the trunk route when an
extension user makes a TIE line call.
It is necessary to make unified tables with all PBXs in the TIE line network.
The routing pattern appropriate for each call is decided by the dialled number.
There are two system programmes for the tables:
TIE Line Routing Table: used to assign the leading numbers (PBX code or extension number) and trunk
group hunt sequence.
→ 17.1 PBX Configuration—[9-1] Private Network—TIE Table—Priority 1–Priority 8—Leading Number
TIE Modify Removed Number of Digits/Added Number: used to remove digits from and add a number to
the dialled number of the TIE line call. This modification may be needed depending on the TIE line network
configuration.
17.1 PBX Configuration—[9-1] Private Network—TIE Table—Priority 1–Priority 8
→ Removed Number of Digits
→ Added Number
→ Trunk Group
[Programming Examples]

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Your PBX is PBX-1 and there are four PBXs in your TIE line network. To identify the trunk route as
illustrated, you should make the following tables.
a. Extension Number Method (Access without PBX Code)
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Other PBX Extension—Dial
TIE Line Network

PBX-4 PBX-3

Extn. 4xxx Extn. 3xxx

b-2nd) 3xxx
If you dial: TRG 2 c) 4xxx
a) 2xxx
b) 3xxx TRG 1
c) 4xxx
Extn. 1xxx a) 2xxx Extn. 2xxx
(2, 3, 4: Other PBX
Extension Number
b-1st) 3xxx
[TIE] in the Flexible
PBX-1 PBX-2
Numbering Plan)

[TIE Line Routing and Modification Table of PBX-1]

Priority 1 Priority 2 ..
Dial Modification Dial Modification ..
Location Leading
No. No. TRG Removed TRG Removed
Added Added
No. of No. of ..
No. No.
Digits Digits
01 2 1 0 ..
02 3 1 0 2 0 ..
03 4 2 0 ..
: : : : : : : : :

Explanation:
Location 01:
The hunt sequence by dialling [2XXX]:
The 1st route—trunk group (TRG) 1 Sending no. to PBX-2: 2XXX
Location 02:
The hunt sequence by dialling [3XXX]:
The 1st route—trunk group (TRG) 1 Sending no. to PBX-2: 3XXX
The 2nd route—trunk group (TRG) 2 Sending no. to PBX-4: 3XXX
Location 03:
The hunt sequence by dialling [4XXX]:
The 1st route—trunk group (TRG) 2 Sending no. to PBX-4: 4XXX

b. PBX Code Method (Access with PBX Code)

→ 10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—TIE Line Access

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4.3.1 TIE Line Service

→ 17.1 PBX Configuration—[9-1] Private Network—TIE Table—Own PBX Code

TIE Line Network
PBX-4 PBX-3
PBX Code 954 PBX Code 953

Extn. xxxx Extn. xxxx

b-2nd) 953#-xxxx
If you dial: TRG 2 c) 954#-xxxx
a) 7-952-xxxx
b) 7-953-xxxx TRG 1
c) 7-954-xxxx
Extn. 1xxx a) 952-xxxx Extn. xxxx
(7: TIE Line Access
Number in the b-1st) 953-xxxx
Flexible PBX-1 PBX-2
Numbering Plan) PBX Code 951 PBX Code 952

[TIE Line Routing and Modification Table of PBX-1]

Priority 1 Priority 2 ..
Dial Modification Dial Modification ..
Location Leading
No. No. TRG Removed TRG Removed
Added Added
No. of No. of ..
No. No.
Digits Digits
01 952 1 0 ..
02 953 1 0 2 3 953# ..
03 954 2 3 954# ..
: : : : : : : : :

Explanation:
Location 01:
The hunt sequence by dialling [7+PBX Code 952+XXXX]:
The 1st route — trunk group (TRG) 1
Sending no. to PBX-2: 952–XXXX
Location 02:
The hunt sequence by dialling [7+PBX Code 953+XXXX]:
The 1st route — trunk group (TRG) 1
Sending no. to PBX-2: 953–XXXX
The 2nd route — trunk group (TRG) 2
Sending no. to PBX-4: 953#–XXXX
Location 03:
The hunt sequence by dialling [7+PBX Code 954+XXXX]:
The 1st route — trunk group (TRG) 2
Sending no. to PBX-4: 954#–XXXX
To Receive a TIE Line Call
a. Extension Number Method (Access without PBX Code)

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[Example] 1 A TIE line call is sent to

PBX-2 from PBX-1. If
TIE Line Network
the number sent from
PBX-4 PBX-3 PBX-1 is an extension
3 number of PBX-2 (e.g.,
2011), the call will be
received at extension
Extn. 3011
"2011". If not, PBX-2
checks the number in
2 3011 the TIE Line Routing
and Modified Table of
1 3011 PBX-2.
2 If the match is found in
Extn. 1011 Extn. 2011 the table, the call will be
modified according to
Dials "3011". PBX-1 PBX-2 the table and send to the
corresponding PBX
(PBX-3).
3 The number sent from
PBX-2 "3011" is an
extension number of
PBX-3. The call is
received at extension
"3011".

Note
When a TIE line call is sent from one PBX to another, the receiving PBX first modifies the received
number according to the assignment for the trunk port: the number of digits removed, and the
number added, are determined by this assignment. Then the PBX checks whether the completed
number is an existing extension number at that PBX.

b. PBX Code Method (Access with PBX Code)

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[Example] 1 A TIE line call is sent

to PBX-2 from PBX-1.
TIE Line Network
If the number sent
PBX-4 PBX-3
PBX Code 954 PBX Code 953 from PBX-1 has the
3
PBX code of PBX-2
"952", the call will be
received at the
Extn. 1011
corresponding
extension of PBX-2
2 953-1011
(e.g., 1011 of PBX-2).
If not, PBX-2 checks
the number in the TIE
1 953-1011
Line Routing and
Modified Table of
Extn. 1011 Extn. 1011 PBX-2.
PBX-1
Dials "7-953-1011". PBX Code 951
PBX-2 2 If the match is found
PBX Code 952
in the table, the call
will be modified
according to the table
and send to the
corresponding PBX
(PBX-3).
3 The number sent from
PBX-2 "953-1011" has
the PBX code of
PBX-3 "953". The call
is received at
extension "1011" of
PBX-3.

Note
When a TIE line call is sent to a PBX from another PBX, first the PBX modifies the number sent to
the PBX according to the assignment for each trunk port of the PBX: the removed number of digits
from and/or added number to the number sent to the PBX is determined by the assignment. Then
the PBX starts to check the number whether the number has the PBX code of the PBX.

TIE Line Routing Flowchart

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[Making a TIE Line Call from an Extension]

A TIE line call is made as follows:

PBX Code Method: 7-abc-xxxx
Extension No. Method: dexx

Is the dialled number

identified as a TIE line access no. No Not treated as
or an other PBX extension no. in
a TIE line call.
the flexible numbering
plan of the own PBX?

Yes:
A TIE line access no.: 7
Other PBX extension no.: de

Is the leading
No
number (abc or de) found in the TIE Line Reorder tone
Routing and Modification Table
of the own PBX?

Yes

Selects the corresponding trunk group, and

the dialled number is modified if a removed
number of digits and/or added number is assigned.

No
Is the trunk group Reorder tone
available?

Yes

No
Is there an idle trunk Busy tone
in the trunk group?

Yes

Routes to other PBX or trunk.

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4.3.1 TIE Line Service

[Receiving a Call through a TIE Line]

<Extension Number Method <PBX Code Method
(Access without PBX Code)> (Access with PBX Code)>
A call is received through a A call is received through a
TIE line as follows: TIE line as follows:
# 1021 ## 0511033

The received number is modified as The received number is modified as

programmed for each trunk port. programmed for each trunk port.

Removed number of digits: 1 Removed number of digits: 3

Added number: None Added number: 9
Received number: # 1021 Received number: ## 0511033

Modified number: # 1021=1021 Modified number: ##0511033= 9511033

Remove the first 1 digit. 1) Remove the first 3 digits. 2) Add "9".

Does the modified number have No

the own PBX code "951"?

Yes: 9511033 Goes to A

Removes the own PBX ( A is in the
code "951". flowchart of [Making
a TIE Line Call from an
1021 1033
Extension].)
Checks the modified number with the flexible numbering plan of the own PBX.

Operator Call No. Extension No. Extension No. Idle Line Others
of the Own PBX of Other PBX Access No.
Directs the call or Sends reorder tone,
Goes to A Trunk Group
to the operator. or sends the call to
( A is in the flowchart of Access No. the operator
Does the corresponding No [Making a TIE Line Call (Intercept Routing
extension exist? from an Extension].) —No Destination).
Is the trunk
Yes group of the outgoing
call from the own PBX enabled No
against the COS of the trunk group of
No Is the corresponding the incoming call
extension idle? to the own PBX?
Reorder
Yes Sends reorder tone
Yes
tone, or sends the
Calls the extension. call to the operator TRS/Barring applies.
(Intercept Routing
Call Waiting —No Destination).
Busy tone Sends the call to
Intercept Routing the trunk.
—Busy/DND

Conditions
• A trunk which is used for a private network should be assigned "Private" as the networking type. (→
2.1.1.1 Incoming Trunk Call Features—SUMMARY)

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• To establish a QSIG network (→ 4.3.4 QSIG Standard Features), each ISDN (QSIG) connection in a TIE
line network must have the port on one PBX assigned as a master port, and the port on the other PBX
assigned as a slave port. PBXs that support this feature are KX-NSX series, KX-NS1000 PBXs, and
KX-NCP series, KX-TDE series, and KX-TDA series PBXs.

[TIE Line Network Connection Example]

Extn.1000 PBX-1 PBX-2 Extn. 2000

(A) Master (A) Slave
: :
Extn.1999 Extn. 2999
QSIG Network
(B) Master (C) Slave

(B) Slave (C) Master

PBX-3 Extn. 3000

:
Extn. 3999

• When a TIE line call arrives at a busy extension which has disabled Call Waiting, the caller will hear a
busy tone. If required, Intercept Routing—Busy/DND can be activated.
• The Inter-digit time can be assigned for TIE line calls.
→ 10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—Incoming Call Inter-digit
Timer—TIE

PC Programming Manual References

9.14.2 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property—DN to Gateway
9.14.2 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property—DN to Gateway—GW
Settings–Main
9.14.1 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property—DN to Call Distribution
Port Group
9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Network Numbering Plan—
Trunk Property
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Network Numbering Plan—
Trunk Property
10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—Incoming Call Inter-digit
Timer—TIE
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ Idle Line Access (Local Access)
→ Trunk Group Access
→ TIE Line Access
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Other PBX Extension
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—TRS—TRS Level—Day,
Lunch, Break, Night
10.9 PBX Configuration—[2-9] System—System Options—Option 4—Private Network—TIE Call by
Extension Numbering
11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings—Main—COS
17.1 PBX Configuration—[9-1] Private Network—TIE Table
18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings
→ DIL—Trunk Property
→ DIL—DIL Destination—Day, Lunch, Break, Night

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4.3.2 Voice over Internet Protocol (VoIP) Network

→ DDI / DID / TIE / MSN

18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—DDI / DID Destination—Day,
Lunch, Break, Night
18.5 PBX Configuration—[10-5] CO & Incoming Call—Miscellaneous—Intercept—Routing to Operator - No
Destination (Destination is not programmed.)

4.3.1.4 Common Extension Numbering for 2 PBXs

Description
Two PBXs can have extensions with the same leading number. TIE Line calling is available using extension
numbering.
PBX-1 PBX-2

TIE Line
Interface Interface

Extn.1011 Extn.1012 Extn.1013

Dials "1013".

Explanation:
If a dialled number is not found at the local PBX, the call can be sent to the remote PBX.
When an extension number is dialled, the PBX first searches local extensions for a matching number. If
there is no match, the PBX then checks the TIE Line Routing Table for a corresponding entry. If an entry is
found, the call is sent to the connected PBX.
Case 1:
Extension 1012 of PBX-1 dials extension number "1011".
→ The dialled number is found at the local PBX, so extension 1012 of PBX-1 is connected to extension
1011 of PBX-1.
Case 2:
Extension 1012 of PBX-1 dials extension number "1013".
→ The dialled number is not found at the local PBX, so the call is redirected to the specified TIE Line, and
extension 1012 of PBX-1 is connected to extension 1013 of PBX-2.

Conditions
• System programming is required to enable this feature.

4.3.2 Voice over Internet Protocol (VoIP) Network

Description
When a PBX is connected to another PBX via a private IP network, voice signals are converted into IP
packets and sent over the network. This is known as Voice over IP (VoIP). This PBX uses the H.323
standard for VoIP communication.
VoIP networks support private network communications using TIE line service.

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[Example]

Telephone Company

TRG 1
PBX-1 PBX-2
Extn.1000 Extn.2000
Private IP
: V-IPGW Network V-IPGW :
Router Router
Extn.1999 Extn.2999
TRG 2

Dials "2999".
PBX-3
Extn.3000
V-IPGW :
Router
Extn.3999

Required Programming
• PBX
For making a call:
ARS programming (→ 2.8.1 Automatic Route Selection (ARS)) or TIE line service programming
For receiving a call:
TIE line service programming

[TIE Line Routing and Modification Table]

Priority 1 Priority 2 ..
Dial Modification Dial Modification ..
Location Leading
No. No. TRG Removed TRG Removed
Added Added
No. of No. of ..
No. No.
Digits Digits
2 ..
01 2 (VoIP 0
port)
02 3 2 0 ..
: : : : : : : : :

[Explanation]
Calls to destinations with leading number "2" or "3" are automatically routed through the VoIP ports,
designated as trunk group 2.
• IP Gateway
IP address assignment for the local PBX and other PBXs.

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4.3.2 Voice over Internet Protocol (VoIP) Network

[Programming Example]
Destination
Leading No. IP Address
2 200.45.11.35
3 199.176.64.1
: :

[Explanation]
Calls are routed to the IP address of each V-IPGW/IP-GW card based on the leading number dialled.
Automatic Rerouting of VoIP Calls to Public Trunks
When a VoIP call cannot be completed successfully, the PBX can automatically attempt to make the call
using a public trunk instead. This provides a backup method of making calls in cases when IP network
transmission cannot be completed successfully.

[Example]

Telephone Company Telephone Company

(area code: 098) (area code: 012)

012-345-1011
PBX-1 PBX-2
V-IPGW Private IP V-IPGW
Extn.1000 Extn.1011
Network
(012-345-1011)
Dials
"7-20-1011". PBX code: 30 PBX code: 20

The leading numbers of extensions accessed through the VoIP network are added as entries to the Quick
Dialling table, in addition to being registered as Other PBX Extension Numbers, as shown below:
[Programming Example]
Number to dial to call an extension at another PBX using VoIP network:
7 (TIE line access number) + 20 (PBX Code) + 1011 (extension number)
Number to dial to call that extension using a public trunk:
9 (trunk access number) + 012-345-1011

[Quick Dialling Table]

Quick Dialling No. Destination No.
720 9012345

If the call cannot be completed using the VoIP network, and the dialled leading number is found in the Quick
Dialling Table, the call will be automatically rerouted to a trunk as specified by the corresponding destination
number.
When a call is made using the VoIP network, if the PBX does not receive a reply from the other PBX within
about 4 seconds of making the call, or an error is returned, the call is rerouted to a public trunk as specified.

Automatic Rerouting of VoIP Calls to Public Trunks using ARS

When dialling an outside party using ARS, the call can be rerouted to a public trunk if the call cannot be
completed successfully.

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Telephone Company
Telephone Company
(area code: 012)
(area code: 098)

012-345-1011
PBX-1 PBX-2
V-IPGW Private IP V-IPGW
Extn.1000 Network (012-345-1011)

Dials
"9-012-345-1011"

Explanation:
1. An extension of PBX-1 dials the Idle Line Access number of PBX-1 "9" and telephone number
"012-345-1011".
2. PBX-1 modifies and routes the call to PBX-2 through a private IP network according to the ARS
programming of PBX-1.
3. The IP network transmission cannot be completed successfully and the call is rerouted via a public
trunk to the outside party "012-345-1011".
[Quick Dialling Table]
The leading number (in this case, "9") of the dialled number is found in the Quick Dialling Table, and the call
is automatically rerouted to the specified trunk group. It is necessary to specify a trunk group to make this
type of call. If the idle line access number is used in a destination number, the call will be rerouted through
the same private IP network according to the ARS programming, and the call will not be completed.
Quick Dialling No. Destination No.
9 802

Sending Faxes Through the Network

Through system programming, it is possible to select whether faxes are sent using analogue or T.38
protocol signals.

Conditions
• Some QSIG services are available. (→ 4.3.4 QSIG Standard Features)
• TRS/Call Barring settings apply to calls rerouted to public trunks. When making a call using System
Speed Dialling, regular TRS/Call Barring settings are applied, even if the TRS/Barring Override by
System Speed Dialling feature is enabled. (→ 2.7.1 Toll Restriction (TRS)/Call Barring (Barring))

Installation Manual References

4.4 Virtual Cards

PC Programming Manual References

9.14.1 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property—DN to Call Distribution
Port Group
9.14 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property
17.1 PBX Configuration—[9-1] Private Network—TIE Table

Feature Manual References

2.8.1 Automatic Route Selection (ARS)
4.3.1 TIE Line Service

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4.3.2 Voice over Internet Protocol (VoIP) Network

4.3.2.1 Gateway Groups

Description
It is possible to automatically reroute outgoing VoIP calls according to preassigned gateway groups.
A gateway device can be assigned a gateway group number based on its IP Address. When a call is made
using a leading number allocated to a gateway group, it is routed to the lowest-numbered available device
within that group. In other words, if the lowest-numbered device is busy or not available, the call slides to the
next available device.

[Example]

Office A (GW Group-1)

Gateway Settings List PBX-2

1. V-IPGW(1): GW Group-1 V-IPGW(1)
2. V-IPGW(2): GW Group-1 Leading Number
Dials "123XXX"
3. V-IPGW(3): GW Group-2 List
4. V-IPGW(4): GW Group-2 1. 1xx: GW Group-1
2. 2xx: GW Group-2
Busy V-IPGW(2)

PBX-1
Rerouted Office B (GW Group-2)
V-IPGW Private IP
Network PBX-3
V-IPGW(3)

V-IPGW(4)

When "123XXX" is dialled, the call is routed to GW Group-1. However the lowest-numbered device (V-IPGW
(1)) is busy or not available, so the call is rerouted to V-IPGW (2).

4.3.2.2 Common Extension Numbering for Multiple PBXs

Description
Multiple PBXs in separate locations, connected in an IP network, can share a common block of extensions
designated in a gateway group.

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 4.3.2 Voice over Internet Protocol (VoIP) Network

2) Extn. 105 not PBX B in Office B

found in PBX B Extn. 102
V-IPGW Extn. 202
Extn. 203

1) Extn. 105 dialled 3) Rerouted

PBX A in Office A
PBX C in Office C
Extn. 101 Private IP
Extn. 105
Extn. 103 V-IPGW Network
V-IPGW Extn. 201
Extn. 104
Extn. 303

4) Check next PBX in

Gateway Group settings 5) Extn. 105
found in PBX C

Explanation:
In the same way as when connected by a TIE Line, if a dialled number is not found at the local PBX, the call
can be sent to other PBXs connected via an IP network. When an extension number is dialled, the PBX first
searches local extensions for a matching number. If there is no match, the PBX then checks the TIE Line
Routing Table for the Gateway Group for a corresponding entry. If an entry is found, the call is sent to the
connected PBX.

Conditions
• System programming is required to enable this feature.
• If the called extension does not exist at the called PBX, the next PBX in the same gateway group is called
automatically.
• The Routing to Operator setting in system programming must be disabled to use this feature.
• To use this feature, all PBXs in the IP network must be KX-NS1000 PBXs, or KX-NCP/KX-TDE series
PBXs with MPR Software Version 3.0000 or later.

PC Programming Manual References

18.5 PBX Configuration—[10-5] CO & Incoming Call—Miscellaneous—Intercept—Routing to Operator - No
Destination (Destination is not programmed.)

4.3.2.3 Call Distribution Port Group

Description
It is possible to set which virtual port receives each call depending on the telephone number of the called
party.
By assigning each port to a Call Distribution Port Group (CDPG), it is possible to select which group
receives each call. CDPG settings cover all V-IPGW cards in the PBX, which allows for ports on different
cards to be assigned to the same CDPG. In other words, two 8-port cards can be used as one 16-port card.
Programming Example:
To enable this feature, it is necessary to programme the following 2 tables through system programming:
• CDPG Table

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4.3.2 Voice over Internet Protocol (VoIP) Network

- in order to assign ports to Call Distribution Port Groups.

• Hunt Pattern Table
- in order to programme a priority list of CDPG destinations for each leading number.

[Programming Example of the CDPG Table]

V-IPGW Card Number Port Number Call Distribution Port Group*1
1 1 CDPG 1
1 2 CDPG 2
1 : :
1 8 CDPG 2
2 1 CDPG 1
2 : :
2 8 CDPG 3

*1 → 9.14 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property—Call Distribution Port Group

[Programming Example of the Hunt Pattern Table]

Leading Call Distribution Port Call Distribution Port Call Distribution Port
No. …
Number*1 Group (1st)*2 Group (2nd)*2 Group (16th)*2
1 10 CDPG 1 CDPG 4 … -
2 20 CDPG 1 CDPG 2 … CDPG 3
: : : : … :
32 300 CDPG 8 CDPG 11 … -

*1 → 9.14.1 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property—DN to Call Distribution Port Group—Hunt

Pattern 1–16—Leading Number
*2 → 9.14.1 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property—DN to Call Distribution Port Group—Hunt
Pattern 1–16—Call Distribution Port Group—1st–16th

Note
The same CDPG can be set for several leading numbers.
Explanation:
As shown in the CDPG table above, CDPG 1 consists of Port 1 of the first card and Port 1 of the second
card. When "1023-456-7890" (leading number: 10) is dialled:

The leading number (10) is The call is routed to The call is rerouted to The call is
searched for in the Hunt Pattern Port 1 of the first card. Port 1 of the second answered.
Table. card.

CDPG 1 is the 1st priority Port 1 of the first card Port 1 of the second
CDPG. is busy. card is available.

If all the ports belonging to CDPG 1 are busy, the call is rerouted to the lowest-numbered available port
belonging to CDPG 4, which is set as the second priority for this leading number.

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 4.3.3 ISDN Virtual Private Network (ISDN-VPN)

4.3.3 ISDN Virtual Private Network (ISDN-VPN)

Description
ISDN Virtual Private Network (ISDN-VPN) is a service provided by the telephone company. It uses an
existing line as if it were a private line. There is no need to set up a private line or to lease a line from the
telephone company. Making and receiving both public and private calls is possible using the same line.
Public/Private Discrimination:
a. When making a call: The public/private discrimination number is required before sending the dialled
number to the telephone company. The public/private discrimination number can be dialled manually, or
automatically by ARS programming (→ 2.8.1 Automatic Route Selection (ARS)) and/or TIE line service
programming.
b. When receiving a call: The telephone company distinguishes the call type. If it is a private call, the call
is received by the TIE line service method. If it is a public call, the call is received by the Incoming Trunk
Call Distribution method (DIL/DDI/MSN) which is assigned on the trunk.
[Example]

Public ISDN

<Public ISDN-VPN
Discrimination>
9-0-01-23-4567
01-23-4567
Public No.

Dials
<Private "01-45-6789".
PBX-1 Discrimination> PBX-2 PBX-3
PBX Code 111 PBX Code 112 PBX Code 113
113-401
Head Branch Branch
Office Private No. Office Office

Extn. 201 Extn. 202 Extn. 301 Extn. 302 Extn. 401 Extn. 402
(DDI No.:
01-45-6789)
Dials "9-01-23-4567". Dials "401".
(ARS) (TIE)
Note: Public Call

Private Call

Conditions
• Each BRI/PRI port can be set to public or VPN through system programming. To use this service, select
VPN.
• Even if the telephone company does not support the ISDN-VPN service, it is possible to use the same
kind of service when making a call by TIE line service programming, and/or Quick Dialling programming
(→ 2.6.6 Quick Dialling).

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4.3.4 QSIG Standard Features

[Quick Dialling Programming Example]

Location No. Quick Dialling No. Desired No.
Quick Dialling 01 2345 (extension no. of other PBX) 9-123-4321 (Public no. of
extension 2345)
: : :

Explanation:
When an extension user dials "2345", he is connected to extension "2345" of other PBX whose public
number is "123-4321".

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Network Numbering Plan—
Trunk Property
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Network Numbering Plan—
Trunk Property
16.2 PBX Configuration—[8-2] ARS—Leading Number—Leading Number
17.1 PBX Configuration—[9-1] Private Network—TIE Table—Priority 1–Priority 8
→ Leading Number
→ Removed Number of Digits
→ Added Number

Feature Manual References

4.3.1 TIE Line Service

4.3.4 QSIG Standard Features

4.3.4.1 QSIG Standard Features—SUMMARY
Description
QSIG is a protocol which is based on ISDN (Q.931) and offers enhanced PBX features in a private network.
The QSIG network supports private communications by the TIE line service method.
The following features are available for an ISDN-QSIG or VoIP private network. For ISDN, system
programming is required to specify whether each feature (excluding Calling Line Identification Presentation
[CLIP]) is available for each port of the private network.

[Service Table]
Service Description & Reference
Calling Line Identification Sends the caller’s number to the QSIG network when making a
Presentation (CLIP) call.

→ 4.3.4.2 Calling/Connected Line Identification Presentation

(CLIP/COLP) and Calling/Connected Name Identification
Presentation (CNIP/CONP)—by QSIG
Connected Line Sends the number of the answered party to the QSIG network
Identification Presentation when answering a call.
(COLP)
→ 4.3.4.2 Calling/Connected Line Identification Presentation
(CLIP/COLP) and Calling/Connected Name Identification
Presentation (CNIP/CONP)—by QSIG

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Service Description & Reference

Calling Line Identification Prevents the caller’s CLI being presented to the called party by
Restriction (CLIR) the caller.

→ 4.3.4.2 Calling/Connected Line Identification Presentation

(CLIP/COLP) and Calling/Connected Name Identification
Presentation (CNIP/CONP)—by QSIG
Connected Line Prevents COLP being sent by the answered party.
Identification Restriction
(COLR) → 4.3.4.2 Calling/Connected Line Identification Presentation
(CLIP/COLP) and Calling/Connected Name Identification
Presentation (CNIP/CONP)—by QSIG
Calling Name Identification Sends the caller’s name to the QSIG network when making a call.
Presentation (CNIP)
→ 4.3.4.2 Calling/Connected Line Identification Presentation
(CLIP/COLP) and Calling/Connected Name Identification
Presentation (CNIP/CONP)—by QSIG
Connected Name Sends the name of the answered party to the QSIG network when
Identification Presentation answering a call.
(CONP)
→ 4.3.4.2 Calling/Connected Line Identification Presentation
(CLIP/COLP) and Calling/Connected Name Identification
Presentation (CNIP/CONP)—by QSIG
Calling Name Identification Prevents the caller’s name being presented to the called party by
Restriction (CNIR) the caller.

→ 4.3.4.2 Calling/Connected Line Identification Presentation

(CLIP/COLP) and Calling/Connected Name Identification
Presentation (CNIP/CONP)—by QSIG
Connected Name Prevents CONP being sent by the answered party.
Identification Restriction
(CONR) → 4.3.4.2 Calling/Connected Line Identification Presentation
(CLIP/COLP) and Calling/Connected Name Identification
Presentation (CNIP/CONP)—by QSIG
Call Forwarding (CF)—by Forwards a call to the QSIG network.
QSIG
→ 4.3.4.3 Call Forwarding (CF)—by QSIG
Call Transfer (CT)—by QSIG Transfers a call to the QSIG network.

→ 4.3.4.4 Call Transfer (CT)—by QSIG

Completion of Calls to Busy Receives callback ringing when a busy called party on the QSIG
Subscriber (CCBS)—by network becomes free.
QSIG
→ 4.3.4.5 Completion of Calls to Busy Subscriber (CCBS)—by
QSIG

Feature Manual References

4.1.2 Integrated Services Digital Network (ISDN) Service Features
4.3.1 TIE Line Service

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4.3.4 QSIG Standard Features

4.3.4.2 Calling/Connected Line Identification Presentation (CLIP/

COLP) and Calling/Connected Name Identification Presentation
(CNIP/CONP)—by QSIG
Description
Calling Line/Name Identification Presentation (CLIP/CNIP):
The PBX can send a preprogrammed extension number and/or name to the QSIG network when an
extension user makes a call. The called party can see the number and/or name on his telephone display
before answering the call.
Connected Line/Name Identification Presentation (COLP/CONP):
The PBX sends a preprogrammed extension number and/or name to the QSIG network when the extension
user answers an incoming call. The caller can see the number and/or name of the answering party on his
telephone display when the call is answered.
[CLIP/CNIP Example]

2) "John
101"
1) Dials "202". is displayed.
PBX-1 CLIP: 101 PBX-2
CNIP: John

Caller Called party

(Extn. No.: 101 (Extn. No.: 202)
Extn. Name: John)

[COLP/CONP Example]

1) Dials "203". Called party

(Extn. No.: 203
PBX-1 PBX-2 Extn. Name: Tom)

FWD, IRNA, etc.

Caller Answering party

COLP: 204
CONP: Paul (Extn. No.: 204
3) "Paul Extn. Name: Paul)
204"
is displayed.
2) Answers the call.

CLIP/COLP Number:
The extension number sent to the QSIG network for CLIP/COLP can be assigned for each extension
through system programming.
CNIP/CONP Name:
The extension name sent to the QSIG network for CNIP/CONP can be assigned for each extension through
system programming.
Calling/Connected Line Identification Restriction (CLIR/COLR):
It is possible for each extension to restrict the sending of its extension number to the QSIG network by
pressing the CLIR button, COLR button, or entering the feature number.
Calling/Connected Name Identification Restriction (CNIR/CONR):

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 4.3.4 QSIG Standard Features

It is possible for each extension to restrict the sending of its extension name to the QSIG network. When
CLIR is activated, CNIR becomes active automatically. When COLR is activated, CONR becomes active
automatically.

Conditions
• These features comply with the following European Telecommunication Standard (ETS) specifications:
– CLIP/COLP: ETS 300 172 Circuit mode basis services.
– CNIP/CONP: ETS 300 238 Name identification supplementary services.
• COLP/CLIR/COLR/CNIP/CONP/CNIR/CONR Assignment for Each Port
Each service can be enabled or disabled on each ISDN (QSIG) port of the PBX.
• CLIR Button and COLR Button
It is possible to switch between CLIP and CLIR by pressing the CLIR button, and COLP and COLR by
pressing the COLR button. A flexible button can be customised as the CLIR or COLR button.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, 3PTY
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911, 3PTY
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features
→ COLR Set / Cancel
→ CLIR Set / Cancel
8.1.1 Users—User Container—Add User/Edit User—User Information
→ Extension Number
→ Display Name

Feature Manual References

2.21.2 Flexible Buttons

Operating Manual References

1.9.4 Displaying Your Telephone Number on the Called Party and Caller’s Telephone (Calling/Connected
Line Identification Presentation [CLIP/COLP])
1.9.5 Preventing Your Telephone Number Being Displayed on the Caller’s Telephone (Connected Line
Identification Restriction [COLR])
1.9.6 Preventing Your Number Being Displayed on the Called Party’s Telephone (Calling Line Identification
Restriction [CLIR])

4.3.4.3 Call Forwarding (CF)—by QSIG

Description
The PBX forwards the call to a destination extension in another PBX in QSIG network. The destination can
be set on your own PBX on an extension basis as the forward destination of trunk calls (→ 2.3.2 Call
Forwarding (FWD)).

If the same trunk group is used for the incoming call and the forwarded call, the following situation will be
possible.

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4.3.4 QSIG Standard Features

[Example] 1 Extension 1000 of PBX-1 dials

extension number "2000", and the call
QSIG is sent to extension "2000" of PBX-2
PBX-1 PBX-2 through QSIG network.
1 Call to 2000 2 The call is forwarded to the forward
destination of trunk calls of extension
2000, which is extension "1001" of
2 Forwarded PBX-1.
to 1001
3 The call between PBX-1 and PBX-2 is
released, and the call is connected
directly to the forward destination of
Extn. 1000 Extn. 1001 Extn. 2000 extension 2000.
(Forward Destination
Dials "2000". of Trunk Calls: 1001)

QSIG

PBX-1 PBX-2

Extn. 1000 Extn. 1001 Extn. 2000

Conditions
• This feature complies with European Telecommunication Standard (ETS) specification ETS 300 257,
Diversion supplementary services.
• This feature can be enabled or disabled on each ISDN (QSIG) port of the PBX.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, 3PTY
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911, 3PTY

Operating Manual References

1.6.1 Forwarding Calls

4.3.4.4 Call Transfer (CT)—by QSIG

Description
The PBX transfers the call to a destination extension in another PBX in QSIG network.

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If the same trunk group is used for the incoming call and the transferred call, the following situation will be
possible.
[Example] 1 Extension 1000 of PBX-1 dials
extension number "2000", and the call
QSIG is sent to extension "2000" of PBX-2
PBX-1 PBX-2 through QSIG network.
1 Call to 2000 2 The call is transferred from extension
2000 to extension "1001" of PBX-1.
3 The call between PBX-1 and PBX-2 is
2 Transferred released, and the call is connected
to 1001
directly to the transfer destination of
extension 2000.

Extn. 1000 Extn. 1001 Extn. 2000

Dials "2000".

QSIG

PBX-1 PBX-2

Extn. 1000 Extn. 1001 Extn. 2000

Conditions
• This feature complies with European Telecommunication Standard (ETS) specification ETS 300 261, Call
transfer supplementary service.
• This feature can be enabled or disabled on an ISDN (QSIG) port basis.
• Call Transfer with Announcement and Call Transfer without Announcement is possible (→ 2.12.1 Call
Transfer).

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, 3PTY
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911, 3PTY

Operating Manual References

1.4.1 Transferring a Call (Call Transfer)

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4.3.4 QSIG Standard Features

4.3.4.5 Completion of Calls to Busy Subscriber (CCBS)—by QSIG

Description
If the call has been made to an extension in another PBX in QSIG network and the called party is busy, an
extension user can set to receive callback ringing when the called party becomes free. When the user
answers the callback ringing, that party’s number is automatically dialled.

Conditions
• This feature complies with European Telecommunication Standard (ETS) specification ETS 300 366, Call
completion supplementary services.
• This feature is available under the following conditions:
a. The caller’s PBX is capable of using CCBS.
b. The called party’s PBX is capable of accepting CCBS.
• To receive and send CCBS, receiving and sending CCBS must be enabled individually on an ISDN
(QSIG) port basis through system programming.
• An extension user can set only one CCBS. The last setting is effective.
• The CCBS setting is cancelled if there is no callback ringing within 60 minutes or callback ringing is not
answered within 10 seconds.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, 3PTY
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Supplementary Service—
COLP, CLIR, COLR, CNIP, CONP, CNIR, CONR, CF (Rerouting), CT, CCBS, AOC-D, AOC-E, E911, 3PTY

Operating Manual References

1.2.4 When the Dialled Line is Busy or There is No Answer

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 Section 5
System Configuration and Administration
Features

Feature Manual 379

5.1 System Configuration—System

5.1 System Configuration—System

5.1.1 Class of Service (COS)
Description
Each extension must belong to a Class of Service (COS). By assigning certain extensions to a COS, it is
possible to control the behaviour and privileges of extension users (allowing or denying certain extensions
access to various features, extensions, and trunks) depending on the duties appointed to them.
Many extensions can belong to the same COS by assigning each extension the same COS number,
allowing the same restrictions and privileges to apply to a group of extensions.
The following features are controlled on a COS basis:
1. 2.1.2.2 Internal Call Block
2. 2.3.2 Call Forwarding (FWD)
3. 2.3.3 Do Not Disturb (DND)—DND Override
4. 2.4.3 Call Pickup
5. 2.5.4.3 Account Code Entry
6. 2.5.5.3 Trunk Access
7. 2.10.2 Executive Busy Override
8. 2.10.3 Call Monitor
9. 2.10.4.3 Off-hook Call Announcement (OHCA)
10. 2.10.4.4 Whisper OHCA
11. 2.7.1 Toll Restriction (TRS)/Call Barring (Barring)
12. 2.7.3 Extension Dial Lock
13. 2.7.5 Walking COS
14. 2.11.8 Trunk Call Limitation
15. 2.12.1 Call Transfer
16. 2.18.2 Door Open
17. 2.16.1 Direct Inward System Access (DISA)
18. 4.1.2.4 Call Forwarding (CF)—by ISDN (P-MP)
19. 2.22.1.1 Station Message Detail Recording (SMDR)—SMDR for Outgoing Trunk Calls
20. 5.1.4 Time Service—Time Service Switching
21. 5.1.6 Manager Features
22. 5.5.2 PT Programming
23. 2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension—SDN Key
mode, SDN Walking COS, and assigning SDN buttons through PT programming
24. 2.29.1 User Container
25. 5.6.8 Congestion Control
COS for Unified Messaging
The Unified Messaging system has its own COS settings for controlling access to various functions.
(→ 3.2.1.11 Class of Service (COS))

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 5.1.2 Group

Conditions
• Walking COS
Extension users can temporarily use their own COS at another extension with a less-privileged COS to
access features, extensions, or trunks that are normally inaccessible due to that extension’s COS.

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings
8.2 Users—Advanced Extension Settings—COS

Feature Manual References

3.2.1 System Features
6.1 Capacity of System Resources

Operating Manual References

1.2.7 Calling without Restrictions

5.1.2 Group
Description
This PBX supports various types of groups.
1. Trunk Group
Trunks can be grouped into a specified number of trunk groups (e.g., for each carrier, trunk type, etc.).
Several settings can be assigned on a trunk group basis. All trunks belonging to a trunk group follow
the assignment determined for that trunk group.
→ 11.1.1 PBX Configuration—[3-1-1] Group—Trunk Group—TRG Settings
One trunk can belong to only one trunk group on a port basis.
Port basis: ELCOT/CLCOT/LCOT/DID/E & M/ISDN-BRI/ISDN-PRI23/ISDN-PRI30/SIPGW
Channel basis: E1/T1
2. Extension User Group
The PBX supports extension user groups, each of which is used to compose the following groups:
a. Tenant (→ 5.1.3 Tenant Service)
b. Call Pickup Group (See below.)
c. Paging Group (See below.)
Every extension must belong to one extension user group, but cannot belong to more than one
extension user group.
→ 11.2 PBX Configuration—[3-2] Group—User Group
Assignable Extensions: PT/SLT/PS/SIP Extension/ISDN Extension/T1-OPX

[Example]
Extension Extension Extension Extension
User Group 1 User Group 2 User Group 3 User Group 4

Extn. 100 Extn. 101 Extn. 102 Extn. 103 Extn. 104 Extn. 105 Extn. 106 Extn. 107

Call Pickup Group

Using the Group Call Pickup feature, extensions can answer any calls within a specified group.
One extension user group can belong to several call pickup groups. (→ 2.4.3 Call Pickup)
→ 11.3 PBX Configuration—[3-3] Group—Call Pickup Group

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5.1.2 Group

→ 11.3.1 PBX Configuration—[3-3-1] Group—Call Pickup Group—Main Setting

[Example]

Call Pickup Group 1 Call Pickup Group 2 Call Pickup Group 3

Extension Extension Extension Extension

User Group 1 User Group 2 User Group 3 User Group 4

Extn. 100 Extn. 101 Extn. 102 Extn. 103 Extn. 104 Extn. 105 Extn. 106 Extn. 107

Paging Group
Using the Paging feature, extensions can make a page to any paging groups or answer a page to their
own groups. One extension user group or external pager can belong to several paging groups.
(→ 2.17.1 Paging)
→ 11.4 PBX Configuration—[3-4] Group—Paging Group
→ 11.4.1 PBX Configuration—[3-4-1] Group—Paging Group—Priority Setting
→ 11.4.2 PBX Configuration—[3-4] Group—Paging Group—External Pager

[Example]

Paging Group 1 Paging Group 2 Paging Group 3

Extension Extension Extension

User Group 1 User Group 2 User Group 3

Extn. 100 Extn. 101 Extn. 102 Extn. 103 Extn. 104 Extn. 105 Pager

3. Idle Extension Hunting Group

If a called extension is busy or in DND mode, Idle Extension Hunting redirects the incoming call to an
idle member of the same idle extension hunting group, which can be programmed through system
programming. Idle extensions are automatically searched according to a preprogrammed hunting type:
Circular Hunting or Terminated Hunting (→ 2.2.1 Idle Extension Hunting).
→ 11.6 PBX Configuration—[3-6] Group—Extension Hunting Group
→ 11.6.1 PBX Configuration—[3-6] Group—Extension Hunting Group—Member List
4. Incoming Call Distribution Group
An incoming call distribution group is a group of extensions which receives incoming calls directed to
the group. Each incoming call distribution group has a floating extension number (default: 6 + two-digit
*1
group number ) and name. One extension can belong to multiple groups.
→ 11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings
→ 11.5.1.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—
Member List
Assignable Extensions: PT/SLT/PS/SIP Extension/ISDN Extension/T1-OPX/PS Ring Group
(→ 2.2.2 Incoming Call Distribution Group Features)

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 5.1.2 Group

[Example]

Incoming Call Incoming Call

Distribution Group 1 Distribution Group 2
(Floating Extn. No. 601, (Floating Extn. No. 602,
Name: Sales 1) Name: Sales 2)

Extn. 103 Extn. 104

Extn. 100 Extn. 101 Extn. 102 Extn. 105 Extn. 106 Extn. 107

*1 The number of digits for Floating Extn. No depends on the value specified for Numbering Plan in Easy Setup.
→ 2.1.4 Easy Setup Wizard

5. UM Group
A UM group is the collection of all Unified Messaging ports of one PBX.
(→ 3.1.1 Unified Messaging System Overview)
→ 11.7.1 PBX Configuration—[3-7-1] Group—UM Group—System Settings
→ 11.7.2 PBX Configuration—[3-7-2] Group—UM Group—Unit Settings
6. PS Ring Group
A PS ring group is a group of PS extensions that receives incoming calls directed to the group. Each
group has a floating extension number and name through system programming. One PS can belong to
multiple groups.
(→ 5.2.4.2 PS Ring Group)
→ 11.8 PBX Configuration—[3-8] Group—PS Ring Group
→ 11.8.1 PBX Configuration—[3-8] Group—PS Ring Group—Member List

PS Ring Group 1 PS Ring Group 2

(Floating Extn. No. 301, (Floating Extn. No. 302,
Name: Sales 1) Name: Sales 2)

PS01 PS04
PS03 PS06
PS02 PS05

7. Conference Group
A conference group is a group of parties that are called when an extension user uses the Conference
Group Call feature (→ 2.15.1 Conference Group Call). When Broadcast Mode is enabled through
system programming, a maximum of 31 parties can be assigned to a group. When Broadcast Mode is
disabled, a maximum of 7 parties can be assigned to a group. A maximum of 48 conference groups
can be programmed.
→ 11.9 PBX Configuration—[3-9] Group—Conference Group
→ 11.9.1 PBX Configuration—[3-9] Group—Conference Group—Member List
8. P2P Group
Devices in the same P2P group can establish peer-to-peer (P2P) connections and communicate (make
calls) without using PBX resources. IP-PTs, SIP extensions, and PBXs are all assigned to P2P groups.

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5.1.2 Group

Connection between devices in the same P2P group

P2P Group 1

PBX PBX

DSP DSP

P2P

Connection between devices in different P2P groups

P2P Group 1 P2P Group 2

PBX PBX
non-P2P

DSP DSP

The PBX group assignment is used when an IP telephone establishes a call with a non-IP telephone
(e.g., an SLT). If the non-IP telephone’s PBX and the IP telephone are in the same P2P group, a P2P
connection is established between the IP telephone and the PBX:

Site 1 Site 2

P2P Group 1 PBX PBX

DSP

Router Router

SLT

→ 11.10 PBX Configuration—[3-10] Group—P2P Group

PC Programming Manual References

11 PBX Configuration—[3] Group
2.1.4 Easy Setup Wizard

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 5.1.3 Tenant Service

Feature Manual References

5.5.7 Floating Extension
6.1 Capacity of System Resources

5.1.3 Tenant Service

Description
This PBX can be shared with a certain number of tenants.
1. Tenant Configuration
Tenant Member
The tenant members consist of extension user groups. One extension user group can belong to only
one tenant. Therefore, one extension can belong to only one tenant.
(→ 5.1.2 Group)
Time Service
Each tenant has a Time Table. The Start and/or End time of each time mode (day/lunch/break/night)
can be set for each day of the week. The Time Table numbers correspond to the tenant numbers
respectively.
(→ 5.1.4 Time Service)

[Example]
Tenant 1 Tenant 2

Extension Extension
User Group 1 User Group 5
Extension Extension
User Group 2 User Group 6
Extension
User Group 3
Extension
User Group 4

Use Time Table 1 Use Time Table 2

2. System Management
Each of the following system management items can be assigned to each tenant.
a. Tenant Operator (extension number/floating extension number of incoming call distribution group/
none) (→ 5.1.5 Operator Features)
→ 14.6 PBX Configuration—[6-6] Feature—Tenant—Operator (Extension Number)
b. ARS Mode (Off/Local Access/All Access/System) (→ 2.8.1 Automatic Route Selection (ARS))
→ 14.6 PBX Configuration—[6-6] Feature—Tenant—ARS Mode
c. Music Source for Music on Hold (System/BGM Number/Tone)
(→ 2.13.4 Music on Hold)
→ 14.6 PBX Configuration—[6-6] Feature—Tenant—Music On Hold
d. System Speed Dialling (System/Tenant Exclusive)
(→ 2.6.4 Speed Dialling—Personal/System)
→ 14.6 PBX Configuration—[6-6] Feature—Tenant—System Speed Dial

[Programming Example]
System Speed
Tenant No. Operator ARS Mode Music Source
Dialling
1 Extn.101 Local Access System*1 System*2

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5.1.3 Tenant Service

System Speed
Tenant No. Operator ARS Mode Music Source
Dialling
Extended/
2 None*3 System*4 Tone
Tenant Exclusive
Floating extn. Extended/
3 Off BGM1
no. 200 Tenant Exclusive
: : : : :

*1 Follows the system assignment of the music source for the Music on Hold.
→ 10.2 PBX Configuration—[2-2] System—Operator & BGM
*2 Follows the system assignment for System Speed Dialling.
→ 14.1 PBX Configuration—[6-1] Feature—System Speed Dial
*3 Follows the system assignment of a PBX operator.
→ 10.2 PBX Configuration—[2-2] System—Operator & BGM—BGM and Music on Hold—Music on Hold
*4 Follows the system assignment of the ARS mode.
→ 16 PBX Configuration—[8] ARS

Conditions
• Multi-Connection Networking (→ 4.2 Multi-connection Networking)
For features whose operation depends on Time Service (→ 5.1.4 Time Service), system programming
specifies which PBX’s clock is used to determine the time mode.
→ 14.6 PBX Configuration—[6-6] Feature—Tenant—Time Service Mode
• ARS Assignment
When "On for Local Access Operation" or "On for Any CO Access Operation" is assigned
as the ARS Mode for a tenant, only a subset of the ARS Leading Number Table is applied to that tenant’s
outgoing calls. Tenants 1 to 20 are assigned a range of 50 of the entries in the Leading Number Table as
follows:
– Tenant 1: Entries 1 to 50
– Tenant 2: Entries 51 to 100
– Tenant 3: Entries 101 to 150
:
– Tenant 100: Entry 4951 to 5000
– Tenant 101 to 128: Entry 1 to 5000
If "Same as System Setting" is selected, then all 5000 entries in the table will be applied to that
tenant’s outgoing calls. All 5000 entries in the table are applied to tenants 101 to 128 when ARS is
enabled, regardless of the tenant’s ARS Mode.
By dividing tenants, specific ARS settings can be applied to specific tenants according to the
requirements of each tenant.
The following example illustrates how the ARS Leading Number Table is applied to tenants:
Tenant No. ARS Mode Applied ARS Entries
1 On for Local Access Operation Entries 1 to 50
2 Off Not applied
3 Same as System Setting (System Entries 1 to 5000
Setting: On)
4 On for Local Access Operation Entries 151 to 200
5 Off Not applied

• Tenant-to-Tenant Call Block

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 5.1.3 Tenant Service

The following features can be restricted based on the COS for each extension (not based on the tenant)
by the Internal Call Block feature (→ 2.1.2.2 Internal Call Block):
– Calling extensions or doorphone(s) in the restricted tenant(s)
– Picking up calls ringing in the restricted tenant(s)
– Retrieving a call held within the restricted tenant(s)
[Example]

Tenant 1 Tenant 2
Extension Extension
User Group 1 User Group 3

Extn. 100 Extn. 101 Extn. 104 Extn. 105

COS 1 COS 2 COS 3 COS 4
Extension
Tenant 3 Extension
User Group 2 Extension User Group 4
User Group 5

Extn. 102 Extn. 103 Extn. 106 Extn. 107

Extn. 108 Extn. 109
COS 5 COS 6
Extension
User Group 6

Extn. 110 Extn. 111

[Programming Example]
Called Party
Caller
COS 1 COS 2 COS 3 COS 4 COS 5 COS 6 ...
COS 1 ...
COS 2 ...
COS 3 ü ü ...
COS 4 ü ü ...
COS 5 ü ü ü ü ...
COS 6 ü ü ü ü ...
: : : : : : : :

ü: Block
Explanation:
1. Assign each extension in a tenant to a certain COS number. Each tenant must have unique COS
numbers.
Tenant 1: COS 1 and COS 2

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5.1.4 Time Service

Tenant 2: COS 3 and COS 4

Tenant 3: COS 5 and COS 6
2. Tenant-to-Tenant Call Block enables by the Internal Call Block feature.
a. Tenant 1 (COS 1 and COS 2) can make calls to both Tenant 2 (COS 3 and COS 4) and Tenant
3 (COS 5 and COS 6) as well as Tenant 1.
b. Tenant 2 (COS 3 and COS 4) can make calls to Tenant 1 (COS 1 and COS 2) and Tenant 2.
c. Tenant 3 (COS 5 and COS 6) can make calls to Tenant 3 itself only.
• An incoming call distribution group must belong to one tenant because the following features are
determined on a tenant basis (→ 2.2.2.1 Incoming Call Distribution Group Features—SUMMARY):
– Music on Hold while a call is waiting in the queue
– The Time Table which determines the overflow destination

PC Programming Manual References

10.2 PBX Configuration—[2-2] System—Operator & BGM
→ PBX Operator—Day, Lunch, Break, Night
→ BGM and Music on Hold—Music on Hold
10.4 PBX Configuration—[2-4] System—Week Table
10.5 PBX Configuration—[2-5] System—Holiday Table
10.7.3 PBX Configuration—[2-7-3] System—Class of Service—Internal Call Block
11.2 PBX Configuration—[3-2] Group—User Group
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Main—
Tenant Number
8.2 Users—Advanced Extension Settings—User Group
13.1 PBX Configuration—[5-1] Optional Device—Doorphone—Tenant Number
13.5 PBX Configuration—[5-5] Optional Device—External Sensor—Tenant No.
14.1 PBX Configuration—[6-1] Feature—System Speed Dial
14.6 PBX Configuration—[6-6] Feature—Tenant
16 PBX Configuration—[8] ARS
16.5.3 PBX Configuration—[8-5-3] ARS—Carrier—Authorisation Code for Tenant
18.2 PBX Configuration—[10-2] CO & Incoming Call—DIL Table & Port Settings—DIL—Tenant Number
18.3 PBX Configuration—[10-3] CO & Incoming Call—DDI / DID Table—Tenant Number
18.4 PBX Configuration—[10-4] CO & Incoming Call—MSN Table—MSN—Tenant Number

Feature Manual References

6.1 Capacity of System Resources

5.1.4 Time Service

Description
This PBX supports day, night, lunch, and break modes of operation. TRS/Barring can be arranged
separately. The destination of incoming calls can be set differently for each mode.
1. Time Service Switching Mode
Day/lunch/break/night mode can be switched either automatically or manually. The switching mode can
be assigned for each tenant.
The switching mode can also be changed by pressing the Time Service Switching Mode (Automatic/
Manual) button. This can be performed by only an extension assigned as the manager, or
preprogrammed extension on a COS basis.

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Type Description
Automatic The PBX will switch mode according to the preprogrammed Time Table.
Manual A manager, or preprogrammed extension on a COS basis can switch
mode by dialling the feature number or pressing the Time Service
button.

The Unified Messaging System Manager can set the time service mode from an outside telephone.
Even while in the Automatic Switching mode, day/lunch/break/night mode can be changed manually.
2. Time Table
Each tenant has a Time Table used for the Automatic Switching mode. The Start and/or End time of
each mode can be set for each day of the week. The Time Table numbers correspond to the tenant
numbers respectively.

[Time Table Example]

Time Table No. (Tenant No.)
Time Schedule
1 2 3 4 …
MON Day 1 start 08:00 11:00 08:00 08:00 …
Lunch start 12:00 NONE 16:00 12:00 …
Day 2 start 13:00 NONE NONE NONE …
Break 1 start NONE NONE NONE NONE …
Break 1 end (Day restart) NONE NONE NONE NONE …
Night start 16:00 20:00 12:00 NONE …
TUE Day 1 start 08:00 11:00 08:00 08:00 …
Lunch start 12:05 NONE 13:00 13:00 …
Day 2 start 13:00 NONE NONE NONE …
Break 1 start NONE NONE NONE NONE …
Break 1 end (Day restart) NONE NONE NONE NONE …
Night start 16:31 20:00 17:00 NONE …
: : : : : : …

<Time Service Image of Monday>

Time Table No. 00:00 08:00 11:00 12:00 13:00 16:00 20:00 24:00 08:00
1 Night Day 1 Lunch Day 2 Night Day 1

2 Night Day 1 Night

3 Night Day 1 Night Lunch Day 1

4 Night Day 1 Lunch Day 1

3. Features Using Time Service

The following features can be set in each time mode (day/lunch/break/night):
a. Destination of incoming trunk calls (DIL/DID/DDI/MSN) (→ 2.1.1 Incoming Trunk Call Features)
b. Destination of the Intercept Routing (→ 2.1.1.6 Intercept Routing)
c. Queuing Time Table for incoming call distribution groups (→ 2.2.2.3 Queuing Feature)
d. Overflow destination for incoming call distribution groups (→ 2.2.2.6 Overflow Feature)
e. Destination of incoming doorphone calls (→ 2.18.1 Doorphone Call)
f. PBX operator (→ 5.1.5 Operator Features)

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5.1.4 Time Service

g. COS for TRS/Barring and for Trunk Access

h. Outgoing Message (OGM) for Timed Reminder (→ 2.24.3 Timed Reminder)
i. Intercept time for Intercept Routing—No Answer (→ 2.1.1.6 Intercept Routing) and for DISA
Intercept Routing—No Answer (→ 2.16.1 Direct Inward System Access (DISA))
j. Service group settings for voice mail (→ 3.2.1.40 Service Group)
k. Using quick dialling to reroute a Multi-Connection network intercom call to a trunk during break
mode (→ 2.6.6 Quick Dialling).
This feature works only when the time mode is set to "break".
[Programming Examples of DID/DDI Table and DIL Table]
DID/DDI table can be programmed for each DID/DDI number, and a tenant (Time Table) number is
assigned to each DID/DDI number. DIL table can be programmed for each trunk, and a tenant (Time
Table) number is assigned to each trunk.

<DID/DDI Table>
Location DID/DDI No. Tenant DID/DDI Destination
(Time Table)
No. Day Lunch Break Night

001 123-4567 1 105 100 (UM) 105 100 (UM)

002 123-2468 1 102 100 (UM) 102 100 (UM)
: : : : : : :

<DIL Table>
Trunk No. Tenant (Time Table) No. DIL Destination
Day Lunch Break Night
01 1 101 100 (UM) 101 100 (UM)
02 2 102 100 (UM) 102 100 (UM)
: : : : : :

Explanation:
If a trunk call with a DID number (123-4567) is received at 20:00;
a. Tenant (Time Table) number 1 will be used.
b. The call is received during night mode in Time Table 1.
c. The call will be routed to extension 100 (UM Group).
4. Holiday Mode
The holiday mode activates automatically using the Automatic Switching mode. Up to 24 holidays can
be stored, and one time mode can be selected for all holidays.
5. Time Service Button
A flexible button can be customised as the following buttons:
a. Day/Night button
b. Day/Night/Lunch button
c. Day/Night/Break button
d. Day/Night/Lunch/Break button
Each of these buttons is used for switching between modes. For example, pressing the Day/Night
button switches between day and night modes. All of these buttons show the current status as follows:

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Light Pattern Status

Off Day mode
Red on Night mode
Green on Lunch mode
Slow green flashing Break mode
Slow red flashing Holiday mode

Note
Any extension user (except extension users allowed to change the mode) can only check the
current status on the display by pressing the Time Service button.

Conditions
• System programming can set the following time periods:
– Day-1 (Day Start time)
– Lunch (Lunch Start time)
– Day-2 (Lunch End time)
– Night (Night Start time)
– Break-1 Start
– Break-1 End (Day restart)
– Break-2 Start
– Break-2 End (Day restart)
– Break-3 Start
– Break-3 End (Day restart)
• Multi-Connection Networking (→ 4.2 Multi-connection Networking)
In Automatic Switching mode, system programming specifies which PBX’s clock is used to determine the
time mode when a time-mode-dependent feature is used. One of the following 2 options can be specified:
– The local PBX of the device or trunk that is using the feature
– The PBX site to which the Time Table’s tenant is assigned
→ 14.6 PBX Configuration—[6-6] Feature—Tenant—Time Service Mode
• Time Service Switching Mode (Automatic/Manual) Button
A flexible button can be customised as the Time Service Switching Mode (Automatic/Manual) button.
• When the Time Service mode is switched, the Idle user can be serviced out automatically by the settings
for each user. For more information, See 2.24.4.3 Automatic Service-out Feature.

PC Programming Manual References

10.4 PBX Configuration—[2-4] System—Week Table
10.5 PBX Configuration—[2-5] System—Holiday Table
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Features—Time Service (Day /
Lunch / Break / Night) Switch
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Manager—Time Service
Switch
8.3.1 Users—Flexible Button—Wired Extension
→ Type
→ Parameter Selection (for Time Service)

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5.1.5 Operator Features

→ Parameter Selection (for Time Service - Automatic/Manual)

→ Optional Parameter (Ringing Tone Type Number) (for Time Service)
8.3.2 Users—Flexible Button—Portable Station
→ Type
→ Parameter Selection (for Time Service)
→ Parameter Selection (for Time Service - Automatic/Manual)
→ Optional Parameter (or Ringing Tone Type Number) (for Time Service)

Feature Manual References

2.6.6 Quick Dialling
2.7.1 Toll Restriction (TRS)/Call Barring (Barring)
2.21.2 Flexible Buttons
3.2.1 System Features
5.1.1 Class of Service (COS)
5.1.3 Tenant Service
6.1 Capacity of System Resources

Operating Manual References

1.9.10 Checking the Time Service Status
2.1.2 Time Service Mode Control
2.2.1 System Manager Features—Remote Time Service Mode Setting

5.1.5 Operator Features

Description
Any extension or Incoming Call Distribution (ICD) group can be designated as an operator.
This PBX supports the following types of operators:
Type Description
PBX operator An extension or incoming call distribution group can be assigned as a
PBX operator for each time mode (day/lunch/break/night).
Tenant operator An extension or incoming call distribution group can be assigned as a
tenant operator. The tenant operator may be the extension or incoming
call distribution group of another tenant.
[Example] Extension 110 in tenant 1 is the tenant operator of tenant 3.

Operator Call:
An extension user can call an operator by dialling the preprogrammed Operator Call feature number. The
destination of the Operator Call depends on the following:
– If the Tenant Service is not in use:
The call is directed to the PBX operator according to the corresponding time mode.
– If the Tenant Service is in use:
The call is directed to the extension’s tenant operator. If a tenant operator is not assigned, the call is
directed to the PBX operator. In this case, the current time mode of the extension’s tenant is used to
determine the PBX operator that the call is directed to.
If neither a tenant operator nor a PBX operator is assigned, the caller will hear a reorder tone.

Conditions
• A single extension or incoming call distribution group can be assigned as both a tenant operator and the
PBX operator.
• Tenant operators can be assigned individually for multiple tenants.

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PC Programming Manual References

10.2 PBX Configuration—[2-2] System—Operator & BGM—PBX Operator—Day, Lunch, Break, Night
14.6 PBX Configuration—[6-6] Feature—Tenant—Operator (Extension Number)

Feature Manual References

5.1.3 Tenant Service
6.1 Capacity of System Resources

Operating Manual References

1.2.1 Basic Calling
1.12.1 Using the Telephones in a Hotel-type Environment (Hospitality Features)

5.1.6 Manager Features

Description
An extension assigned as the manager (manager extension) is allowed to use the specified features. COS
programming determines the extensions which can use the following manager features:
Manager
Feature Description & Reference
Password
Manager Manager Changes the manager password. Required
Programming Password
Change
Call Charge Sets, displays, clears, and prints the call Required
Management charge data.

→ 2.22.3 Call Charge Services

Verification Code Sets a verification code PIN for each Required
Personal verification code.
Identification
Number (PIN) Set → 2.7.6 Verification Code Entry
Remote PIN Clears the extension PIN of an extension Required
Clear remotely, and a verification code PIN. PIN
Lock is also unlocked.

→ 2.24.1 Extension Personal Identification

Number (PIN)
→ 2.7.6 Verification Code Entry
Remote Sets or cancels the Extension Dial Lock on Required
Extension Dial an extension remotely.
Lock
→ 2.7.3 Extension Dial Lock
Dial Tone Transfer Changes the TRS/Barring level of the Not required
extension temporarily.
[Example] An extension user can call a
manager to release the restricted outgoing
call (e.g., international call).

→ 2.7.4 Dial Tone Transfer

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5.1.6 Manager Features

Manager
Feature Description & Reference
Password
Outgoing Message (OGM) Records and plays back outgoing Not required
messages (OGMs).

→ 2.28.2 Outgoing Message (OGM)

Time Service Switches the time mode (day/lunch/break/ Not required
night) manually.

→ 5.1.4 Time Service

BGM—External Sets the External BGM on and off. Not required

→ 2.28.1 Background Music (BGM)

Trunk Busy Out Clear Clears the Busy Out status of a trunk. Not required

→ 2.5.4.6 Trunk Busy Out

Conditions
CAUTION
There is a risk that fraudulent telephone calls will be made if a third party discovers a personal
identification number (PIN) (verification code PIN or extension PIN) of the PBX.
The cost of such calls will be billed to the owner/renter of the PBX.
To protect the PBX from this kind of fraudulent use, we strongly recommend:
1. Keeping PINs secret.
2. Selecting complex, random PINs that cannot be easily guessed.
3. Changing PINs regularly.
• Manager Password
One manager password can be assigned per PBX.

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Manager—Manager
19.1 PBX Configuration—[11-1] Maintenance—Main—Password—Manager Password - PT Programming—
Prog *1

Feature Manual References

5.1.1 Class of Service (COS)

Operating Manual References

2.1 Control Features
4.1.2 Manager Programming

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5.2 System Configuration—Extensions

5.2.1 IP Proprietary Telephone (IP-PT)
Description
IP Proprietary Telephones (IP-PTs) are telephones that make and receive calls using IP and that connect to
the PBX over a LAN. Except for their use of IP, they function almost identically to traditional proprietary
telephones. Each IP-PT is always registered to a specific User Container and must belong to one User
Group.
IP-PTs have two Ethernet ports for connection, primary and secondary. They are connected to the PBX
through a network hub or other splitting device, and can have a PC connected to the secondary Ethernet
port.

[Connection Example]

PBX
IP-PT

V-IPEXT Private IP
Network
IP-PT PC

IP-PT

Primary Secondary
Ethernet Port Ethernet Port

Main Office Branch Office

Conditions
IP-PTs that can be integrated into the KX-NSX are the KX-NT343, KX-NT346, KX-NT366, KX-NT543,
KX-NT546, KX-NT560, KX-NT511, KX-NT551, KX-NT553 and KX-NT556.
• IP-PT registration is required through system programming before an IP-PT can be used with the PBX.
An IP-PT cannot be used unless an extension number is assigned. However, depending on system
programming, registration may occur completely automatically, or may require only inputting the desired
extension number.
For details on how to register IP-PTs, refer to the Installation Manual.
• MGCP-TLS / SRTP
For each IP-PT (KX-NT600 series only) port, the MGCP signalling is encrypted. At the same time, the
Voice-IP packet data is encrypted by SRTP. Since the IP-PT with MGCP-TLS / SRTP set to Enable
makes a call with the other PT after terminating the Voice-IP packets in the PBX, P2P calls are not
possible.
• DSP Resource Usage
Making a call from an IP-PT requires a certain number of DSP resources, depending on the codec used.
If all DSP resources are in use, this operation cannot be performed. To ensure a minimum level of
performance, DSP resources can be reserved for VoIP communication. (→ 5.5.3 DSP Resource Usage)
However, DSP resources are not required for P2P calls. (→ 5.2.3 Peer-to-Peer (P2P) Connection)
• KX-NT307(PSLP1528) Bluetooth Module
A Bluetooth wireless headset can be registered to a KX-NT300 series (except KX-NT321) IP-PT
containing the KX-NT307(PSLP1528) Bluetooth Module through personal programming. When Headset

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5.2.2 SIP (Session Initiation Protocol) Extension

Mode is off, the Bluetooth headset can be used to answer calls or redial. In this case, Headset Mode will
turn on automatically, and will turn off after you hang up.
This Bluetooth Module is also compatible with KX-DT343/KX-DT346 DPTs.
• Calls made using a Bluetooth wireless headset will not be disconnected immediately when the user
wanders out of range. However, if the Bluetooth wireless headset user remains out of range for a
specified time period, the call will be disconnected.
• The following features, available when a legacy gateway is connected (→ 5.3.1 Stacking Connection),
cannot be used with an IP-PT:
– XDP
– Digital XDP
– OHCA

Installation Manual References

4.4 Virtual Cards

PC Programming Manual References

9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Main—IP Terminal Registration
Mode
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Port Number
9.15 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Card Property
9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property

5.2.2 SIP (Session Initiation Protocol) Extension

Description
This PBX supports the connection of SIP-compatible IP telephones (hardphones and softphones). SIP
extensions make and receive calls using Internet Protocol (IP).
For more information about SIP extension compatibility with feature numbers, see 5.5.6 Flexible Numbering/
Fixed Numbering.
For more information about Panasonic SIP phones (P-SIP), see 5.2.2.2 Panasonic SIP Phones (P-SIP).
Below are the features supported by SIP extensions:
• Absent Message (→ 2.20.2 Absent Message)
• Account Code Entry (→ 2.5.4.3 Account Code Entry)
• Automatic Route Selection (ARS) (→ 2.8 Automatic Route Selection (ARS) Features)
• Call Forwarding (FWD) (→ 2.3.2 Call Forwarding (FWD))
• Call Hold (→ 2.13.1 Call Hold)
• Call Park*1 (→ 2.13.2 Call Park)
• Call Pickup (→ 2.4.3 Call Pickup)
• Call Transfer with Announcement (→ 2.12.1 Call Transfer)
• Call Waiting Tone (→ 2.1.3.3 Call Waiting)
• COLR/CLIR/CLIP/COLP (→ 4.1.2.2 Calling/Connected Line Identification Presentation (CLIP/COLP))
• Conference*2 (→ 2.14 Conference Features)
• Conference Group Call (→ 2.15.1 Conference Group Call)
• Data Line Security (→ 2.11.5 Data Line Security)
• Dial Information (CTI) (→ 2.26.1 Computer Telephony Integration (CTI))

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• Direct Inward System Access (DISA) (→ 2.16.1 Direct Inward System Access (DISA))
• Door Open (→ 2.18.2 Door Open)
• Doorphone Call (→ 2.18.1 Doorphone Call)
• Emergency Call (→ 2.5.4.2 Emergency Call)
• Executive Busy Override Deny (→ 2.10.2 Executive Busy Override)
• Extension Dial Lock (→ 2.7.3 Extension Dial Lock)
• Extension Feature Clear (→ 2.24.2 Extension Feature Clear)
• Extension PIN (→ 2.24.1 Extension Personal Identification Number (PIN))
• External BGM On/Off (→ 2.28.1 Background Music (BGM))
• FWD/DND (→ 2.3 Call Forwarding (FWD)/Do Not Disturb (DND) Features)
• Group FWD (→ 2.3.2 Call Forwarding (FWD))
• Idle Line Access (→ 2.5.5.3 Trunk Access)
• Log-in/Log-out (→ 2.2.2.7 Log-in/Log-out)
• Message Waiting (→ 2.20.1 Message Waiting)
• Not Ready (→ 2.2.2.7 Log-in/Log-out)
• Operator Call (→ 5.1.5 Operator Features)
• Paging (→ 2.17.1 Paging)
• Personal Speed Dialling (→ 2.6.4 Speed Dialling—Personal/System)
• Quick Dialling (→ 2.6.6 Quick Dialling)
• Redial (→ 2.6.3 Last Number Redial)
• Remote Station Lock (→ 2.7.3 Extension Dial Lock)
• S-CO Line Access (→ 2.5.5.3 Trunk Access)
• System Speed Dialling (→ 2.6.4 Speed Dialling—Personal/System)
• TIE Line Call (→ 4.3.1 TIE Line Service)
• Time Service (→ 5.1.4 Time Service)
• Timed Reminder (→ 2.24.3 Timed Reminder)
• Trunk Group Access (→ 2.5.5.3 Trunk Access)
• Verification Code (→ 2.7.6 Verification Code Entry)
*1 SIP extensions can retrieve parked calls but cannot park calls.
*2 As a member only (not as an originator).

Conditions
[General]
• This PBX supports SIP devices that use RFC 3261, 3264, 3310, 2327, or 4028.
• Some SIP phones may not be compatible with this PBX.
• Before a SIP extension can be used with the PBX, the IP address of the mother board, password, and
extension number must be assigned on the SIP extension and on the PBX. Even if the IP terminal
registration mode has been set to full automatic mode or extension input mode, general SIP extensions
must be registered manually.
For details on how to register, refer to the Installation Manual.
• When registering the SIP extension, the user ID must be the extension number of the SIP extension.

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5.2.2 SIP (Session Initiation Protocol) Extension

• When a SIP extension uses the Call Hold feature, the target call is put on Consultation Hold.
• DSP Resource Usage
Making a call from a SIP extension requires a certain number of DSP resources, depending on the codec
used. If all DSP resources are in use, this operation cannot be performed. To ensure a minimum level of
performance, DSP resources can be reserved for VoIP communication. (→ 5.5.3 DSP Resource Usage)
However, DSP resources are not required for P2P calls. (→ 5.2.3 Peer-to-Peer (P2P) Connection)

Installation Manual References

4.4 Virtual Cards

PC Programming Manual References

9.17 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Card Property
9.18 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Port Property

5.2.2.1 Simple Remote Connection

Description
Using the Built-in Media Relay Gateway
Using the Built-in Media Relay Gateway feature, the following IP phones can be connected to a remote
location without an additional device, such as an SBC (Session Border Controller):
• KX-NT500 series IP-PT
• IP-Softphone by Panasonic
• IP-CS (KX-NS0154)
• KX-HDV series SIP phones
• Mobile Softphone (KX-UCMA)
• General SIP phones

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Mobile Softphone

Mobile Internet

PBX
(+ Built-in Media Relay Internet
Gateway) Switching Hub Switching Hub

Router Router

PC IP-PT IP-PT SIP phone

Main Site Remote Site

: LAN
: WAN

Note
• Voice data (RTP packets) from remote extensions connected to the PBX via the built-in Media Relay
Gateway cannot be sent via a P2P connection. Instead, these calls are connected using the RTP
switch method.
• IP Softphones connected from a remote location via the Media Relay Gateway do not support the
following features:
– Paging (2.17.1 Paging)
– Background Music (2.28.1 Background Music (BGM))
• KX-HDV series SIP phones, Mobile Softphones and general SIP phones can be connected to the
Built-in Media Relay Gateway by following the method for using an SBC.
Using an SBC (Session Border Controller)
General SIP phones support simple remote connectivity when the KX-NSX Series is networked with an SBC
(session border controller). Simple remote connectivity means that even if the SIP phone is located behind a
NAT router, firewall, or both, specialised settings such as NAT traversal settings do not need to be
configured for each remote extension.
There are 2 scenarios for configuring and connecting a SIP phone:
a. The SIP phone is connected and registered to the PBX on the PBX’s local network. The necessary
settings are configured automatically by the PBX.
b. The remote IP settings of the SIP phone are configured without first connecting the phone to the PBX.
Once programmed, the SIP phone is sent to the remote location, connected to the network and will
automatically connect to the PBX.

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Installation Manual References

5.9.3 Installing SIP Phones at a Remote Site with an External SBC
5.9.4 Installing IP Phones at a Remote Site with a Built-in Media Relay Gateway

PC Programming Manual References

9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—SIP Extension
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Port Number—Port Number
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Media Relay
→ Common—NAT - External IP Address / FQDN
→ IP Extension / IP-CS—NAT - MGCP Server Port No.
→ IP Extension / IP-CS—NAT - MGCP Server Port No. for IP-CS
→ IP Extension / IP-CS—Keep Alive Packet Type
→ IP Extension / IP-CS—Keep Alive Packet Type for IP-CS
→ IP Extension / IP-CS—Keep Alive Packet Sending Interval Time (s)
→ IP Extension / IP-CS—NAT - FTP Server Port No.
→ SIP Extension—NAT - SIP Proxy Server Port No.
→ Option—NAT - RTP IP Address
→ Option—NAT - SIP Proxy Server IP Address
9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property—Remote Place
(Location / P2P)
9.18 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Port Property—Remote Place
(Location / P2P)
9.20 PBX Configuration—[1-1] Configuration—Slot—V-IPCS16—Port Property—Remote Place

5.2.2.2 Panasonic SIP Phones (P-SIP)

Description
• Panasonic SIP Phones (P-SIP) are treated as the following models:
– KX-HDV series
– KX-TGP600
– KX-NTV150 (Communication IP camera)
– KX-NTV160 (Video door phone)
– Mobile Softphone by Panasonic
• P-SIP phones are registered and de-registered to/from the PBX in the same way as general SIP phones.
• Video communication is available for internal call between the following phones:
– KX-HDV430
– KX-NTV150 (Communication IP camera)
– KX-NTV160 (Video door phone)
– Mobile Softphone by Panasonic
• During the video communication, the following operations are not available:
– Transfer (→2.12 Transferring Features)
– Hold (→2.13 Holding Features)
– Conference (→2.14 Conference Features)
• You can get more detailed information on the Panasonic web site at:
– KX-HDV series/KX-TGP600:

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https://panasonic.net/cns/pcc/support/sipphone/download/
– KX-NTV series:
https://panasonic.net/cns/pcc/support/pbx/download/ntv/index.html
– Mobile Softphone:
https://panasonic.net/cns/pcc/support/pbx/download/mobilesip/index.html

Conditions
• Requirement:
A V-SIPEXT card is required to use a P-SIP phone.
• P-SIP phones are counted as general SIP phones and, therefore, are included in the maximum capacity
for general SIP phones.

[KX-NTV series]
You can open a door that is associated with a KX-NTV series device (except the trunk call from another PBX
to the KX-NTV series.)

[KX-TGP600]
• A KX-TGP600 can be registered to up to 8 V-SIPEXT ports.
• In addition, one KX-TGP600 can support up to 8 wireless handset conversations simultaneously.
However, since the number of wireless handsets is controlled by the KX-TGP600, from the PBX's
perspective it appears that the number of occupied ports does not match the number of extensions (i.e.,
the PBX can only see the KX-TGP600 base unit).
• Depending on the registration status of the wireless handsets, up to 8 activation keys for IP-PT/P-SIP
may be required.

Installation Manual References

4.4 Virtual Cards

PC Programming Manual References

9.17 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Card Property
9.18 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Port Property

5.2.3 Peer-to-Peer (P2P) Connection

Description
This PBX automatically establishes peer-to-peer communication between peer-to-peer compatible IP
extensions (i.e., IP-PTs and SIP extensions) that belong to the same P2P group. With peer-to-peer calls, the
call is routed directly from one IP extension to another without going through a DSP card, which means that
P2P calls are established without using the PBX's resources.
P2P Scenarios
P2P calls can be established between telephones in a single site or across sites. The following illustrations
show P2P calls established in a variety of network configurations. It is assumed that all the devices are
connected over a private IP network.

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5.2.3 Peer-to-Peer (P2P) Connection

P2P connection within a site

Site 1 Site 2

PBX PBX

Router Router
P2P P2P

Calls between IP telephones within a site establish a P2P connection.

Conditions
[General]
• Three codecs are used for peer-to-peer calls: G.722, G.711, and G.729A. The speech quality of the
codecs varies as follows: (High) G.722, G.711, G.729A (Low).
When the preferred codec of each party differs, the call will be established using the lower codec. For
example, if the caller prefers G.711 while the called party prefers G.729A, the call will be established
using G.729A.
• G.722 is only available for calls between KX-NT300 series IP-PTs, KX-NT500 series IP-PTs, and some
SIP extensions that support this codec during peer-to-peer communication.
• Through system programming, it is possible to assign the preferred codec to use for IP-PTs.
• For General SIP and P-SIP SIP extensions, the priority of the codec that will be used can be specified via
the telephone itself.
• For non-peer-to-peer calls via the DSP card, calls cannot be made or received when all of the card’s
resources are being used.
• IP-CSs do not support P2P connections.
• P2P communication using the T.38 protocol is supported for calls between IP extensions.
• If all of the following conditions are met, it will be possible to continue a P2P call at time of a PBX failure.
– The terminal performing the P2P call is a KX-NT550 series terminal (Version 02.000 or later)
– The terminal performing the P2P call is not integrated in the Expansion Gateway
[P2P Groups]
• Telephones must belong to the same P2P group to establish a P2P connection.
Telephones at different sites can be assigned to the same P2P group.
• Calls between IP extensions in different P2P groups are established via the DSP card in "DSP-through
mode". DSP-through mode is where only 1 DSP resource is required, regardless of the codec being used.
• Each PBX in a Multi-Connection network is also assigned to a P2P group. This assignment is used when
a non-IP telephone (e.g., an SLT) is on a call with an IP telephone at another branch. If the PBX and the
IP telephone both belong to the same P2P group, the connection is established from the PBX to the IP
telephone directly.

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 5.2.4 Portable Station (PS) Features

Site 1 Site 2

P2P Group 1 PBX PBX

DSP

Router Router

SLT

• Calls between extensions in different P2P groups will consume DSP resources, even if both extensions
belong to the same PBX (site).

Installation Manual References

4.3.3 DSP S Card (KX-NS0110), DSP M Card (KX-NS0111), DSP L Card (KX-NS0112)

PC Programming Manual References

9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—VoIP-DSP Options
9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property—Location / P2P—P2P
Group
9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property—Location / P2P—P2P
Group Name
9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property—Option—IP Codec
Priority
9.18 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Port Property—Main—P2P Group
9.18 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Port Property—Main—P2P Group
Name
9.18 PBX Configuration—[1-1] Configuration—Slot—V-SIPEXT128—Port Property—FAX/T.38

Feature Manual References

5.2.1 IP Proprietary Telephone (IP-PT)
5.1.2 Group

5.2.4 Portable Station (PS) Features

5.2.4.1 Portable Station (PS) Connection
Description
This PBX supports the connection of a PS. Cell Stations (CSs) allow PSs to receive reception within a
designated area. It is possible to use the PBX features using the PS like a PT.

Conditions
• The PS registration is required through the system programming. To avoid unexpected registration to
another PBX, the Personal Identification Number (PIN) for the PBX is necessary to register a PS. The
registration can be cancelled.
• Handover
Even if a PS user moves during a conversation, the PS will automatically switch between cells without
disconnecting the call (Handover).

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5.2.4 Portable Station (PS) Features

Handover is available in any of the following cases:

1. During a conversation with an extension or outside party.
2. While a call is ringing at the PS.
3. In a stacking connection, the CSs are connected to different PBXs. (→ 5.3.1 Stacking Connection)

PBX

CS
CS Handover:
Interface
Calls will not be
disconnected.

CS

However, Handover is not available in any of the following cases:

1. When the new (Handover) CS is busy.
2. When there is no CS within range.
3. While the Live Call Screening (LCS) or Two-way Record is activated (→ 3.2.2.19 Live Call
Screening (LCS) and 3.2.2.34 Two-way Record/Two-way Transfer).
4. While the PS user is paging other extensions (→ 2.17.1 Paging).
5. While the PS user is dialling digits to make a trunk call.
6. During a conversation with an extension using PC Console or PC Phone, to which the conversation
is being recorded.
• DSP Resource Usage
Making a call through an IP-CS with a PS requires a certain number of DSP resources, depending on the
codec used. If all DSP resources are in use, this operation cannot be performed. To ensure a minimum
level of performance, DSP resources can be reserved for VoIP communication. (→ 5.5.3 DSP Resource
Usage)
• When a caller has dialled the extension number of a PS but the CS is busy, the caller hears a busy tone.
• For more information about connecting PSs to CSs, see the Quick Installation Guide for the relevant CS.
• The number of digits allowed for a PS extension number is determined by the PS model. See your PS
documentation for details.

PC Programming Manual References

9.30 PBX Configuration—[1-2] Configuration—Portable Station

Feature Manual References

5.5.6 Flexible Numbering/Fixed Numbering

5.2.4.2 PS Ring Group

Description
A PS ring group is a group of PS extensions that receives incoming calls. Each group has a floating
extension number and name. One PS can belong to multiple groups.

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 5.2.4 Portable Station (PS) Features

[Programming Example]
PS Ring Group 01 PS Ring Group 02 PS Ring Group 03 ..
Floating Extn. No. 301 302 303 ..
Group Name Sales 1 Sales 2 Sales 3 ..
Incoming Trunk Call Called Party’s Name/ Caller’s Name/ Caller’s Name/ ..
Information Display Number Number Number
PS01 ü ..
PS02 ü ..
PS03 ü ..
PS04 ü ü ..
PS05 ü ..
PS06 ü ..
PS07 ü ..
: : : : :

ü: Constituent

PS Ring Group 1 PS Ring Group 2

(Floating Extn. No. 301 (Floating Extn. No. 302
Name: Sales 1) Name: Sales 2)

PS01 PS04
PS03 PS06
PS02 PS05

Conditions
• PS Ring Group
A maximum of 32 groups can be created.
• Compatible PSs
The following PSs can be assigned to PS ring groups:
– KX-TCA175
– KX-TCA185
– KX-TCA275
– KX-TCA285
– KX-TCA385
– KX-TCA364
– KX-WT115
• Incoming trunk call information is shown on a PS display when a trunk call arrives at a PS ring group
which the PS joins. The display information can be selected on a PS ring group basis through system
programming: Called Party’s Name/Number or Caller’s Name/Number.

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5.2.4 Portable Station (PS) Features

• Calling Multiple PSs Simultaneously

There are two methods to call multiple PSs simultaneously using the floating extension number assigned
to the following groups:
Method Assignment Merit Demerit
Incoming Call Assign all desired PSs to All PS users in the The CS may often be
Distribution Group one incoming call group can use the busy as each PS in
distribution group, and set Log-in/Log-out the group uses one
the group call distribution feature, Wrap-up channel when a call
method for the group to feature, and ICD arrives at the group.
"Ring". Group button for the
group.
→ 2.2.2 Incoming Call
Distribution Group
Features
PS Ring Group Assign all desired PSs to Only one channel is PS users in the
one PS ring group. used when a call group cannot use the
arrives at the group. Log-in/Log-out and
Wrap-up features.

• When a PS joins a PS ring group, the following personal settings are disregarded:
1. When the PS ring group is called:
– Delayed Ringing
– Display information when the incoming calls arrive;
The settings (e.g., display priority) are disregarded.
– The setting which is assigned on the PS (e.g., FWD)
– The status of the PS (e.g., busy)
2. Log-in/log-out setting (from the PS ring group/from the incoming call distribution group which the PS
ring group belongs to). (→ 2.2.2.7 Log-in/Log-out)

Note
Log-in/log-out setting of the PS ring group from the incoming call distribution group is also
disregarded.

PS Ring Group

Log-out

Log-in

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 5.2.4 Portable Station (PS) Features

PS Ring
Group
Log-out

Incoming Call
Distribution Group

PS Ring Group
Log-in

Log-out

Log-in

• When the PS ring group is called using the floating extension number, the group becomes busy to other
callers using the floating extension number. However, the individual group members may be called
directly using their extension number.
• If a PS in a PS ring group has set the DND feature for trunk calls, the PS will not ring when an intercom
call or a trunk call arrives at the PS ring group. (→ 2.3.3 Do Not Disturb (DND))
• For calls directed to PS ring groups, the PBX will handle at most two calls simultaneously. The third call
cannot arrive at a PS ring group until one of the first two calls is answered or a caller hangs up.
• If the PS becomes the Service-out state automatically while receiving a PS Ring Group call, the PS will
keep ringing. However, the PS user cannot answer the incoming call. (→ 2.24.4.3 Automatic Service-out
Feature)

PC Programming Manual References

11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Main—
Distribution Method
11.8 PBX Configuration—[3-8] Group—PS Ring Group
11.8.1 PBX Configuration—[3-8] Group—PS Ring Group—Member List

Feature Manual References

6.1 Capacity of System Resources

5.2.4.3 PS Directory
Description
A PS user can store numbers and/or names in the directory. A stored number is dialled by selecting a name
or number in the directory.
Depending on the PS model, the PS user can use the following directories for easy operation:
Type Description
PS Dialling Directory Makes a call by selecting from a private directory of names and
telephone numbers.

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5.2.5 ISDN Extension Features

Type Description
System Speed Dialling Directory Makes a call by selecting from a common directory of names and
numbers.
PBX Extension Dialling Directory Makes a call by selecting from a common directory of extension
names.
Shortcut Directory Accesses a feature by selecting from a private directory of feature
names and numbers.
Quick Dialling Makes a call or accesses a feature easily by selecting from a
private directory of names and numbers.

PC Programming Manual References

8.1.1 Users—User Container—Add User/Edit User—User Information—Display Name
14.1 PBX Configuration—[6-1] Feature—System Speed Dial
→ Name
→ CO Line Access Number + Telephone Number

5.2.4.4 PS Feature Buttons

Description
A PS user can use PBX features using a combination of buttons (button + a specified number, , or #)
and/or display operation. The flexible buttons and the display are customised through PS programming. The
button assignment is the same for the PT (→ 2.21.2 Flexible Buttons). Some special feature buttons (e.g.,
WAVESEARCH button) may be customised depending on the PS type.

5.2.5 ISDN Extension Features

5.2.5.1 ISDN Extension
Description
An ISDN (BRI/PRI) port can be used for either trunk or extension connection. When extension connection is
enabled, ISDN terminal devices (e.g., ISDN telephone, G4 fax machine, PC) or a behind PBX can be
connected to the port.
If the ISDN port is in point-to-point (P-P) configuration, one terminal device can be connected to the port. If
the ISDN port is in point-to-multipoint (P-MP) configuration, a maximum of eight terminal devices can be
connected to the port. However, only a maximum of two devices can be used simultaneously.
Individual terminal devices on a behind PBX can be addressed with Multiple Subscriber Number (MSN).
The MSN consists of the ISDN extension number and an additional digit, 0 through 9 or 00 through 99.

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 5.2.5 ISDN Extension Features

[Example]

Extn. No.: 1234

ISDN Port for
EXTN. (P-MP)
1234:
Up to eight terminals
... can be connected to
the port and receive
PBX (1234) (1234) ... (1234) a call simultaneously.

Extn. No.: 13X (MSN) 131–139:

ISDN Port for
Up to eight terminals
EXTN. (P-MP)
can be connected to
the port.
...
130:
(131) (132) ... (139) All terminals receive
a call simultaneously.
ISDN Port for Extn. No.: 14XX (MSN)
EXTN. (P-P/P-MP)

ISDN Port for

EXTN. (P-P/P-MP)
Behind
PBX 1400–1499:
Extn. No.: 15XX Each terminal
(MSN) ...
receives a call.
(Use the DDI/MSN
(1400) (1401) ... (1499) feature of the
Idle Extension behind PBX.)
Hunting is available.

Conditions
• Subaddressing
Subaddressing is possible between ISDN terminal devices. The subaddress goes through the PBX to the
ISDN terminal devices directly.
• Call Transfer (PBX feature) is available only for ISDN extensions in P-MP configuration. (→ 2.12.1 Call
Transfer)
• ISDN extensions can belong to an incoming call distribution group (→ 2.2.2 Incoming Call Distribution
Group Features) or idle extension hunting group (→ 2.2.1 Idle Extension Hunting). In this case, an MSN
can be assigned.
• If the last digit of the MSN is "0", all terminal devices on the same ISDN port receive the call
simultaneously, if the specification of each terminal device is available. To use the MSN whose last digit is
"0" as an individual MSN, system programming is required for each ISDN port.
• ISDN terminal devices that receive power over the telephone line are not supported.

PC Programming Manual References

9.26 PBX Configuration—[1-1] Configuration—Slot—Port Property - BRI Port—ISDN Extension
9.27 PBX Configuration—[1-1] Configuration—Slot—Port Property - PRI Port—Extension Setting

Feature Manual References

2.1.1.4 Multiple Subscriber Number (MSN) Ringing Service

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5.2.6 IP-CS (KX-NS0154)

5.2.6 IP-CS (KX-NS0154)

Description
The KX-NS0154 is an IP-CS that supports many network features supported by IP-PTs. It also supports 2
types of signal synchronisation.
Easy Setup
This IP-CS can be automatically registered to the PBX by connecting it to the network.
For details, refer to "5.9.1 Registering IP Telephones" in the Installation Manual.
It also supports remote connectivity with the Media Relay Gateway feature.
For details, refer to "5.9.4 Installing IP Phones at a Remote Site with a Built-in Media Relay Gateway" in the
Installation Manual.
Synchronisation Methods
CSs can synchronise their communication frames in order to enable handover when a PS user moves
between coverage areas of different CSs. There are 2 types of synchronisation: air synchronisation and LAN
synchronisation.
In air synchronisation, CSs synchronise frames wirelessly. In LAN synchronisation, CSs synchronise frames
over a LAN. LAN synchronisation has the benefit of providing synchronisation, even if wireless
communication is not possible or reliable between CSs.
For details about configuring synchronisation, refer to the documentation for the CS.
Statistics and Diagnostics
The IP-CS compiles statistics about its performance for diagnostic purposes. The following logs can be
viewed in Web Maintenance Console:
• IP-CS statistics for wireless channel and air synchronisation
→ 7.5.6.1 Utility—Report—IP-CS Information—IP-CS Statistical Information
• LAN Sync Group performance statistics
→ 7.5.6.2 Utility—Report—IP-CS Information—LAN Sync Information
• PS Call Disconnect Log
→ 7.5.6.3 Utility—Report—IP-CS Information—PS Call Disconnect Log

Conditions
[General]
• For details about setting up and operating the IP-CS, refer to the documentation of the IP-CS.
• If the number of DSP resources is insufficient for handling calls, the CS operates in the same way as
when it does not have any available channels.
For details about DSP resources, see "5.5.3 DSP Resource Usage".
• DECT communication between the CS and a PS is encrypted. However, encryption is not supported
when using a DECT repeater (except for KX-A406). In this case, the CS Repeater Mode setting must be
set to Normal. For details, refer to the documentation for the IP-CS and the repeater.
• KX-NSXE001, KX-NSXE010, KX-NSXE050 (Activation Key for IP-CS channel expansion) increases the
number of CS channels from 4 to 8 for the specified number of CSs.
[Registration]
• IP-CSs that support LAN synchronisation (KX-NS0154 only) are automatically assigned a LAN
Synchronisation Group number, according to the site they are registered to. If the maximum number of
CSs have already been registered, no group number will be assigned.
[Synchronisation]
• Slave IP-CSs can be added to a LAN synchronisation group as desired. They will automatically
synchronise themselves with the other IP-CSs.

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 5.2.7 Internal Call Encryption

• To enable easy setup, LLDP is supported by IP-CSs (KX-NS0154 only). If this feature causes network
congestion, it can be disabled.
• Only KX-NS0154 IP-CSs support LAN synchronisation.
• Even in LAN synchronisation, it is possible to also use air synchronisation. However, this is only possible
between KX-NS0154 IP-CSs.
• You must specify a LAN synchronisation group number or an air synchronisation group number for each
IP-CS. Do not leave both of these settings unspecified.
• In air synchronisation, if the CS loses its synchronisation signal, it will continue to run independently. Any
current calls will remain connected. Unlike previous IP-CS models (e.g., KX-NCP0158), the Watching
Degeneracy timer does not apply to the KX-NS0154 IP-CS.

PC Programming Manual References

7.4.4.1 Utility—Monitor/Trace—CS Status Monitor—Air Sync Group
7.4.4.2 Utility—Monitor/Trace—CS Status Monitor—LAN Sync Group
7.5.6 Utility—Report—IP-CS Information
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Main
→ LLDP Packet Sending Ability
→ IP-CS Registration Mode
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Port Number—Firmware
Update Port No. for IP-PT/IP-CS (Media Relay)
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Media Relay
→ IP Extension / IP-CS—NAT - MGCP Server Port No. for IP-CS
→ IP Extension / IP-CS—Keep Alive Packet Type for IP-CS
→ IP Extension / IP-CS—NAT - FTP Server Port No.
9.19 PBX Configuration—[1-1] Configuration—Slot—V-IPCS16—Card Property
9.20 PBX Configuration—[1-1] Configuration—Slot—V-IPCS16—Port Property
10.3 PBX Configuration—[2-3] System—Timers & Counters—Miscellaneous—Air Synchronisation—
Watching Degeneracy (x60s)
19.2 PBX Configuration—[11-2] Maintenance—CS Synchronisation

5.2.7 Internal Call Encryption

Description
This feature encrypts intercom calls to prevent eavesdropping.

Conditions
• Calls between specific extensions are encrypted. SIP signalling (SIP-TLS) and voice data (SRTP, P2P
calls only) can be encrypted.
• SIP signalling encryption (SIP-TLS) for voice data is performed between the main unit and devices.
• A SIP-TLS connection is established through authentication with the NSX server. For more information,
refer to 5.9.5 Encryption Settings for Internal Calls in Installation Manual.
• When in an SRTP call, the KX-HDV series / KX-TGP600’s LCD will display a key-shaped icon. When an
SRTP call is necessary for information security reasons, confirm that the key-shaped icon is displayed on
the KX-HDV series / KX-TGP600’s LCD. When in an RTP call or when switching from an SRTP call to an
RTP call after using an additional service (e.g. after a call transfer), the key-shaped icon will disappear.
For more information on the key-shaped icon, check the KX-HDV series / KX-TGP600’s corresponding
manual.
• When performing a call transfer, unless all of the transferred extension, transferring extension, and
transfer destination are internal SRTP-compatible lines, the call will be an RTP call after the call transfer.

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5.2.7 Internal Call Encryption

Notice
Even if the call is an SRTP call before the call transfer, depending on the combination, the call may be
an RTP call after the call transfer.
• Only calls made after using the additional services below support SRTP encryption.
Call Forwarding (refer to 2.3.2 Call Forwarding (FWD))
Call Transferring (refer to 2.12.1 Call Transfer)
Intercept Routing (refer to 2.1.1.6 Intercept Routing)
• The encryption (SRTP) of the video phone call’s media stream will not function.
• Intercom call encryption is available for extension calls through the Media Relay Gateway.
• Calls to telephones that do not support this feature will be unencrypted (RTP).

PC Programming Manual References

6.14 Tool—P-SIP Option
7.2.2 Utility—File—File Transfer PBX to PC
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Port Number
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—Media Relay
9.5.1 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main—SIP Extension

Installation Manual References

5.9.5 Encryption Settings for Internal Calls

Feature Manual References

2.1.1.6 Intercept Routing
2.3.2 Call Forwarding (FWD)

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 5.3 Legacy Device Connection

5.3 Legacy Device Connection

5.3.1 Stacking Connection
Description
KX-TDA and KX-TDE series PBXs can be connected to an Expansion Gateway (KX-NS1000) via a stacking
connection and used to connect telephones, devices, and trunks that are not natively compatible with the
KX-NSX Series. Such items include DPTs and APTs, DSS Consoles, digital CSs (i.e., non-IP), E1 trunks,
and T1 trunks.
A PBX connected through a stacking connection is referred to as a "legacy gateway".
All programming can be done through Web Maintenance Console on the KX-NSX Series.

Conditions
• Hardware Requirement: STACK-M card (KX-NS1000) and a STACK-S card (for each legacy gateway).
• The KX-TDA15 and KX-TDA30 are not supported for stacking.
• The following features are available through a legacy gateway:
– Parallelled Telephone
– E1 Line Service
– T1 Line Service
– KX-T7710 One-touch Dialling
For details about these features and how to configure the necessary settings, refer to the documentation
of the PBX you will be connecting.
• IP extensions and trunks, including SIP trunks, are not supported through legacy gateways.

PC Programming Manual References

9.2 PBX Configuration—[1-1] Configuration—Slot—Summary—Legacy Gateway 1 Shelf/Legacy Gateway 2
Shelf
9.28 PBX Configuration—[1-1] Configuration—Slot—Legacy-GW1/Legacy-GW2
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—KX-T7710

5.3.2 Trunk Adaptor Connection

Description
IP trunks (V-SIPGW and V-IPGW) can be connected to a Trunk Adaptor (KX-NS8188/KX-NS8290), which
allows the KX-NSX Series to connect to E1 trunks and PRI lines.

Conditions
[General]
• One IP trunk port corresponds to 1 channel in the Trunk Adaptor.
• An activation key is not required for ports whose Connection Attribute setting is set to Trunk Adaptor.
Ports without this setting can still be connected to a Trunk Adaptor, but they will require an activation key.
• One KX-NSX Series can connect to multiple Trunk Adaptors.
• For details about configuring settings for E1 trunks and PRI lines, refer to the documentation for the Trunk
Adaptor.
[Connection via SIP Trunk]

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5.3.2 Trunk Adaptor Connection

• The SIP trunk authentication ID and the authentication ID of the IP interface on the Trunk Adaptor must
be the same.
• The channel attribute of SIP trunks connecting to a Trunk Adaptor must be set to Basic channel.

PC Programming Manual References

9.11 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Port Property—Programming Port
Properties—Trunk Adaptor
9.11 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Port Property—Main
→ Connection Attribute
→ Channel Attribute
9.14 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property
→ Trunk Adaptor
→ Connection Attribute

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 5.4 E-mail Notification Features

5.4 E-mail Notification Features

5.4.1 E-mail Notification for Extension Users
Description
An e-mail can be sent to extension users, notifying them of events such as when they receive a new voice
message. Notifications can be sent for the following events:
• Missed trunk call
• New UM voice message
• New fax message
• New Automatic 2Way recoding message
• User password expiration notification

Conditions
• For extension users to receive notifications of new voice messages and/or faxes, they must have a
mailbox assigned to their extensions. (→ 20 UM Configuration—[1] Mailbox Settings)
• For extension users to receive notifications of missed trunk calls, the following settings must be
configured:
→ "Contact—Email 1–3" in 8.1.1 Users—User Container—Add User/Edit User
→ "Use for missed call notification" in 8.1.1 Users—User Container—Add User/Edit User
• To send e-mail notifications, the SMTP server settings must be configured.
• Up to 3 e-mail addresses can be registered for each extension user.
• The maximum file size of the e-mail attachment is 30 MB. Files that exceed this size cannot be sent as an
attachment.

PC Programming Manual References

8.1.1 Users—User Container—Add User/Edit User
20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Notification Parameters—E-mail/Text
Message Device—Device No. 1, 2, 3—E-mail Address
28.2.5 Network Service—[2-7] Server Feature—SMTP

Feature Manual References

3.2.1.29 Message Waiting Notification—E-mail Device

Operating Manual References

3.2.1 User Programming—Editing Settings in My Portal

5.4.2 E-mail Notification of System-level Events

Description
An e-mail can be sent to administrators or other specified e-mail addresses when certain system-level
events occur. Notifications can be sent for the following events:

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5.4.3 E-mail Notification of Sensor Alarm

Event Details
System alarm An e-mail is sent to users registered as an administrator, and to up to
two additional e-mail addresses.
Reported alarm information includes the following
• Error message which is detected by the KX-NSX main unit
• Expiration notification of the AK maintenance (you will be notified 10
days before expiration)
Software update Notifications can be sent for the following four types of software update
events:
1. A software update has become available on the update FTP
server.
2. A software update has been downloaded from the update FTP
server.
3. A software update has been successfully installed.
4. A software licence is about to expire.

Conditions
• To send e-mail notifications, the SMTP server settings must be configured.
• The system name in the e-mail notification can be configured through system programming.
→ 28.3.3 Network Service—[3-3] Client Feature—SNMP Agent—MIB info—SysName

PC Programming Manual References

5.1 System Control—Program Update
7.7 Utility—Email Notification
8.1.1 Users—User Container—Add User/Edit User
28.2.5 Network Service—[2-7] Server Feature—SMTP

Feature Manual References

5.5.8 Software Upgrading

5.4.3 E-mail Notification of Sensor Alarm

Description
An e-mail can be sent to a specified e-mail address when the external sensor detects an alarm. You can put
comments in the e-mail. If you put a URL for the network camera in the comments, the recipient of the
notification can monitor the video feed from the network camera by clicking URL in the comments.

Conditions
• Hardware Requirement:
An external sensor and a DOORPHONE card with Expansion Gateway. To connect an external sensor to
a legacy gateway (→ 5.3.1 Stacking Connection), refer to the documentation of the legacy gateway.
• To send e-mail notifications for a sensor alarm, a DOORPHONE card must be installed.
• A maximum 128 sensors can be connected to the PBX. An e-mail address can be set for each external
sensor.
• For users to receive sensor alarm notifications, the following settings must be configured:
– Notification e-mail for sensor alarm must be enabled

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13.5 PBX Configuration—[5-5] Optional Device—External Sensor—Sensor Alarm—Email notification

– E-mail address
13.5 PBX Configuration—[5-5] Optional Device—External Sensor—Sensor Alarm—Email Address
– Comment for e-mail
13.5 PBX Configuration—[5-5] Optional Device—External Sensor—Sensor Alarm—Email Comment

PC Programming Manual References

13.5 PBX Configuration—[5-5] Optional Device—External Sensor

Feature Manual References

2.18.3 External Sensor

5.4.4 E-mail Notification for Manager

Description
A manager can receive an e-mail notification when Automatic Two-way Recording is used. The recording
can be converted to an audio file and then sent as an e-mail attachment.
The destination of the e-mail is the e-mail address set for the Automatic Two-way Recording mailbox.

Conditions
• KX-NSXU004 (Activation Key for Two-way Recording Control) is required to use this feature.
• The e-mail is sent when the Automatic Two-way Recording has finished.
• If the destination e-mail address is changed during recording, the change will take effect immediately and
the recorded conversation will be sent to the new e-mail address when recording finishes.
• The maximum file size of the e-mail attachment is 30 MB. Files that exceed this size cannot be sent as an
attachment.
• The feature is enabled by setting an e-mail address for the mailbox. E-mail notifications will not be sent if
this address is not set.
→ 20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Notification Parameters

PC Programming Manual References

7.5.5 Utility—Report—E-mail Report
20.1.1 UM Configuration—[1-2] Mailbox Settings—Basic Setting—Notification Parameters
→ E-mail/Text Message Device—Device No. 1, 2, 3—User Name
→ E-mail/Text Message Device—Device No. 1, 2, 3—E-mail Address
→ E-mail/Text Message Device—Device No. 1, 2, 3—Only Urgent Messages
→ E-mail/Text Message Device—Device No. 1, 2, 3—Title Order
→ E-mail/Text Message Device—Device No. 1, 2, 3—Title String
→ E-mail/Text Message Device—Device No. 1, 2, 3—Callback Number
→ E-mail/Text Message Device—Device No. 1, 2, 3—Send Wait Time [0-120 min]
→ E-mail/Text Message Device—Device No. 1, 2, 3—Attach Voice File
→ E-mail/Text Message Device—Device No. 1, 2, 3—Use Mode
24.4 UM Configuration—[5-4] System Parameters—Parameters—E-mail Option

Feature Manual References

3.2.1.4 Automatic Two-way Recording for Manager

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5.5 System Data Control

5.5 System Data Control

5.5.1 PC Programming
Description
There are three levels of authorisation for programming the PBX, where each level controls which settings
the programming is allowed to access and change. The four levels are as follows:
Level Description Number of Accounts/Network
Installer For dealers and system installers 1
User For on-site managers 128
(Administrator)
User For individual site managers 32
(Sub-Administrator)
User (User) For end users KX-NSX1000: 1000
KX-NSX2000: 2000

Installers can assign Sub-Administrators for system maintenance of each site in a Multi-connection network.
Sub-Administrators can log in to Web Maintenance Console at the same time and programme the settings
for which they have authority.
In addition, the Installer can assign authority to each Sub-Administrator for the following settings:
– User Group
– DDI / DID Table
– Incoming Call Distribution Group
– Settings for Portable Station

Conditions
• Each account is assigned a password that is required to log in.
• Users can be added using the Add User Wizard.

CAUTION
To the Administrator or Installer regarding account passwords
1. Please provide all system passwords to the customer.
2. To avoid unauthorised access and possible abuse of the PBX, keep the passwords secret, and
inform the customer of the importance of the passwords, and the possible dangers if they become
known to others.
3. The PBX has no passwords set initially. For security, select an installer password as soon as the
PBX system is installed at the site.
4. Change the passwords periodically.
5. It is strongly recommended that passwords of 10 numbers or characters be used for maximum
protection against unauthorised access.

Installation Manual References

5.3 Starting Web Maintenance Console
5.5 Programming a Multi-connection Network

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PC Programming Manual References

2.1.1 Web Maintenance Console Accounts
2.1.2 Access Levels
8 Users/Supervisors/Maintenance Personnel
8.9 Maintenance Personnel—Sub Administrator Setting

Operating Manual References

3.2 System Programming Using Web Maintenance Console

5.5.2 PT Programming
Description
A PT user can perform the following programming:
a. Personal Programming: Customising the extension according to his needs.
b. Manager Programming: Customising specified frequently changing items (e.g., Charge Management
and Remote Extension Dial Lock).

Conditions
• COS programming determines whether personal programming can be performed.
• The extension(s) assigned as the manager COS can perform manager programming.
• During programming, the PT is considered to be busy.
• Only one manager programmer is allowed to perform manager programming at one time. The maximum
number of simultaneous programmers that each PBX supports is as follows:
– one manager programmer + 127 personal programmers
– 128 personal programmers
• Personal Programming Data Default Set
A user can return the items programmed on the telephone to default.

PC Programming Manual References

10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Manager—PT Programming
Mode
19.1 PBX Configuration—[11-1] Maintenance—Main—Password—Manager Password - PT Programming—
Prog *1

Feature Manual References

5.1.1 Class of Service (COS)
5.1.6 Manager Features
6.1 Capacity of System Resources

Operating Manual References

1.9.12 Clearing Features Set at Your Extension (Extension Feature Clear)
3.1 Customising Your Phone (Personal Programming)
4.1 Manager Programming

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5.5.3 DSP Resource Usage

5.5.3 DSP Resource Usage

Description
To digitally process audio signals, such as a telephone call, the PBX must use a certain number of DSP
(Digital Signal Processing) resources. DSP resources are provided by the DSP cards installed in the PBX.
Since there are a limited number of DSP resources, no further operations (e.g., telephone calls, playing an
OGM) can be performed if all resources are being used.
The following list shows some of the basic operations that require DSP resources.
• IP extension call
• IP trunk call
• Conference
• Accessing the Unified Messaging system (including recording calls)
• OGM playback
• Echo canceller (for trunk-to-trunk analogue calls)
For IP extension and trunk calls, the number of required resources differs depending on the codec (G.711 or
G.729) used.

Note
The examples in this section are intended to illustrate the concept of DSP resource usage. More
complex situations may necessitate additional resources, and in some cases fewer resources may be
necessary.
Examples of DSP resource usage
Fundamentally, the number of resources required for a given situation is the sum of the resources required
for each individual operation. The following examples illustrate DSP resource usage in various situations.

[IP trunk to IP extension]

X Y
PBX

IP Trunk G.729A G.711

IP-PT

If an IP trunk call using the G.729 codec requires X number of resources, and an IP extension call using the
G.711 codec requires Y number of resources, then the number of resources required for a call from an IP
extension to an IP trunk requires X + Y number of resources.

[Unified Messaging access]

X
Y
UM

G.711

PBX IP-PT

Playing back messages from or recording messages to the Unified Messaging (UM in the figure above)
system requires DSP resources, X in this example, in addition to the resources required for the G.711 codec
(Y). The total cost is X + Y.

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[Conference call]
X
W
IP Trunk G.711

X G.729A
Y
G.711 IP-PT

X Conference
Z
G.711
X Analogue

IP Trunk G.711 SLT

PBX

A conference requires additional resources, Y, for handling the multiple voice channels. Also, in standard
two-way conversations, analogue lines generally do not require any DSP resources, but in a conference
they do. In addition, IP trunks in a conference require additional DSP resources.
For this example, then, the number of required resources is X + X + X + X + Y + Z + W.
DSP usage graph
The PBX keeps a record of the maximum DSP usage per hour for each of the following features/services. A
graph can then be displayed in Web Maintenance Console showing trends in DSP usage over time, as well
as the number of calls and operations that could not be performed due to lack of resources.
• VoIP (IP trunk, IP extension and IP-CS usage)
• Conference
• Unified Messaging
• OGM
• Two-way Recording
The graph also shows the amount of free resources and the total resource use.
DSP resources can be reserved for and, therefore, guaranteed to be available for each service. For details
about reserving DSP resources, see 5.5.3.1 DSP Resource Reservation. If the DSP resources reserved for
a service are all used, free (unreserved) DSP resources can be used. The DSP resource usage graph
shows the DSP resources reserved for each service, the free DSP resources, and the total DSP resources
available on a per-hour basis. An example of the DSP resource usage graph is shown below.

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5.5.3 DSP Resource Usage

[Example]

Conditions
[General]
• Most internal VoIP calls require no resources because IP phones connect over a P2P connection
(→ 5.2.3 Peer-to-Peer (P2P) Connection). That is, the phones themselves do the signal processing
without consuming PBX resources. The PBX simply performs the initial connection.
• Telephones and trunk lines connected via a legacy gateway (→ 5.3.1 Stacking Connection) use the same
number of resources as telephones and trunks connected directly to the PBX.
[DSP usage graph]
• The most recent 30 days of DSP usage is recorded.
• The scale of the graph can be set to 1-hour, 4-hour, or 24-hour intervals.
• The maximum value of the vertical axis is the maximum recorded DSP usage value, and not the available
DSP resource capacity.
• To view the number of times DSP resources are measured as busy, see Counting of DSP busy times.
• The state of DSP resource usage can be outputted as a CSV file by clicking Export. In this data it is
possible to view the busy status of reserved DSP resources for each service, as well as the data shown in
the graph.
• If DSP resources are not reserved for each service, all DSP resources will be taken from free DSP
resources, and will be treated as free resources in the graph. In this case, the lines for free DSP
resources and total DSP resources will be the same. To accurately measure the amount of DSP
resources used for each service, reserve DSP resources for each service.

Installation Manual References

2.3.2 System Capacity—DSP Resources
4.3.3 DSP S Card (KX-NS0110), DSP M Card (KX-NS0111), DSP L Card (KX-NS0112)

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PC Programming Manual References

9.14.2 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Port Property—DN to Gateway—GW
Settings–Codec—IP Codec Priority—1st, 2nd, 3rd
9.5 PBX Configuration—[1-1] Configuration—Slot—Site Property—Main
→ DSP CODEC G.711 only (SIP extension)
→ DSP CODEC G.711 only (IP-GW)
→ DSP CODEC Priority-1 value only (others)
9.12 PBX Configuration—[1-1] Configuration—Slot—V-IPGW128—Card Property
→ Voice Codec Priority 1st
→ Voice Codec Priority 2nd
→ Voice Codec Priority 3rd
9.11 PBX Configuration—[1-1] Configuration—Slot—V-SIPGW128—Port Property—Voice/FAX—IP Codec
Priority—1st, 2nd, 3rd
9.16 PBX Configuration—[1-1] Configuration—Slot—V-IPEXT128—Port Property—Option—IP Codec
Priority
9.20 PBX Configuration—[1-1] Configuration—Slot—V-IPCS16—Port Property—Option—IP Codec Priority
9.33 PBX Configuration—[1-5] Configuration—DSP Resource

Feature Manual References

2.14.2 Conference
2.28.2 Outgoing Message (OGM)
3.2 System and Subscriber Features
3.2.2.34 Two-way Record/Two-way Transfer
3.2.1.4 Automatic Two-way Recording for Manager
5.2.1 IP Proprietary Telephone (IP-PT)
5.2.2 SIP (Session Initiation Protocol) Extension

5.5.3.1 DSP Resource Reservation

Description
A number of resources can be reserved for particular features or services to guarantee a minimum level of
service. Resources reserved for a particular service (e.g., conferencing) cannot be used for another service
(e.g., Unified Messaging).
For example, one may want to reserve resources for OGM (Outgoing Message) to ensure that recorded
messages can be played to incoming calls.
The resources can be reserved for the following types of services:
• VoIP (G.711)
• Conference trunk
• Unified Messaging
• Two-way Recording
• OGM

Note
It is not necessary to reserve resources for a feature to use it. In normal operation, free DSP resources
are allocated on a first-come first-serve basis. Resources should be reserved only if a minimum level of
performance is required for your system.
For example, reserving resources for Two-way Recording (→ 3.2.1.4 Automatic Two-way Recording for
Manager, → 3.2.2.34 Two-way Record/Two-way Transfer) also reserves UM ports. This can have the
unintended effect of blocking access to the Unified Messaging system even when no recording is being

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5.5.3 DSP Resource Usage

performed. Therefore, resources for Two-way Recording should be reserved only if it is necessary to
guarantee that Two-way Recording can be performed. (→ 3.1.1 Unified Messaging System Overview)
Resource Reservation Example
The following table shows the number and types of resources that must be reserved for a given workload.
To calculate the number of free (i.e., non-reserved) resources, a DSP M card (127 DSP resources) is
assumed.
Minimum Resources per unit of No. of required DSP
Service
performance*1 performance resources
VoIP call (G.711) 40 calls 1 40
Unified Messaging *2
5 operations 1.3 6.5
Two-way Recording 3 recordings 2.3 6.9
OGM 10 playbacks 2 20
Conference trunk 10 conferences 0.5 5
Unified Messaging
2 tones 2 2
tone*3
Total Reserved Resources 80.4
Free Resources *4
47

*1 Minimum performance refers to the minimum number of simultaneous operations.

*2 Unified Messaging operations include operations such as users (subscribers) accessing their mailboxes and outside callers
leaving messages in subscribers’ mailboxes.
*3 The resources for Unified Messaging tones (used for Two-way Recording) are reserved automatically and cannot be released.
*4 Free Resources = DSP capacity – Total Reserved Resources, rounded to the nearest whole number.

If the PBX’s resources are reserved as shown in the example above, the resources required to meet the
numbers of operations listed in the "Minimum performance" column are guaranteed to be available. Note,
however, that for an operation such as a conference call, DSP resources are required for each individual
conference party in addition to the resources required for the conference trunk itself. Therefore, if all 40 VoIP
units as well as all free resources are being used, a new conference call cannot be established, even if
sufficient conference resources are available.

Conditions
• The total number of resources provided by each type of DSP card is as follows:
– DSP S card: 63
– DSP M card: 127
– DSP L card: 254

PC Programming Manual References

9.33 PBX Configuration—[1-5] Configuration—DSP Resource

5.5.3.2 DSP Resource Advisor

Description
Web Maintenance Console provides a tool for calculating the number of resources required for a given set
of operating conditions. The Web Maintenance Console user provides information such as the number of
ports for a given resource (e.g., 16 extension ports using the G.729 codec) and the expected usage load
(e.g., 50% busy), and the resource advisor calculates the number of DSP resources required to meet those
conditions.

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This tool can be used in offline mode to simulate various PBX configurations and usage cases to help
determine the number and size of DSP cards required.
This tool will also recommend which and how many resources to reserve for various features
(→ 5.5.3.1 DSP Resource Reservation). The recommended settings can be applied immediately from the
resource advisor tool.
The resource usage can be calculated using the following types of services and features:
Service/Feature DSP Resources per Unit
Trunk using G.729 codec 2.2
Trunk using G.711 codec 1
Non-IP trunk (ISDN trunk, analogue trunk, etc.) 1
Extension using G.729 codec 2.2
Extension using G.711 codec 1
IP-CS extension using G.729 codec 2.2
IP-CS extension using G.711 codec 1
Unified Messaging *1
1.3
Two-way Recording*1*2 2.3
OGM *1
2
Conference trunk*1 0.5

*1 The DSP costs of the extensions/trunks involved in the operation are not included in the per-unit DSP resource count.
*2 Two-way Recording also requires a conference trunk. For an example of the required DSP resources, see 5.5.3 DSP Resource
Usage.
Example 1: Small Office
In a small office (e.g., 32 employees), the necessary number of trunks and extensions is likely to be
relatively small. In addition, the expected load on the system will also be small.
(For clarity, unused services are not included in the table.)
Service Number of Ports Load (Busy Ratio %) DSP Cost*1
Trunk using G.729 codec 3 5% 0.3
Non-IP trunk 1 5% 0.05
Extension using G.729 codec 32 10% 7.0
Unified Messaging 4 — 5.2
OGM 2 — 4.0
Conference 4 — 2.0
Total DSP Cost 18.55

*1 DSP Cost = Number of Ports × Resource cost per port (unit) × Load

In the example above, the total DSP cost is 18.55. In such an environment, a PBX with a DSP S card (max.
63 DSP resources) would be sufficient.
Example 2: Call Centre
In a call centre, both the number of trunks and number of extensions are likely to be very high. Also, since
employees are constantly receiving calls, the system load will be high. Furthermore, calls are often recorded
at call centres to provide quality-of-service monitoring.

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5.5.3 DSP Resource Usage

Service Number of Ports Load (Busy Ratio %) DSP Cost*1

Trunk using G.729 codec 8 50% 8.8
Trunk using G.711 codec 128 80% 102.4
Extension using G.729 codec 32 50% 35.2
Extension using G.711 codec 128 80% 102.4
IP-CS using G.729 codec 8 50% 8.8
Unified Messaging 8 — 10.4
Two-way Recording 4 — 9.2
OGM 4 — 8.0
Conference 12 — 6.0
Total DSP Cost 291.2

*1 DSP Cost = Number of Ports × Resource cost per port (unit) × Load

In this example, the total DSP cost is 291.2. In this case, two DSP cards are necessary: at minimum one
DSP L card (max. 254 DSP resources) and one DSP S card (max. 63 DSP resources).

Conditions
• Calls that are established via P2P (→ 5.2.3 Peer-to-Peer (P2P) Connection) do not use the PBX’s DSP
resources, so they may be excluded from the usage calculation.

PC Programming Manual References

9.33 PBX Configuration—[1-5] Configuration—DSP Resource

5.5.3.3 Resource Depletion Indication Monitoring

Description
The resource usage for the past a week can be collected and viewed using Web Maintenance Console.
In addition, if the NSX server detects a system error indication, a notification can be sent to the System
Administrator to prevent a system failure.

Conditions
• KX-NSX2201 (Maintenance Annual Key) is required to use this feature.
If the activation key is not installed or expires, the ability to reference stored information, record error logs,
and receive e-mail notification of failures through Web Maintenance Console is not available.
• The peak value of the resource usage for each hour is recorded.
• When the peak rate (resource usage peak / available resources) exceeds a specified threshold, an error
will be recorded in the error log.
The threshold can be programmed for each type of resources through Web Maintenance Console.
(Default: 80%)
To disable the recording of the error log, set the threshold to None. For more information, refer to 7.4.12.1
Utility—Monitor/Trace—System Resource Usage—Setting in PC Programming Manual.
• Errors are logged periodically until the resource usage rate becomes equal to or less than the threshold.
The interval for logging errors can be programmed through Web Maintenance Console. (Default: 24 hour)
• If an error occurs, an e-mail can be sent to the System Administrator.
See 5.4 E-mail Notification Features.

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• If the system is initialised, the recorded data will be cleared. If the system is shut down or restarted, the
data will not be cleared.

Resource information to record

• The collected resource information is recorded for each type of resource.
• Data for the NSX server or expansion gateways is recorded depending on the resource type. All the
recorded data is managed on the NSX server.
• The following table indicates what information is collected for each type of resource:
Conditions for
Resource type Recorded information Solutions
error log recording
SMDR Buffer Total capacity: Total When the peak rate Output the SMDR
(Only information of NSX capacity of the buffer exceeds the information using an
server is recorded) Peak value: The amount of threshold. SMDR device to
information stored in the clear the buffer.
buffer
Hotel Billing Buffer*1 When the room
Peak rate: Buffer usage
(Only information of NSX status of the
rate (Peak value / Total
server is recorded) extension is switched
capacity)
to Check-in, the hotel
billing information of
the extension will be
cleared.
DSP VoIP (G. Assigned resources: The When the peak rate • Change the
Resource 711) number of DSP resources (Peak value / allocation of DSP
(Recorded reserved for each service Assigned resources) resources.
Unified
for each Free resources: The exceeds the
message • Add another DSP
site) number of DSP resources threshold.
card.
Two-way that are not reserved
Recording Peak value: Total number Note
(Automatic of resources used for each The peak rate
recording service (including DSP may exceed
and resources that are used 100%.
conversation from the free resources)
recording for
each user)
OGM
Conference
trunk
Total DSP Total resources: Total When the peak rate
Resource number of DSP resources (Peak value / Total
Free resources: The resources) exceeds
number of DSP resources the threshold.
that are not reserved
Peak value: Total number
of used DSP resources

*1 If the ACD report feature is enabled, the hotel billing buffer will not be recorded.

PC Programming Manual References

7.4.12.1 Utility—Monitor/Trace—System Resource Usage—Setting

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5.5.4 Automatic Setup

7.4.12.2 Utility—Monitor/Trace—System Resource Usage—Display

7.7.1 Utility—Email Notification—Alert
9.33 PBX Configuration—[1-5] Configuration—DSP Resource

5.5.4 Automatic Setup

Description
There are two automatic setup features as follows:
1. Automatic ISDN Configuration
2. Automatic Time Adjustment

1. Automatic ISDN Configuration

The ISDN (BRI) port configuration can be set automatically through system programming.
The following items will be set by making and receiving a call using the subscriber number assigned for
each ISDN (BRI) port:
a. L1 Active Mode
b. L2 Data Link Mode
c. Access Mode (Point-to-Point/Point-to-Multipoint)
d. TEI Assignment Mode (Fix 00–63/Automatic)
2. Automatic Time Adjustment
It is possible to adjust the PBX clock automatically in the following two ways:
a. Summer Time (Daylight Saving Time) Setting:
The start and end dates of the summer time can be programmed. The PBX clock will be adjusted
(one hour forward or backward) at 2:00 AM of the programmed date, if enabled through system
programming. It means 2:00 AM will become 3:00 AM on the start date of the summer time, and
2:00 AM will become 1:00 AM on the end date.

Note
If the Timed Reminder (Wake-up call) is set;
– On the summer time start date, the setting between 2:00 AM and 3:00 AM will not happen.
– On the summer time end date, the setting between 1:00 AM and 2:00 AM will ring twice.
b. Time Information from Telephone Company:
Time information can be received on the following calls:
• An incoming or outgoing call through an ISDN line
• An incoming call through an analogue line with Caller ID which includes the time information.
The PBX clock will be adjusted every day with the first call after 3:05 AM, if enabled through
system programming.

Note
If the Timed Reminder (Wake-up call) is set, the setting will not happen or will ring twice
depending on the adjustment.
c. Time Information through Network Time Protocol (NTP):
By connecting the PBX to an NTP server, it is possible to receive and update the time setting.
The PBX clock will be adjusted every day at 3:05 AM, if enabled through system programming.

Conditions
[General]

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 5.5.5 Dynamic Host Configuration Protocol (DHCP) Server

• Through system programming, it is possible to specify NTP, ISDN, or neither method as the selected
method of automatic time adjustment.
• SMDR will record the call information using the PBX clock so that the recording time will be overlapped at
the end of summer time. (→ 2.22.1.1 Station Message Detail Recording (SMDR))
[NTP Time Information]
• The time set through NTP will apply the same to all PTs connected to the PBX, regardless if an IP
extension is located in another time zone.

PC Programming Manual References

6.2 Tool—BRI Automatic Configuration
10.1.1 PBX Configuration—[2-1-1] System—Date & Time—Date & Time Setting
10.1.2 PBX Configuration—[2-1-2] System—Date & Time—SNTP / Daylight Saving
10.1.2.1 PBX Configuration—[2-1-2] System—Date & Time—SNTP / Daylight Saving—Daylight Saving
28.2.4 Network Service—[2-6] Server Feature—NTP—NTP server

Feature Manual References

4.1.2.1 Integrated Services Digital Network (ISDN)—SUMMARY
2.24.3 Timed Reminder

5.5.5 Dynamic Host Configuration Protocol (DHCP) Server

Description
The PBX has a built-in DHCP server. When the DHCP server is enabled, the PBX will automatically assign
IP addresses to other devices on the network, such as IP-PTs.
Using a DHCP server simplifies network management by removing the need to assign IP addresses to
devices manually.

Conditions
• The DHCP Server feature cannot be used if the PBX’s IP address assignment mode is set to DHCP.
• If the PBX’s DHCP server is enabled, make sure that no other DHCP servers are running on the same
network. Having more than one DHCP server on a network can result in network errors.
• For the following settings, the PBX delivers the settings of its LAN port to devices: subnet mask, default
gateway address, and DNS server addresses.

PC Programming Manual References

28.2.1 Network Service—[2-1] Server Feature—DHCP(LAN)

5.5.6 Flexible Numbering/Fixed Numbering

Description
To dial another extension user or to access PBX features, the access numbers (extension numbers or
feature numbers) are required.
There are three types of numbering plans:
1. Flexible Numbering (available while a dial tone is heard)
2. Flexible Numbering (available while a busy, DND, or ringback tone is heard)
3. Fixed Numbering (available while dialling or talking)

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1. Flexible Numbering (available while a dial tone is heard)

Extension numbers and feature numbers which are available while a dial tone is heard can be
customised for easy use. The numbers must not conflict. It is also possible to use the default settings
shown in the following table.
a. Extension Numbers: An Extension Numbering Plan of 3 digits, 4 digits or 5 digits can be
selected as the system basis. However, only 3 digits, or 4 digits can be specified during Easy
Setup. Extension numbers consist of leading numbers and additional numbers. Extension
numbers (consisting of "0" through "9") can be assigned as follows:
• Numbering schemes: 1-64
• Leading number: up to three digits
• Additional number: up to two digits (default: two digits)
b. Feature Numbers: A number of up to four digits, consisting of "0" through "9", " ", and "#"

c. Other PBX Extension Numbers (Other PBX Extension Number [TIE] -1 through 16): A
number of up to three digits, consisting of "0" through "9", " ", and "#"
[Flexible Numbering Table (available while a dial tone is heard)]
• Feature Numbers and Other PBX Extension Numbers
• Extension Numbers
Default

Feature When 3-digit When 4-digit

numbers are numbers are
selected selected
Extension Numbering Scheme 1—Leading Number 1/2 10/20
Extension Numbering Scheme 2—Leading Number 2/3 11/21
Extension Numbering Scheme 3—Leading Number 3/4 12/22
Extension Numbering Scheme 4—Leading Number 4/1/9 13/23
Extension Numbering Scheme 5—Leading Number None 14/24
Extension Numbering Scheme 6—Leading Number None 15/25
Extension Numbering Scheme 7—Leading Number None 16/26
Extension Numbering Scheme 8—Leading Number None 17/27
Extension Numbering Scheme 9—Leading Number None 18/28
Extension Numbering Scheme 10—Leading Number None 19/29
Extension Numbering Scheme 11—Leading Number None 20/30
Extension Numbering Scheme 12—Leading Number None 21/31
Extension Numbering Scheme 13—Leading Number None 22/32
Extension Numbering Scheme 14—Leading Number None 23/33
Extension Numbering Scheme 15—Leading Number None 24/34
Extension Numbering Scheme 16—Leading Number None 25/35
Extension Numbering Scheme 17—Leading Number None 26/36
Extension Numbering Scheme 18—Leading Number None 27/37

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 5.5.6 Flexible Numbering/Fixed Numbering

Default

Feature When 3-digit When 4-digit

numbers are numbers are
selected selected
Extension Numbering Scheme 19—Leading Number None 28/38
Extension Numbering Scheme 20—Leading Number None 29/39
Extension Numbering Scheme 21—Leading Number None 50
Extension Numbering Scheme 22—Leading Number 5 51
Extension Numbering Scheme 23—Leading Number None 52
Extension Numbering Scheme 24—Leading Number None 53
Extension Numbering Scheme 25—Leading Number None 54
Extension Numbering Scheme 26—Leading Number None 55
Extension Numbering Scheme 27—Leading Number None 56
Extension Numbering Scheme 28—Leading Number None 57
Extension Numbering Scheme 29—Leading Number None 58
Extension Numbering Scheme 30—Leading Number None 59
Extension Numbering Scheme 31—Leading Number 6 60
Extension Numbering Scheme 32-64—Leading Number None None

Feature Default
Operator Call *1
9/0
Idle Line Access (Local Access)*1 0/9
Trunk Group Access *1
8
TIE Line Access*1 7
Redial*1 #
Speed Dialling—System/Personal*1
Personal Speed Dialling—Programming*1 30
Doorphone Call*1 31
Conference Group Call*1 32
Group Paging *1
33
External BGM on/off*1 35
Outgoing Message (OGM) playback/record/clear 36
S-CO Line Access*1 37
Parallel Telephone (Ring) Mode set/cancel 39
Group Call Pickup*1 40
Directed Call Pickup *1
41
TAFAS—Calls through an External Pager 42

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5.5.6 Flexible Numbering/Fixed Numbering

Feature Default
Group Paging answer*1 43
Automatic Callback Busy cancel/CCBS cancel 46
User Remote Operation/Walking COS/Verification Code Entry 47
Account Code Entry *1
49
Call Hold/Call Hold Retrieve 50
Call Hold Retrieve—Specified with a Holding Extension Number*1 51
Call Park/Call Park Retrieve*1*2 52
Call Hold Retrieve—Specified with a Held Trunk Number*1 53
Door Open*1 55
External Relay 56
External Feature Access 60
SIP Refer (Blind)*3 61
ISDN Hold 62
COLR set/cancel*1 7 0
CLIR set/cancel *1
7 1
Switch CLIP/COLP of the Trunk/Extension*1 7 2
MCID 7 3
ISDN-FWD set/cancel/confirm 7 5
Message Waiting set/cancel/callback 70
FWD/DND set/cancel—Both*1 710
FWD/DND set/cancel—External *1
711
FWD/DND set/cancel—Internal*1 712
FWD/DND No Answer Timer set *1
713
Group FWD set/cancel—Both*1 714
Group FWD set/cancel—External *1
715
Group FWD set/cancel—Internal*1 716
Call Pickup Deny set/cancel*1 720
Paging Deny set/cancel*1 721
Main device Service-in 722
Sub device Service-in 723
PS Service-in 724
Service-out 725
Device in use (for CTI)*4 726
Data Line Security set/cancel *1
730
Manual Call Waiting for Extension Call off/BSS/OHCA/Whisper OHCA*1 731

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 5.5.6 Flexible Numbering/Fixed Numbering

Feature Default
Automatic Call Waiting set/cancel*1 732
Executive Busy Override Deny set/cancel *1
733
Not Ready Mode on/off*1 735
Log-in/Log-out *1
736
Incoming Call Queue Monitor 739
Hot Line programme/set/cancel 740
Absent Message set/cancel*1 750
BGM set/cancel 751
Remote Wake-up Call 76
Timed Reminder set/cancel 760
Printing Message 761
Extension Dial Lock set/cancel*1 77
Time Service Switch *1
780
Remote Extension Dial Lock off*1 782
Remote Extension Dial Lock on *1
783
Trunk Busy Out Clear 785
Extension Feature Clear *1
790
Extension Personal Identification Number (PIN) set/cancel*1 799
Dial Information (CTI) None
Other PBX Extension Number (TIE) 1–16 None
Quick Dialling *5
None

2. Flexible Numbering (available while a busy, DND, or ringback tone is heard)

Feature numbers which are available while a busy, DND, or ringback tone is heard can be customised
for easy use. The numbers should be one digit ("0" through "9", " ", or "#") and must not conflict. For
default, refer to the following table:
[Flexible Numbering Table (available while a busy, DND, or ringback tone is heard)]
Feature Default
Call Waiting/DND Override*6 1 or 2*7
Executive Busy Override*6 3
Message Waiting set*6 4
Call Monitor*6 5
Automatic Callback Busy/CCBS 6
Alternate Calling—Ring/Voice

3. Fixed Numbering (available while dialling or talking)

The features which are available while dialling or talking have fixed numbers as shown in the following
table:

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5.5.6 Flexible Numbering/Fixed Numbering

[Fixed Numbering Table (available while dialling or talking)]

Feature Fixed Numbering
Pulse to Tone Conversion
Conference*8 3
Door Open*9 5

*1 SIP extension users can use these feature numbers except Parallel Telephone Mode, incoming call queue monitor, BGM Set/
Cancel, Timed Reminder.
*2 From a SIP extension, this feature number can be used only for Call Park Retrieve.
*3 This feature number is used for the SIP-carrier transfer feature. For more information, see 2.12.2 SIP Refer Transfer.
*4 You cannot configure a public device as a CTI Control target.
*5 It is possible to register Quick Dialling numbers that overlap with other registered numbers. This is used for the Automatic
Rerouting of VoIP Calls To Public Trunk feature.
*6 SIP extensions cannot perform DND Override, Executive Busy Override, Message Waiting, Parallel Telephone Mode, Alternate
calling, or Call Monitor, but can be the recipient of them.
*7 To use Call Waiting/DND Override, both "1" and "2" are available by default.
*8 SIP extensions cannot establish conferences but can participate in them.
*9 SIP extension users can use these feature numbers.

Conditions
• All features have a default feature number.
• The following are examples of feature number conflicts: 1 and 11, 0 and 00, 2 and 21, 10 and 101, 32 and
321, etc.
• Feature number + Additional number (Parameter)
Some flexible feature numbers require additional digits to make the feature active. For example, to set
Call Waiting, the feature number for "Call Waiting" must be followed by "1" and to cancel it, the same
feature number should be followed by "0".
• If a feature number includes " " or "#", rotary SLT users cannot use it.
• ISDN extension users cannot use the following features:
– OGM playback/record
– Call Hold/Call Hold Retrieve (held at its own extension)
– ISDN Hold
– MCID
– Call Waiting
– Hot Line
– Timed Reminder
– Executive Busy Override
– Call Monitor
– Automatic Callback Busy/CCBS
• PS users cannot use the following features:
– Personal Speed Dialling
– OGM playback/record
– S-CO Line Access

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 5.5.7 Floating Extension

– Parallel Telephone Mode set/cancel

– BGM set/cancel
– Timed Reminder
• SIP extension users cannot use the following features:
– Parallel Telephone (Ring) Mode set/cancel
– Outgoing Message (OGM) playback/record/clear
– TAFAS—Calls through an External Pager
– Call Hold/Call Hold Retrieve
– Call Park/Call Park Retrieve
– External Relay
– External Feature Access
– ISDN Hold
– MCID
– ISDN-FWD set/cancel/confirm
– Main device Service-in
– Sub device Service-in
– PS Service-in
– Service-out
– Hot Line programme/set/cancel
– Printing Message
– Trunk Busy Out Clear

PC Programming Manual References

10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main
→ Extension
→ Features
→ Other PBX Extension
10.6.2 PBX Configuration—[2-6-2] System—Numbering Plan—Quick Dial
10.6.3 PBX Configuration—[2-6-3] System—Numbering Plan—B/NA DND Call Feature

Feature Manual References

3.1.1 Unified Messaging System Overview
5.2.4.1 Portable Station (PS) Connection
6.1 Capacity of System Resources

Operating Manual References

5.3.1 Feature Number Table

5.5.7 Floating Extension

Description
Virtual extension numbers can be assigned to resources to make them appear as extensions. This feature is
also known as Floating Station.

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5.5.8 Software Upgrading

These numbers are defined as floating extension numbers and can be assigned as a destination of
incoming calls etc.
Resource Description Default
Device External Pager Used as the destination for TAFAS feature. 600 or 6000*1
(→ 2.17.2 Trunk Answer From Any Station
(TAFAS))
Outgoing Message Used to send messages for DISA feature. 536–599 or
(OGM) (→ 2.16.1 Direct Inward System Access 58 + two-digit
(DISA)) OGM
number*1
Group Incoming Call Used to call an incoming call distribution 6 or 60 +
Distribution Group group. (→ 2.2.2.1 Incoming Call Distribution two-digit
Group Features—SUMMARY) group
number*1*2
PS Ring Group Used to call a PS ring group. (→ 5.2.4.2 PS
—
Ring Group)
UM Group Used to call a UM group. 500 or 5000*1

*1 The default floating extension number depends on the value specified for Numbering Plan in Easy Setup.
*2 A default floating extension number is provided only up to group 64. The floating extension number for groups 65 and higher must
be set explicitly.

Conditions
• It is possible to give names to floating extension numbers.

PC Programming Manual References

2.1.4 Easy Setup Wizard
10.6.1 PBX Configuration—[2-6-1] System—Numbering Plan—Main—Extension
11.5.1 PBX Configuration—[3-5-1] Group—Incoming Call Distribution Group—Group Settings—Main
→ Floating Extension Number
→ Group Name
11.7.2 PBX Configuration—[3-7-2] Group—UM Group—Unit Settings—Floating Extension No.
11.8 PBX Configuration—[3-8] Group—PS Ring Group—Floating Extension Number
13.2 PBX Configuration—[5-2] Optional Device—External Pager—Floating Extension Number
13.3.2 PBX Configuration—[5-3-2] Optional Device—Voice Message—DISA Message—Floating Extension
Number

Feature Manual References

3.1.1 Unified Messaging System Overview
5.5.1 PC Programming

5.5.8 Software Upgrading

Description
The main software of the PBX, as well as the software of other connected devices can be updated either
manually or automatically.
• Obtaining updates manually

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 5.5.8 Software Upgrading

Obtaining software updates (downloading the update to the PBX) can be done manually via Web
Maintenance Console. In this case, software updates can be obtained from an FTP server, a USB
memory device connected to the PBX, a NAS, or a PC that can access Web Maintenance Console.
• Obtaining updates automatically
The PBX can automatically check for and download updates from an FTP server. Also, a notification e-
mail can be sent to specified e-mail addresses when an update becomes available and when it is
downloaded.
Also, the PBX can be configured to check for updates automatically, but not to download them.
The installation of the updates needs to be performed manually at the Web Maintenance Console.
The software of the following types of devices and components can be updated:
Data Type Description
Main software data Operating system data area on the PBX’s mother board.
The main software data of other PBXs connected to a
Multi-Connection network as Expansion Gateways is also
updated.
LPR (software on a slot card) software Flash ROM on a slot card (e.g., BRI4+SLC2)
data This includes the LPR software of legacy gateways
(→ 5.3.1 Stacking Connection).
Cell Station (CS) and Portable Station Flash ROM on a CS and/or PS
(PS) software data*1 This includes the Flash ROM on CSs connected through a
legacy gateway (→ 5.3.1 Stacking Connection).
IP-PT software data*2 Firmware of supported IP-PTs
Expansion gateway software data In a Multi-Connection network (→ 4.2 Multi-connection
Networking), the main software of Expansion gateways
and the software of devices connected to Expansion
Gateways can be transferred from the Master unit.
Also software can be upgraded directly from Web
maintenance console of Expansion gateway(Manual
upgrade only).
Software of FAX card is updated at the time that software
of Expansion gateways is updated manually only.

*1 Only available for certain PSs.

*2 Only Panasonic telephones are supported. For details about a specific telephone, refer to the telephone’s documentation.
PS Air Download
The firmware of certain PSs can be updated via an IP-CS. The following equipment supports this feature:
• IP-CS: KX-NS0154
• PS: KX-TCA185, KX-TCA285, KX-TCA385

Conditions
• The software version of the mother board can be confirmed through system programming.
• In a Multi-Connection network (→ 4.2 Multi-connection Networking), the main software of all units (Master
units and Expansion gateways) must be the same.

Installation Manual References

4.3.1 Mother Board

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5.5.8 Software Upgrading

PC Programming Manual References

5.1 System Control—Program Update
9.2 PBX Configuration—[1-1] Configuration—Slot—Summary—Physical Shelf

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 5.6 Fault Recovery/Diagnostics

5.6 Fault Recovery/Diagnostics

5.6.1 Reserved
5.6.2 Power Failure Restart
Description
When turning the electricity back on, the PBX restarts the stored data automatically and the PBX will record
the event (System Restart) in the error log.

Conditions
• In the event of a power failure, PBX memory is protected by a factory-provided lithium battery. There is no
memory loss except the memories of Automatic Callback Busy (Camp-on) (→ 2.10.1 Automatic Callback
Busy (Camp-on)) and Call Park (→ 2.13.2 Call Park).

5.6.3 Local Alarm Information

Description
When a PBX error occurs and the PBX detects it, the System Alarm button light on the PT of an extension,
which is allowed to use this feature through system programming (a maximum of two extensions per PBX),
turns on red. Pressing the button will show the error number on the display. If multiple errors occur, the error
number will be displayed in order of highest priority to lowest. The System Alarm button light turns off
automatically after all error numbers have been displayed.
For details about the errors and their solutions, refer to the Installation Manual.
[Error Example]

ERR #100 (01 10000)

(1) (2)

[Explanation]
Number in the Example Item Description
(1) Error Code Shows three-digit error code.
(2) Sub Code Shows 10-digit sub code (BBBWXYYZZZ).
BBB: Site number (000 to 031)
W: Slot type (Physical shelf: blank, Virtual shelf: *)
X: Unit number/Non-PBX process code
YY: Slot number/Process code
ZZZ: Port number/Process number

Conditions
• System Alarm Button

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5.6.4 Simple Network Management Protocol (SNMP) System Monitor

A flexible button can be customised as the System Alarm button.

• The alarm information will be recorded on SMDR, if enabled through system programming.
• The PBX can be automatically diagnosed at a preprogrammed time every day.

Installation Manual References

7.1.4 Troubleshooting by Error Log
6.4.2 Multi-connection Networking Survivability

PC Programming Manual References

7.3.1 Utility—Log—Error Log
8.3.1 Users—Flexible Button—Wired Extension—Type
8.3.2 Users—Flexible Button—Portable Station—Type
19.1 PBX Configuration—[11-1] Maintenance—Main
→ SMDR—Print Information—Error Log
→ Maintenance—Local Alarm Display—Extension 1, Extension 2
→ Maintenance—Daily Test Start Time—Set
→ Maintenance—Daily Test Start Time—Hour
→ Maintenance—Daily Test Start Time—Minute

Feature Manual References

2.21.2 Flexible Buttons
2.22.1.1 Station Message Detail Recording (SMDR)

5.6.4 Simple Network Management Protocol (SNMP) System

Monitor
Description
It is possible for a PC assigned as an SNMP manager to manage and receive PBX system status
information, such as alarm information and general system activity using SNMP. Management Information
Bases (MIBs) are sent to a PC (i.e., the SNMP manager) connected to the PBX over a LAN and can then be
stored and analysed using SNMP manager software.
The two features for managing information using SNMP are as follows:
– Polling:
A bilateral transaction of information. Polling allows the manager to request information from the PBX.

PBX
Request
PC

Response Manager

– TRAP:
An automatic relay of information from the PBX when a status change occurs or an alarm is detected.

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 5.6.4 Simple Network Management Protocol (SNMP) System Monitor

PBX
Sends Information
PC

Manager

TRAP Implementation
The PBX will send the two types of TRAP as follows:
Type TRAP Name Description
Standard TRAP coldStart Information is sent after turning on the power of the
PBX or resetting the PBX.
Authentication Failure Information is sent when an unregistered Community
Name and/or Manager IP address is entered.
Enterprise Specific Major Alarm Information is sent when a major alarm is detected.
TRAP*1
Minor Alarm Information is sent when a minor alarm is detected.

*1 Enterprise Specific TRAPs contain information exclusive to the PBX model (Enterprise Specific MIB).

Conditions
• Through system programming, it is possible to enable or disable this feature.
• Up to 2 SNMP managers can be assigned.
• This PBX supports SNMP Protocol Version 1.0, 2.0c, 3.0 and SNMP Version 1.0-TRAP.
• This PBX can only receive read-only MIBs. Write MIBs are not supported.
• This PBX supports MIB II.
• For more information regarding major and minor alarms, refer to the Installation Manual.
• For a list of the MIB object groups supported by this PBX, refer to 6.4 Supported Management
Information Base (MIB) Table in the Appendix.
• Through system programming, it is possible to select whether each type of TRAP (e.g., coldStart) is sent
to the SNMP manager or not.

Installation Manual References

7.1.4 Troubleshooting by Error Log

PC Programming Manual References

28.3.3 Network Service—[3-3] Client Feature—SNMP Agent

Feature Manual References

5.6.3 Local Alarm Information

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5.6.5 Dynamic Host Configuration Protocol (DHCP) Assignment

5.6.5 Dynamic Host Configuration Protocol (DHCP) Assignment

Description
It is possible to assign this PBX as a Dynamic Host Configuration Protocol (DHCP) client, allowing IP
addresses to be received from a DHCP server over a LAN.

Conditions
• It is possible to enable this feature through system programming.

Notice
It is important to set your DHCP server to not change the IP addresses of the mother board and DSP
cards once IP telephones are registered to the PBX. The IP telephones will not operate properly if these
IP addresses are changed.

PC Programming Manual References

28.1 Network Service—[1] IP Address/Ports—Basic Settings
→ LAN Setting—Obtain an IP address automatically/Use the following IP address
→ LAN Setting—IP Address
→ DSP IP Setting—Obtain DSP IP address automatically/Use the following DSP IP address

5.6.6 PING Confirmation

Description
It is possible for this PBX to confirm the connection of IP telephones, routers, and hubs within or outside the
private network using PING. The PBX will send an Internet Control Message Protocol (ICMP) echo request
through the PC programming terminal and receive an ICMP message confirming connection.

Conditions
• This PBX performs PING as follows:
– Test packet length: 56 bytes
– Ping attempts: 5
– Time out length: 1 second
– Ping interval time: 1 second

PC Programming Manual References

7.1.2 Utility—Diagnosis—Ping

5.6.7 System Monitoring

Description
The KX-NSX series, automatically monitors the operational status of the system, and provides a feature
which can acquire said operational status by various means. The methods to acquire operational statuses
are as follows.
• CSV File Output
Outputs the monitoring status of the target item as a CSV file. CSV files can be acquired through the
Web-MC operation.
• Web-MC Display
The monitoring status of the target items is displayed in the Web-MC.

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 5.6.8 Congestion Control

• Acquisition by MIB
The monitoring status of the target items can be acquired from the Management Information Bases
(MIBs).
(→ 5.6.4 Simple Network Management Protocol (SNMP) System Monitor)
• Alarm Notification
If the target item is in a congested state, it is recorded in the Syslog, alarm notification will be performed.
(→ 2.22.1.2 Syslog Record Management)

Below the possible acquisition of items per acquisition method is shown.

CSV File Web-MC Acquisition by Alarm

Item
Output Display MIB Notification
CPU Usage (%) ü ü ü
PBX Internal Memory Free Block ü ü
Usage (%)
Memory Usage (%) ü ü ü ü
SWAP memory empty capacity ü ü
(virtual memory usage) (KB)
Page In/Out Speed (Kbytes/s) ü ü
Disc Usage (%) ü ü ü ü
Internal Storage Lifespan ü ü ü
The amount of sent/received data ü ü
of each virtual NIC (Byte)
The amount of sent/received Loss ü ü
Packets of each virtual NIC
VoIP Call Quality Recording (IPPT ü ü
Terminal Call Quality Recording)

[Integrated Terminal Monitoring]

The operational status of terminals integrated to the system (IP telephone, Expansion Gateway) can also be
monitored with the KX-NSX series models. IP telephones, Expansion Gateways and DSP can be monitored.
The monitor method accumulates the results of the keep-alive every hour. The success rate is displayed
24/7 on the Web-MC. Furthermore, stored information can be obtained from the CSV files by use of the
Web-MC.

PC Programming Manual References

9.4 PBX Configuration—[1-1] Configuration—Slot—System Property—System Status
7.2.2 Utility—File—File Transfer PBX to PC
7.5.7 Utility—Report—Expansion Gateway Keep Alive Information

5.6.8 Congestion Control

Description
If a certain load or more is applied to the system, the system will take certain measures to continue to
provide service.
If the load for call control exceeds a certain value, the system will be considered congested. While the
system is congested, new incoming/outgoing calls will be restricted to prevent services from being
completely stopped.

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5.6.9 Hot Standby

If congestion occurs, it is recorded in the error log.

Conditions
• The Call Control feature and Web Maintenance Console manage the congestion state independently.
[Congestion states and operating restrictions]
• The 2 types of congestion and how operations are restricted in response to each are as follows:
System congestion
– Outgoing intercom calls, access to the Unified Messaging system, and the use of feature numbers are
restricted. (This restriction does not apply to current conversations.)
– For extensions that cannot make calls, guidance will be provided to notify users about the system
congestion.
→ 13.3.1 PBX Configuration—[5-3-1] Optional Device—Voice Message—DISA System
– Through COS programming, certain users can be allowed to make calls even while the system is
congested.
→ 10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Miscellaneous
– Access to Web Maintenance Console is restricted (User-level accounts only).
Severe system congestion
– Outgoing intercom calls, access to the Unified Messaging System, and the use of feature numbers are
prohibited. (This restriction does not apply to current conversations.)
– Extension users who try to make a call will hear a reorder tone or a busy tone.
– Even extension users who have permission to make calls during system congestion through COS
programming cannot make calls.
– All incoming calls from trunks (public lines/private lines [e.g., H.323 and Q-SIG]) are restricted.
– Access to Web Maintenance Console is prohibited (User-level accounts only).
• If the NSX server is switched through the Hot Standby feature, the congestion state is not transmitted to
the other NSX server.

[Recovery from a congestion state]

• As the system load decreases, the congestion state will be gradually released as follows:
Severe system congestion → System congestion → Normal
• The system will return to normal if it is restarted.

PC Programming Manual References

13.3.1 PBX Configuration—[5-3-1] Optional Device—Voice Message—DISA System
10.7.1 PBX Configuration—[2-7-1] System—Class of Service—COS Settings—Miscellaneous

5.6.9 Hot Standby

Keeps a secondary NSX server (Secondary server) running and synchronised with the main NSX sever
(Primary server). These NSX servers monitor each other's status, so if a fault occurs in the Primary server,
the Secondary server immediately takes over operation.
Because the servers are continuously synchronised, switching operation from one NSX server to the other
can be done quickly.

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 5.6.9 Hot Standby

Keep-alive Monitoring
• Keep-alive monitoring starts automatically after the NSX server starts up and its status is specified as
either Active or Standby.
The NSX servers monitor each other by sending and receiving packets.
If no transmissions are received from Primary server for a specified length of time, the Secondary server
switches its status.
The following timeout settings can be programmed:
Item Description
System The length of time after which one NSX server considers the other NSX server to be off-
Failure line. If the Primary server goes off-line, the Secondary server changes its status from
Detection Standby to Active.
Time
System The length of time to wait after restarting to establish a connection to the other NSX
Recovery server. If the NSX server receives packets from the other NSX server, the NSX server
Detection switches to standby mode. If no packets are received after this timer expires, the NSX
Time server switches to Active mode.
Alive Check The interval for sending keep-alive monitoring packets.
Interval

Conditions for Transferring the Synchronised data

• Transferring synchronisation data is always performed automatically.
• Synchronisation data is not compressed nor encrypted.
• If a system shutdown is performed while synchronisation data is being transferred, the system first
completes transferring the data safely and then performs the shutdown procedure.
• You can stop or restart the synchronisation of data through Web Maintenance Console. The
synchronisation settings are retained even after the system restarts. The operations of stopping and
restarting data synchronisation are recorded as information logs. For details about transferring
synchronisation data, refer to 5.6 System Control—Hot Standby—Data Transfer in PC Programming
Manual.
• By using the system settings, it is possible to set the system setting data to be saved and synchronised
automatically at a predetermined time (refer to "10.9 PBX Configuration—[2-9] System—System Options
—Option 10" in PC Programming Manual).
• The following table shows which data is synchronised and the timing of synchronisation:
Category Items Synchronisation Timing
System Setting • Network Service • An Administrator or the Installer logs out of
Data Web Maintenance Console.
• PBX Configuration
• Router Configuration • An Administrator or the Installer saves the
settings.
• Automatic Two-way Recording,
• Automatic data saving/synchronisation is
ACD Monitor
performed at a predetermined time
• UM Configuration according to the system settings.
Incoming/ • Incoming/Outgoing call logs • A system setting data is changed.
Outgoing call log

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5.6.9 Hot Standby

Category Items Synchronisation Timing

User Container • User Container • A user logs out of the User Portal.
• User's Device
• User's Flexible Buttons
• User's Personal Speed Dial
• User's CLIP ID Table
UM voice data • System Guidance (8 languages) • Custom guidance is recorded.
(System
• Custom Service Menu
Guidance)
• Company Greeting
• Other menus (Company Name,
Language Select Menu, Hold
Announce Menu, and other
guidance that users can record)
UM voice data • Owner Name • A user records custom guidance.
(Personal
• Personal Greetings
Greetings)
• Caller Information
UM voice data • Messages recorded in users' • A message is recorded to a mailbox.
(messages in mailboxes
users' mailboxes
• Messages recorded in the
and the
Automatic Two-way Recording
Automatic Two-
mailbox
way Recording
mailbox) • Messages recorded in the
Manager's mailbox
SMDR • Trunk call information (incoming/ • A trunk call finishes.
Information outgoing)
• Log-in/Log-out information • A user logs in or logs out.
• Data for Hotel Application • Timed reminder is executed.
• A room status is changed.
• Intercom call information • An extension call finishes.
(outgoing)
Billing • Call Billing • Billing information is changed.
Information
ACD Report • Call Log, Agent Log • Export-related data (call information and
Data agent information) is recorded.
• Report Profile • A report profile is created.
• Information for ACD Report • An ACD report is created.
Target file for file • LPR Programmes • A programme or language data is
transfer downloaded.
• IP-PT Programmes
• CS/PS Programmes
• Language data

446 Feature Manual

 5.6.9 Hot Standby

Category Items Synchronisation Timing

Audio files • MOH • An MOH file is downloaded.
• OGM • An OGM file is downloaded.
• An OGM file is recorded from an extension.

Providing Service while Switching the Status of NSX Servers

When the status of the NSX server is switched, services and device operation are temporarily stopped. After
the NSX server status has finished switching, services and operation resume.
The actions of services and devices while the NSX server's status is being changed is as follows:
Category Items Action
Services Call Control All services and features are temporary
unavailable. All trunks, extensions, and the
other devices are in an idle state. Current calls
will be disconnected. However, calls over a
peer-to-peer connection will continue. After
switching is complete, service resumes.
Web Maintenance service All Web Maintenance Console sessions are
forcefully terminated. Any users will need to
log in to Web Maintenance Console again
after switching is complete.
CTI service All CSTA sessions are forcefully terminated.
When the NSX server finishes switching its
status, any CTI servers automatically try to
reconnect to the NSX server. After a
connection is established, applications that
communicate with a CTI server are available.
Devices IP terminals connected to the NSX The devices stop their current operations.
server directly After the status switch is complete, each
device automatically reconnects to the NSX
IP-PTs connected to the Media
server.
Relay Gateway
IP-CSs connected to the NSX
server directly
IP trunks connected to the NSX Communication with SIP servers is
server directly terminated. After the status switch is
complete, the NSX server reconnects to the
Trunk adaptors
SIP servers. If the previous call information is
still retained by the other party, a busy tone
will be heard.
Expansion gateways Expansion gateways stop operation. When the
status switch is complete, expansion
gateways reconnect to the NSX server. After
establishing the connection, all devices under
the expansion gateway are available to use.

Feature Manual 447

5.6.9 Hot Standby

Conditions for Accessing Web Maintenance Console

When users access Web Maintenance Console, they automatically connect to the NSX server whose status
is Active. To connect to an NSX server whose status is Standby, you need to enter the IP address of that
NSX server.

Providing Service while Standby the Status of NSX Servers

The NSX server in Standby state will discontinue all the services, except for the following.

[Services provided in Standby state]

Type Function Remark
Server Features Web Maintenance Console Operations limited for specified menus
SSH
FTP
SNMP
Client Features SMTP E-mail Notification
SYSLOG
DHCP Acquisition of IP address
DNS
FTP Acquisition of software
NTP Time correction
Others Remote Maintenance

PC Programming Manual References

5.6 System Control—Hot Standby

Installation Manual References

6.7 Hot Standby

448 Feature Manual

Section 6
Appendix

Feature Manual 449

6.1 Capacity of System Resources

6.1 Capacity of System Resources

System
Item Capacity
Absent Message—Extension 1 × 16 characters
Absent Message—System 8 × 16 characters
Call Park Zone 100/System, 100 for each tenant
Conference 3-32 parties per conference
256 parties total per KX-NSX series PBX
COS 512
DID/DDI Table 32 digits, 7000 entries
Extension number 1 – 5 digits (PS:1-4 digits)
Extension Personal Identification Number 8 digits, 1 entry/extension
(PIN)
Host PBX Access Code 10 digits, 10 entries/trunk group
Multiple Subscriber Number (MSN) 10 entries per ISDN-BRI port
Number of Characters of Name 20
Printing message 8
Queuing Time Table 256
Ring Tone Pattern Plan 8
Simultaneous Programmers • one manager programmer + 127 personal programmers
(PT Programming)
• 128 personal programmers
Simultaneous Programmers • one Installer level account
(Web Maintenance Console)
• 1000/2000 User (User) level accounts
• 128 User (Administrator) level accounts
• 32 User (Sub-Administrator) level accounts
SMDR Call Storage 160000 calls
Special Carrier Access Code 16 digits, 100 entries
Tenant 128
Time Service Holiday 24
Verification Code 4 digits, 2000 entries
Verification Code Personal Identification 10 digits, 2000 entries
Number (PIN)
Charge Record for HOTEL 40000 calls

Dialling
Item Capacity
Emergency Call 32 digits, 20 entries

450 Feature Manual

 6.1 Capacity of System Resources

Item Capacity
Hot Line 32 digits
Key Pad Protocol Dial (ISDN Service 32 digits
Access)
Personal Speed Dialling 32 digits, 100 entries/extn.
Quick Dialling 8 digits, 4000/System or 1000/Tenant
Redial 32 digits
System Speed Dialling 64,000/System
*System wide: 5,000 entries
*Tenant wide: 59,000 entries
Max for 1 tenant: 5,000 entries
One-touch Dialling 32 digits,
10000 entries/system

Groups
Item Capacity
Conference Group 48 (31 members/group for Conference Group Mode,
31 members/group for Broadcast Mode)
User Group 512
Call Pickup Group 200
Idle Extension Hunting Group 256 (16 extensions/group)
Incoming Call Distribution Group 256 (128 extensions/group)
Paging Group 200
PS Ring Group 32
Trunk Group 128
P2P Group 512

TRS/Barring
Item Capacity
TRS/Barring Level 7
TRS/Barring Denied Code 16 digits, 600 entries/level
TRS/Barring Exception Code 16 digits, 600 entries/level

ARS
Item Capacity
Routing Plan Table 240 entries
Leading Number Table 16 digits, 5000 entries
Leading Number Exception Table 16 digits, 1000 entries
ARS Carrier 128

Feature Manual 451

6.1 Capacity of System Resources

Item Capacity
Itemised Billing Code 10 digits
Authorisation Code for Tenant 16 digits
Authorisation Code for Trunk Group 10 digits

Call Log and Message Waiting

Item Capacity
Outgoing Call Log 100 records/user
10,000 records/system
Incoming Call Log 100 records/user
20,000 records/system
Incoming Call Log—PS + Incoming Call
10240 records/system
Distribution Group
Message Waiting—PS + Incoming Call KX-NSX1000: 1000
Distribution Group KX-NSX2000: 2000
Message Waiting—PT + SLT KX-NSX1000: 1000
KX-NSX2000: 2000

Outgoing Message (OGM)

Item Capacity
Outgoing Message (OGM) 128
OGM Total Recording Time Approx. 64 minutes
OGM Port 128

Hospitality and Charge Management Features

Item Capacity
Hotel Operator 4
Charge Rate 7 digits including a decimal
Charge Denomination 3 currency characters/symbols

Networking
Item Capacity
Multi-Connection Networking 1 Main Unit
31 Expansion Gateway
TIE Line Routing and Modification Table 32 entries
Leading Number 3 digits
PBX Code 7 digits

452 Feature Manual

 6.1 Capacity of System Resources

Unified Messaging
Maximum capacity
ITEM
NSX1000 NSX2000 Note
Number of Users 1,000 2,000 1 user always owns 1
(Number of general mailboxes) mailbox
Total number of UM mail boxes 1,608 2,608 Maximum number of users +
number of ICDG + Tenant
management + automatic
2WAY recording
Tenant 128 128
Class of Service 512 512
UM Channel 128 128
Fax Channel 31 31 Assume 1 channel per site
IMAP Channel 72 72
UM recording time 1,600 hours*1 1,600 hours*1
Message length 1–60 min 1–60 min Unlimited can only be used
/Unlimited /Unlimited during 2Way-rec
Number of simultaneously usable 2Way- 85 85
rec
(Number of conference rooms)
Auto 2Way-rec manager 96 96
UM System Manager 1/Tenant 1/Tenant
UM Message Manager 1/Tenant 1/Tenant
UM Service Group 384/Tenant 384/Tenant
Custom Service 200/Tenant 200/Tenant
Mailbox Group 40 40
Mailbox Group 200 200
Member
Maximum Messages 9,000 9,000
(per mailbox)

*1 Recording time for each Storage Memory Card

Storage Memory Card capacity Call recording time

Storage Memory Card-S KX-NSX2135 8 GB 200 hours

Storage Memory Card-M KX-NSX2136 16 GB 400 hours

Storage Memory Card-L KX-NSX2137 32 GB 800 hours

Storage Memory Card-LL KX-NSX2138 64 GB 1600 hours

Message Resource
ITEM Capacity
System Guidance 1432

Feature Manual 453

6.1 Capacity of System Resources

ITEM Capacity
Custom Service Menu 200 items/Tenant
Company Greeting 32
Mailbox Group List Voice Label 20
System Caller ID Caller Name 1200

Greeting recording length

System Greeting Recording time (seconds)
Custom Service Menu 60
Company Name 10
Company Greeting 45
System Mailbox Group Voice Label 10
System Caller Name 10
Prompt Selection 30
Hold Announce Menu 30

Mailbox Greeting Recording time (seconds)

Mailbox Owner 10
Personal Greeting 45
Interview Message 10
Personal Group List 10
Personal Caller-ID 10
EMD List Member Name 10

Call Centre
Call Centre Supervisor 32
Auto 2way Rec Manager 96
Number of Call log for ACD report 1200000

Web Maintenance Console Accounts

Item Capacity
Users (User) KX-NSX1000: 1000 Accounts
KX-NSX2000: 2000 Accounts*1
User (Sub-Administrator) 32 Accounts
Users (Administrator) 128 Accounts
Installer 1 account
Password (all account types) 4 – 16 characters

*1 The total number of administrator and user accounts combined cannot exceed Users (User) maximum Accounts.

454 Feature Manual

 6.2 Tones/Ring Tones

6.2 Tones/Ring Tones

6.2.1 Tones/Ring Tones
Tone Patterns (Default)
1s

Confirmation Tone 1

Confirmation Tone 2

Confirmation Tone 3

Confirmation Tone 4

Confirmation Tone 5

Dial Tone 1

Dial Tone 2

Dial Tone 3

Dial Tone 4

Busy Tone

Reorder Tone

Ringback Tone 1

Ringback Tone 2

DND Tone

Trunk Call Limit Warning

Tone

Feature Manual 455

6.2.1 Tones/Ring Tones

Tone Patterns (Default)

15 s

Hold Alarm Tone

Call Waiting Tone 1

1s

Call Waiting Tone 2

OR

Ring Tone Patterns (Default)*

1s

Single

Double

Triple

S-Double

* The duration of a ring tone may vary by country/area.

456 Feature Manual

 6.3 Features that Require Activation Keys

6.3 Features that Require Activation Keys

Feature Required Activation Keys
Announces queuing status (number of calls in the waiting Activation Key for Call Centre Feature
queue/number of calls in the waiting queue and estimated Enhancement
waiting time) (KX-NSXF002 or KX-NSXF021)
(→ 2.2.2.3 Queuing Feature)
ACD Supervisory Feature Activation Key for Call Centre Feature
(→ 2.2.2.9 Supervisory Feature (ACD)) Enhancement
(KX-NSXF002 or KX-NSXF022)
Parallel Ringing When Forwarding to Trunk • Activation key for Normal user
(→ 2.3.2 Call Forwarding (FWD)) (KX-NSUN001, KX-NSUN010,
KX-NSUN050, KX-NSUN100,
KX-NSUN500)
• Activation key for Mobile user
(KX-NSUM001, KX-NSUM010,
KX-NSUM050, KX-NSUM100,
KX-NSUM500)
• Activation key for Advance user
(KX-NSUA001, KX-NSUA010,
KX-NSUA050, KX-NSUA100,
KX-NSUA500)
DISA Automatic Walking COS • Activation key for Normal user
(→ 2.16.1 Direct Inward System Access (DISA)) (KX-NSUN001, KX-NSUN010,
KX-NSUN050, KX-NSUN100,
KX-NSUN500)
• Activation key for Mobile user
(KX-NSUM001, KX-NSUM010,
KX-NSUM050, KX-NSUM100,
KX-NSUM500)
• Activation key for Advance user
(KX-NSUA001, KX-NSUA010,
KX-NSUA050, KX-NSUA100,
KX-NSUA500)
Computer Telephony Integration (CTI) Activation Key for CTI interface
(→ 2.26.1 Computer Telephony Integration (CTI)) (KX-NSXF004)
Built-in FOS interface Activation Key for CTI interface
(→ 2.23.4 Built-in FOS Interface) (KX-NSXF004)
Scheduled Backup for Unified Messaging Activation Key for Message Backup
(→ 3.1.2.4 System Backup/Restore) (KX-NSXU003)
Automatic Two-way Recording for Manager Activation Key for Two-way Recording
(→ 3.2.1.4 Automatic Two-way Recording for Manager) Control
(KX-NSXU004)

Feature Manual 457

6.3 Features that Require Activation Keys

Feature Required Activation Keys

Message Waiting Notification—E-mail Device Activation Key for Unified Messaging E-
(→ 3.2.1.29 Message Waiting Notification—E-mail Device) mail Notification
(KX-NSUN001, KX-NSUN010,
KX-NSUN050, KX-NSUN100,
KX-NSUN500 or KX-NSUM001,
KX-NSUM010, KX-NSUM050,
KX-NSUM100, KX-NSUM500 or
KX-NSUA001, KX-NSUA010,
KX-NSUA050, KX-NSUA100,
KX-NSUA500)
IMAP Integration Activation Key for Unified Messaging E-
(→ 3.3.1 IMAP Integration) mail Notification
(KX-NSUN001, KX-NSUN010,
KX-NSUN050, KX-NSUN100,
KX-NSUN500 or KX-NSUM001,
KX-NSUM010, KX-NSUM050,
KX-NSUM100, KX-NSUM500 or
KX-NSUA001, KX-NSUA010,
KX-NSUA050, KX-NSUA100,
KX-NSUA500)
System Speed Dial External Activation Key for LDAP
(→ 2.6.5 Speed Dialling—External (LDAP)) (KX-NSXF005)

When an Ultimate Activation Key for Panasonic Terminal Users (KX-NSXP101) is installed, some features
can be used without installing an activation key. For details, refer to "3.1.1 Type and Maximum Number of
Activation Keys" in the Installation Manual.

458 Feature Manual

 6.4 Supported Management Information Base (MIB) Table

6.4 Supported Management Information Base (MIB)

Table
System Group (1.3.6.1.2.1.1)
Object ID Item Description
1 sysDescr Information of Hardware type and Software version of the Device.
2 sysObjectID Object identifier of this product.
3 sysUpTime Elapsed time since the system was restarted.
4 sysContact Device Administrator.
5 sysName Name of Device.
6 sysLocation Installation Location of Device.
7 sysService Support Layer.

Interface Group (1.3.6.1.2.1.2)

Object ID Item Description
1 ifNumber The number of Network Devices.
2.1.1 ifIndex Each interface identifier.
2.1.2 ifDescr Explanation of Interface.
2.1.3 ifType Type of Interface.
2.1.4 ifMtu Maximum Datagram Length which can be sent/received.
2.1.5 ifSpeed Maximum Transfer Speed.
2.1.6 ifPhysAddress Physical Address (MAC Address).
2.1.7 ifAdminStatus The desired state of the interface.
2.1.8 ifOperStatus The current operational state of the interface.
2.1.9 ifLastChange The value of sysUpTime at the time the interface entered its
current operational state (up or down).
2.1.10 ifInOctets The number of Octets received.
2.1.11 ifInUcastPkts The number of Unicast Packets delivered to a higher-layer
protocol.
2.1.12 ifInNUcastPkts The number of Non Unicast Packets delivered to a higher-layer
protocol.
2.1.14 ifInErrors The number of inbound Packets that contained errors.
2.1.15 ifInUnKnownProtos The number of Packet received which are discarded because of
an unknown/unsupported protocol.
2.1.16 ifOutOctets The number of Octets transmitted.
2.1.17 ifOutUcastPkts The number of Unicast Packets which are received from upper
protocol.
2.1.18 ifOutNUcastPkts The number of Non Unicast Packets which are received from
upper protocol.

Feature Manual 459

6.4 Supported Management Information Base (MIB) Table

Object ID Item Description

2.1.21 ifOutQLen The length of the output packet queue (in packets).
2.1.22 ifSpecific Relevant MIB object identifier.

IP Group (1.3.6.1.2.1.4)
Object ID Item Description
1 ipForwarding The value which indicates operation availability as a router
(whether Datagram is transferred or not).
2 ipDefaultTTL Default value for IP Packet TTL (Time to Live).
3 ipInReceives The total number of Packets received (including packet received
in error).
4 ipInHdrErrors The number of Packets discarded due to errors in their header.
5 ipInAddrError The number of Packets discarded because IP Address of the
destination was invalid.
7 ipInUnknownProtos The number of Packets discarded because the protocol was
unknown/unsupported.
8 ipInDiscards The number of incoming Packets discarded because of an
insufficient reception buffer.
9 ipInDelivers The total number of Packets received (including ICMP) normally.
10 ipOutRequests The total number of IP Packets (ICMP) which are tried to be
transmitted (relay Packet is not included).
13 ipReasmTimeout The maximum number of seconds required in the buffer to rebuild
a fragmented Packet.
14 ipReasmReqds The number of Packets that required rebuilding from a
fragmented state.
15 ipReasmOKs The number of Packets that were rebuilt correctly from a
fragmented state.
16 ipReasmFails The number of Packets that could not be rebuilt correctly from a
fragmented state.
17 ipFragOKs The number of Packets that were fragmented correctly.
18 ipFragFails The number of Packets that could not be fragmented correctly.
19 ipFragCreates The number of IP datagrams created due to fragmentation.
20 ipAddrTable (NA) Management Table of addressing information relevant to this
entity’s IP addresses.
20.1 IpAddrEntry (NA) Components of ipAddrTable.
20.1.1 IpAdEntAddr IP Address.
20.1.2 IpAdEntIfindex Index value of the Interface which is assigned to IP address.
20.1.3 IpAdEntNetMask The Subnet Mask associated with IP address.
20.1.4 ipAdEntBcastAddr Broadcast Address Value associated with IP Address.
21 ipRouteTable Management table of IP routing information

460 Feature Manual

 6.4 Supported Management Information Base (MIB) Table

Object ID Item Description

21.1 ipRouteEntry ipRouteTable component
21.1.1 ipRouteDest IP address of the routing information destination. The value is
aaa.bbb.ccc.ddd.
21.1.2 ipRouteIfIndex Index number of the interface related to routing information.
The same value as the ifIndex at the bottom of the interfaces
subtree
21.1.3 ipRouteMetric1 Metric value of the routing information (priority ranking)
21.1.7 ipRouteNextHop IP address that becomes the next arrival destination of the
routing information (IP address of the gateway)
21.1.8 ipRouteType Routing type of the routing information
21.1.9 ipRouteProto Routing protocol
21.1.11 ipRouteMask When the subnet mask value default gateway of the routing
information is 0.0.0.0
21.1.13 ipRouteInfo The OID of the object that indicates the additional information of
the routing information. 0.0 when there is no additional
information
22 ipNetToMediaTable Management table of the conversion table for IP address –
physical address
22.1 ipNetToMediaEntry Component of the ipNetToMediaTable
22.1.1.x ipNetToMediaIfIndex The index number of the interface related to the conversion table
22.1.2.x ipNetToMediaPhysAdd Physical address associated with the IP address (MAC address
ress etc.)
22.1.3.x ipNetToMediaNetAddr Actual IP address. The same as aaa.bbb.ccc.ddd
ess
22.1.4.x ipNetToMediaType Support settings method for the IP address and the physical
address
23 ipRoutingDiscards It was valid when using a dynamic routing protocol the total
number of routing tables which have been discarded

ICMP Group (1.3.6.1.2.1.5)

Object ID Item Description
1 cmpInMsgs The total number of ICMP messages received (excluded, with
error).
2 icmpInErrors The total number of ICMP messages received which contained
error.
3 icmpInDestUnreachs The total number of received ICMP unreachable messages
4 icmpInTimeExcds The total number of received ICMP overtime messages
5 icmpInParmProbs The total number of received ICMP parameter error messages
6 icmpInSrcQuenchs The total number of received ICMP source loss messages
7 icmpInRedirects The total number of received ICMP redirection messages

Feature Manual 461

6.4 Supported Management Information Base (MIB) Table

Object ID Item Description

8 icmpInEchos The total number of ICMP echo request messages received.
9 icmpInEchoReps The total number of ICMP echo answering messages received.
10 icmpInTimestamps The total number of received ICMP Time stamp request
messages
11 icmpInTimestampReps The total number of received ICMP timestamp reply messages
12 icmpInAddrMasks The total number of received ICMP address mask request
messages
13 icmpInAddrMaskReps The total number of received ICMP address mask reply
messages
14 icmpOutMsgs The number of ICMP messages which were sent.
15 icmpOutErrors The number of ICMP messages which were not sent because of
error.
16 icmpOutDestUnreachs The number of sent ICMP unreachable destination messages
17 icmpOutTimeExcds The number of sent ICMP overtime messages
18 icmpOutParmProbs The number of sent ICMP parameter error messages
19 icmpOutSrcQuenchs The number of sent ICMP source loss messages
20 icmpOutRedirects The number of sent ICMP redirection messages
21 icmpOutEchos The number of ICMP Echo request messages sent.
22 icmpOutEchoReps The number of ICMP Echo Reply messages sent.
23 icmpOutTimestamps The number of sent ICMP Time stamp request messages
24 icmpOutTimestampRe The number of sent ICMP timestamp reply messages
ps
25 icmpOutAddrMasks The number of sent ICMP address mask request messages
26 icmpOutAddrMaskRep The number of sent ICMP address mask reply messages
s

TCP Group (1.3.6.1.2.1.6)

Object ID Item Description
1 tcpRtoAlgorithm The algorithm used to determine the timing of retransmitting
when a response was unacknowledged.
2 tcpRtoMin Minimum value permitted for retransmission timeout (in
milliseconds).
3 tcpRtoMax Maximum value permitted for retransmission timeout (in
milliseconds).
4 tcpMaxConn Maximum number of TCP connections which can be supported.
5 tcpActiveOpens The total number of Active open TCP connections.
6 tcpPassiveOpens The total number of Passive open TCP connections.
7 tcpAttemptFails The total number of connections error.
8 tcpEstabResets The total number of resets.

462 Feature Manual

 6.4 Supported Management Information Base (MIB) Table

Object ID Item Description

9 tcpCurrEstab The number of connections when the current status is
ESTABLISHED (connecting) or CLOSE-WAIT (waiting to
disconnect)
10 tcpInSegs The total number of segments received.
11 tcpOutSegs The total number of segments sent.
12 tcpRetransSegs The total number of segments retransmitted.
13.1 tcpConnEntry Component of the tcpConnTable
13.1.1 tcpConnState Connection status
13.1.2 tcpConnLocalAddress Local IP address The same as aaa.bbb.ccc.ddd
13.1.3 tcpConnLocalPort Local port number The same as ppp
13.1.4 tcpConnRemAddress Remote IP address The same as AAA.BBB.CCC.DDD
13.1.5 tcpConnRemPort Remote port number The same as PPP
14 tcpInErrs The total number of segments received in error.
15 tcpOutRsts The total number of TCP segments sent containing the RST flag
(reset connection).

UDP Group (1.3.6.1.2.7)

Object ID Item Description
1 udpInDatagrams The total number of UDP Datagrams received.
2 udpNoPorts The total number of received UDP Datagrams for which there
was no application at the destination port.
3 udpInErrors The total number of received UDP Datagrams which contained
error.
4 udpOutDatagrams The total number of UDP Datagrams sent.
5 udpTable Standby information management table for UDP datagram
5.1 udpEntry Component of the udpTable
5.1.1 udpLocalAddress udpLocalAddress IP address of the local connection waiting on
the UDP datagram
The same as aaa.bbb.ccc.ddd. All IP addresses are 0.0.0.0 while
in Standby Mode.
5.1.2 udpLocalPort Local port number of the connection waiting on the UDP
datagram The same as ppp

SNMP Group (1.3.6.1.2.1.11)

Object ID Item Description
1 snmpInPkts The total number of SNMP messages received.
2 snmpOutPkts The total number of SNMP messages sent.
3 snmpInBadVersions The total number of received SNMP messages of which version
is unsupported.

Feature Manual 463

6.4 Supported Management Information Base (MIB) Table

Object ID Item Description

4 snmpInBadCommunity The total number of SNMP messages with unknown Community
Names Name.
5 snmpInBadCommunity The number of SNMP messages trying to operate without having
Uses authority
6 snmpInASNParseErrs The total number of SNMP messages with incorrect OID type.
8 snmpInTooBigs The total number of received messages with a "tooBig" error
9 snmpInNoSuchNames The total number of received messages with a "noSuchName"
error
10 snmpInBadValues The total number of received messages with a "badValue" error
11 snmpInReadOnlys The total number of received messages with a "readOnly" error
12 snmpInGenErrs The total number of received messages with a "genErr" error
13 snmpInTotalReqVars The total number of Objects which have been retrieved value
successfully.
15 snmpInGetRequests The total number of Get-Request which have been accepted and
processed (e.g. Data extract by using snmpget/snmpwalk
command).
16 snmpInGetNexts The total number of Get-Next which have been accepted and
processed (for at the time of following a layer-tree by using
snmpwalk command).
17 snmpInSetRequests Total number of processed Set-Requests (such as the
configuration of the data with the snmpset command)
18 snmpInGetResponses Total number of processed Get-Responses
19 snmpInTraps Total number of processed Traps
20 snmpOutTooBigs The total number of sent SNMP messages which returned an
error of "TooBig".
21 snmpOutNoSuchNam The total number of sent SNMP messages which returned an
es error of "NoSuchName".
22 snmpOutBadValues "badValue" Total number of sent SNMP messages where an
error occured
24 snmpOutGenErrs The total number of sent SNMP messages which returned an
error of "GenErr".
25 snmpOutGetRequests Total number of sent Get-Requests
26 snmpOutGetNexts Total number of sent Get-Nexts
27 snmpOutSetRequests Total number of sent Set-Requests
28 snmpOutGetResponse The total number of GetResponse sent.
s
29 snmpOutTraps The total number of TRAP sent.
30 snmpEnableAuthenTra Indicates whether the SNMP agent process is permitted to
ps generate authentication failure traps.

464 Feature Manual

 6.4 Supported Management Information Base (MIB) Table

Related to System Monitoring

1. Cumulative Total Value of CPU Operating Time

1.3.6.1.4.1.2021.11
Object ID Item Description
50 ssCpuRawUser User mode CPU time
52 ssCpuRawSystem System CPU time
53 ssCpuRawIdle Idle CPU time
54 ssCpuRawWait Wait CPU time
64 ssCpuRawStael I/O Wait CPU time

2. PBX Internal Memory Free Block Utilisation Rate (average utilisation rate% of N seconds)

1.3.6.1.4.1.258.601.4.2.1
Object ID Item Description
1 panaPbxMemFbkSmall The average value of N seconds of the utilisation
rate (Small) (%)
2 panaPbxMemFbkMiddle The average value of N seconds of the utilisation
rate (Middle) (%)
3 panaPbxMemFbkLarge The average value of N seconds of the utilisation
rate (Large) (%)
4 panaPbxMemFbkLLSmall The average value of N seconds of the utilisation
rate (LL-Small) (%)
5 panaPbxMemFbkLLMiddle The average value of N seconds of the utilisation
rate (LL-Middle) (%)

3. Memory Usage (used/buffer/cached)

1.3.6.1.4.1.2021.4
Object ID Item Description
5 memTotalReal Total capacity of the actual CPU memory (kB)
6 memAvailReal Available memory capacity of the actual CPU memory (kB)
14 memBuffer Amount of buffer memory (kB)
15 memCached Amount of cache memory (kB)

4. Available SWAP Memory Capacity (virtual memory usage) (KB)

1.3.6.1.4.1.2021.4
Object ID Item Description
3 memTotalSwap Total Swap capacity (kB)
4 memTotalSwap Available Swap capacity (kB)

5. Page In/Out Speed (KB/s)

Feature Manual 465

6.4 Supported Management Information Base (MIB) Table

1.3.6.1.4.1.2021.11
Object ID Item Description
3 ssSwapIn Average Swap in amount of 1 minute (kB)
4 ssSwapOut Average Swap out amount of 1 minute (kB)

6. RW Completion Waiting Time

1.3.6.1.4.1.2021.9.1.
Object ID Item Description
1.i dskIndex Index
2.i dskPath Path name
3.i dskDevice Device name
9.i dskPercent Utilisation rate (%) (e.g.) when 30%, 30

7. Transmission and reception amount (in bytes) per virtual NIC, transmission and reception loss packet
amount per Virtual NIC

1.3.6.1.2.1.2.2.1
Object ID Item Description
1.i ifIndex Index
2.i ifDescr Description "eth0" etc.
10.i ifInOctets Total number of received bytes
16.i ifOutOctets Total number of sent bytes
13.i ifInDiscards Number of received_discarded packets
14.i ifInErrors Number of received_error packets
15.i ifInUnknownProtos Number of received_unknown protocol packets
19.i ifOutDiscards Number of sent_discarded packets
20.i ifOutError Number of sent_error packets

8. Internal Storage Service Life (Panasonic Original)

1.3.6.1.4.1.258.601.4
Object ID Item Description
4.1 panaPbxDsklife1 Storage service life indicator of storage 1 of the slave
The index value is indicated by the numbers 0-100.
100-1: Normal, 0: Abnormal (the higher the number the
longer the remaining lifespan)
255: unimplemented
4.2 panaPbxDsklife2 Storage service life indicator of storage 1 of the UM
The meaning of the values is the same as with
panaPbxDsklife1.

9. Alive Monitoring of the UDP Port

466 Feature Manual

 6.5 Revision History

1.3.6.1.2.1.7
Object ID Item Description
7.1 udpEndpointProcess UDP port monitoring by the monitoring server (Zabbix) as
described in network maintenance

6.5 Revision History

6.5.1 PNMPR Software File Version 002.00xxx
Changed Contents
• Introduction
• Functional Limitation
• 2.1.1.1 Incoming Trunk Call Features—SUMMARY
• 2.1.1.6 Intercept Routing
• 2.2.2.2 Group Call Distribution
• 2.2.2.9 Supervisory Feature (ACD)
• 2.5.5.3 Trunk Access
• 2.6.1 Memory Dialling Features—SUMMARY
• 2.6.5 Speed Dialling—External (LDAP)
• 2.9.1 Primary Directory Number (PDN)/Secondary Directory Number (SDN) Extension
• 2.11.4 Headset Operation
• 2.14.1 Conference Features—SUMMARY
• 2.16.1 Direct Inward System Access (DISA)
• 2.19.2 Incoming Call Log
• 2.20.1 Message Waiting
• 2.22.1.1 Station Message Detail Recording (SMDR)
• 2.23.3 Call Billing for Guest Room
• 2.24.4 Smart Desk Features
• 2.24.4.1 Service-in Feature
• 2.24.4.2 Service-out Feature
• 2.29.1 User Container
• 2.29.1.1 User Container Configuration
• 2.29.1.2 Individual User Information Management of the Construction and Maintenance Personnel
• 2.29.1.6 Linking with LDAP User Information
• 2.29.2.2 Sub Device
• 2.29.2.5 Public (Mobile Phone) Device
• 2.29.2.6 UM (Mailbox)
• 3.1.1 Unified Messaging System Overview
• 3.1.3 Fax Server

Feature Manual 467

6.5.2 PNMPR Software File Version 002.01xxx

• 3.2.1.40 Service Group

• 4.2.1 Multi-connection Networking Overview
• 4.2.3 Multi-connection Networking Survivability
• 4.2.3.1 Simplified Isolated Mode
• 5.1.1 Class of Service (COS)
• 5.2.3 Peer-to-Peer (P2P) Connection
• 5.2.7 Internal Call Encryption
• 5.3.1 Stacking Connection
• 5.5.1 PC Programming
• 5.5.3.1 DSP Resource Reservation
• 5.5.3.2 DSP Resource Advisor
• 5.5.3.3 Resource Depletion Indication Monitoring
• 5.5.8 Software Upgrading
• 5.6.8 Congestion Control
• 5.6.9 Hot Standby
• 6.1 Capacity of System Resources
• 6.3 Features that Require Activation Keys
• 6.4 Supported Management Information Base (MIB) Table

6.5.2 PNMPR Software File Version 002.01xxx

Changed Contents
• 5.2.7 Internal Call Encryption

6.5.3 PNMPR Software File Version 002.03xxx

Changed Contents
• Introduction-Functional Limitation
• 2.1.1 Incoming Trunk Call Features
• 2.1.1.6 Intercept Routing
• 2.1.3.2 Ring Tone Pattern Selection
• 2.4.4 Hands-free Answerback
• 2.6.5 Speed Dialling—External (LDAP)
• 2.11.1 Hands-free Operation
• 2.11.5 Data Line Security
• 2.17.1 Paging
• 2.19.1 Caller ID
• 2.29.1.6 Linking with LDAP User Information
• 3.2.1.4 Automatic Two-way Recording for Manager
• 5.2.7 Internal Call Encryption

468 Feature Manual

 6.5.4 PNMPR Software File Version 002.10xxx

• 5.5.3 DSP Resource Usage

• 5.6.9 Hot Standby

6.5.4 PNMPR Software File Version 002.10xxx

Changed Contents
• 2.6.5 Speed Dialling—External (LDAP)
• 2.15.1 Conference Group Call
• 2.18.5 Communication IP Camera/Video Door Phone
• 2.24.4 Smart Desk Features
• 2.24.4.3 Automatic Service-out Feature
• 4.2.1 Multi-connection Networking Overview
• 5.1.3 Tenant Service
• 5.1.4 Time Service
• 6.1 Capacity of System Resources

6.5.5 PNMPR Software File Version 003.00xxx

New Contents
• 2.23.4 Built-in FOS Interface
• 2.26.2 CA (Communication Assistant) Operator Console

Changed Contents
• Introduction
– About this Feature Manual
– Functional Limitation
• List of Abbreviations
• 2.2.2.9 Supervisory Feature (ACD)
• 2.13.2 Call Park
• 2.16.1 Direct Inward System Access (DISA)
• 2.17.1 Paging
• 2.18.5 Communication IP Camera/Video Door Phone
• 2.20.1 Message Waiting
• 2.21.4 Display Information
• 2.22.1.1 Station Message Detail Recording (SMDR)
• 2.22.1.2 Syslog Record Management
• 2.24.1 Extension Personal Identification Number (PIN)
• 2.24.3 Timed Reminder
• 2.26.1 Computer Telephony Integration (CTI)
• 2.27.1 Cellular Phone Features—SUMMARY

Feature Manual 469

6.5.5 PNMPR Software File Version 003.00xxx

• 2.28.2 Outgoing Message (OGM)

• 2.29.1.1 User Container Configuration
• 2.29.1.2 Individual User Information Management of the Construction and Maintenance Personnel
• 2.29.2.1 Main Device/Device Common
• 2.3.2 Call Forwarding (FWD)
• 2.4.4 Hands-free Answerback
• 2.6.2 One-touch Dialling
• 2.7.6 Verification Code Entry
• 2.8.1 Automatic Route Selection (ARS)
• 3.1.1 Unified Messaging System Overview
• 3.2.1.11 Class of Service (COS)
• 3.2.1.12 Company Greeting
• 3.2.1.15 Custom Service
• 3.2.1.21 Holiday Service
• 3.2.1.25 Interview Service
• 3.2.1.26 List All Names
• 3.2.1.3 Automated Attendant (AA)
• 3.2.1.4 Automatic Two-way Recording for Manager
• 3.2.1.46 Voice Mail Service
• 3.3.1 Integration with Microsoft Outlook
• 4.2.1 Multi-connection Networking Overview
• 4.3.1.2 TIE Line and Trunk Connection
• 5.1.2 Group
• 5.1.3 Tenant Service
• 5.1.4 Time Service
• 5.2.2 SIP (Session Initiation Protocol) Extension
• 5.2.2.1 Simple Remote Connection
• 5.2.2.2 Panasonic SIP Phones (P-SIP)
• 5.2.3 Peer-to-Peer (P2P) Connection
• 5.2.7 Internal Call Encryption
• 5.3.1 Stacking Connection
• 5.6.9 Hot Standby
• 6.1 Capacity of System Resources
• 6.3 Features that Require Activation Keys
• 6.4 Supported Management Information Base (MIB) Table

470 Feature Manual

 6.5.6 PNMPR Software File Version 004.00xxx

6.5.6 PNMPR Software File Version 004.00xxx

Changed Contents
• Introduction
• 2.10.1 Automatic Callback Busy (Camp-on)
• 2.18.3 External Sensor
• 2.18.5 Communication IP Camera/Video Door Phone
• 2.20.1 Message Waiting
• 2.21.2 Flexible Buttons
• 2.24.3 Timed Reminder
• 2.29.2.4 Wireless Device
• 3.2.1.45 Trunk Service (Universal Port)
• 3.2.2.10 Caller ID Callback
• 4.2.3.1 Simplified Isolated Mode
• 5.2.1 IP Proprietary Telephone (IP-PT)
• 5.2.2.2 Panasonic SIP Phones (P-SIP)
• 5.2.3 Peer-to-Peer (P2P) Connection
• 6.1 Capacity of System Resources

Feature Manual 471

Panasonic Corporation
1006, Oaza Kadoma, Kadoma-shi, Osaka 571-8501, Japan

http://www.panasonic.com/

© Panasonic Corporation 2015

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