SV9100 Networking Manual GE 5 0

Networking Manual
A50-035910-001 GE
Issue 5.0
Contents of this manual are subject to change without prior notice at the discretion of NEC
Corporation. This document has been prepared for the use of employees and customers of
NEC Corporation and may not be reproduced without prior written approval of NEC
Corporation.
UNIVERGE is a registered trademark of NEC Corporation. Windows is a registered trademark

of Microsoft Corporation. Ekahau Site Survey is a trademark of Ekahau, Inc. AIRPCAP is a
trademarks or registered trademarks of Riverbed
Copyright 2014 - 2016
NEC Corporation
TABLE OF CONTENTS
Introduction
Section 1 General Overview .................................................................................... 1-1
Section 2 Manual Organization ............................................................................... 1-1
Section 3 Common Terms ....................................................................................... 1-1
Book 1 – SV9100 K-CCIS

Chapter 1 General Information <US Only>
Section 1 System Outline ........................................................................................ 1-1
1.1 Common Channel Handler (GCD-CCTA) ............................................ 1-2
Section 2 DTI Specifications ................................................................................... 1-3
2.1 Characteristics .................................................................................... 1-3
Section 3 Network Structure Considerations ........................................................ 1-7
3.1 Determining System Configurations .................................................... 1-7
Chapter 2 Hardware Installation <US Only>

Section 1 Installation Precautions .......................................................................... 2-1
1.1 Busying Out Extension/Line Blades ..................................................... 2-2
Section 2 Installing an Extension or Trunk Blade ................................................. 2-2
2.1 Installing the Blades ............................................................................. 2-2
2.2 Order of Installing Extension Blades .................................................... 2-3
2.3 Order of Installing Trunk Blades .......................................................... 2-4
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2.3.1 Installing COIU-LS1/LG1, GCD-4ODTA, GCD-4DIOPA, or GCD-2BRIA

Blades .............................................................................................. 2-4
Section 3 Installing the GCD-CCTA (CCIS Trunk Interface) ................................. 2-5
3.1 Description ........................................................................................... 2-5
3.2 Installation ........................................................................................... 2-6
3.3 LED Indications ................................................................................... 2-6
3.4 Connectors .......................................................................................... 2-7
Chapter 3 System Data Programming <US Only>

Section 1 K-CCIS Programming ............................................................................. 3-1
1.1 Digital Trunk Data Assignment ............................................................ 3-1
1.2 CCIS Assignment ................................................................................ 3-3
1.3 Numbering Plan Assignment ............................................................... 3-4

1.4 Programming for Closed Numbering Plan ........................................... 3-4
1.5 Programming for Open Numbering Plan ............................................. 3-5
1.6 Closed Number Programming Example .............................................. 3-7
1.7 Open Number Programming Example .............................................. 3-11
1.8 Dedicated Tandem CO Trunk Calls ................................................... 3-15
1.9 Shared Tandem CO Trunk Calls ....................................................... 3-17
Chapter 4 Features and Specifications

Section 1 General Information ................................................................................ 4-1
Section 2 System Availability ................................................................................. 4-2
2.1 Terminal Type ...................................................................................... 4-2
2.2 Required Components ......................................................................... 4-2
2.3 Operating Procedures ......................................................................... 4-2
Section 3 Service Conditions ................................................................................. 4-3
Section 4 Related Feature List ................................................................................ 4-5
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Section 5 K-CCIS Features ...................................................................................... 4-5

Automatic Recall – K-CCIS ............................................................................ 4-6
Brokerage Hotline – K-CCIS ........................................................................... 4-8
Call Forwarding – All Calls – K-CCIS ........................................................... 4-10
Call Forwarding – Busy/No Answer – K-CCIS .............................................. 4-14
K-CCIS Call Rerouting .................................................................................. 4-18
Call Park Retrieve – K-CCIS ........................................................................ 4-28
Call Transfer – All Calls – K-CCIS ................................................................ 4-34
Calling Name Display – K-CCIS ................................................................... 4-37
Calling Number Display – K-CCIS ................................................................ 4-39
Calling Party Number (CPN) Presentation from Station – K-CCIS ............... 4-41
Centralized Billing – K-CCIS ......................................................................... 4-43
Centralized BLF (K-CCIS) ............................................................................ 4-47
Centralized Day/Night Mode Change – K-CCIS ........................................... 4-52
Centralized E911 – K-CCIS .......................................................................... 4-57
Dial Access to Attendant – K-CCIS .............................................................. 4-60
Direct Inward Dialing – K-CCIS .................................................................... 4-65
Dual Hold – K-CCIS ...................................................................................... 4-67
Elapsed Time Display – K-CCIS ................................................................... 4-69
Flexible Numbering of Stations – K-CCIS .................................................... 4-71
Handsfree Answerback – K-CCIS ................................................................ 4-74
Hot Line – K-CCIS ........................................................................................ 4-76
Link Reconnect – K-CCIS ............................................................................. 4-78
Multiple Call Forwarding – All Calls – K-CCIS .............................................. 4-82
Multiple Call Forwarding – Busy/No Answer - K-CCIS ................................. 4-87
Paging Access – K-CCIS .............................................................................. 4-92
Quick Transfer to Voice Mail – K-CCIS ...................................................... 4-100
Station-to-Station Calling – K-CCIS ............................................................ 4-103
Uniform Numbering Plan – K-CCIS ............................................................ 4-105
Voice Call – K -CCIS .................................................................................. 4-107
Voice Mail Integration – K-CCIS ................................................................. 4-109
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Chapter 5 SV9100 IP K-CCIS

Section 1 System Outline ........................................................................................ 5-1
1.1 IP K-CCIS Application using the GPZ-IPLE ........................................ 5-1
1.1.1 CCIS Networking via IP (Non Peer-to-Peer Connections Basis) ..... 5-1
1.2 Description ........................................................................................... 5-2
1.3 Systems Requirements ....................................................................... 5-3
1.3.1 Hot Swap ......................................................................................... 5-4
1.3.2 Connectors ...................................................................................... 5-4
1.3.3 Ethernet Status (IPLE) ..................................................................... 5-4
1.3.4 Environmental Conditions ................................................................ 5-4
1.3.5 Installation Procedure ...................................................................... 5-4
1.3.6 Switch Settings ................................................................................ 5-4
1.3.7 LED Indications ............................................................................... 5-5
1.3.8 Connectors ...................................................................................... 5-8
1.4 IP Addressing ..................................................................................... 5-9
1.4.1 General IP Configuration ............................................................... 5-10
1.4.2 ToS Settings (Layer 3 QoS) .......................................................... 5-10
1.4.3 Bandwidth ...................................................................................... 5-14
1.4.4 Bandwidth Calculations ................................................................. 5-15
1.4.5 Payload Calculation Voice ............................................................ 5-15
1.4.6 Some Network Considerations ...................................................... 5-17
Section 2 UNIVERGE SV9100 Chassis Programming ........................................ 5-20
2.1 Digital Trunk Assignment ................................................................... 5-20
2.2 VoIP IP Address Assignments ........................................................... 5-23
2.3 Local Numbering Plan Assignment ................................................... 5-24
2.4 Closed Numbering Plan – Using Closed Number Blocks .................. 5-24
2.5 Open Numbering Plan – Using ARS Table 1, 2, or 3 ........................ 5-25
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2.6 IP K-CCIS Assignment ...................................................................... 5-27
2.7 IP K-CCIS (Peer-to-Peer) Assignment .............................................. 5-27
2.8 CCIS over IP CODEC Information Setup .......................................... 5-28
2.9 SIP-MLT CODEC Information Fixed Mode Setup ............................. 5-31
2.10 MTU Assignment .............................................................................. 5-32
Section 3 Programming Example ......................................................................... 5-33
3.1 SV9100 IP K-CCIS Programming Example 1 .................................... 5-33

3.1.1 Card Interface Slot Assignment (PRG 10-03) ................................ 5-34
3.1.2 Digital Trunk Assignments ............................................................. 5-34
3.1.3 VoIP Address Assignments ........................................................... 5-35
3.1.4 IP K-CCIS Availability .................................................................... 5-35
3.1.5 IP K-CCIS Assignment ................................................................... 5-35
3.1.6 IP K-CCIS (Peer-to-Peer) Assignment ........................................... 5-36
3.1.7 Local Numbering Plan Assignment ................................................ 5-36
3.1.8 Closed Numbering Plan – Using Closed Number Blocks .............. 5-37
3.1.9 Tandem Connections ..................................................................... 5-38
3.2 UNIVERGE SV9100 IP K-CCIS and Electra Elite IPK II Programming Ex-
ample 2 .............................................................................................. 5-39
3.2.1 Chassis Programming Assignments .............................................. 5-39
3.2.2 Digital Trunk Assignments ............................................................. 5-41
3.2.3 CCH Assignment ........................................................................... 5-42
Section 4 Port Designations ................................................................................. 5-43
Section 5 CCIS Networking via IP (Peer-to-Peer Connections Basis) .............. 5-44
Section 6 Service Conditions ................................................................................ 5-45
6.1 Restrictions ........................................................................................ 5-46
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Book 2 – SV9100 IP Networking

Chapter 1 General Information
Section 1 UNIVERGE SV9100 IP Networking ........................................................ 1-1
Section 2 Voice Over IP .......................................................................................... 1-1
Chapter 2 IP Networking
Section 1 Introduction ............................................................................................. 2-1
Section 2 UNIVERGE SV9100 IP K-CCIS Network SV9100 ................................... 2-1
Section 3 IP Trunks .................................................................................................. 2-2
3.1 Configure IP Trunks ............................................................................ 2-2
Section 4 Example Configurations ........................................................................ 2-3
4.1 Network Configurations ....................................................................... 2-3
4.2 UNIVERGE SV9100 IP K-CCIS Programming Example 1 .................. 2-4

4.2.1 Card Interface Slot Assignment (PRG 10-03 ETU Setup) ............... 2-5
4.2.2 Digital Trunk Assignments ............................................................... 2-5
4.2.3 VoIP Address Assignments ............................................................. 2-6
4.2.4 CCIS Availability .............................................................................. 2-6
4.2.5 IP CCISoIP Assignment .................................................................. 2-6
4.2.6 CCIS Assignment ............................................................................ 2-7
4.2.7 Centralized Day Night Switching Assignments ................................ 2-8
4.2.8 Centralized BLF Assignments ......................................................... 2-9
4.2.9 Local Numbering Plan Assignment ............................................... 2-10
4.2.10 Closed Numbering Plan - using Closed Number Blocks ............... 2-10
4.2.11 Tandem Connections .................................................................... 2-11
Section 5 DTMF Relay ........................................................................................... 2-11
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Chapter 3 General IP Configuration

Section 2 Network Addressing Overview .............................................................. 3-1
2.1 IP Address ........................................................................................... 3-1
2.2 Subnet Mask ........................................................................................ 3-2
2.3 DHCP ................................................................................................... 3-2
Section 3 Configuration Examples ......................................................................... 3-3
3.1 Example Configuration 1 - Existing Network with Static Addressing ... 3-3
3.2 Example Configuration 2 - New Network with Dynamic Addressing ... 3-5
Section 4 Testing the UNIVERGE SV9100 Network Connection ......................... 3-8
Chapter 4 Programming
Section 1 Before You Start Programming ............................................................ 4-1
Section 2 How to Enter Programming Mode ......................................................... 4-2
Section 3 How to Exit Programming Mode ............................................................ 4-3
Section 4 Using Keys to Move Around in the Programs ...................................... 4-3
Section 5 Programming Names and Text Messages ............................................ 4-4
Section 6 Using Softkeys For Programming ......................................................... 4-5
Section 7 What the Softkey Display Prompts Mean ............................................. 4-6
Section 8 Programs ................................................................................................. 4-6
Chapter 5 Network Design Considerations

Section 1 Introduction ............................................................................................ 5-1
Section 2 QoS ........................................................................................................... 5-1
2.1 QoS Definitions .................................................................................... 5-1
2.2 Voice Quality Improvements ................................................................ 5-3
2.3 Types of Classifications for Traffic for QoS ......................................... 5-4
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Section 3 Internet Based Connections (xDSL, Cable, etc.) ................................. 5-5
Section 4 Firewalls and NAT .................................................................................. 5-6
4.1 Understanding the Infrastructure ......................................................... 5-6
4.2 Firewall Integration .............................................................................. 5-7
4.3 Virtual Private Network (VPN) Tunnelling ............................................ 5-8
Section 5 CODEC and Bandwidth ....................................................................... 5-10
5.1 CODECs ............................................................................................ 5-10
5.2 Bandwidth .......................................................................................... 5-12
Section 6 DSP Resource Calculation .................................................................. 5-13
Section 7 Quality of Service (QoS) Implementation ........................................... 5-14
7.1 Prioritization ....................................................................................... 5-14
7.2 Layer 2 QoS (802.1pq) ...................................................................... 5-17
7.3 Layer 3 QoS ...................................................................................... 5-21
7.4 IP Precedence ................................................................................... 5-23
7.5 Diffserv (Differentiated Service) ......................................................... 5-24
7.6 Comparison of IP Precedence and Diffserv Values .......................... 5-25
7.7 Programming QoS in the UNIVERGE SV9100 System .................... 5-28
7.7.1 Marking Voice Traffic - Program 84-10-XX .................................... 5-28
7.7.2 UNIVERGE SV9100 Voice Protocols ............................................ 5-28
7.7.3 Configuring Diffserv ....................................................................... 5-29
7.7.4 Configuration Examples for Classification and Queuing ............... 5-30
Section 8 Port Designations ................................................................................. 5-33
Chapter 6 SIP Trunking

Section 1 Description .............................................................................................. 6-1
Section 2 SIP Trunk E.164 Support ........................................................................ 6-4
Section 3 SIP Trunk E.164 CLIP Enhancement ..................................................... 6-5
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Section 4 Video Support over SIP Trunks ............................................................. 6-6
Section 5 Guide to Feature Programming ............................................................. 6-8
Section 6 SIP Trunk Keep Alive using OPTION Message .................................. 6-38
Section 7 TLS Support on SIP .............................................................................. 6-42
Section 8 Guide to Feature Programming ........................................................... 6-48
Section 9 Multi-Gateway Address Support .......................................................... 6-51
Chapter 7 H.323 Trunking

Section 2 H.323 Trunk Implementation .................................................................. 7-2
2.1 Determine Numbering Plan ................................................................. 7-2
2.2 Configure VoIPDB Trunk Ports ............................................................ 7-3
2.2.1 H.323 Setup ..................................................................................... 7-3
2.2.2 Configure VoIPDB Networking Information ...................................... 7-4
2.2.3 VoIPDB (DSP) Basic Setup ............................................................. 7-4
2.3 Configure Signaling Information (H.323) ............................................. 7-5
2.3.1 H.323 GateKeeper Setup ................................................................ 7-5
2.3.2 H.323 Alias Address Setup .............................................................. 7-6
2.4 Configure Call Routing for H.323 ......................................................... 7-7
2.4.1 H.323 Call Routing ........................................................................... 7-7
2.4.2 Caller ID (H.323) .............................................................................. 7-8
2.5 CODEC Selection for IP Networking ................................................... 7-9

2.5.1 CODEC Selection for IP Networking (H.323) ................................. 7-10
2.6 H.323 TIE Line Programming Example ............................................. 7-12
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Chapter 8 IP Multiline Station (SIP)

Section 2 IP to TDM Conversion ............................................................................. 8-2
2.1 DT800/DT700 IP Multiline Telephones (ITL/ITZ) ................................. 8-2
2.2 Conditions ............................................................................................ 8-3
Section 3 Power Fail Adapter [PSA-L ( ) UNIT] .................................................... 8-3
3.1 Connecting to an IP Telephone ........................................................... 8-3
3.2 Operation During Power Failure .......................................................... 8-5
Section 4 LAN Connection ...................................................................................... 8-6
Section 5 Providing Power ..................................................................................... 8-7
5.1 Local Power ........................................................................................ 8-7
5.2 802.3af Power Over Ethernet (PoE) ................................................... 8-7
Section 6 Peer-to-Peer ............................................................................................. 8-7
Section 7 Miscellaneous ......................................................................................... 8-8
7.1 System Tones and Ring Tones .......................................................... 8-8
7.2 Music on Hold ..................................................................................... 8-8
Section 8 Configuration Examples ........................................................................ 8-8
8.1 IP Addressing ...................................................................................... 8-8
8.2 Example Configuration 1 - Static IP Addressing, One LAN ................. 8-9
8.3 Example Configuration 2 - Dynamic IP Addressing, One LAN .......... 8-10
8.4 Example Configuration 3 - Static IP Addressing, Routed WAN ......... 8-12
Section 9 IP Phone Programming Interface ....................................................... 8-13
Section 10 DHCP Server Configuration ................................................................ 8-14
Section 11 Auto Config for IP Terminals SV9100 ................................................. 8-14
11.1 Required Equipment .......................................................................... 8-14
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11.2 Building Config File ............................................................................ 8-15
11.3 Configuring an FTP Server ................................................................ 8-19
11.4 DHCP Server Setup Windows Server ............................................... 8-21
Section 12 IP Telephone Registration and Deletion ............................................ 8-22
Section 13 System IP Phones and Analog Trunks .............................................. 8-23
Section 14 Firmware Upgrade Procedure .............................................................. 8-23
14.1 Manually Upgrading Firmware ........................................................... 8-23
14.2 Checking the Firmware Version ......................................................... 8-24
14.3 Upgrading Automatically .................................................................... 8-24
Section 15 IP Station (SIP Multiline Telephone) .................................................... 8-26
15.1 IP Phone Registration Modes ............................................................ 8-26
15.2 General IP Configuration ................................................................... 8-28
15.3 VLANs ................................................................................................ 8-29
15.3.1 Logging In on the PC ..................................................................... 8-30
15.3.2 Tagging Voice Packets Using IP Phone ........................................ 8-31
15.3.3 Tagging Data Packets Using IP Phone .......................................... 8-33
15.3.4 Entering VLAN Settings by Phone (Voice Packets Only) .............. 8-36
15.3.5 Entering VLAN Settings for PC Port by Phone (Data Packets Only) ....
........................................................................................................8-36
15.4 ToS Settings (Layer 3 QoS) ............................................................... 8-37
15.5 Bandwidth .......................................................................................... 8-44
15.6 Some Network Considerations .......................................................... 8-47
15.7 Guide to Feature Programming ......................................................... 8-51
15.8 SIP MLT Quick Startup Guide ........................................................... 8-61
15.8.1 Plug and Play ................................................................................. 8-61
15.8.2 Automatic Registration ................................................................... 8-66
15.8.3 Manual Registration ....................................................................... 8-70
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15.9 IP Phone Relocation .......................................................................... 8-73
Chapter 9 IP Single Line Telephone (SIP)

Section 2 VoIP and CPU LAN Link - Speed and Duplex Mode ............................ 9-3
Section 3 Standard SIP-TCP Connection Support ................................................ 9-5
Section 4 STD SIP Conference ............................................................................... 9-8
Section 5 STD SIP Transfer–Unattended ............................................................. 9-11
Section 6 NAT Mode for Standard SIP Telephone .............................................. 9-12
Section 7 Programming ......................................................................................... 9-18
7.1 Configure VoIPDB Networking Information ....................................... 9-18

7.1.1 VoIPDB (DSP) Basic Setup ........................................................... 9-19
7.1.2 VoIP ToS Setup ............................................................................. 9-19
7.1.3 SIP Peer-to-Peer ........................................................................... 9-19
7.1.4 IP Extension Numbering ................................................................ 9-20
7.1.5 SIP Extension CODEC Information ............................................... 9-20
7.1.6 SIP Extension Basic Information Setup ......................................... 9-23
7.1.7 IP Phone Configuration ................................................................. 9-23
7.2 SIP Phone Example .......................................................................... 9-25
Chapter 10 SV9100 NetLink

Section 1 Introduction ........................................................................................... 10-1
Section 2 SV9100 Requirements .......................................................................... 10-3
2.1 Hardware ........................................................................................... 10-3
2.2 Capacity ............................................................................................. 10-3
2.3 License .............................................................................................. 10-3
2.4 Protocol ............................................................................................. 10-3
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2.5 Determining System ID ...................................................................... 10-3
2.6 Determining Primary System ............................................................. 10-4
2.7 Software Information .......................................................................... 10-4
Section 3 Installation ............................................................................................. 10-4
3.1 NetLink Connection ........................................................................... 10-4
3.2 Limitations .......................................................................................... 10-9
3.2.1 Features ......................................................................................... 10-9
3.2.2 IP Terminal .................................................................................... 10-9
3.2.3 IP Trunks ....................................................................................... 10-9
3.2.4 General limitations ......................................................................... 10-9
3.3 DSP ................................................................................................. 10-15
3.4 E911 ................................................................................................ 10-16
3.5 Music on Hold .................................................................................. 10-16
3.6 Bandwidth ........................................................................................ 10-17
3.7 Conference Block ............................................................................ 10-18
3.8 Voice Mail ........................................................................................ 10-18
3.9 FoIP ................................................................................................. 10-18
Section 4 List of Supported Features in Secondary System ........................... 10-18
Section 5 NetLink Functions ............................................................................... 10-26
5.1 Programming ................................................................................... 10-26
Section 6 Fail-Over ............................................................................................... 10-31
Section 7 Top Priority Primary System (Primary System Automatic Integration) ..

................................................................................................................10-36
7.1 Forced Change Primary System ...................................................... 10-37
7.2 IP Settings ....................................................................................... 10-38
Section 8 Communication Server Database Synchronization ......................... 10-38
8.1 Asynchronous Settings .................................................................... 10-39
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8.2 SRAM Database .............................................................................. 10-40
Section 9 Create Recovery Database ................................................................. 10-41
Section 10 Licensing Control ............................................................................... 10-41
Section 11 MAC Address Authorization .............................................................. 10-42
Section 12 Maintenance Features ........................................................................ 10-43
Section 13 Port Assignment ................................................................................. 10-43
Section 14 Slot Control ......................................................................................... 10-44
Section 15 Time Zone Setup ................................................................................. 10-46
Section 16 VoIP Handling ...................................................................................... 10-47
Section 17 NetLink Multi-SIP Carrier .................................................................... 10-52
Section 18 How to Enter Programming Mode ..................................................... 10-54
Section 19 How to Exit Programming Mode ........................................................ 10-55
Section 20 Using Keys to Move Around in the Programs .................................. 10-55
Section 21 Programming Names and Text Messages ........................................ 10-57
Chapter 11 NAPT
Section 1 NAPT ...................................................................................................... 11-1
1.1 Introduction ........................................................................................ 11-1

1.2 Requirements .................................................................................... 11-2
1.2.1 Main Software ................................................................................ 11-2
1.2.2 Hardware ....................................................................................... 11-2
1.2.3 Capacity ......................................................................................... 11-2
1.2.4 License .......................................................................................... 11-3
1.3 Installation ......................................................................................... 11-3
1.3.1 Settings for Terminals .................................................................... 11-3
1.4 Programming Example .................................................................. 11-5
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Section 2 Conditions ........................................................................................... 11-12
Section 3 Restrictions – Static NAT ................................................................... 11-13
Section 4 Minimum Required Programming ..................................................... 11-14
Chapter 12 All DSP Busy Indication

Section 1 Introduction ........................................................................................... 12-1
Section 2 Service Conditions ................................................................................ 12-2
Section 3 Related Features ................................................................................... 12-3
Chapter 13 AspireNet
1.1 What is AspireNet? ............................................................................ 13-1
1.2 SV9100 PCBs .................................................................................... 13-2
1.3 Available Features ............................................................................. 13-3
1.4 Using This Manual ............................................................................. 13-4
1.5 Unique Considerations ...................................................................... 13-4
2.1 Required System Programming ......................................................... 13-5
2.2 AspireNet ISDN ................................................................................. 13-5
2.3 AspireNet IP ....................................................................................... 13-7
2.3.1 System IP Address ........................................................................ 13-8
2.3.2 IPLE Card IP Address .................................................................... 13-8
2.3.3 IP Address of other systems within the network ............................ 13-9
2.3.4 Networking TCP Port ..................................................................... 13-9
2.3.5 Networking TCP Port ................................................................... 13-10
2.4 AspireNet Multi Site ......................................................................... 13-11
2.5 Numbering Plan ............................................................................... 13-12
3.1 ARS/F-Route ................................................................................... 13-17
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3.2 Barge In ........................................................................................... 13-19
3.3 Call Forwarding ............................................................................... 13-20
3.4 Call Forwarding/ Do Not Disturb Override ...................................... 13-21
3.5 Call Forward/ Off-Premise ............................................................... 13-21
3.6 Call Forwarding with Follow Me ....................................................... 13-22
3.7 Camp On ......................................................................................... 13-24
3.8 Caller ID Display .............................................................................. 13-25
3.9 Central Office Calls, Placing: Seizing a trunk in a networked system .......
..........................................................................................................13-26
3.10 Conference ...................................................................................... 13-27
3.11 Department Calling .......................................................................... 13-29
3.12 Department Step Call ...................................................................... 13-30
3.13 Direct Inward Dialing (DDI) .............................................................. 13-31
3.14 Direct Inward Line (DIL) ................................................................... 13-32
3.15 Direct Inward System Access (DISA) .............................................. 13-33
3.16 Hold ................................................................................................. 13-34
3.17 Hotline / Direct Station Selection (DSS) .......................................... 13-35
3.18 Intercom ........................................................................................... 13-36

3.19 Keep Alive Operation ....................................................................... 13-37
3.20 Last Number Redial ......................................................................... 13-38
3.21 Message Waiting ............................................................................. 13-39
3.22 Operator, Centralized ...................................................................... 13-40
3.23 Paging ............................................................................................. 13-41
3.24 Park ................................................................................................. 13-43
3.25 Ringdown Extension, Internal/External ............................................ 13-44
3.26 Selectable Display Messaging ......................................................... 13-45
3.27 Toll Restriction ................................................................................. 13-46
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3.28 Transfer ........................................................................................... 13-47
3.29 Voice Mail, Centralized .................................................................... 13-49
4.1 Before Reading This Section ........................................................... 13-54
4.2 How to Use This Section ................................................................. 13-54
4.3 How to Enter Programming Mode ................................................... 13-55
4.4 How to Exit Programming Mode ...................................................... 13-56
4.5 Using Keys to Move Around in the Programs .................................. 13-56
4.6 Programming Names and Text Messages ...................................... 13-57
4.7 Using Softkeys For Programming .................................................... 13-58
4.8 What the Softkey Display Prompts Mean ........................................ 13-59
5.1 AspireNet - ISDN BRI .................................................................... 13-164
5.2 AspireNet - ISDN PRI .................................................................... 13-165
5.3 AspireNet - IP ................................................................................ 13-166
5.3.1 VoIP Networking Example ......................................................... 13-166
5.3.2 Multi-Site Networking Example ................................................. 13-168
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LIST OF FIGURES
Introduction

Figure 1-1 K-CCIS System Outline .................................................................................................1-1
Figure 1-2 Clock Supply Route .......................................................................................................1-2
Figure 1-3 K-CCIS System Configuration .......................................................................................1-3
Figure 1-4 Frame Configuration of 24-DTI (12 Multiframe) .............................................................1-4
Figure 1-5 Frame Configuration of 24-DTI (24 Multiframe) .............................................................1-6
Figure 1-6 Star Topology (KTS-to-KTS or PBX-to-KTS) .................................................................1-8
Figure 1-7 Tree Topology (KTS-to-KTS or PBX-to-KTS) ................................................................1-9
Figure 1-8 Mesh Topology (PBX-to-KTS) .....................................................................................1-10
Figure 1-9 K-CCIS Routes ............................................................................................................1-11
Figure 1-10 Assigning One Common Signaling Channel Between Two Systems ..........................1-11
Figure 1-11 One GCD-CCTA Assigned per Common Signaling Channel ......................................1-12
Figure 1-12 Common Signaling Channel - Example of Incorrect Connection .................................1-12
Figure 1-13 Point Code Determinations ..........................................................................................1-13
Figure 1-14 CCH Linking .................................................................................................................1-14
Figure 1-15 Circuit Identification Codes (CIC) .................................................................................1-16
Figure 1-16 Closed Numbering Plan Example ................................................................................1-18
Figure 1-17 Open Numbering Plan Example ..................................................................................1-19

Figure 2-1 Inserting Blades in the 19” Chassis ...............................................................................2-2
Figure 2-2 GCD-CCTA Blade ..........................................................................................................2-5
Figure 2-3 GCD-CCTA LED Indication Pattern of Layer 1 on T1 Unit ............................................2-7
SV9100 Networking Manual xix

Issue 5.0

Figure 4-1 K-CCIS Call Rerouting ................................................................................................ 4-18
Figure 4-1 Link Reconnect for Station Calls ................................................................................. 4-79
Figure 4-2 Link Reconnect for Trunk Tandem Calls ..................................................................... 4-80
Figure 4-3 Multiple Call Forwarding over K-CCIS Links for All Calls ............................................ 4-85
Figure 4-4 Multiple Call Forwarding over K-CCIS Links for Busy/No Answer .............................. 4-90

Figure 5-1 Peer-to-Peer One IP to TDM ......................................................................................... 5-2
Figure 5-2 GPZ-IPLE Daughter Board ........................................................................................... 5-3
Figure 5-3 GCD-CP10 Blade with Daughter Board Installed .......................................................... 5-5
Figure 5-4 GCD-CP10 LED Locations ............................................................................................ 5-6
Figure 5-5 VoIP Connections ......................................................................................................... 5-8
Figure 5-6 Connecting a VOIPDB to a Network/PC ....................................................................... 5-9
Figure 5-7 System Data 84-10 ToS Setup ................................................................................... 5-13
Figure 5-8 Multiple Gateways ....................................................................................................... 5-17
Figure 5-9 Two SV9100 Systems Connected via the WAN ......................................................... 5-18
Figure 5-10 VPN Network ............................................................................................................... 5-19
Figure 5-11 Programming Example 1 ............................................................................................. 5-33
Figure 5-12 TEL 302 (SIP MLT) makes call to TEL 202 (SIP MLT) via IP K-CCIS ........................ 5-38
Figure 5-13 Programming Example 2 ............................................................................................. 5-39

Figure 2-1 Example IP Network Configuration ............................................................................... 2-3
Figure 2-2 Programming Example 1 ............................................................................................... 2-4

Figure 3-1 Example Configuration 1 - Existing Network with Static IP Address ............................. 3-3
Figure 3-2 Example Configuration 1 - Adding the UNIVERGE SV9100 Chassis to the Network ... 3-5
Figure 3-3 Example Configuration 2 - New Network with Dynamic Addressing ............................. 3-6
xx List of Figures
Issue 5.0
Figure 3-4 TCP/IP Properties Screen ..............................................................................................3-7

Figure 3-5 Testing the Network Connection ....................................................................................3-8

Figure 5-1 Layer 2 Diagram (802.1Q) .............................................................................................5-4
Figure 5-2 Virtual Private Network (VPN) Example .........................................................................5-9
Figure 5-3 Network Bottleneck Example .......................................................................................5-15
Figure 5-4 Voice and Data Network Implementation .....................................................................5-16
Figure 5-5 Priority Queuing on Voice and Data Networks .............................................................5-16
Figure 5-6 Protocol Structure for Layer 2 QoS ..............................................................................5-18
Figure 5-7 Ethernet Frame Example - Layer 2 QoS Enabled .......................................................5-19
Figure 5-8 Layer 3 QoS Example ..................................................................................................5-21
Figure 5-9 Ethernet Frame Example - Containing One RTP (Speech) Packet .............................5-24
Figure 5-10 Common Network with Cisco Router ...........................................................................5-30

Figure 6-17 TLS Support on SIP .....................................................................................................6-42
Figure 6-18 SIP Server Authentication ............................................................................................6-42
Figure 6-19 Certificate Registration .................................................................................................6-49
Figure 6-19 Uploading Certificate Registration ...............................................................................6-49
Figure 6-20 Certificate Registration Error Display ...........................................................................6-50
Figure 6-21 Deleting a Registered Certificate .................................................................................6-50
Figure 6-22 Multi-Gateway Address Support ..................................................................................6-51
Figure 6-23 Multi-Gateway Setup ...................................................................................................6-55
Figure 6-24 UPnP and Multi-Gateway Setup .................................................................................6-56

Figure 7-1 H.323 TIE Line Programming Example .......................................................................7-12

Figure 8-1 DT800/DT700 IP Telephone (ITL) .................................................................................8-2
Figure 8-2 Power Fail Adapter Connection .....................................................................................8-4
Figure 8-3 IP Telephone Connection ..............................................................................................8-5
SV9100 Networking Manual xxi

Issue 5.0
Figure 8-4 Typical Network IP Connection ..................................................................................... 8-6

Figure 8-5 Example Configuration 1 - Static IP Addressing, One LAN .......................................... 8-9
Figure 8-6 Example Configuration 2 - Dynamic IP Addressing, One LAN .................................... 8-10
Figure 8-7 Example Configuration 3 - Static IP Addressing, Routed WAN .................................. 8-12
Figure 8-8 IP Phone Manager ...................................................................................................... 8-15
Figure 8-9 Config Setup APL ........................................................................................................ 8-16
Figure 8-10 Assign IP Address ....................................................................................................... 8-17
Figure 8-11 Network Settings - 1st Server Port .............................................................................. 8-17
Figure 8-12 Save As/Name File Window ........................................................................................ 8-18
Figure 8-13 Quick ‘n Easy FTP Server ........................................................................................... 8-19
Figure 8-14 Quick ‘n Easy FTP Server Configuration .................................................................... 8-20
Figure 8-15 DT800/DT700 Encryption ............................................................................................ 8-28
Figure 8-16 Log In to IP Phone ...................................................................................................... 8-30
Figure 8-17 LAN Port Settings Window .......................................................................................... 8-31
Figure 8-18 VLAN Mode ................................................................................................................. 8-32
Figure 8-19 VLAN ID ...................................................................................................................... 8-32
Figure 8-20 VLAN Priority ............................................................................................................... 8-32
Figure 8-21 PC Port Settings Window ............................................................................................ 8-33
Figure 8-22 Port VLAN Mode ......................................................................................................... 8-34
Figure 8-23 Port VLAN ID ............................................................................................................... 8-34
Figure 8-24 Port VLAN Priority ....................................................................................................... 8-34
Figure 8-25 Save Network Settings ................................................................................................ 8-35
Figure 8-26 Save Confirmation Window ......................................................................................... 8-35
Figure 8-27 84-10: ToS Setup ........................................................................................................ 8-39
Figure 8-28 SIP DT800/DT700 Basic Setup ................................................................................... 8-40
Figure 8-29 Type of Service Window ............................................................................................. 8-42
Figure 8-30 RTP - Voice Packets ................................................................................................... 8-43
Figure 8-31 SIP - Signalling Packets .............................................................................................. 8-43
Figure 8-32 SV9100 Network Example No. 1 ................................................................................. 8-47
Figure 8-33 SV9100 Network Example No. 3 ................................................................................. 8-49
Figure 8-34 System Data 10-12: GCD-CP10 Network Setup ......................................................... 8-62
Figure 8-35 System Data 84-26: IPLE Basic Setup (DSP) ............................................................. 8-63
Figure 8-36 System Data 11-02: Extension Numbering ................................................................. 8-64
Figure 8-37 PCPro Programming Unregistered Phones ................................................................ 8-65
xxii List of Figures

Issue 5.0
Figure 8-38 IP Phone List ................................................................................................................8-65

Figure 8-39 PCPro Programming Unregistered Phones .................................................................8-67
Figure 8-40 IP Phone List ................................................................................................................8-67
Figure 8-41 Automatic Registration Setting .....................................................................................8-68
Figure 8-42 Automatic Registration Basic Setup .............................................................................8-68
Figure 8-43 Automatic Registration Personal ID Index ...................................................................8-68
Figure 8-44 Automatic Registration User Name and Password Assignment ..................................8-69
Figure 8-45 Manual Registration .....................................................................................................8-71
Figure 8-46 IP Phone Relocation Flow Chart ..................................................................................8-75

Figure 10-1 NetLink Fail-Over Example ..........................................................................................10-2
Figure 10-2 Two SV9100 Systems Connected Via the WAN ..........................................................10-6
Figure 10-3 VPN Network ...............................................................................................................10-7
Figure 10-4 NetLink with IP Phones connected via NAPT ..............................................................10-8
Figure 10-5 PRG 51-18-01 with setting of “4” ...............................................................................10-32
Chapter 11 NAPT
Figure 11-1 NAPT Configuration Example ......................................................................................11-2
SV9100 Networking Manual xxiii

Issue 5.0
xxiv List of Figures

LIST OF TABLES
Introduction
Table I-1 Common Terms and Associated Abbreviations ......................................................... 1-1

Table 1-1 12-Multiframe Bit Assignment .................................................................................... 1-5
Table 1-2 24-Multiframe Bit Assignment .................................................................................... 1-6

Table 2-1 Extension Blade Installation Order Example .............................................................. 2-3
Table 2-2 Adding Daughter Board to Chassis Example ............................................................. 2-3
Table 2-3 Trunk Blade Installation Order Example ..................................................................... 2-4
Table 2-4 GCD-CCTA LED Indications ...................................................................................... 2-6
Table 2-5 GCD-CCTA RJ-45 Cable Connector Pin-Outs ........................................................... 2-7

Table 3-1 Digital Trunk Data Programming Assignments .......................................................... 3-1
Table 3-2 CCIS Programming Assignments ............................................................................... 3-3

Table 3-3 Numbering Plan Assignments .................................................................................... 3-4
Table 3-4 Closed Numbering Plan Assignments ........................................................................ 3-4
Table 3-5 Open Numbering Plan Assignments .......................................................................... 3-5

Table 4-1 Programming Example ............................................................................................. 4-20
SV9100 Networking Manual xxv

Issue 5.0

Table 5-1 VOIPDB LED Indications ........................................................................................... 5-6
Table 5-2 VOIPDB LED CN1 Transmit/Receive Data Indications .............................................. 5-7
Table 5-3 IP Precedence Values .............................................................................................. 5-11
Table 5-4 ToS Field with IP Precedence Bits ........................................................................... 5-11

Table 5-5 DSCP Code Points and Binary Values .................................................................... 5-12
Table 5-6 ToS Field with Diffserv sBits .................................................................................... 5-13
Table 5-7 Digital Trunk Assignments ....................................................................................... 5-20

Table 5-8 VoIP IP Address Assignments ................................................................................. 5-23
Table 5-9 Local Numbering Plan Assignments ........................................................................ 5-24
Table 5-10 Closed Numbering Plan Assignments ...................................................................... 5-24

Table 5-11 Open Numbering Plan Assignments ........................................................................ 5-25
Table 5-12 IP K-CCIS Assignments ........................................................................................... 5-27
Table 5-13 IP K-CCIS (Peer-to-Peer) Assignments ................................................................... 5-27

Table 5-14 CCIS over IP CODEC Information Assignments ...................................................... 5-28

Table 1-1 VoIP Specifications .................................................................................................... 1-2
Table 4-1 Keys for Entering Data ............................................................................................... 4-3
Table 4-2 Keys for Entering Names ........................................................................................... 4-4
Table 4-3 Softkey Display Prompts ............................................................................................ 4-6
xxvi List of Tables

Issue 5.0

Table 5-1 Type of Service Field (IP Precedence - i Ref. REC 1349) ....................................... 5-23
Table 5-2 Diffserv Parameters ..................................................................................................5-24

Table 5-3 IP Precedence and Diffserv Values Comparison ..................................................... 5-26
Table 5-4 Voice Protocols ........................................................................................................ 5-28
Table 5-5 DIffserv Configuration ...............................................................................................5-29
Table 5-6 Cisco Router Configuration Example ....................................................................... 5-31
Table 5-7 Port Designations for IP Applications ....................................................................... 5-33

Table 6-1 SIP INVITE Header Fields ........................................................................................ 6-36
Table 6-2 Delete + from Incoming SIP INVITE ......................................................................... 6-36

Table 6-3 Delete + and Country Code from Incoming SIP INVITE ........................................... 6-37
Table 6-4 Delete + and Country Code from Incoming SIP INVITE ........................................... 6-37
Table 6-5 SIP Packet Sequence .............................................................................................. 6-41

Table 6-6 Multi-Gateway Programming .................................................................................... 6-55
Table 6-7 Multi-Gateway Programming .................................................................................... 6-57

Table 7-1 H.323 Setup Assignments .......................................................................................... 7-3
Table 7-2 VoIPDB Network Configuration Assignments ............................................................ 7-4
Table 7-3 VoIPDB (DSP) Basic Assignments ............................................................................ 7-4
Table 7-4 H.323 GateKeeper Setup Assignments ..................................................................... 7-5
Table 7-5 H.323 Alias Address Setup Assignments ................................................................... 7-6
Table 7-6 H.323 Call Routing Assignments ................................................................................ 7-7
Table 7-7 H.323 Caller ID Assignments ..................................................................................... 7-8
Table 7-8 IP Networking (H.323) CODEC Assignments .......................................................... 7-10
SV9100 Networking Manual xxvii

Issue 5.0
Table 8-1 IP Phone Programming Options User Menu ............................................................ 8-13
Table 8-2 Common IP Precedence/Diffserv Values and Hexadecimal Equivalent .................. 8-41
Table 8-3 IP Phone Relocation ................................................................................................ 8-74

Table 10-1 The Data Which Isn’t a Target of a Replication ...................................................... 10-11
Table 10-1 VoIP Resource Chart ........................................................................................... 10-13
Table 10-2 CPU Reset Programs ............................................................................................. 10-14

Table 10-3 Supported Features in Secondary System ............................................................ 10-18
Table 10-4 The Data Which Isn’t a Target of a Replication ...................................................... 10-39
Table 10-5 Keys for Entering Data ........................................................................................... 10-55
Table 10-6 Keys for Entering Names ....................................................................................... 10-57
Chapter 11 NAPT

Table 12-1 Alarm Types ............................................................................................................. 12-1
Table 13-2 Keys for Entering Data ........................................................................................... 13-56
Table 13-3 Keys for Entering Names ....................................................................................... 13-57
Table 13-4 Softkey Display Prompts ........................................................................................ 13-59
Table 13-5 System Numbering Default Settings ...................................................................... 13-83
xxviii List of Tables

UNIVERGE® SV9100
Introduction
When the GPZ-IPLE daughter board is installed, half-duplex connections are not supported. For
troubleshooting purposes, a managed switch capable of port mirroring is required to capture
packet data from the SV9100 IPLE Ethernet port.
IMPORTANT
SECTION 1 GENERAL OVERVIEW
This manuals provides information for networking for the UNIVERGE SV9100
systems. Networking can be accomplished using one of the following methods:
 SV9100 K-CCIS (US Only)
 SV9100 IP K-CCIS
 SV9100 NetLink
SECTION 2 MANUAL ORGANIZATION
This manual is divided into two books to describe the networking functions: the K-
CCIS networking/K-CCIS IP networking and IP networking.
SECTION 3 COMMON TERMS
The following terms and the associated abbreviations or alternate nomenclature may
be found throughout this document.
Table I-1 Common Terms and Associated Abbreviations
Term Abbreviation Description
GCD-CP10 GCD-CP10 SV9100 CPU

SV9100 CPU
CPU
GPZ-IPLE VOIPDB VoIP Daughter Board, mounted on
VOIP Daughter Board GCD-CP10
SV9100 Networking Manual I-1

Issue 5.0
Table I-1 Common Terms and Associated Abbreviations (Continued)
Term Abbreviation Description
GCD-2BRIA BRIU 2-port Basic Rate Interface blade

2BRIU
GPZ-2BRIA 2BRIUDB 2-port basic rate daughter board, mounted on
BRIUDB the GCD-2BRIA
GCD-CCTA CCHU CCIS Trunk Interface blade
CCIS CCHU
GCD-4COTA/C 4COIU 4-port loop/ground start trunk blade
GPZ-4COTE/G 4COIUDB 4-port loop/ground start daughter board,
4COIU Daughter Board mount on GCD-4COTA/C blade
GCD-8DLCA 8ESIU 8 digital station interface blade
GCD-16DLCA 16ESIU 16 digital station interface blade
GPZ-8DLCB 8ESIUDB 8 digital station interface daughter board,
8ESIU Daughter Board mounted on the GCD-8DLCA
CD-PVAA PVAA Conference Bridge blade
GCD-RGA RGA Application Gateway
DT400/DT300 Series DT400/DT300 Digital SV9100/SV9300 Digital Multiline Terminals.
Multiline Terminal Telephone equipment names begin with DTL/
DTZ/DTL Telephones DTZ. For example: DTL-12D-1 TEL, DTL-
Digital Multiline 24D-1 TEL, DTL-32D-1 TEL, DTZ-12D-3 TEL,
Telephones DTZ-24D-3 TEL, etc.
DT800/DT700 Series DT800/DT700 IP Digital SV9100/SV9300 IP Digital Multiline
Multiline Terminal Terminals. Telephone equipment names
ITZ/ITL Telephones begin with ITL/ITZ. For example: ITZ-12D-3
IP Digital Multiline TEL, ITZ-24D-3 TEL, ITL-32D-1 TEL, etc.
Telephones
GCD-4DIOPB DIOPU 4-port DID/OPX blade
4DOIPU
GCD-4ODTB TLIU 4-port E&M tie line blade
4TLIU
I-2
UNIVERGE® SV9100
General Information <US Only>

Chapter 1
This chapter provides a system outline, the name and functions of the circuit cards
required, system capacity, time slot assignments, system specifications and network
structure considerations for UNIVERGE SV9100 K-CCIS.
SECTION 1 SYSTEM OUTLINE
Key-Common Channel Interoffice Signaling (K-CCIS) can interface this system with
a Public Network. The system is configured with the (GCD-CCTA) for CCIS Trunk
Interface/CCH. The GCD-CCTA is the Digital Trunk Interface required for receiving/
transmitting common signaling data from/to the distant office. A Phase Locked
Oscillator (PLO) for digital network synchronization is built into the GCD-CP10.
The system can provide a variety of interoffice service features such as Link
Reconnect, Virtual Look Ahead Routing, Centralized Voice Mail/Auto Attendant
Integration, Call Forwarding, Voice Call with Hands free Answerback and Caller ID
Display. For a more detail description refer to Chapter 4 . For a diagram of the
system outline, refer to Figure 1-1 K-CCIS System Outline.
Figure 1-1 K-CCIS System Outline
SV9100 Networking Manual 1-1

Issue 5.0
1.1 Common Channel Handler (GCD-CCTA)
The Common Channel Handler (GCD-CCTA) blade provides a common channel

signal to a UNIVERGE SV9100 network. It is responsible for signaling between
the Key Telephone System (KTS) and the UNIVERGE SV9100 network under
control of the GCD-CP10.
1.2 Phase Locked Oscillator (PLO)
The Phase Locked Oscillator (PLO) provides synchronization between the TDSW
and other offices and is built into the GCD-CP10. The clock generates a
synchronized clock signal according to the source clock signals supplied from the
source office in the network, and supplies the generated clock signal to the
TDSW. The clock is supplied with clock signals extracted from the GCD-CCTA.
Figure 1-2 Clock Supply Route
1.3 Common Channel Interoffice Signaling (K-CCIS)
The PBX can provide connection to the UNIVERGE SV9100 using a digital
network. The network requires the Common Channel Handler (GCD-CCTA) to
control the common signaling between offices.
Digital Network
When UNIVERGE SV9100 is provided through a digital network, the CCH and T-1
functionality is combined on the GCD-CCTA to provide a fixed path in the TDSW
and to transmit and receive common signaling data to/from the distant office
through a predetermined channel. Voice signals are transmitted and received per
line through other channels.
1-2 General Information <US Only>

Issue 5.0
Figure 1-3 K-CCIS System Configuration shows the system configuration of

K-CCIS provided using a digital network.
Figure 1-3 K-CCIS System Configuration
SECTION 2 DTI SPECIFICATIONS
The following specifications apply to the GCD-CCTA.
2.1 Characteristics
Output
Line Rate 1.544 Mbps + 50 ppm
Line Code AMI with ZCS/B8ZS 
Line Impedance 100 
Pulse Amplitude (Base to Peak) 3 volts ± 0.6 volts
Pulse Width 3214 ns ± 30 ns

Issue 5.0
Input
Line Rate 1.544 Mbps ± 200 ppm (130 ppm)
Pulse Amplitude (Base to Peak) 1.5 volts ~ 3 volts
Frame Synchronization Pattern 100011011100
Input Jitter ITU-T Fig.1/G743
Wander ±138UI, –193UI or –138UI, +193UI

Cable Length from SV9100 to Maximum 196m (655 ft.) with
MDF or External Equipment (22 AWG) twisted-pair cable
 AMI: Alternate Mark Inversion
ZCS: Zero Code Suppression
B8ZS: Bipolar Eight Zero Substituting
2.2 Frame Configuration for 24 DTI
According to the AT&T Specifications for 24-channel transmission, there are two
frame configurations: 12 Multiframe (D4) and 24 Multiframe (ESF).
12 Multi-Frame (D4)
This frame has 12 Multiframes, and each Multiframe has a 24-channel PCM
signal (8 bits/channel) and an S (Superframe) bit. Figure 1-4 Frame Configuration
of 24-DTI (12 Multiframe) shows the frame configuration, and Figure 1-5 Frame
Configuration of 24-DTI (24 Multiframe) shows the frame bit assignment.
Figure 1-4 Frame Configuration of 24-DTI (12 Multiframe)
CH1 CH2 CH3 CH24
s 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8
s FRAME 1 FRAME 2 FRAME 3 FRAME 12
128 S
S: Superframe Bit

Issue 5.0
Table 1-1 12-Multiframe Bit Assignment
S-Bit
Frame
Number Terminal Signal
Synchronization (FT) Synchronization (FS)
1 1
2 0
3 0
4 0
5 1
6 1
7 0
8 1
9 1
10 1
11 0
12 0
 The S-bit is the first bit in each frame.
Frames are repeated in the order shown in this table.

Issue 5.0
24-Multiframe (Extended Superframe – ESF)
This frame has 24 Multiframes and each Multiframe has a 24-channel PCM signal
(8 bits/channel) and an S (Superframe) bit.
Figure 1-5 Frame Configuration of 24-DTI (24 Multiframe)
CH1 CH2 CH3 CH24
s 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8 1 2 3 4 5 6 7 8
s FRAME 1 FRAME 2 FRAME 3 FRAME 24
128 S
S: Superframe Bit
Table 1-2 24-Multiframe Bit Assignment
S-Bit
Frame
Number Frame 4 kbps
CRC
Synchronization Data Link
1 m
2 CB1
3 m
4 0
5 m
6 CB2
7 m
8 0
9 m
10 CB3
11 m
12 1

Issue 5.0
Table 1-2 24-Multiframe Bit Assignment (Continued)
S-Bit
Frame
Number Frame 4 kbps
CRC
Synchronization Data Link
13 m
14 CB4
15 m
16 0
17 m
18 CB5
19 m
20 1
21 m
22 CB6
23 m
24 1
 The S-bit is the first bit in each frame.
Frames are repeated in the order shown in this table.
The letter m in the 4 kbps Data Link column indicates the frame is usually
assigned to 1.
SECTION 3 NETWORK STRUCTURE CONSIDERATIONS
3.1 Determining System Configurations
The configuration of the network and the number of lines (channels) is determined
by the traffic between each office.
The topologies listed in this section are supported in the UNIVERGE SV9100
system KTS-to-KTS structure.
 Star Topology – Refer to Figure 1-6 Star Topology (KTS-to-KTS or

PBX-to-KTS).
 Tree Topology – Refer to Figure 1-7 Tree Topology (KTS-to-KTS or

PBX-to-KTS).

Issue 5.0
 Mesh Topology is supported only when the KTS is the end-point in a

PBX-to-KTS network. Refer to Figure 1-8 Mesh Topology (PBX-to-KTS).
Figure 1-6 Star Topology (KTS-to-KTS or PBX-to-KTS)
Main Hub
B C D E

Issue 5.0
Figure 1-7 Tree Topology (KTS-to-KTS or PBX-to-KTS)
3 5
Tree Topology supports a total of 255 systems. Even though 255 systems are al-
lowed, only five hops* are permitted. Software does not limit the number of hops.
The limitation is due to the CCH message delay through each tandem system.
 Hops - Tandem through another system

Issue 5.0
Figure 1-8 Mesh Topology (PBX-to-KTS)
NEAX2000/ NEAX2000/
NEAX2400 NEAX2400
NEAX2000/
NEAX2400
NEAX2000/
NEAX2400
NEAX2000/
NEAX2400
NEAX2000/
NEAX2400
UNIVERGE SV9100
UNIVERGE SV9100
A UNIVERGE SV9100 can be connected to a PBX only as a remote

office. The UNIVERGE SV9100 software can register a maximum of
255 point codes in the UNIVERGE SV9100 network. The network must
consist of a PBX-to-KTS structure with the KTS programmed as a
remote office. The KTS must be located as the end-point in the
UNIVERGE SV9100 network.
3.2 Determining Number of K-CCIS Routes
When the system is a Central Office or Tandem Office, two or more routes to other
offices are required. Each GCD-CCTA can support one K-CCIS link. Up to eight
GCD-CCTA blades can be installed in a UNIVERGE SV9100 system. The KTS
requires the GCD-CCTA to support a K-CCIS interface.
One Common Signaling Channel (CSC) can support up to 127 voice channels, if
needed.

Issue 5.0
One GCD-CCTA provides 24 channels. One Common Signaling Channel port

must be assigned on the Digital Trunk Interface in case two systems are
connected by one Digital Link.
Figure 1-9 K-CCIS Routes
UNIVERGE SV9100 23 Voice Channels UNIVERGE SV9100
GCD-CCTA
GCD-CCTA
One Common Signaling Channel
System A has one GCD-CCTA.
One Common Signaling Channel can be assigned even if two Digital Links are
connected between two systems. Only one Common Signaling Channel can be
assigned per GCD-CCTA in each system.
Figure 1-10 Assigning One Common Signaling Channel Between Two Systems
24 Voice Channels
GCD-CCTA
GCD-CCTA
UNIVERGE SV9100 UNIVERGE SV9100
GCD-CCTA
GCD-CCTA
23 Voice Channels

Issue 5.0
One GCD-CCTA can be assigned to one Common Signaling Channel. Tandem

K-CCIS connections require one GCD-CCTA for every connection as shown in
Figure 1-11 One GCD-CCTA Assigned per Common Signaling Channel.
Figure 1-11 One GCD-CCTA Assigned per Common Signaling Channel
23 Voice Channels
UNIVERGE SV9100 One Common Signaling Channel
GCD-CCTA
GCD-CCTA
UNIVERGE SV9100
UNIVERGE SV9100
GCD-CCTA
GCD-CCTA
23 Voice Channels
Common Signaling Channels cannot be connected to another GCD-CCTA. They

must be assigned to their own physical link.
Figure 1-12 Common Signaling Channel - Example of Incorrect Connection

GCD-CCTA
UNIVERGE SV9100
GCD-CCTA
Common Signaling Channels cannot be connected to another GCD-CCTA.
3.3 Determining the Type of Transmission Lines
The transmission lines, available on the UNIVERGE SV9100 system, are digital
only (GCD-CCTA).

Issue 5.0
3.4 Determining which Systems Should be the Central Office
If using a KTS-to-KTS only network and features such as Voice Mail Integration –
K-CCIS are used, the key system that has the voice mail system installed, must
be programmed as the Central/Originating Office. All other key systems must be
programmed as Remote/Destination offices.
3.5 Determining Point Codes
Point Codes are used to distinguish an originating office from a destination office
in the K-CCIS network. A Point Code is assigned to each office in the K-CCIS
network. The following guidelines apply when determining the Point Codes:
 The Point Code cannot be assigned to more than one office.
 The same origination Point Code must be assigned to each K-CCIS channel in
the same system.
 The maximum number of Point Codes that can be assigned is 255 (a maximum
of 256 offices can be connected in the same network).
Figure 1-13 Point Code Determinations
Point Code: 00001 A CCIS Route B Point Code: 00002

CCIS Route 1:00002 CCIS Route 1 CCIS Route 1:00001
CCIS Route 2:00003 CCIS Route
CCIS Route 1
C Point Code: 00003

CCIS Route 1:00001
 Data Assignment for System A
CCIS
PRG Point Code Remarks
Route ID
50-02-03 1 00001 Assign the Originating Point

Code for System A
2 00001
50-02-04 1 00002 Assign the Destination Point
Code for CCH 1
2 00003 Assign the Destination Point
Code for CCH 2

Issue 5.0
 Data Assignment for System B
CCIS
Route ID
50-02-03 1 00002 Assign the Originating Point Code for

System B
50-02-04 1 00001 Assign the Destination Point Code
for CCH 1
 Data Assignment for System C
CCIS
Route ID
50-02-03 1 00003 Assign the Originating Point Code

for System C
50-02-04 1 00001 Assign the Destination Point Code
for CCH 1
3.6 Determining CCH Link to Send Messages
The tandem office must be programmed with the proper information to indicate
how the CCH (in its own system) is connected to other offices in the network.
Every office linked to the CCH must be identified by assigning the Point Codes
that it accepts in the network. Refer to Figure 1-14 CCH Linking. When using
features such as the Link Reconnect – K-CCIS, this must be programmed using
Program 50-06-01. Program 50-03-01 allows the links between each system and
the CCH, through which it is directed, to be assigned.
Figure 1-14 CCH Linking
Point Code: 00002

CCIS Route 1:00001
CCIS Route 2:00003
1:PC00001.Route 1
Point Code: 00004
B
2:PC00003.Route 2 D CCIS Route 1:00003
3:PC00004.Route 2 All of CCIS use Route 1
CCIS
CCIS CCIS
Route
Route Route
CCIS CCIS
CCIS Route Route
Route Point Code: 00003
C CCIS Route 1:00002
A Point Code: 00001 CCIS Route 2:00004
CCIS Route 1:00002 1:PC00001.Route 1
All of CCIS use Route 1 2:PC00002.Route 1
3:PC00004.Route 2

Issue 5.0
 Data Assignment for System A
CCIS
Route ID
50-03-01 None None It is unnecessary to assign system

data for this office.
 Data Assignment for System B
CCIS
Route ID
50-03-01 1 00001 Assign the data to create a link

between Office A and Office B
2 00003 using the CCH1 and a link
2 00004 between Office B and C using
CCH2.
 Data Assignment for System C
CCIS
Route ID
50-03-01 1 00001 Assign the data to create a link

between Office B and Office C
1 00002 using the CCH1 and a link
2 00004 between Office C and D using
CCH2.
 Data Assignment for System D
CCIS
Route ID
50-03-01 None None It is unnecessary to assign system

data for this office.
3.7 Determining Circuit Identification Code (CIC)
The GCD-CCTA trunk must distinguish between Voice Path and Common
Signaling channel. The trunks using Voice Path are assigned a CIC number for
each T1 trunk. The CIC numbers must match those in the connected system.
Refer to Figure 1-15 Circuit Identification Codes (CIC). The maximum value of a
CIC that can be assigned is 127.

Issue 5.0
Figure 1-15 Circuit Identification Codes (CIC)

Issue 5.0
3.8 Determining Numbering Plan
The Uniform Numbering Plan is the numbering plan in the K-CCIS network. The
F-Route (Flexible Route Selection) and the Automatic Route Selection (ARS)
feature provide the Open Numbering Plan.
When an outgoing call is placed through a K-CCIS link, the originating station
number (Office Code and Station Number) and a terminating Station Number are
transmitted across the link to the destination office. The originating station
number consists of the office number assigned in Program 50-04-01 and the
station number assigned in 11-02-01 for the station. These features can be used
to edit the dialing string and specify the dialing digits to add to the programmed
data to specify the maximum dialing digits in an F-Route table setting.
In a Closed Numbering Plan network, the user can make a call to another station
by dialing the station number of the distant system extension. When a Closed
Numbering plan is used the extension in the network cannot have the same prefix
numbering. Example: System A extensions are 100, System B extensions are
200.
Figure 1-16 Closed Numbering Plan Example and Figure 1-17 Open Numbering
Plan Example provide examples of Station Numbering (Closed Numbering) and
Office Code and Station Numbering (Open Numbering).

Issue 5.0
Figure 1-16 Closed Numbering Plan Example
KTS B
KTS A KTS C
2500 2501
2100 2101 2800 2801
When a call is originated from Office A to Office C, 2800 is dialed.

28 00
Station Number
Office Location (Access Code
analyzed by Program 44-02-01)
 When using a Closed Numbering Plan, the station numbers can have two to
eight digits.

Issue 5.0
Figure 1-17 Open Numbering Plan Example
KTS B
Office Code 57
KTS A KTS C
Office Code 56 Office Code 58
2100 2101
2100 2101 2100 2101
 When a call is originated from Office A to Office C, 8 + 58 + 2100 is dialed.

8 = Access Code for F-Route in PRG 11-01-01
858 = Office Code for System C in PRG 44-02-01 (F-Route Dial Analyze Table)
2100 = Station Number
The Station Number can have two to eight digits.
Program 50-04-01 allows a maximum of four digits, including the Access Code and the
Office Code.
 When using the Open Numbering Plan, the following combination of digits can be used:
 When the Access Code is set for two digits, the Office Code can have only two digits.
Access Code = XX
Office Code = XX
Station Number = XXXX
 When the Access Code is set for one digit, the Office Code can have two or three digits.
Access Code = X
Office Code = XX or XXX
Station Number = XXXXX

Issue 5.0

UNIVERGE® SV9100
Hardware Installation <US Only> Chapter 2
SECTION 1 INSTALLATION PRECAUTIONS
Preinstallation planning is essential. Advanced planning minimizes installation time,

cost, and disruption of the customer business activities.
Observe the following precautions when installing the blades to avoid static
electricity damage to hardware or exposure to hazardous voltages.
WARNING
 Never install telephone wiring during a lightning storm.

 Never install telephone jacks in wet locations unless the jack is specifically designed
for wet locations.
 Never touch uninsulated telephone wires or terminals unless the telephone line is
disconnected at the network interface.
 Use caution when installing or modifying telephone lines.
 Ground the Controlling and Expansion chassis before installing or removing the
blades.
 The Expansion Chassis must be installed with the system power Off.
 Do not touch the blade components.
 Carry the blade in a conductive polyethylene bag to prevent static electricity until
ready to install the blade.
 When installing or removing the blades from the chassis, the installer must wear a
grounded wrist strap to protect the blade from static electricity.
 Although it is recommended to install the blades with the system power off, most
blades can be installed hot.
Ensure the power is turned OFF before installing the following blades:
 GCD-CP10
 GPZ-BS10 and GPZ-BS11

Issue 5.0
1.1 Busying Out Extension/Line Blades
The extension/trunk blades may "busy-out" idle circuits. Extensions/lines cannot

make or receive calls during this condition. Calls in progress before the blade is
"made-busy" are not affected. The blade can be pulled out without interrupting a
call in progress.
An extension/line blade LED status:

 Normally flashes
 Lights steady when "made-busy" when an extension/line is in use
 Goes out when the all extensions/lines are "made-busy" (idle)
SECTION 2 INSTALLING AN EXTENSION OR TRUNK BLADE
2.1 Installing the Blades
To install an extension/trunk blade with the system running:
1. Insert the blade in the guide rail and push the blade securely into position.
Tighten the thumb screw on each side of the blade.
2. The Status LED starts flashing when the blade starts processing (15
seconds).
Figure 2-1 Inserting Blades in the 19” Chassis
2-2 Hardware Installation <US Only>

Issue 5.0
2.2 Order of Installing Extension Blades
The order in which the station blades (ESIU and SLIU) are
physically inserted determines the numbering plan.
To avoid unexpected extension/trunk numbering if the VoIP or Voice
Mail Daughter Board register with the system first, install the
CAUTION GCD-CP10 blade after the other types of extension and trunk blades
are installed.
For example, when a digital station blade (GCD-16DLCA) is in Slot #1 (ext.

301~316) and three additional digital station blades are installed in the following
order, the numbering plan below applies:
Table 2-1 Extension Blade Installation Order Example
Order of Blade Slot Extension

Blade
Installation Number Numbers
1 1 GCD-16DLCA 101~116
2 2 GCD-16DLCA 117~132
3 4 GCD-8DLCA 133~148
GPZ-8LCE
4 3 GCD-8DLCA 149~164
After the initial power up of the system, subsequent power ups or resets do not
change the slot identification. System programming (Program 90-05) must be
performed to change the slot identification.
Adding any daughter board to increase the available ports or going to a higher
capacity blade (e.g., GCD-16DLCA) may require that the slot be deleted in
programming, and the blade reinstalled. In the following example, to add a
daughter board to slot 2, the blade must be removed, deleted in
Program 90-05-01, then reinstalled with the daughter board attached, otherwise
the additional ports are not recognized. This however, uses new ports for the
combined blade – the initial ports (ports 17~24 using the example below) are not
used.
Table 2-2 Adding Daughter Board to Chassis Example
Initial Blade Updated Blade
Blade Extension Blade Extension

Blade Blade
Slot # Numbers Slot # Numbers
1 GCD-16DLCA 101~116 1 GCD-16DLCA 101~116

2 GCD-8DLCA 117~124 2 — —
(no daughter
board)

Issue 5.0
Table 2-2 Adding Daughter Board to Chassis Example
3 GCD-16DLCA 125~140 3 GCD-16DLCA 125~140

4 GCD-8DLCA 141~156
(with daughter
board)
The system automatically recognizes each Blade installed in the system. If a

Blade was previously installed in a slot and another type of Blade is to be installed
in that same slot, the Blade must be removed from the chassis and then the slot
definition removed using Program 90-05 prior to installing the new Blade.
This same condition applies to extensions and other devices connected to the
system. For example, if a port was previously used for a telephone and a DSS
Console is to be installed in that same port, the telephone must first be undefined
in Program 10-03 before the console is connected.
2.3 Order of Installing Trunk Blades
2.3.1 Installing COIU-LS1/LG1, GCD-4ODTA, GCD-4DIOPA, or

GCD-2BRIA Blades
The order in which trunk blades are physically inserted determines the
numbering plan. :
To avoid unexpected extension/trunk numbering if the VoIP or

Voice Mail blades register with the system first, install
CAUTION GCD-CP10 blade after the other types of extension and trunk
blades are installed.
For example, if four blades are installed in the following order, the numbering
plan below would apply.
Table 2-3 Trunk Blade Installation Order Example
Order of Blade Slot

Blade Line Circuits
Installation Number
1 4 4COIU 1~4
2 5 4COIU 5~8
3 7 4TLIU 9~12
4 6 4TLIU 13~16

Issue 5.0
SECTION 3 INSTALLING THE GCD-CCTA (CCIS Trunk Interface)
Figure 2-2 GCD-CCTA Blade
3.1 Description
The GCD-CCTA blade is common to both UNIVERGE SV9100 and SV9300

systems.
The Common Channel Handler Interface blade is a digital trunk blade that
terminates FT1 trunks (up to 24 DS-0 channels) providing a common channel
signal interface.
The GCD-CCTA (Common Channel Handler) is an optional blade that provides a

common channel signal through the GCD-CCTA to a K-CCIS network and
controls the signaling between the KTS and the CP10. Each GCD-CCTA blade
supports one K-CCIS link. Eight GCD-CCTA blades can be installed per system.
The T1 interface has a single 24 channel 64kb/s digital signal circuit that can be
configured for T1 trunking.

Issue 5.0
3.2 Installation
Install the GCD-CCTA in any universal slot.
3.3 LED Indications
LED indications for the GCD-CCTA are listed in Table 2-4 GCD-CCTA LED
Indications. Each LED is listed with its associated function and LED and
Operational status. Refer to Figure 2-3 GCD-CCTA LED Indication Pattern of
Layer 1 on T1 Unit on page 2-7 for LED pattern information.
Table 2-4 GCD-CCTA LED Indications
Alarm Details of the Alarm LED Indication Pattern
LOS Loss of Signal (Red Alarm) Following an alarm blink (red,

No Signal (Analog Interface) green, red, green) a Red LED will
light.
AIS Alarm Indication Signal Following an alarm blink (red,
(Blue Alarm) green, red, green) a Red LED
slowly flashes On and Off twice.
OOF Out Of Frame (Red Alarm) Following an alarm blink (red,
green, red, green) a Red LED
and Green LED flash On and Off
three times simultaneously.
RAI Remote Alarm Indication (Yellow Following an alarm blink (red,
Alarm) green, red, green) a Green LED
flashes On and Off twice.
No Alarm The system does the LED control.
 The order of priority is set up to alarm in the order LOS>AIS>OOF>RAI.

Issue 5.0
Figure 2-3 GCD-CCTA LED Indication Pattern of Layer 1 on T1 Unit
3.4 Connectors
Table 2-5 GCD-CCTA RJ-45 Cable Connector Pin-Outs shows the pin-outs for the
RJ-45 connector. Refer to Figure 2-2 GCD-CCTA Blade on page 2-5 for an
illustration showing the location of the connectors on the GCD-CCTA blade.
Table 2-5 GCD-CCTA RJ-45 Cable Connector Pin-Outs
RJ-45 Cable Connector – CN2
Pin No. Connection
1 RA
2 RB
3 —
4 TA
5 TB
6 —
7 —
8 —

Issue 5.0

UNIVERGE® SV9100
System Data Programming <US Only> Chapter 3
SECTION 1 K-CCIS PROGRAMMING
This chapter lists the Programs that must be assigned to support K-CCIS. The
Programming used depends on the K-CCIS features that are used. The tables
provided in this section provide a complete list of the required Programs that support
the function (e.g., Digital Trunk Assignment, CCH Assignment, Numbering Plan
Assignment).
At the end of this section, programming samples are provided for Open and Closed
Numbering Plans.
1.1 Digital Trunk Data Assignment
Use these programming assignments to indicate to the system where (which slot)
the GCD-CCTA blade is located, the signaling format the GCD-CCTA blade uses,
and to assign other information relating to the trunks.
Table 3-1 Digital Trunk Data Programming Assignments
Program/
Description/Selection Assigned Data Comments
Item No.
10-03-01 Blade Setup – Logical Port 0 ~ 400 The GCD-CP10 automatically

Number defines each blade during
installation.
10-03-02 Blade Setup – Frame Type 0 = D4 (default) If 56 K K-CCIS is used. 24

Setup (12 Multiframe) Multiframe (ESF) must be
1 = ESF assigned.
(24 Multiframe)
10-03-03 Blade Setup – Zero Code 0 = B8ZS (default)

Suppression Setup 1 = AMI/ZCS
ZCS_B8ZS
10-03-04 Blade Setup – DTI<-> 0 = 0~ 133 feet (default) Maximum distance back-to-back
CSU Distance Setup 1 = 133 ~ 266 feet T1s can be connected without
2 = 266 ~ 399 feet CSU/DSU service.
3 = 399 ~ 533 feet
4 = 533 ~ 655 feet
10-03-05 Blade Setup – T1 Clock 0 = Internal (default) Define the Master (Internal) or
Source Master/Slave 1 = External Slave (External) clock source.

Issue 5.0
Table 3-1 Digital Trunk Data Programming Assignments (Continued)
Program/
Item No.
10-03-06 Blade Setup – Number Auto

of Ports 4 Ports
8 Ports
12 Ports
16 Ports
20 Ports
10-03-07 Blade Setup – Wiring Auto

Type Cross
Straight
14-05-01 Trunk Group Trunk Group = 1~100 Default priorities for trunks
Assign Trunk to Trunk Default is 1 1 ~ 400 is 1 ~ 400.
Priority = 1 ~ 400
Groups/Outbound Priority
22-02-01 Incoming Call Trunk Setup 0 = Normal (default) Set the feature type for the trunk
1 = VRS (second dial you are programming.
tone if no VRS installed)
2 = DISA
3 = DID
4 = DIL
5 = E&M Tie Line
6 = Delayed VRS
7 = ANI/DNIS
8 = DID(DDI) Mode
Switching
34-01-01 E&M Tie Line Basic Setup 0 = 2nd Dial Tone Set the signaling mode for DID
– DID/E&M Start Signaling 1 = Wink and Tie trunks. DID and Tie
2 = Immediate trunks can use either Immediate
3 = Delay start or Wink start signaling.
Default is 1
3-2 System Data Programming <US Only>

Issue 5.0
1.2 CCIS Assignment
Use these programming assignments to set the availability of CCIS.
Table 3-2 CCIS Programming Assignments
Program/ Description/
Assigned Data Comments
Item No. Selection
50-01-01 CCIS System Setting – 0=Disable Any CCIS settings lose
CCIS Availability 1=Enable functionality if this setting is set
Default is 0 to 0.
50-02-01 Connecting System 0~400 Specify the Trunk port to send

Settings – Port Number of D-channel information. This
Default is 0 program is available for using
Common Signaling DTI package.
Channel (T1)
50-02-02 Connecting System 0=64Kbps If 56K K-CCIS is used (1), 24

Settings – Common 1=56Kbps (Default) Multi-Frame (ESF) must
2=48Kbps(1) be assigned in PRG 10-03-02.
Signaling Channel Data 3=48Kbps(2)
Speed Assignment (T1) Default is 1
50-02-03 Connecting System 0 ~ 16367 Default settings for each CCIS

Settings – Originating Route ID is not assigned.
Point Code This Program is for DTI Only
and must be the same for all
CCIS Route IDs.
Default is 0
50-02-04 Connecting System 0 ~ 16367 DPC must be what

Settings – Destination the OPC is on the
opposite side of the
Point Code (T1) Default is 0 link.
50-02-05 Connecting System 0=Disable Enable to receive name display

Settings – Calling Name 1=Enable across K-CCIS.
Default is 1
Indication (T1)
50-04-01 CCIS Office Code Assign up to four Only used in an open

Assignment – CCIS Office digits. numbering plan network. This
should include the Trunk
Code Access Code and Office Code
Default is No Setting number.
14-14-01 CCIS Trunk CIC 0 = Not Assigned Assign a Circuit Identification

Assignment – Trunk 1~127 = CIC Numbers Code (CIC) number to each
Default is 0 trunk number used for voice
Group Number 1~200 channel.

Issue 5.0
1.3 Numbering Plan Assignment
Use these programming assignments to indicate to the system the number of

digits that are assigned to stations, the number of digits assigned to Access
Codes, and to assign stations to ports.
Table 3-3 Numbering Plan Assignments
Item No. Selection
11-01-01 System Numbering 0 = Not Used1 = Service Default for 1X, 2X, and 3X is 2.
Code
2 = Extension Number
3 = Trunk Access
4 = Special Trunk Access
5 = Operator Access
6 = Flexible Routing
9 = Dial Extension Analyze
11-02-01 Extension Numbering Assign Station Numbers to Defaults for Ports 1 ~ 960:
Port Numbers 101 ~ 199
3101 ~ 3961
11-04-01 Virtual Extension Assign Station Numbers to Defaults for Ports

Numbering Port Number 1 ~ 99 = 201~ 299
100 ~ 512 = Not Assigned
1.4 Programming for Closed Numbering Plan
Use these programs to assign an Access Code to the Intercom Function

Numbers, to assign F-Routes to the appropriate Trunk Routes, and specify the
digits that are added to the dialed number.
Table 3-4 Closed Numbering Plan Assignments
Item No. Selection
11-01-01 System Numbering 2 = Extension Number Default for 1X, 2X, and 3X is 2.
44-02-01 Dial Analysis Table for Up to eight digits can be Assign the digits to be dialed
ARS/F-Route Access – assigned. across the K-CCIS link. These
digits were assigned as F-Route
Dial in Program 11-01-01.
Number of digits to be
analyzed by the system. (Use line key l for “Don’t Care”
Default is No Setting. digit @.)
44-02-02 Dial Analysis Table for 2 = ARS/F-Route Table Service type 2 assigns the digits
ARS/F-Route Access – to be dialed to an F-Route.
Program 44-02-03 assigns the
Service Type F-Route to be used.
Default is 0

Issue 5.0
Table 3-4 Closed Numbering Plan Assignments (Continued)
Item No. Selection
44-02-03 Dial Analysis Table for 2 = 0 ~ 500 When setting data is 2, refer to
ARS/F-Route Access – (0 = No Setting) Program 44-05 for further routing
options.
Additional Data Default is 0
44-05-01 ARS/F-Route Table – 0-100,101-150,255 Select the trunk group to be used

Trunk Group Number 0 = Not Set for the outgoing K-CCIS call.
1~100 = Trunk Group from
PRG14-05
101~150 = Networking
255 = Extension Call
Default is 0
1.5 Programming for Open Numbering Plan
Use these programs to assign the number of digits to Access Code and to make
ARS assignments.
Table 3-5 Open Numbering Plan Assignments
Item No. Selection
11-01-01 System Numbering 2 = Extension Number Default for 1X, 2X, and 3X is 2
44-02-01 Dial Analysis Table for Up to eight digits can be Assign the digits to be dialed
digits were assigned as F-Route
Number of digits to be in Program 11-01-01.
analyzed by the system
Use line key l for “Don’t Care”
Default is No setting digit @.
44-02-02 Dial Analysis Table for 2 = ARS/F-Route Table Service type 2 assigns the digits
ARS/F-Route Access – to be dialed to an F-Route.
Default is 0
44-02-03 Dial Analysis Table for 2 = 0 ~ 500 When setting data is (2), refer to
ARS/F-Route Access – (0 = No Setting) Program 44-05.
Additional Data Default is 0

PRG14-05
Default is 0

Issue 5.0
Table 3-5 Open Numbering Plan Assignments (Continued)
Item No. Selection
44-05-02 ARS/F-Route Table – 0 = No Setting Enter the number of digits to be

Delete Digits 1 ~ 255 = Number of digits deleted from the dialed number.
to delete
(255 = Delete All)
Default is 0
44-05-03 ARS/F-Route Table – 0 = No Setting Enter the table number defined in

Additional Dial Number 1 ~ 1000 = Add Table in Program 44-06 for additional
Program 44-06-01 digits to be dialed.
Table Default is 0
44-05-04 ARS/F-Route Table – 0 = Off Select whether or not a beep is

Beep tone 1 = On heard if a lower priority trunk
Default is 0 group is used to dial out.
44--05-05 ARS/F-Route Table – 0 = No Setting Select the gain table number to

Gain Table Number for 1 ~ 500 be used for the internal call
Default is 0 defined in Program 44-07.
Internal Calls
44-05-06 ARS/F-Route Table – 0 = No Setting Select the gain table number to

Gain Table Number for 1 ~ 500 be used for the internal call
Default is 0 defined in Program 44-07.
Tandem Connections
44-05-07 ARS/F-Route Table – 0 = No Setting Select the ARS Class of Service

ARS Class of Service 1 ~ 16 to be used for the table. An
extension ARS COS is
Default is 0 determined in Program 26-04-01.
44-05-08 ARS/F-Route Table – 0 = No Setting If a Dial Treatment is selected,

Dial Treatment 1 ~ 15 Programs 44-05-02 and
44-05-03 are ignored and the
Dial Treatment defined in
Default is 0 Program 26-03-01 is used
instead.
44-05-09 ARS/F-Route Table – 0 = No Maximum Assign the Maximum Digits for K-

Maximum Digit 1 ~ 24 CCIS or ISDN Calls.
Default is 0
Assignment

Issue 5.0
1.6 Closed Number Programming Example
This section provides the steps needed to program a closed numbering plan.
Step 1: T1 Tie Lines
The following diagram is an example of Programs that should be assigned for T1

Tie lines. The example assumes that the SV9100 system is defaulted with the
GCD-CCTA card installed.
Abbreviations used in the diagram:
MB = Memory Block TRK = Trunk TG = Trunk Group
CC = Clear Channel TT = Trunk Type
PRG 10-03-03 PRG 10-03-03 PRG 10-03-03

CC = B8ZS CC = B8ZS CC = B8ZS
PRG 10-03-05 PRG 10-03-05 PRG 10-03-05

External
Internal External
Internal
PRG 22-02-01 PRG 22-02-01 PRG 22-02-01
TT = E&M TT = E&M TT = E&M
PRG 14-05-01 PRG 14-05-01 PRG 14-05-01

TRK 001~024 = TG 10 TRK 001~024 = TG 10 TRK 001~024 = TG 10
TRG 025~048 = TG 11
PRG 14-01-13 PRG 14-01-13 PRG 14-01-13

TRK 001~024 = Yes TRK 001~048 = Yes TRK 001~024 = Yes
PRG 34-01-01 PRG 34-01-01 PRG 34-01-01

TRK 001~024 = Wink TRK 001~048 = Wink TRK 001~024 = Wink
PRG 11-01-01 PRG 11-01-01 PRG 11-01-01

1X = 3 digit; Intercom 1X = 3 digit; F-Route 1X = 3 digit; F-Route
2X = 3 digit; F-Route 2X = 3 digit; Intercom 2X = 3 digit; F-Route
3X = 3 digit; F-Route 3X = 3 digit; F-Route 3X = 3 digit; Intercom
PRG 11-02-01 PRG 11-02-01 PRG 11-02-01

Extensions = 1XX Extensions = 2XX Extensions = 3XX
PRG 11-04-01 PRG 11-04-01 PRG 11-04-01

Unassign Virtuals = 2XX Unassign Virtuals = 2XX Unassign Virtuals = 2XX

Issue 5.0
Step 2: Closed Numbering Plan
The following diagram provides an example of Programs that should be assigned

for Closed Numbering. The example assumes that Step 1: T1 Tie Lines was
completed.
AC = Access Code BLK = Closed Numbering Block MB = Memory Block
ACG = Access Item Code TT = Trunk Type
Calling Calling Calling

200~230 100~130 100~130
300~330 300~330 200~230
PRG 11-01-01 PRG 11-01-01 PRG 11-01-01
Dial 1X = 3 digit; Intercom Dial 1X = 3 digit; F-Route Dial 1X = 3 digit; F-Route
Dial 2X = 3 digit; F-Route Dial 2X = 3 digit; Intercom Dial 2X = 3 digit; F-Route
Dial 3X = 3 digit; F-Route Dial 3X = 3 digit; F-Route Dial 3X = 3 digit; Intercom
PRG 44-02-01 PRG 44-02-01 PRG 44-02-01
TBL 1 = 2 TBL 1 = 1 TBL 1 = 1
TBL 2 = 3 TBL 2 = 3 TBL 2 = 2
PRG 44-02-02 PRG 44-02-02 PRG 44-02-02
TBL 1 = F-Route TBL 1 = F-Route TBL 1 = F-Route
TBL 2 = F-Route TBL 2 = F-Route TBL 2 = F-Route
PRG 44-02-03 PRG 44-02-03 PRG 44-02-03
TBL 1 = F-Route 1 TBL 1 = F-Route 1 TBL 1 = F-Route 1
TBL 2 = F-Route 1 TBL 2 = F-Route 2 TBL 2 = F-Route 1
PRG 44-05-01 PRG 44-05-01 PRG 44-05-01
F-Route TBL 1 = TG 10 F-Route TBL 1 = TG 10 F-Route TBL 1 = TG 10
F-Route TBL 2 = TG 11
PRG 44-05-02 PRG 44-05-02 PRG 44-05-02
F-Route TBL 1 Delete Digit = 0 F-Route TBL 1 Delete Digit = 0 F-Route TBL 1 Delete Digit = 0
F-Route TBL 2 Delete Digit = 0
PRG 44-05-03 PRG 44-05-03 PRG 44-05-03
F-Route TBL 1 Add Dial = 0 F-Route TBL 1 Add Dial = 0 F-Route TBL 1 Add Dial = 0
F-Route TBL 2 Add Dial = 0
PRG 44-05-09 PRG 44-05-09 PRG 44-05-09
F-Route TBL 1 Max. Digit = 3 F-Route TBL 1 Max. Digit = 3 F-Route TBL 1 Max. Digit = 3
F-Route TBL 2 Max. Digit = 3
 Before moving on to the next step, test the T1 Tie lines and the Closed Numbering Plan.

Issue 5.0
Step 3: K-CCIS Activation
The following diagram provides an example of Programs that should be assigned

for K-CCIS. The example assumes that Step 1: T1 Tie Lines and Step 2: Closing
Number Plan are completed.
DSTCCH = Destination Point Code CCH = Control Channel Handler TRK = Trunk
ORGCCH = Originating Point Code TG = Trunk Group
UNIVERGE UNIVERGE Electra Elite

SV9100 SV9100 IPK II
TG 10 TG 10 TG 11 TG 10
TG 00 TG 00 TG 00 TG 00
CCH CCH
(101s) (201s) (301s)
PC00001 PC00002 PC00003
101~130 201~230 301~330
PRG 14-13-01 PRG 14-13-01 PRG 14-13-01

TG10 = CCIS Route ID 1 TG10 = CCIS Route ID 1 TG10 = CCIS Route ID 1
TG11 = CCIS Route ID 2
PRG 14-14-01 PRG 14-14-01 PRG 14-14-01

TRK 001~023 = 1~23 TRK 001~023 = 1 ~ 23 TRK 001~023 = 1~23
TRK 025~047 = 1 ~ 23
PRG 14-05-01 PRG 14-05-01 PRG 14-05-01

TRK 001~024 = 10 TRK 001~024 = 10 TRK 001~024 = 10
Priority = TRK 001~024 = 024~001 Priority = TRK 001~024 = 024~001 Priority = TRK 001~024 = 024~001
TRK 025~048 = 11
TRK 048 = 00
Priority = TRK 025~048 = 048~025
PRG 22-02-01 PRG 22-02-01 PRG 22-02-01

TRK 001~024 = Tie TRK 001~024 = Tie TRK 001~024 = Tie
TRK 025~048 = Tie
PRG 50-01-01 PRG 50-01-01 PRG 50-01-01

CCIS Availability = Yes CCIS Availability = Yes CCIS Availability = Yes
PRG 50-02-01 PRG 50-02-01

CCIS Route ID1 = TRK 024 CCIS Route ID1 = TRK 024
CCIS Route ID2 = TRK 048
PRG 50-02-02 PRG 50-02-02 PRG 50-02-02

CCIS Route ID1 = 56K CCIS Route ID1 = 56K CCIS Route ID1 = 56K
CCIS Route ID2 = 56K
PRG 50-02-03 PRG 50-02-03 PRG 50-02-03

CCIS Route ID1 = Org. Point Code 1 CCIS Route ID1 = Org. Point Code 2 CCIS Route ID1 = Org. Point Code 3
CCIS Route ID2 = Org. Point Code 2

Issue 5.0
PRG 50-02-04 PRG 50-02-04 PRG 50-02-04

CCIS Route ID1 = Dest. Point Code 2 CCIS Route ID1 = Dest. Point Code 1 CCIS Route ID1 = Dest. Point Code 2
CCIS Route ID2 = Dest. Point Code 3
PRG 50-02-06
CCIS Route ID1 = CCH1
PRG 50-03-01 PRG 50-03-01 PRG 50-03-01

N/A System ID1 = 00001 N/A
System ID2 = 00003
PRG 50-04-01 PRG 50-04-01 PRG 50-04-01

Vacant Vacant Vacant

Issue 5.0
1.7 Open Number Programming Example
This sections provides the steps needed to program an open numbering plan.
Step 1: T1 Tie Lines
The following diagram provides an example of Program and item numbers that
should be assigned for T1 Tie lines. The example assumes that each UNIVERGE
SV9100 and Electra Elite IPK II system is defaulted with the GCD-CCTA blades
installed.
CC = Clear Channel TT = Trunk Type

Tie Lines Tie Lines

25 26 27
TG 10 TG 10 TG 11 TG 10
(101s) (201s) (301s)
101~130 101~130
101~130
PRG 10-03-05 PRG 10-03-05 PRG 10-03-05

External
Internal External
Internal
PRG 10-03-03 PRG 10-03-03 PRG 10-03-03

CC = B8ZS CC = B8ZS CC = B8ZS
PRG 22-02-01 PRG 22-02-01 PRG 22-02-01

TRK 025~048 = Tie
PRG 14-05-01 PRG 14-05-01 PRG 14-05-01

TRK 001~024 = TG 10 TRK 001~024 = TG 10 TRK 001~024 = TG 10
TRK 025~048 = TG 11
PRG 14-02-13 PRG 14-02-13 PRG 14-02-13

TRK 001~024 = Yes TRK 001~024 = Yes TRK 001~024 = Yes
TRK 025~048 = Yes
PRG 22-02-01 PRG 22-02-01 PRG 22-02-01

TRK 024~048 = Tie
PRG 34-01-01 PRG 34-01-01 PRG 34-01-01

TRK 001~024 = Wink TRK 001~024 = Wink TRK 001~024 = Wink
TRK 025~048 = Wink

Issue 5.0
Step 2: Open Numbering Plan
The following diagram provides an example of Programs and item numbers that
should be assigned for Open Numbering. The example assumes that Step 1: T1
Tie Lines was completed.
AC = Access Code ACG = Access Item Code MB = Memory Block
RT = Route TB = ARS Table PRG =Program

SV9100
25 SV9100 IPK II
Tie Lines Tie Lines
25 26 27
TG 10 TG 10 TG 11 TG 10
(101s) (101s) (101s)
101~130 101~130 101~130
Calling Calling Calling

101 ~ 130 101 ~ 130 101 ~ 130
PRG 11-01-01 PRG 11-01-01 PRG 11-01-01

Dial 1X = 3 Digits; Intercom Dial 1X = 3 Digits; Intercom Dial 1X = 3 Digits; Intercom
Dial 8X = 1 Digit, F-Route Access Dial 8X = 1 Digit, F-Route Access Dial 8X = 1 Digit, F-Route
PRG 11-02-01 PRG 11-02-01 PRG 11-02-01

Port 1 = 101 Port 1 = 101 Port 1 = 101
PRG 44-02-01 PRG 44-02-01 PRG 44-02-01

F-Route TBL1 = 825 TBL1 = 825 F-Route TBL1 = 825
PRG 44-02-02 PRG 44-02-02 PRG 44-02-02

TBL1 = F-Route TBL1 = F-Route TBL1 = F-Route
PRG 44-02-03 PRG 44-02-03 PRG 44-02-03
TBL1 = F-Route TBL1 TBL1 = F-Route 1 TBL1 = F-Route 1
PRG 44-05-01 PRG 44-05-01 PRG 44-05-01

TBL1 = TG 255 TBL1 = TG 10 TBL1 = TG 10
TBL2 = TG 10 TBL2 = TG 255 TBL2 = TG 10
TBL3 = TG 11 TBL3 = TG 255

Issue 5.0
PRG 44-05-02 PRG 44-05-02 PRG 44-05-02

F-Route TBL1; Delete Digit = 3 F-Route TBL2: Delete Digit = 3 F-Route TBL2; Delete Digit = 3
PRG 44-05-09 PRG 44-05-09 PRG 44-05-09

F-Route TBL2; Max Digit = 6 F-Route TBL1: Max Digit = 6 F-Route TBL2; Max Digit = 6
F-Route TBL3: Max Digit = 6
 Before moving to the next step, test the T1 Tie lines and the Open Numbering Plan.
Step 3: K-CCIS Activation
are assigned for K-CCIS. The example assumes that Step 1: T1 Tie Lines and
Step 2: Open Number Plan are completed.
DSTCCH = Destination Point Code CCH = Control Channel Handler TRK = Trunk
ORGCCH = Originating Point Code TG = Trunk Group PRG = Program

TG 10 TG 10 TG 11 TG 10
TG 10 TG 10 TG 11 TG 10
CCH CCH
(101s) (201s) (301s)
PC00001 PC00002 PC00003
101~130 201~230 301~330
PRG 14-13-01 PRG 14-13-01 PRG 14-13-01

TG10 = CCIS Route ID 1 TG10 = CCIS Route ID 1 TG10 = CCIS Route ID 1
TG11 = CCIS Route ID 2
PRG 14-14-01 PRG 14-14-01 PRG 14-14-01

TRK 001~023 = 1~23 TRK 001~023 = 1 ~ 23 TRK 001~023 = 1~23
TRK 025~047 = 1 ~ 23
PRG 14-05-01 PRG 14-05-01 PRG 14-05-01

TRK 001~024 = 10 TRK 001~023 = 10 TRK 001~024 = 10
TRK 024 = 00 TRK 024 = 00 TRK 024 = 00
Priority = TRK 001~024 = 024~001 Priority = TRK 001~024 = 024~001 Priority = TRK 001~024 = 024~001
TRK 025~048 = 11
TRK 048 = 00
Priority = TRK 025~048 = 048~025
PRG 22-02-01 PRG 22-02-01 PRG 22-02-01

TRK 025~048 = Tie
PRG 50-01-01 PRG 50-01-01 PRG 50-01-01

CCIS Availability = Yes CCIS Availability = Yes CCIS Availability = Yes
PRG 50-02-01 PRG 50-02-01

CCIS Route ID1 = TRK 024 CCIS Route ID1 = TRK 024
CCIS Route ID2 = TRK 048

Issue 5.0
PRG 50-02-02 PRG 50-02-02 PRG 50-02-02

CCIS Route ID1 = 56K CCIS Route ID1 = 56K CCIS Route ID1 = 56K
CCIS Route ID2 = 56K
PRG 50-02-03 PRG 50-02-03 PRG 50-02-03

CCIS Route ID1 = Org. Point Code 1 CCIS Route ID1 = Org. Point Code 2 CCIS Route ID1 = Org. Point Code 3
CCIS Route ID2 = Org. Point Code 2
PRG 50-02-04 PRG 50-02-04 PRG 50-02-04

CCIS Route ID1 = Dest. Point Code 2 CCIS Route ID1 = Dest. Point Code 1 CCIS Route ID1 = Dest. Point Code 2
CCIS Route ID2 = Dest. Point Code 3
PRG 50-02-06
PRG 50-03-01 PRG 50-03-01 PRG 50-03-01

N/A System ID1 = 00001 N/A
System ID2 = 00003
PRG 50-04-01 PRG 50-04-01 PRG 50-04-01

825 826 827

Issue 5.0
1.8 Dedicated Tandem CO Trunk Calls
The following diagram provides an example of SV9100 Programs that should be

assigned when all Local and Long Distance CO calls, from the REMOTE site, are
routed through the MAIN site using Flexible Routing (F-Routes) and Automatic
Route Selection (ARS).
The example assumes that the UNIVERGE SV9100 systems are connected via
K-CCIS and using Closed Numbering Plan. All local calls are 10-digit dial and all
1+ calls are 11-digit dial. Only one CO Trunk is set for 911 in the REMOTE
system.
RT = ARS Route DN = Dial Number PRG = Program
PRG 11-01-01 PRG 11-01-01

Dial 8 = 1 Digit: 2nd Trunk Access Dial 9 = 1 Digit; Trunk Access
Dial 9 = 1 Digit; Trunk Access Dial 3 = 3 Digit; Intercom
Dial 3 = 1 Digit; F-Route Dial 0 = 1 Digit; Operator Access
Dial 0 = 1 Digit; F-Route
PRG 11-09-01
Trunk Access Code = 9
PRG 11-09-02
2nd Trunk Access Code = 8
PRG 26-01-01
ARS Service = On
PRG 26-02-01
ARS TBL1 = @
PRG 26-02-02
ARS TBL1 = F-Route
PRG 26-02-03
ARS TBL1 = F-Route TBL 1
PRG 44-02-01
Analysis TBL1 = 0
Analysis TBL2 = 3

Issue 5.0
PRG 44-02-02
Analysis TBL1 = F-Route
Analysis TBL2 = F-Route
PRG 44-02-03
Analysis TBL1 = Data F-Route 2
Analysis TBL2 = Data F-Route 3
PRG 44-05-01
F-Route TBL1 = TG 10
PRG 44-05-08
F-Route TBL1 = ARS Treatment TBL1
PRG 44-05-09
F-Route TBL1 = Max Digit 10
PRG 26-03-01
ARS Treatment TBL1 = D019RE

Issue 5.0
1.9 Shared Tandem CO Trunk Calls
should be assigned when two sites share CO lines for reducing Long Distance
calls using Automatic Route Selection (ARS).
The example assumes that the UNIVERGE SV9100 systems are connected via
K-CCIS and using Closed Numbering Plan. All local calls are 10-digit dial and all
1+ calls are 11-digit dial.

TB = ARS Table DN = Dial Number RT = ARS Route
UNIVERGE
SV9100 UNIVERGE
Remote SV9100
Main
CO Trunks CO Trunks
K-CCIS Link
TG10 TG10
Area Code 817 Area Code 214
Area Code 940 Area Code 972
101s 301s
PRG 11-01-01 PRG 11-01-01

Dial 9 = 1 digit: Trunk Access Dial 9 = 1 digit: Trunk Access
PRG 11-09-01 PRG 11-09-01

Trunk Access Code = 9 Trunk Access Code = 9
PRG 11-09-02 PRG 11-09-02

nd
2 Trunk Access = 8 2nd Trunk Access = 8
PRG 26-01-01 PRG 26-01-01

ARS Service = On ARS Service = On
PRG 26-02-01 PRG 26-02-01

ARS Analysis TBL1 = 1214 ARS Analysis TBL1 = 1817
PRG 26-02-02 PRG 26-02-02

ARS Analysis TBL1 = F-Route ARS Analysis TBL1 = F-Route
ARS Analysis TBL2 = F-Route ARS Analysis TBL2 = F-Route
ARS Analysis TBL3 = TRG (Trunk Group) ARS Analysis TBL3 = TRG (Trunk Group)

Issue 5.0
PRG 26-02-03 PRG 26-02-03

ARS Analysis TBL1 = F-Route TBL1 ARS Analysis TBL1 = F-Route TBL1
ARS Analysis TBL2 = F-Route TBL1 ARS Analysis TBL2 = F-Route TBL1
ARS Analysis TBL3 = Add Data TG 01 ARS Analysis TBL3 = Add Data TG 01
PRG 44-05-01 PRG 44-05-01

F-Route TBL1 = TG 10 F-Route TBL1 = TG 10
PRG 44-05-08 PRG 44-05-08

F-Route TBL1 = ARS Edit TBL1 F-Route TBL1 = ARS Edit TBL1
PRG 44-05-09 PRG 44-05-09

F-Route TBL1 = Max Digit 11 F-Route TBL1 = Max Digit 11
PRG 26-03-01 PRG 26-03-01

ARS treatment TBL1 = D019RE ARS treatment TBL1 = D019RE

UNIVERGE® SV9100
Features and Specifications Chapter 4
SECTION 1 GENERAL INFORMATION
Key-Common Channel Interoffice Signaling (K-CCIS) allows multiple systems to be

connected together to provide additional feature compatibility, above what normal Tie
Lines provide. The system is configured with the 24-channel GCD-CCTA (T-1/CCH)
for receiving or transmitting common signaling data from/to a distant office. The
system can provide a variety of interoffice service features such as Calling Name
display, Centralized Voice Mail Integration, or Link Reconnect.
The following features are provided:

 Automatic Recall – K-CCIS on page 4-6
 Brokerage Hotline – K-CCIS on page 4-8
 Call Forwarding – All Calls – K-CCIS on page 4-10
 Call Forwarding – Busy/No Answer – K-CCIS on page 4-14
 Call Park Retrieve – K-CCIS on page 4-28
 Call Transfer – All Calls – K-CCIS on page 4-34
 Calling Name Display – K-CCIS on page 4-37
 Calling Number Display – K-CCIS on page 4-39
 Calling Party Number (CPN) Presentation from Station – K-CCIS on page 4-41
 Centralized Billing – K-CCIS on page 4-43
 Centralized BLF (K-CCIS) on page 4-47
 Centralized Day/Night Mode Change – K-CCIS on page 4-52
 Centralized E911 – K-CCIS on page 4-57
 Dial Access to Attendant – K-CCIS on page 4-60
 Direct Inward Dialing – K-CCIS on page 4-65
 Dual Hold – K-CCIS on page 4-67
 Elapsed Time Display – K-CCIS on page 4-69
 Flexible Numbering of Stations – K-CCIS on page 4-71
 Handsfree Answerback – K-CCIS on page 4-74

Issue 5.0
 Hot Line – K-CCIS on page 4-76

 Multiple Call Forwarding – All Calls – K-CCIS on page 4-82
 Multiple Call Forwarding – Busy/No Answer - K-CCIS on page 4-87
 Paging Access – K-CCIS on page 4-92
 Quick Transfer to Voice Mail – K-CCIS on page 4-100
 Station-to-Station Calling – K-CCIS on page 4-103
 Uniform Numbering Plan – K-CCIS on page 4-105
 Voice Call – K -CCIS on page 4-107
 Voice Mail Integration – K-CCIS on page 4-109
SECTION 2 SYSTEM AVAILABILITY
2.1 Terminal Type
All stations
2.2 Required Components
GCD-CCTA
- OR -
GPZ-IPLE
2.3 Operating Procedures
Normal call handling procedures apply.
The following table shows the chassis system software compatibility with GCD-CCTA
firmware.
Chassis Software GCD-CCTA
SV9100 V1.00 X
X = Compatible
– = Not compatible
4-2 Features and Specifications

Issue 5.0
SECTION 3 SERVICE CONDITIONS
General:
 Each UNIVERGE SV9100 system can have up to eight K-CCIS routes.
 One GCD-CCTA is required to support each K-CCIS link. A maximum of eight
K-CCIS links are supported.
 The K-CCIS feature shares the CO/PBX/Tie/DID trunks available for the
system.
 When assigning a Closed Numbering Plan and DID conversion across K-CCIS
is required, the UNIVERGE SV9100 uses the ARS/F-Route Tables.
 The UNIVERGE SV9100 uses the F-Route Tables to assign an Open
Numbering Plan.
 When all K-CCIS voice channels are busy, the UNIVERGE SV9100 originator
of a K-CCIS call hears a busy tone from the system.
 Outgoing CO calls in a K-CCIS network can be routed over the K-CCIS link and
use the distant system CO lines.
 Distant system extension numbers in the K-CCIS network can be assigned to
One Touch keys and Speed Dial buffers.
 When a K-CCIS trunk is on hold, the Specified Line Seizure access codes can
be used to retrieve the call from its held state.
Restrictions:
 The UNIVERGE SV9100 can support only 1~8-digit station numbers.
 Station Numbers are assigned by the 10s group for 4-digit station numbers,
100s group for 5-digit station numbers, 1000s group for 6-digit station numbers,
10000s group for 7-digit station numbers.
 When Voice Mail Message Waiting status must be sent across the K-CCIS to a
remote system, F-Routes must be used.
 For a Closed Numbering Plan network using F-Routes, a maximum of 120
F-Route Tables are available allowing a maximum of 121 connected systems
per K-CCIS network.
 When a Closed Numbering Plan Network is used, a user can call another
station by dialing the distant extension number, but extensions in the network
cannot have the same prefix.
 For an Open Numbering Plan network, a user can dial another station by
dialing the office location number plus an extension number and the extension
number can have the same prefix, but the office location cannot be the same.
 When an UNIVERGE SV9100 system is a tandem system (in the middle)
between systems with higher K-CCIS feature support (including NEAX PBXs),
only the K-CCIS features supported by the UNIVERGE SV9100 tandem system

Issue 5.0
are passed through and supported.

 An UNIVERGE SV9100 K-CCIS network should never have more than five
hops (tandem connections) because of the message delay through each
tandem system.
 A Star topology network supports up to eight systems.
 A Tree topology network is supported. The maximum number of systems
depends on the Numbering Plan used and the maximum number of hops
(tandem connections).
 A Mesh topology network is not supported.
 K-CCIS requires assigning a point code for each office. Point codes
differentiate between an originating office and a destination office in the K-CCIS
Network. Assigning point codes requires the following considerations:
 The point code must be unique in the network.
 When a system has two or more CCH channels, the same originating point code
must be assigned to all channels in the system.
 The UNIVERGE SV9100 can have a maximum of 255 codes assigned to distant
systems.
 Using an UNIVERGE SV9100-to-UNIVERGE SV9100 network, centralized
voice mail is not supported when an Open Numbering Plan is used.
 Centralized E911 – K-CCIS is supported.
 When Centralized E911 – K-CCIS is not used, each UNIVERGE SV9100
system in a K-CCIS network must have at least one trunk for Emergency 911
calls.
 Using a NEAX-to-UNIVERGE SV9100 network, the PBX must supply
centralized voice mail.
 Multiline terminals must have an available Call Appearance (CAP) key to
originate or answer a K-CCIS trunk call.
 Direct access of K-CCIS voice or data channels using Line keys or Specified
Line Seizure access codes is prohibited.
 The Recall key or Drop key does not function on K-CCIS calls. When either
key is pressed, operation is ignored, and the call continues.
 Trunk queuing is prohibited on a K-CCIS trunk route.
 The ability to route an incoming DID call directly across a K-CCIS link (Direct
Inward Dialing – K-CCIS) is supported only when a Closed Numbering Plan
using F-Routes is used.
 This feature is not supported by the GCD-4ODTA Analog Line interface.
 Eight GCD-CCTA can be assigned per system.
 Extension numbers cannot start with 0 or 9.

Issue 5.0
 Internal Calls, transferred calls, and K-CCIS calls do not provide Caller ID to
single line telephones.
 Caller ID Call Return feature is not supported with K-CCIS calls.
SECTION 4 RELATED FEATURE LIST
 T1 Connections
 Universal Slots
SECTION 5 K-CCIS FEATURES
The remainder of this chapter provides detailed information for the available K-CCIS
features.

Issue 5.0
Automatic Recall – K-CCIS
FEATURE DESCRIPTION
This feature allows a call to be release transferred to another station in another office
in the K-CCIS network and recall back to the originator of the transfer after a
programmed time.
SYSTEM AVAILABILITY
All Terminals
Required Components:
GCD-CCTA
- OR -
GPZ-IPLE
OPERATING PROCEDURES
Using a multiline terminal with a call in progress (Closed Numbering Plan):
1. Press Transfer. Internal dial tone is heard. The call is placed on Non-Exclusive Hold.
2. Dial the distant K-CCIS station number where the call is to be transferred.
3. Wait for the ringback tone.
4. Hang up.
- OR -
1. When the party answers, announce the transfer.

2. Restore the handset (transfer is completed).
Using a multiline terminal with a call in progress (Open Numbering Plan):
1. Press Transfer, and receive internal dial tone. The call is placed on Non-Exclusive
Hold.
2. Dial the trunk Access Code (normally 8).
3. Dial the Office Code number.

Issue 5.0
6. Hang up.
SERVICE CONDITIONS
 If PRG 34-07-05 is left at default (30) the transferred call recalls to the station
that performed the transfer when not answered.
 A UNIVERGE SV9100 station can receive a K-CCIS transferred call as a
camp-on call if allowed by Class of Service.
Restrictions:
 PRG 34-07-05 cannot be set based on a Timer Class of Service in PRG 20-31.
 Trunk-to-Trunk Transfer must be allowed in Program 14-01-13 (Trunk-to-Trunk
Transfer Yes/No Selection).
 A blind transfer across a K-CCIS link cannot be completed until ringback tone is
received at the transferring station.
RELATED FEATURE LIST

 Link Reconnect – K-CCIS
 Station-to-Station Calling – K-CCIS
 Uniform Numbering Plan – K-CCIS
 Call Transfer – All Calls – K-CCIS
GUIDE TO FEATURE PROGRAMMING
Program/
Item No.
34-07-05 E&M Tie Line Timer – Trunk 0 ~ 64800 seconds Determine the amount of time the call should
Answer Detect Timer for E&M ring a station in another office before recalling
Default is 30 back to the originator of the transfer.

Issue 5.0
Brokerage Hotline – K-CCIS
FEATURE DESCRIPTION
This feature provides a ringdown connection between two stations, each using a
multiline terminal, in different offices in the CCIS network.
SYSTEM AVAILABILITY
All Terminals
GCD-CCTA
- OR -
GPZ-IPLE
To use this feature at any terminal:
1. Lift the handset or press Speaker.

2. Press the line/feature key associated with the pre-assigned station.
The destination station is automatically dialed, ring back tone is heard and the
destination station answers.
3. After completion of conversation, hang up or press Speaker
To make another Brokerage-Hot Line-CCIS call immediately, press another line/

feature key without going on hook and off hook.
SERVICE CONDITIONS
 Either multiline terminal in a Brokerage - Hot Line – (K-CCIS) pair may transfer
a Hot Line call to another station in the K-CCIS network using the Call Transfer
– All Calls - (K-CCIS) feature.
Restrictions:
None

Issue 5.0

 Call Transfer – All Calls - K-CCIS
Program/
Item No.
15-07-01 Programmable Function Keys 01 = DSS / One-Touch Assign a DSS/One Touch key of the station in
the Distant Office.

Issue 5.0
Call Forwarding – All Calls – K-CCIS
FEATURE DESCRIPTION
This feature allows all calls destined for a particular station to be routed to another
station or to an Attendant, in another office in the K-CCIS network, regardless of the
status (busy or idle) of the called station. The activation and cancellation of this
feature may be accomplished by either the station user or an Attendant position if
allowed by Class of Service (COS). Attendant Positions can be used to cancel Call
Forward – All Call system-wide.
For more details, refer to the UNIVERGE SV9100 Features and Specifications
Manual.
SYSTEM AVAILABILITY
Terminal Type:
All multiline terminals
Required Components
GCD-CCTA
- OR -
GPZ-IPLE
To set Call Forward – All Calls – K-CCIS from a multiline telephone (Closed
Numbering plan):
1. Press the Call Forward – All ON/OFF key.

2. Dial 1 to set, then enter the remote K-CCIS station number.
3. Press Speaker.
- OR -

2. Dial Access Code 848 (default), then 1 to set.
3. Dial the remote K-CCIS station number.
4. Restore handset or press Speaker.

Issue 5.0
To set Call Forward – All Calls – K-CCIS from a Multiline Telephone (Open
Numbering Plan):

2. Dial 1 to set.
5. Dial the distant K-CCIS station number.
- OR -

3. Dial the trunk Access Code (normally 9 ).
To cancel Call Forward – All Calls – K-CCIS from a Multiline Telephone:
1. Press Call Forward – All ON/OFF key.

2. Dial 0 to cancel.
3. Press Speaker.
- OR -

2. Dial Access Code 848 (default), then dial 0.
3. Restore the handset or press Speaker.
SERVICE CONDITIONS
General:
 Any station or Call Arrival (CAR) key can be set for Call Forwarding – All
Calls – K-CCIS.
Restrictions:
 Call Forward – Off-Premise must be allowed in PRG 20-11-12 (Class of Service
External Call Forward) to set call forwarding to a remote K-CCIS station

Issue 5.0
number.
 Trunk-to-Trunk Transfer must be allowed in PRG 14-01-13 (Trunk-to-Trunk
 A Single Line Telephone user can transfer a trunk call to another internal station
that is set for Call Forwarding – All Calls – K-CCIS, however, when the distant
party answers the call, a conference cannot be established.
 The destination station in the distant system is the only station that can call a
station with Call Forwarding – All Calls – K-CCIS set.
 Call Forwarding with Both Ringing (All Calls) is not supported.
 Call Forward Split Internal/External is not supported.
 Forwarding to Voice Mail is not Included in the Maximum Hop Count.
 Call Forward continues to operate to a DT800/DT700 that has been removed.

 Call Forwarding – Busy/No Answer – K-CCIS
 Multiple Call Forwarding – All Calls – K-CCIS
 Multiple Call Forwarding – Busy/No Answer – K-CCIS
This guide provides a list of associated Programs that support this feature.
Item No. Selection
20-11-12 Class of Service Options 0 = Off To enable per class of service.
(Hold Transfer Service) – 1 = On
Call Forwarding
Off-Premise (External Call
Forwarding Default is 0
20-06-01 Class of Service for 0~15 All Extensions are in Class 1. (default)
Extensions
20-09-07 Class of Service Options 0 = Off Must be Off for Call Forward – Busy to operate.
(Incoming Call Service) – 1 = On
Call Queuing Default is 0

Issue 5.0
Item No. Selection
(Supplementary Service) – 1 = On
Automatic Off Hook
Signaling (Automatic
Override) Default is 0
15-07-01 Programmable Function 10 = Call Forward – Service Codes:

Keys Immediate 848
11 = Call Forward – Busy
12 = Call Forward – No 843
Answer
13 = Call Forward – Busy 845
No Answer
844
14-01-13 Basic Trunk Data Setup – 0 = Disable Must be enabled for Trunk-to-Trunk Transfer, Call
Trunk-to-Trunk Transfer 1 = Enable Forward – Off-Premise, and tandem trunking.
Loop Supervision Default is 0

50-06-01 CCIS Feature Availability – 0 = Not Available To enable or disable Link Reconnect.
Link Reconnect 1 = Available
Default is 1
50-05-01 CCIS Maximum Call 0~7 Hops Sets Maximum Hops allowed in a CCIS network.
Forwarding Hop Counter Default is 5

Issue 5.0
Call Forwarding – Busy/No Answer – K-CCIS
FEATURE DESCRIPTION
This feature permits a call to a Busy or unanswered station to be forwarded to

another station or an Attendant, in another office in the K-CCIS network. The
activation and cancellation of this feature may be accomplished by either the station
user or an Attendant position, if allowed by Class of Service (COS).
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
To set Call Forward – Busy/No Answer – K-CCIS from a Multiline Telephone

(Closed Numbering Plan):
1. Press the Call Forward – Busy/No Answer ON/OFF key.
2. Dial 1 to set.
4. Press Speaker.
- OR -


Issue 5.0
To set Call Forward – Busy/No Answer - K-CCIS from a Multiline Telephone

(Open Numbering Plan):
2. Dial 1 to set.
6. Press Speaker.
- OR -

2. Dial Access Code 844 (default), then Dial 1 to set.
To cancel Call Forward – Busy/No Answer – K-CCIS from a Multiline Telephone:

1. Press Call Forward – Busy/No Answer On/Off key.
3. Press Speaker.
- OR -

2. Dial Access Code 844 (default), then Dial 0 to cancel.
SERVICE CONDITIONS
General:
 Any station or Call Arrival (CAR) key can be set for Call Forwarding – Busy/No
Answer – K-CCIS.
Restrictions:
 Call Forward – Off-Premise must be allowed in Class of Service (PRG
20-11-12) External Call forward to set call forwarding to a remote K-CCIS
station number.

Issue 5.0
 Trunk-to-Trunk Transfer must be allowed in PRG 14-01-13 for each trunk

(Trunk-to-Trunk Transfer Yes/No Selection).
 A Single Line Telephone user can transfer a trunk call to another internal station
that is set for Call Forwarding – All Calls - K-CCIS, however, when the distant
party answers the call, a conference cannot be established.
 The destination station in the distant systems is the only station that can call a
station with Call Forwarding – All Calls – K-CCIS set.
 Call Forwarding with Both Ringing (All Calls) is not supported.
 Call Forward Split Internal/External is not supported.
 Forwarding to Voice Mail is not Included in the Maximum Hop Count.
 Call Forward continues to operate to a Dterm that has been removed.

 Call Forwarding – Busy/No Answer – K-CCIS
 Multiple Call Forwarding – All Calls – K-CCIS
 Multiple Call Forwarding – Busy/No Answer – K-CCIS
 Link Reconnect – K-CCIS
 Uniform Numbering Plan – K-CCIS
Guide to Feature Programming
Item No. Selection
20-11-12 Class of Service Options 0 = Off To enable per class of service.
(Hold Transfer Service) – 1 = On
Call Forwarding
Off-Premise (External Call
Extensions
Automatic Off Hook

Issue 5.0
Item No. Selection
Keys Immediate 848
11 = Call Forward – Busy
12 = Call Forward – No 843
Answer
No Answer
844
14-01-13 Basic Trunk Data Setup – 0 = Disable Must be enabled for Trunk-to-Trunk Transfer, Call
Trunk-to-Trunk Transfer 1 = Enable Forward – Off-Premise, and tandem trunking.

50-06-01 CCIS Feature Availability – 0 = Not Available Enable/Disable Link Reconnect.
Default is 1
50-05-01 CCIS Maximum Call 0~7 Hops Set Maximum Hops allowed in a CCIS network.

Issue 5.0
K-CCIS Call Rerouting
FEATURE DESCRIPTION
The CCIS Call Rerouting feature allows a system to use multiple call routing priorities
when remote system trunks are all busy. The four priorities can be local or remote
trunks. For example using ARS and F-Route table priorities the system can try up to
four remote systems or up to three remote systems and a local trunk to route the
outbound call to its destination. If an outbound route is unavailable for any reason the
call will fall through to the next priority.
Figure 4-1 K-CCIS Call Rerouting
TRG1 All Busy
0
2) TRG10 3) TRG10
TRG1 isIdle
SYSTEM AVAILABILITY
Terminal Type:
All Terminals
GCD-CP10
GPZ-IPLE
K-CCIS License (0016)

Issue 5.0
K-CCISoIP License (5012)
SERVICE CONDITIONS
General:
 The originating system must have a dial treatment of D019RE where 9 is the
ARS trunk access code in the destination system for this feature to work.
 ARS must be enabled in all systems for this feature to work.
 The CCIS Call Rerouting feature is not supported on stations using 3rd party
CTI.
 The CCIS Call Rerouting feature is not supported on a station which is
controlled by 1st party CTI.
 The CCIS Call Rerouting feature will not work if Line Load Control has been
triggered on the destination system.
 The CCIS Call Rerouting feature is only supported on CCISoIP with SV9100
systems.
 The CCIS Call Rerouting feature can only use the four routing options
contained in one F-Route table.
 The CCIS Call Rerouting feature cannot fall through from one F-Route table to
a second F-Route table.
 If none of the four F-Route table route priorities are available the outbound call
will fail.

Issue 5.0
Programming Example
The following example will use the first two priorities of System A to route 10 digit
local calls out trunk group one of System B and if that fails the call is routed out trunk
group one of System A.
This example assumes the following:

 System A has local trunks 1-8 in trunk group 1.
 Both systems are already configured for the LAN on which they reside and
have a network route to each other.
 Both systems are configured for K-CCISoIP.
 System A is assigned point code 1 and 1XX extensions.
 System B is assigned point code 2 and 2XX extensions.
 While dial analysis can be configured in all systems, for this example only
System A has ARS dial analysis configured.
Table 4-1 Programming Example
System A System B
Program Description
Setting Setting
10-68-01 IP Trunk Mode CCIS 3 3

10-68-02 Starting CCIS Trunk Number 9 9
10-68-03 Number of CCIS Trunks to Create.
This will create CCISoIP trunks 9- 8 8
16.
14-05-01 First setting data. Put CCISoIP
10 10
trunks in trunk group 10.
14-05-01 Second setting data. Set priorities 1-
1~8 1~8
8 for trunks 9-16.
26-01-01 Enable ARS Services 1 1
26-02-01 For table entry 1 enter 5 wildcard
@@@@@ N/A
characters.
26-02-02 For table entry 1 set service type to
2 N/A
F-Route (2)
26-02-03 For table entry 1 set additional data
2 N/A
to F-Route Table 2.
26-03-01 Set ARS Treatment 1 to sent the
number including the trunk access D019RE N/A
code of 9 to System B.
44-02-01 Closed numbering setup - Set to first
digit of remote system extension 2 1
number

Issue 5.0
Table 4-1 Programming Example (Continued)
System A System B
Program Description
Setting Setting
44-02-02 Closed numbering setup – Set to F-

2 2
Route table (2).
44-02-03 Closed numbering setup – Set to
use F-Route table 1 for intercom 1 1
calls.
44-05-01 Closed numbering setup – Set table
1 first priority to use CCIS trunk 10 10
group 10.
1 first priority maximum digit length 3 3
to 3.
1 first priority to remote system point 2 1
code.
44-05-01 ARS Routing – Set table 2 first
10 N/A
priority to CCIS trunk group 10.
1 N/A
priority to dial treatment 1
11 N/A
priority maximum dialing digit 11.
2 N/A
priority to System B point code (2).
44-05-01 ARS Routing – Set table 2 second
1 N/A
priority to local trunk group 1.
44-05-09 ARS Routing – Set table 2 second
10 N/A
priority maximum dialing digit 10.

Issue 5.0
This guide provides a list of associated Programs that support this feature. .
Program
Program Name/Description Input Data Default
Number
10-12-01 GCD-CP10 Network Setup - IP Address 0.0.0.0 ~ 126.255.255.254 Default =

Should be set to 0.0.0.0 when using PRG 10- 128.0.0.1 ~ 191.254.255.254 192.168.0.10
12-09. 192.0.0.1 ~ 223.255.255.254
10-12-03 GCD-CP10 Network Setup – Default 0.0.0.0 ~ 126.255.255.254 Default = 0.0.0.0

Gateway 128.0.0.1 ~ 191.254.255.254
Assign the default gateway IP address for the 192.0.0.1 ~ 223.255.255.254
GCD-CP10.
10-12-09 GCD-CP10 Network Setup – IP Address 0.0.0.0 ~ 126.255.255.254 Default =

Set IP Address for GPZ-IPLE. 128.0.0.1 ~ 191.254.255.254 172.16.0.10
192.0.0.1 ~ 223.255.255.254
10-19-01 VoIP DSP Resource Selection 0 = Common use for both IP Resource 1 = 1
Select type of GPZ-IPLE DSP Resource. This extensions and trunks Resource
program setting has no affect on the terminal/ 1 = IP Extension 2 ~ 256 = 0
trunk port assignment or usage. 2 = SIP Trunk
3 = Networking/CCIS
4 = Use for NetLink
5 = Blocked
10-50-01 License Information – License Name N/A

Read Only program used to confirm license
information that is stored in a system.
10-54-01 License Configuration for Each Package – 1 ~ 255 resource licenses No Setting
License Code
Assign VoIP resource licenses (5103) to the
GCD-CP10 slot (1).
10-68-01 IP Trunk Availability – Trunk Type 0 = None Default = 0

Set trunk type to CCISoIP for interconnecting 1 = SIP
trunks. 2 = H.323
3 = CCIS
10-68-02 IP Trunk Availability – Start Port 0 ~ 400 Default = 0

Set the trunk port number to start the
assignment from.
10-68-03 IP Trunk Availability – Number of Ports 0 ~ 400 Default = 0

Set the number of ports to assign from the
starting point set in 10-68-02.
11-01-01 System Numbering – Service Code Refer to the

The Call reroute feature only supports closed Programming
numbering. Manual for
This program is used to customize the system default values.
internal (Intercom) numbering plan.

Issue 5.0
Program
Number
11-02-01 Extension Numbering Maximum of eight digits. 1 200

Assign extension numbers to extension ports. 2 201
The telephone programming identity follows the ~~
port number – not the extension number. 300 499
301 5000
~~
960 5659
14-01-38 Trunk Basic Setup – Outgoing CLI Selection 0: None

Read Only program used to confirm the trunk 1: SIP
type. 2: H.323
3: CCIS
14-05-01 Trunk Group – Trunk Group Number Trunk Group Number: (default = Trunk
Assign CCISoIP trunks to trunk groups (1~100). 0~100 Priority Number: Group 1, with
1~400 priority in
ascending
order.)
20-09-08 Class of Service Options (Incoming Call 0 = Off Default COS 1 ~

Service) – Calling Party Information 1 = On 15 = 0
Turn Off or On an extension user ability to have
calls queued if a call rings the extension when it
is busy.
20-14-01 Class of Service Options for DISA/E&M – 0 = Off Default COS 1 ~

First Digit Absorption (Delete First Digit 1 = On 15 = 0
Dialed)
This feature must be enabled in the system
where the outgoing trunks reside for the Class
of Service (COS) sent be the originating
system, at default this is COS 1.
Turn Off or On a DISA or tie trunk caller ability
to dial 9 for Trunk Group Routing or Automatic
Route Selection.
22-02-01 Incoming Call Trunk Setup Trunks 1 ~ 400 Default = 0

Set the CCISoIP trunks to 5 (Tie Line). 0 = Normal
1 = VRS (second dial tone if no
VRS installed)
2 = DISA
3 = DID
4 = DIL
5 = E&M Tie line
6 = Delayed VRS
7 = ANI/DNIS
8 = DID(DDI) Mode Switching
26-01-01 Automatic Route Selection Service – ARS 0 = Disabled (ARS service is Default = 0
Service Off)
ARS must be enabled in all system for this 1 = Enabled (ARS service is On)
feature to work.

Issue 5.0
Program
Number
26-01-06 Automatic Route Selection Service – Class 0 = Disable (Off) Default = 0

of Service Match Access 1 = Enable (On)
With the ARS Class of Service Match Access
feature, you can determine whether or not the
system should allow a call based on the COS
assigned to the Dial Analysis Table (Program
26-02).
This change can be used to create a tenant-like
application. It then uses the trunk group set in
the Additional Entry in Program 26-02-03 to
place the out-going call.
When this feature is enabled, the calls are
routed in sequential order, and forward –
provided the Class of Service for the trunk
groups match.
26-01-07 Automatic Route Selection Service – 0 = F-Route Default = 0

F-Route Access COS Reference 1 = ARS
Define where the system looks for the
Automatic Route Selection (ARS) class of
service.
26-02-01 Dial Analysis Table for ARS/LCR – Dial Dial a maximum of 16 digits Default = No
Enter the digits (16 digits maximum: 1~9, 0, #, (0 ~ 9, #, @) Setting
@; 800 separate entries) for the Dial Analysis
Table which is analyzed by ARS/LCR.
This table is checked after any programmed F-
Route operations have completed.
The system then refers to Program 26-02-02
and Program 26-02-03 to determine the routing
for the call.
To enter a wild card/don’t care digit, press Line
Key 1 to enter an @ symbol. It is important to
remember that the system checks the table
numbers in numerical order.
This means that entries for specific numbers
should be entered first (such as your local area
codes), then enter the items containing wild
card digits. If the system sees an entry of
2@@, any table entries which follow are
ignored.
For example, if 268, 269, and 270 are local
exchanges, these would be the first three table
entries which route according to the settings
made in Program 26-02-02 and Program 26-02-
03 for each of the table entries. If the next entry
is 2@@, the system checks no further in this
program and routes all other 2xx numbers
according to the entries made in Program 26-
02-02 and Program 26-02-03 for this table
entry.

Issue 5.0
Program
Number
26-02-02 Dial Analysis Table for ARS – ARS Service 0 = No Service (Call Restricted) Default = 0
Type 1 = Route to Trunk Group
For each Dial Analysis Table used (1~2000), 2 = Select F-Route Access
select Service Type 2 – F-Route Selected to
have the dialed number controlled by the F-
Route
table. If Service Type 2 is selected and
F-Route operation is on, the F-Route table used
is determined by Program 44-04.
26-02-03 Dial Analysis Table for ARS – Additional F-Route Table 0~500 Default = 0
Data
For each Dial Analysis Table (1~2000), enter
the F-Route table to use (1~500)
26-02-04 Dial Analysis Table for ARS – ARS Class of Class = 0 ~ 50 Default = 0
Service
For each Dial Analysis Table (1 ~ 2000), set the
Automatic Route Selection (ARS) Class of
Service (0 ~ 50).
26-03-01 ARS Dial Treatments – Treatment Code Maximum of 24 characters Default = Blank
For the originating system a treatment code
must be used for any route that will use
outbound trunks in a remote system.
The recommended treatment code is D019RE
where 9 is the trunk access code in the
destination system.
26-04-01 ARS Class of Service Class = 0 ~ 50 Default = 0

Set class of service for ARS. Used when 26-01-
07 is set to ARS.
34-02-01 E&M Tie Line Class of Service Day/Night Mode Default = 1

This program can be used to assign the Tie 1~8
Line Class of Service (1 ~ 15) for inbound Class: 1 ~ 15
trunks.
Use Program 20-14-01 to set the Tie Line Class
of Service options. You cannot use Program 20-
06 to assign Class of Service to Tie Lines.
44-02-01 Dial Analysis Table for ARS/F-Route Access Up to eight digits max. Default = No
– Dial Setting
Set the Dial digits for the Pre-Transaction Table
for selecting ARS/F-Route (eight digits
maximum: 1 ~ 9, 0, #, @). To enter a wild card/
don’t care digit, press Line Key 1 to enter an @
symbol.
44-02-02 Dial Analysis Table for ARS/F-Route Access 0 = No Setting (None) Default = 0
– Service Type 1 = Extension Call (Own)
Set the Service Type (0 ~ 3) for the 2 = ARS/F-Route Table
Pre-Transaction Table for selecting ARS/ (F-Route)
F-Route. 3 = Dial Extension Analyze
Table (Option)

Issue 5.0
Program
Number
44-02-03 Dial Analysis Table for ARS/F-Route Access 1 = Delete Digit = 0 ~ 255 Default = 0
– Additional Data (255: Delete All Digits)
If a Service Type is set to F-Route in Program 2 = 0 ~ 500
44-02-02, set which F-Route table to use. (0 = No Setting)
Table Number = 0 ~ 4
(0 = No Setting)
44-05-01 ARS/F-Route Table – Trunk Group Number 0-100,101-150,255 Default = 0

Select the trunk group number to be used for 0 = Not Set
the outgoing ARS call. 1~100 = Trunk Group from
For this feature you would use trunk groups for PRG14-05
the CCIS connections. 101~150 = Networking
44-05-04 ARS/F-Route Table – Beep Tone 0 = Off (No Beep) Default = 0

Enable this option if the customer wants to hear 1 = On (Beep)
an audible beep every time a F-Route priority is
used.
44-05-07 ARS/F-Route Table – ARS Class of Service 0 ~ 50 Default = 0

For each ARS/F-Route table (1 ~ 500) and
each priority number (1 ~ 4) you can assign a
Class of Service to be used for ARS calls. This
COS is sent to the destination system.
Extension ARS COS is determined in Program
26-04-01.
44-05-08 ARS/F-Route Table – Dial Treatment 0 ~ 15 Default = 0

For each ARS/F-Route table (1 ~ 500) and
priority number (1 ~ 4) you must assign a dial
treatment to use.
The recommended treatment code is D019RE
where 9 is the trunk access code in the
destination system.
The Dial Treatments are defined in Program 26-
03-01.
44-05-09 ARS/F-Route Table – Maximum Digit 0 ~ 24 Default = 0

Set the maximum number of digits to send
when using the F-Route.
44-05-10 ARS/F-Route – CCIS over IP Destination 0 ~ 16367 Default = 0

Point Code
For each ARS/F-Route table (1 ~ 500). Set the
CCIS over IP Destination Point Code (0 ~
16367).
50-01-01 CCIS System Setting – 0 = Disable Default = 0

CCIS Availability 1 = Enable
Any CCIS settings lose functionality if this
setting is set to 0.
50-02-03 Connecting System Settings – Originating 0 ~ 16367 Default = 0

Point Code
For Route ID 9 set the Origination point code.

Issue 5.0
Program
Number
50-03-01 CCIS Destination System Settings – 0 ~ 16367 Default = 0

Destination Point Code
Assign the destination transfer point code for
Tandem KTS.
50-03-03 CCIS Destination System xxx.xxx.xxx.xxx Default = 0.0.0.0

Settings – IP Address (IP (xxx = 0 ~ 255)
only) Default is 0.0.0.0
Assign remote system IP network information
84-26-01 IPLE Basic Setup – IP Address XXX.XXX.XXX.XXX Default:

Assign the IPLE IP Address. Slot 1 =
172.16.0.20
84-26-02 IPLE Basic Setup – RTP 0 ~ 65534 Default = 10020

Port Number
If needed the RTP port number can be set here.
84-26-03 IPLE Basic Setup – 0 ~ 65534 Default = 10021

RTCP Port Number
If needed the RTCP port number can be set
here.
(RTP Port Number + 1)

Issue 5.0
Call Park Retrieve – K-CCIS
FEATURE DESCRIPTION
This feature allows a station user to retrieve Parked Calls at remote sites across
K-CCIS. Locally parked calls can be retrieved from a remote system, connected via
K-CCIS, by dialing the Call Park Hold Group Number, plus the park orbit location.
SYSTEM AVAILABILITY
Terminal Type:
All Terminals
GCD-CCTA
- OR -
GPZ-IPLE
Retrieve Park Call at remote location:

1. Go off-hook, and wait for internal dial tone.
2. Dial the Call Park Retrieve Access Code 861 (default) + Park Hold Group
Number (1~64) + Park Orbit Number (1~64) of the call to be retrieved. Call
Park Retrieve access codes cannot be the same at all locations in the K-
CCIS Network.
3. Talk with party.
SERVICE CONDITIONS
General:
 A different Call Park Retrieve Access Code must be programmed for each
system in the K-CCIS network.
 The Park Group Number and Park Orbit Number must be dialed immediately
following the Park Retrieve Service Code.
 When two or more stations attempt to retrieve the parked call, only one station
can retrieve the call.

Issue 5.0
 A station connected to a PBX can retrieve a parked call in an UNIVERGE

SV9100, but the station connected to the UNIVERGE SV9100 system cannot
retrieve a parked call in a PBX.
 A Park Hold key cannot be used to retrieve a parked call from a distant system.
 F-Routes are required to route Call Park Retrieve Access Code to proper
system in the K-CCIS network.
 When a remote location retrieves a call from another location, the call is treated
as if it were transferred from the distant location.
 SMDR reports the retrieved call from the distant location as if it were a
transferred call.
 When a call that has Caller ID Information is retrieved at the distant location the
Caller ID information is treated as if it were a transferred call.
 Link Reconnect operates when the trunk is retrieved back to the origination
system.
Restrictions:
 A Call cannot be placed into remote systems Call Park Location.
 Call Park Retrieve – K-CCIS is only a Key System-to-Key System supported
feature.
 The digit (# or ) cannot be used in conjunction with IP K-CCIS.
 When the UNIVERGE SV9100 is connected to the Electra Elite IPK II, the
maximum digits assignment in the UNIVERGE SV9100 is determined by
Program 44-05-09.
 Call Park Searching is supported in the local system only.

 Call Park – System

Issue 5.0
Park Originate System
Program/
Item No.
11-12-32 Answer for Park 861 (default) This varies based on the K-CCIS network
configuration and PRG 11-01-01.
20-14-12 Class of Service for DISA/ 0 = Off (default) Enable Retrieve Park Hold feature per Class of
E&M – Retrieve Park Hold 1 = On Service.
24-03-01 Park Group 01~64 Assigns an extension to a Park Group.
1 = Default
Remote System (Call Park Retrieve)
Program/
Item No.
11-01-01 System Numbering
11-12-32 Service Code Setup (for Default is 861

Service Access) – Answer
for Park Hold
20-14-12 Class of Service Options 0 = Off Enable Retrieve Park Hold feature per Class of
for DISA/E&M – Retrieve 1 = On Service.
Park Hold Default is 1
24-03-01 Park Group 01~64 Assigns an extension to a Park Group.
Default is 1
44-02-01 Dial Analysis Table for ARS/ Analysis Table 1 = 861 Assigns access code used to retrieve the parked
F-Route Access –Dial Default is No Setting call.
44-02-02 Dial Analysis Table for ARS/ 0 = No setting (None) To assign the service type used.
F-Route Access –Service 1 = Extension Call (Own)
Type 2 = ARS/F-Route Table
(F-Route)
3 = Dial Extension.
Analyze Table (Option)
Default is 2

Issue 5.0
Program/
Item No.
44-02-03 Dial Analysis Table for ARS/ 0 = No setting Enter additional data required for the Service Type
F-Route Access – 1 = Delete Digits = 0~255 selected in PRG 44-02-02.
Additional Data (255 = delete all
digits)
2 = 0~500
3 = Dial Extension
Analyze Table
Number = 0~4
Default is 0
44-05-01 ARS/F-Route Table – Trunk 0 = Not Set Select trunk group number used for outgoing ARS
Group Number 1 ~ 100 = Trunk Group calls.
from PRG14-05 Setting of 255 = Internal Extension Call.
101 ~ 150 Networking
Default is 0
44-05-02 ARS/F-Route Table – 0 = No setting Enter number of digits to delete from the dialed
Delete digits 1~254 number.
255 = Delete all Digits
Default is 0
44-05-03 ARS/F-Route Table – 0 = No setting Enter table number (defined in 44-06) for additional
Additional Dial Number 1~1000 digits to be dialed.
Table Default is 0
44-05-09 ARS/F-Route Table – Max 0 = No Max Assign Max digits for the Call Park Retrieve Access
Digit 1~24 Code.
Default is 0

Issue 5.0
Programming Example:
For the following example, to retrieve a call which is parked, use the following access
codes from any system:
SV9100
SV9100 Electra Elite IPK II
T1 T1
DPC 1 CCIS DPC 2 CCIS DPC 3
CD-CCTA CD-CCTA CD-CCTA DTI

Cabinet 1 Cabinet 1 Cabinet 1 Cabinet 1
Slot 1 TG 10 TG 10 Slot 1 Slot 4 TG 11 TG 10 Slot 1
System A System B System C

(101s) (201s) (301s)
Call in Park
Group 01   
Call Park Retrieve

Call Parked At Notes
Access Codes
501 = System A Call Park Retrieve Access Code

System A 501+05
05 = Park Orbit Number
502 = System B Call Park Retrieve Access Code
System B 502+01+05 01 = Park Hold Group Number
503 = System C Call Park Retrieve Access Code
System C 503+01+05 01 = Park Hold Group Number
System A (101s) System B (201s) System C (301s)
PRG 11-01-01 PRG 11-01-01 PRG 11-01-01

50 = 2 digits; F-Route 50 = 2 digits; F-Route 50 = 2 digits; F-Route
PRG 11-12-32 PRG 11-12-32 PRG 11-12-32
Ans Park Hold = 861 Ans Park Hold = 861 Ans Park Hold = 861
(Park Retrieve System A) (Park Retrieve System B) (Park Retrieve System C)
PRG 44-02-01 PRG 44-02-01 PRG 44-02-01
Table 1 = Dial 501 Table 1 =Dial 502 Table 1 =Dial 503
(Park Retrieve System A) (Park Retrieve System B) (Park Retrieve System C)

Issue 5.0
PRG 44-02-01 PRG 44-02-01 PRG 44-02-01

Table 2 = Dial 502 Table 2 = Dial 501 Table 2 = Dial 501
(ARS Table #4) (Park Retrieve System A) (Park Retrieve System A)
(Park Retrieve System B)
PRG 44-02-01 PRG 44-02-01 PRG 44-02-01
(Park Retrieve System C) (Park Retrieve System C) (Park Retrieve System B)
PRG 44-02-02 PRG 44-02-02 PRG 44-02-02
Table 1 = F-Route Table 1 = F-Route Table 1 = F-Route
PRG 44-02-03 PRG 44-02-03 PRG 44-02-03
Table 1 = F-Route 1 Table 1 = F-Route 1 Table 1 = F-Route 1
PRG 44-05-01 PRG 44-05-01 PRG 44-05-01
Table 1 = TRK GP 255 Table 1 = TRK GP 255 Table 1 = TRK GP 255
PRG 44-05-02 PRG 44-05-02 PRG 44-05-02
Table 1 = Delete Digits 3 Table 1 = Delete Digits 3 Table 1 = Delete Digits 3
PRG 44-05-03 PRG 44-05-03 PRG 44-05-03
Table 1 = Add Dial 1 Table 1 = Add Dial 1 Table 1 = Add Dial 1
PRG 44-05-09 PRG 44-05-09 PRG 44-05-09
Table 1 = Max Digit 0 Table 1 = Max Digit 0 Table 1 = Max Digit 0
* PRG 44-05-10 * PRG 44-05-10
Table 2 = DPC Table1 = 1 Table 2 = DPC Table1 = 1
Table 2 = DPC Table2 = 3 Table 2 = DPC Table2 = 2
PRG 44-06-01 PRG 44-06-01 PRG 44-06-01
* For CCISoIP Programming only.

Issue 5.0
Call Transfer – All Calls – K-CCIS
FEATURE DESCRIPTION
This feature allows a station user to transfer incoming or outgoing Central Office,
intraoffice, and interoffice calls to another station in the K-CCIS network without
Attendant assistance.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
Using a Multiline Terminal with a call in progress (Closed Numbering Plan):

1. Press Transfer, and internal dial tone is heard. The call is placed on
Non-Exclusive Hold.]
4. Hang up.
- OR -


Issue 5.0
Using a Multiline Terminal with a call in progress (Open Numbering Plan):

1. Press Transfer, and receive internal dial tone. The call is placed on
Non-Exclusive Hold.
6. Hang up.
- OR -

SERVICE CONDITIONS
General:
 A UNIVERGE SV9100 station can receive a K-CCIS transferred call as a
camp-on call if allowed by Class of Service.
Restrictions:
 A blind transfer across a K-CCIS link cannot be completed until ringback tone is
received at the transferring station.


Issue 5.0
Program/
Item No.
20-09-07 Class of Service Options 0 = Off To enable receiving Call Queuing.

20-06-01 Class of Service for 0~ 15 All Extensions are in Class 1. (default)
Extensions
20-09-01 Class of Service Options 0 = Off Turn Off or On the extension ability to receive the
(Incoming Call Service) – 1 = On second call.
Second Call for DID/DISA/
DIL/E&M Override Default is 0
20-13-06 Class of Service Options 0 = Manually Allow a busy extension user to manually or
(Supplementary Service) – 1 = Automatically automatically receive off-hook signals.
Automatic Off Hook
Signaling (Automatic Default is 0
Override)
14-01-13 Basic Trunk Date setup – 0 = Disable Must enable for Trunk-to-Trunk Transfer,
Trunk-to-Trunk Transfer 1 = Enable Call-Forward – Off-Premise, or tandem trunking.

50-06-01 CCIS Feature Availability – 0 = Not Available Enable Link Reconnect.
Link Reconnect 1 = Enable
Default is 1

Issue 5.0
Calling Name Display – K-CCIS
FEATURE DESCRIPTION
This feature permits the station name of a calling or called party at another switching
office to be displayed on a multiline terminal, through the K-CCIS network.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
SERVICE CONDITIONS
General:
 Both the caller/calling station number name and number can be displayed on
an UNIVERGE SV9100 station if allowed by Class of Service.
 For incoming or outgoing K-CCIS calls, the Calling/Called Name and Number
are displayed for the entire length of the call including the Elapsed Call Time.
RESTRICTIONS:
 In the UNIVERGE SV9100 system, only 12 digits/characters can be entered for
each station name.

Issue 5.0

 Calling Number Display – K-CCIS
Item No. Selection
14-02-10 Analog Trunk Date Setup – 0 = No Enable/Disable a trunk ability to receive Caller ID.
Caller ID 1 = Yes
Default is 0
20-09-02 Class of Service Options 0 = Off Control the Caller ID Display at an extension.
Caller ID Display Default is 1
20-09-08 Class of Service Options 0 = Off Enable receiving Calling Party Information from
(Incoming Call Service) – 1 = On K-CCIS.
Calling Party Information Default is 1
Extensions
50-02-05 Connecting System 0 = Disable Enable receiving Calling Name indication from
Settings – Calling Name 1 = Enable K-CCIS.
Indication (T1) Default is 1
15-01-01 Basic Extension Data Setup Up to 12 Characters Set the extension/Virtual extension name.
– Extension Name Default:
Sta 200 101 = Ext 200
101
Sta 201 102 = Ext 201
102
etc.

Issue 5.0
Calling Number Display – K-CCIS
FEATURE DESCRIPTION
This feature permits the number of a calling or called party at another switching
office, to be displayed on a multiline terminal through the K-CCIS network.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
SERVICE CONDITIONS
General:
 Both the caller/calling station number and name can be displayed on an
UNIVERGE SV9100 station if allowed by Class of Service.
 For incoming or outgoing K-CCIS calls, the Calling/Called Name and Number
are displayed for the entire length of the call including the Elapsed Call Time.
 For an open numbering plan the Office Code number and station number are
displayed for caller/calling station number.
Restrictions:
 The UNIVERGE SV9100 supports 2~8-Digit station numbers.
 When calling over a K-CCIS tandem connection, the calling party number
(CPN) is transferred to the ISDN network.

Issue 5.0

 Calling Name Display – K-CCIS
Item No. Selection
14-02-10 Analog Trunk Date Setup – 0 = Disable (No) Enable/Disable a trunk ability to receive Caller ID.
Caller ID 1 = Enable (Yes)
Default is 0
20-09-02 Class of Service Options 0 = Off Control the Caller ID Display at an extension.
20-09-08 Class of Service Options 0 = Off Enable receiving Calling Party Information from
(Incoming Call Service) – 1 = On K-CCIS.
Calling Party Information Default is 1
20-06-01 Class of Service for 0~15 All Extensions are in Class 1. .(default)
Extensions
50-02-05 Connecting System 0 = Disable Enable receiving Calling Name indication from
Settings – Calling Name 1 = Enable K-CCIS.
Indication (T1) Default is 1
15-01-01 Basic Extension Data Setup Up to 12 Characters Set the extension/Virtual extension name.
– Extension Name Default:
Sta 200 101 = Ext 200
101
Sta 201 102 = Ext 201
102
etc.

Issue 5.0
Calling Party Number (CPN) Presentation from

Station – K-CCIS
FEATURE DESCRIPTION
Calling Party Number (CPN) Presentation from Station K-CCIS feature allows each
station of the remote systems a unique 10-digit number (the DID number of the
originating station) to be sent out over the PRI circuit of the main system.
SYSTEM AVAILABILITY
Terminal Type:
All Terminals
GCD-CCTA
- OR -
GPZ-IPLE
Placing a call with CPN:

2. Dial 9 + the number.
3. Converse with caller.
4. Hang up the handset.
SERVICE CONDITIONS
Restrictions:
 A maximum of 16 digits can be assigned as the Calling Party Number (CPN) in
Program 21-12-01 and Program 21-13-01.
 The PRI provider must provision for the CPN used for E911. The CPN must be
within the allowable range. For more information please contact your local
ISDN provider regarding allowable ranges.
 The Calling Party Number (CPN) is sent only to the network when the calling
party from the remote system dials a trunk access code of 9 when making an

Issue 5.0
outbound call.
 The Calling Party Number (CPN) is not sent to the network when the originating
station of the remote system calls a station in the main system that is call
forwarded off site.

 ISDN Compatibility
 Automatic Route Selection
 Central Office Calls, Placing
 T1 Trunking (with ANI/DNIS Compatibility)

Item No. Selection
Basic Trunk Data Setup – 0 = Off Enable Outgoing Caller ID through Mode for
Trunk-to-Trunk 1 = On each CCIS trunk to enable CPN information
14-01-24 Outgoing Caller ID to pass through the Tandem Office.
through Mode
Default is 0 Tandem System Only
Class of Service Options 0 = Off Determine if the ISDN calling line identity
(Outgoing Call Service) – 1 = On presentation and screening indicators are to
20-08-13
ISDN CLIP be allowed.
Default is 1
ISDN Calling Party Number Up to 16 digits max. Assign each trunk a Calling Party Number.
Setup for Trunks
When a call is made by an extension which
does not have an Extension Calling Number
21-12-01
assigned
(Program 21-13), the system sends the
calling number for the ISDN trunk defined in
Default is No Setting 21-12.
ISDN Calling Party Number Up to 16 digits max. Assign each extension a Calling Party
21-13-01 Setup for Extensions Number.
Default is No Setting

Issue 5.0
Centralized Billing – K-CCIS
FEATURE DESCRIPTION
This feature sends the billing information from local systems to a billing center office
for central management of all billing information in the network. The UNIVERGE
SV9100 can send billing information to a billing center office (NEAX2000/2400), but
cannot receive the billing information as the billing center office.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
Required Components
GCD-CCTA
- OR -
GPZ-IPLE
Not Applicable
SERVICE CONDITIONS
General:
 The Station Message Detail Recording (SMDR) feature and Centralized Billing
Feature can be used at the same time.
 Centralized Billing – K-CCIS feature supports the following calls:
 Incoming CO Calls (using the main system and another system trunk/K-CCIS
trunk)
 Outgoing CO Call (using the main system and another system trunk/K-CCIS trunk)
 Call Transfer of CO calls (using the main system and another system trunk/K-CCIS
trunk)
 Conference Calls (using the main system and another system trunk/K-CCIS trunk)

Issue 5.0
Restrictions:
 In a K-CCIS network, the PBX must be the main system where billing
information is sent. Centralized billing cannot be used in a KTS-to-KTS
network.
 Station-to-station calls in their own system are not reported to the billing center
office with UNIVERGE SV9100.
 The information storage capacity of the local UNIVERGE SV9100 office is
approximately 300 calls. If the K-CCIS link is down due to network trouble, the
billing information is stored by the GCD-CP10. When the maximum calls
exceed this amount, the oldest call information is overwritten by the latest
(newest) call.
 With the UNIVERGE SV9100, trunk type (e.g., analog or ISDN) information is
not reported to the billing center office.
 When the K-CCIS link is down due to network trouble, the UNIVERGE SV9100
system does not provide an SMDR alarm indication.
 Account Codes cannot exceed 10 digits.

 Account Code – Forced/Verified/Unverified
 Account Code Entry
 Authorization Code
 Centralized Day/Night Mode Change – K-CCIS
 Station Message Detail Recording (SMDR)
 Voice Mail Integration – K-CCIS

Issue 5.0
For Centralized Billing Installation
Item No. Selection
Basic Trunk Data Setup – 0 = Off Enable Outgoing Caller ID through Mode for
Trunk-to-Trunk 1 = On each CCIS trunk to enable CPN information
14-01-24 Outgoing Caller ID to pass through the Tandem Office.
through Mode
Default is 0 Tandem System Only
Class of Service Options 0 = Off (Default) Determine if the ISDN calling line identity
20-08-13 (Outgoing Call Service) – 1 = On presentation and screening indicators are to
ISDN CLIP Default is 1 be allowed.
ISDN Calling Party Number Up to 16 digits max. Assign each trunk a Calling Party Number.
Setup for Trunks
When a call is made by an extension which
does not have an Extension Calling Number
21-12-01
assigned
calling number for the ISDN trunk defined in
Program/
Item No.
CCIS System Setting – 0 = Disable All CCIS settings lose functionality when 0 is
50-01-01 CCIS Availability 1 = Enable selected.
Default is 0
CCIS Destination System 0~16367 Assign the destination transfer point code for
50-03-01 Settings – Destination Tandem KTS.
Point Code Default is 0
CCIS Centralized Billing 0~16367 Assign point code for the Centralized Billing Office
50-07-01 Center Office – Destination PBX.

Issue 5.0
For Station Message Detail Recording (SMDR)
Program/
Item No.
System Options for 0~64800 seconds The system waits for this time to expire before
21-01-03 Outgoing Calls – Trunk Default is 5 placing the call.
Interdigit Time (External)
SMDR Options – Omit 0 = Not Applied The number of leading digits that do not print on
35-01-04 Digits 1~24 the SMDR report.
Default is 0
For Centralized Billing K-CCIS
Program/
Item No.
SMDR Options – Minimum 0 = All A call must be longer than this duration to be
35-01-06 Call Duration 0~65535 seconds included in the SMDR report.
Default is 0
SMDR Output Options – 0 = Not Displayed Select whether or not incoming calls are displayed
35-02-08 Incoming Call 1 = Displayed on the SMDR report.
Default is 1
Basic Extension Data Setup 0 = Not printed on SMDR Use to include the extension being programmed in
– SMDR Printout report the SMDR report.
15-01-03 1 = Included on SMDR
Report
Default is 1

Issue 5.0
Centralized BLF (K-CCIS)
FEATURE DESCRIPTION
This feature provides a busy indication for another station across the K-CCIS
network on programmed Direct Station Selection/Busy Lamp Field (DSS/BLF) keys.
The busy indication is a red LED associated with a Feature Access or One-Touch key
programmed for Centralized BLF (K-CCIS). Pressing the Centralized DSS/BLF key
allows direct access to the station through the K-CCIS network. Do Not Disturb and
Voice Mail Message Waiting on Line key indication are also supported.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
Attendant Add-On Console
GCD-CCTA
- OR -
GPZ-IPLE
To program a Feature Access or One-Touch key for Centralized DSS/BLF:

1. Press Speaker.
2. Dial 851 .
3. Press the key to be programmed.
4. Dial 01 to assign a DSS/One Touch Key.
5. Dial the station number.
6. Press Hold.

Issue 5.0
Using a Feature Access or a One-Touch key programmed for Centralized DSS/BLF:

1. Press the programmed Feature Access or One-Touch key. Hear ringback
tone.
2. When the called party answers, lift the handset or talk using handsfree if
allowed.
SERVICE CONDITIONS
General:
 Voice Mail Message Waiting on Line Key indication is supported for Centralized
DSS/BLF keys if PRG 20-13-41 [Class of Service Options (Supplementary
Service)] – VM Message Indication on DSS/BLF key (VMS Message Indication)
is allowed.
 If Voice Mail Message Waiting on Line Key indication is allowed and a VM
Message Waiting indication is provided (a new message is stored), pressing
the Centralized DSS/BLF key performs the following;
 At system with Voice Mail installed, the user is logged into the owner mail box.
 At remote systems, the station is called.
BLF Sending Service Conditions:

 The maximum number of destination offices for sending BLF messages is eight
per system.
 Up to 120 Extension Numbers (entries into the table) can be assigned for
sending BLF messages. With each assigned Extension Number, up to eight
destination offices can be selected until a maximum of 240 total sending
Extension Numbers are assigned.
 A maximum of 240 total sending Extension Numbers (BLF messages) can be
assigned. If 30 Extension Numbers (entries into the table) are assigned with
each set for all eight groups (systems), the 240 limit is reached and no more
Extension Numbers can be entered.
 The BLF messages are sent in a four-second cycle (at default), so some delay
occurs to change the indication in the destination office. In the network
configured with two systems, it can take about four to five seconds (at default)
to change the BLF indication in the destination office.
 This feature is provided with DSS/One-Touch Keys on multiline terminals and
with an Attendant Add-On Console.
 When the button on the Attendant Add-On Console has a Mail Box button of a
remote user programmed and the button is pressed the call is placed to the
station on the remote side.

Issue 5.0
BLF Receiving Service Conditions:

 BLF information can be received for up to 120 remote extensions per system.
 All multiline terminals in the system can assign Centralized DSS/BLF keys for
the supported remote extensions.
 The LED indication of the DSS/BLF button on a multiline terminal is as follows:
 Idle – No lamp indication
 Busy – Steady red lamp
 Do Not Disturb – Flashing red lamp
 VM Message Waiting – Fast flashing red lamp
 The LED indication of the DSS/BLF button on the Attendant Add-on Console is as
follows:
 Idle No lamp indication
 Busy Steady red lamp
 Do Not Disturb Flashing red lamp
 VM Message Waiting Fast flashing red lamp

 The BLF Information expels when data cannot be sent because of link
disconnect. Status changes of BLF information while the system could not send
data are not indicated on restoration.
 The Voice Mail MSG Waiting has priority over any other state of the flashing line key or
One-Touch key.
Restrictions:
 This feature is not supported between UNIVERGE SV9100 and NEAX PBXs.
 This feature is supported with a Closed Numbering Plan only (not available with
an Open Numbering plan).
 The same extension line from a remote site can be assigned to multiple DSS/
One Touch keys.
 The BLF information is expelled when data cannot be sent if the K-CCIS link is
down. The UNIVERGE SV9100 does not send BLF information again when the
K-CCIS link is restored.
 BLF messages can be forwarded up to eight times in the network. When
designing the K-CCIS network, this should be a consideration.
 When a Centralized DSS/BLF key is first programmed on a Feature Access or
One Touch key, the BLF status does not change (update) until new BLF
information is received from the remote system.

Issue 5.0

 Do Not Disturb (DND)
 Feature Access – User Programmable
 Voice Mail Message Indication on Line Keys
GUIDE TO FEATURE PROGRAMMING (FOR MAIN SYSTEM)
(For Sending System)
Program/
Item No.
DSS Console Operating 0 = Business Mode

Mode 1 = Hotel Mode
30-01-01
2 = ACD Monitor Mode
3 = Business/ACD Mode
DSS Console Extension Up to eight digits Identifies extensions with DSS Consoles attached.
30-02-01
Assignment Default is No Setting.
DSS Console Key Keys No. 001~114 Customize key assignments for DSS Consoles
assignment 00~99 = General 1~32.
Functional Level
30-03-01 *00~*99 = Appearance
Functional Level
Default = extensions.
200~259
CCIS Centralized BLF 0~16367 Define the Point Code of Billing Center Office.
Sending Group Assignment 1~8+Destination Point
50-08-01
– Destination Point Code Code
Default is 0
CCIS Centralized BLF 0~8 Define the CCIS Route ID to send Billing Center
Sending Group Assignment Office.
50-08-02
– CCIS Route ID  Not used with IP K-CCIS.
Default is 0
CCIS Centralized BLF Up to eight digits BLF message is indicated when the status of the
Sending Extension Number specified extension number is changed.
50-09-01
Assignment – Extension
number Default is No setting.
CCIS Centralized BLF Select Tables 1~ 120 Enable sending BLF to Send Group 1 assigned in
Sending Extension Number 0 = Disable PRG 50-08-XX.
50-09-02
Assignment – Send to 1 = Enable
Sending Group 1 Default is 0

Issue 5.0
Program/
Item No.
CCIS Centralized BLF 0 = 4 seconds Assign BLF sending interval to each sending
Interval Time Assignment – 1 = 8 seconds system.
50-10-01 Type of Interval Time 2 = 12 seconds
3 = 16 seconds
Default is 0
(For Receiving System)
Program/
Item No.
Class of Service Options 0 = Off Allow the DSS/BLF to indicate when the extension
(Supplementary Service) – 1 = On has a new message waiting in VM.
20-13-41
Voice Mail Message
Indication on DSS key Default is 0
Programmable Function LK01 = 01(Trunk Line Enter extension number up to eight digits.
15-07-01
keys Key)
Class of Service for 0~15 All Extensions are in Class 1. .(default)

20-06-01
Extensions
DSS Console Key 1~114 0~99 (General Functional)
30-03-01 Assignment 00~99 (Appearance Functional Level)
95 = Page Switching

Issue 5.0
Centralized Day/Night Mode Change – K-CCIS
FEATURE DESCRIPTION
This feature switches the Day/Night mode of a remote office that is linked to a main
office using K-CCIS, in accordance with the Day/Night mode switching from an
Attendant Position at the main office.
When a UNIVERGE SV9100 system is connected to another UNIVERGE SV9100

system, the main office can control remote offices.
When connected to a NEAX2400, the UNIVERGE SV9100 can be used only as a

remote office.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
Required Components
GCD-CCTA
- OR -
GPZ-IPLE
To set or cancel Night Transfer system-wide from an Attendant Position:
Main Office:
1. Press Speaker.
2. Dial Access Code 818 .
3. Dial the Night service Code:
1 Day 1 mode 5 Day 2 mode
2 Night 1 mode 6 Night 2 mode
3 Midnight 1 mode 7 Midnight 2 mode
4 Rest 1 mode 8 Rest 2 mode
4. Press Speaker or hang up.

Issue 5.0
- OR -
1. Press the Night Transfer key on the Attendant Add-On Console.
Remote Office:
No manual operation is required.
SERVICE CONDITIONS
General:
 A maximum of 16 remote offices can be controlled by one main office.
 If Automatic Day/Night Mode Switching is assigned in the main office, all
remote offices change the mode, if assigned.
 If the remote office is to be restricted from overriding the Day/Night Mode
setting, the following Memory Blocks should be assigned:
 12-01-01 Night Mode Function Setup – Manual Night Service enable
 20-07-01 Class of Service Options (Administrator Level) – Manual Night Service

enable
 When the remote office is in Night Mode (as assigned in the Centralized
Day/Night Mode - K-CCIS feature), normal Night Mode indications are
provided.
 The Night Mode indication is the first word (Night) on the second row of the
multiline terminal LCD. The LED for any Feature Access key assigned for Night
Mode transfer and the Night Mode key on the Attendant console are On.
 If the K-CCIS link is not available due to network trouble, the UNIVERGE
SV9100 main office resends the K-CCIS Day/Night Mode switch command
every 16 minutes.
Restrictions:
 Centralized Day/Night Mode switching from a main office can send a
system-wide K-CCIS Day/Night mode switch command only. Individual Night
Service Groups Mode switching is not supported.
 When an UNIVERGE SV9100 receives the K-CCIS Day/Night Mode switch
command from a main office, the remote office changes all Night Service
Groups to the requested mode.
 Program 50-03-01 (Destination Point Code Transfer Assignment) must be set
for all offices for the Centralized Day/Night Mode feature.
 A NEAX2000 cannot be used as a main office or a tandem office for the
Centralized Day/Night Mode Change (K-CCIS) feature.

Issue 5.0

 Assigned Night Answer (ANA)
 Authorization Code
 Automatic Day/Night Mode Switching
 Centralized Billing – K-CCIS
 Code Restriction
 Dial Access to Attendant – K-CCIS
 Direct Inward Termination (DIT)
 Flexible Ringing Assignment
 Night Call Pickup
 Night Chime
 Night Transfer
 Voice Mail Integration – K-CCIS
Centralized Day/Night Mode Change Installation
Program/
Item No.
CCIS System Setting – 0 = Disable All CCIS settings lose functionality when this
50-01-01 CCIS Availability 1 = Enable setting is 0.
Default is 0
CCIS Destination System 0~16367 Enable for all offices for Centralized Day/Night
50-03-01 Settings – Destination Point mode change.
Code Default is 0
CCIS Feature Availability – 0 = Disable If this data is set to 0, Link Reconnect does not run.
Centralized Day/Night 1 = Enable
50-06-02
Switching (for message
receiver side) Default is 1
CCIS Centralized Day/Night Send Group (1~16) Select the Remote Office to send Day/Night
Switching Sending Group Point Code (1~16367) Switching control message.
50-11-01
Assignment – Destination 0 = No Setting

Issue 5.0
Program/
Item No.
CCIS Centralized Day/Night Send Group (1~16) Select the Remote Office to send Day/Night
Switching Sending Group CCIS Route ID (0~8) Switching control message.
50-11-02
Assignment – CCIS Route 0 = No Setting
ID Default is 0
CCIS Centralized Day/Night Day Mode = Mode 1~8 Set the mode for Day/Night Switching
Mode to System Mode Default is 1 Synchronized Day/Night Group.
50-12-01 Assignment – Switching Night Mode = Mode 1~8
Synchronized Day/Night Default is 2
Mode Group
For Night Transfer Feature
Program/
Item No.
Night Mode Function Setup 0 = Off Allow user to activate Night Service by dialing a
12-01-01 – Manual Night Mode 1 = On service code.
Switching Default is 1
Automatic Night Service Night Mode Group (1~32) Set to define the daily pattern for the auto night
Patterns Time Pattern Number mode switch setting.
(01~10)
Set Time Number
12-02-01
(01~20)
Default = All Groups, All
patterns: 00:00~00.00 for
Mode 1
Weekly Night Service Night Mode Service Set to define a weekly schedule of night switch
Switching Group Number = 01~32 settings.
Default:
01 Sunday = Pattern 2
02 Monday = Pattern 1
12-03-01 03 Tuesday = Pattern 1
04 Wednesday = Pattern
1
05 Thursday = Pattern 1
06 Friday = Pattern 1
07 Saturday = Pattern 2
Holiday Night Service Night Mode Service Set to define a yearly schedule for holiday.
12-04-01 Switching Group Number = 1~32

Issue 5.0
Program/
Item No.
DSS Console Key Key Number (001~114) Customize key assignments for DSS Consoles
Assignment 00~99 = General 1~32.
Functional Level
30-03-01 *00~*99 = Appearance
Functional level
Default is extensions.
101~160
Class of Service Options 0 = Off
(Administrator Level) – 1 = On
20-07-01 Manual Night Service
Enabled Default is 0 for COS 1~15
Programmable Function 09 = Day/Night Mode Enter 0 to toggle night mode.

15-07-01
Keys Switch
Class of Service for 0~15 All Extensions are in Class 1. .(default)

20-06-01
Extensions

Issue 5.0
Centralized E911 – K-CCIS
FEATURE DESCRIPTION
This feature allows a remote system to transmit a Calling Party Number to the 911
Emergency System over a K-CCIS direct or tandem connection.
SYSTEM AVAILABILITY
Terminal Type:
All Stations
Required Components
GCD-CCTA
- OR -
GPZ-IPLE
To use this feature at any terminal:

1. Lift the handset, and wait for internal dial tone.
Dial 911.
- OR -
1. Dial 9 911.
SERVICE CONDITIONS
General:
 If you want to send your phone number via CCIS, please refer to Calling Party
Number (CPN) Presentation from Station – K-CCIS on page 4-41.
 The Calling Party Number (CPN) is sent only to the network when the remote
system accesses an ISDN – PRI trunk in the distant system and the ISDN –
PRI trunk has Calling Party Number (CPN) Presentation and Screening service
enabled from the network.
 If Program 21-01-10 is programmed with an entry other than 0, a call does not
have a talk path unless the user dials at least the number of digits entered in
this option when placing an out going call. This means that an entry of 4 or
higher in this program causes a problem when dialing 911. Since it is only a 3-
digit number, the call does not have a talk path preventing the emergency

Issue 5.0
dispatcher from hearing the caller. It is recommended that this option be kept at
its default setting of 0 to prevent any problems with dialing 911.
 The attendant receives a notification each time a co-worker dials an emergency
911 call. This notification is the co-worker name and number display optionally
accompanied by an audible alarm. Notification occurs regardless of whether
the attendant is idle or busy on a call. You can optionally extend this ability to
other supervisory extensions as well.
 The PRI provider must provision for the CPN used for E911. The CPN must be
within the allowable range. For more information please contact your local
ISDN provider regarding allowable ranges.
 Virtual Extensions notify the attendant with the stations name and number
when an emergency 911 call is originated from the Virtual Extension.
Restrictions:
 Centralized E911 (outgoing with CES-ID) is not supported.
 A maximum of 16 digits can be assigned as the Calling party Number (CPN) in
Program 21-13-01.
 CAMA trunks are not supported.
 If Virtual Extensions are used to make E911 calls, they provide the information
for the VE key.

 ISDN Compatibility
 Calling Party Number (CPN) Presentation from Station K-CCIS
 E911 Compatibility
 Automatic Route Selection

Issue 5.0

Item No. Selection
Class of Service Options 0=Off Determine if the ISDN calling line

20-08-13 (Outgoing Call Service) – ISDN 1=On identity presentation and screening
CLIP Default is 0 indicators are to be allowed.
ISDN Calling Party Number Up to 16 digits max. Assign each trunk a Calling Party
Setup for Trunks Number.
When a call is made by an extension
21-12-01 which does not have an Extension
Calling Number assigned
calling number for the ISDN trunk
defined in 21-12.
Basic Trunk Data Setup – Trunk- 0 = Disable (Off) Enable CPN information to pass through
to-Trunk Outgoing Caller ID 1 = Enable (On) the Tandem Office.
14-01-24 Through Mode
Default is 0

Issue 5.0
Dial Access to Attendant – K-CCIS
FEATURE DESCRIPTION
This feature allows a station user to call an Attendant by dialing a call code through
the K-CCIS network.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
To call an Attendant Position:

2. Dial 0 or the operator call code if it is different from 0 .
SERVICE CONDITIONS
General:
 The operator call code must be for an individual Attendant Access Code
number.
 When calling to a UNIVERGE SV9100 Attendant Position, the SV9100 sends
Operator to the call originator as the name.
 Shows Office Code if using Open Numbering Plan.
 If using an Open Numbering Plan, and a call is made to an UNIVERGE SV9100
Attendant Position, the operator office code is included with the name.
 When making a call from a UNIVERGE SV9100 Attendant Position across a
K-CCIS network, the Caller ID Name and Number display is the same as for a
station-to-station call.

Issue 5.0
 This feature is also available when the Attendant Console is in a NEAX2000

IVS2 or NEAX2400 in the CCIS network.
 When an UNIVERGE SV9100 station calls a NEAX Desk Console Attendant
Position, OPERATOR is displayed on the LCD during the incoming ring. If
using an Open Number Plan, the office code of the Desk Console is also
displayed.
 In a Closed Numbering Plan network, a station can call an Attendant in the
K-CCIS network by dialing 0.
 In an Open Numbering Plan network, a station user can call an Attendant within
the K-CCIS network by dialing: Access Code + Office Code + 0.
Restrictions:
 When a PBX is in Night Mode, calls to a NEAX Desk Console are restricted.
When an UNIVERGE SV9100 station calls a NEAX Desk Console Attendant
Position that is set to Night Mode, ERROR is displayed in the calling station
LCD and the call is rejected.
 When a NEAX Desk Console Attendant Position calls an UNIVERGE SV9100
station, the UNIVERGE SV9100 station does not store the call in the Caller ID
Scrolling feature. This call record can be printed on SMDR without Caller ID
information.
 Operator Calling, PRG 20-14-05, does not keep a Tie Line caller from dialing 0
for the operator.

 Attendant Positions
 Centralized Day/Night Mode Change – K-CCIS
 Voice Calls – K-CCIS
For Main System
Program/
Item No.
11-01-01 System Numbering Default is Refer to UNIVERGE SV9100 Programming

0 = 1 digit Intercom Manual for all options and default settings.

Issue 5.0
Program/
Item No.
20-17-01 Operator Extension – Up to eight digits Define extension numbers that are used as
Operator Extension Default is 200 . operators. Assign only in KTS-to-KTS network.
Number
For Remote System
Program/
Item No.
11-01-01 System Numbering Default is Refer to UNIVERGE SV9100 Programming

0 = 1 digit; F-Route Manual for all options and default settings.
44-02-01 Dial analysis Table for ARS/ Analysis Table 1 = 0 Assign access code (up to eight digits) to dial the
F-Route Access – Dial Attendant.
Default is No Setting.
44-02-02 Dial analysis Table for ARS/ 0 = No Setting (None) Assign the service type to be used.
F-Route Access – Service 1 = Extension Call (Own)
(F-Route)
3 = Dial Extension
Analyze Table
(Option)
Default is 0
44-02-03 Dial analysis Table for ARS/ 0 = No Setting Enter additional data for Service Type selected in
F-Route Access – 1 = Delete Digit = 0~255 PRG 44-02-02.
Additional Data (255: Delete all digits)
2 = 0~500
3 = Dial Extension
Analyze Table
Number (0~4)
Default is 0
44-05-01 ARS/F-Route Table –Trunk 0-100,101-150,255 Select trunk group number for outgoing calls.
Group Number 0 = Not Set
1~100 = Trunk Group
from PRG14-05
Default is 0
44-05-02 ARS/F-Route Table –Delete 0 = No Setting Enter number of digits to delete from the dialed
Digits 0~255 number.
(255 = Delete all digits)
Default is 0
44-05-03 ARS/F-Route Table – 0~1000 Enter Table Number defined in PRG 44-06.
Additional Dial Number
Table Default is 0

Issue 5.0
Program/
Item No.
44-05-09 ARS/F-Route Table – 0~24 Assign Max. digits for Call Park Retrieve Access
Maximum Digit Default is 0 Code.
44-05-10 ARS/F-Route Table – CCIS 0~16367 Assign remote IP Destination Point Code.
over IP Destination Point
Code Default is 0
Programming Example (Dest PC)
T1 T1
UNIVERGE SV9100 UNIVERGE CCIS Electra Elite
CCIS IPK II
SV9100
GCD-CCTATI GCD- DTI DTI

Cabinet 1 CCTATI Cabinet Cabinet 1
Slot 1 TG10 TG10 Cabinet 1 TG11 TG10
CCIS 1 Slot 4 CCIS Slot 1
Slot 1
System A System C
(101s) System B (301s)
(201s)
 
(Attendant)
200

Program 11-01-01 Program 11-01-01 Program 11-01-01

0 = 1 digits; Intercom 0 = 1 digits; F-Route 0 = 1 digits; F-Route
Operator1 = Oper.Ext.No. 200 Table 1 =Dial 0 Table 1 =Dial 0
(Dial Attendant System A) (Dial Attendant System A)
Program 44-02-02 Program 44-02-02
Table 1 = F-Route Table 1 = F-Route
Table 1 = F-Route 1 Table 1 = F-Route 1
Table 1 = TRK GP 10 Table 1 = TRK GP 10
Table 1 = Delete Digits 0 Table 1 = Delete Digits 0
Table 1 = Add Dial 0 Table 1 = Add Dial 0

Issue 5.0

Table 1 = Max Digit 1 Table 1 = Max Digit 1
* Program 44-05-10 *Program 44-05-10
F-Route Table 1 DPC = 1 F-Route Table 1 DPC = 1
* For CCISoIP Programming only.

Issue 5.0
Direct Inward Dialing – K-CCIS
FEATURE DESCRIPTION
This feature allows an incoming DID call (centralized DID) to be routed directly
across a K-CCIS link to reach a station in the remote system without Attendant
assistance.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
All Stations
GCD-CCTA
- OR -
GPZ-IPLE
SERVICE CONDITIONS
General:
 Call billing to the outside party starts when the incoming call connects to the
K-CCIS trunk.
 When an incoming DID call from the CD-PRTA card with Caller ID information
is transferred to the station in K-CCIS network, the Caller ID Name and Number
follow across the K-CCIS network to the distant system.
 This feature is supported when a Closed Numbering Plan or Open Numbering
is used.
 The UNIVERGE SV9100 system supports DID Digit Conversion when using
station numbers with 2~8 digits.
 An extension on a remote system can be the destination for the DID Received
Vacant Number Operation Assignment (Program 22-09-02).

Issue 5.0
Restrictions:
 Program 20-02-15 (Caller ID Display Mode) must be set to 0 to display the DID
Name on incoming DID calls.
Refer to the Key-Common Channel Interoffice Signaling (K-CCIS) feature for more
details related to Single Line Telephone and IP (K-CCIS) support.

 Key-Common Channel Interoffice Signaling (K-CCIS)
 Flexible Numbering of Stations – K-CCIS
Program/
Item No.
22-09-01 DID Basic data Setup – 1~8 Assign number of digits the Table expects to
Expected Number of Digits receive from Telco. Use this program to make the
Default is 4 system compatible with three- and four- digit DID
service.
22-10-01 DID Translation Table Setup 0 = No setting Program the Translation Table size.
– Conversion Table Area 1~2000
Number Default is 0
22-11-01 DID Translation Number Maximum eight digits Assign the received number to the Conversion
Conversion – Received Table number.
Number Default is No Setting
22-11-02 DID Translation Number Maximum 24 digits Assign the destination extension based on the
Conversion – Target digits received.
Number Default is No Setting.
22-11-03 DID Translation Number Maximum 12 digits Assign the DID Name based on the digits received.
Conversion – DID Name

Issue 5.0
Dual Hold – K-CCIS
FEATURE DESCRIPTION
This feature allows two connected Multiline Telephones to be placed on hold

simultaneously over the K-CCIS link. This enables the held parties to answer or
originate a call from a secondary line or intercom path.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
SERVICE CONDITIONS
General:
 This feature is available for interoffice calls through K-CCIS.
 Both Non-Exclusive Hold and Exclusive Hold can be used for Dual
Hold – K-CCIS.
 The K-CCIS call is held on a Call Appearance key.
Restrictions:
None


Issue 5.0
Program/
Item No.
24-01-01 System Options for Hold – 0~64800 seconds A call on Hold recalls to the extension that placed
Hold Recall Time Default is 90 the call on hold after this time expires.
24-01-03 System Options for Hold – 0~64800 seconds A call left on Exclusive Hold recalls to the extension
Exclusive Hold Recall Time that placed it on hold after this time expires.
Default is 90
20-29-01 Timer Class for Extensions 0~15 Assign Timer Class (0~16) to each extension for
– Day/Night Mode 1~8, 0 = Not Assigned night mode. Virtual extension numbers are
Class Number Default is 0 included.
20-30-01 Timer Class for Trunks – 0~15, #,  Assign Timer Class (0~16) to each trunk for night
Day/Night Mode 1~8, Class 0 = Not Assigned mode.
Number Default is 0
20-31-01~23 Timer Class Timer Refer to Flexible Assign times. These timers are referred when a
Assignment Timeouts in the class is set to any number from 1 to 16 in PRG
UNIVERGE SV9100 20-29-01/20-30-01.
Features and
Specifications Manual for
more Flexible Time
details.

Issue 5.0
Elapsed Time Display – K-CCIS
FEATURE DESCRIPTION
This feature provides an Elapsed Call Time on the LCD which shows the duration of
time that a multiline terminal is connected to any call through the K-CCIS network.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
All Display multiline terminals
GCD-CCTA
- OR -
GPZ-IPLE
No manual operation is required.
SERVICE CONDITIONS
General:
 When a call is retrieved from Exclusive Hold and/or Non-Exclusive Hold from
the same station, the elapsed call timer begins at 0.
 When a call is transferred, the elapsed time of the party receiving the transfer
begins at zero.
Restrictions:
 For calls across a K-CCIS link, the Elapsed Call timer begins only after
receiving answer supervision from the distant system.
 For Voice Calls across the K-CCIS link, the Elapsed Call timer does not begin
until the distant station answers.
 For conference calls established across a K-CCIS link, the elapsed call timer
does not start during an active conference call.

Issue 5.0

Program/
Item No.
20-13-36 Class of Service Options 0 = Off Turn off the extension Call Timer. The system
(Supplementary Service) – 1 = On waits for the interdigit time (21-01-01) before this
Call Duration Timer Display Default is 1 time begins.
20-09-06 Class of Service Options 0 = Off Turn on the Incoming Time and Date display on the
(Incoming Call Service) – 1 = On LCD while the Telephone is ringing.
Incoming Time Display Default is 0

Issue 5.0
Flexible Numbering of Stations – K-CCIS
FEATURE DESCRIPTION
This feature allows telephone numbers to be assigned to any stations in the K-CCIS
network, based solely upon numbering plan limitations.
Station numbers can be assigned by the 10's group for 4-digit station numbers, 100's
group for 5-digit, 1,000's group for 6-digit station numbers, and 10,000's group for
7-digit station numbers.
Example:
Station Numbering Plan Site A Site B Site C
4-digit station numbers 1000 ~ 1009 1010 ~ 1019 1020 ~ 1029

8-digit station numbers 10000000 ~ 10000010 ~ 10000020 ~
10000009 10000019 10000029
Manual.
SYSTEM AVAILABILITY
Terminal Type:
All Stations
GCD-CCTA
- OR -
GPZ-IPLE

Issue 5.0
SERVICE CONDITIONS
General:
 Give careful consideration to the network numbering plan to avoid needless
loss of Access Codes or duplication of telephone numbers.
 The first digit or first two digits of a telephone number distinguishes one system
from another system.
 Station Numbering Plan can have 2~8 digits.
Restrictions:
 Tenant service is not provided, i.e., numbers cannot be duplicated for different
tenants.
 Extension numbers should not start with 0,9,* or #.
For non-K-CCIS feature support, refer to the UNIVERGE SV9100 Features and
Specifications Manual, Flexible Numbering Plan feature.

Program/
Item No.
11-11-01 System Numbering Default is 1 = 3-digit; Refer to the UNIVERGE SV9100 Programming
Intercom Manual for all options and default settings.
11-02-01 Extension Numbering –Dial Default : Assign up to eight digits for Extension Numbers.
(up to eight digits) Port 1 ~ 300 = 200 ~ 499
Port 301 ~ 960 = 5000 ~
5659

Issue 5.0
Program/
Item No.
11-20-01 Dial Extension Analyze Valid entries:

Table – Dial (up to eight 0, 1~9, #, 
digits)
Use tables 01~128 to
assign the digits to be
dialed using the Dial
Extension Analyze Tables.
These tables are used
when Program 11-01-01 is
set to option 9 = Dial
Extension Analyze.
(Up to eight digits can be Default is not assigned.
assigned with valid entries:
0, 1~9, #, ).
11-20-02 Dial Extension Analyze 0 = Not used
Table – Type of Dials 1 = Service Code
2 = Intercom
Assign the type of dial for 5 = Operator
the Dial Extension Analyze 6 = F-Route
Table from Program 11-20-
01.
(Svc Code, Intercom,
operator, or F-Route) Default is Not Set.

Issue 5.0
Handsfree Answerback – K-CCIS
FEATURE DESCRIPTION
This feature allows Multiline Telephone station users to respond to voice calls
through a K-CCIS network without lifting the handset.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
To turn the microphone On/Off:

1. Press Feature.
2. Dial 1.
- OR -
Press the programmable line key assigned as the MIC On/Off key.
SERVICE CONDITIONS
Restrictions
 Handsfree Answerback – (K-CCIS) can be used only when responding to Voice
Calls – (K-CCIS) from a remote user.
 After a user changes ring back tone to voice call, it cannot be changed back to
ringing.
 Voice Call cannot be set as the initial call across K-CCIS. The initial call must
be a ringing call.

Issue 5.0

 Voice Calls – K-CCIS
Program/
Item No.
11-16-03 Single Digit Service code Default is 1 Customize the one-digit Service Code used when a
Setup – Switching of Voice/ busy or ring back signal is heard.
Signal Call
11-12-06 Service Code Setup (for Default is 812 Toggle an ICM call between Handsfree
Service Access) – Answerback or Forced Intercom Ringing for
Switching of Voice Call and outgoing Intercom Calls.
Signal Call
20-08-10 Class of Service Options 0 = Off Turn Off or On the ability to force Handsfree
(Outgoing Call service) – 1 = On Answerback or Forced Intercom Ringing for
Signal/Voice Call Default is 1 outgoing Intercom Call on/off.
20-08-11 Class of Service Options 0 = Off When extension is set for ICM calls, enable this
(Outgoing Call service) – 1 = On option to prevent callers from changing to voice
Protect for the Call Mode announce mode.
Switching from Caller
(Internal Call) Default is 0

Issue 5.0
Hot Line – K-CCIS
FEATURE DESCRIPTION
This feature allows two stations at different nodes in the K-CCIS network to be
mutually associated on automatic ringdown through the K-CCIS network.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
To execute at any station programmed for Hot Line:

The remote K-CCIS station is called.
SERVICE CONDITIONS
General:
 Any multiline terminal (a maximum number of 896 stations) can be assigned for
Hot Line – (K-CCIS).
 Either multiline terminal in a Hot Line – (K-CCIS) pair may transfer a Hot Line
call to another station in the K-CCIS network using the Call Transfer – All Calls
- (K-CCIS) feature.
Restrictions:
None

Issue 5.0


Program/Item Description/
No. Selection
Basic Extension Data Setup – 0 = Off If enabled, the extension user gets
15-01-02 Outgoing Trunk Line Preference 1 = On trunk dial tone when the handset is
Default is 0 lifted.
Class of Service for Extensions 1 ~ 15 Assign a Class of Service to an
20-06-01 Default: All Extensions are in extension
Class 1.
Class of Service Options 0 = Off Turn Off or On Ringdown Extension
20-08-09 (Outgoing Call Service) – Hotline/ 1 = On for extensions with this COS.
Extension Ringdown Default is 0
Class of Service Options 0 = Off When Hotline is programmed and 20-
(Outgoing Call Service) – Hotline 1 = On 08-19 is turned ON (1), the user can
20-08-19
for SPK press the speaker button and the
Default is 0 Hotline destination is dialed.
System Options for Outgoing Calls 0~64800 seconds A Ringdown extension automatically
21-01-09 – Ringdown Extension Timer calls its programmed destination after
(Hotline Start) Default is 5 this time.
Extension Ringdown (Hotline) Maximum 24 digits Define the Hotline Destination number
Assignment – Hotline Destination for each extension number.
Number (0~9, , #, Pause, Hook
21-11-01 Flash, and @(code to wait
for answer supervision)

Issue 5.0
Link Reconnect – K-CCIS
FEATURE DESCRIPTION
This feature provides the system that is connected to a K-CCIS network with the
ability to release the redundant K-CCIS link connections and reconnect the link with
the system for efficient usage of the K-CCIS trunks.
SYSTEM AVAILABILITY
Terminal Type:
Required Components
GPZ-IPLE
SERVICE CONDITIONS
General:
 The link reconnect ability is provided for a station call over K-CCIS that is
transferred or forwarded to another station or trunk in the same office as the call
originating station. (Refer to Figure 4-1 Link Reconnect for Station Calls.)

Issue 5.0
Figure 4-1 Link Reconnect for Station Calls
System A 1 3 System B
4
2
101 102 201
System A System B
After Reconnect
Before Reconnect
201
100 101
Stations 101, 102, and 201 must be multiline terminals.
 When Station A holds the call or is in the conference, Link Reconnect is not provided.
 A trunk call (CO/PBX/TIE/DID/K-CCIS) over a K-CCIS network is transferred or

forwarded to another station or trunk within the same office as the original
incoming trunk. (Refer to Figure 4-2 Link Reconnect for Trunk Tandem Calls.)

Issue 5.0
Figure 4-2 Link Reconnect for Trunk Tandem Calls
Call Forward to CO Line
PSTN
3
System A 2 System B
Call Forward to
External Party
Voice Path after Call Forwarding
PSTN
System A System B
Call Forward to
External Party
After Reconnect
Before Reconnect
 Link reconnect occurs after answering a transferred or forwarded K-CCIS call.
Restrictions:
 Answer supervision is required for Link Reconnect to occur. For outgoing calls
on analog trunks, Answer supervision is based on the Elapsed Call Time -
Program 21-01-03 (Trunk Interdigit Time).
 When a call is on hold, or in a conference, and is transferred back across the
K-CCIS link, Link Reconnect is not provided.

Issue 5.0
 When Connecting a SV9100 to a NEAX PBX, Link reconnect needs to be

turned off in the PBX to the SV9100.

 Call Forwarding – All Calls – K-CCIS
Program/
Item No.

Default is 1
21-01-03 System Options for 0 = Disabled The system waits for this time to expire before
Outgoing Calls – Trunk 1~64800 seconds placing the call (Call time starts after this time
Interdigit Time (External) Default is 10 expires).

Issue 5.0
Multiple Call Forwarding – All Calls – K-CCIS
FEATURE DESCRIPTION
This feature allows a Multiple Call Forwarding – All Calls sequence to be forwarded
over a K-CCIS network to a station in another office.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
To set Call Forward – All Calls – K-CCIS from a Multiline Telephone (Closed
Numbering Plan):
2. Dial 1 to set. Then enter the K-CCIS station number.
3. Press Speaker.
- OR -

2. Dial Access Code 848 (set as default). Then Dial 1 to set.

Issue 5.0
To set Call Forward – All Calls – K-CCIS from a Multiline Telephone (Open
Numbering Plan):
1. Press the Call Forward – All Call ON/OFF key, and Dial 1 to set.
5. Press Speaker.
- OR -

2. Dial Access Code 848 (set as default), and Dial 1 to set.
To cancel Call Forward – All Calls – K-CCIS from a Multiline Telephone:

1. Press Call Forward – All Call On/Off key.
3. Press Speaker.
- OR -

2. Dial Access Code 848 (set as default), and Dial 0 to cancel.
SERVICE CONDITIONS
General:
 Multiple Call Forwarding – All Calls – K-CCIS can forward a call up to seven
times across K-CCIS links (up to seven hops) depending on system data.
 Multiple Call Forwarding over a K-CCIS link is combined with Multiple Call
Forwarding – All Calls/Busy/No Answer.
 If the calling station is set as the destination in a multiple hop scenario, Multiple
Call Forwarding – All Calls – K-CCIS is not performed (i.e., an infinite loop does
not occur).

Issue 5.0
 For multiple Call Forwarding – All Calls/Busy (Immediate) calls, the display on
the calling party Multiline Telephone displays the terminating station user name
and the station number for the first station of a distant system in the Multiple
Call Forwarding group. For the terminating station, the telephone display
indicates the name and the number of the calling party and the trunk number of
the incoming call.
 When a calling station has been Call Forwarding – All Calls – K-CCIS to the
maximum times assigned in Program 50-05-01 (K-CCIS Maximum Call
Forwarding Hop Assignment) and encounters another Call Forwarding – All
Calls – K-CCIS condition, the calling station is not forwarded and rings at the
last destination.
 If the destination station in a Multiple Call Forwarding – All Calls – K-CCIS
situation is busy and has not set Call Forwarding – Busy and has Call Alert
Notification disabled, the calling party receives busy tone.
 When combining Call Forwarding – Busy and Call Forwarding – All Calls
– K-CCIS, if the destination station is busy and has Call Alert Notification
disabled, the calling party hears busy tone after the maximum hops assigned in
Program 50-05-01 (K-CCIS Maximum Call Forwarding Hop Assignment).
 Multiple Call Forwarding – All Calls -K-CCIS and Call Forwarding – Busy
– K-CCIS may be mixed; up to seven combined multiple forwardings may
occur.
 An example of Multiple Call Forwarding over a K-CCIS link is shown in
Figure 4-3 Multiple Call Forwarding over K-CCIS Links for All Calls.

Issue 5.0
Figure 4-3 Multiple Call Forwarding over K-CCIS Links for All Calls
Offic Offic Offic Offic
K- K- K-
Coun Coun Coun


   
Number of Call
Forwards over
Multiple Call Forwarding K-CCIS = 2
Over K-CCIS
Office Allowed
A
Office Allowed
B
Office Allowed
C
Office Allowed
D
 The counter is reduced by one with each hop (tandem connection).


Issue 5.0
Program/Item
No.
20-06-01 Class of Service for 0~15 Default: All Extensions are in Class 1. .
Extensions
20-09-07 Class of Service Options 0 = Off Must be Off for Class of Service for Call Forward
(Incoming Call Service) – 1 = On Busy to operate.
20-11-12 Class of Service Options 0 = Off Enable Call Forward – Off-Premise per Class of
(Hold/Transfer Service) – 1 = On Service.
Call Forwarding Off
Premise (External Call
Automatic Off Hook

Keys Immediate 848
12 = Call Forward – No
Answer 845
13 = Call Forward – B/NA 844
14-01-13 Basic Trunk Data Setup – 0 = Disable Must be enabled for Trunk-to-Trunk Transfer;
Trunk-to-Trunk Transfer 1 = Enable Call Forward – Off- Premise, or Tandem
Trunking.
Default is 1
50-05-01 CCIS Maximum Call 0~7 Hops Set Maximum Hops.

Issue 5.0
Multiple Call Forwarding – Busy/No Answer - K-CCIS
FEATURE DESCRIPTION
This feature allows a Multiple Call Forwarding – Busy/No Answer sequence to be

forwarded over a K-CCIS network to a station in another office.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
To set Call Forward – Busy/No Answer - K-CCIS from a Multiline Telephone (Closed
Numbering Plan):
2. Dial 1 to set.
4. Press Speaker.
- OR -


Issue 5.0
To set Call Forward – Busy/No Answer - K-CCIS from a Multiline Telephone (Open
Numbering Plan):
2. Dial 1 to set.
6. Press Speaker.
- OR -

2. Dial Access Code 844 (default), then dial 1 to set.
To cancel Call Forward – Busy/No Answer – K-CCIS from a Multiline Telephone:

1. Press Call Forward – Busy/No Answer On/Off key.
3. Press Speaker.
- OR -

2. Dial Access Code 844 (default), then Dial 0 to cancel.
To set for any station for Attendant Positions only (Closed Numbering Plan):
2. Dial the Call Forward Busy/No Answer for any Extension to Destination
Service Code (default: 794 ).
3. Dial 1 (Set).
4. Dial the extension number to be forwarded and then the destination number.

Issue 5.0
To cancel for any station for Attendant Positions only:

1. Pick up the handset or press Speaker.
2. Dial the Call Forward Busy/No Answer for any Extension to Destination
Service Code (default: 794 ).
3. Dial 0 (Cancel).
4. Dial the station number, which is forwarded.
SERVICE CONDITIONS
General:
 Multiple Call Forwarding – Busy/No Answer Calls - K-CCIS can forward a call
up to seven times across K-CCIS links (up to seven hops) depending on
systems data.
 Multiple Call Forwarding over a K-CCIS link is combined with Multiple Call
Forwarding – All Calls/Busy/No Answer.
 If the calling station is set as the destination in a multiple hop scenario, Multiple
Call Forwarding – Busy/No Answer Calls - K-CCIS are not performed, i.e., an
infinite loop does not occur.
 For multiple Call Forwarding – All/Busy (Immediate) calls, the display on the
calling party’s Multiline Telephone indicates the terminating station user’s name
and the station number for the first station of a distant system in the Multiple
Call Forwarding group. For the terminating station, the telephone display
indicates the name and the number of the calling party and the trunk number of
the incoming call.
 For multiple Call Forwarding – No Answer/Busy (Delay) calls, the display on the
calling party’s Multiline Telephone indicates the name and number of the first
station of a distant systems in the Multiple Call Forwarding group. For the
terminating station, the telephone display indicates the name and the number
of the calling party and the trunk number of the incoming call.
 When a calling station has been Call Forwarding – Busy/No Answer Calls –
K-CCIS to the maximum times assigned in Program 50-05-01 (K-CCIS
Maximum Call Forwarding Hop Assignment) and encounters another Call
Forwarding – Busy/No Answer Calls – K-CCIS condition, the calling station is
not forwarded and rings at the last destination.
 When the destination station in a Multiple Call Forwarding – Busy/No Answer
Calls – K-CCIS situation is busy and has not set Call Forwarding – Busy and
has Call Alert Notification disabled, the calling party receives busy tone.
 When combining Call Forwarding – Busy and Call Forwarding – All Calls –
K-CCIS and the destination station is busy and has Call Alert Notification
disabled, the calling party hears a busy tone after the maximum hops assigned

Issue 5.0
in Program 50-05-01 (K-CCIS Maximum Call Forwarding Hop Assignment).

 Multiple Call Forwarding – All Calls – K-CCIS can forward a call a maximum of
seven times across K-CCIS link (maximum of seven hops) depending on
system data assignments.
 An example of Multiple Call Forwarding over a K-CCIS link as shown Figure 4-4
Multiple Call Forwarding over K-CCIS Links for Busy/No Answer.
Figure 4-4 Multiple Call Forwarding over K-CCIS Links for Busy/No Answer
Offic Offic Offic Offic
K- K- K-
Coun Coun Coun 
   
Number of Call Forwards
over K-CCIS = 2
Multiple Call Forwarding Over K-CCIS
Office A Allowed
Office B Allowed
Office C Allowed
Office D Allowed
 The counter is reduced by one with each hop (tandem connection).
Restrictions:
 Trunk-to-Trunk Transfer must be allowed in Program 20-11-14 [Class of Service
Options (Hold/Transfer Service) Trunk-to-Trunk Transfer Restriction].


Issue 5.0
Program/Item
No.
20-06-01 Class of Service for 0~15 Default: All Extensions are in Class 1. .
Extensions
20-09-07 Class of Service Options 0 = Off Must be Off for Class of Service for Call Forward
(Incoming Call Service) – 1 = On Busy to operate.
20-11-12 Class of Service Options 0 = Off Enable Call Forward – Off-Premise per Class of
(Incoming Call Service) – 1 = On Service.
Call Forwarding Off
Forwarding) Default is 0
20-11-14 Class of Service Options 0 = Off Turn Off or On the Trunk-to-Trunk Transfer
(Incoming Call Service) – 1 = On Restriction. If enabled (turned on),
Trunk-to-Trunk Transfer Trunk-to-Trunk Transfer is not possible.
Restriction Default is 0
Automatic Off Hook

Keys Immediate 848
12 = Call Forward – No
Answer 845
13 = Call Forward –B/NA 844
14-01-13 Basic Trunk Data Setup – 0 = Disable (No) Must be enabled for Trunk-to-Trunk Transfer;
Trunk-to- 1 = Enable (Yes) Call Forward – Off- Premise, or Tandem
Trunk Transfer Trunking.

Default is 1
50-05-01 CCIS Maximum Call 0~7 Hops Set Maximum Hops.

Issue 5.0
Paging Access – K-CCIS
FEATURE DESCRIPTION
This feature allows users to access internal or external paging from remote sites
across the K-CCIS network. Local stations where the external paging equipment is
installed can use the Meet-Me Answer feature to answer the page and establish a
station-to-station K-CCIS call.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
To access internal or external paging across a K-CCIS network:

2. Dial the Access Code for the required zone, or press the programmed
Feature Access or One-Touch key.
SERVICE CONDITIONS
General:
 The single external paging zone output built into the basic B64-U20 KSU can
be used for Paging Access – (K-CCIS).
 Program 31-01-02 (Paging Announcement Duration) applies to
Paging Access – (K-CCIS).
 If a user dials during Paging Access – (K-CCIS), DTMF tones are heard from
the external paging equipment at the remote site.

Issue 5.0
 Program 31-02-01 (Internal Paging Group Number) applies to

 Program 31-02-02 (Internal All Call Paging Receiving) applies to
Restrictions:
 Amplifiers and speakers must be locally provided.
 Combined Paging is not supported over K-CCIS.
 Internal Paging across K-CCIS is supported only between UNIVERGE SV9100
and UNIVERGE SV9100.

 Background Music Over External Speakers
 External Zone Paging (Meet-Me)
 Internal Zone Paging (Meet-Me)
For Paging Installation
Program/
Item No.
10-03-01 ETU Setup – Terminal Type 0 = Not Set Assign ESI port for External Paging.
(B1) 1 = Multiline Terminal
3 = Bluetooth Cordless
Handset
6 = PGD (Paging)
7 = PGD (Tone Ringer)
8 = PGD (Door Box)
9 = PGD (ACI)
10 = DSS Console
11 = ---Not Used---
11-01-01 System Numbering Default is 51 = 2 digit: F- This Service Code must be assigned also in PRG
Route. 11-12-20.
11-12-20 Service Code Setup (for Default is 803 Setting 703 should be changed based on the
Service Access) – External K-CCIS network configuration.
Paging

Issue 5.0
Program/
Item No.
11-12-22 Service Code Setup Default is 865 The Service Code assigned in this Program is used
(Service Access) – Meet for Meet Me Answer to External Paging.
Me Answer to External
Paging
15-07-01 Programmable Function 19 = External. Group Set the functions of programmable extension
Keys Paging 1~8 Function Keys.
20 = External. All Call
Paging
21 = Internal. Group
Paging 1~64
22 = Internal. All Call
Paging
20-10-06 Class of Service Options 0 = Off Enable Meet Me Conferencing and Paging.
(Answer Service) – Meet 1 = On
Me Conference and Paging
Default is 1
20-14-07 Class of Service Options for 0 = Off Allow DISA or Tie Line Trunk user to use External
DISA/E&M – External 1 = On Paging.
Paging Default is 1
31-01-01 System Options for Internal/ Up to 12 Characters Assign a name to each All Call Internal Paging
External Paging –All Call Default is Group All Zone. The name is displayed to make the
Paging Zone Name announcement.
31-01-02 System Options for Internal/ 0~64800 seconds Set the length of Paging Announcements.
External Paging –Page
Announcement Duration
Default is 1200
31-01-04 System Options for Internal/ 0~64800 seconds After user initiates a Meet Me or Voice Call
External Paging –Privacy conference, the system waits this time for the
Release Time Default is 90 Paged Party to answer the call.
31-04-01 External Paging Zone 0 = No Setting Assign each External Paging Zone to an External
Group Default: Paging Group.
Speaker 1 = Group 1
Speaker 2 = Group 2
Speaker 3 = Group 3
Speaker 4 = Group 4
Speaker 5 = Group 5
Speaker 6 = Group 6
Speaker 7 = Group 7
Speaker 8 = Group 8
Speaker 9 = CPU Group
1

Issue 5.0
Program/
Item No.
31-06-01 External Speaker Control – 0 = No Tone Enable Splash Tone before paging.
Broadcast Splash Tone 1 = Splash Tone
before Paging (Paging Start 2 = Chime Tone
Tone) Default is 2
31-06-02 External Speaker Control – 0 = No Tone Enable Splash Tone after paging.
Broadcast Splash Tone 1 = Splash Tone
After Paging (Paging End 2 = Chime Tone
Time Default is 2
31-06-03 External Speaker Control – 0 = Both Ways (Duplex) Establish whether or not the external speaker is
Speech Path 1 = One Way used for talkback.
PGD  SPK (Simplex) (Not Available on CPU page port 9)
Default is 1
For Remote System
Program/
Item No.
11-01-01 System Numbering Default is 51 = 2 digit: F- Assign also in PRG 11-12-20.

Route
11-12-20 Service Code Setup (for Default is 803 Change 803 Setting based on the K-CCIS network
Service Access) – External configuration.
Paging
44-02-01 Dial Analysis Table for ARS/ Analysis Table 1 = 511 Assign the access code used to page.
F-Route Access) – Dial Up to eight digits can be assigned.
44-02-02 Dial Analysis Table (ARS/F- 0= No Setting (None) Assign Service Type.
Route Access) – Service 1 = Extension Call (Own)
(F-Route)
3 = Dial Extension
Analyze Table
(Option)
Default is 0
44-02-03 Dial Analysis Table (ARS/F- 0= No Setting For the Service Type selected in 44-02-02, enter
Route Access) –Additional 1 = Delete Digits 0~255 the additional data required.
data (255: Delete All
Digits)
2 = 0~500
3 = Dial Extension
Analyze Table
Number (0~4)
Default is 0

Issue 5.0
Program/
Item No.
44-05-01 ARS/F-Route Table – Trunk 0 = Not Set Select trunk group number used for outgoing ARS
Group Number 1 ~ 100 = Trunk Group calls.
from PRG14-05 Setting 255 = Internal Extension Call.
101 ~ 150 Networking
Default is 0
44-05-02 ARS/F-Route Table – 0 = No setting Enter number of digits to delete from the dialed
Delete digits 0~255 number.
255 = Delete All Digits
Default is 0
44-05-03 ARS/F-Route Table – 0= No Setting Enter Table Number (defined in PRG 44-06) for
Additional Dial Number 0~1000 additional digits to dial.
Table Default is 0
44-05-09 ARS/F-Route Table – 0 = No Maximum Assign max digits for the Paging Access Code.
Maximum Digit 0~24
Default is 0

Issue 5.0
Programming Example (External)
Paging Access for System A = 511(0~9)
Paging Access for System B = 512(0~9)
System A (101s) System B (201s)
PRG 11-01-01 PRG 11-01-01

51 = 2 digits: F-Route 51 = 2 digits: F-Route
PRG 11-12-20 PRG 11-12-20
External Paging = 803 External Paging = 803
(Page System A) (Page System B)
PRG 44-02-01 PRG 44-02-01
Table 1 = Dial 511 Table 1 = Dial 512
PRG 44-02-01 PRG 44-02-01

(Page System B) (Page System A)
PRG 44-02-02 PRG 44-02-02
PRG 44-02-03 PRG 44-02-03
PRG 44-05-01 PRG 44-05-01
PRG 44-05-02 PRG 44-05-02

Issue 5.0
System A (101s) System B (201s)
PRG 44-05-03 PRG 44-05-03

PRG 44-05-09 PRG 44-05-09
PRG 44-06-01 PRG 44-06-01
Programming Example (Internal)

Paging Access for System A = 521
Paging Access for System B = 522
System A (101’s) System B (201’s)

52 = 2 digits; F-Route 52 = 2 digits; F-Route
Internal Paging = 801 Internal Paging = 801
Table 1 = Dial 521 Table 1 =Dial 522
(Page System B) (Page System A)

Issue 5.0
System A (101’s) System B (201’s)


Issue 5.0
Quick Transfer to Voice Mail – K-CCIS
FEATURE DESCRIPTION
A station user transferring a call can force the call to be transferred to the called party
voice mail box after the transferred call recalls, after an internal station number is
dialed while performing a screened transfer, or during intercom calls.
SYSTEM AVAILABILITY
Terminal Type:
All multiline terminals allow either operation.
Single line telephones may perform the Quick Transfer only during screened transfer
operations. They may not perform Quick Transfer after recall.
Required Components
GCD-CCTA
- OR -
GPZ-IPLE
Quick Transfer Across K-CCIS:
To Quick Transfer a call while talking with an outside or internal party:

1. Press Transfer, and wait for an internal dial tone.
2. Enter a station number, and wait for a ring back tone.
3. Dial the Quick Transfer Access Code (default: No Setting ). The outside party
is transferred to the station user Voice Mail box.
4. Hang up.
5. The Voice Mail answers.
To leave a message using Quick Transfer to voice mail during an intercom call:
1. Make the intercom call.
2. Dial the Quick Transfer Access Code (default: No Setting ).
3. Leave a voice mail message.
4. Hang up.

Issue 5.0
SERVICE CONDITIONS
General:
 The Quick Transfer to Voice Mail feature is allowed when:
 Listening to the ring back tone (RBT)
 Listening to the call waiting tone (CWT)
 In Handsfree Answerback Mode
 In Voice Over Mode
 This feature is allowed from a single line telephone (SLT) until the PBR times
out (default: 10 sec).
 An SLT may perform the Quick Transfer only during screened transfer
operations.
 The InMail is supported for centralized voice mail in a CCIS network.

 Digital Voice Mail
Program/
Item No.
11-16-09 Single Digit Service Code Assign access code for Quick Transfer to Voice
Setup – Access to Voice Mail.
Mail Default is No Setting
11-07-01 Department Group Pilot Tel Groups 1~64 Assign pilot number to each Department Group set
Numbers Dial up to eight digits. up in PRG 16-02. Extension numbers are assigned
Default is No Setting in PRGs 11-02, 11-04, 11-06, and 11-08.
15-03-01 Single Line Telephone 0 = DP Tell the system which dialing is used by the
Basic Data Setup – SLT 1 = DTMF connected telephone users. For each Voice Mail
Signaling Type Default is 1 extension this option must be 0.
15-03-03 Single Line Telephone 0 = Normal Enter 1 to allow a single line port to receive DTMF
Basic Data Setup – 1 = Special tones after the initial call setup.
Terminal Type Default is 0
45-01-01 Voice Mail Integration 0 = No Voice Mail Assign Department Group Number as the Voice
Options – Voice Mail 0~64 Groups Mail Group.
Department Group Number
Default is 0

Issue 5.0
Program/
Item No.
45-01-14 Voice Mail Integration Up to eight digits Assign the CCIS Centralized Voice Mail Pilot
Options – CCIS Centralized Default not assigned. Number for Remote Sites.
Voice Mail Number
Programming Example
T1 CCIS T1 CCIS
UNIVERGE UNIVERGE SV9100 Electra Elite IPK II
SV9100
Voice Mail DTI DTI DTI DTI

190 Cabinet 1 Cabinet 1 Cabinet 1 Cabinet 1
Slot 1 Slot 1 Slot 4 Slot 1
TG10 TG10
CCIS TG11 TG10
System A System B CCIS System C
(100s) (200s) (300s)
  

Issue 5.0
Station-to-Station Calling – K-CCIS
FEATURE DESCRIPTION
This feature permits any multiline terminal user to dial another multiline terminal
directly through a K-CCIS network.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
SERVICE CONDITIONS
General:
 If the called station is off-hook and has Call Queuing disabled, the originating
station receives a busy tone. If the called station is idle, the called station rings
and the caller hears ringback tone.
 If the called station is off-hook on a call and has Call Queuing enabled, the
originating station receives ringback tone and the called station receives call
alert tone.
 Station-to-Station Calling between tenants in the K-CCIS network is not
restricted.
 The release process is First Party Release.
Restrictions:
 The same telephone numbers cannot be duplicated in the same system.

Issue 5.0

 Dual Hold – K-CCIS
 Elapsed Time Display – K-CCIS
 Hands-Free Answerback – K-CCIS
This guide provides a list of associated Memory Blocks that support this feature.
Program/
Item No.
11-01-01 System Numbering – 1X = 3 Digit; Intercom Assign the system internal (intercom) numbering
Extension Number plan.
11-02-01 Extension Numbering Default: Port 1 ~ 300 = Assign the Extension Numbers.
200 ~ 499
Port 301 ~ 960 = 5000 ~
5659

Issue 5.0
Uniform Numbering Plan – K-CCIS
FEATURE DESCRIPTION
In a K-CCIS network, a Uniform Numbering Plan enables a multiline terminal user to

call any other multiline terminal in the network. Two types of numbering plans are
provided. In the first plan, the station user dials any telephone number from two to
eight digits. The location of the office is identified by the first digit or first two digits of
the telephone number. In the second plan, the station user dials a one-, two- or
three-digit office code and a telephone number from two to eight digits.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
To call a station at another office using Numbering Plan 1 (Closed Numbering Plan):
To call a station at another office using Numbering Plan 2 (Open Numbering Plan):

Issue 5.0
SERVICE CONDITIONS
General:
 In a closed numbering plan, the location of the office can be identified by the
first digit or first two digits of the telephone number.
 In an open numbering plan, each office in the K-CCIS network is assigned a
one-, two- or three-digit office code and each station in the office is assigned
telephone numbers from two to eight digits.
 In the same office, a station-to-station call is made by dialing the telephone
number of the desired station.
Restrictions:
 For a Closed Numbering Plan network, a maximum of 255 systems can be
connected per K-CCIS Network.
 When a Closed Numbering plan is used the extensions in the network cannot
have the same prefix number.
 For an Open Numbering Plan network, the Automatic Route Selection (ARS)/
Flexible Routing (F-Route) feature must be used to place Station-to-Station
calls over K-CCIS.
 When an Open Numbering plan is used, the extensions in the network can
have the same prefix number, however the office location number cannot be
the same.

Refer to System Data Programming <US Only>.

Issue 5.0
Voice Call – K -CCIS
FEATURE DESCRIPTION
This feature provides a voice path, through the K-CCIS network, between a MLT in
one office and a MLT in another office. This path is established from the calling party
to the called party built-in speaker. If the called party MIC is on, the called party can
converse hands-free.
SYSTEM AVAILABILITY
Terminal Type:
GCD-CCTA
- OR -
GPZ-IPLE
From one MLT to another MLT:

1. The originating MLT user dials the desired station number in a different office
and receives ring back tone.
2. Calling party dials 1. A signal tone is transmitted over the K-CCIS network to
the called party speaker.
3. The called party can press MIC, or press FEATURE and dial 1 (if the MIC
LED is not on) to allow two-way conversation with the calling party.
SERVICE CONDITIONS
General:
 The UNIVERGE SV9100 can assign a Feature Access/One Touch Button as a
Voice Call key. This performs the same operation as pressing 1.
 Any station in the same system can use Directed Call Pick Up to retrieve the
Voice Call over K-CCIS.
 When a Voice Call is sent to a station that is unable to receive voice
announcement, RST is displayed on the originator display.

Issue 5.0
 During Voice Call, the ICM Key is flashing (Red).
Restrictions:
 The calling party must wait for at least one ring back before Voice Call is
attempted.
 After the calling party changes ring back to Voice Call, it cannot be changed
back to tone.
 Voice Call cannot be set as the initial call across K-CCIS.
 Group Call Pick Up is not allowed to retrieve voice calls over K-CCIS.
 Single Line terminals can be used to originate a Voice Call Over K-CCIS.
However, they are not allowed to receive a voice call.

 Handsfree Answerback – K-CCIS
Program/
Item No.
11-12-06 Service Code Setup (for Assigns the access code used to toggle ICM call
Service Access) –Switching between Handsfree Answerback and Forced
Voice Call and Signal Call Intercom Ringing for Outgoing Intercom calls.
Default is 812

Issue 5.0
Voice Mail Integration – K-CCIS
FEATURE DESCRIPTION
This feature allows any station user in the K-CCIS network to use the Voice Mail
System (VMS) in another office in the K-CCIS network.
Manual.
SYSTEM AVAILABILITY
Terminal Type:
All Stations
REQUIRED COMPONENTS:
CD-VM00
GCD-CCTA
- OR -
GPZ-IPLE
To access voice mail from a Multiline Telephone in the Main system:

2. Dial pilot number for voice mail.
3. When voice mail answers use softkeys to navigate.
- OR -
4. Wait for softkeys to time out and listen to voice prompts to navigate.
5. When finished hang up.
To access voice mail from a Multiline Telephone in the Remote system:

2. Dial extension number for voice mail.
3. When voice mail answers listen to voice prompts to navigate.

Issue 5.0
1. When finished hang up.
To program a One-Touch/Feature Access key for easy message access:

1. Press Feature.
2. Dial 851 .
3. Press One-Touch/Feature Access key.
4. Dial 1, followed by Voice Mail extension number.
5. Press Hold.
SERVICE CONDITIONS
General:
 Any station or Call Arrival (CAR) key can be set for Call Forwarding – Busy/No
Answer to voice mail.
 The following features are supported for voice mail users in remote systems:
 Message Waiting Indication
 Automated Attendant
 Auto Login
 Call Forward – Busy/No Answer to voice mail
 Call Forward – All Call to voice mail
 A voice mail with at least eight ports should be used in any K-CCIS system with
a shared voice mail.
 For InMail remote CCIS extensions are not supported in a centralized directory.
Restrictions:
 In the voice mail, only release transfer type is supported for mail boxes of
stations in Remote systems.
 In a KTS to KTS Network, only digital voice mails are supported for K-CCIS.

Issue 5.0
 In a KTS to KTS network, Centralized Voice Mail is supported only via closed
numbering plan and only up to 7-digit station numbers.
 In a PBX to KTS network, Centralized Voice Mail is supported only via closed
numbering plan.
 In a PBX to KTS network, Centralized Voice Mail is supported using the PBX
voice mail.
 When a call is forwarded to voice mail by multiple call forwarding, the message
is left in the mailbox of the first forwarded station.
 Call Forward – Off-Premise must be allowed in Class of Service Feature
Selection to set call forwarding to main K-CCIS voice mail.
 A remote system can have only Message Waiting LED on Line key for
extensions in the remote system. Remote system users cannot press a
flashing Line key to route to voice mail or the message box of an extension on
Message Waiting LED on the Line key.
 The following features are not supported for voice mail users in remote
systems:
 Live Record
 Live Monitoring
 Caller ID Display
 Softkeys
 Await Answer transfer from voice mail
 Call Holding
 Constant Message Count Indication
 Call Back to VM
 Live Transfer (Caller ID Return Call)
 The Dial Access Code for Single Line Telephone Hookflash is supported for
trunk calls into the main system only.
 The voice mail must be installed in the PBX (NEAX system) when 5-, 6-, or
7-digit station numbers are used.
 Centralized Voice Mail and Local Voice Mail cannot be mixed in a K-CCIS
network.

Issue 5.0

 Call Forwarding – All Calls - K-CCIS
 Call Forwarding – Busy/No Answer - K-CCIS
 Multiple Call Forwarding – All Calls - K-CCIS
 Multiple Call Forwarding – Busy/No Answer - K-CCIS
This guide provides a list of associated Memory Blocks that support this feature.
Program/
Item No.
11-01-01 System Numbering 1 = Service Code Defaults for 1X, 2X, and 3X = 2 Extension
2 = Extension Number Number.
3 = Trunk Access
4 = Special Trunk access
5 = Operator Access
7 = Not Used
8 = Networking Access System
11-02-01 Extension Numbering Assign Station Numbers to Port Defaults for Ports 1~960:
Numbers 200-499
5000-5659
16-01-02 Department Group Basic 0 = Normal Routing (Priority) Set the call routing for Department Calling.
Data Setup – Department 1 = Easy – UCD Routing
Calling Cycle (Circular)
Default is 0
16-01-03 Department Group Basic 0 = Normal (Intercom caller to Assign how the system routes an Intercom
Data Setup – Department busy department member hears Call to a busy Department Group member.
Routing When busy busy)
1 = Circular (Intercom callers to
busy department member route
to idle member)
Default is 0
16-01-04 Department Group Basic 0 = Last extension is called and Assign the action taken when a call reaches
Data Setup – Hunting Mode hunting is stopped the last extension in the Department Group.
1 = Circular
Default is 0

Issue 5.0
Program/
Item No.
16-02-01 Department Group Groups 1~64 Assign the Department Groups. The initial
Assignment for Extensions Priority 1~999 priority value becomes the numerical port
Default = 1-XXX order assigned in PRG 11-02 and 11-04
(Ports 1~256).
11-11-01 Service Code Setup (for Assign the Call Forward – All Call service
Setup/Entry Operation) – Set /Cancel Service Code.
Call Forward – All Call Default is 848
11-11-02 Service Code Setup (for Assign the Call Forward – Busy Set/Cancel
Setup/Entry Operation) – Service Code.
Call Forward – Busy Default is 843
11-11-03 Service Code Setup (for Assign the Call Forward – No Answer Set/
Setup/Entry Operation) – Cancel Service Code.
Call Forward – No Answer Default is 845
11-11-04 Service Code Setup (for Assign the Call Forward – Busy/No Answer
Setup/Entry Operation) – Set/Cancel Service Code.
Call Forward – Busy/No
Answer Default is 844
11-07-01 Department Group Pilot UP to eight digits can be Assign pilot numbers to each Department
Numbers – Dial assigned. Group set up in PGM 16-02.
11-16-09 Single Digit Service Code Assign single digit access code for Quick
Setup – Access to Voice Transfer to Voice Mail.
Mail Default is No Setting
20-06-01 Class of Service for 0~15 Default is All Extensions are in Class 1.
Extensions
20-09-02 Class of Service Options 0 = Off Enable the Caller ID display at an
(Incoming Call Service) – 1 = On extension.
20-11-01 Class of Service Options 0 = Off Disable Call Forward – All Call at an
(Hold/Transfer Service) – 1 = On extension.
Call Forward All Default is 1
20-11-02 Class of Service Options 0 = Off Disable Call Forward – Busy at an
Call Forward When Busy Default is 1
20-11-03 Class of Service Options 0 = Off Disable Call Forward – No Answer at an
Call Forwarding When
Unanswered Default is 1

Issue 5.0
Program/
Item No.
20-11-12 Class of Service Options 0 = Off Enable Call Forward – Off-Premise at an

Call Forwarding Off
Forwarding) Default is 0
20-11-13 Class of Service Options 0 = Off Turn Off or On an extension user ability to
(Hold/Transfer Service) – 1 = On have a call which recalls from hold transfer
Operator Transfer After to the operator.
Hold Callback Default is 0
20-11-14 Class of Service Options 0 = Off Enable Trunk-to-Trunk Transfer Restriction

(Hold/Transfer Service) – 1 = On at an extension.
Trunk-to-trunk Transfer
Restriction Default is 0
15-07-01 Programmable Function Service Codes:

Keys 10 = Call Forward – Immediate 848
12 = Call Forward – No Answer 845
13 = Call Forward – Busy/No
Answer 844
14-01-13 Basic Trunk Data Setup – 0 = Disable (No) Must be enabled for Trunk-to-Trunk
Trunk-to-Trunk Transfer 1 = Enable (Yes) Transfer, Call Forward – Off-Premise, and
Tandem Trunking.
14-01-22 Basic Trunk Data Setup – 0 = Disable (No) Enable the system to send Caller ID Digits
Caller ID to Voice Mail 1 = Enable (Yes to Voice Mail.
Default is 0
22-02-01 Incoming Call Trunk Setup 0 = Normal Set the feature for the trunk being
1 = VRS (second dial tone if no programmed.
VRS installed)
2 = DISA
3 = DID
4 = DIL
5 = E&M Tie Line
6 = Delayed VRS
7 = ANI/DNIS
Default is 0
22-07-01 DIL Assignment Extension Number/ Pilot Number Assign destination extension or Department
(eight digits maximum) Group pilot number programmed in PRG

Issue 5.0
Program/
Item No.
30-03-01 DSS Console Key Key Numbers 001~114 Customize Key Assignments for DSS
Assignment 00~99 = General Functional Consoles 1~32.
Level
*00~*99 = Appearance Function
Level
Default is Extensions. 200~259
45-01-01 Voice Mail Integration 0 = No Voice Mail Assign an Extension Department Group as
Options – Voice Mail 0~64 Groups the Voice Mail Group.
Department Group Number
Default is 0
45-01-14 Voice Mail Integration Up to eight digits Assign the CCIS Centralized Voice Mail
Options – CCIS Centralized Pilot number for remote sites.
Voice Mail Number

Issue 5.0

UNIVERGE® SV9100
SV9100 IP K-CCIS Chapter 5
This chapter describes the system outline, Hardware installation, and programming
procedures for providing IP K-CCIS using the GPZ-IPLE on the UNIVERGE SV9100
system.
SECTION 1 SYSTEM OUTLINE
1.1 IP K-CCIS Application using the GPZ-IPLE
The system uses the GPZ-IPLE to connect multiple systems together over a Data
Communication IP Network (Intranet). Key-Common Channel Interoffice Signaling
(K-CCIS) is used to provide telephony services between the UNIVERGE SV9100
and another UNIVERGE SV9100 or a NEAX PBX system.
1.1.1 CCIS Networking via IP (Non Peer-to-Peer Connections Basis)

 IP trunk connections over CCIS Networking via IP is used to provide
telephony services between the UNIVERGE SV9100 and UNIVERGE
SV9100 and a NEAX IPS, IPX, SV7000, UNIVERGE SV8100 and
UNIVERGE SV9300.
 The SV9100 uses the NEC proprietary CCIS Peer-to-Peer protocol

over IP to communicate between system to system.

Issue 5.0
 The GPZ-IPLE is required for connections between IP terminals and

IP trunks. A maximum of one GPZ-IPLE can be accommodated per
system with a maximum of 256 DSP Resources per system.
Figure 5-1 Peer-to-Peer One IP to TDM
1.2 Description
The GPZ-IPLE board is an optional interface package for converting the Real
Time Transfer Protocol (RTP) packets on the IP network to PCM highway. IP
telephones are connected directly to the IP bus. When IP telephones are required
to be connected to conventional PCM based digital circuit, the GPZ-IPLE board
converts IP packet signals. The GPZ-IPLE provides the digital signal processors
(DSPs) for IP stations and trunks.
A DSP provides format conversion from circuit switched networks (TDM) to

packet switched networks (IP). Each voice channel from the circuit switched
network is compressed and packetized for transmission over the packet network.
In the reverse direction, each packet is buffered for de-jittering, decompressed,
and sent to the circuit switched network. Each DSP converts a single speech
channel from IP to TDM and vice versa.
The following are examples of DSP allocation.

 Calling from an IP phone to a TDM phone uses one DSP.
 Calling from an IP phone to another IP phone that is registered to the same

CPU uses no DSPs.
 Calling from a TDM phone to a TDM phone uses no DSPs.
5-2 SV9100 IP K-CCIS

Issue 5.0
 Calling from a TDM phone and out a IP trunk uses one DSP.
 Calling from a TDM phone across IP K-CCIS to another TDM phone uses one
DSP.
 Calling from an IP Phone across IP K-CCIS to another IP Phone uses two DSP
resources at each location.
1.3 Systems Requirements
Only voice (RTP/RTCP) processing functions are mounted among VOIP functions
on the GPZ-IPLE and all call control functions are handled by the GCD-CP10.
Only one GPZ-IPLE blade can be mounted on the GCD-CP10 at any given time.
The GPZ-IPLE daughter board has Layer2 Switch ability, along with a Gigabit
Ethernet LAN interface and RTP/RTCP packet is transmitted and received
directly.
The number of ports supported by the IP K-CCIS (Peer-to-Peer) application

depends on the number of ports licensed in the GCD-CP10. For example, if the
GCD-CP10 is installed with a GPZ-IPLE and the GCD-CP10 is licensed for 64
VoIP Channels. The maximum configuration supported by the CCISoIP
Application is 64 CCISoIP channels.
Figure 5-2 GPZ-IPLE Daughter Board
The GPZ-IPLE board can be mounted only on the GCD-CP10. Only one
GPZ-IPLE board at a time can be mounted on the GCD-CP10.

Issue 5.0
1.3.1 Hot Swap
The GPZ-IPLE is not hot swappable and cannot be removed from the
GCD-CP10 without first powering down the chassis and removing the
GCD-CP10.
1.3.2 Connectors
The GPZ-IPLE has the following connectors:
 CN1 - RJ-45 Gigabit Ethernet LAN Interface
 CN3 - Connects to the GCD-CP10
1.3.3 Ethernet Status (IPLE)
Four LEDs (two red, one green and one yellow) on the GPZ-IPLE indicate
Ethernet connection status. The yellow LED is on when the Ethernet link is
up. The green LED flashes to indicate activity and the two red LEDs are
solid to indicate 10 Base-T/100 Base-TX or 1000 Base-T link speed.
1.3.4 Environmental Conditions
Temperature: +32 to 104 degrees Fahrenheit
Humidity: 10% ~ 90% (noncondensing)
Cable length of Ethernet cable: Up to 100m
1.3.5 Installation Procedure
Perform the following steps to install a GPZ-IPLE on the GCD-CP10:
Do not remove or install the GCD-CP10 with the power on.

WARNING
1. Turn off the system power.
2. Remove the GCD-CP10, and install the GPZ-IPLE on it.
3. Insert the GCD-CP10 in slot 1 in the Controlling Chassis.
4. Figure 5-3 GCD-CP10 Blade with Daughter Board

InstalledConnect the GPZ-IPLE to the CD-RTB or to an
external switching hub using an ethernet cable. Refer to the
UNIVERGE SV9100 Programming Manual for detailed
programming instructions.
1.3.6 Switch Settings
The daughter board has no switches to set.

Issue 5.0
1.3.7 LED Indications
Refer to Figure 5-3 GCD-CP10 Blade with Daughter Board Installed.
Figure 5-3 GCD-CP10 Blade with Daughter Board Installed
GPZ-IPLE Blade
Battery

Issue 5.0
See Figure 5-4 GCD-CP10 LED Locations for the location of the LEDs on
the blade.
Figure 5-4 GCD-CP10 LED Locations
LED indications for the GPZ-IPLE daughter board are shown in Table 5-1
VOIPDB LED Indications. Each LED is listed with its associated function,
LED status and Operation status.
Table 5-1 VOIPDB LED Indications
LED
LED Function Operation Status
Status
Link 10 (LED 10) 10 Base-T link speed indicator On Red 10 Base-T link up
Link 100 (LED
100 Base-T link speed indicator On Red 100 Base-T link up
100)
Link 1000
1000 Base-T link speed indicator On Yellow 1000 Base-T link up
(LED 1G)
ACT (LED A) Link activity or data transmission On Green Link up completed
and reception.

Issue 5.0
The following table shows the LED indication when transmitting or receiving
data on CN1.
Table 5-2 VOIPDB LED CN1 Transmit/Receive Data Indications
Link Up
Auto Negotiation Mode Force Mode

LED
100Mbps 10Mbps 100Mbps 10Mbps
1000M 1000M
bps bps
Half Full Half Full Half Full Half Full
ACT (LED A) ON ON ON ON ON ON ON ON ON ON
LINK1000 (LED 1G) ON OFF ON OFF ON ON OFF ON OFF ON
LINK 100 (LED 100) ON OFF OFF OFF OFF ON OFF OFF OFF OFF
LINK10 (LED 10) OFF ON ON OFF OFF OFF ON ON ON ON

Issue 5.0
1.3.8 Connectors
Figure 5-5 VoIP Connections shows a typical connection layout. Figure 5-6
Connecting a VOIPDB to a Network/PC on page 5-9 illustrates how to
connect a VoIP Daughter Board to a Network or PC.
Figure 5-5 VoIP Connections
GCD-CP10
GPD-8/16DLCB

Issue 5.0
1.4 IP Addressing
Figure 5-6 Connecting a VOIPDB to a Network/PC shows how the VOIPDB is

connected to a network or PC.
Figure 5-6 Connecting a VOIPDB to a Network/PC
One valid IP address must be assigned for all the DSP's that are used in IPLE.

Issue 5.0
The GPZ-IPLE needs only one IP Address.
When assigning the IP address to the IPLE card, the address must be
in the same network (subnet). If the CPU is also connected to the
network, it requires a separate IP address in a different network
IMPORTANT
(subnet).
When you have an IPLE card attached to the CPU, the CPU NIC is
no longer required. All connections that previously terminated to the
CPU NIC card can now be terminated to the IPLE NIC.
For example, PC Pro, Web Pro, ACD, etc. terminate to the IPLE NIC
card, when installed. Both the IPLE and CPU NIC share the same
gateway assignment. The default gateway command in 10-12-03 is
used by both NICs, allowing only one device, IPLE or CPU, to route
outside of its own network.
1.4.1 General IP Configuration
The voice quality of VoIP depends on variables such as available

bandwidth, network latency, and quality of service initiatives (QOS), all of
which are controlled by the network and Internet service providers.
Because these variables are not in NEC control, it cannot guarantee the
performance of the users IP based voice solution. Therefore, NEC
recommends connecting the VoIP equipment through a local area network
using private IP addresses.
For a network to be suitable for VoIP it must pass specific requirements. To

make sure that the site meets these requirements an IP ready check and a
site survey must be completed at each site before VoIP implementation.
 One way delay must not exceed 100 ms
 Round Trip delay must not exceed 200 ms
 Packet loss must not exceed 1%
 Data switches must be manageable
 Routers must provide QOS
 Adequate bandwidth for estimated VoIP traffic (refer to 1.4.3
Bandwidth)
Depending on how QOS policies are built in the network, assignments

might be needed in the CPU.
1.4.2 ToS Settings (Layer 3 QoS)
The marking of packets at layer 3 is done by marking the ToS byte in the IP
header of the voice packet. The SV9100 supports two methods for marking
the ToS byte.
 IP precedence
 DSCP (Diffserv)

Issue 5.0
IP Precedence
IP precedence uses the first three bits of the ToS field to give eight possible
precedence values (0~7). Under normal circumstances, the higher the
number, the higher the priority. However, the administrator can assign these
precedence values with the lower values having a higher priority.
The following table shows the eight common values for IP precedence.
Table 5-3 IP Precedence Values
Precedence Higher Number Value Correlates

TOS Field
Value to Higher Priority
000 0 Routine or Best Effort
001 1 Priority
002 2 Immediate
003 3 Flash
004 4 Flash Override
005 5 Critical
006 6 Internetwork Control
007 7 Network Control
Working in conjunction with IP precedence, the next four bits in the ToS field
influence the delivery of data based on the following:
 delay
 throughput
 reliability
 cost
However these fields are usually not used. The following table shows the
eight bit ToS field and the associated IP precedence bits.
Table 5-4 ToS Field with IP Precedence Bits
IP Throughp Reliabilit Not

IP Precedence IP Precedence Delay Cost
Precedence ut y Used
1(on) here = value of 4 1(on) here = value of 2 1(on) here = value of 1

Issue 5.0
DSCP
DSCP stands for Differential Services Code Point (or Diffserv for short). It
uses the first 6 bits of the ToS field, therefore, giving 64 possible values.
The following table lists the most common DSCP code points, the binary
value, and the associated name.
Table 5-5 DSCP Code Points and Binary Values
DSCP Code Points DSCP Value Name
000000 08 Best Effort (BE)
001000 8 Class Selector 1 (CS1)
001010 10 Assured Forwarding 11 (AF11)
100110 38 Assured Forwarding 43 (AF 43)
101110 46 Expedited Forwarding (EF)

Issue 5.0
The following table shows the 8 bit ToS field and the associated Diffserv
bits.
Table 5-6 ToS Field with Diffserv sBits
Not
Not
Diffserv Diffserv Diffserv Diffserv Diffserv Diffserv Use
Used
d
1(on) here = 1(on) here = 1(on) here = 1(on) here = 1(on) here = 1(on) here =
value of 32 value of 16 value of 8 value of 4 value of 2 value of 1
Assignments for the IP Precedence/Diffserv values in the system are submitted in

command 84-10.
RTP/RTCP = voice packets

CCIS= signaling packets
Figure 5-7 System Data 84-10 ToS Setup

Issue 5.0
1.4.3 Bandwidth
The bandwidth required for VoIP calls depends on several factors.
 Layer 2 media
 CODEC
 Packet Size
 RTP Header Compression
 Voice Activity Detection (VAD)
 Number of simultaneous calls
 Possibly add encryption after research.
Layer 2 media is concerned with moving data across the physical links in
the network. A few of the most common layer 2 media types are Ethernet,
PPP, and Frame Relay.
CODEC stands for Coder/Decoder and is the conversion of the TDM signal
into an IP signal and vice versa. A CODEC can also compress/decompress
the voice payload to save on bandwidth.
Packet Size is the amount of audio in each PDU (protocol data unit)
measured in milliseconds. The larger the packet the less bandwidth used.
This is because sending larger packets (more milliseconds of voice)
requires, overall, less packets be sent. The downside of this practice is if a
packet is dropped/lost a larger piece of voice is missing from the
conversation as the system waits the additional delay for the next packet
arrival.
RTP Header Compression compacts the RTP header from 40 bytes in size
to 2 ~ 4 Bytes in size. RTP header compression is used only on low speed
links. Regularly on every voice packet there is an IP/UDP/RTP header that
is 40 bytes in length. Compressing this header, down to 2 ~ 4 bytes, can
save a considerable amount of bandwidth. The following is an example of a
VoIP packet without RTP header compression and one of a packet with
RTP header compression.
Notice that the overall packet size, when using RTP header compression, is
considerably smaller.
 VOIP packet without RTP header compression

IP Header UDP Header RTP-Header
Voice-Payload
20-bytes 8-bytes 12-bytes

Issue 5.0
 VOIP packet with RTP header compression

Compressed
Voice-Payload
Header 2 ~ 4-bytes
Voice Activity Detection (VAD) is suppression of silence packets from

being sent across the network. In a VoIP network all conversations are
packetized and sent, including silence. On an average a typical
conversation contain anywhere from 35% ~ 45% silence. This can be
interrupted as 35% ~ 45% transmission of VoIP packets, having no audio,
use valuable bandwidth. With the VAD option enabled, the transmitting of
packets stops after a threshold is met determining silence. The receiving
side then injects comfort noise into the call so it does not appear the call has
dropped.
1.4.4 Bandwidth Calculations
The first step in calculating the bandwidth of a call is determining how many
bytes the voice payload is going to use. The amount is directly affected by
the CODEC and packet size. Below are the supported default CODEC
speeds for CCISoIP.
 G.711 = 64000 bps
 G.722 = 64000 bps
 G.729 = 8000 bps
 G.726 = 32000 bps
 G.723 = 5400 bps
1.4.5 Payload Calculation Voice
(Packet size * Codec bandwidth) / 8 = Voice Payload in Bytes
Example of G.711 with a 20 ms packet size

(.020 * 64000) /8 = 160 Bytes
Example of G.729 with a 30 ms packet size

(.030 * 8000) /8 = 30 Bytes
Now that you have the voice payload in bytes you can now calculate the
overall bandwidth including the layer 2 media. Below are some of the
common layer 2 media types and their overhead.
 Ethernet = 18 Bytes
 802.1Q/P Ethernet = up to 32 bytes
 PPP = 9 Bytes
 Frame Relay = 6 Bytes
 Multilink Protocol = 6 Bytes

Issue 5.0
Bandwidth Calculation
( [ Layer 2 overhead + IP/UDP/RTP header + Voice Payload ] / Voice

Payload ) * Default CODEC speed = Total Bandwidth
Example of a G.711 call over Ethernet using a 20 ms packet size and not
using RTP header compression
(.020 * 64000) / 8 = 160 Bytes for Voice Payload
( [ 18 + 40 + 160] / 160 ) * 64000 = 87200 bps
If VAD is not enabled, each side of the conversation would be streaming

87.2 Kbps in one direction for a total of 174.4 kbps.
Below is a chart showing the supported CODECs, with a few different

packet sizes over PPP and Ethernet.
CODEC Packet Size PPP Ethernet
G.711 10 103.2 kbps 110.4 kbps

G.711 20 83.6 kbps 87.2 kbps
G.711 30 77.1 kbps 79.5 kbps
G.711 40 73.9 kbps 75.6 kbps
G.722 10 103.2 kbps 110.4 kbps

G.722 20 83.6 kbps 87.2 kbps
G.722 30 77.1 kbps 79.5 kbps
G.722 40 73.9 kbps 75.6 kbps
G.729 10 47.2 kbps 54.4 kbps

G.729 20 27.6 kbps 31.2 kbps
G.729 30 21.1 kbps 23.5 kbps
G.729 40 17.8 kbps 19.6 kbps
G.729 50 15.9 kbps 17.3 kbps
G.729 60 14.5 kbps 15.7 kbps
G.726 10 71.2 kbps 78.4 kbps

G.726 20 51.6 kbps 55.2 kbps

Issue 5.0
G.726 30 45.1 kbps 47.5 kbps

G.726 40 41.8 kbps 43.6 kbps
G.723 30 18.5 kbps 20.8 kbps

G.723 60 11.9 kbps 13.2 kbps
1.4.6 Some Network Considerations
Before adding the UNIVERGE SV9100 to a customer network, a detailed

network diagram of the existing network must be obtained from the
customer. This diagram provides information about any network condition
that can prevent or hinder the VoIP equipment from functioning correctly.
Multiple Gateways
Figure 5-8 Multiple Gateways shows the UNIVERGE SV9100 assigned a

default Gateway of 192.168.1.1/24. The existing network has two routers in
the same broadcast domain connecting different networks. With regular
data communications this is not an issue. However, the VOIP cannot
function correctly in this type of network.
Figure 5-8 Multiple Gateways
IPLE Signaling Address = 192.168.15.5/24

IPLE DSP = 192.168.15.6/24
IPLE Default Gateway = 192.168.15.1/24
Internet
IP Address
192.168.1.1/24 IP Address
192.168.15.1/24
IP Address
192.168.20.1/24
IP Address
IPLE Signaling Address = 192.168.15.5/24 192.168.1.2/24
IPLE DSP = 192.168.15.6/24

IPLE DSP = 192.168.20.6/24

Issue 5.0
Firewall
Another regular device in customer networks that can hinder VoIP

performance is a firewall. Most corporate LANs connect to the public
Internet through a firewall. A firewall is filtering software built into a router or
a stand alone server unit. It is used to protect a LAN it from unauthorized
access, providing the network with a level of security. Firewalls are used for
many things, but in its simplest form, a firewall can be thought of as a one
way gate. It allows outgoing packets from the local LAN to the Internet but
blocks packets from the Internet routing into the local LAN, unless they are
a response to query.
A firewall must be configured to allow specific traffic from the Internet to

pass through onto the LAN.
Figure 5-9 Two SV9100 Systems Connected via the WAN shows two
SV9100 systems. One on the corporate local LAN and one on a Remote
network connected via the WAN. The remote site cannot call the MAIN site,
therefore it is not working.
Figure 5-9 Two SV9100 Systems Connected via the WAN
Headquarters
WAN
Local LAN
Remote Network
Firewall Firewall
The green arrow in Figure 5-9 Two SV9100 Systems Connected via the
WAN represents the data packets leaving the REMOTE IPLE card destined
for the SV91000 on the Headquarters LAN. The firewall on the
Headquarters network is not configured to recognize the TCP/UDP ports
utilized by the NEC equipment thus blocking them resulting in registration
failure. To solve this issue the ports used by the NEC VoIP equipment have
to be opened in the firewall allowing the NEC traffic to pass through to the
SV9100.
The ports, 57000 and 59000 (TCP) for signaling and the voice ports, are
required to be open at each location.

Issue 5.0
VPN
Another common feature is the use of the Internet as the WAN between
customer locations. When this is done VPNs are typically used between the
locations. A VPN (Virtual Private Network) is a private data network that
maintains privacy through the use of tunneling protocols and security
features over the public Internet. This allows for remote networks (with
private addresses), residing behind NAT routers and/or firewalls, to
communicate freely with each other. When building the VPN tunnels,
throughout the network, they must be assigned as a fully meshed network.
This means that every network is allowed direct connection to each and
every other network in the topology.
The following diagram shows three sites connected together via VPN. This
network is not fully meshed due to the lack of a VPN tunnel between Sites B
and C.
Figure 5-10 VPN Network
SITE B
VPN Tunnel
Between Sites A and B
SITE A
IPLE 32
IPLE DSP = 192.168.15.6/24
Internet
IPLE 32
IPLE DSP = 192.168.15.6/24
VPN Tunnel SITE C

Between Sites A and C
IPLE 32
IPLE DSP = 192.168.20.6/24
Site A can communicate with phones on sites B and C. Phones on sites B

and C can not communicate with each other.
To correct this issue, another VPN connection between sites B and C is

required.
MTU Size
In some network environment the MTU size of CCPU or IPL NIC may need
to be changed. With Version 2.00 or lower the MTU size was fixed to 1500
for both CCPU & IPL NIC. With Version 3.00 or higher now the MTU size
can be changed for both CCPU & IPL NIC.

Issue 5.0
If data to be send is greater than the defined MTU, the data will be
transmitted into two or more packets as per defined MTU.
Conditions
 MTU size defined in PRG 10-12-18 is applied on the packets

transmitted from IP address define in PRG 10-12-01.
 MTU size defined in PRG 10-12-19 is applied on the packets

transmitted from IPL IP address define in PRG 10-12-09.
 MTU size value is applied after logging out from WebPro, PcPro or
TelPro.
 MTU is not applied on RTP / RTCP packets generated from DSP of

VoIPDB.
SECTION 2 UNIVERGE SV9100 CHASSIS PROGRAMMING
The following data programs are used when installing the GPZ-IPLE daughter board
for UNIVERGE SV9100 IP (K-CCIS).
 If any address or NIC setting is changed, the system must be reset for the changes to take
affect.
2.1 Digital Trunk Assignment
Use these programs to make digital trunk assignments when an IPLE daughter
board is installed. .
Table 5-7 Digital Trunk Assignments
Program/
Item No.
10-68-01 IP Trunk Availability - 0 = None Define the IP Trunk Type

Trunk Type 1 = SIP
2 = H.323
3 = CCIS
Default is 0
10-68-02 IP Trunk Availability - 0 ~ 400 Assign the Start Port for

Start Port IP Trunks.
Default is 0
10-68-03 IP Trunk Availability - 0 ~ 400 Define the number of

Number of Ports ports.
Default is 0

Issue 5.0
Table 5-7 Digital Trunk Assignments (Continued)
Program/
Item No.
10-12-03 GCD-CP10 Network 0.0.0.0 ~ 126.255.255.254 Assign the Default

Setup – Default Gateway 128.0.0.1 ~ 191.255.255.254 Gateway for the
192.0.0.1 ~ 223.255.255.254 GCD-CP10.
Default is 0.0.0.0 When using the IPLE,

the IP Address and
Subnet Mask in
PRG 10-12-01 and
PRG 10-12-02 remain
default.
10-12-09 GCD-CP10 Network 0.0.0.0 ~ 126.255.255.254 Assign the IP Address of
Setup – IP Address 128.0.0.1 ~ 191.255.255.254 the IPLE.
192.0.0.1 ~ 223.255.255.254
Default is 0.0.0.0
10-12-10 GCD-CP10 Network 255.255.0.0 Assign the Subnet Mask

Setup – Subnet Mask of the IPLE
Default is 255.255.0.0
10-19-01 VoIP DSP Resource Slot Number 1:

Selection – VoIP DSP
Resource Selection DSP Resource Number:
01 ~ 256
0 = Common use for both IP

extensions and trunks
1 = IP Extension
2 = SIP Trunk
3 = Networking/CCIS
4 = NetLink
5 = Blocked
6 = Common without Unicast
paging
7 = Multicast Paging
8 = Unicast Paging
Default:
Resource 1 = 1
Resources 2~256 = 0

Issue 5.0
Table 5-7 Digital Trunk Assignments (Continued)
Program/
Item No.
10-54-01 License Configuration for 1 ~ 255 Resource Licenses

Each Package - License
Code
Assign VoIP Resource
Licenses (5103) to GCD-
CP10 slot (1). If applying
more than 255 licenses to a
slot, the licenses must be
applied across multiple
indexes. For example,
assigning 256 VoIP resource
licenses (5103) to the CPU
slot could be assigned using
different methods, as long as
the total for the CPU slot is
256. 1. Index 1 has 128 of
feature code 5103 and index
2 also has 128 of feature
code 5103 for a total of 256.
2. Index 1 has 255 of feature
code 5103 and index 2 has 1
of feature code 5103 for a
total of 256.
14-05-01 Trunk Group 0 ~ 100 Assign Trunk-to-Trunk

Default is 1 Groups.
22-02-01 Incoming Call Trunk 001 ~ 400 Set the feature type for
Setup – Trunk Type the trunk you are
0 = Normal programming.
1 = VRS (second dial tone if no
VRS installed)
2 = DISA
3 = DID
4 = DIL
5 = E&M Tie line
6 = Delayed VRS
7 = ANI/DNIS
8 = DID(DDI) Mode
Switching
Default is 0

Issue 5.0
2.2 VoIP IP Address Assignments
For this feature, the GPZ-IPLE, daughter board is installed on the GCD-CP10
blade. The GPZ-IPLE daughter board reduces the maximum capacity of trunks in
the system.
Table 5-8 VoIP IP Address Assignments
Program/
Item No.
84-26-01 IPL Basic Setup – IP XXX.XXX.XXX.XXX Assign the VOIPDB Gateway IP

Address Address.
Slot 1 = 172.16.0.20
84-26-02 IPL Basic Setup – RTP 0~65534 Default:

Port Number
RTP Port = 10020
84-26-03 IPL Basic Setup – RTCP 0~65534 Default:

Port Number
(RTP Port Number + 1) RTCP = 10021

Issue 5.0
2.3 Local Numbering Plan Assignment
Use these programs to assign system, extension and virtual extension numbering
for the Local Numbering Plan.
Table 5-9 Local Numbering Plan Assignments
Program/
Item No.
11-01-01 System Numbering 0 = Not Used Defaults for 1X, 2X, 3X 4X, 5X,
1 = Service Code 6X= 2 Extension Number.
3 = Trunk Access
5 = Operator Access
7 = Not Used
8 = Networking System
Access
11-02-01 Extension Numbering Assign Station Numbers to Defaults for Ports

Port Numbers Port 1 ~ 300 = 200 ~ 499
Port 301 ~ 960 = 5000 ~ 5659

Port Numbers All Virtual Extension Ports =
Numbering No Setting
2.4 Closed Numbering Plan – Using Closed Number Blocks
Use these programs to assign system numbering and ARS/F-Route dialed digits
for a Closed Numbering Plan.
Table 5-10 Closed Numbering Plan Assignments
Program/
Item No.
11-01-01 System Numbering 2 = Extension Number Defaults for 1X, 2X, 3X 4X, 5X,
6X= 2 Extension Number.
44-02-01 Dial Analysis Table for Up to 8 digits can be assigned. Assign the digits to be dialed
ARS/F-Route Access – across the K-CCIS link. These
Use line key 1 for Don’t Care digits were assigned as
Dial F-Route in Program 11-01-01.
digit, @

Issue 5.0
Table 5-10 Closed Numbering Plan Assignments (Continued)
Program/
Item No.
44-02-02 Dial Analysis Table for 2 = ARS/F-Route Table The service type 2 assigns the
ARS/F-Route Access – (F-Route) digits to be dialed to an
F-Route.
Service Type
Default is No Setting Program 44-02-03 assigns the
F-Route to be used.
44-02-03 Dial Analysis Table for 2 = 0~500 When setting data is (2), refer
ARS/F-Route Access – 0 = No Setting to Program 44-05.
Additional Data
Default is 0
44-05-01 ARS/F-Route Table – 0-100,101-150,255 Select the trunk group used for
Trunk Group Number 0 = Not Set the outgoing K-CCIS call.
PRG 14-05
Default is 0
2.5 Open Numbering Plan – Using ARS Table 1, 2, or 3
Use these programs to assign system numbering, ARS/F-Route dialed digits and
trunk groups as well as other ARS/F-Route table information for an Open
Numbering Plan.
Table 5-11 Open Numbering Plan Assignments
Program
/Item No.
11-01-01 System Numbering 2 = Extension Number Defaults for 1X, 2X, 3X 4X, 5X,
6X= 2 Extension Number.
44-02-01 Dial Analysis Table for Up to 8 digits can be Assign the digits to be dialed
digits were assigned as F-Route in
Use line key 1 for Don’t Care Program 11-01-01.
digit, @
44-02-02 Dial Analysis Table for 2 = ARS/F-Route Table The service type (2) assigns the
ARS/F-Route Access – (F-Route) digits to be dialed to an F-Route.
Default is 0
44-02-03 Dial Analysis Table for 0~500 When setting data is (2), refer to
ARS/F-Route Access – (0 = No Setting) Program 44-05.
Additional Data
Default is 0

Issue 5.0
Table 5-11 Open Numbering Plan Assignments (Continued)
Program
/Item No.

PRG 14-05
Default is 0
44-05-02 ARS/F-Route Table – 0 = No Setting Enter the number of digits to be

Delete Digits 1~255 = Number of digits to deleted from the dialed number.
delete
(255 = Delete All
Default is 0
44-05-03 ARS/F-Route Table – 0 = No Setting

Additional Dial Number 0~1000
Table Add Digits in Program
44-06-01
Default is 0
44-05-04 ARS/F-Route Table – 0 = Off Select whether or not a beep is

Beep Tone 1 = On heard if a lower priority trunk
Default is 0 group is used to dial out.
44-05-05 ARS/F-Route Table – 0 = No Setting Select the gain table number to be

Gain Table Number for 0 ~ 500 used for the internal call (defined
in Program 44-07).
Internal Calls
Default is 0
44-05-06 ARS/F-Route Table – 0 = No Setting Select the gain table number to be

Gain Table Number for 0 ~ 500 used for the tandem call (defined
in Program 44-07).
Tandem Connections
Default is 0
44-05-07 ARS/F-Route Table – 0 = No Setting Select the ARS Class of Service

ARS Class of Service 1 ~ 16 to be used for the table. An
extensions ARS COS is
determined in Program 26-04-01.
Default is 0
44-05-08 ARS/F-Route Table – Dial 0 = No Setting If a Dial Treatment is selected,

Treatment 0 ~ 15 Programs
44-05-02 and 44-05-03 are
ignored and the Dial Treatment
Default is 0 defined in Program 26-03-01 is
used instead.
44-05-09 ARS/F-Route Table – 0 = No Maximum Assign the Max Digits for K-CCIS
Maximum Digit 0 ~ 24 or ISDN Calls.
Default is 0
44-05-10 ARS/F-Route Table – 0~16367 seconds

CCIS over IP Destination

Issue 5.0
2.6 IP K-CCIS Assignment
Use these programs to assign CCIS availability, origination and destination point
codes for IP K-CCIS.
Table 5-12 IP K-CCIS Assignments
Program/
Item No.
50-01-01 CCIS System Setting – 0 = Disable

CCIS Availability 1 = Enable
Default is 0
50-02-03 Connecting System 0 ~ 16367 Assign the Originating Point

Settings – Originating Code for Route ID 9 of own side.
50-03-01 CCIS Destination System 0 ~ 16367 Assign remote system IP network

Settings – Destination information.
PRG 50-03 assignments are only
used only for CCISoIP.
50-03-03 CCIS Destination System xxx.xxx.xxx.xxx Assign remote system IP network

Settings – IP Address (IP (xxx = 0 ~ 255) information.
only)
Default is 0.0.0.0 PRG 50-03 assignments are only
used for CCISoIP.
50-03-04 CCIS Destination System 0 : Disable

Settings – Point Code 1 : Enable
Availability
Default is 1
2.7 IP K-CCIS (Peer-to-Peer) Assignment
Use these programs to assign the CCIPS over IP connection method and the TCP
server port number.
Table 5-13 IP K-CCIS (Peer-to-Peer) Assignments
Program/
Item No.
50-15-02 CCIS over IP Basic 0 ~ 65535

Information Setting –TCP
Server Port Number
Default is 57000
50-15-03 CCIS over IP Basic 0 ~ 65535

Information Setting –
TCP Client Base Port
Number Default is 59000

Issue 5.0
2.8 CCIS over IP CODEC Information Setup
Use these programs to assign CCIS over IP CODEC information, including the
number of G.711/G.723/G.729 type, number of audio frames, voice activity
detection mode, jitter buffers and other CODEC related information.
Table 5-14 CCIS over IP CODEC Information Assignments
Program/
Item No.
84-21-01 CCIS over IP CODEC 1~4

Information Basic Setup – 1 = 10 ms
Number of G.711 Audio 2 = 20 ms
Frames 3 = 30 ms
4 = 40 ms
Default is 3
84-21-02 CCIS over IP CODEC 0 = A-law

Information Basic Setup – 1 = -law
G.711 Type
Default is 0
84-21-03 CCIS over IP CODEC 0 = Disable

Information Basic Setup – 1 = Enable
G.711 Voice Activity
Detection Mode Default is 0
84-21-04 CCIS over IP CODEC 0 ~ 300 ms

Information Basic Setup –
G.711 Jitter Buffer Min Default is 30

G.711 Jitter Buffer
Average Default is 60

G.711 Jitter Buffer Max
Default is 120
84-21-07 CCIS over IP CODEC – 1~6

G.729 Audio Frame
Number 1 = 10 ms
2 = 20 ms
3 = 30 ms
4 = 40 ms
5 = 50 ms
6 = 60 ms
Default is 3

Issue 5.0
Table 5-14 CCIS over IP CODEC Information Assignments (Continued)
Program/
Item No.
84-21-08 CCIS over IP CODEC 0 = Disable


G.729 Jitter Buffer Min Default is 30

G.729 Jitter Buffer
Average Default is 60

G.729 Jitter Buffer Max Default is 120
84-21-19 CCIS over IP CODEC 0 = G.711 PT

Information Basic Setup – 2 = G.729 PT
1st Priority of Audio 3 = G.722
Capability 4 = G.726
Default is 0
84-21-20 CCIS over IP CODEC 0 = G.711 PT

Information Basic Setup – 2 = G.729 PT
2nd Priority of Audio 3 = G.722 PT
Capability 4 = G.726 PT
Default is 2
84-21-22 CCIS over IP CODEC 1 = Static

Information Basic Setup – 2 = Not Used
Jitter Buffer Mode 3 = Self adjusting
Default is 3
84-21-23 CCIS over IP CODEC 0 = Self adjustment

Information Basic Setup – 1 = -19dBm (-49dBm)
Voice Activity Detection :
Threshold 20 = 0dbm (-30dBm)
:
29 = +9dbm (-21dBm)
30 = +10dbm (-20dBm)
Default is 20

Issue 5.0
Program/
Item No.

G.722 Audio Frame 1 = 10 ms
Number 2 = 20 ms
3 = 30 ms
4 = 40 ms
Default is 3

G.722 Jitter Buffer (min)
Default is 30

G.722 Jitter Buffer
(Average) Default is 60

G.722 Jitter Buffer (max)
Default is 120

G.726 Audio Frame 1 = 10 ms
Number 2 = 20 ms
3 = 30 ms
4 = 40 ms
Default is 3
84-21-33 CCIS over IP CODEC 0 = Disabled

Information Basic Setup – 1 = Enabled

Default is 30

G.726 Jitter Buffer

Issue 5.0
Program/
Item No.

Default is 120
84-21-37 ---Not Used---
2.9 SIP-MLT CODEC Information Fixed Mode Setup
Use these programs to assign the fixed mode audio capacity and number of audio
frames for the SIP MLT..
Program/
Item No.
84-29-01 SIP-MLT CODEC Type:

Information Fixed Mode 1~5
Setup – Audio Capability 1 = MultiCast
2 = Reserved
3 = Reserved
4 = Reserved
5 = Reserved
Codec:
1 = G.711 A-law
2 = G.711 -law
3 = G.729
5 = G.722
Default is 1

Issue 5.0
Program/
Item No.
84-29-02 SIP-MLT CODEC Type:

Information Fixed Mode 1~6
Setup – Number of Audio 1 = MultiCast
Frames 2 = Reserved
3 = Reserved
4 = Reserved
5 = Reserved
Size
1 = 10ms
2 = 20ms
3 = 30ms
4 = 40ms
5 = 50ms
6 = 60ms
Default is 2
2.10 MTU Assignment
In some network environment the MTU size of CCPU or IPL NIC may need to be
changed. With Version 2.00 or lower the MTU size was fixed to 1500 for both
CCPU & IPL NIC. With Version 3.00 or higher now the MTU size can be changed
for both CCPU & IPL NIC.
Program/
Item No.
10-12-18 GCD-CP10 Network 1000~1500 Define the MTU size for the packets sent from
Setup - CCPU MTU Default is 1450 IP address defined in PRG 10-12-01.
10-12-19 GCD-CP10 Network 1000~1500 Define the MTU size for the packets sent from
Setup - IPL MTU Default is 1450 IP address defined in PRG 10-12-09.

Issue 5.0
SECTION 3 PROGRAMMING EXAMPLE
3.1 SV9100 IP K-CCIS Programming Example 1
This is an example of UNIVERGE SV9100 program data assignments for a 4-digit

Closed Numbering Plan using Closed Number Blocks. The following system
configurations are used for all four systems:
 Each system is a single chassis with one CHS2U.
 Each system has 16 multiline terminals.
 Each system has eight analog trunks.
Figure 5-11 Programming Example 1
System B
Point Code 00002

GCD-CP10:
192.168.1.20
System A
IP Network GW1: 192.168.1.21
Point Code 00001
GCD-CP10:
192.168.1.10 1100s
GW1: 192.168.1.11
TEL 1100 TEL 1101 TEL 1102
1000s
TEL 1000 TEL 1001 TEL 1002
Assume that the systems are defaulted (first power on) with the following cards
installed as described in the following table.

Issue 5.0
3.1.1 Card Interface Slot Assignment (PRG 10-03)
The following slot assignments were set using program 10-03.
CARD TYPE AND PORTS
Card Type During 1st System A System B

Slot
Power On
Card Type Ports Card Type Ports
1 GCD-CP10/GPZ-IPLE GCD-CP10/GPZ-IPLE Trunks 5 ~ 28 GCD-CP10/GPZ-IPLE Trunks 5 ~ 28
2 GCD-8DLCA GCD-8DLCA Station 1 ~ 8 GCD-8DLCA Station 1 ~ 8
3 GCD-8DLCA GCD-8DLCA Station 9 ~ 16 GCD-8DLCA Station 9 ~ 16
4 None None N/A None N/A
5 GCD-4COTA/C GCD-4COTA/C Trunks 1 ~ 4 GCD-4COTA/C Trunks 1 ~ 4
3.1.2 Digital Trunk Assignments
This example shows digital trunk assignments made for systems A and B.
System A System B
Insert GPZ-IPLE on GCD-CP10 in Chassis 1, Slot Insert GPZ-IPLE on GCD-CP10 in Chassis 1,
1. Slot 1.
Verify PRG 10-03-01 (ETU Setup) Verify PRG 10-03-01 (ETU Setup).

Assign DSP Resources 001~024 as CCIS (LK4). Assign DSP Resources 001 ~ 024 as CCIS
(LK4).

Assign the Trunk Type as CCIS Assign the Trunk Type as CCIS

Assign the Start Port as 5 Assign the Start Port as 5

Assign 24 as number of ports Assign 24 as number of ports

Assign Trunks 5 ~ 28 in Trunk Group 10 Assign Trunks 5 ~ 28 in Trunk Group 10.

Set Trunks 5 ~ 28 as Tie Line (LK6). Set Trunks 5 ~ 28 as Tie Line (LK6).

Issue 5.0
3.1.3 VoIP Address Assignments
This example shows VoIP address assignments made for systems A and
B.
System A System B

IPL IP Address: 192.168.1.10 IPL IP Address: 192.168.1.20
Slot 1, Assign IP Address Slot 1, Assign IP Addresses
GW: 192.168.1.11 GW: 192.168.1.20
3.1.4 IP K-CCIS Availability
This example shows IP K-CCIS availability assignments made for systems

A and B.
System A System B

Select Option 1 (Enable) Select Option 1 (Enable)
Enables the CCIS Feature Enables the CCIS Feature

Assign Route ID 9 = 01 Assign Route ID 9 = 02
(Route ID 9 is designated for Host (Route ID 9 is designated for Host
Originating Point Code) Originating Point Code)
3.1.5 IP K-CCIS Assignment
This example shows IP K-CCIS assignments made for systems A and B.
System A System B

System ID 1 ~ 255 – Enter Destination System ID 1 ~ 255 – Enter Destination
Point Code Point Code
System ID 1 = DPC 02 System ID 1 = DPC 01

System ID 1 ~ 255 – Enter Destination System ID 1 ~ 255 – Enter Destination
System IP Address. System IP Address
System ID 1 = 192.168.1.20 System ID 1 = 192.168.1.10

Issue 5.0
3.1.6 IP K-CCIS (Peer-to-Peer) Assignment
This example shows IP K-CCIS peer-to-peer assignments made for

systems A and B.
System A System B

Enable/Disable Connection Method Enable/Disable Connection Method
0 = P2P Disable 0 = P2P Disable
1 = P2P Enable (Default) 1 = P2P Enable (Default)

Assign the TCP Server Port Number Assign the TCP Server Port Number.
0 ~ 65535 0 ~ 65535
57000 (Default) 57000 (Default)

Assign the TCP Client Base Port Number Assign the TCP Client Base Port Number.
0 ~ 65535 0 ~ 65535
59000 (Default) 59000 (Default)
3.1.7 Local Numbering Plan Assignment
This example shows Local Numbering Plan assignments made for systems
A and B.
System A System B

1 = Service Code 1 = Service Code
2 = Extension Number 2 = Extension Number
3 = Trunk Access Code 3 = Trunk Access Code
4 = 2nd Trunk Access Code 4 = 2nd Trunk Access Code
5 = Operator Access 5 = Operator Access
6 = ARS/F-Route Access 6 = ARS/F-Route Access
Assign Station Numbers to Port Assign Station Numbers to Port
Numbers. Numbers.
Defaults for Ports 1 ~ 960: Defaults for Ports 1 ~ 960:
200 ~ 499 200 ~ 499
5000 ~ 5659 5000 ~ 5659

Issue 5.0
3.1.8 Closed Numbering Plan – Using Closed Number Blocks
This example shows Closed Numbering Plan assignments made for

systems A and B.
System A System B

Dial 10 = 4 Digit; Extension Dial 10 = 4 Digit; F-Route
Dial 11 = 4 Digit; F-Route Dial 11 = 4 Digit; Extension

Assign Station Numbers to Port Number Assign Station Numbers to Port Number.
Defaults for Ports 1 ~ 960 Defaults for Ports 1 ~ 960

200 ~ 499 200 ~ 499
5000 ~ 5659 5000 ~ 5659
TBL 1 = 11 TBL 1 = 10

TBL 1 = F-Route TBL 1 = F-Route

F-Route TBL 1 = 1 F-Route TBL 1 = 1

F-Route TBL 1 = TG10 F-Route TBL 1 = TG10

F-Route TBL 1 Delete Digit = 0 F-Route TBL 1 Delete Digit = 0

F-Route TBL 1 Add Dial = 0 F-Route TBL 1 Add Dial = 0

F-Route TBL 1 Max Digit = 4 F-Route TBL 1 Max Digit = 4

F-Route TBL1 = DPC 2 F-Route TBL1 = DPC 1

Issue 5.0
3.1.9 Tandem Connections
This example shows Tandem connections assignments made for systems A

and B.
System A System B

Trunk to Trunk Transfer Trunk to Trunk Transfer
0 = Disable (Default) 0 = Disable (Default)

1 = Enable 1 = Enable
Trunk to Trunk Outgoing Caller ID Through Trunk to Trunk Outgoing Caller ID Through
Mode Mode
0 = Disable (Off) (Default) 0 = Disable (Off) (Default)

1 = Enable (On) 1 = Enable (On)
Figure 5-12 TEL 302 (SIP MLT) makes call to TEL 202 (SIP MLT) via IP
K-CCIS illustrates calls between two SIP multiline terminals over IP-CCIS.
Figure 5-12 TEL 302 (SIP MLT) makes call to TEL 202 (SIP MLT) via IP K-CCIS
System
RTP/Control
IP Network
System
RTP/Control
TEL 200 TEL 201 TEL 202
TEL 300 TEL 301 TEL 302

Issue 5.0
3.2 UNIVERGE SV9100 IP K-CCIS and Electra Elite IPK II

Programming Example 2
The following example provides programming details for three systems connected
by the IP CCH for NEAX application and one system connected by legacy (T1)
K-CCIS.
Point Code 00002

Address:
Point Code 00001 1st PVA: 192.168.1.10
2nd PVA: 192.168.1.11

Electra Elite
IPK II
T1 for K-CCIS
System A UNIVERGE
(1000) SV9100
System B
(1100)
IP Network
(Intranet)
UNIVERGE
SV9100
System C
(1200)
Point Code 00003

Address: 192.168.1.20
3.2.1 Chassis Programming Assignments
For this example, the following system configurations are used:
 A four-digit Closed Numbering Plan using Closed Number Blocks is

used.
 System A is a single-cabinet Electra Elite IPK II Expanded port

package with one B64-U20 KSU, one DTI-U( ) ETU, and a CCH(4)-U(
) ETU.
 System B is a single-chassis UNIVERGE SV9100 with one CHS2U

and one GPZ-IPLE installed on the GCD-CP10 and one GCD-CCTA
blade installed.
 System C is a single-cabinet UNIVERGE SV9100 port package with

one CHS2U and GPZ-IPLE Daughter Board installed on the
GCD-CP10.

Issue 5.0
 Each system has 16 multiline terminals.
 Each system has eight analog trunks.
Assume that the systems were defaulted (first power on) with the following
cards installed as described below.
Card Interface Slot Assignment (PRG 10-03)
SYSTEM A
Card Type And Ports

Card Type During 1st
Slot
Power On
Card Type Ports
1 ESI(8)-U10 ETU ESI(8)-U10 ETU Station 1 ~ 8
2 ESI(8)-U10 ETU ESI(8)-U10 ETU Station 9 ~ 16
3 None None N/A
4 COI(8)-U10 ETU COI(8)-U10 ETU Trunks 1 ~ 8
5 DTI-U( ) ETU DTI-U( ) ETU Trunks 9 ~ 32
6 CCH(4)-U( ) ETU CCH(4)-U( ) ETU N/A
7 None None N/A
8 None None N/A
0A None None N/A
System B
Card Type And Ports

Slot
Power On
Card Type Ports
GCD-CP10 with GPZ-IPLE GCD-CP10 with GPZ-IPLE Trunks 29 ~ 52

1 Blade/Daughter Board Blade/Daughter Board
2 GCD-8DLCA Blade GCD-8DLCA Blade Station 1 ~ 8
4 None None N/A
5 GCD-4COTA/C Blade GCD-4COTA/C Blade Trunks 1 ~ 4
6 GCD-CCTA Blade GCD-CCTA Blade Trunks 5 ~ 28

Issue 5.0
System C
Card Type And Ports

Slot
Power On
Card Type Ports
GCD-CP10 with GPZ-IPLE GCD-CP10 with GPZ-IPLE Trunks 5 ~ 28

1 Blade/Daughter Board Blade/Daughter Board
4 None None N/A
5 GCD-4COTA/C Blade GCD-4COTA/C Blade Trunks 1 ~ 4
6 None None None
This example shows digital trunk assignments for systems A, B and C.
Notice the Logical Trunk Notice the Logical Trunk Select Slot 1.
Number Number Notice the Logical Trunk
Number.
Program 10-03-02 Program 10-03-02 Program 10-19-01

0 = D4 (Multiframe) 0 = D4 (Multiframe) Assign DSP Resources
(Default) (Default) 001 ~ 024 as CCIS (LK 4).
1 = ESF (24 Multiframe) 1 = ESF (24 Multiframe)
0 = B8ZS (Default) 0 = B8ZS (Default) 0 = Disable (Default)

1 = ANI/ZCS 1 = ANI/ZCS 1 = Enable
Program 10-03-05 Program 10-03-05 *Program 10-40-04
Assign Clocking Assign Clocking Assign Number of Ports for

0 = Internal 0 = Internal CCIS for Slot 1 = 24.
1 = External (Default) 1 = External (Default)
Program 14-05-01 Program 10-19-01 Program 14-05-01
Trunks 9 ~ 32 Assign DSP Resources Trunks 5 ~ 28

Assign Trunk Group 10 001 ~ 024 as CCIS (LK 4). Assign Trunk Group 10

Issue 5.0
Program 22-02-01 Program 10-40-04 Program 22-02-01
Trunks 9 ~ 32 Assign Number of Ports for Trunks 5 ~ 28

Assign LK 6 (Tie Line) CCIS for Slot 1 = 24 Assign LK 6 (Tie Line)
Trunks 9 ~ 32 Trunks 5 ~ 28 Trunks 5 ~ 28

Assign LK 2 (Wink) Assign Trunk Group 11 Assign LK 2 (Wink)
Trunks 29 ~ 52
Assign Trunk Group 10
Program 22-02-01
Trunks 5 ~ 52
Assign LK 6 (Tie Line)
Program 34-01-01
Trunks 5 ~ 52
Assign LK 2 (Wink)
For CCISoIP only
3.2.3 CCH Assignment
This example shows CCH assignments for systems A, B and C.
Program 14-13-01 Program 44-05-10 Program 44-05-10
Select Trunk Group 10 F-Route Table 1 : 3 F-Route Table 1 : 2

Assign CCIS Route ID 1 F-Route Table 2 : 4 F-Route Table 2 : 4
Trunks 9 ~ 32 Select Trunk Group 10 Enable CCIS

Assign CIC 1 ~ 24 Assign CCIS Route ID 1
Enable CCIS Trunks 5 ~ 28 Route ID 9 : 3

Assign CIC 1 ~ 24
Program 50-02-01 Program 50-01-01 Program 50-03-01
Select CCIS Route ID 1 Enable CCIS System ID 1 : 2

Assign Channel Port 28 System ID 2 : 4

Issue 5.0
1. Select CCIS Route ID 1. Select CCIS Route ID 1

1 2. Assign Channel Port
2. Assign Data Speed LK 28
2 (56k)
Program 50-02-02
1. Select CCIS Route ID 1

2. Assign Data Speed LK
2 (56k)
Program 50-02-03
Route ID 9 : 2
Program 50-03-01
1. System ID 1 : 3
2. System ID 2 : 4
 For CCISoIP only
SECTION 4 PORT DESIGNATIONS
This section provides port number designations for IP applications.
IP Application IP Port Numbers Comments
SIP Trunk
SIP Trunk Signaling 5060 UDP
SIP Trunk Voice with GPZ-IPLE 10020~10531 UDP
3rd Party SIP
SIP SLT Signaling 5070 UDP
SIP SLT Voice with GPZ-IPLE 10020~10531 UDP
H.323 Trunk
H.323 Signaling 1718/1719/1720 TCP
H.323 Trunk Voice with GPZ-IPLE 10020~10531 UDP

Issue 5.0
SV9100 IP - K-CCIS
IP K-CCIS Signaling 57000/59000 TCP
IP Trunk Voice with GPZ-IPLE 10020~10531 UDP
SV9100 NetLink
Primary System Signaling 58000/58001 TCP
Secondary System Signaling 58002 TCP
IP DSP Resource with GPZ-IPLE 10020~10531 UDP
NEC Proprietary SIP (SIP MLT)
SIP MLT Signaling 5080 UDP
SECTION 5 CCIS NETWORKING VIA IP (PEER-TO-PEER CONNECTIONS BASIS)
Description
IP-KCCIS supports Peer-to-Peer calls between IP Terminals residing in different

offices, without using DSP resources. Two DSP resources in each office/system are
consumed for calls between an IP Terminal and an IP Terminal.
The number in each Main System Remote System

box indicates how
many VoIP CO CO Conf. CO Conf. CO CO Conf. CO Conf.
TDM IP TDM IP
Analog/ IP TDM Analog/ IP TDM
resources are used Terminal Terminal Terminal Terminal
Digital/SIP Terminal Terminal Digital/SIP Terminal Terminal
Main TDM 0 M: 1 0 M: 1 0 M: 1 M: 1 0 M: 1 M: 1
System Terminal R: 1 R: 1
IP M: 1 0 M: 1 M: 2 M: 1 R: 1 0 R: 1 M: 2 M: 2
Terminal R: 1
Remote TDM M: 1 R: 1 M: 1 R: 1 M: 1 0 R: 1 0 R: 1 M: 1
System Terminal R: 1 R: 1 R: 1 R: 1
IP M: 1 0 M: 1 R: 2 M: 1 R: 1 0 R: 1 R: 2 M: 1
Terminal R: 2 R: 2
M = Main K-CCIS System

R = Remote K-CCIS System

Issue 5.0
 For InMail remote CCIS extensions are not supported in a centralized directory.
 DT800/DT700 terminals are supported for Peer-to-Peer connections via a P2P
CCIS call.
 Standard SIP terminals are not supported for Peer-to-Peer connection.
 If either Programs15-05-50 or 50-15-04 are set to 0 (Disable) in system A, Peer-to-
Peer is disabled for system A and any remote systems when calling system A.
 When port translation is done through a NAT router, Peer-to-Peer is disabled.
 When RTP encryption is enabled, Peer-to-Peer is disabled.
 When connecting to other SV9100s using CCISoIP, Program 84-34-01 (Type 5
CCIS) must be set the same in all systems.
 SV9100 K-CCIS-IP to another SV9100, for calls from an IP terminal to a TDM
terminal/trunk via Peer-to-Peer, the IP Terminal's CODEC must match the CCIS
CODEC and the packet size is auto negotiated based on the receiving sides packet
size.
 SV9100 K-CCIS-IP to another SV9100 or to a NEAX PBX (SV9300, SV9500, etc.),
for calls from an IP terminal to another IP terminal/trunk via Peer-to-Peer, the IP
Terminal’s CODEC must match and the packet size is auto negotiated.
 SV9100 K-CCIS-IP to a NEAX PBX (SV9300, SV9500, etc.), for calls from an IP
terminal to a TDM terminal via Peer-to-Peer, the IP Terminal's CODEC and packet
size (Program 84-24-XX) need to match the NEAX PBX CCIS CODEC and packet
size settings.
 For this feature, the GPZ-IPLE is installed on the GCD-CP10. Only one GPZ-IPLE
can be installed per system for a maximum of 256 VoIP resources.
 The audio quality of speech connections depends on the available bandwidth
between the GPZ-IPLE daughter boards in the data network. As Internet is an
uncontrolled data network compared to an intranet, using this application in intranet
WAN environment with known, or controlled and assured, Quality of Service (QoS),
is highly recommended.
 The GCD-CCTA blade is not required to support this feature. It can be installed and
used in a system using traditional K-CCIS with point to point T1 lines allowing both
IP K-CCIS and traditional K-CCIS to be used with the same system.
 The LAN connection is provided by a 10 Base-T/100 Base-TX/1000 Base-T, Auto
sensing, full duplex Ethernet.
 If single lines for fax machines are set to Special (PRG 15-03-03), faxing across IP
CCIS will always use G.711 CODEC.
 SV9100 to SV9100, IP terminal to TDM via Peer-to-Peer, the IP Terminal's CODEC
must match the CCIS CODEC and the packet size will be auto negotiated based on
the receiving sides packet size.

Issue 5.0
 SV9100 to SV9100 or NEAX PBX (SV9300, SV9500, etc.), IP terminal to IP

terminal via Peer-to-Peer, the IP Terminals' CODEC must match and the packet size
is auto negotiated.
 SV9100 to NEAX PBX (SV9300, SV9500, etc.), IP terminal to TDM via Peer-to-
Peer, the IP Terminal's CODEC and packet size (PRG 84-24-XX) need to match the
NEAX PBX CCIS CODEC and packet size settings.
 Any calls across CCISoIP (station to station or stations transferring trunks) that use
Quick Transfer to voice mail will require an extra CAP key. The initial call across the
CCISoIP link will use the first CAP key and when the digit 8 is pressed, to perform
the Quick Transfer, a second CAP key is accessed.
 Verified Account Codes for Toll calls across a CCIS network will not be restricted
when a trunk access code is added to the number to allow ARS routing through
another K-CCIS T1/IP networked site. This access code (typically a 9) precedes the
dialed “1” which is used by the system to identify a LD call. As a result the call is no
longer considered LD and the account code will not be required.
 All IP trunks (SIP, CCIS, or H.323) must be contiguous. If any IP trunks are added to
a system that already has IP trunks installed, and the next set of trunks is not
contiguous, then all IP trunks are moved to a new set of contiguous trunk numbers.
6.1 Restrictions
 Call Forward Busy No Answer must be disabled for a phone to receive a CCIS
Call Back request.
 The UNIVERGE SV9100 can send billing information to a billing center office
(NEAX2000/2400/SV9300), but it cannot receive the billing information as the
billing center office.
 Voice Calls – K-CCIS, voice announce is not supported for a forwarded call
across IP K-CCIS to NEAX.
 Not all data networks are suitable to support Voice over Internet Protocol (VoIP).
A good VoIP solution requires a low-latency, low jitter and low packet loss
network. Accordingly, a network must be evaluated for latency, packet loss, and
jitter to qualify and determine if it can provide toll-quality speech paths.
 The GPZ-IPLE can support up to 256 VoIP resources dependent on the number
of VoIP Channel Licenses installed on the GCD-CP10 which may be configured
to support SIP MLT, 3rd Party SIP Stations and Trunks or IP K-CCIS
Applications.
 A GCD-CP10 license is required for IP K-CCIS trunks.
 A GCD-CP10 license is required for SIP Clients to include SIP stations and SIP
trunks.
 The GPZ-IPLE daughter board may configure VoIP DSP usage criteria with
SV9100 Program 10-19-01 DSP Resources. Each GPZ-IPLE can flag individual
DSP resources as:
■ IP Ext – IP Extensions (includes SIP MLT Station or 3rd Party SIP Stations)
■ SIP Trk – SIP Trunks

Issue 5.0
■ IP K-CCIS – CCIS Networking

■ Common – Common usage for CCIS Networking, 3rd Party SIP Station, SIP MLT
Stations, SIP Trunks.
■ Pressing Feature + 4 from any MLT Terminal shows the following:
 What type of GPZ-IPLE daughter board is installed on the GCD-CP10.
 How many VoIP resources are Active (call in progress) and Reserved (call
setup in progress).
 The MAC Address of the GPZ-IPLE.
 One way delay must not exceed 100 ms.
 Round trip delay must not exceed 200 ms.
 Packet loss must not exceed 1%.
 When an SIP MLT phone establishes a call via IP K-CCIS to another SIP MLT
phone in a remote system, a VoIP Resource is used from both systems.
 When a TDM phone establishes a call via IP K-CCIS and calls another TDM
phone in a remote system, a VoIP resource is used from both systems.
 When using InMail in a CCIS or Netlink network, 8-digit extensions and

mailboxes are not supported.

Issue 5.0

UNIVERGE® SV9100
General Information Chapter 1
SECTION 1 UNIVERGE SV9100 IP NETWORKING
UNIVERGE SV9100 is an enterprise IP Telephony solution that allows businesses

and organizations to converge their voice and data network to secure the many
advantages of IP telephony, while enjoying the hundreds of features that they have
come to expect from the telephone systems.
SECTION 2 VOICE OVER IP
Voice over IP (VoIP) is a technology that converts speech into data packets and
transmits these packets over TCP/IP networks. The technology also facilitates
compression and silence suppression to reduce network bandwidth demands.
As most organizations already have existing data networks in place, considerable

cost savings can be achieved by utilizing spare bandwidth on these networks for
voice traffic.
UNIVERGE SV9100 supports the use of IP Phones. These telephones provide the
same functionality as a multiline telephone but use the data network rather then the
traditional telecoms infrastructure. This can reduce costs and allow the use of
UNIVERGE SV9100 telephones in locations that would not normally be supported by
multiline telephones.
UNIVERGE SV9100 can also use VoIP technologies to connect two or more
telephone systems together. This can eliminate inter-site call charges, and can also
simplify calling between sites (as desk-to-desk dialing is possible).
Table 1-1 VoIP Specifications lists the specifications for various aspects of
UNIVERGE SV9100 VoIP system.

Issue 5.0
Table 1-1 VoIP Specifications
Category Feature Notes
IP Address DHCP Server GCD-CP10

DHCP Client IP Phone
QoS 802.1p/1q GCD-CP10
L3 QoS (ToS) Diffserv/IP Precedence
Maintenance HTTP Server GCD-CP10
VLAN Tag and port-based VLAN
VoCoder G.711 µ-law/A-law
G.729a
Jitter Buffer Size Set by system programming
RTP Length Set by system programming
Echo Canceller Tail Size Set by system programming
Level Adjustment Set by system programming
IP Phone SIP Phone SIP Phone
SIP Trunk SIP Trunk Maximum 400 Trunks
IP CCIS IP CCIS Trunks Maximum 400 Trunks
1-2 General Information

UNIVERGE® SV9100
IP Networking Chapter 2
SECTION 1 INTRODUCTION1
IP Networking uses VoIP technology to connect two or more telephone systems

together. This allows calls to be made between sites without using the public
telephone network. This saves considerable money, and makes communication
between sites much easier.
Two types of IP Networking are available on the UNIVERGE SV9100 CCIS Network
and SIP trunks. These methods are explained below.
SECTION 2 UNIVERGE SV9100 IP K-CCIS NETWORK SV9100
The SV9100 IP K-CCIS package provides a seamless connection of multiple

systems into a single virtual communications system using VoIP lines with a unified
numbering plan.
The SV9100 IP K-CCIS Network allows many offices to connect their UNIVERGE
SV9100 systems so they appear as one. This gives them the ability to have only one
operator to manage the system and share one voice mail system in the network.
An extension user in the network can easily dial another extension or transfer a call
in the SV9100 IP K-CCIS Network system. Calls are passed from network node to
network node using a protocol that contains information about the source of the call,
the type of call and the destination of the call.
SV9100 IP K-CCIS is explained in detail in the UNIVERGE SV9100

Key-Common Channel Interoffice Signaling (K-CCIS) section. Please refer to this for
a complete description and installation instructions.
1. The voice quality of VoIP is dependent on variables such as available bandwidth, network latency and Quality of Ser-
vice (QoS) initiatives, all of which are controlled by the network and Internet service providers. Because these vari-
ables are not in NEC control, it cannot guarantee the performance of the user’s IP-based remote voice solution.
Therefore, NEC recommends connecting VoIP equipment through a local area network using a Private IP address.

Issue 5.0
SECTION 3 IP TRUNKS
The SIP Trunks method of networking allows connection to SIP devices. This could
be a PBX system or a third-party product. When using SIP, the feature set is limited
and the advanced networking features cannot be used. If these features are
required, use IP K-CCIS.
Refer to SIP Trunking for a a detailed description of SIP trunking and for set up
instructions.
To set up IP trunks:
1. Connect the system to the Data Network. (Refer to General IP Configuration

for detailed instructions.)
2. Configure the IP trunks.

3. Configure the IP K-CCIS or SIP information.
4. Configure the F-Route.
3.1 Configure IP Trunks
When installing a GPZ-IPLE daughter board in the UNIVERGE SV9100 system,

external line ports are allotted in accordance with the number of Licensed ports for
the particular IP Application.
The UNIVERGE SV9100 system now has the required information about the
remote destinations and the SIP/IP K-CCIS configuration is complete. The only
remaining task is to configure F-Route to route calls to remote destinations via the
IP trunks. F-Route configuration is discussed in detail in the Automatic Route
Selection (ARS) feature in the UNIVERGE SV9100 Programming Manual. A basic
list of the programming commands required for F-Route is shown in the example
below.
2-2 IP Networking
Issue 5.0
SECTION 4 EXAMPLE CONFIGURATIONS
4.1 Network Configurations
Figure 2-1 Example IP Network Configuration shows four sites networked via IP
trunks. Each site has a Point Code and an IP address.
The programing for Office A and C is shown below. This would be sufficient
programming to make a call from Office A to Office C.
Figure 2-1 Example IP Network Configuration
Ext. 1000 Office A Office B Ext 1100
IP Address (GCD-CP10): IP Address (GCD-CP10):

192.168.1.10 192.168.1.20
Point Code: 00001 Point Code: 00002
IP Network
(Intranet)
PBX PBX
Office D
Office C
Ext. 1200 Ext. 1300

IP Address: 192.168.1.30 IP Address: 192.168.1.40

Issue 5.0
4.2 UNIVERGE SV9100 IP K-CCIS Programming Example 1
This is an example of UNIVERGE SV9100 program data assignments for a 4-digit

Closed Numbering Plan using Closed Number Blocks. The following system
configurations are used for all four systems:
 Each system is a single cabinet expanded port package with one CHS2U-xx.
 Each system has 16 multiline terminals.
 Each system has eight analog Trunks.

Address: GCD-CP10: 192.168.1.10 Address: 1st GCD-CP10 Resource: 192.168.1.20
UNIVERGE UNIVERGE
SV9100 SV9100
System A System B
(1000) (1100)
IP Network
(Intranet)
NEAX NEAX
System C System D
(1200) (1300)

Address: 192.168.1.30 Address: 192.168.1.40
 Systems C and D are NEAX systems. Refer to the applicable system

manuals for programming details.
2-4 IP Networking
Issue 5.0
It is assumed that the systems were defaulted (first power on) with the

NOTE
following blades installed as described below.:
4.2.1 Card Interface Slot Assignment (PRG 10-03 ETU Setup)
The following table provides information for assigning the blade interface
slots.
Card Type and Ports

Slot System A System B
Power On
Card Type Ports Card Type Ports
1 GCD-CP10/GPZ-IPLE GCD-CP10/GPZ-IPLE Trunks 5 ~28 GCD-CP10/GPZ-IPLE Trunks 5~28
2 GCD-8DLCA GCD-8DLCA Station 1~8 GCD-8DLCA Station 1~8
3 GCD-8DLCA GCD-8DLCA Station 9~16 GCD-8DLCA Station 9~16
5 GCD-4COTA/C GCD-4COTA/C Trunks 1~4 GCD-4COTA/C Trunks 1~4
Use the table below to make the appropriate assignments for digital trunks.
System A System B

Insert GPZ-IPLE on GCD-CP10 in Chassis 1, Slot 1. Insert GPZ-IPLE on GCD-CP10 in Chassis 1, Slot 1.
Verify PRG 10-03-01 (ETU Setup). Verify PRG 10-03-01 (ETU Setup).

Assign DSP Resources 001~024 as CCIS (LK4). Assign DSP Resources 001~024 as CCIS (LK4).
Program 10-68 Program 10-68

Assign the Trunk Type as CCIS with the Start Port Assign the Trunk Type as CCIS with the Start Port
as 5 and Number of Port 24. as 5 and Number of Port 24.

Assign Trunks 5~28 in Trunk Group 10. Assign Trunks 5~28 in Trunk Group 10.

Set Trunks 5~28 as Tie Line (LK6). Set Trunks 5~28 as Tie Line (LK6).

Issue 5.0
4.2.3 VoIP Address Assignments
Use the table below to make the appropriate assignments for VoIPDB
(GPZ-IPLE) addresses.
Item No. Selection
84-26-01 IPLE Basic Setup – IP XXX.XXX.XXX.XXX Assign the IPLE IP

Address Address
Default:
Slot 1 = 172.16.0.20
84-26-02 IPLE Basic Setup – RTP 0~65534 Default:

Port Number
RTP Port = 10020
84-26-03 IPLE Basic Setup – 0~65534 Default:

RTCP Port Number
(RTP Port Number + 1) RTCP Port = 10021
4.2.4 CCIS Availability
Use the table below to make the appropriate assignments for CCIS
availability.
System A System B

Select Option 1 (Enable) Select Option 1 (Enable)
Enables the CCIS Feature Enables the CCIS Feature

Assign Route ID 9 = 01 Assign Route ID 9 = 02
(Route ID 9 is designated for Host (Route ID 9 is designated for Host
Originating Point Code) Originating Point Code)
4.2.5 IP CCISoIP Assignment
Use the table below to make the appropriate assignments for CCIS over IP.
Item No. Selection
50-03-01 CCIS Destination System 0~16367 seconds PRG 50-03 assignments are
Settings – only used for CCISoIP.
Destination Point Code Default is 0
50-03-03 CCIS Destination System xxx.xxx.xxx.xxx

Settings – (xxx = 0~255) PRG 50-03 assignments are
IP Address (IP only) Default is 0.0.0.0 only used for CCISoIP.
50-03-04 CCIS Destination System 0:Disable

Settings – Point Code 1:Enable
Availability Default is 1
2-6 IP Networking
Issue 5.0
4.2.6 CCIS Assignment
Use the table below to make the appropriate system-wide CCIS

assignments.
System A System B

1. System ID 1~255 – Enter Destination 1. System ID 1~255 – Enter
Point Code Destination Point Code
2. System ID 1 = DPC 02 2. System ID 1 = DPC 01
1. System ID 1~255 – Enter Destination 1. System ID 1~255 – Enter
System IP Address. Destination System IP Address
2. System ID 1 = 192.168.1.20 2. System ID 1 = 192.168.1.10
3. System ID 2 = 192.168.1.21 3. System ID 2 = 192.168.1.11
CCIS Office Code Assignment Assign CCIS Office Code Assignment Assign
up to four digits. up to four digits.
Default not assigned. Default not assigned.

CCIS Centralized Response Timeout CCIS Centralized Response Timeout
Assignment - IAI Response Timer Assignment - IAI Response Timer
Enter 00~99. Enter 00~99.

CCIS Feature Availability - Link CCIS Feature Availability - Link
Reconnect Reconnect
0 = Not Available 0 = Not Available

1 = Available (default) 1 = Available (default)
CCIS Maximum Call Forwarding Hop CCIS Maximum Call Forwarding Hop
Counter Counter
1~7 1~7
7 = Default 7 = Default

Issue 5.0
4.2.7 Centralized Day Night Switching Assignments
Use the table below to make the appropriate assignments for centralized
day and night switching.
System A System B
PRG 50-06-02 CCIS Feature PRG 50-06-02 CCIS Feature

Availability - Centralized Day/Night Availability - Centralized Day/Night
Switching Switching
0 = Disable (default) 0 = Disable (default)
PRG 50-11-02 CCIS Centralized Day/ PRG 50-11-02 CCIS Centralized Day/
Night Switching Sending Group Night Switching Sending Group
Assignment - CCIS Route ID Assignment - CCIS Route ID
Send Group (1~8) Send Group (1~8)
CCIS Route ID CCIS Route ID
0 = No Setting (default) 0 = No Setting (default)
1~4 1~4
PRG 50-12-01 CCIS Centralized Day/ PRG 50-12-01 CCIS Centralized Day/Night
Night Mode to System Mode Mode to System Mode Assignment - Day
Assignment - Day Mode Mode
Day Mode = Mode 1~8 Day Mode = Mode 1~8
1 = default 1 = default
Night Mode = Mode 1~8 Night Mode = Mode 1~8
2 = default 2 = default
2-8 IP Networking
Issue 5.0
4.2.8 Centralized BLF Assignments
Use the table below to make the appropriate assignments for centralized
BLF.
System A System B
PRG 50-08-01 CCIS Centralized BLF PRG 50-08-01 CCIS Centralized BLF
Sending Group Assignment – Sending Group Assignment –
Destination Point Code Destination Point Code
Select Group ID (1~8) + Destination Select Group ID (1~8) + Destination
Point Code. Point Code.
Sending Group Assignment – CCIS Sending Group Assignment – CCIS
Route ID Route ID
Select Group ID 1~8 + CCIS Route ID. Select Group ID 1~8 + CCIS Route ID.
1~16367 1~16367
0 = Not Assigned (default) 0 = Not Assigned (default)
1~16367 1~16367
Sending Extension Number Assignment Sending Extension Number
– Extension Number Assignment– Extension Number
Select Table Number 1 ~120 and Enter Select Table Number 1 ~120 and Enter
Extension Number. Extension Number.
PRG 50-09-02 Centralized BLF Sending PRG 50-09-02 Centralized BLF
Extension Number Assignment – Send Sending Extension Number
to Sending Group 1 Assignment -Send to Sending Group 1
Select Table Number 1 ~120. Select Table Number 1 ~120.
0 = Disable (default) 0 = Disable (default)
Interval Time Assignment– Type of Interval Time Assignment– Type of
Interval Time Interval Time
0 = 4 seconds (default) 0 = 4 seconds (default)
1 = 8 seconds 1 = 8 seconds

Issue 5.0
4.2.9 Local Numbering Plan Assignment
Use the table below to make local numbering plan assignments.
Item No. Selection
11-01-01 System Numbering 0 = Not Used Defaults for 1X, 2X, 3X 4X, 5X, 6X=
1 = Service Code 2 Extension Number.
3 = Trunk Access
5 = Operator Access
11-02-01 Extension Assign Station Numbers to Defaults for Ports

Numbering Port Numbers Port 1 ~ 300 = 200 ~ 499
Port 301 ~ 960 = 5000 ~ 5659

Numbering Port Numbers All Virtual Extension Ports = No
Setting
4.2.10 Closed Numbering Plan - using Closed Number Blocks
Use the table below to make closed numbering plan assignments.
Item No. Selection
11-01-01 System Numbering 2 = Extension Number Defaults for 1X, 2X, 3X 4X, 5X, 6X=
2 Extension Number.
44-02-01 Dial Analysis Table for Up to eight digits can be Assign the digits to be dialed across
ARS/F-Route Access – assigned. the K-CCIS link. These digits were
Number of digits to be assigned as F-Route in Program
analyzed by the system 11-01-01
(Use line key 1 for “Don’t Care”
digit, @)
44-02-02 Dial Analysis Table for 0 = No Setting (None) The service type (2) assigns the
ARS/F-Route Access – 2 = ARS/F-Route Table digits to be dialed to an F-Route.
Service Type (F-Route) Program 44-02-03 assigns the F-
Route to be used.
Default is 0
44-02-03 Dial Analysis Table for 2 = 0~500 When setting data is (2), refer to
ARS/F-Route Access – (0=No Setting) Program 44-05.
Additional Data
Default is 0
44-05-01 ARS/F-Route Table – 0 = No Setting Select the trunk group to be used

Trunk Group Number 1 ~ 100 = Trunk Group for the outgoing K-CCIS call.
from PRG14-05
101 ~ 150 = Networking
Default is 0
2-10 IP Networking
Issue 5.0
4.2.11 Tandem Connections
Use the table below to make assignments for tandem connections.
System A System B
Trunk to Trunk Transfer Trunk to Trunk Transfer
0 = Disable 0 = Disable
1 = Enable (Default) 1 = Enable (Default)
Trunk to Trunk Outgoing Trunk to Trunk Outgoing

Caller ID Through Mode Caller ID Through Mode
0 = Disable (Off) (default) 0 = Disable (Off) (default)

1 = Enable (On) 1 = Enable (On)
SECTION 5 DTMF RELAY
When sending DTMF over an SV9100 CCISoIP network it may be necessary to

configure the item below. This allows the sending UNIVERGE SV9100 to detect
DTMF tones and convert them to data, then regenerate the DTMF tones at the
receiving side.
 84-34-01 VoIPDB Setup DTMF Relay Mode (05-CCIS over IP)

When using G.729, NEC recommends setting this item to (1) = RFC2833

Issue 5.0
2-12 IP Networking
UNIVERGE® SV9100
General IP Configuration Chapter 3
SECTION 1 INTRODUCTION
The UNIVERGE SV9100 system uses IP for various applications, including:

 System Programming
 Voice Over IP
 Switching
This section describes the procedure for connecting the UNIVERGE SV9100 system
to an existing data network and configuring TCP/IP. This is the first step in
implementing VoIP and other IP applications.
SECTION 2 NETWORK ADDRESSING OVERVIEW
Before connecting the system to a data network, it is necessary to obtain the relevant
IP Addressing information. This information is supplied by the IT Manager or Network
Administrator at the customer site.
2.1 IP Address
All equipment/devices used in the LAN setup must have an IP address

assignment. An IP address assigns a unique address for each device. There are
two types of IP addresses: Private and Global. A Private IP address is not
accessible through the Internet; a Global IP address can be accessed through the
Internet.
In most cases, a Private address is used, as LAN devices are not usually directly
connected to the Internet. Private addresses are usually taken from the following
ranges:
 Class A 10.0.0.0 ~ 10.255.255.255
 Class B 172.16.0.0. ~ 172.31.255.255
 Class C 192.168.0.0 ~ 192.168.255.255

Issue 5.0
A Public address is normally only used when a device is directly connected to the
Internet. This is unlikely in the case of the equipment. If Public addressing is
used, the numbers are normally allocated by an ISP.
2.2 Subnet Mask
As the IP address includes information to identify both the network and the final
destination, the Subnet Mask sets apart the network and destination information.
The default subnet masks are:
 Class A 255.0.0.0
 Class B 255.255.0.0
 Class C 255.255.255.0
The Subnet Mask is made up of four groups of numbers. When a group contains
the number 255, the router ignores or masks that group of numbers in the IP
address as it is defining the network location of the final destination.
For example, if the IP address is: 172.16.0.10 and the Subnet Mask used is Class
B (255.255.0.0), the first two groups of numbers (172.16) are ignored once they
reach the proper network location. The next two groups (0.10) are the final
destination within the LAN to which the connection is to be made.
 For sub-netted networks, the subnet mask may be different from the default subnet
masks listed above.
2.3 DHCP
Dynamic Host Configuration Protocol (DHCP) assigns a dynamic IP address.

Network control may be easier with DHCP as there is no need to assign and
program individual IP addresses for the LAN equipment. To use a dynamic IP
address, a DHCP server must be provided. The UNIVERGE SV9100 system
GCD-CP10 blade provides an internal DCHP server, enabling the ability to use
DHCP.
When equipment, which is connected to the LAN (the DHCP client), is requesting
an IP address, it searches the DHCP server.
When the request for an address is recognized, the DHCP server assigns an IP
address, Subnet mask, and the IP address of the router, based on UNIVERGE
SV9100 system programming.
Note that the GCD-CP10 blade must always have a static IP address. This
address is set in Program 10-12-01 : GCD-CP10 Network Setup – IP Address
(default: 192.168.0.10).
3-2 General IP Configuration

Issue 5.0
SECTION 3 CONFIGURATION EXAMPLES
The following configuration examples illustrate a typical network configuration for an

existing network that uses a static address and a typical configuration for a new
network that uses a dynamic address.
3.1 Example Configuration 1 - Existing Network with Static

Addressing
Figure 3-1 Example Configuration 1 - Existing Network with Static IP Address on

page 3-3 shows a typical network configuration that uses Static IP Addressing.
Each client device has a manually assigned IP address in the 192.168.1.0/24

network (i.e., 192.168.1.1 to 192.168.1.254 with a subnet mask of
255.255.255.0). They also have a default gateway address configured
(192.168.1.254) this allows the device to route packets to destinations that exist
outside of their own LAN.
Figure 3-1 Example Configuration 1 - Existing Network with Static IP Address
WAN,
Internet, etc.
Router
(Default Gateway)
192.168.1.254
Switch
(Switched Hub)
192.168.10.11 192.168.1.10
192.168.1.50
192.168.1.32

Issue 5.0
Assume that a UNIVERGE SV9100 is added to the existing data network. The
Network Administrator (or IT Manager) should provide the following:
 IP Address (for the GCD-CP10 blade)
 IP Addresses (for the GPZ-IPLE daughter board)
 Subnet Mask
 Default Gateway
 A spare switch/hub port
First, program the UNIVERGE SV9100:

 192.168.1.200
 255.255.255.0
 PRG10-12-03: 192.168.1.254
 A system reset is required for the IP Address changes to take effect.
Now connect the GCD-CP10 blade Ethernet Port to the switch/hub port, using a
standard Cat-5 patch cable. The UNIVERGE SV9100 is now configured on the
network and should be accessible by other devices on the network. Refer to
Figure 3-2 Example Configuration 1 - Adding the UNIVERGE SV9100 Chassis to
the Network.

Issue 5.0
Figure 3-2 Example Configuration 1 - Adding the UNIVERGE SV9100 Chassis to the
Network
WAN,
Internet, etc.
UNIVERGE SV9100
Chassis with
GCD-CP10 Installed
Router
(Default Gateway)
192.168.1.254
Switch
(Switched Hub)
192.168.1.200
192.168.1.50 192.168.10.11 192.168.1.10
192.168.1.32
3.2 Example Configuration 2 - New Network with Dynamic

Addressing
Figure 3-3 Example Configuration 2 - New Network with Dynamic Addressing on

page 3-6 shows a typical network configuration using Dynamic IP Addressing,
and the UNIVERGE SV9100 Internal DHCP server. In most cases, the customer
would use an external DHCP server (for example on a Windows Server) or static
addressing (as illustrated in Figure 3-2 Example Configuration 1 - Adding the
UNIVERGE SV9100 Chassis to the Network on page 3-5). However, if the
UNIVERGE SV9100 is to be installed on a new network the Network
Administrator may want to use the UNIVERGE SV9100 internal server (this is
called inDHCP).
In this example, the client PCs get an IP address, subnet mask, and default
gateway from the inDHCP server. The server also uses DHCP, but should always
be given the same IP address (192.168.1.32).

Issue 5.0
The Network Administrator (or IT Manager) should provide the following:

 IP Address (for the GCD-CP10 blade)
 IP Addresses (for the GPZ-IPLE)
 Subnet Mask
 Default Gateway
 Range(s) of IP Addresses to assign
 List of permanent IP addresses, with corresponding MAC Addresses
 A spare switch/hub port
Figure 3-3 Example Configuration 2 - New Network with Dynamic Addressing
WAN,
Internet, etc.
Router
(Default
Gateway)
192.168.1.254
Switch
(Switched Hub)
UNIVERGE SV9100
Chassis with GCD-CP10
Installed
192.168.1.200
DHCP Client DHCP Client DHCP Client
Server
(should always have
192.168.1.32 assigned)
Now connect the UNIVERGE SV9100 GPZ-IPLE Ethernet Port to the switch/hub
port, using a standard CAT-5 patch cable. The UNIVERGE SV9100 is now
configured on the network and its DHCP server is ready to allocate IP addresses.
The client PCs should be set to Obtain IP Address Automatically. Refer to Figure
3-4 TCP/IP Properties Screen on page 3-7.

Issue 5.0
Figure 3-4 TCP/IP Properties Screen
If the client PCs are now connected to the network (and restarted), they should be
assigned an IP address in the range 192.168.1.50 to 192.168.1.150, a subnet
mask of 255.255.255.0 and a default gateway of 192.168.1.254. When the server
tries to obtain an IP address, the inDHCP server allocates IP address
192.168.1.32, as it is statically assigned to the server MAC address.

Issue 5.0
SECTION 4 TESTING THE UNIVERGE SV9100 NETWORK CONNECTION
To test the UNIVERGE SV9100 network connection, it is possible to use the ICMP
(Internet Control Message Protocol) Ping command. This basically sends a small
amount of data from one device to another and then waits for a reply. This test
confirms that the IP addressing and physical connection are good. To perform this
test, from a Windows PC:
1. Click Start.
2. Click Run... .
3. In the Open dialogue box, enter command.
4. Click OK. A Command prompt window opens.
5. Type ping 192.168.1.200.
Figure 3-5 Testing the Network Connection shows that the UNIVERGE SV9100
system has replied to the Ping request – this indicates that the UNIVERGE SV9100
system is correctly connected to the network.
Figure 3-5 Testing the Network Connection

UNIVERGE® SV9100
Programming Chapter 4
SECTION 1 BEFORE YOU START PROGRAMMING
This chapter provides you with detailed information about the UNIVERGE SV9100
program blocks that may be required to connect the SV9100 to a data network and to
configure the VoIP function. The configuration and programming examples, found in
the earlier chapters, can be a useful reference when programming the data.
When using this chapter, note that the information on each program is subdivided
into the following headings:
Description describes what the program options control. The Default Settings for
each program are also included. When you first install the system, it uses the Default
Setting for all programs. Along with the Description are the Conditions which
describe any limit or special consideration that may applies to the program.
The reverse type (white on black) just beneath the Description heading is the
program access level. You can use only the program if your access level meets or
exceeds the level the program requires. Refer to Section 2 How to Enter
Programming Mode on page 4-2 for a list of the system access levels and
passwords.
Feature Cross Reference provides you with a table of all the features affected by
the program. You should keep the referenced features in mind when you change a
program. Customizing a feature may have an effect on another feature that you did
not intend.
Telephone Programming Instructions shows how to enter the program data into
system memory. For example:
To enter the programming mode:
1. 15-07-01
15-07-01 TEL
KY01 = *01

Dial 150701 from the telephone dial pad. See the message 15-07-01 TEL on the
first line of the telephone display. This indicates the program number (15-07), item
number (01), and that the options are being set for the extension.

Issue 5.0
The second row of the display KY01 = *01 indicates that Key 01 is being
programmed with the entry of *01. The third row allows you to move the cursor to
the left or right, depending on which arrow is pressed.
To learn how to enter the programming mode, refer to Section 2 How to Enter
Programming Mode below.
SECTION 2 HOW TO ENTER PROGRAMMING MODE
To enter programming mode:
1. Go to any working display telephone.

 In a newly installed system, use extension (port 1).
2. Do not lift the handset.
3. Press Speaker.
4. ##
Password
5. Dial the system password + Transfer.

Refer to the following table for the default system passwords. To change the
passwords, use 90-02 : Programming Password Setup.
User
Password Level Programs at this Level
Name
---- ---- 1 (MF) Manufacturer (MF):

All programs
12345678 tech 2 (IN) Installation (IN):
All programs in this section not listed below for SA
and SB
0000 ADMIN1 3 (SA) System Administrator - Level 1 (SA):
10-01, 10-02, 10-12, 10-13, 10-14, 10-15, 10-16,
10-17, 10-18, 10-22, 12-02, 12-03, 12-04, 15-01,
15-07, 15-09, 15-10, 15-11, 20-16, 21-07, 21-14,
22-04, 22-11, 25-08, 30-03, 32-02, 40-02, 41-02,
41-03, 41-04, 41-05, 41-06, 41-07, 41-08, 41-09,
41-10, 41-11, 41-12, 41-13, 41-14, 41-15, 41-16,
41-17, 41-18, 90-03, 90-04, 90-06, 90-07, 90-18,
90-19
9999 ADMIN2 4 (SB) System Administrator - Level 2 (SB):
13-04, 13-05, 13-06
4-2 Programming
Issue 5.0
SECTION 3 HOW TO EXIT PROGRAMMING MODE
To exit the programming mode:
To exit programming mode, first exit the programming options mode.
1. Press Answer to exit program options, if needed.
Program Mode
Base Service OP1 OP2
2. Press Speaker. If changes were made to the system programming, Saving System
Data is displayed.
3. When completed, the display shows Complete Data Save and exits the telephone to
idle.
 To save a customer database, a blank USB Drive is required. Insert the USB Drive
into the GCD-CP10 and, use Program 90-03, to save the software to the USB Drive.
(Use Program 90-04 to reload the customer data if necessary). A USB Drive can
hold only one customer database. Each database to be saved requires a separate
drive.
SECTION 4 USING KEYS TO MOVE AROUND IN THE PROGRAMS
Once you enter the programming mode, use the keys in the following chart to enter
data, edit data and move around in the menus.
Table 4-1 Keys for Entering Data
Keys for Entering Data
Use this key... When you want to...
0~9 and  Enter data into a program.

Transfer Complete the programming step you just made (e.g., pressing Enter on a
PC keyboard). When a program entry displays, press Transfer to bypass
the entry without changing it.
Conf Delete the entry to the left (e.g., pressing Backspace on a PC keyboard).
Hold Delete or clear all characters to the right of the cursor.
Answer Exit one step at a time from the program window currently being viewed.
For example, if programming item 5 in 15-03, pressing Answer allows you
to enter a new option in program 15-03. Pressing Answer again allows
you to select a new program in the 15-XX series. Pressing Answer a third
time allows you to enter a new program beginning with 1. Pressing
Answer one last time brings you to the beginning program display,
allowing you to enter any program number.

Issue 5.0
Table 4-1 Keys for Entering Data (Continued)
Redial Switch between the different input data fields by pressing Redial. The
cursor moves up to the top row of the display. Pressing Redial again
moves the cursor back to the middle row.
Line Keys Use preprogrammed settings to help with the program entry. These
settings vary between programs from LINE 1 = 0 (off) and LINE 2 = 1 (on)
to preset values for timers where LINE 1 = 5, LINE 2 = 10, LINE 3 = 15,
etc.
For programs with this option, the line key, which currently matches the
programmed setting, lights steady.
The display can also indicate Softkey, which will allow you to select the
values as well (-1 and +1 will step through these pre-programmed
settings.)
Line Key 1 Program a pause into an Speed Dialing bin.
Line Key 2 Program a recall/flash into an Speed Dialing bin.
Line Key 3 Program an @ into an Speed Dialing bin.
VOL  Scroll backward through a list of entry numbers (e.g., from extension etc.)
or through entries in a table (e.g., Common Permit Table).
If you enter data and then press this key, the system accepts the data
before scrolling forward.
VOL  Scroll forward through a list of entry numbers (e.g., from extension etc.) or
through entries in a table (e.g., Common Permit Table).
before scrolling backward.
SECTION 5 PROGRAMMING NAMES AND TEXT MESSAGES
Several programs (e.g., Program 20-16: Selectable Display Messages) require you
to enter text. Use the following chart when entering and editing text. When using the
keypad digits, press the key once for the first character, twice for the second
character, etc. For example, to enter a C, press the key 2 three times. Press the key
six times to display the lower case letter. The name can be up to 12 digits long.
Table 4-2 Keys for Entering Names
Use this keypad digit . . . When you want to. . .
1 Enter characters:
1 @ [ ¥ ] ^ _ ` { | }  Á À Â Ã Ç É Ê ì ó
2 Enter characters: A-C, a-c, 2.

3 Enter characters: D-F, d-f, 3.
4-4 Programming
Issue 5.0
4 Enter characters: G-I, g-i, 4.

5 Enter characters: J-L, j-l, 5.
6 Enter characters: M-O, m-o, 6.
7 Enter characters: P-S, p-s, 7.
8 Enter characters: T-V, t-v, 8.
9 Enter characters: W-Z, w-z, 9.
0 Enter characters:
0 ! “ # $ % & ’ ( ) ô Õ ú ä ö ü 
 Enter characters:
 + , - . / : ; < = > ? B E  S ¢ £
# # = Accepts an entry (only required if two letters on the same key
are needed – ex: TOM). Pressing # again = Space. (In system
programming mode, use the right arrow Softkey instead to accept
and/or add a space.)
Conf Clear the character entry one character at a time.
Hold Clear all the entries from the point of the flashing cursor and to the
right.
SECTION 6 USING SOFTKEYS FOR PROGRAMMING
Each UNIVERGE SV9100 display telephone provides interactive Softkeys for

intuitive feature access. The options for these keys will automatically change
depending on where you are in the system programming. Simply press the Softkey
located below the option you wish and the display will change accordingly.
_
Program Mode

Pressing the VOLUME  or VOLUME  will scroll between the menus.

Issue 5.0
SECTION 7 WHAT THE SOFTKEY DISPLAY PROMPTS MEAN
When using a display telephone in programming mode, various Softkey options are
displayed. These keys will allow you to easily select, scan, or move through the
programs.
Table 4-3 Softkey Display Prompts
Softkey Display Prompts
If you press this The system will. . .

Softkey . . .
back Go back one step in the program display.

You can press VOLUME  or VOLUME  to scroll forward or
backward through a list of programs.
 Scroll down through the available programs.
 Scroll up through the available programs.
select Select the currently displayed program.
 Move the cursor to the left.
 Move the cursor to the right.
-1 Move back through the available program options.
+1 Move forward through the available program options.
SECTION 8 PROGRAMS
This sections describes the programs used to connect the SV9100 to a data network
and to configure the VoIP functions.
4-6 Programming
Issue 5.0
Program 10 : System Configuration Setup

10-03 : ETU Setup
Level:
IN
Description
Use Program 10-03 : ETU Setup to setup and confirm the Basic Configuration data
for each blade. When changing a defined terminal type, first set the type to 0 and
then plug the new device in to have the system automatically define it or you may
have to reseat the blade.
The items highlighted in gray are read only and cannot be changed.
IMPORTANT
Input Data
For CNF PKG Setup

Physical Port Number 01~16
Item
Item Input Data Default
No.
01 Logical Port Number 0~960 0
For DLCA PKG Setup

Item
No.
01 Terminal Type (B1) 0 = Not set 0

1 = Multiline Terminal
3 = Bluetooth Cordless Handset
6 = PGDAD (Paging)
7 = PGDAD (Tone Ringer)
8 = PGDAD (Door Box)
9 = PGDAD (ACI)
10 = DSS Console
11 = -- Not Used ---

Issue 5.0
Item
No.
02 Logical Port Number 0 = Not set 0

(B1) 1 = Multiline Terminal (1~960)
6 = PGDAD (Paging) (1~8)
7 = PGDAD (for Tone Ringer)
(1~8)
8 = PGDAD (for Door Box) (1~8)
9 = PGDAD (for ACI) (1~96)
10 = DSS (1~32)
11 = --- Not Used ---
03 --- Not Used ---
B-Channel 2
Item
No.
06 Terminal Type (B2) 0 = Not set 0

6 = PGDAD (Paging)
7 = PGDAD (Tone Ringer)
8 = PGDAD (Door Box)
9 = PGDAD (ACI)
12 = APR(B2 Mode)
07 Logical Port Number (B2) 0 = Not set 0

6 = PGDAD (Ext. Speaker)
7 = PGDAD (Paging/Tone Ringer) (1~8)
8 = PGDAD (for Door Box) = (1~8)
9 = PGDAD (ACI) = (1~96)
12 = APR (for B2 mode) (193~512)
08 Multiline Telephone Type 0 = DT3** 0

1 = DT4**
09 Side Option Information 0 = No option 0

1 = 8LK Unit
2 = 16LK Unit
3 = 24ADM
10 Bottom Option 0 = No option 0

Information (Only applies 1 = APR
to DTL style telephones)
2 = ADA
3 = BHA
4 = Not Used
5 = BCA
4-8 Programming
Issue 5.0
B-Channel 2
Item
No.
11 Handset Option 0 = No option 0

Information 1 = PSA/PSD
2 = Bluetooth Cordless Handset
For LCA PKG Setup

Item
No.
03 Transmit Gain Level (S-Level) 1~57 (-15.5 +12.5dB) 32 (0dB)
04 Receive Gain Level (R-Level) 1~57 (-15.5 +12.5dB) 32 (0dB)
For COTA/C Unit Setup

Item
No.

Issue 5.0
For ODTB PKG Setup

Item
No.
02 2/4 Wire 0 = 2 Wire 1

1 = 4 Wire
03 E&M Line Control Method 0 = TYPE I 0

1 = TYPE V
For DIOP PKG Setup

Item
No.
01 LD/OPX Specification 0 = LD Trunk 0

1 = OPX
02 Logical Port Number 0 = 1~400 (LD Trunk) 0

1 = 1~960 (OPX)
4-10 Programming
Issue 5.0
For BRIA PKG Setup

ISDN Line Number 01~04
Item
No
01 ISDN Line Mode 0 = Not Used 1

1 = T-Point:1~400
2 = S-Point:1~960
3 = NW Mode (Leased Line): 1~256
4 = NW Mode (Interconnected Line):
1~256
5 = NW Mode (Interconnected Line,
Fixed Layer1=NT): 1~256
6 = S-Point (Leased Line): 1~960
02 Logical Port Number 0 = Not Used 0

 The starting port number of a BRI 1 = For T-Bus (1~400)
line is displayed. Two logic ports are
automatically assigned to a BRI line.
03 Connection Type 0 = Point-to-multipoint 0

1 = Point-to-Point
04 Layer 3 Timer Type 1~5 1

 Each timer value of Layer 3 is set up
for every type using Program 81-06
(T-Bus)
05 CLIP Information Announcement 0 = Disable 1

Based on this setting, the system 1 = Enable
includes a Presentation Allowed (1) or
Presentation Restricted (0) in the Setup
message to allow or deny the Calling
Party Number. Program 15-01-04 must
also be set to 1 if this option is enabled.
06 S-Point Connection Bus Mode 0 = Extended Passive Bus 0

1 = Short Passive Bus
07 S-Point DID Digit Range 0 ~ 4 0
08 Dial Sending Mode 0 = Enblock Sending 1

ISDN Protocol definition 1 = Overlap Sending
09 Dial Information Element 0 = Keypad Facility 1

ISDN Protocol definition 1 = Called Party Number
[Only when Dialing Sending Mode
(10-03-08) is set for 1 (Overlap Sending]
10 Master/Slave System 0 = Slave System 0

If set to 0, system is synchronized to 1 = Master System
network clock. If set to 1, system is not
synchronized to the network clock.
11 NW Mode Networking System Range 0 ~ 50 0

Issue 5.0
Item
No
14 S-Point Service Protocol 0 = Keypad Facility 0

1 = Specified Protocol
15 S-Point Call Busy Mode 0 = Alerting Message 0

1 = Disconnect Message
17 ISDN Line Ringback Tone 0 = Disable 0

If Telco does not provide ringback tone, 1 = Enable
SV9100 can if set to 1:Enable.
18 Type of Number 0 = Unknown 0

ISDN Protocol definition 1 = International number
2 = National number
3 = Network specific number
4 = Subscriber number
5 = Abbreviated number
19 Numbering Plan Identification 0 = Unknown 0

ISDN Protocol definition 1 = ISDN numbering plan
2 = Data numbering plan
3 = Telex numbering plan
4 = National standard numbering plan
5 = Private numbering plan
22 --- Not Used ---
23 Straight/Cross Wiring 0 = Auto 0

1 = Manual (Cross)
2 = Manual (Straight)
24 S-Point Power Feeding 0 = Disable 0

1 = Enable
25 S-Point DID Digit 0 = Disable 0

1 = Enable
4-12 Programming
Issue 5.0
For PRTA PKG Setup

Item
No.
01 --- Not Used ---
02 Logical Port Number 1 = for T-Bus 1~400 1

 The start port number of a PRI line
is displayed.
03 --- Not Used ---

 Each timer value of Layer 3 is set up
for each type in
Program 81-06 (T-Bus)
05 CLIP Information 0 = Disable 1

Based on this setting, the system 1 = Enable
06 Length of Cable 0 = Level 1 2

1 = Level 2
2 = Level 3
3 = Level 4
4 = Level 5
07 --- Not Used ---
08 Dial Sending Mode 0 = Enbloc Sending 1

ISDN Protocol definition 1 = Overlap Sending
09 Dial Information Element 0 = Keypad Facility 0

ISDN Protocol definition 1 = Called Party Number
(Only when Dialing Sending Mode
(10-03-08) is set for 1 (Overlap
Sending)
10 --- Not Used ---
11 --- Not Used ---
12 --- Not Used ---

Issue 5.0
Item
No.
13 Loss-Of-Signal Detection Limit 0 = Level 0 (lowest sensitivity) 2

If the transmit/receive voltage is less 1 = Level 1
than the setting in 10-03-13, the system 2 = Level 2
considers this as Loss-Of-Signal and
3 = Level 3
the PRTA does not come up. Note that
there are different values based on the
4 = Level 4
setting in 10-03-12 for the PRI. 5 = Level 5
6 = Level 6
7 = Level 7 (highest sensitivity)
14 --- Not Used ---
15 --- Not Used ---
16 --- Not Used ---
17 --- Not Used ---
18 Type of Number 0 = Unknown 0

ISDN Protocol definition. 1 = International number
Select the number type for the ISDN 2 = National number
circuit. 3 = Network Specific number
4 = Subscriber number
5 = Abbreviated number
19 Numbering Plan Identification 0 = Unknown 0

ISDN Protocol definition. 1 = SDN numbering plan
Select the Numbering Plan used for the 2 = Data numbering plan
ISDN circuit. 3 = Telex numbering plan
4 = National standard numbering
plan
5 = Private numbering plan
20 Network Exchange Selection 0 = Standard (same as NI-2) 0

Select the ISDN protocol for the ISDN 1 = reserved
circuit. 2 = reserved
3 = DMS (A211)
4 = 5ESS
5 = DMS (A233)
6 = 4ESS
7 = NI-2
21 Number of Ports 0 = Auto 0

1 = 4 Ports
2 = 8 Ports
3 = 12 Ports
4 = 16 Ports
5 = 20 Ports
22 --- Not Used ---
4-14 Programming
Issue 5.0
Item
No.

1 = Manual (Cross)
For DTI (T1) PKG Setup

Item
No.

The start port number of a T1 line is
displayed, and 24 logic ports are
automatically assigned to a DTI (T1)
line.
02 T1 Signal Format Selection 0 = D4 (12 Multi Frame) 0

1 = ESF (24 Multi Frame)
03 Zero Code Suppression 0 = B8ZS 0

1 = AMI/ZCS
04 Line Length Selection 0 = 0 feet ~ 133 feet 0

1 = 133 feet ~ 266 feet
2 = 266 feet ~ 399 feet
3 = 399 feet ~ 533 feet
4 = 533 feet ~ 655 feet
05 T1 Clock Source 0 = Internal 1

1 = External

1 = 4 Ports
2 = 8 Ports
3 = 12 Ports
4 = 16 Ports
5 = 20 Ports

1 = Manual (Cross)

Issue 5.0
For IPLE PKG Setup

Item
No.
01 VOIP Type 0 = Not Used None (will be

1 = Not Used determined
2 = Not Used when
3 = IPLE Installed)
02 Number of Channels 0 ~ 256 (Read Only) 0
03 Number of Voice Channels 0 ~ 256 (Read Only) 0
For VM00 PKG Setup

Item
No.
For CCTA PKG Setup

Item
No.

line.
02 T1 Signal Format Selection 0 = D4 (12 Multi Frame) 1

1 = ESF (24 Multi Frame)
03 Zero Code Suppression 0 = B8ZS 0

1 = AMI/ZCS
04 Line Length Selection 0 = 0 feet ~ 133 feet 0

1 = 133 feet ~ 266 feet
2 = 266 feet ~ 399 feet
3 = 399 feet ~ 533 feet
4 = 533 feet ~ 655 feet
05 T1 Clock Source 0 = Internal 1

1 = External
4-16 Programming
Issue 5.0
Item
No.

1 = 4 Ports
2 = 8 Ports
3 = 12 Ports
4 = 16 Ports
5 = 20 Ports

1 = Manual (Cross)
Conditions
 When changing a defined terminal type, first set the type to 0 and then plug the
new device in to have the system automatically define it, or redefine the type
manually.
 The system must have a blade installed to view/change the options for that type
of blade.
Feature Cross Reference

 Universal Slots

Issue 5.0

10-04 : Music On Hold Setup
Level:
IN
Description
Use Program 10-04 : Music on Hold Setup to set the Music on Hold (MOH) source.
For internal Music on Hold, the system can provide a service tone callers on hold or
one of eleven synthesized selections.
Input Data
Item
Item Input Data Default Description
No.
01 Music on Hold 0 = Internal MOH(IC) 2 The Music on Hold (MOH)

Source Selection 1 = External MOH source can be internal
2 = Service Tone (synthesized) or from a
customer-provided music
3 = VMDB
source.
The customer-provided source
can connect to a PGDAD or
the connector on the side of
the Base Cabinet MOH/IN
connection.
Trunk MOH and Extension
MOH music source use the
same Music on Hold source.
02 Music on Hold [In case Item 1 is 0]

Tone Selection 1 = Download File1
2 = Download File2
3 = Download File3
[In case Item 1 is 1, 2,
or 3]
1~100 = VRS Message
Number
03 Audio Gain Setup 1~57 (-15.5 ~ +12.5dB) 32

(0dB)
Conditions
None
4-18 Programming
Issue 5.0

 Analog Communications Interface (ACI)
 Background Music
 Music on Hold

Issue 5.0

10-12 : GCD-CP10 Network Setup
Level:
SA
Description
Use Program 10-12 : GCD-CP10 Network Setup to setup the IP Address,
Subnet-Mask, and Default Gateway addresses.
Input Data
Item
No.
01 IP Address 0.0.0.0 ~ 126.255.255.254 192.168.0.10 Set for GCD-CP10.

128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
02 Subnet 128.0.0.0 192.0.0.0 224.0.0.0 255.255.255.0 The setting of Subnet

Mask 240.0.0.0 248.0.0.0 252.0.0.0 Mask is invalid when
all Host Addresses
254.0.0.0 255.0.0.0 255.128.0.0
are 0.
255.192.0.0 255.224.0.0 255.240.0.0
If the network section
255.248.0.0 255.252.0.0 255.254.0.0
is:
255.255.0.0 255.255.128.0 255.255.192.0
0,
255.255.224.0 255.255.240.0 255.255.248.0
127,
255.255.252.0 255.255.254.0 255.255.255.0
255.255.255.128 255.255.255.192 255.255.255.224 128.0,
255.255.255.240 255.255.255.248 255.255.255.252 191.255,
255.255.255.254 255.255.255.255 192.0.0,
223.255.255
The setting of Subnet
Mask is invalid.
03 Default 0.0.0.0 ~ 126.255.255.254 0.0.0.0 IP Address for Router.

Gateway 128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
4-20 Programming
Issue 5.0
Input Data (Continued)
Item
No.
04 Time Zone 0~24 (0 = -12 Hours and 24 = +12 Hours) +7 Determine the offset
(-5 hours) from Greenwich Mean
Time (GMT) time.
Then enter its
respective value. For
example, Eastern
Time (US and
Canada) has a GMT
offset of -5. The
program data would
then be 7 (0= -12, 1=
-11, 2= -10, 3= -9, 4= -
8, 5= -7, 6= -6,
7= -5, ……24= +12)
05 NIC 0 = Auto Detect 0 NIC Auto Negotiate

Interface 1 = 100Mbps, Full-Duplex (GCD-CP10)
2 = 100Mbps, Half-Duplex
3 = 10Mbps, Full-Duplex
5 = 1Gbps, Full Duplex
07 NAPT 0.0.0.0 ~ 126.255.255.254 0.0.0.0 Set the IP address on

Router IP 128.0.0.1 ~ 191.255.255.254 the WAN side of
Address router.
(Default 192.0.0.1 ~ 223.255.255.254
Gateway
[WAN])
08 ICMP 0= (Enable) 0 When receiving ICMP

Redirect 1= (Disable) redirect message, this
determines if the IP
Routing Table
updates automatically
or not.
09 IP Address 0.0.0.0 ~ 126.255.255.254 172.16.0.10 Set for IPLE.

128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
10 Subnet 128.0.0.0 192.0.0.0 224.0.0.0 255.255.0.0 Set for IPLE.

Mask 240.0.0.0 248.0.0.0 252.0.0.0
254.0.0.0 255.0.0.0 255.128.0.0
255.192.0.0 255.224.0.0 255.240.0.0
255.248.0.0 255.252.0.0 255.254.0.0
255.255.0.0 255.255.128.0 255.255.192.0
255.255.224.0 255.255.240.0 255.255.248.0
255.255.252.0 255.255.254.0 255.255.255.0
255.255.255.128 255.255.255.192 255.255.255.224
255.255.255.240 255.255.255.248 255.255.255.252
255.255.255.254 255.255.255.255

Issue 5.0
Item
No.
11 NIC Setup 0 = Auto Detect 0 Set for IPLE.

5 = 1 Gbps, Full-Duplex
13 DNS 0.0.0.0 Set for adding a

Primary function for DNS.
Address 0.0.0.0 ~ 126.255.255.254
128.0.0.1 ~ 191.255.255.254
14 DNS 192.0.0.1 ~ 223.255.255.254
Secondary
Address
15 DNS Port 0 ~ 65535 53
17 IPL NIC 0 = MDI 0

Port Setting 1 = MDI-X
Conditions
None

 Voice Over Internet Protocol (VoIP)
4-22 Programming
Issue 5.0

10-13 : In-DHCP Server Setup
Level:
SA
Description
Use Program 10-13 : In-DHCP Server Setup to setup the DHCP Server built into
the GCD-CP10 blade.
Input Data
Item
No.
01 DHCP Server 0 = Disable 0 Enable/Disable the built-in

Mode 1 = Enable DHCP Server.
02 Lease Time Days 0~255 0 day Lease Time of the IP address to

a client.
Hour 0~23 0 hour  Press Transfer to increment

to the next setting data.
Minutes 0~59 30 minutes
05 Last DHCP Data 0 = Disable 1 If 10-13-01 is enabled, Enable/

1 = Enable Disable DHCP resource.
Conditions
 The System must be reset in order for these changes to take affect.


Issue 5.0

10-14 : Managed Network Setup
Level:
SA
Description
Use Program 10-14 : Managed Network Setup to set up the range of the IP
address which the DHCP Server leases to a client.
Item
No.
01 The range of the IP Minimum: 172.16.0.100 When Maximum has not been
address to lease 0.0.0.0 ~ 126.255.255.254 entered, the maximum value equals
the minimum value.
128.0.0.1 ~ 191.255.255.254
When Single is selected in 10-13-04,
192.0.0.1 ~ 223.255.255.254
only 1 scope range can be entered.
Maximum: 172.16.5.254 When Divide Same Network is
0.0.0.0 ~ 126.255.255.254 selected in 10-13-04, a maximum of
10 scope ranges can be entered.
128.0.0.1 ~ 191.254.255.254
192.0.0.1 ~ 223.255.255.254
Conditions
None

4-24 Programming
Issue 5.0

10-15 : Client Information Setup
Level:
SA
Description
Use Program 10-15 : Client Information Setup to set up the client information
when the DHCP server needs to assign a fixed IP address to clients.
Input Data
Client Number 1~960
Item
No.
MAC: 00-00-00-00-00-00 ~ 00-00-00-00-00-00

The IP address should be FF-FF-FF-FF-FF-FF
assigned out of the scope
01 range set up in Program 10- 1.0.0.0 ~ 126.255.255.254 0.0.0.0
14. 128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
Conditions
None


Issue 5.0

10-16 : Option Information Setup
Level:
SA
Description
Use Program 10-16 : Option Information Setup to set up the option given from the
DHCP server to each client.
Input Data
Item
No.
01 Router Code number 0~255 3 (Fixed)

Set the Router IP address.
IP address 0.0.0.0
0.0.0.0 ~ 126.255.255.254
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
02 DNS Server Code number 0~255 6 (Fixed)

Set IP address of DNS Server.
IP address 0.0.0.0
0.0.0.0 ~ 126.255.255.254
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
03 TFTP Server Code number 0~255 66 (Fixed)

Set the name for the TFTP
Server. Maximum 64 character strings No setting
05 MGC Code number 0~255 129 (Fixed)

IP address 172.16.0.10
0.0.0.0 ~ 126.255.255.254
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
06 Client Host Name Code number 0~255 12 (Fixed)

Set the Client Host Name.
Maximum 64 character strings No setting
07 DNS Domain Name Code number 0~255 15 (Fixed)
Set the DNS Domain Name.
08 Download Protocol Code number 0~255 43 (Fixed)
Set Download Protocol used
for AutoConfig (for DT800/ Sub code number 163
DT700 Series).
1 = FTP 1
2 = HTTP
4-26 Programming
Issue 5.0
Item
No.
09 Encryption Information Code number 0~255 43 (Fixed)

Set an Encryption Information
used for AutoConfig (for Sub code number 164
DT800/DT700 series).
Maximum 128 character No setting
strings
10 FTP Server Address Code number 0~255 43 (Fixed)

Set a FTP Server Address
used for AutoConfig. Sub code number 141
IP address 0.0.0.0
0.0.0.0 ~ 126.255.255.254
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
11 Config File Name Code number 0~255 43 (Fixed)

Set a File Name used for
AutoConfig. Sub code number 151
12 Vender Class ID Code number 0~255 60 (Fixed)
Maximum 256 character NECDT700
strings
13 SNMP Server Code number 0~255 69 (Fixed)

IP address 0.0.0.0
0.0.0.0 ~ 126.255.255.254
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
14 POP3 Server Code number 0~255 70 (Fixed)

IP address 0.0.0.0
0.0.0.0 ~ 126.255.255.254
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
16 SIP Server (IP Address) Code number 0~255 120 (Fixed)

IP address
0.0.0.0 ~ 126.255.255.254
172.16.0.10
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
17 SIP Server (Domain Name) Code number 0~255 120 (Fixed)


Issue 5.0
Item
No.
18 FTP Server Code number 0~255 141 (Fixed)

IP address
0.0.0.0 ~ 126.255.255.254
0.0.0.0
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
19 Config File Name Code number 0~255 151 (Fixed)

20 LDS Server 1 Code number 0~255 162 (Fixed)
IP address
0.0.0.0 ~ 126.255.255.254
0.0.0.0
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254

IP address
0.0.0.0 ~ 126.255.255.254
0.0.0.0
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254

IP address
0.0.0.0 ~ 126.255.255.254
0.0.0.0
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254

IP address
0.0.0.0 ~ 126.255.255.254
0.0.0.0
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
24 Next Server IP Address IP address

0.0.0.0 ~ 126.255.255.254
0.0.0.0
128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
27 SIP Server Receive Port Code number 0~255 168 (Fixed)

Port: 1~65535 5080
Code Number 0~255 43 (Fixed)
28 Config File Name Subcode Number 0~255 152 (Fixed)

Up to 15 characters No setting
4-28 Programming
Issue 5.0
Conditions
None


Issue 5.0

10-19 : VoIP DSP Resource Selection
Level:
SA
Description
Use Program 10-19 : VoIP DSP Resource Selection to define the criteria for each
DSP resource on the VoIP blade.
Input Data
Slot Number 1
Input Data
DSP Resource Number 01~256
Input Data
Item
No.
01 VoIP DSP Resource Selection 0 = Common use for both IP Resource 1 = 1

extensions and trunks Resource 2~256 = 0
1 = IP Extension
2 = SIP Trunk
3 = Networking/CCIS
4 = Use for NetLink
5 = Blocked
paging
7 = Multicast paging
8 = Unicast paging
Conditions
None

 None
4-30 Programming
Issue 5.0

10-54 : License Configuration for Each Package
Level:
IN
Description
Use Program 10-54 : License Configuration for Each Package to set the license
information for each unit.
Input Data
Slot Number 1~24
License Index Number 1~32
Item
Item Read Data
No.
01 License Code 0000~9999
02 License Quantity 0~255
Conditions
 If applying more than 255 licenses to a slot the licenses must be applied across
multiple indexes. For example assigning 256 VoIP resource licenses (5103) to
the CPU slot could be assigned using different methods as long as the total for
the CPU slot is 256:
1. Index 1 has 128 of feature code 5103 and index 2 also has 128 of feature
2. Index 1 has 255 of feature code 5103 and index 2 has 1 of feature code
5103 for a total of 256.
 When using IP devices IP Resource licenses (5103) must be assigned to the
CPU Slot (1) for them to be available for use. If this is not done, IP related
features will not work.

None

Issue 5.0
Program 20 : System Option Setup

15-05 : IP Telephone Terminal Basic Data Setup
Level:
IN
Description
Use Program 15-05 : IP Telephone Terminal Basic Data Setup to set up the basic
settings for an IP telephone.
Input Data
Extension Number Maximum eight digits
Item Related
No. Item Input Data Default Description Program
01 Terminal Type 1 = H.323 3 Viewing Only – No changes

2 = SIP permitted
3 = None
4 = DT800/DT700
02 Terminal MAC MAC address 00-00-00-00-00-00 MAC Address of registered 15-05-01

Address MLT SIP phone is stored
00-00-00-00-00-00 to
FF-FF-FF-FF-FF-FF and/or can input the MAC
address of an MLT SIP
phone so when it comes
online it will be provided with
the extension in which the
MAC address matches.
04 Nickname Up to 48 characters No setting Nickname section on Invite

message.
07 Using IP 0.0.0.0~255.255.255.255 0.0.0.0 Informational Only 15-05-01

Address
registered IP Phones
09 Call Procedure 0 ~ 65535 0

Port
11 DT800/DT700 0 ~ 65535 0
C/CTR Port
15 CODEC Type 1-Type 1 1 Assign the CODEC Type of 84-24-XX

2-Type 2 the MLT SIP.
3-Type 3
4-Type 4
5-Type 5
16 Authentication Up to 24 characters None Assign the authentication 15-05-01

Password password for SIP single line
telephones.
4-32 Programming
Issue 5.0
Item Related
No. Program
18 IP Duplication 0 = Disable 0 When this program is 1 = 10-03-09

Allowed Group Enable, the duplication of an 15-05-22
1 = Enable
IP address is allowed at the
time of SIP/DT700 terminal
registration.
19 Side Option 0 = No Option 0 This is a read only program 10-03-09

Information that shows what type of Line 15-05-22
1 = 8LK Unit
2 = 16LK Unit Key unit is installed on the
ITH-style telephone.
3 = 24ADM
4 = EHS
20 Bottom Option 0 = No Option 0 This is a read only program 10-03-10

Information that shows what type of
1 = ADA
2 = BHA adapter is installed on the
3 = Not Used
4 = BCA
21 Handset 0 = Normal Handset 0 This is a read only program 10-03-11

Option that shows what type of 15-05-23
Information
1 = Handset for power
failure (PSA/PSD) Handset is installed on the
2 = BCH
22 Side Option 0 = No Setting 0 This is a read only program 30-01

Additional Data that shows the DSS console 30-02
1~32 = DSS Console
number number when one is 30-03
installed on a ITH-style 30-04
telephone. 30-05
30-06
23 Handset 0 = No Setting 0
Option 1~16 = Terminal
Additional
Information equipment number
(TEN) of Bluetooth
Determine to
use TEN or not. Cordless Handset
(BCH)
24 Protection 0 = Not Used 0 Enable this to allow the MLT 90-40-01

Service SIP telephones to use the 90-40-02
1 = Used
security key. If disabled, and
the key is pressed, nothing
happens.

Issue 5.0
Item Related
No. Program
26 D800/DT700 0 = Not Set 0

Terminal Type
1 = ITL-**E-1D/IP-*E-1
2 = ITL-**D-1D/ITL-
12BT1D/ITL- 12PA-1D
[without 8LKI(LCD)-L]
3 = ITL-**D-1D/ITL-12BT-
1D/ITL-12PA-1D [with
8LKI(LCD)-L]
4 = ITL-320C-1
5 = Softphone
6 = CTI
7 = AGW
8 = IP3PH-8WV
9 = Not Used
10 = ITL-**DG-3
11 = ITL-**CG-3
12 = ITL-2CR-1
13 = ITZ-**D-*D/ITZ-
**PD-*D/ITZ-**pA-
*D/ITZ-**DG
14 = ITZ-*CG
15 = ITZ-**LDG
16 = ITY-6D
17 = ITY-8LDX
27 Personal ID 0~960 0 Use when the SIP Multiline 84-22-XX

Index telephone is using manual/
auto registration. Assign
each phone a unique
personal index. Then go to
command 84-22 to assign
the user name and
password.
28 Addition 0 = Disable 0
Information
Setup
1 = Enable
Select whether
to inform of
additional
information or
not.
29 Terminal WAN- 0.0.0.0~255.255.255.255 0.0.0.0

side IP
Address
30 DTMF play 0 = Do Not Play 0

during
conversation 1 = Play
at Receive
Extension
4-34 Programming
Issue 5.0
Item Related
No. Program
31 Alarm Tone 0 = Not Ringing 1

during
conversation 1 = Ringing
(RTP packet
loss alarm)
32 Key Calling 0 = Not Used 0

1 = Used
33 LAN Side IP 0.0.0.0~255.255.255.255 0.0.0.0. Read-only

Address of
Terminal
34 Terminal Touch 0 = Off 1 Set whether the touch

Panel On/Off 1 = On screen used on ITL-320C-1
(BK) TEL can be used (On)
or cannot be used (Off).
35 Encryption 0 = Off 0
Mode
1 = On
36 DT800/DT700 00.00.00.00~FF.FF.FF.FF 00.00.00.00

Firmware
Version
37 DT800/DT700 2 = Red 2 Sets LED color for internal

Large LED 3 = Green Intercom call. In DT800/
Illumination DT700 local terminal setting
Setup 4 = Blue
menu, illumination setting
5 = Yellow
must be ‘Automatic’,
6 = Purple otherwise the terminal will
7 = Light Blue ignore PRG 14-01-35, PRG
8 = White 15-05-37 and PRG 15-23
9 = Rotation settings.
38 Paging 0 = Multicast 0 Sets the protocol mode for

Protocol Mode the Paging function.
1 = Unicast
2 = Auto
39 CTI Override 0 = Disable 0 Sets the override function

Mode against the terminal that is
1 = Enable
controlled by the CTI.
40 Calling Name 0 = Both name and 0 Sets the incoming calling

Display Info via number name display type on a
Trunk for standard SIP terminal. Trunk
Standard SIP 1 = Name only
name is the first priority and
2 = Number only
abbreviated (SPD) name is
3 = None second priority.
41 Time Zone 0~24 (-12~+12) 12 Sets the time difference from

(Hour) the system time set in
Program 10-01. Input hour(s)
based on this Program.

Issue 5.0
Item Related
No. Program
43 Video Mode 0 = Disable 0 This Program is used to

1 = Enable select the video function with
the standard SIP terminal. If
the standard SIP terminal
supports the video function,
the SV9100 transfers the
video CODEC in SDP
information.
44 Using 0 = Disable 0 This Program is used to

Standard SIP Enable or Disable the
Display for
1 = Enable
system to send INVITE from
CPN
tag Display attribute which is
sent from a standard SIP
terminal as CPN to ISDN,
and if there is no Display
attribute from standard SIP
terminal, the system will not
refer to either PRG 21-12-01
or 21-13-01 and no CPN will
be sent.
45 NAT Plug & 0 = Disable 0 Select sending RTP port 10-46-14

Play number to remote router. (0)
1 = Enable
uses result from negotiation
result, (1) from received RTP
packet. Effective only when
10-46-14 is to NAT Mode.
46 Door Phone 0 = Not assigned 0 Indicates automatically 10-03

Number (Read 1 ~ 8 = Door Phone assigned IP Door Phone (DLCA)
Only) Number after system PGD
Number
registers the Door Phone
port. System assigns the
number not to duplicate with
the Door Phone connected
to PGDAD . (Read Only)
47 Registration 0 = Disable 180 On a per station basis, this 10-46-14

Expiration setting defines the SIP
Timer for NAT
60 ~ 65535(sec)
registration expiry timer. If
this value is set to 0, for a
NAPT terminal, the value in
PRG 84-23- 01 is applied.
48 Subscribe 0 = Disable 180 On a per station basis, this 10-46-14

Expire Timer setting defines the SIP
for NAT
60 ~ 65535(sec)
Subscribe expiry timer. If this
value is set to 0, for a NAPT
terminal, the value in PRG
84-23-02 is applied.
4-36 Programming
Issue 5.0
Item Related
No. Program
49 Receiving SIP 0 = Disable 1 Select whether or not

INFO system can receive DTMF
1 = Allowed any time
2 = Allowed while RTP is from standard SIP phone via
SIP INFO message. There
not available
are two receive types.
1=’Allowed any time' can
receive a SIP INFO
message from a standard
SIP phone as a dial
information any time.
'2=Allowed while RTP is not
available' can receive a SIP
INFO message before
establishing RTP
connection.
50 Peer to Peer 0 = Disable On On a per station basis

Mode enables P2P.
1 = Enable
Conditions
 15-05-04 – Nickname must be unique in the system.
 The system programming must be exited before these program options take
affect.


Issue 5.0
Program 84 : Hardware Setup for VoIP

84-09 : VLAN Setup
Level:
IN
Description
Use Program 84-09 : VLAN Setup to set up the VLAN data.
Input Data
Item
No.
01 VLAN 0 = Disable (Off) 0

1 = Enable (On)
02 VLAN ID 1~4094 0
03 VLAN Priority 0~7 0
Conditions
 System programming must be exited before these program options take affect.

4-38 Programming
Issue 5.0

84-10 : ToS Setup
Level:
IN
Description
Use Program 84-10 : ToS Setup to set up the Type of Service data.
Input Data
Protocol Type 1 = Not Used

2 = Not Used
3 = Voice Control
4 = H.323
5 = RTP/RTCP
6 = SIP
7 = CCISoIP
8 = SIP MLT
9 = SIP Trunk
10 = Net-Link
11 = Video RTP/RTCP
Item
No.
01 ToS Mode 0 = Disable (Invalid) 0 When Input Data is set to 1,

1 = IP Precedence Item No. 07 is invalid.
2 = Diffserv When Data is set to 2, Item
No. 02 ~ 06 are invalid.
02 Priority, IP Precedence 0~7 0 1 = Router queuing priority

0 = Low
7 = High
03 Low Delay 0~1 0 1 = Optimize for low delay

0 = Normal Delay, Low routing
Delay
04 Wideband 0~1 0 1 = Optimize for high

(Throughout) 0 = Normal Throughput bandwidth routing
1 = High Throughput
05 High Reliability 0~1 0 1 = Optimize for reliability

0 = Normal Reliability routing
1 = Low Reliability
07 Priority (D.S.C.P. - 0~63 0 DSCP (Differentiated

Differentiated Services Services Code Point)
Code Point)

Issue 5.0
Conditions
 The system must be reset for these program options to take affect.

4-40 Programming
Issue 5.0

84-21 : CCIS over IP CODEC Information Basic Setup
Level:
IN
Description
Use Program 84-21 : CCIS over IP CODEC Information Basic Setup to set the
CODEC parameters of the GPZ-IPLE.
Input Data
Item
No.
01 Number of G.711 Audio Frames 1~4 3

1 = 10ms
2 = 20ms
3 = 30ms
4 = 40ms
02 G.711 Type 0 = A-law 0

1 = -law
03 G.711 Voice Activity Detection Mode 0 = Disable 0

1 = Enable
04 G.711 Jitter Buffer (min) 0~300ms 30
05 G.711 Jitter Buffer (average) 0~300ms 60
06 G.711 Jitter Buffer (max) 0~300ms 120
07 G.729 Audio Frame Number 1~6 3

1 = 10ms
2 = 20ms
3 = 30ms
4 = 40ms
5 = 50ms
6 = 60ms
08 G.729 Voice Activity Detection Mode 0 = Disable 0

1 = Enable
10 G729 Jitter Buffer (average) 0~300ms 60
11 G729 Jitter Buffer (max) 0~300ms 120

Issue 5.0
Item
No.
19 1st Priority of Audio Capability 0 = G.711 PT 0

2 = G.729 PT
3 = G.722
4 = G.726
20 2nd Priority of Audio Capability 0 = G.711 PT 2

2 = G.729 PT
3 = G.722 PT
4 = G.726 PT
22 Jitter Buffer Mode 1 = Static 3

3 = Self Adjusting
23 Voice Activity Detection Threshold 0 ~ 30 (-20dB ~ +10dB) 20

0 = -20dB (-50dBm)
1 = -19dBm (-49dBm)
:
20 = 0dBm (-30dBm)
:
29 = +9dBm (-21dBm)
30 = +10dBm (-20dBm)

1 = 10ms
2 = 20ms
3 = 30ms
4 = 40ms

1 = 10ms
2 = 20ms
3 = 30ms
4 = 40ms
33 G.726 Voice Activity Detection Mode 0 = Disabled 0

1 = Enabled
43 RTP Filter 0 = Disable 1

To avoid incorrect voice pass connection, this 1 = Enable
Program checks the sending side address from 2 = Enable (include SSRC)
received RTP packet at VoIPDB.
4-42 Programming
Issue 5.0
Conditions
None


Issue 5.0

84-33: Fax Over IP Setup
Level:
IN
Description
Use Program 84-33 : Fax Over IP Setup to set up the parameters of the Fax Over
IP function.
Index 1
Type 1 = H.323 Trunk

2 = Networking
3 = SIP Trunk
4 = SIP Extension
5 = CCIS over IP
6 = NetLink
Input Data
Item
Item Input Data Default Profile 1 - 6
No.
01 FAX Relay Mode 0 = Disable 0

1 = Enable
2 = Each Port Mode
02 T.38 Protocol Mode 1 = R/U 1

2 = U/R
3 = RTP
4 = UDPTL
04 Jitter Buffer (max) 0 ~ 300 160
05 T.38 RTP Format Payload 96 ~ 127 100

Number
06 T.38 Fax Maximum Speed 0 = V.27ter, 4800bps 2

1 = V.29, 9600bps
2 = V.17, 14400bps
07 T.38 Data Error Correction Mode 0 = Redundancy 0

1 = FEC
08 T.38 Error protection depth for 0~2 0

Signaling
09 T.38 Error protection depth for 0~2 0

Data
4-44 Programming
Issue 5.0
Input Data
Item
Item Input Data Default Profile 1 - 6
No.
10 T.38 TCF Method 1 = VOIPDB 1

2 = G3FE
11 T.38 ECM (Error Correction 0 = Disable 1

Mode)
1 = Enable
12 FAX CODEC 1 = G.711 a-law 1

2 = G.711 u-law
3 = G.726
13 Payload Size 1 ~ 4 (10ms base) 2

2 = Self adjusting
15 Minimum Jitter Buffer 0 ~ 300 80
16 Average Jitter Buffer 0 ~ 300 120
17 Maximum Jitter Buffer 0 ~ 300 160
18 FAX RTP Payload Type 97 ~ 127 103
19 FAX over IP Type 1 = Type 1 1

2 = Type 2
Conditions
None

 Voice Over Internet Protocol (VoIP)

Issue 5.0
Program 90 : Maintenance Program

90-23 : Deleting Registration of IP Telephones
Level:
IN
Description
This program is available only via telephone programming, Web programming

NOTE
and not through PC Programming).
Use Program 90-23 : Deleting Registration of IP Telephones to delete the

registered IP telephone from the system.
Input Data
Extension Number Up to 8 digits
Ite
m Item Input Data
No.
01 Delete IP Telephone [Delete?] : Dial 1 + press Transfer

This assignment removes the station number (Press Transfer to cancel.)
association with the MAC address of the IP station.
Conditions
None

4-46 Programming
Issue 5.0

90-34 : Firmware Information
Level:
IN
Description
Use Program 90-34 : Firmware Information to list the package type and firmware
blades installed in the system.
Input Data
Slot No. 1~24
Item
Item Display Data
No.
01 Pkg Name PKG Name
02 Firmware Version Number 00.00~15.15
03 VoIPDB Version DEV/PR/REL-00.00

00.00.00.00~ FF.FF.FF.FF
04 DSP Project Number 00000000~ FFFFFFFF
05 Vocallo F/W Version 00.00.00.00~ FF.FF.FF.FF
06 OCT1010ID Version 00.00.00.00~ FF.FF.FF.FF
Conditions
 These Programs are ‘Read Only.’

None

Issue 5.0
4-48 Programming
UNIVERGE® SV9100
Network Design Considerations Chapter 5
This chapter explains some issues that should be considered when planning a
UNIVERGE SV9100 VoIP installation. This is a generalized explanation and
therefore does not discuss vendor-specific issues and solutions. Typically, different
solutions are implemented by different manufacturers.
SECTION 2 QOS
Quality of Service (QoS) is one of the most important factors for VoIP. This refers to
the perceived quality of speech and the methods used to provide good quality
speech transmission. Several factors that affect speech quality and several
mechanisms can be used to ensure QoS.
This chapter also describes the problems that can occur and some possible
solutions. Each network equipment manufacturer (NEC, 3Com, Cisco, etc.) has
slightly different methods of implementing QoS and these are not discussed in this
document. This chapter provides an overview to classify voice traffic on the
UNIVERGE SV9100 so that the network equipment can impose QoS.
2.1 QoS Definitions
This section lists common definitions used with QoS for VoIP.
Latency (Delay):
If at any point the usage on the network exceeds the available bandwidth, the
user experiences delay, also called latency. In more traditional uses of an IP data
network, the applications can deal with this latency. If a person is waiting for a
web page to download, they can accept a certain amount of wait time. This is not
so for voice traffic. Voice is a real time application, which is sensitive to latency. If
the end-to-end voice latency becomes too long (250ms, for example), the call
quality is usually considered poor. It is also important to remember that packets
can get lost. IP is a best effort networking protocol. This means the network tries
to get the information there, but there is no guarantee.
Delay is the time required for a signal to traverse the network. In a telephony
context, end-to-end delay is the time required for a signal generated at the
talker's mouth to reach the listener's ear. Therefore, end-to-end delay is the sum

Issue 5.0
of all the delays at the different network devices and across the network links
through which voice traffic passes. Many factors contribute to end-to-end delay,
which are covered next.
The buffering, queuing, and switching or routing delay of IP routers primarily

determines IP network delay. Specifically, IP network delay is comprised of the
following:
 Packet Capture Delay

Packet capture delay is the time required to receive the entire packet before
processing and forwarding it through the router. This delay is determined by the
packet length and transmission speed. Using short packets over high-speed
networks can easily shorten the delay but potentially decrease network
efficiency.
 Switching/Routing Delay
Switching/routing delay is the time the router takes to switch the packet. This
time is needed to analyze the packet header, check the routing table, and route
the packet to the output port. This delay depends on the architecture of the
switches/routers and the size of the routing table.
 Queuing Time
Due to the statistical multiplexing nature of IP networks and to the
asynchronous nature of packet arrivals, some queuing, thus delay, is required at
the input and output ports of a packet switch. This delay is a function of the
traffic load on a packet switch, the length of the packets and the statistical
distribution over the ports. Designing very large router and link capacities can
reduce but not completely eliminate this delay.
Jitter
Delay variation is the difference in delay exhibited by different packets that are
part of the same traffic flow. High frequency delay variation is known as jitter.
Jitter is caused primarily by differences in queue wait times for consecutive
packets in a flow, and is the most significant issue for QoS. Certain traffic types,
especially real-time traffic such as voice, are very intolerant of jitter. Differences
in packet arrival times cause choppiness in the voice.
All transport systems exhibit some jitter. As long as jitter falls within defined
tolerances, it does not impact service quality. Excessive jitter can be overcome
by buffering, but this increases delay, which can cause other problems. With
intelligent discard mechanisms, IP telephony/VoIP systems try to synchronize a
communication flow by selective packet discard, in an effort to avoid the
walkie-talkie phenomenon caused when two sides of a conversation have
significant latency. UNIVERGE SV9100 incorporates a Jitter Buffer to avoid
these problems.
Packet Loss
During a voice transmission, loss of multiple bits or packets of stream may cause
an audible pop that can become annoying to the user. In a data transmission,
loss of a single bit or multiple packets of information is almost never noticed by
5-2 Network Design Considerations

Issue 5.0
users. If packet drops become epidemic, the quality of all transmissions

degrades. Packet loss rate must be less than five percent for minimum quality
and less than one percent for toll quality.
2.2 Voice Quality Improvements
This section describes various techniques that can be used to improve the voice
quality.
 Increase available bandwidth:
This can sometimes be the most basic solution and the easiest of the solutions.
If running a System IP Phone using G.711 with a 30ms fill time over Ethernet, for
only one call, 90Kbps of bandwidth is needed. If that same user only has a 64K
line, they do not have a decent IP voice call. The user can increase the
available bandwidth to slightly exceed the 90Kbps requirements and their voice
quality dramatically increases. This solution might not be viable if no more
bandwidth is available.
 Use a different CODEC:

The CODEC contains possible compression algorithms to be used on the voice.
Lets take the example above again. The user only wants one voice line over a
64Kbps data connection. They also want to maintain their current fill time of
30ms. Change to a G.729. For one line, only 34Kbps is required for a call. This
fits well within the 64Kbps of available bandwidth.
 Increase the number of frames per packet:

To continue with the example above, the user has moved to a G.729 CODEC.
But now, the user wishes to add two more System IP Phones. Their current
64Kbps line can handle one call, because it is only 34Kbps. Two more System
IP Phones would increase the total to 102Kbps so obviously there is not
sufficient bandwidth.
The user can now increase the fill time to 50ms. This reduces the bandwidth per
call to 19.8Kbps (3x 19.8 = 59.4Kbps). The savings in bandwidth comes from
the fact that with a longer fill time, fewer packets are needed to send the voice.
With fewer packets, less header information needs to be attached and
transmitted.
 Change Layer 2 Protocols:
Ethernet is most commonly used for IP packets. Unfortunately, Ethernet has a
fairly large overhead of 34 bytes. So every IP voice packet going over Ethernet
has a 34-byte Ethernet header attached to it. As the number of packets add up,
this header data can become significant. Frame Relay has a 7-byte header and
Point-to-Point Protocol (PPP) has a 6-byte header. With this decrease in header
length at layer 2, some significant savings in bandwidth use can be achieved.
The down side to this is that most networks may not have these services
available, where Ethernet is very widely used. This is usually outside the control
of the installer and therefore NEC strongly advises users to do more research
on other layer 2 protocols before trying to implement them in their voice
network.
 Implement Quality of Service (QOS):
Now, assume a derivative of the above example. The user needs only one voice
line over their 64Kbps connection. They are using G.729 with a 30ms fill time.
This requires 34Kbps of their available bandwidth. Also assume that this line is

Issue 5.0
used at certain times of the day for data connectivity. This data connectivity is
very light, only 20Kbps or so during most of the day, but does spike to 50Kbps
during certain points of the day. This data is not time sensitive like the voice
data, so if necessary it could be forced to wait.
Therefore, the user can implement a Quality of Service mechanism on the IP
network. At its most basic form, this denotes certain IP packets as being more
important than others. So they would tell this 64Kbps line that IP packets with
voice deserve a higher priority than those without voice. This allows the network
devices to give priority to the other data, so the quality of the call is not
compromised.
2.3 Types of Classifications for Traffic for QoS
Classification uses information from a packet (or frame) to define the type of data
and therefore how the data should be handled for QoS on the network. Using
packet classification, you can partition network traffic into multiple priority levels or
Types of Service (ToS). UNIVERGE SV9100 supports methods of marking a
packet with its classification information in the Layer 2 or 3 headers.
 VLAN (802.1Q):
Virtual LANs work at Layer 2 of the OSI model and can be equated to a
broadcast domain. More specifically, VLANs can be seen as a group of end
stations, perhaps on multiple physical LAN segments that are not constrained
by their physical location and therefore, communicate as if they were on a
common LAN. Packets can be marked as important by using layer 2 classes of
service (CoS) settings in the User Priority bits of the 802.1Pq header. Refer to
Program 84-09 : VLAN Setup on page 5-30 for information for VLAN
configuration.
Figure 5-1 Layer 2 Diagram (802.1Q)
Layer 2
802.1Q/p
TAG
PREAM SFD DA SA Type PT Data FCS
4 Bytes
PRI CFI VLAN ID
Three bits used for CoS

(user priority)

Issue 5.0
 IP Precedence - Layer 3 QoS:

Allows you to specify the class of service for a packet. You use the three
precedence bits in the Ipv4 header type of service (ToS) field for this purpose.
Using the ToS bits, you can define up to six classes of service. Other devices
configured throughout the network can then use these bits to determine how to
treat the packet in regard to the type of service to grant it. These other QoS
features can assign appropriate traffic-handling policies including congestion
management and bandwidth allocation. By setting IP Precedence levels on
incoming traffic and using them in combination with QoS queuing features, you
can create differentiated service. (Refer to Program 84-10 : ToS Setup on page
5-31 for detailed programming information.)
 Differentiated service (Diffserv) - Layer 3 QoS:

Provides services differentiated on performance using weighted priority
queuing. DiffServ requires that edge routers classify traffic flows into a member
from a set of categories based on the TCP/IP header fields in what is called a
micro flow. Because the Diffserv is present in every packet header, each node
can provide differentiated services per-hop. Refer to Program 84-10 : ToS
Setup on page 5-31 for detailed programming information.
SECTION 3 INTERNET BASED CONNECTIONS (XDSL, CABLE, ETC.)
Internet-based connections are becoming increasingly popular. This is mainly due to

the speed and cost of xDSL and cable modem connections. For data applications,
these types of connection are generally acceptable. For Voice over IP applications
several issues should be taken into consideration.
Asymmetric Data Rates

On many Internet based connections, there are different data rates for upstream and
downstream. For example 1Mbps down and 256Kbps up. This works well for Internet
access, as generally you download files from the Internet to your PC and transmit
less information in the other direction. For VoIP, speech uses the same amount of
bandwidth in both directions, which means that the amount of simultaneous calls can
not exceed the amount of “upstream” bandwidth available.
Contention
Most Internet based connections specify a contention ratio. This is typically 50:1 for
home users or 20:1 for business users. This specifies the number of users
subscribed to a single connection to the Internet Service Provider (ISP). This
indicates how many users share the bandwidth with other users on the Internet,
which means that the speeds that you are quoted are not necessarily accurate – you
receive less than these figures.
 It is unlikely that all subscribers are using a connection at the same time, so these figures are
not quite as bad as they first seem.

Issue 5.0
Network Address Translation (NAT)

Usually, the equipment that your ISP provides (cable modem, ADSL router, etc.)
uses Network Address Translation. This allows several devices to share one public
IP address. The issues relating to the use of NAT are outlined in Firewalls and NAT
below.
VPN
Due to the use of NAT, and non-routable IP addressing, it is necessary to implement
a VPN solution. This is outlined in VPN Tunneling below. (Refer to 4.3 Virtual
Private Network (VPN) Tunnelling on page 5-8.)
QoS
As discussed earlier, it is essential to have some form of Quality of Service
implemented. With Internet based connections, we are not in control of the many
routers, switches and other network hardware that reside between our two VoIP
endpoints. This means that we cannot specify any QoS parameter on these devices.
The only point where the QoS can be controlled is at the VPN or firewall. This allows
VoIP traffic to be prioritized over any other data that is sent out to the Internet. This
helps to maintain reasonable quality speech – but once the data has exited the local
router/cable modem it is at the mercy of the Internet.
When implementing UNIVERGE SV9100 IP over Internet based connections it is

very important that these factors are considered, and that the customer is made
aware that neither the installer nor NEC are held responsible for any quality issues
experienced.
SECTION 4 FIREWALLS AND NAT
The ways in which networks are designed to be secure (firewall, VPN services, proxy
servers, etc.) and integration of NAT create problems for VoIP. This is due in part, to
the endless number of different scenarios for non-real time protocols and their limited
solutions.
4.1 Understanding the Infrastructure
The networks in place today look very different than the networks of yesterday. In
the past, only computers and servers were connected to the network. The network
was built to be as a best effort delivery mechanism, where delay and lost of
information between devices was something we dealt with. Today, there is an over
saturation of devices needing to gain access to the IP network. Desktop
computers, fax machines, wireless PDAs, Servers, home appliances, video
servers and now VoIP terminals all are fighting for bandwidth, precedence, and
addresses on this converged network.

Issue 5.0
It is necessary to create some kind of Intranet environment (across the Internet),

with fixed network characteristics, where VoIP solutions can tolerate some minor
variations. IT personnel have been tasked with implementing different
mechanisms in the network to support the new demands required on the
converged network. Some solutions that have been implemented are:
 QoS devices to support precedence settings of voice packets.
 Elimination of hubs in place of switches to support 100Mbps full-duplex

transmission.
 Firewall integration to protect the internal network from external attack.
 Network Address Translation (NAT) devices are widely deployed to support the
addressing issues.
 Virtual Private Network (VPN) Servers were added to Enterprise networks to

support the security and connectivity issues for remote users.
Some solutions, such as the hub replacement and integration of QoS, are done
behind the scenes and should have no effect on the voice application. Other
solutions such as NAT and Firewall cause major disturbance to VoIP.
Implementing a VPN is the only way to resolve these issues.
4.2 Firewall Integration
Network security is always a concern when connecting the Local Area Network
(LAN) to the Wide Area Network (WAN). There are many ways to integrate
security in the network – the most popular are Firewalls and Proxy servers.
 Firewalls
Firewalls can be implemented in both hardware and software, or a combination
of both. Firewalls are frequently used to prevent unauthorized Internet users
from accessing private networks connected to the Internet, especially intranets.
All messages entering or leaving the intranet pass through the firewall, which
examines each message and blocks those that do not meet the specified
security criteria.
 Proxy Server
Proxy server intercepts all messages entering and leaving the network. The
proxy server effectively hides the true network address.
What should be noted is that no matter which security measure is implemented,

the VoIP must have TCP/UDP ports open in the security wall (e.g., firewall/proxy)
for the media and control streams to flow. If any point in the network prevents the
ports from flowing from end-to-end, the VoIP application does not work.
The ports that need to be open on the firewall/proxy vary depending on the
particular application being used. A list of these ports is shown below, however it
should be noted that the preferred solution would be to allow all ports on the
UNIVERGE SV9100 device to be open, or to place the SV9100 outside of the
firewall.

Issue 5.0
UNIVERGE SV9100
Applications Rx Port
Programming
PC Programming 7
DHCP Server 67
SIP MLT Listening Port 5080 10-46-06
SIP Trunk Listening Port 5060 10-29-04
SIP Single Line Stations 5070 84-20-01

Realtime Transport Protocol 10020 84-26-02
Realtime Transport Control 10021 84-26-03
Protocol (RTCP)
4.3 Virtual Private Network (VPN) Tunnelling
A Virtual Private Network is a private data network that maintains privacy through
using a tunneling protocol and security procedures. Allowing for remote networks
(including VoIP devices), which reside behind NATs and/or Firewalls to
communicate freely with each other. In UNIVERGE SV9100 VoIP networks,
implementation of VPNs can resolve the issues with NAT that are described in the
previous section.
The idea of the VPN is to connect multiple networks together using public (i.e.,
Internet) based connections. This type of connection is ideal for those commuters,
home workers, or small branch offices needing connectivity into the corporate
backbone. It is possible to connect these remote networks together using private
links (such as leased lines, ISDN, etc.) but this can be very expensive and there is
now a high demand for low cost Internet connectivity.
Companies today are exploring the use of VPN for a variety of connectivity
solutions, such as:
 Remote User to Corporate Site VPN
Allows employees to use their local ISP fastest connection such as cable
modems, DSL, and ISDN. For traveling users, all they need to do is dial into
their ISP local phone number.
 Site-to-site VPN
Allows companies to make use of the Internet for the branch-to-branch
connections, cutting the cost of the expensive point to point leased line service.
 Extranet
Extranet describes one application using VPN technology. The concept allows a
company and a vendor/supplier to access network resources at each site. For
example, a customer may have access to a suppliers intranet for access to
product information.

Issue 5.0
VPNs can be implemented in hardware or software. Single users, such as

traveling sales personnel, may have a software based VPN client on their laptop
computer. This connects back to the Head Office VPN server. For larger sites, the
VPN is typically implemented using a hardware VPN – this is often incorporated in
to a firewall solution.
The diagram below is example of how a VPN tunnel may be implemented. The
red lines in the diagram show the tunnels that are created through the Internet.
Each network can connect to the others as though they are connected with private
connections (kilostream, etc.), without the issues relating to NAT.
Figure 5-2 Virtual Private Network (VPN) Example
Internet
SL AD
SL
AD
ADSL ADSL
Router Router
DSL / Cable / Dialup
Firewall/ Firewall/
VPN VPN
Home Office LAN
UNIVERGE SV9100
Head Office LAN
Mobile Workers
(Software VPN Client)
When IP address translation is applied to a VoIP packet, the application fails and
the communication path is broken. VoIP packets contain the IP address
information and the ports used as part of its payload. When NAT is applied, only
the header parameter is changed, not the payload data that affects the process of
data packets within the VoIP switch and terminal.

Issue 5.0
The common scenario for remote IP deployment is:

 Implementation of an IP Phone with a public IP address talking with an
UNIVERGE SV9100 behind NAT. An example would be a telecommuter.
Implementation of an IP Phone behind a NAT, which connects to the Internet,
terminates in a UNIVERGE SV9100 behind a different NAT.
When selecting VPN equipment it is important to consider Quality of Service.
Generally, VPN hardware is connected to Internet connections which are
unreliable and out of the control of the customer. However, it is possible to set
prioritization on some VPN units for voice traffic. This does not solve the
unreliability of the Internet, but helps to ensure that the data traffic to and from
the LAN do not impair the quality of the voice traffic. (Refer to Section 2 QoS on
page 5-1).
 NEC strongly recommends that any VPN hardware used for VoIP has the
facility to prioritize voice traffic.
SECTION 5 CODEC AND BANDWIDTH
This section describes CODEC and bandwidth and their application with the
UNIVERGE SV9100 system.
5.1 CODECs
CODEC (COder/DECoder) uses the technology of encoding and decoding a

signal. For VoIP, this specifically refers to the algorithm used to convert analog
speech to digital data for transmission on an IP network.
The UNIVERGE SV9100 system supports three different CODECs:

 G.711
This is the ITU-T recommendation for coding of speech at 64kbps using PCM
(pulse code modulation). This CODEC is often described as uncompressed as it
uses the same sampling rate as Time-Division Multiplexing (TDM). G.711 has a
MOS2 score of 4.2 but uses a large bandwidth for transmission. This CODEC is
not commonly used due to the bandwidth required, although it can be
acceptable in LAN environment (i.e., IP Phones connected over a 100Mbps
LAN).
 G.722
G.722 is an ITU standard CODEC that provides 7kHz wideband audio at data
rates from 48 to 64kbps. This is useful in a fixed network Voice Over IP
applications, where the required bandwidth is typically not prohibitive, and offers
a significant improvement in speech quality over older narrowband CODECS
such as G.711, without an excessive increase in implementation complexity.
2. The Mean Opinion Score (MOS) provides a numerical measure of the quality of human speech at the destination end
of the circuit. The scheme uses subjective tests (opinionated scores) that are mathematically averaged to obtain a
quantitative indicator of the system performance.

Issue 5.0
 G.726
G.726 is an ITU-T ADPCM speech CODEC standard covering voice
transmission at rates of 16, 24, 32, and 40kbit/s. It was introduced to supersede
both G.721, which covered ADPCM at 32kbit/s, and G.723, which described
ADPCM for 24 and 40kbit/s. G.726 also introduced a new 16kbit/s rate. The four
bit rates associated with G.726 are often referred to by the bit size of a sample
as 2-bits, 3-bits, 4-bits, and 5-bits respectively.
 G.729A
This ITU-T recommendation describes the algorithm for coding of speech
signals at 8kbps using CS-ACELP (conjugate-structure algebraic code-excited
linear prediction). This CODEC samples the analog signal at 8000Hz and uses
a frame size of 10ms. This CODEC has a MOS score of 4.0.
G.729 is the most commonly used CODEC for UNIVERGE SV9100 VoIP
installations. This is due to the fact that it offers high compression (and therefore
low bandwidth) while maintaining good speech quality.
 G.723
This ITU-T recommendation describes a very low bit-rate compression
algorithm. The standard describes two versions 5.3Kbps and 6.4Kbps.
UNIVERGE SV9100 uses the higher bit rate. This CODEC offers low bandwidth
speech transmission, but has a lower MOS score of 3.9. This CODEC is not
commonly used on the UNIVERGE SV9100, but is particularly suited to low
bandwidth WAN connections.
 iLBC
The iLBC CODEC is an algorithm that compresses each basic frame (20ms or
30ms) of 8000 Hz, 16-bit sampled input speech, into output frames with rate of
400 bits for 30ms basic frame size and 304 bits for 20ms basic frame size. This
CODEC is suitable for real-time communications such as, telephony and video
conferencing, streaming audio, archival and messaging.
Packet Size:
Each CODEC has a set frame length. This is the time that the frame
encapsulates. For G.729 and G.711 the frame length is 10ms and for G.723 the
frame length is 30ms. It is possible to configure the packet size in the UNIVERGE
SV9100 programming. To do this, we tell the UNIVERGE SV9100 how many
frames to encapsulate into each packet for transmission.
For example, the G.729 has a frame length of 10ms - the packet size is set to 3 (in
Program 84-11-01). This gives a 10ms x 3 = 30ms packet.

Issue 5.0
5.2 Bandwidth
The bandwidth required for VoIP calls depends on several factors, including:
 Number of simultaneous calls
 CODEC used
 Frame Size
 Data Networking Protocol used
 UNIVERGE SV9100 CCIS Network parameters
The more frames encapsulated into each packet, the less bandwidth is required.
This is because each packet transmitted has the same header size. Therefore, if
numerous very small packets are sent then bandwidth is also being used for a
large amount of header information. If we add several frames to the packet, less
packets are transmitted and therefore have less header information sent.
If we add many voice frames to each packet, less bandwidth is being used.
However, this does have disadvantages. If there is a large packet size, and a
particular voice packet is lost, this has a greater impact on the speech quality. If a
small quantity of voice frames per packet is being used, the effect of losing a
packet is reduced.
As a general rule: The more frames per packet, the less bandwidth is used, but
the quality is also lower.
Examples:
Example 1: CODEC: G.729 Frame Size: 10ms Voice Frames per Packet: 2 Voice
Sample Size: 20ms (frame size x Voice Frames) Bandwidth Required: 24Kbps
Example 2: CODEC: G.729 Frame Size: 80ms Voice Frames per Packet: 8 Voice
Sample Size: 80ms (frame size x Voice Frames) Bandwidth Required: 12Kbps

Issue 5.0
SECTION 6 DSP RESOURCE CALCULATION
Voice over IP (NECi SIP, SIP stations, SIP trunks) requires DSP resources to be able
to convert from TDM3 to IP technologies. DSPs (Digital Signal Processors) take a
TDM signal and convert to Realtime Transport Protocol (RTP) for transmission as
VoIP, and vice versa. Each IP to TDM conversion requires a DSP resource.
DSP resources are provided by the GPZ-IPLE It can be difficult to work out how
many DSP resources are required in an UNIVERGE SV9100 system, because:
 not all IP Extensions/trunks are used at the same time
 peer-to-peer calls do not use a DSP resource
GPZ-IPLE IP Addressing
The GPZ-IPLE requires two IP Addresses, one for Signaling (PRG 10-12-09), and
one for the DSP Resources (PRG 84-26-01).
 When assigning the IP addresses to the GPZ-IPLE card, they must be in the same network
(subnet). If the CPU will be connected to the network it requires a separate IP address in a
different network (subnet). When an GPZ-IPLE card is attached to the CPU, using the CPU
NIC is no longer required. All connections that previously terminated to the CPU NIC card
can now be terminated to the GPZ-IPLE NIC. For example, PC PRO, Web Pro, and ACD all
terminate to the GPZ-IPLE NIC card when installed. Both the GPZ-IPLE and CPU NIC
share the same gateway assignment. The default gateway command in Program 10-12-03 is
used by both NICs, allowing only one device, GPZ-IPLE or CPU, to route outside of its own
network.
The following chart shows the minimum and maximum number of IP addresses used
with different GPZ-IPLE card configurations.
Minimum IP Maximum IP
Card Notes
Addresses Addresses
The number of DSP channels
GPZ-IPLE 1 2 depends on the VOIP license
loaded to GCD-CP10 up to 256.
3. TDM = Time Division Multiplexing - traditional circuit-based telephony

Issue 5.0
To calculate the maximum DSPs required:
The manual calculations listed below are used in the UNIVERGE SV9100.
Non-Peer-to-Peer Mode (Program15-05-50=0):
It is easy to calculate the maximum number of DSPs for a system that is not
peer-to-peer. This is a simple addition of:
VoIP extensions (VoIPE) + VoIP trunks (VoIPT)
Combine the resource figures:
Combine the (extension resource figure x DSPs required for extensions) + (trunk
resource figure x DSPs required for trunks) equals the total card resource required.
nTotalCardResourceRequired = (nExtCardResourceFactor x
nDSPsForExt)+(nTrkCardResourceFactor x nDSPsForTrk) 4
SECTION 7 QUALITY OF SERVICE (QOS) IMPLEMENTATION
Section 2.2 Voice Quality Improvements on page 5-3 discusses some of the
problems associated with voice quality. This section describes how QoS can be
implemented on data networks to provide the “best case” for VoIP traffic.
Not all network hardware supports QoS and each manufacturer has their own
methods of implementing QoS. The explanations below are as generic as possible.
The installer/maintainer of the data network should be familiar with the QoS
characteristics of their equipment and should be able to configure the equipment
accordingly.
Quality of Service is commonly used to describe the actual implementation of

prioritization on network hardware. This prioritization (at Layer 2 and Layer 3 of the
OSI model) is described in Figure 5-1 Layer 2 Diagram (802.1Q) on page 5-4.
7.1 Prioritization
When data is transmitted through a network, bottlenecks can occur causing the
available bandwidth to be reduced or the data to increase. This impacts the
packet delivery.
Consider data communication between the two computers shown in the diagram
Figure 5-1 Layer 2 Diagram (802.1Q). The Hosts can transmit data at 100 Mbps.
When a packet from Host A, destined for Host B, reaches the router, the available
bandwidth is reduced to 256Kbps and the packet flow must be reduced. Figure
5-3 Network Bottleneck Example shows a diagram of this condition.
4. This figure is different only to the number of required DSPs if the CODECS used are the faster ones. All other
CODECS are a multiplication factor of 1 thus not effecting the calculation.

Issue 5.0
Figure 5-3 Network Bottleneck Example
Host A Host B
100Mbps
100Mbps
256Kbps
Private Circuit
(Leased Line)
100Mbps 100Mbps
Switched Hub Router Router Switched Hub
For this example, each end of the network has only.one host Typically, many hosts
are sending data over the narrow bandwidth. The routers buffer packets and
transmit them over the WAN lines as efficiently as possible. When this occurs,
certain packets are dropped by the router and some packets are delayed.
For most data applications this packet loss/delay is not critical.For example, a
delay of one to five seconds to transmit an email is imperceptible. When VoIP is
implemented, this loss/delay has a massive impact on the voice quality. The
resulting gaps in speech, distortion and delay are unacceptable for voice traffic.
To avoid this problem, it is possible to prioritize the VoIP packets. The router
examines all packets received, determines the priority level of the packet, and
forwards it accordingly. The data5 is assigned lower priority and the voice is
transmitted before the data. This can have a negative impact on the data network
if a lot of voice is transmitted.
5. This description discusses voice and data. These terms are commonly used when describing QoS, although in the
case of VoIP, the voice is actually converted to IP and transmitted as data. Therefore, everything transmitted on a
Data Network is data, but logically we think of this as voice and data traffic.

Issue 5.0
Figure 5-4 Voice and Data Network Implementation shows how a voice and data
network can be implemented.
Figure 5-4 Voice and Data Network Implementation
Host A Host B
100Mbps
100Mbps
256Kbps
Private Circuit
(Leased Line)
100Mbps 100Mbps
Managed Switch Router Router Managed Switch

100Mbps
100Mbps
Telephone Telephone
System A System B
After the router is configured for QoS, it examines incoming packets and allocates a
priority to the packet. Figure 5-5 Priority Queuing on Voice and Data Networks
shows the affect priority queuing has on voice and data networks. The packets arrive
randomly. They are processed and output according to the QoS policy. The VoIP
traffic is output first.
Figure 5-5 Priority Queuing on Voice and Data Networks
Packet from VoIP Device

Packet from PC
Ethernet (100Mbps) Leased Line (256Kbps)
Router with QoS

Policy Applied
Direction of IP Traffic

Issue 5.0
To enable this type of queuing it is necessary to:
a Configure the VoIP equipment to mark its packets with a specific value so that the
switches/routers can identify that it is voice – Called Marking.
b Configure the network equipment to recognize the difference between the
different Marked packets – Called Classification. (i.e., informs the router what a
voice packet looks like.
c Configure the network equipment to give priority to the packets that have been
classified as voice – Called Priority Queuing.
7.2 Layer 2 QoS (802.1pq)
QoS is most commonly implemented at Layer 3 of the OSI model. This layer deals
with IP addresses, and is usually handled by Routers. However, sometimes it is
necessary to implement Layer 2 QoS – usually in large LAN environments with
many IP phones.
Layer 2 devices work with Ethernet frames (encapsulated IP packets) rather than
IP addresses. These devices are usually Switched Hubs (Switches). As the IP
header information is encapsulated, the Switched Hubs cannot reference the
Type of Service (Layer 3 QoS) field in the IP header to determine the priority of a
frame.
Layer 2 QoS uses the Priority field of the Ethernet frame. This field has three bits
and can have eight possible values (000 to 111 in binary). Some switches can be
configured to prioritize traffic based on these values. This field is available only if
the Ethernet device is configured for VLAN (IEEE 802.1q) operation (VLAN is
outside the scope of this document).

Issue 5.0
Protocol Structure - IEEE 802.1p: LAN Layer 2 QoS
Figure 5-6 Protocol Structure for Layer 2 QoS illustrates the format of an Ethernet
frame and the User Priority field that is used for Layer 2 QoS.
Figure 5-6 Protocol Structure for Layer 2 QoS
IEEE 802.1Q Tagged Frame for Ethernet
7 1 6 6 2 2 2 42~1496 4
Bytes Byte Bytes Bytes Bytes Bytes Bytes Bytes Bytes
Preamble SFD DA SA TPID TCI Type Data CRC
Length
Expanded View of TCI Field
3 Bits 1 Bit 12 Bits

User Priority CFI Bits of VLAN ID (VID) to identify possible VLA
Ns
The following define the fields used for the protocol structure:
Preamble (PRE) - The PRE is an alternating pattern of ones and zeros that tells
receiving stations a frame is coming, and synchronizes frame-reception portions
of receiving physical layers with the incoming bit stream.
Start-of-frame delimiter (SFD) - The SOF is an alternating pattern of ones and

zeros, ending with two consecutive 1-bits indicating that the next bit is the left-
most bit in the left-most byte of the destination address.
Destination Address (DA) - The DA field identifies which station(s) should

receive the frame.
Source Addresses (SA) - The SA field identifies the sending station.
Tag Protocol Identifier (TPID) - The defined value of SV9100 in hex. When a
frame has the EtherType equal to SV9100, this frame carries the tag IEEE 802.1Q
/ 802.1P.
Tag Control Information (TCI) - The field including user priority, Canonical
format indicator and VLAN ID.

Issue 5.0
User Priority - Defines user priority, giving eight priority levels. IEEE 802.1P
defines the operation for these three user priority bits.
CFI - Canonical Format Indicator is always set to zero for Ethernet switches. CFI
is used for compatibility reason between Ethernet type network and Token Ring
type network.
VID - VLAN ID is the identification of the VLAN, which is basically used by the
standard 802.1Q. It allows the identification of 4096 VLANs.
Length/Type - This field indicates either the number of MAC-client data bytes that
are contained in the data field of the frame, or the frame type ID if the frame is
assembled using an optional format.
Data - Is a sequence of bytes of any value. The total frame minimum is 64 bytes.
Frame Check Sequence (FCS) - This sequence contains a 32-bit cyclic

redundancy check (CRC) value, which is created by the sending MAC and is
recalculated by the receiving MAC to check for damaged frames.
Example Ethernet Frame with Layer 2 QoS Enabled
The example below shows an Ethernet Frame containing one RTP (speech)
packet. The Frame is VLAN tagged, has a VLAN ID of 99 and a VLAN Priority of
5. It is also possible to see that the Layer 3 QoS has not been set.
Figure 5-7 Ethernet Frame Example - Layer 2 QoS Enabled
Source Destination Protocol

172.16.0.101 172.16.0.21 RTP
Info
Payload type=ITU-T G.729, SSRC=701655963, Seq=28165, Time=21520
Frame 160 (78 bytes on wire, 78 bytes captured)

Arrival Time: Jan 18, 2005 13:55:44.842738000
Time delta from previous packet: 0.008241000 seconds
Time since reference or first frame: 2.910072000 seconds
Frame Number: 160
Packet Length: 78 bytes
Capture Length: 78 bytes
Ethernet II, Src: 00:60:b9:c6:6e:45, Dst: 00:60:b9:c1:ab:a3
Destination: 00:60:b9:c1:ab:a3 (Nitsuko_c1:ab:a3)
Source: 00:60:b9:c6:6e:45 (Nitsuko_c6:6e:45)
Type: 802.1Q Virtual LAN (0xSV9100)

Issue 5.0
Figure 5-7 Ethernet Frame Example - Layer 2 QoS Enabled (Continued)
802.1q Virtual LAN

101. .... .... .... = Priority: 5 (Layer 2 Priority = 5)
...0 .... .... .... = CFI: 0
.... 0000 0110 0011 = ID: 99
Type: IP (0x0800)
Internet Protocol, Src Addr: 172.16.0.101 (172.16.0.101), Dst Addr:
172.16.0.21 (172.16.0.21)
Version: 4
Header length: 20 bytes
Differentiated Services Field: 0x00 (DSCP 0x00: Default; ECN: 0x00)
0000 00.. = Differentiated Services Codepoint: Default
(0x00)
.... ..0. = ECN-Capable Transport (ECT): 0
.... ...0 = ECN-CE: 0
Total Length: 60
Identification: 0x0086 (134)
Flags: 0x00
0... = Reserved bit: Not set
.0.. = Don't fragment: Not set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 30
Protocol: UDP (0x11)
Header checksum: 0x4391 (correct)
Source: 172.16.0.101 (172.16.0.101)
Destination: 172.16.0.21 (172.16.0.21)
User Datagram Protocol, Src Port: 10022 (10022), Dst Port: 10020
(10020)
Source port: 10022 (10022)
Destination port: 10020 (10020)
Length: 40
Checksum: 0x0581 (correct)
Real-Time Transport Protocol
Stream setup by SDP (frame 1)
Setup frame: 1
Setup Method: SDP
10.. .... = Version: RFC 1889 Version (2)
..0. .... = Padding: False
...0 .... = Extension: False
.... 0000 = Contributing source identifiers count: 0
0... .... = Marker: False
.001 0010 = Payload type: ITU-T G.729 (18)
Sequence number: 28165
Timestamp: 21520
Synchronization Source identifier: 701655963
Payload: 76AC9D7AB6ACE2510B3A3338646DA738...

Issue 5.0
7.3 Layer 3 QoS
QoS is most commonly implemented at Layer 3. This allows the VoIP packets to
be prioritized by routers, before they are forwarded to their next hop.
Layer 3 QoS uses the Type of Service (ToS) field of the IP packet. This is an 8-bit
field in the header of the IP packet. The field can be used by Diffserv or IP
Precedence. Although these are two different standards, the actual field in the IP
packet is the same – Only the method of evaluating the bits differs.
QoS does not function only by using the ToS field (i.e., Marking the VoIP packets).
It is an end-to-end process and requires configuration on all networking devices.
Packet Marking is the first step in this process and is often the only step that the
NEC dealer performs.
Figure 5-8 Layer 3 QoS Example
Protocol Structure - IP/Pv4 Header (Internet Protocol Version 4)

4 Bits 4 Bits 8 Bits 16 Bits
Version IHL Type of Service Total Length
Identification Flags Fragment Offset
Time to Live Protocol Header Checksum
Source Address
Destination Address
Option + Padding
Data
Type of Service Field (Diffserv)

6 Bits 2 Bits
Differentiated Services Code Point ECN
(Not QoS related)
Type of Service Field (IP Precedence - Ref. RFC 1349)

3 Bits 1 Bit 1 Bit 1 Bit 1 Bit 1 Bit
IP Precedence Value Delay Throughput Reliability Cost MBZ
(must be
zero)

Issue 5.0
Listed below are the fields used in Figure 5-8 Layer 3 QoS Example.
Version – the version of IP currently used.
IP Header Length (IHL) – datagram header length. Points to the beginning of the
data. The minimum value for a correct header is 5.
Type-of-Service – Indicates the quality of service desired by specifying how an

upper-layer protocol would like a current datagram to be handled, and assigns
datagrams various levels of importance. This field is used for the assignment of
Precedence, Delay, Throughput and Reliability.
Total Length – Specifies the length, in bytes, of the entire IP packet, including the
data and header. The maximum length specified by this field is 65,535 bytes.
Typically, hosts are prepared to accept datagrams up to 576 bytes.
Identification – Contains an integer that identifies the current datagram. This field
is assigned by sender to help receiver to assemble the datagram fragments.
Flags – Consists of a 3-bit field of which the two low-order (least-significant) bits
control fragmentation. The low-order bit specifies whether the packet can be
fragmented. The middle bit specifies whether the packet is the last fragment in a
series of fragmented packets. The third or high-order bit is not used.
Fragment Offset – This 13-bit field indicates the position of the fragment data
relative to the beginning of the data in the original datagram, which allows the
destination IP process to properly reconstruct the original datagram.
Time-to-Live – This is a counter that gradually decrements down to zero, at

which point the datagram is discarded. This keeps packets from looping endlessly.
Protocol – Indicates which upper-layer protocol receives incoming packets after

IP processing is complete.
Header Checksum – Helps ensure IP header integrity. Since some header fields
change, e.g., Time To Live, this is recomputed and verified at each point that the
Internet header is processed.
Source Address – Specifies the sending node.
Destination Address – Specifies the receiving node.
Options – Allows IP to support various options, such as security.
Data – Contains upper-layer information.

Issue 5.0
7.4 IP Precedence
IP Precedence is a QoS method that combines a priority value with different on/off
parameters; Delay, Throughput, Reliability and Cost. The MBZ (Must be Zero) bit
is not used.
Using the ToS bits, you can define up to eight classes of service. Other devices
configured throughout the network can then use these bits to determine how to
treat the packet in regard to the type of service to grant it. These other QoS
features can assign appropriate traffic-handling policies including congestion
management and bandwidth allocation. By setting IP Precedence levels on
incoming traffic and using them in combination with QoS queuing features, you
can create differentiated service.
Table 5-1 Type of Service Field (IP Precedence - i Ref. REC 1349)
3 Bits 1 Bit 1 Bit 1 Bit 1 Bit 1 Bit

IP Precedence Value Delay Throughput Reliability Cost MBZ
(must be
zero)
IP Precedence Value Throughput

Binary Value Description
Value Description
Value 0 Normal Throughput
0 000 Routine 1 High Throughput
1 001 Priority
2 010 Immediate
3 011 Flash. Reliability
4 100 Flash Override Value Description
5 101 CRITIC/ECP 0 Normal Reliability
6 110 Internetwork Control 1 High Reliability
7 111 Network Control
Delay Cost
Value Description Value Description
0 Normal Delay 0 Normal Cost
1 Low Delay 1 Low Cost

Issue 5.0
7.5 Diffserv (Differentiated Service)
Differentiated Services (Diffserv) uses the ToS field in an IP header. Diffserv is

now commonly used instead of IP Precedence (refer to 7.4 IP Precedence on
page 5-23) as it provides greater flexibility. This method uses six bits of the ToS
field to determine the priority – which provides up to 64 possible values. The
combination of binary digits is known as the Diffserv Codepoint (DSCP).
Table 5-2 Diffserv Parameters
6 bits 2 bits
Differentiated Services Code Point ECN
(Not QoS related)
The example below shows an Ethernet Frame containing one RTP (speech)
packet. The IP Packet has the ToS field set to 101000 (binary) which is the
equivalent of Class Selector 5. The router(s) in this network should be
programmed to prioritize based on CS5.
Figure 5-9 Ethernet Frame Example - Containing One RTP (Speech) Packet
Source Destination Protocol

172.16.0.21 172.16.0.101 RTP
Info
Payload type=ITU-T G.729, SSRC=732771006, Seq=30885, Time=20560
Frame 159 (65 bytes on wire, 65 bytes captured)

Arrival Time: Jan 18, 2005 13:55:44.834497000
Time delta from previous packet: 0.000445000 seconds
Time since reference or first frame: 2.901831000 seconds
Frame Number: 159
Packet Length: 65 bytes
Capture Length: 65 bytes
Ethernet II, Src: 00:60:b9:c1:ab:a3, Dst: 00:60:b9:c6:6e:45
Destination: 00:60:b9:c6:6e:45 (Nitsuko_c6:6e:45)
Source: 00:60:b9:c1:ab:a3 (Nitsuko_c1:ab:a3)
Type: IP (0x0800)
Internet Protocol, Src Addr: 172.16.0.21 (172.16.0.21), Dst Addr:
172.16.0.101 (172.16.0.101)
Version: 4
Header length: 20 bytes
Diff Services Field: 0xa0 (DSCP 0x28: Class Selector 5; ECN: 0x00)
1010 00.. = Diff Services Codepoint: Class Selector 5 (0x28)
.... ..0. = ECN-Capable Transport (ECT): 0
.... ...0 = ECN-CE: 0

Issue 5.0
Figure 5-9 Ethernet Frame Example - Containing one RTP (Speech) Packet (Continued)
Total Length: 44
Identification: 0x0069 (105)
Flags: 0x00
0... = Reserved bit: Not set
.0.. = Don't fragment: Not set
..0. = More fragments: Not set
Fragment offset: 0
Time to live: 30
Protocol: UDP (0x11)
Header checksum: 0x431e (correct)
Source: 172.16.0.21 (172.16.0.21)
Destination: 172.16.0.101 (172.16.0.101)
User Datagram Protocol, Src Port: 10020 (10020), Dst Port: 10022
(10022)
Source port: 10020 (10020)
Destination port: 10022 (10022)
Length: 24
Checksum: 0x5293 (correct)
Real-Time Transport Protocol
Stream setup by SDP (frame 112)
Setup frame: 112
Setup Method: SDP
10.. .... = Version: RFC 1889 Version (2)
..1. .... = Padding: True
...0 .... = Extension: False
.... 0000 = Contributing source identifiers count: 0
0... .... = Marker: False
.001 0010 = Payload type: ITU-T G.729 (18)
Sequence number: 30885
Timestamp: 20560
Synchronization Source identifier: 732771006
Payload: 3ED0
Padding data: 00
Padding count: 2
7.6 Comparison of IP Precedence and Diffserv Values
As stated earlier, IP Precedence and Diffserv use the same 8-bit ToS field in the
IP header to mark packets. It is possible to have the same ToS value for either
method which means that the two methods can work alongside each other.
For example, if the VoIP equipment supports IP Precedence and the router can
prioritize only using the DSCP they can be set to the same value. Refer to Table
5-3 IP Precedence and Diffserv Values Comparison for the values.

Issue 5.0
Table 5-3 IP Precedence and Diffserv Values Comparison
DSCP DSCP
IP Precedence Description
Decimal Binary
0 000000 0 Class Selector 0

1 000001
2 000010
3 000011
4 000100
5 000101
6 000110
7 000111
9 001001
10 001010 AF11 (Assured Forwarding)
11 001011
13 001101
15 001111
17 010001
19 010011
21 010101
23 010111
25 011001
27 011011

Issue 5.0
Table 5-3 IP Precedence and Diffserv Values Comparison (Continued)
DSCP DSCP
Decimal Binary
29 011101
31 011111
33 100001
35 100011
37 100101
39 100111
41 101001
42 101010
43 101011
44 101100
45 101101
46 101110 EF (Expedited Forwarding)
47 101111
49 110001
50 110010
51 110011
52 110100
53 110101
54 110110
55 110111
57 111001
58 111010

Issue 5.0
Table 5-3 IP Precedence and Diffserv Values Comparison (Continued)
DSCP DSCP
Decimal Binary
59 111011
60 111100
61 111101
62 111110
63 111111
7.7 Programming QoS in the UNIVERGE SV9100 System
7.7.1 Marking Voice Traffic - Program 84-10-XX
Before programming the UNIVERGE SV9100 system, discuss the

requirements with the network engineering staff or the managed network
provider. If the ToS markings that are used are not specifically configured
into the network equipment, the voice traffic is handled by the default queue
and is given lowest priority.
7.7.2 UNIVERGE SV9100 Voice Protocols
The UNIVERGE SV9100 system supports the following types of VoIP traffic
(refer to the Input Data section of Program 84-10 : ToS Setup on page
5-32).
Table 5-4 Voice Protocols
Number Protocol Type Description

1 DRS Not Used
2 Protims Not Used
3 Voice Control (H.245) Communication from GPZ-IPLE to GPZ-IPLE
H.323 Communication from (TIE Lines) GPZ-IPLE to GPZ-IPLE

4 -- or --
Communication from GPZ-IPLE to Network
5 RTP/RTCP Voice (RTP) and Call Quality Data (RTCP)
6 SIP Communication from GPZ-IPLE to Standard SIP device.
7 CCISoIP CCIS Signaling Messages
8 SIP MLT Communication from GPZ-IPLE to SIP Multiline Terminal.
9 SIP Trunks Communication from GPZ-IPLE to Network
10 NetLink Communication from GPZ-IPLE to GPZ-IPLE

Issue 5.0
7.7.3 Configuring Diffserv
Use Program 84-10-10 to select the logic for marking the ToS field (refer to
Program 84-10 : ToS Setup on page 5-32). The choices are:
Table 5-5 DIffserv Configuration
Number ToS Mode Programs Enabled
0 None None – ToS bits are: 00000000
1 IP 84-10-02 Priority – 0=Lowest ~ 7=Highest

Precedence (ToS bits: 0~2)
84-10-03 Delay – 0=Normal, 1=Low
(Tos Bit: 3)
84-10-04 Throughput – 0=Normal, 1=High
(Tos Bit: 4)
84-10-05 Reliability – 0=Normal, 1=Low
(ToS Bit: 5)
84-10-06 Cost – 0=Normal, 1=Low
(ToS Bit: 6)
ToS Bit 7: Always 0
 Typically, only one of bits 3~6 is set to 1 and the other three bits
are set to 0. For example, to maximize route reliability, set
84-10-05 to 1 and leave 84-10-03, 84-10-04 and 84-10-06 at 0
(default).
2 Diffserv 84-10-07 DSCP Value in Decimals: 0~63 (ToS bits: 0~5)

ToS Bits 6 & 7 are not evaluated

Issue 5.0
7.7.4 Configuration Examples for Classification and Queuing
Figure 5-10 Common Network with Cisco Router shows a typical network
scenario and an example of a Cisco router configuration.
 This document provides a general description of VoIP technology, but it

does not discuss individual manufacturer solutions. This sample
configuration is provided as a common scenario. It is a good example of
how QoS can be implemented on a router.
NEC does not endorse or provide support on any third

party equipment unless it is supplied by NEC.
IMPORTANT
Figure 5-10 Common Network with Cisco Router
PC PC
192.168.1.50 192.168.2.50
Dallas Fort Worth

100Mbps
100Mbps
256Kbps
Private Circuit
(Leased Line)
100Mbps 10.0.0.1 10.0.0.2 100Mbps
192.168.1.1 192.168.2.1
Managed Switch Cisco 2621 Cisco 2621 Managed Switch
100 Mbps
100Mbps
Telephone System Telephone System

192.168.1.10 192.168.2.10
Refer to Table 5-6 Cisco Router Configuration Example for configuration

information about the Cisco 2621 router. A description of key commands
follows.

Issue 5.0
Table 5-6 Cisco Router Configuration Example
Current Configuration : 2023 bytes

version 12.3
hostname Cisco2621
|
class-map match-any VoIPClass (1)
match ip dscp cs5 (2)
policy-map VoIPPolicy (3)
class VoIPClass (4)
priority 50 (5)
class class-default (6)
fair-queue (7)
|
interface FastEthernet0/0
description Connects to Dallas LAN
ip address 192.168.1.1 255.255.255.0
|
interface Serial0/0
description Connects to Fort Worth via Kilostream
bandwidth 256 (8)
ip address 10.0.0.1 255.255.0.0
service-policy output VoIPPolicy (9)
encapsulation ppp
|
ip route 0.0.0.0 0.0.0.0 10.0.0.2
Configuration Example Explanation:
1. Defines a Class Map called VoIPClass.

2. Matches any packets that have the ToS field set to IP
Precedence 5 / DSCP 40 and assigns them to VoIPClass.
3. Defines a Policy Map called VoIPPolicy.
4. Creates a Class called VoIPClass and assigns this to the
VoIPPolicy.
5. Allocates 50Kbps of bandwidth to the VoIPClass.
6 & 7.Determines that any data that does not match VoIPClass
should be processed using the “fair-queue” method (i.e., No
Prioritization).
8. Determines the amount of bandwidth available on the Serial
interface – essential for the QoS calculations.
9. Applies the VoIP Policy to any packets that exit the serial
interface. This means that data being received (input) does
not use this policy.

Issue 5.0

84-10 : ToS Setup
Level:
IN
Description
Use Program 84-10 : ToS Setup to set up the Type of Service data.
Input Data
Protocol Type 1 = Not Used

2 = Not Used
3 = Voice Control
4 = H.323
5 = RTP/RTCP
6 = SIP
7 = CCISoIP
8 = SIP MLT
9 = SIP Trunk
10 = NetLink
11 = Video RTP/RTCP
Item
No.
01 ToS Mode 0 = Disable (Invalid) 0 When Input Data is set to 1,

1 = IP Precedence Item No. 07 is invalid.
2 = Diffserv When Data is set to 2, Item
No. 02 ~ 06 are invalid.
02 Priority, IP Precedence 0~7 0 1 = Router queuing priority

0 = Low
7 = High
03 Low Delay 0~1 0 1 = Optimize for low delay

0 = Normal Delay, Low routing
Delay
04 Wideband 0~1 0 1 = Optimize for high

(Throughout) 0 = Normal Throughput bandwidth routing
1 = High Throughput
05 High Reliability 0~1 0 1 = Optimize for reliability

0 = Normal Reliability routing
1 = Low Reliability

Issue 5.0
Item
No.
07 Priority (D.S.C.P. - 0~63 0 DSCP (Differentiated

Differentiated Services Services Code Point)
Code Point)
Conditions
 The system must be reset for these program options to take affect.

SECTION 8 PORT DESIGNATIONS
The following lists the port numbers for supported IP applications.
Table 5-7 Port Designations for IP Applications
SIP Trunk
SIP Trunk Signaling 5060 UDP
3rd Party SIP
SIP SLT Signaling 5070 UDP
SIP SLT Voice with GPZ-IPLE 10020~10531 UDP
H.323 Trunk
H.323 Signaling 1718/1719/1720 TCP
H.323 Trunk Voice with GPZ-IPLE 10020~10536 UDP
SV9100 IP - K-CCIS
IP K-CCIS Signaling 57000/5900 TCP

Issue 5.0
Table 5-7 Port Designations for IP Applications (Continued)
IP Trunk Voice with GPZ-IPLE 10020~10531 UDP
SV9100 NetLink
Primary System Signaling 58000/58001 TCP
Secondary System Signaling 58002 TCP
IP DSP Resource with GPZ-IPLE 10020~10531 UDP
NEC Proprietary SIP (SIP MLT)
SIP MLT Signaling 5080 UDP

UNIVERGE® SV9100
SIP Trunking Chapter 6
SECTION 1 DESCRIPTION
The UNIVERGE SV9100 IP Trunk SIP package sends the real time voice over the
corporate LAN or WAN. The voice from the telephone is digitized and then put into
frames to be sent over a network using Internet protocol.
(With CPU Version 2.00 and Lower):
With the SV9100 you can have two SIP Profiles allowing you to connect to two
different SIP Carriers, or allow you to have a SIP System Interconnection and
connection to a SIP Carrier.
(With CPU Version 3.00 and Higher):
With the SV9100 you can have six SIP Profiles allowing you to connect to multiple
SIP Carriers and SIP System Interconnection at the same time.
Using VoIP equipment at a gateway (a network point that acts as an entrance to

another network), the packetized voice transmissions from users in the company are
received and routed to other parts of the company Intranet (local area or wide area
network) or they can be sent over the Internet using CO lines to another gateway.
The GPZ-IPLE Daughter Board interface can provide IP trunks and Tie Lines that
can operate in the following modes:

Issue 5.0
 COI
 COID
 DID
 TLI
 DTI
Depending on the requirements and resource allocation in the LAN/WAN/Internet,

the GPZ-IPLE - SIP can be configured to use any of the following voice
compressions:
 G.711 u/A-Law – Highest Bandwidth
 G.729 (a) – Most often used
 The LAN/WAN or Internet connection is provided by a 10 Base-T/100 Base-TX
Ethernet.
For a list of vendors that have successfully completed interoperability certification go

to http://www.necntac.com and refer to Technical Documentation.
Conditions
 The option to set the SIP trunk Codec to G711 or G729 Fixed is
supported in Program 84-13-28.
 A maximum of 400 IP Trunks are supported in the SV9100.
 The SV9100 supports G.711 or T.38 for FAX.
 The SV9100 support fallback to G.711 from G.729/G.726 for data (FAX)
calls over SIP Trunk.
 A transferred call can not use T.38 at the transferred destination.
 SIP trunks are assigned in increments of four.
 Calling Party Name is not provided for outgoing calls on SIP trunks.
 SV9100 can setup six SIP profiles. Multiple SIP carriers or SIP System
can be used at the same time.
 SIP Profile is set to each SIP trunk in Program 14-18-05.
 VoIP DSP is required if a call is made via SIP trunk. Program 10-19(IPL
DSP Resource Selection) should set either "Use for SIP trunks" or
"Common use".
 With SV9100 software and GPZ-IPLE daughter board installed, half
duplex connections are not supported. For troubleshooting purposes, a
managed switch capable of port mirroring is required to capture packet
data from the SV9100 IPLE Ethernet port.
6-2 SIP Trunking

Issue 5.0
 The SV9100 software enables multiple SIP trunk carriers to be utilized

when NetLink is configured. Refer to Chapter 10 SV9100 NetLink for
additional details.
 Each SIP Profile will require a different SIP Listen Port assigned in
Program 84-14-06.
 Each SIP Profile will have to connect using one route because the
SV9100 can have only one Gateway.
 If SIP Trunks are programmed in a NetLink Secondary system, only SIP
Profile 1 can be utilized.
 The SV9100 can only use one DNS Server, two SIP Carriers that
require their own DNS Server can't be used.VBD is supported on the
GPZ-IPLE.
 VBD on SIP trunk is only supported on SLT terminals.
 VBD supports G.711/G.729 (PCMU/PCMA) CODEC.
 The selection of (FAX or VBD) mode is on a per terminal basis.
❍ The FAX connection may not work properly when in VBD Mode.
 VBD supports one way switching from the Voice session to VBD. VBD
to Voice session is not supported. When VBD session ends, the session
is closed.
 The VBD feature is not dependent on Carrier Type (Program 10-29-14).
 VBD is only supported on analog terminals and SIP trunks within the
same system.
 VoIPDB cancels the VAD and Echo canceler automatically when
changed into the VBD CODEC.
 When using VAD on SDP the setting is effective for G.711 and G.729
CODEC types.
 SIP Centrex Transfer is not supported.
Default Settings
None
System Availability
Terminals
All Multiline Terminals

Issue 5.0
Required Component(s)
 GPZ-IPLE
 R3 Enhancement License (0413)
 System Port License (0300)
 VoIP Resource License (5103)
 IP Trunk License (5001)
SECTION 2 SIP TRUNK E.164 SUPPORT
Description
With SIP Trunk E.164 Support enabled, the PBX is able to support SIP configurations
where the number presentation within the SIP messages is formatted using the
E.164 international numbering scheme. Specifically the system is able to handle the
+ digit when required as the International Access Code.
For example, a normal international SIP call can be dialed and displayed as follows:
Number dialed = 00441202223344
Request-URI: Invite sip: [email protected] SIP/2.0
With SIP Trunk E.164 Support enabled, the SIP call can be displayed once dialed as:
Request-URI: Invite sip:[email protected] SIP/2.0
This display is a requirement of certain SIP ITSPs (Internet Telephony Service

Providers) and may require that PBX handle these calls and modify any SIP
messages to the correct format accordingly.
This feature uses the following SIP header fields:

Request-URI
To
From
P-Asserted Identity
P-Preferred Identity
Conditions
 E.164 support is applied on the SIP trunk interface.
 E.164 is supported for all carrier choices (Program 10-29-14).
 Netlink multi-carrier support uses E.164 support across all carrier
configurations at the secondary nodes.
6-4 SIP Trunking

Issue 5.0
Default Settings
Disabled
System Availability
Terminals
Trunks
IP SIP
 GCD-CP10
 GPZ-IPLE
SECTION 3 SIP TRUNK E.164 CLIP ENHANCEMENT
Description
With the SIP Trunk E.164 CLIP Enhancement enabled, when an incoming SIP call
from an external ITSP is presented at the system with a + in the From header field as
the international access code, it is recognized and displayed as an international call
at the terminal display and also logged in the terminals incoming caller history,
allowing any outbound calls made from a multiline terminals caller history possible
using this numbering scheme.
This presentation can be a requirement of certain SIP ITSPs (Internet Telephony

Service Providers) so it is necessary the PBX can handle these calls and modify any
SIP messages to the correct format accordingly.

Issue 5.0
Conditions
 E.164 Enhancement is applied for the SIP trunk interface.
 Outgoing call from caller history of incoming calls is only possible from
multiline terminals.
 Netlink systems deployed in multiple countries using this feature may
not work correctly because the system will not know which international
code should be added at each node.
Default Settings
Disabled
System Availability
Terminals
Trunks
IP SIP
 GCD-CP10
 GPZ-IPLE
SECTION 4 VIDEO SUPPORT OVER SIP TRUNKS
Description
The SV9100 can support video calling over SIP interconnection trunks. The IP Trunk
license (5001/5103), IP Station licenses (5111) and Video MCU license (0042) are
required.
6-6 SIP Trunking

Issue 5.0
Conditions
 Calls over SIP Interconnection while in P2P mode cannot be put on
hold.
 Calls over SIP Interconnection while in P2P mode cannot be
transferred, i.e. an internal call cannot be transferred to a SIP
Interconnection trunk.
 A video caller cannot use CTI/OAI at the same time as the CTI/OAI
feature needs P2P to be set to off.
 When the video interconnection using a SIP trunk is configured, other
SIP connections, such as a SIP carrier connection is not supported in
the same system.
 Video capability in the initial invite message is required for the Video
Terminal.
 When using an MCU, the SV9100 requires the Carrier Type Setting
(Program 10-29-14) to be set to 0 = Standard.
 When using an MCU, the same video capability must be set between
the MCU and the Video SIP Terminal.
Default Settings
This feature is not enabled at default.
System Availability
Terminals
 Polycom HDX4003
 Polycom VVX1500D
GCD-CP10 with GPZ-IPLE installed.
Related Features
None

Issue 5.0
SECTION 5 GUIDE TO FEATURE PROGRAMMING
The Level 1, Level 2 and Level 3 columns indicate the programs that are assigned
when programming this feature in the order they are most commonly used. These
levels are used with PCPro and WebPro wizards for feature programming.
 Level 1 – these are the most commonly assigned programs for this feature.
 Level 2 – these are the next most commonly assigned programs for this
feature.
 Level 3 – these programs are not often assigned and require an expert level
working knowledge of the system to be properly assigned.
IP Trunk – (SIP) Session Initiation Protocol:
Level
Program
Number
1 2 3
10-12-03 GCD-CP10 Network Setup – Default 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Gateway 128.0.0.1 ~ 191.255.255.254
Assign the default gateway IP address for the

192.0.0.1 ~ 223.255.255.254
GCD-CP10.
10-12-05 GCD-CP10 Network Setup – NIC Interface 0 = Auto Detect 0

Set up the NIC Interface. 1 = 100Mbps, Full Duplex
2 = 100Mbps, Half Duplex
3 = 10Mbps, Full Duplex

5 = 1Gbps, Full Duplex
10-12-07 GCD-CP10 Network Setup – NAPT Router IP 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Address 128.0.0.1 ~ 191.255.255.254
(Default Gateway [WAN])
Define the IP Address of the WAN side of the
192.0.0.1 ~ 223.255.255.254 
router.
 Only used when Program 10-20-21 is enabled.
10-12-08 GCD-CP10 Network Setup – ICMP Redirect 0= (Enable) 0
When receiving ICMP redirect messages, this 1= (Disable)
determines if the IP Routing Table updates

automatically or not.
10-12-09 GCD-CP10 Network Setup – IP Address 0.0.0.0 ~ 126.255.255.254 172.16.0.10

Set IP address for GPZ-IPLE. 128.0.0.1 ~ 191.255.255.254 
192.0.0.1 ~ 223.255.255.254
6-8 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
10-12-10 GCD-CP10 Network Setup – Subnet Mask 128.0.0.0 | 192.0.0.0 | 224.0.0.0 255.255.0.0
Define the Media Gateway Subnet Mask |240.0.0.0 | 248.0.0.0 | 252.0.0.0
Address. |254.0.0.0 | 255.0.0.0
|255.128.0.0 | 255.192.0.0
|255.224.0.0 | 255.240.0.0
|255.248.0.0 | 255.252.0.0
|255.254.0.0 | 255.255.0.0
|255.255.128.0 | 255.255.192.0
|255.255.224.0 | 255.255.240.0
|255.255.248.0 | 255.255.252.0 
|255.255.254.0 | 255.255.255.0
|255.255.255.128
|255.255.255.192
|255.255.255.224
|255.255.255.240
|255.255.255.248
|255.255.255.252
|255.255.255.254
|255.255.255.255
10-12-11 GCD-CP10 Network Setup – NIC Setup 0 = Auto Detect 0

Define the LAN interface Speed and Mode of 1 = 100Mbps, Full Duplex
the VoIP Application supported. 3 = 10Mbps, Full Duplex 
 IPLE daughter board does not support half 5 = 1Gbps, Full Duplex
duplex connection.
10-19-01 VoIP DSP Resource Selection 0 = Common use for both IP 0

Select type of GPZ-IPLE DSP Resource. This extensions and trunks
program setting has no affect on the terminal/ 1 = IP Extension
trunk port assignment or usage. 2 = SIP Trunk
3 = CCIS/Networking
4 = Use for NetLink 
5 = Blocked
Paging
8 = Unicast Paging
10-23-01 SIP System Interconnection Setup – System 0 = No (Disable) 0

Interconnection 1 = Yes (Enable)
Determine if the system is interconnected to

another system.
10-23-02 SIP System Interconnection Setup – IP 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Address 128.0.0.1 ~ 191.255.255.254
Define the IP Address of the SIP System

192.0.0.1 ~ 223.255.255.254
Interconnection.
10-23-04 SIP System Interconnection Setup – Dial Maximum of 12 digits. No Setting

Number
Define Dial Number for the SIP System

Interconnection.

Issue 5.0
Level
Program
Number
1 2 3
10-23-06 SIP System Interconnection Setup – SIP Version 3.00 or higher CPU 1
Profile software:
Assign the Interconnection to a SIP Profile. 1 = Profile 1
2 = Profile 2
3 = Profile 3
4 = Profile 4 
5 = Profile 5
6 = Profile 6
Note: With Version 2.00 or lower
CPU software only two SIP
Profiles are supported.
10-28-01 SIP System Information Setup – Domain Maximum of 64digits. No Setting

Name
Define the Domain name. This information is Profile (1~2) (Version 2.00 or
generally provided by the SIP carrier. lower)

Profile (1~6) (Version 3.00 or
higher)
10-28-02 SIP System Information Setup – Host Name Maximum of 48 digits. No Setting
Define the Domain name. This information is
generally provided by the SIP carrier. Profile (1~2) (Version 2.00 or
lower)

higher)
10-28-03 SIP System Information Setup – Transport 0 = UDP 0

Protocol 1 = TCP
Define the Transport type. This option is always 2 = TLS (With Version 5.00 or
set to UDP. higher)

lower)
higher)
10-28-05 SIP System Information Setup – Domain 0 = IP Address 0

Assignment 1 = Domain Name
Define the Domain Assignment. This entry is
determined by what information the SIP carrier
provides. If the SIP carrier provides a server
Profile (1~2) (Version 2.00 or 
lower)
name: [email protected], then
the domain is: L0.cbeyond.net and the host
higher)
name is SIPconnect-sca.
10-28-06 SIP System Information Setup – IP Trunk 0 = Disable 0

Port Binding 1 = Enable
Enable/Disable IP Trunk Port binding.
lower)
higher)
6-10 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
10-29-01 SIP Server Information Setup – Default 0 = Off 0

Proxy (Outbound) 1 = On
Define the SIP Proxy setup, Default Proxy
(Outbound). When SIP trunking is used, this Profile (1~2) (Version 2.00 or
must be on. lower)
 If entries are made in Program 10-29-xx for an
SIP Server and the SIP Server is removed or not
higher)
used, the entries in Program 10-29-xx must be set
back to their default settings. Even if Program
10-29-01 is set to 0 (off), the UNIVERGE SV9100
still checks the settings in the remaining 10-29
programs.
10-29-02 SIP Server Information Setup – Default 0 = Off 0

Proxy (Inbound) 1 = On
Define the Default Proxy (Inbound).
lower)
higher)
10-29-03 SIP Server Information Setup – Default 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Proxy IP Address 128.0.0.1 ~ 191.255.255.254
Enter the default Proxy IP Address if the SIP 192.0.0.1 ~ 223.255.255.254
carrier is using an IP address for the proxy. In
most cases, this is left at the default entry as 
the domain name is used.
lower)
higher)
10-29-04 SIP Server Information Setup – Default 0 ~ 65535 5060

Proxy Port Number
Define the Proxy Port Number. Profile (1~2) (Version 2.00 or
lower)

higher)
10-29-05 SIP Server Information Setup – Registrar 0 = None 0

Mode 1 = Manual
This program is used to Enable/Disable the
ability to Register to the ITSP. Profile (1~2) (Version 2.00 or 
 In Registration mode (IP or Domain Name) lower)
this MUST be enabled. Profile (1~6) (Version 3.00 or
higher)
10-29-06 SIP Server Information Setup – Registrar IP 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Address 128.0.0.1 ~ 191.255.255.254
Define the Registrar IP Address. The carrier 192.0.0.1 ~ 223.255.255.254
may provide an IP address. In most cases, a
domain name is used so this entry is left at the 
default.
lower)
higher)

Issue 5.0
Level
Program
Number
1 2 3
10-29-07 SIP Server Information Setup – Registrar 0 ~ 65535 5060

Port Number
Define the Registrar Port Numbers. Profile (1~2) (Version 2.00 or
lower)

higher)
10-29-11 SIP Server Information Setup – Registrar Maximum of 128 characters. No Setting
Domain Name
Define the Registrar Domain Name (normally Profile (1~2) (Version 2.00 or
provided by the SIP carrier). lower)

For example: mysipserver.sipprovider.com Profile (1~6) (Version 3.00 or
higher)
10-29-12 SIP Server Information Setup – Domain Maximum of 64 characters. No Setting

Name
Define the domain name of the SIP PROXY Profile (1~2) (Version 2.00 or
Server provided by the SIP Carrier. lower)
For example: if the SIP Proxy server address is Profile (1~6) (Version 3.00 or 
proxy.sipprovider.com, you would assign higher)
sipprovider.com in this program.
 If no SIP Proxy address is provided, use the SIP
Registration address as the proxy address.
10-29-13 SIP Server Information Setup – Proxy Host Maximum of 48 characters. No Setting
Name
Assign the Proxy Host Name of the SIP PROXY Profile (1~2) (Version 2.00 or
Server provided by the SIP Carrier. lower)
For example: if the SIP Proxy server address is Profile (1~6) (Version 3.00 or 
proxy.sipprovider.com, you would assign proxy higher)
in the program.
 If no SIP Proxy address is provided, use the SIP
Registration address as the proxy address.
10-29-14 SIP Server Information Setup – SIP Carrier 0 ~ 26 0

Choice 1 ~ 26 =
Define the SIP Carrier Choice. Carrier Type A ~ Carrier Type Z
 Selecting Carrier B automatically sets
Program 10-29-16 to on (1). Program 10-29-16
MUST be set to off for incoming calls to route using
lower)
the lowest available trunk port. Profile (1~6) (Version 3.00 or
 Each certified vendor my use a different carrier higher)
type. Visit NTAC website (http://www.necntac.com)
to verify the proper setting per vendor.
6-12 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
10-29-15 SIP Server Information Setup – Registration 120 ~ 65535 seconds 3600
Expiry (Expire) Time
This program defines the SIP Trunk Profile (1~2) (Version 2.00 or
Registration timer. lower)
This timer is negotiated between the Carrier Profile (1~6) (Version 3.00 or
and the SV9100 during the registration process. higher) 
The Carrier will make the final decision on the
value to be used which means the value
specified in this program may be ignored.
 When half of this timer expires, the SV9100 will
re-register itself with the Carrier.
10-29-20 SIP Server Information Setup – 0~9 1

Authentication Trial 0 = No Authentication (1 Authentication
Define the number of times the SV9100 will Attempt)
attempt to authenticate before timing out and
not completing the registration process.
lower)
 Recommend changing this to at least 2 in case Profile (1~6) (Version 3.00 or
the first attempt did not succeed due to network higher)
problems or any other issues.
10-29-21 SIP Server Information Setup – NAT Router 0 = Disabled 0

If the SV9100 is connecting to the SIP Carrier 1 = Enabled
using NAT Translations, this setting must be
enabled. Profile (1~2) (Version 2.00 or 
lower)
higher)
10-36-01 SIP Trunk Registration Information Setup – 0 = Disable 0

Registration 1 = Enable
Enable/Disable the SIP trunk registration.
lower)
higher)
10-36-02 SIP Trunk Registration Information Setup – Maximum of 32 characters. No Setting

User ID
Assign the SIP User ID provided by your SIP Profile (1~2) (Version 2.00 or
Carrier. In most cases this is your 10 digit main lower)
billing number. This field is also used to send
outbound caller ID information when not
higher)
programmed on a per station or per trunk basis.
 For non-registration and SIP Tie Lines to
another system, you MUST have a USER ID
entered.

Authentication User ID
Define the Authentication USER ID for the SIP Profile (1~2) (Version 2.00 or
Trunk. lower)

higher)

Issue 5.0
Level
Program
Number
1 2 3

Authentication Password
Define the Authentication Password for the SIP Profile (1~2) (Version 2.00 or
Trunk. lower)

higher)
10-37-01 UPnP Setup – UPnP Mode 0 = Disable 0

Enable/Disable UPnP. 1 = Enable

10-37-02 UPnP Setup – Retry Time 0, 60 ~ 3600 60
Define the retry time for UPnP. (1 ~ 59 cannot be input)

10-54-01 License Configuration for Each Package – 1 ~ 255 Resource Licenses 0
License Code
Assign VoIP Resource Licenses (5103) to

GCD-CP10 slot (1).
10-67-01 SIP Stack Configuration Setup – DNS Mode 0 = Disable 0

Select whether DNS mode is enabled. 1 = Enable

10-67-02 SIP Stack Configuration Setup – DNS IP XXX.XXX.XXX.XXX 0.0.0.0
Address
Set the IP Address of DNS Server. The SV9100 
can only use one DNS Server, two SIP Carriers
requiring their own DNS Server can’t be used.
10-67-03 SIP Stack Configuration Setup – DNS Port 0 ~ 65535 53

Assign the DNS port.

10-67-04 SIP Stack Configuration Setup – DNS 0 ~ 65535 53
Source Port 
Assign the DNS source port.
10-68-01 IP Trunk Availability – Trunk Type 0 = None 0

Assign the trunk type as (1) SIP. 1 = SIP
2 = H.323 
3 = CCIS
(NetLink System ID 0 ~ 50)
10-68-02 IP Trunk Availability – Start Port 0 ~ 400 0

Assign the Start Port for your SIP trunks. (NetLink System ID 0 ~ 50)

10-68-03 IP Trunk Availability – Number of Port 0 ~ 400 0
Assign the number of SIP port trunks. (NetLink System ID 0 ~ 50)

Set the system internal (Intercom) numbering Programming
plan. The numbering plan assigns the first and Manual for
second digits dialed and affects the digits an default values.
extension user must dial to access other
extensions and features, such as service codes

and trunk codes. If the default numbering plan
does not meet the site requirements, use this
program to tailor the system numbering to the
site.
6-14 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
14-01-24 Basic Trunk Data Setup – Trunk-to-Trunk 0 = Disable (No) 0

Outgoing Caller ID through Mode 1 = Enable (Yes)
Enable/Disable the ability to send the original 
Caller ID through when the call is Forward
Off-Premise.
14-02-01 Analog Trunk Data setup – Signaling Type 0 = Dial Pulse (10 PPS) 2
(DP/DTMF) 1 = Dial Pulse (20 PPS) 
Set the outgoing signaling type for the tie trunk. 2 = DTMF
14-05-01 Trunk Group – Trunk Group Number Trunk Port 1 ~ 400 = Default = Trunks
Assign trunks to trunk groups. Priority 1 ~ 400 1 ~ 400 assigned
to trunk group 1
with priorities
equal to the 
trunk number.
Trunk 1 = Priority
1 Trunk 400 =
Priority 400.
14-18-05 IP Trunk Data Setup – SIP Profile (SIP Trunk) Profile 1 Profile 1
Assign each SIP Trunk to either SIP Profile. Profile 2
Profile 3
Profile 4

Profile 5
Profile 6
15-03-18 Single Line Telephone Basic Data Setup – Type: 0

Select Special Terminal 0 = FAX
Used for selecting Special terminal type (FAX or 1 = Modem
Modem). This setting influences how data is 
transmitted via SIP trunk.
 Program 15-03-03 must be set to 1 (Special) to
use this feature.
15-16 SIP Register ID Setup for Extensions None, 0~31

Define the SIP Register ID for Extensions (Extension 001~960)
Profile (1~2) (V2.00 or lower)

Profile (1~6) (V3.00 or higher)
21-17-01 IP Trunk (SIP) Calling Party Number Setup 0 ~ 9, , # Maximum of 16 digits No Setting
for Trunk
Assign the Caller Party Number for each IP
trunk. The assigned number is sent to the
central office when the caller places an 
outgoing call. If the Calling Party Number is
assigned by both Program 21-17 and Program
21-18/ 21-19, the system uses the entry in
Program 21-18/21-19.

Issue 5.0
Level
Program
Number
1 2 3
21-19-01 IP Trunk (SIP) Calling Party Number Setup 0 ~ 9, , # Maximum of 16 digits No Setting
for Extension
Assign the Calling Party Number for each Profile (1~2) (V2.00 or lower)
extension. The assigned number is sent to the Profile (1~6) (V3.00 or higher)
21-18/Program 21-19, the system uses the data
in Program 21-18/Program 21-19.
22-02-01 Incoming Call Trunk Setup Trunks 1 ~ 400 0

Assign the incoming trunk type for each trunk. 0 = Normal
1 = VRS (second
dial tone if no
VRS installed)
2 = DISA
3 = DID 
4 = DIL
5 = E&M Tie line
6 = Delayed VRS
7 = ANI/DNIS
8 = DID(DDI) Mode
Switching
44-02-01 Dial Analysis Table for ARS/F-Route Access Maximum of eight digits. No Setting
– Dial (Use line key 1 for a ‘Don’t Care’
Set the Dial digits for the Pre-Transaction Table digit, @)
for selecting ARS/F-Route (eight digits max:

1 ~ 9, 0  #, @). To enter a wild card/don’t care
digit, press Line Key 1 to enter an @.
44-02-02 Dial Analysis Table for ARS/F-Route Access 0 = No Setting (None) 0

Set the Service Type (0 ~ 3) for the Pre- 2 = ARS/F-Route Table
Transaction Table for selecting ARS/F-Route. (F-Route)

Table (Option)
44-02-03 Dial Analysis Table for ARS/F-Route Access 1 = Delete Digit = 0 ~ 255 0
If a Service Type is selected in Program 2 = 0 ~ 500
44-02-02, set the additional data, if required, for
the Pre-Transaction Table for selecting ARS/F-
(0 = No Setting) 
Route (24 digits max: 1 ~ 9, 0  #, @). To enter Table Number = 0 ~ 4
a wild card/don’t care digit, press Line Key 1 to
(0 = No Setting)
enter an @.
44-05-01 ARS/F-Route Table – Trunk Group Number 0-100,101-150,255 0

Assign the trunk group to be used by the ARS/ 0 = Not Set
F-Route Table. 1~100 = Trunk Group from
PRG14-05

6-16 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
44-05-09 ARS/F-Route Table – Maximum Digit 0 ~ 24 0

Input the maximum number of digits to send 
84-10-01 ToS Setup – ToS Mode 0 = Disable (Invalid) 0

When Input Data is set to 1, Protocol 7 is 1 = IP Precedence
invalid. When Data is set to 2, Protocols 2 ~ 6

2 = Diffserv
are invalid.
84-10-02 ToS Setup – Priority, IP Precedence 0~7 0

1 = Router queuing priority. 0 = Low 
7 = High
84-10-03 ToS Setup – Low Delay 0~1 0

1 = Optimize for low delay routing. 0 = Normal Delay, Low Delay

84-10-04 ToS Setup – Wideband (Throughout) 0~1 0
1 = Optimize for high bandwidth routing. 0 = Normal Throughput 
1 = High Throughput
84-10-05 ToS Setup – High Reliability 0~1 0

1 = Optimize for reliability routing. 0 = Normal Reliability 
1 = Low Reliability
84-10-07 ToS Setup – Priority (D.S.C.P. - 0 ~ 63 0

Differentiated Services Code Point) 
DSCP (Differentiated Services Code Point).
84-13-01 SIP Trunk CODEC Information Basic Setup – 1~4 2

Number of G.711 Audio Frames 1 = 10ms
Set the G.711 Audio Frame Number. 2 = 20ms
3 = 30ms
4 = 40ms


84-13-02 SIP Trunk CODEC Information Basic Setup – 0 = Disable 0

G.711 Voice Activity Detection Mode 1 = Enable
Enable/Disable the G.711 VAD Detection Mode. 
84-13-03 SIP Trunk CODEC Information Basic Setup – 0 = A-law 0

G.711 Type 1 = -law
Define the G.711 type. 
84-13-04 SIP Trunk CODEC Information Basic Setup – 0 ~ 300ms 20

Set the minimum G.711 Jitter Buffer. Profile (1~2) (V2.00 or lower)


Issue 5.0
Level
Program
Number
1 2 3

G.711 Jitter Buffer (Average)
Set the average G.711 Jitter Buffer. Profile (1~2) (V2.00 or lower)


Set the maximum G.711 Jitter Buffer. Profile (1~2) (V2.00 or lower)

84-13-07 SIP Trunk CODEC Information Basic Setup – 1 = 10ms 2

Set the G.729 Audio Frame Number. 3 = 30ms
4 = 40ms
5 = 50ms 
6 = 60ms


G.729 Voice Activity Detection Mode 1 = Enable
Enable/Disable the G.729 VAD Detection Mode. 

Set the minimum G.729 Jitter Buffer. Profile (1~2) (V2.00 or lower)


G.729 Jitter Buffer (Average)
Set the average G.729 Jitter Buffer. Profile (1~2) (V2.00 or lower)


Set the maximum G.729 Jitter Buffer. Profile (1~2) (V2.00 or lower)

84-13-17 SIP Trunk CODEC Information Basic Setup – 1 = Static 3

Jitter Buffer Mode 2 = Self Adjusting 
Set the Jitter Buffer Mode.
6-18 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
84-13-18 SIP Trunk CODEC Information Basic Setup – 0 ~ 30 = -19dB ~ +10dB 20

VAD Threshold 1 =-19dB (-49dBm)
Set the VAD (Voice Activity Detection) :
threshold. 2 =0dB (-30dBm)
:
29 =9dBm (-21dBm)

30 =0dBm (-20dBm)

84-13-28 SIP Trunk CODEC Information Basic Setup – 0 = G.711_PT 0

Audio Capability Priority 2 = G.729_PT
Define the Codec Priority. 3 = G.722_PT
4 = G.726_PT
6 = G711_Fix 
7 = G729_Fix


Define the number of G.722 Audio Frames. 2 = 20ms
3 = 30ms
4 = 40ms



Define the minimum level for the G.722 jitter Profile (1~2) (V2.00 or lower)

buffer. Profile (1~6) (V3.00 or higher)

G.722 Jitter Buffer (average)
Define the average level for the G.722 Jitter Profile (1~2) (V2.00 or lower)

Buffer. Profile (1~6) (V3.00 or higher)

Define the Max level for the G.722 Jitter buffer. Profile (1~2) (V2.00 or lower)


Issue 5.0
Level
Program
Number
1 2 3

Define the number of G.726 audio frames. 2 = 20ms
3 = 30ms
4 = 40ms



G.726 VAD Mode 1 = Enable
Enable/Disable the VAD mode for G.726. 

Define the minimum level for the G.726 jitter Profile (1~2) (V2.00 or lower)


G.726 Jitter Buffer (average)
Define the average level for the G.726 jitter Profile (1~2) (V2.00 or lower)


Define the max level for the G.726 jitter buffer. Profile (1~2) (V2.00 or lower)


RTP Filter 1 = Enable
To avoid incorrect voice pass connection, this 
program checks the sending side address from Profile (1~2) (V2.00 or lower)
received RTP packet at VoIPDB.

VAD Negotiation on SDP 1 = Enable
Used to determine the VAD method for setting
VAD information on SDP. Profile (1~2) (V2.00 or lower) 
This is effective when VAD is enabled on each Profile (1~6) (V3.00 or higher)
CODEC.
 Supports G.711 and G.729.
Voice Band Data (VBD) 1 = Enable
Used to Disable or Enable VBD. 
Program 15-03-03 must be set to 1 (Special) to Profile (1~2) (V2.00 or lower)
use this feature. Profile (1~6) (V3.00 or higher)
6-20 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
84-13-67 SIP Trunk CODEC Information Basic Setup – 96 ~ 127 97

VBD Payload Type Profile (1~2) (V2.00 or lower) 
Used to specify the payload type for VBD. Profile (1~6) (V3.00 or higher)
84-13-70 SIP Trunk CODEC Information Basic Setup – Mode 1,Mode 2 Mode 1
Video Quality Mode Profile (1~2) (V2.00 or lower)
This program specifies the SIP trunk video Profile (1~6) (V3.00 or higher)
quality mode. Use this program in conjunction
with 84-27-20 for Mode 1 and 84-27-21 for 
Mode 2 video quality settings.
Mode 1 = CIF (352x288)
Mode 2 = VGA (640x480)
84-13-71 SIP Trunk CODEC Information Basic Setup – H.264 H.264

Video CODEC Profile (1~2) (V2.00 or lower)
This program specifies the video CODEC. At

this time only H.264 is supported.
84-13-72 SIP Trunk CODEC Information Basic Setup – Static, Self-adjusting Self-adjusting
Jitter Buffer Mode for Video Profile (1~2) (V2.00 or lower)
This program sets the jitter buffer size
adjustment. At default this is set to self
Profile (1~6) (V3.00 or higher) 
adjusting and should only be changed when
directed by support.

Minimum Jitter Buffer Mode for Video Profile (1~2) (V2.00 or lower)
This program sets the minimum value of jitter Profile (1~6) (V3.00 or higher)
buffer for the video stream. This value must be 
smaller than the value of the maximum jitter
buffer. This value should only be changed for if
needed for highly congested networks.

Initial Jitter Buffer Mode for Video Profile (1~2) (V2.00 or lower)
This program sets the initial value of jitter buffer Profile (1~6) (V3.00 or higher)
for the video stream. This value must be
smaller than the value of the maximum jitter

buffer and bigger than the value of the minimum
jitter buffer. This value should only be changed
for if needed for highly congested networks.

Maximum Jitter Buffer Mode for Video Profile (1~2) (V2.00 or lower)
This program sets the maximum value of jitter Profile (1~6) (V3.00 or higher)
buffer for the video stream.
It is used only when 84-19-72 (Jitter Buffer
Mode for video) is set to 1:Fixed.

This value must be bigger than the value of the
minimum jitter buffer and should only be
changed if needed for highly congested
networks.

Issue 5.0
Level
Program
Number
1 2 3
84-14-06 SIP Trunk Basic Information Setup – SIP 1 ~ 65535 Profile 1 = 5060
Trunk Port Number Profile 2 = 5062
Set the SIP UA (User Authorized) Trunk port Profile 3 = 5090
number (Receiving Transport for UNIVERGE Profile 4 = 5092
Profile (1~6) (V3.00 or higher) Profile 5 = 5094
SV9100 SIP).
Profile 6 = 5096

 Each SIP Profile will need to have a different Profile(1-2)
SIP Listen Port. (V2.00 or Lower)
Profile(1-6)
(V3.00 or Higher
84-14-07 SIP Trunk Basic Information Setup – 1 ~ 65535 seconds 0

Session Timer Value
The Session Timer, when enabled, will allow Profile (1~2) (V2.00 or lower)
periodic refreshing of the SIP session using the Profile (1~6) (V3.00 or higher)
SIP Re-Invite message. The refreshing will
allow both the SV9100 and the Carrier to
determine if the session is still active. The

periodic refresh can be used to release calls
that are not active anymore when the SIP BYE
message may have been lost.
 When this timer is set to 0, the session timer is
NOT included in the SIP Invite message.
84-14-08 SIP Trunk Basic Information Setup – 1 ~ 65535 seconds 1800

Minimum Session Timer Value
This timer controls the minimum accepted value Profile (1~2) (V2.00 or lower)
of the Session timer that the SV9100 will allow. Profile (1~6) (V3.00 or higher)

If the Session timer from the Carrier is lower
than the value defined here, the SV9100 will
deny the call.
84-14-09 SIP Trunk Basic Information Setup – Called 0 = Request URI 0

Party Information 1 = To Header
This Program is used for inbound calls only.
When set to a 0 (Request-URI), the SV9100 will Profile (1~2) (V2.00 or lower)
look into the SIP "Request-URI" field to route Profile (1~6) (V3.00 or higher)

the inbound call.
When set to a 1 (To Header), the SV9100 will
look into the SIP "To" field to route the inbound
call.
84-14-10 SIP Trunk Basic Information Setup – URL 0 = SIP-URL 0

Type 1 = TEL-URL
Define the URL type for SIP trunks. Profile (1~2) (V2.00 or lower)

6-22 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
84-14-11 SIP Trunk Basic Information Setup – URL/To 0 = Proxy Server Domain 0
HeaderSetting Information 1 = SIP UA Domain
When set to a 0 (Proxy Server Domain), the
SV9100 will use the proxy settings in Programs
10-29-12 and 10-29-13 within the SIP Request-
URI and To headers. If neither of these Profile (1~6) (V3.00 or higher)
programs are assigned, the value in program
10-12-11 is used.
When set to a 1 (SIP UA Domain), the SV9100
will use the domain settings in Program
10-28-02 within the SIP Request-URI and To
headers.
84-14-15 SIP Trunk Basic Information Setup – 100rel 0 = Use Default Settings (100rel 0
Settings included)
This program specifies if the 100rel message is 1 = Use Opposite Settings
included or not. (100rel not included)
When set to a 0 (Use Default Settings), the
100rel will be included in the initial SIP Invite Profile (1~2) (V2.00 or lower)
and any provisional 1XX responses (excluding Profile (1~6) (V3.00 or higher)
the 100 trying).
When set to a 1 (Use opposite settings), the
100rel will NOT be included in the initial SIP
Invite and provisional 1XX responses
(excluding the 100 trying)
84-26-01 IPL Basic Setup – IP Address xxx.xxx.xxx.xxx 172.16.0.20

Assign the IP address for each DSP on the 
GPZ-IPLE.
84-26-02 IPL Basic Setup – RTP Port Number 0 ~ 65534 10020

Assign the RTP port number to be used for
each DSP on the GPZ-IPLE.

 Only even numbered ports are supported.
84-26-03 IPL Basic Setup – RTCP Port Number (RTP 0 ~ 65534 10021
Port Number + 1)
Define the TCP port number for RTCP to use

for each DSP.
84-31-01 VoIPDB Echo Canceler Setup-Echo 0 = Disable 1 = Enable

Canceler Mode 1 = Enable
Enable/Disable echo canceler mode. Profile (1~2) (V.2.00 or lower)


Canceler NLP Mode(2W) 1 = Enable
Set echo canceler NLP mode. Profile (1~2) (V2.00 or lower)


Canceler Comfort Noise Mode 1 = Enable
Enable/Disable Echo Canceler Comfort Noise Profile (1~2) (V2.00 or lower)

mode. Profile (1~6) (V3.00 or higher)

Issue 5.0
Level
Program
Number
1 2 3
84-31-05 VoIPDB Echo Canceler Setup-Echo 0 ~ 87 0

Canceler Comfort Noise Mode Profile (1~2) (V2.00 or lower) 
Define echo canceler tail displacement. Profile (1~6) (V3.00 or higher)
84-31-06 VoIPDB Echo Canceler Setup-Echo 1=32ms 128

Canceler tail Length 2=48ms
Select echo Canceler tail length. 3=64ms
4=80ms
5=96ms 
6=112ms
7=128ms
84-31-07 VoIPDB Echo Canceler Setup-Echo 0 ~ 6(-9db ~ 9db) 6db

Canceler Default ERL level 0 = -9db
Select echo canceler default ERL level. 1 = -6db
2 = -3db
3 = 0db
4 = 3db

5 = 6db
6 = 9db
84-31-08 VoIPDB Echo Canceler Setup-Echo 1 = Line E.C. Line E.C.

Canceler Default Echo type 2 = Acou.E.C.
Select echo canceler echo type. Profile (1~2) (V2.00 or lower)

84-31-10 VoIPDB Echo Canceler Setup-Tx Level 0 = Disable TXHLC mode

Control 1 = TxL.C. mode
Select Tx level control. 2 = TxA.L.C. mode
3 = TxHLC mode

84-31-11 VoIPDB Echo Canceler Setup-Tx Level 0 ~ 16(-24 ~ 24db) 0.0db

Control Level 0 = -24db
Select Tx level control level. 1 = -21db
2 = -18db
:
8 = 0db
:

14 = 18db
15 = 21db
16 = 24db
6-24 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
84-31-12 VoIPDB Echo Canceler Setup-Tx Automatic 0 ~ 12(-42 ~ -6dBm) -21dBm

Level Control Level 0 = -42dBm
Select Tx Automatic Level Control Level. 1 = -39dBm
:
7 = -21dBm
:

11 = -9dBm
12 = -6dBm
84-31-13 VoIPDB Echo Canceler Setup-Tx HLC 0-42 (-42 ~ 0dBm) 1dBm (41)
Threshold 0 = -42dBm
Select Tx HLC threshold. 1 = -41dBm
: 
42 = 0dBm
Profile (1~6) (V3.00 or higher).
84-31-14 VoIPDB Echo Canceler Setup-Tx Signal 0 = Disable Enable

Limiter Mode 1 = Enable
Enable/Disable Tx Signal Limiter Mode. Profile (1~2) (V2.00 or lower)

84-31-15 VoIPDB Echo Canceler Setup-Tx Signal 0-42 (-42 ~ 0dBm) 1dBm (41)
Limiter Threshold 0 = -42dBm
Select Tx signal limiter threshold. 1 = -41dBm
: 
42 = 0dBm
84-31-16 VoIPDB Echo Canceler Setup-Rx Level 0 = Disable Disable

Control 1 = RxL.C.mode
Select Rx Level Control. 2 = RxA.L.C.Flg 
84-31-17 VoIPDB Echo Canceler Setup-Rx Level 0 ~ 16(-24 ~ 24db) 0db

Control Level 0 = -24db
Select Rx Level Control Level. 1 = -21db
2 = -18db
:
8 = 0db
:

14 = 18db
15 = 21db
16 = 24db

Issue 5.0
Level
Program
Number
1 2 3
84-31-18 VoIPDB Echo Canceler Setup-Rx Automatic 0 ~ 12 (-42 ~ -6dBm) -21

Level Control Level 0 = -42dBm
Select Rx Automatic Level Control Level. 1 = -39dBm
:
7 = -21dBm
:

11 = -9dBm
12 = -6dBm
84-31-37 VoIPDB Echo Canceler Setup-Echo 0 = Disable Enable

Canceler NLP Mode(4W) 1 = Enable
Select Echo Canceler NLP Mode(4W). Profile (1~2) (V2.00 or lower)

84-34-01 VoIPDB DTMF Setup-DTMF Relay Mode 0 = Disable Disable

Select DTMF Relay Mode. 1 = RFC2833
2 = H.245 
84-34-02 VoIPDB DTMF Setup-DTMF Payload Number 96 ~ 127 110

Define DTMF Payload Number. Profile (1~2) (V2.00 or lower) 
Profile (1~6) (V.3.00 or higher)
84-34-03 VoIPDB DTMF Setup-DTMF Detection Type 1~5 1

Define DTMF Detection Type. Profile (1~2) (V2.00 or lower) 
84-34-04 VoIPDB DTMF Setup-DTMF Transmit Type 1~5 1

Define DTMF Transmit Type. Profile (1~2) (V2.00 or lower) 
84-34-05 VoIPDB DTMF Setup-DTMF Relay (inband) 1~5 1

Retransmit Type Profile (1~2) (V2.00 or lower) 
Define DTMF Relay (inband) Retransmit Type. Profile (1~6) (V3.00 or higher)
84-38-01 VoIPDB Network Side Echo Canceler Setup- 0 = Disable Disable

Echo Canceler Mode 1 = Enable
Enable/Disable echo canceler mode. Profile (1~2) (V2.00 or lower)


Echo Canceler NLP Mode. 1 = Enable
Set echo canceler NLP mode. Profile (1~2) (V2.00 or lower)

84-38-03 VoIPDB Network Side Echo Canceler Setup- 0 = Disable Enable

Echo Canceler Comfort Noise Mode 1 = Enable
Enable/Disable Echo Canceler Comfort Noise Profile (1~2) (V2.00 or lower)

Mode. Profile (1~6) (V3.00 or higher)
6-26 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
84-38-05 VoIPDB Network Side Echo Canceler Setup- 0 ~ 87 (0ms ~ 870ms) 0

Echo Canceler Tail Displacement Profile (1~2) (V2.00 or lower) 
Define echo canceler tail displacement. Profile (1~6) (V3.00 or higher)
84-38-06 VoIPDB Network Side Echo Canceler Setup- 1=32ms 128

Echo Canceler Tail Length 2=48ms
Select Echo Canceler tail length. 3=64ms
4=80ms
5=96ms 
6=112ms
7=128ms
84-38-07 VoIPDB Network Side Echo Canceler Setup- 0 ~ 6(-9db ~ 9db) 6db
Echo Canceler Default ERL Level 0 = -9db
Select echo canceler default ERL level. 1 = -6db
2 = -3db
3 = 0db
4 = 3db

5 = 6db
6 = 9db
84-38-08 VoIPDB Network Side Echo Canceler Setup- 1 = Line E.C. Line E.C.
Echo Canceler Default Echo type 2 = Acou.E.C.
Select echo canceler echo type. Profile (1~2) (V2.00 or lower)


Tx Level Control 1 = Manual
Select Tx level control. 2 = Auto
3 =HLC

84-38-11 VoIPDB Network Side Echo Canceler Setup- 0 ~ 16(-24 ~ 24db) 0.0db
Tx Level Control Level 0 = -24db
Select Tx level control level. 1 = -21db
2 = -18db
:
8 = 0db
:

14 = 18db
15 = 21db
16 = 24db

Issue 5.0
Level
Program
Number
1 2 3
84-38-12 VoIPDB Network Side Echo Canceler Setup- 0 ~ 12(-42 ~ -6dBm) -21dBm
Tx Automatic Level Control Level 0 = -42dBm
Select Tx Automatic Level Control Level. 1 = -39dBm
:
7 = -21dBm
:

11 = -9dBm
12 = -6dBm
84-38-13 VoIPDB Network Side Echo Canceler Setup- 0-42 (-42 ~ 0dBm) 1dBm (41)
Tx HLC Threshold 0 = -42dBm
Select Tx HLC threshold. 1 = -41dBm
: 
42 = 0dBm

Tx Signal Limiter Mode 1 = Enable.
Enable/Disable Tx Signal Limiter Mode. Profile (1~2) (V2.00 or lower)

84-38-15 VoIPDB Network Side Echo Canceler Setup- 0-42 (-42 ~ 0dBm) 1dBm (41)
Tx Signal Limiter Threshold 0 = -42dBm
Select Tx signal limiter threshold. 1 = -41dBm
: 
42 = 0dBm

Rx Level Control 1 = Manual
Select Rx Level Control. 2 = Auto 
84-38-17 VoIPDB Network Side Echo Canceler Setup- 0 ~ 16(-24 ~ 24db) 8 (0db)
Rx Level Control Level 0 = -24db
Select Rx Level Control Level. 1 = -21db
2 = -18db
:
8 = 0db
:

14 = 18db
15 = 21db
16 = 24db
6-28 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
84-38-18 VoIPDB Network Side Echo Canceler Setup- 0 ~ 12 (-42 ~ -6dBm) -21
Rx Automatic Level Control Level 0 = -42dBm
Select Rx Automatic Level Control Level. 1 = -39dBm
:
7 = -21dBm
:

11 = -9dBm
12 = -6dBm
90-10-01 System Alarm Setup – Alarm Type 0 = Not Set No Setting

Define if Alarms are Minor, Major, or Not Set. 1 = Major Alarm 
2 = Minor Alarm
SIP Trunk E.164 Support
Level
Program
Number
1 2 3
10-02-01 Location Setup – Country Code Dial (maximum of four digits) 1

Enter the country code. 0 ~ 9, , #

10-02-02 Location Setup – International Access Dial (maximum of four digits) No Setting
Codes 0 ~ 9, , # 
Enter the international access code.
44-01-02 System Options for ARS/F-Route – Dial Dial (maximum of one digit) No Setting
Tone Simulation 0 ~ 9, , # cannot be used
When first dialed, digit matches the data set in
this Program, system sends simulated DT to Profile (1~2) (V2.00 or lower)

calling party after receiving first digit.
Numbering plan for the dial needs to be
configured as F-Route in Program 11-01.
44-01-03 System Options for ARS/F-Route – Tone 0 = Internal Dial Tone 0

Type 1 = External Dial Tone
Set simulated DT to type which can change the 
tone used in Program 44-01-02 and Program
44-02-04.

Issue 5.0
Level
Program
Number
1 2 3
84-14-13 SIP Trunk Basic Information Setup – 0 = Off 0

Incoming/ Outgoing SIP Trunk for E.164 1 = Mode 1
When this data is set to 1, then for any 2 = Mode 2
outbound SIP calls a + is added as a prefix to 3 = Mode 3
the Request-URI, To and From header fields of
the SIP message.
When it is set to 2 then if the dialed international
Profile (1~2) (V2.00 or lower) 
access code matches the value in Program Profile (1~6) (V3.00 or higher)
10-02-02 this value is removed from the
number dialed and the + added as a prefix to
the SIP Message.
SIP Trunk E.164 CLIP Enhancement
Level
Program
Number
1 2 3
10-02-01 Location Setup – Country Code Dial (maximum of four digits) 1

Enter the country code. 0 ~ 9, , #

10-02-02 Location Setup – International Access Dial (maximum of four digits) No Setting
Codes 0 ~ 9, , # 
Enter the international access code.
10-02-03 Location Setup – Other Area Access Code Dial (maximum of two digits) 9
Enter the other area access code. 0 ~ 9, , #

84-14-13 SIP Trunk Basic Information Setup – 0 = Off 0
Incoming/ Outgoing SIP Trunk for E.164 1 = Mode 1
When this data is set to 1, then for any 2 = Mode 2
outbound SIP calls a + is added as a prefix to 3 = Mode 3
the SIP message.
When it is set to 2 then if the dialed international
Profile (1~2) (V2.00 or lower) 
access code matches the value in Program Profile (1~6) (V3.00 or higher)
10-02-02 this value is removed from the
number dialed and the + added as a prefix to
the SIP Message.
6-30 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
84-14-16 SIP Trunk Basic Information Setup – SIP 0 = Disable 0

Trunk SIP-URI E.164 Incoming Mode 1 = Mode 1
When this data is set to 1, then for any inbound 2 = Mode 2
SIP calls that include a + and a country code
not defined in Program 10-02-01, delete the +
and add the International Access Code in
Program 10-02-02. If the country code is a Profile (1~6) (V3.00 or higher)
match then delete both the + and country code
but do not add the International Access code.

When it is set to 2, then for any inbound SIP
calls that include a + and a country code not
defined in Program 10-02-01, delete the + and
add the International Access Code in Program
10-02-02. If the country code is a match then
delete both the + and country code and add the
Caller ID Edit Code from Program 10-02-03.
Video Support over SIP Trunks
Program Name/Description Level

Program
Input Data Default
Number
1 2 3

It is recommended to set this program to 128.0.0.1 ~ 191.255.255.254
0.0.0.0. All connections to the system are made

192.0.0.1 ~ 223.255.255.254
through the IPLE.
10-12-03 GCD-CP10 Network Setup – Default 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Gateway 128.0.0.1 ~ 191.254.255.254
Assign the default gateway IP address for the

192.0.0.1 ~ 223.255.255.254
GCD-CP10.
10-12-08 GCD-CP10 Network Setup – ICMP Redirect 0= (Enable) 0

When receiving ICMP redirect messages, this 1= (Disable)
determines if the IP Routing Table updates


Set IP address for IPLE. 128.0.0.1 ~ 191.255.255.254
 The IP Address assigned in Program 10-12-01 192.0.0.1 ~ 223.255.255.254 
cannot start with the same leading digits as the IP
Address assigned here.

Issue 5.0

Program
Input Data Default
Number
1 2 3
10-12-10 GCD-CP10 Network Setup – Subnet Mask 128.0.0.0 | 192.0.0.0 | 224.0.0.0 255.255.0.0
Define the Media Gateway Subnet Mask |240.0.0.0 | 248.0.0.0 | 252.0.0.0
Address. |254.0.0.0 | 255.0.0.0
|255.128.0.0 | 255.192.0.0
|255.224.0.0 | 255.240.0.0
|255.248.0.0 | 255.252.0.0
|255.254.0.0 | 255.255.0.0
|255.255.128.0 | 255.255.192.0
|255.255.224.0 | 255.255.240.0
|255.255.248.0 | 255.255.252.0 
|255.255.254.0 | 255.255.255.0
|255.255.255.128
|255.255.255.192
|255.255.255.224
|255.255.255.240
|255.255.255.248
|255.255.255.252
|255.255.255.254
|255.255.255.255
10-12-11 GCD-CP10 Network Setup – NIC Setup 0 = Auto Detect 0

Define the LAN interface Speed and Mode of 1 = 100Mbps, Full Duplex
the VoIP Application supported. 3 = 10Mbps, Full Duplex 
 IPLE daughter board does not support half 5 = 1Gbps, Full Duplex
duplex connection.
10-23-01 SIP System Interconnection Setup – System 0 = No (Disable) 0

Interconnection 1 = Yes (Enable)
Determine if the system is interconnected to
another system.

 For the SIP System Interconnection set to
1 (Yes).
10-23-02 SIP System Interconnection Setup – IP 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Address 128.0.0.1 ~ 191.255.255.254
Define the IP Address for the SIP System

192.0.0.1 ~ 223.255.255.254
Interconnection.
10-23-04 SIP System Interconnection Setup – Dial Maximum of 12 digits (0 ~ 9) No Setting

Number 
Define the Dial Number for the SIP System.
10-23-06 SIP System Interconnection Setup – SIP 1 = Profile 1 1

Profile 2 = Profile 2
4 = Profile 4

5 = Profile 5
6 = Profile 6
6-32 SIP Trunking

Issue 5.0

Program
Input Data Default
Number
1 2 3

User ID
Assign the SIP User ID provided by your SIP Profile (1~2) (V2.00 or lower)
Carrier. In most cases this is your 10 digit main Profile (1~6) (V3.00 or higher)
billing number. This field is also used to send
outbound caller ID information when not

programmed on a per station or per trunk basis.
 For non-registration and SIP Tie Lines to
another system, you MUST have a USER ID
entered.
10-54-01 License Configuration for Each Package – 1 ~ 255 Resource Licenses 0

License Code
Assign VoIP Resource Licenses (5103) to

GCD-CP10 slot (1)

Programming
Assign the 1st and 2nd dial digit to F-Route for
Remote System Extension Numbering. Manual for

default values.
11-02-01 Extension Numbering – Dial (Up to 8 Digits) Maximum of eight digits. Port 1 ~ 300 =
Assign extension numbers to Extension ports. 200 ~ 499
Port 301 ~ 960 =
5000 ~ 5659

14-05-01 Trunk Group – Trunk Group Number Trunk Port 1 ~ 400 = Default = Trunks
Assign SIP Trunks to same Trunk Group. Priority 1 ~ 400 1 ~ 400 assigned
to trunk group 1
with priorities
equal to the 
trunk number.
Trunk 1 = Priority
1 Trunk 400 =
Priority 400.
14-18-03 IP Trunk Data Setup – P2P Mode 0 = Disable 0

Enable/Disable P2P Mode. 1 = Enable
 For Video Call via the System Interconnection

set to 1 (Enable).
14-18-04 IP Trunk Data Setup – Video Mode 0 = Disable 0

Enable/Disable Video Mode. 1 = Enable

set to 1 (Enable).
14-18-05 IP Trunk Data Setup – SIP Profile (SIP Trunk) 1 = Profile 1 Profile 1
Assign each SIP Trunk to either Profile 1 or 2 = Profile 2
Profile 2. 3 = Profile 3
4 = Profile 4

5 = Profile 5
6 = Profile 6

Issue 5.0

Program
Input Data Default
Number
1 2 3
15-05-43 IP Telephone Terminal Basic Data Setup – 0 = Disable 0

Video Mode 1 = Enable
Enable/Disable Video Mode for Standard SIP
terminals.

set to 1 (Enable).
15-05-50 IP Telephone Terminal Basic Data Setup – 0 = Disable 1

Peer to Peer Mode 1 = Enable
On a per station basis enable or disable Peer to 
Peer mode.
 For Video Call to function set to 1 (Enable).
20-08-13 Class of Service Options (Outgoing Call 0 = Off 0
Service) – ISDN CLIP 1 = On
Determine if the ISDN calling line identity 
presentation and screening indicators should
be allowed.
21-17-01 IP Trunk (SIP) Calling Party Number Setup Maximum of 16 digits No Setting
for Trunk (0 ~ 9, , #)
Assign the Caller Party Number for each IP
trunk. The assigned number is sent to the
21-18/ 21-19, the system uses the entry in
Program 21-18/21-19.
21-19-01 IP Trunk (SIP) Calling Party Number Setup Maximum of 16 digits No Setting
for Extension (0 ~ 9, , #)
Assign the Calling Party Number for each
extension. The assigned number is sent to the Profile (1~2) (V2.00 or lower)
central office when the caller places an

21-18/Program 21-19, the system uses the data
in Program 21-18/Program 21-19.
22-02-01 Incoming Call Trunk Setup Trunks 1 ~ 400 0

Assign the incoming trunk type for each trunk. 0 = Normal
 For the SIP System Interconnection, set each 1 = VRS (second dial tone if no
trunk to 5 (E&M Tie Line). VRS installed)
2 = DISA
3 = DID 
4 = DIL
5 = E&M Tie line
6 = Delayed VRS
7 = ANI/DNIS
6-34 SIP Trunking

Issue 5.0

Program
Input Data Default
Number
1 2 3
44-02-01 Dial Analysis Table for ARS/F-Route Access Maximum of eight digits. No Setting
– Dial
Set the Dial digits for the Pre-Transaction Table
for selecting ARS/F-Route (eight digits 
maximum: 1~9, 0 * #, @).
 To enter a wild card/don’t care digit, press Line
Key 1 to enter an @.
44-02-02 Dial Analysis Table for ARS/F-Route Access 0 = No Setting (None) 0

Set the Service Type (0~3) for the Pre- 2 = ARS/F-Route Table
Transaction Table for selecting ARS/F-Route. (F-Route)

 For the SIP System Interconnection, set each 3 = Dial Extension Analyze
Dial Digit to 2 (F-Route). Table (Option)
44-02-03 Dial Analysis Table for ARS/F-Route Access 1 = Delete Digit = 0 ~ 255 0
This is the F-Route Table set in Program 44-05. 2 = 0 ~ 500
(0 = No Setting) 
Table Number = 0 ~ 4
(0 = No Setting)
44-05-01 ARS/F-Route Table – Trunk Group Number 0-100,101-150,255 0

Assign the Trunk Group to be used by the 0 = Not Set
F-Route Table. 1~100 = Trunk Group from
 This is the Trunk Group assigned to the SIP PRG14-05

System Interconnection trunks in Program 14-05-01. 101~150 = Networking
44-05-09 ARS/F-Route Table – Maximum Digit 0 ~ 24 0

Input the maximum number of digits to send 
84-26-01 IPL Basic Setup – IP Address xxx.xxx.xxx.xxx 172.16.0.20

Assign the IP address for each DSP on the 
IPLE.
Operation
SIP Trunk E.164 Support
To make a call using E.164 number format:

2. Dial 00441202223344#.
¸ The system automatically modifies the required header fields of the SIP INVITE
message using the configuration settings in the table below before forwarding to
the ITSP.

Issue 5.0
Table 6-1 SIP INVITE Header Fields
Description
Program Program Program
Calling Party Number = 441509555123
84-14-13 10-02-01 10-02-02
Called Party Number = 00441202223344
0 44 – Request-URI: Invite sip: [email protected] SIP/2.0

To header: To:sip:[email protected]
From header: From<sip:[email protected]>
No Setting – Request-URI: Invite sip: [email protected] SIP/2.0

1 44 – Request-URI: Invite sip:[email protected] SIP/2.0

No Setting – Request-URI: Invite sip:[email protected] SIP/2.0

2 – 00 Request-URI: Invite sip:[email protected] SIP/2.0

To header: To:<sip:[email protected]>
P-Asserted-Identity: [email protected]>
P-Preferred-Identity: [email protected]>
– No Setting No Function
SIP Trunk E.164 CLIP Enhancement
Delete the + only from an incoming SIP INVITE using E.164 numbering scheme:
Table 6-2 Delete + from Incoming SIP INVITE
Program 84-14-16 Program 84-14-13 Description
0: Off When a + is presented as the international access code in a SIP INVITE for
0: Off Or
1: On incoming calls then delete the + only.
<Example Output>
Incoming call from: +4902131795770
Displayed in terminal incoming caller history as:
Original
6-36 SIP Trunking

Issue 5.0
Delete and replace the + and matched country code from an incoming SIP INVITE
using E.164 numbering scheme:
Table 6-3 Delete + and Country Code from Incoming SIP INVITE
With a SIP INVITE for incoming calls. When a + is presented as the international
access code along with a country code that DOES NOT match the value in
Program 10-02-01, then delete the + and add the international access code value
in Program 10-02-02 only.
1: Mode 1 1: On - Or -
access code along with a country code that DOES match the value in Program
10-02-01, then delete the + and country code but DO NOT add the international
access code value.
<Example Output>
Program 10-02-02 = 00
Original Program 10-02-01 = 0 Program 10-02-01 = 49
Delete and replace the + and matched country code from an incoming SIP INVITE
using E.164 numbering scheme:
Table 6-4 Delete + and Country Code from Incoming SIP INVITE
access code along with a country code that DOES NOT match the value in
Program 10-02-01, then delete the + and add the international access code value
in Program 10-02-02 only.
2: Mode 2 1: On - Or -
access code along with a country code that DOES match the value in Program
10-02-01, then delete the + and country code but DO NOT add the international
access code value.

Issue 5.0
<Example Output>
Program 10-02-02 = 00
Program 10-02-03 = 9
Original Program 10-02-01 = 0 Program 10-02-01 = 49
Making an outgoing call from history of incoming calls:

1. From an idle multiline terminal.
2. Press soft key List.
3. Press soft key CID.
4. Press Speaker.
SECTION 6 SIP TRUNK KEEP ALIVE USING OPTION MESSAGE
Description
The SV9100 provides support for “SIP Trunk Keep Alive” using OPTION message for
all six SIP Profiles, applicable to both IP System Interconnection and SIP Carrier
mode.
Conditions
 OPTION Keep Alive works for all SIP Carrier Type (PRG 10-29-14).
 Program 10-29-19 must be enabled for making OPTION Keep Alive to
work for SIP Carrier mode.
 Program 10-23-05 must be enabled for making OPTION keep alive to
work for IP system Interconnection mode.
 SV9100 sends the OPTION message at the interval of the value set in
program 84-14-18.
 OPTION Keep Alive can be sent to the SIP carrier or IP system
Interconnection of Net Link Secondary System.
6-38 SIP Trunking

Issue 5.0
 OPTION Keep Alive Call Restriction and Alarm of Net Link Secondary
System are not supported.
 If SV9100 does not receive the 200-OK response from the SIP server
then SV9100 would retry sending of OPTION message for 32 seconds.
Default Settings
None
System Availability
Terminals
None
Trunks
SIP Trunks
 GPZ-IPLE
 System Version R3 License (0413)

Program
Input Data Default
Number
1 2 3
10-23-05 SIP System Interconnection Setup – Keep 0 = Disable 0

Alive Mode for SIP
Enable/Disable OPTION Keep Alive Mode for
1 = Option Keep Alive Mode 
IP system interconnection.
10-23-06 SIP System Interconnection Setup – SIP 1 = Profile 1 1

Profile 2 = Profile 2
4 = Profile 4

5 = Profile 5
6 = Profile 6

Issue 5.0

Program
Input Data Default
Number
1 2 3
10-29-19 SIP Server Information Setup – Keep Alive 0 = Disable 0

by OPTION Message 1 = Enable
Enable/Disable support of keep alive by Profile (1~2) (Version 2.00 or
OPTION Message for SIP carrier mode. lower)

higher)
84-14-18 SIP Server Basic Information Setup - Keep 60 ~ 3600 (seconds) 180
Alive by OPTION Interval Timer Profile (1~2) (Version 2.00 or
Define Keep Alive by OPTION Interval Timer. lower) 
SV9100 sends the OPTION message at the Profile (1~6) (Version 3.00 or
interval of the value set in this program. higher)
84-14-19 SIP Server Basic Information Setup - Keep 1~5 1

Alive by OPTION Fail Limit Profile (1~2) (Version 2.00 or
Define Keep Alive by OPTION fail limit. lower) 
higher)
84-14-20 SIP Server Basic Information Setup - Option Max. 32 String ping
Keep Alive User ID Profile (1~2) (Version 2.00 or
Define Keep Alive by OPTION User ID which is lower) 
set in SIP URL of OPTION message. Profile (1~6) (Version 3.00 or
higher)
Operation
None
6-40 SIP Trunking

Table 6-5 SIP Packet Sequence
UA (User Agent) SIP Server

Register
401
Register + Auth If authentication required.
200 OK
INVITE
401 or 407
ACK
If authentication required.
INVITE + Auth
100 Trying
1,180 Ringing 180 Ringing
-> Local RBT
2,180 Ringing + SDP
-> Far side in-band RBT PRACK
3,183 Session Progress + SDP 200 OK If 100rel option supported.
-> Far side in-band RBT
Answer
200 OK
ACK
RTP
BYE
200 OK
Issue 5.0
SECTION 7 TLS SUPPORT ON SIP
Description
With Version 5.00 or higher, TLS protocol is supported on SIP trunks. The SIP
message encryption acts only between “SV9100 and a SIP server” or “SV9100 and
SV9100 (SIP Interconnection)”.
Figure 6-17 TLS Support on SIP
Figure 6-18 SIP Server Authentication

Issue 5.0
Conditions
 NAT Traversal is not supported.
 The packets between “SV9100 and SIP terminals” are not encrypted.
 TLS does not work with carrier type D, J and M.
TLS with SIP Server

 Trusted connection is possible by authentication of SIP server. For
authentication, it is necessary to register the root certificate of a SIP
server to SV9100.
 Even if no root certificate is registered to SV9100, creating a TLS
connection is possible but it would not be trusted.
 Only the certificate consisting of "PEM format" is supported. The
extension of certificate file is “.pem” or “.PEM”.
 The file’s name can consists of a maximum of 32 characters
including extension.
 The SV9100 system retains certificate files even after system
restart.
 The certificates can be checked, added and deleted by WebPro
under “Certificate Registration”.
 Only MF level or IN level User can use “Certificate Registration” on
WebPro.
 “Certificate Registration” is not displayed when SA level or UA level
user logs in.
 If the user set something in program 84-14-22, the SIP Server is
authenticated and if it fails, the TLS connection would not be
possible.
 If the user set nothing in program 84-14-22, the authentication is
skipped, and SV9100 tries to create an untrusted connection.
 In the below cases, the trunk cannot be used :
• If the input certificate is invalid (ex. an error in the term of
validity)
• If the input certificate doesn’t exist.
 If the certificate set in program 84-14-22 has expired or been
deleted, the new TLS connection cannot be created and the trunk
cannot be used but it doesn't affect already established
communication.
6-43 SIP Trunking

Issue 5.0
TLS with SIP Interconnection

 To compose SIP interconnection, there are following cases:
• No authentication
The setting of certificate is not necessary. If no CA certificate is
set, it is connected unconditionally without authentication. The
encryption is also done in this case.
• Authenticate as SV9100 interconnection Group
All of systems have a same server certificate & key and the CA
certificate.
• Authenticate each system
The systems have respective server certificates & keys. The
other systems have the corresponded CA certificate.
 Both SV9100 server certificate and private key are used when
SV9100 receive a request of TLS connection.
 When server certificate is not registered, SV9100 uses self-signed
certificate.
 Only the certificate consisting of "PEM format" is supported. The
extension of certificate file is “.pem” or “.PEM”.
 The extension of private key file is “.key” or “.KEY”.
 The file’s name can consists of a maximum of 32 characters
including extension.
 The SV9100 system retains certificate files even after system
restart.
 The certificates can be checked, added and deleted by WebPro
under “Certificate Registration”.
 Only MF level or IN level User can use “Certificate Registration” on
Webpro. “Certificate Registration” is not displayed when SA level or
UA level user logs in.
 Certificate file name is set in program 10-72-01 and private key file
name is set in program 10-72-02, if the same certificate doesn’t
exist in SV9100 system, the trunk cannot be used.
 When the server certificate is set at program 10-72-01, the Private
Key must be set at program 10-72-02. If the Private Key is invalid
or doesn’t exist, the trunk cannot be used.

Issue 5.0
SRTP Support
 To encrypt voice packets, it is necessary to enable SRTP Mode
(PRG84-27-03 “enable”).
 If TLS protocol is not used, an encryption of voice packets does not
work, even if you set SRTP Mode “enable”.
 In the case of SIP interconnection, if P2P is off between two
standard SIP terminals while they are communicating, the voice
packets are encrypted if SRTP is enabled. (P2P settings are
PRG14-18-03: 0/ PRG15-05-50: 0).
 In case of SIP interconnection, if two standard SIP Terminals have
a P2P external call, the voice packets aren’t encrypted even though
SRTP is enabled because they don’t go through SV9100. (P2P
settings are PRG14-18-03: 1/ PRG15-05-50: 1).
 Display Alarm on Telephone (PRG90-50-01) are as follows:
Error Cause Recover Display (28 char)
License is Encryption When the

Insufficient license is not license check is
installed. passed. (When
the TLS
connection is
re-established
or a TLS SIP
call is created.)
Error in CA Failed to When TLS

Certificate validate a CA connection is
(PRG 84- Certificate. re-executed,
14-22) after setting
right file at
PRG 84-14-22
(Profile: XX).
6-45 SIP Trunking

Issue 5.0
Error Cause Recover Display (28 char)

Error in Failed to When SV9100
Server validate a received Offer
Certificate Server of TLS
(PRG 10- Certificate. connection,
72-01) after setting
right file at
PRG 10-72-01.

Private Key validate a received Offer
(PRG 10- Private Key. of TLS
72-02) connection,
after setting
right file at
PRG 10-72-02.

Server authenticate can create TLS
Authenticati Server. connection.
on

Issue 5.0
Authentication Result
Code Error Reason
01 Certificate is not yet The certificate of PRG 84-14-22 is not yet

valid. valid or is expired. or the certificate received
from SIP server is not yet valid or is expired.
02 The certificate has
expired.
03 Self-signed The certificate received from the SIP server
certificate. is a Self-Signed certificate and same
certificate is not set at PRG 84-14-22.
04 Can’t prove server. Can’t prove the server with using a
certificate of PRG 84-14-22.
Default Settings
None
System Availability
Terminals
None
Required Components
 Encryption License (0030)
Related Hardware
 GPZ-IPLE
6-47 SIP Trunking

Issue 5.0
feature.
Level
Program
Number
1 2 3
10-28-03 SIP System Information Setup –Transport 0 = UDP 0

Protocol 1 = TCP 
Define the Transport Protocol type. 2 = TLS
10-72-01 Network Security Setup – Server Certificate Up to 32 characters No setting

Assign the Sever Certificate file’s name for 
SV9100.
10-72-02 Network Security Setup – Private Key Up to 32 characters No setting

Assign the Private Key file’s name for SV9100.

84-14-21 SIP Trunk Basic Information Setup – SIP 0 ~ 65535 Default is as Follows:
Trunk TLS Port Number 5061 = SIP Profile 1
Assign the SIP Trunk TLS Port Number. 5063 = SIP Profile 2
5091 = SIP Profile 3 
5093 = SIP Profile 4
84-14-22 SIP Trunk Basic Information Setup –TLS Up to 32 characters No setting

Certificate 
Assign the Certificate file’s name for TLS.
84-27-03 IPL Basic Setup-SRTP Mode 0 = Disable 0

Enable/disable SRTP mode to enable/disable 1 = Enable 
voice packet encryption.
90-10-01 System Alarm Setup-Alarm Type 0 = Not Set 0

(Alarm 74) Set system alarm for TLS SIP error information. 1 = Major 
2 = Minor

Issue 5.0
Certificate Registration from WebPro

 Login to WebPro by MF level or IN level.
 Click on “Certificate Registration” under Administration.
Figure 6-19 Certificate Registration
 Select the certificate file, then click the Upload button.
Figure 6-19 Uploading Certificate Registration
6-49 SIP Trunking

Issue 5.0
 If no file is selected, or an invalid extension is selected, then the following error

displays:
Figure 6-20 Certificate Registration Error Display
 A maximum of ten files can be registered and if registered files are 10 or more, the
"Upload" button becomes deactivated. So, now user can not register a new file.
 There is a list of the Certificate and Private Key files registered with SV9100 on the
Certificate Registration page. “Not Before” and “Not after” columns of Private Key
file are displayed as blank.
 The page is reloaded if the certificate has been uploaded and the certificate file's
name is displayed in the list.
 Certificate can be deleted by using the “Delete” button as follows:
Figure 6-21 Deleting a Registered Certificate

Issue 5.0
SECTION 9 MULTI-GATEWAY ADDRESS SUPPORT
Description
With Version 5.00 or higher, SV9100 supports Multi-Gateway address support. With
this enhancement, the Default Gateway Address can be set for every SIP Profile. SIP
signaling and media traversal would follow the configuration settings of respective
SIP Profile.
Figure 6-22 Multi-Gateway Address Support
Conditions
 When Default Gateway in program 10-29-22 for every SIP Profile is not
set, then the Default Gateway set in program 10-12-03 is used.
 When UPnP is ON, the WAN IP address of the router is automatically
set in program 10-12-07.
 UPnP and Multi-Gateway can be used at the same time.
 It is mandatory for IPLE to know the MAC address of the default
gateway for the working of Multi Gateway.
 If UPnP and Multi gateway support is used simultaneously, the MAC
Address is automatically updated in program 10-29-23 and 10-12-21 as
per router set in program 10-29-22 and 10-12-03 respectively. Program
10-29-23 and 10-12-21 are in read only mode.
6-51 SIP Trunking

Issue 5.0
 When Default Gateway is not set for every SIP Profile in program 10-29-
22, the function of Multi Gateway doesn't work. In this case, program
10-12-03 is used and MAC address of router (program 10-12-03) is not
updated in program 10-12-21.
 If SIP trunk is configured with the NetLink secondary system, only SIP
Profile 1 can be used with program 10-12-03 and 10-12-07.
 Multi-Gateway is supported with TLS.
Default Settings
None
System Availability
Terminals
None
Required Components
 GPZ-IPLE
 VoIP Resourse License (5103)
Related Feature
None

Issue 5.0
feature.
Level
Program
Number
1 2 3
10-12-03 GCD-CP10 Network Setup – Default 0.0.0.0 ~ 0.0.0.0

Gateway 126.255.255.254
Assign the default gateway IP address for the 128.0.0.1 ~
GCD-CP10. 191.255.255.254

192.0.0.1 ~
223.255.255.254
10-12-07 GCD-CP10 Network Setup – NAPT Router IP 0.0.0.0 ~ 0.0.0.0

Address (Default Gateway [WAN]) 126.255.255.254
Assign the WAN address of the router that the 128.0.0.1 ~
CCPU is using for NAT. 191.255.255.254

192.0.0.1 ~
223.255.255.254
10-12-21 GCD-CP10 Network Setup – MAC Address 00-00-00-00-00-00 ~ 00-00-00-00-00-00

MAC Address of the router set in PRG 10-12-03 FF-FF-FF-FF-FF-FF 
(Read Only)
10-29-14 SIP Server Information Setup – SIP Carrier 0 ~ 26 0

Choice 1 ~ 26 =
Define the SIP Carrier Choice. Carrier Type A ~
- Selecting Carrier B automatically sets Carrier Type Z
Program 10-29-16 to on (1). Program 10-29-16 Profile 1 ~ 2
MUST be set to off for incoming calls to route (Version 2.00 or

using the lowest available trunk port. lower)
- Each certified vendor my use a different Profile 1 ~ 6
carrier type. Visit NTAC website http://
(Version 3.00 or
www.necntac.com) to verify the proper setting
higher)
per vendor.
10-29-21 SIP Server Information Setup – NAT Router 0 = Not Used 0

If the SV9100 is connecting to the SIP Carrier 1 = Used
using NAT Translations, this setting must be Profile 1 ~ 2
enabled. (Version 2.00 or
lower)

Profile 1 ~ 6
(Version 3.00 or
higher)
6-53 SIP Trunking

Issue 5.0
Level
Program
Number
1 2 3
10-29-22 SIP Server Information Setup – Default 0.0.0.0 ~ 0.0.0.0

Gateway 126.255.255.254
Assign the default gateway IP address for the 128.0.0.1 ~
GCD-CP10. 191.255.255.254

192.0.0.1 ~
223.255.255.254
10-29-23 SIP Server Information Setup – MAC 00-00-00-00-00-00 ~ 00-00-00-00-00-00

Address FF-FF-FF-FF-FF-FF
MAC Address of the router set in PRG 10-29-22

(Read Only)
10-29-24 SIP Server Information Setup – NAPT Router 0.0.0.0 ~ 0.0.0.0

IP Address 126.255.255.254
Assign the WAN address of the router that the 128.0.0.1 ~
CCPU is using for NAT. 191.255.255.254

192.0.0.1 ~
223.255.255.254
10-37-01 UPnP Setup – UPnP Mode 0 = Disable 0

Enable/Disable UPnP. 1 - Enable

14-18-05 IP Trunk Data Setup – SIP Profile (SIP Trunk) 1 = Profile 1 Profile 1
Assign each SIP Trunk to Profile. 2 = Profile 2
3 = Profile 3
4 = Profile 4
5 = Profile 5 
6 = Profile 6
With Version 2.00 or
lower the SV9100
only supports two
SIP Profiles.

Issue 5.0
Operation
The following two scenarios depict the Multi-Gateway and Upnp And Multi-
Gateway setup.
1. Multi-Gateway
Figure 6-23 Multi-Gateway Setup
Table 6-6 Multi-Gateway Programming
Program Name Data
10-29-14 SIP Server Information Setup- SIP - SIP Profile 1

Carrier Choice SIP Carrier Choice = 1 (Carrier
A)
- SIP Profile 2
SIP Carrier Choice = 2 (Carrier
B)
- SIP Profile 3
C)
10-29-21 SIP Server Information Setup - NAT Used

Router
6-55 SIP Trunking

Issue 5.0
Table 6-6 Multi-Gateway Programming (Continued)
Program Name Data
10-29-22 SIP Server Information Setup - - SIP Profile 1

Default Gateway 172.16.10.10
- SIP Profile 2
172.16.20.10
- SIP Profile 3
172.16.30.10

NAPT Router IP Address 10.1.1.10
- SIP Profile 2
10.1.2.10
- SIP Profile 3
10.1.3.10
14-18-05 IP Trunk Data Setup - SIP Profile -Trunk Port: 1-8

SIP Profile = 1 (Profile 1)
-Trunk Port: 9-16
-Trunk Port: 17-24
2. UPnP and Multi-Gateway
Figure 6-24 UPnP and Multi-Gateway Setup

Issue 5.0
Table 6-7 UPnP and Multi-Gateway Programming
Program Name Data
10-12-03 Default Gateway 172.16.50.10
10-29-14 SIP Server Information Setup- SIP - SIP Profile 1

Carrier Choice SIP Carrier Choice = 1 (Carrier
A)
- SIP Profile 2
B)
- SIP Profile 3
C)
10-29-21 SIP Server Information Setup - NAT Used

Router

- SIP Profile 2
172.16.20.10
- SIP Profile 3
0.0.0.0

NAPT Router IP Address 10.1.1.10
- SIP Profile 2
10.1.2.10
- SIP Profile 3
0.0.0.0
10-37-01 UPnP Mode 1 = Enable
14-18-05 IP Trunk Data Setup - SIP Profile -Trunk Port: 1-8

-Trunk Port: 9-16
-Trunk Port: 17-24
6-57 SIP Trunking

Issue 5.0

UNIVERGE® SV9100
H.323 Trunking Chapter 7
H.323 is an International Telecommunication Union (ITU) standard for Packet Based

Multimedia Communication Systems. The UNIVERGE SV9100 can use H.323 to
connect to another UNIVERGE SV9100 system or a third party product.
The feature set is limited. When using H.323, it is not possible to use the advanced
networking features. If these features are required, use IP K-CCIS.
The UNIVERGE SV9100 Voice over IP Trunk Daughter Board H.323 package sends
the real-time voice over the corporate LAN or WAN. The voice from the telephone is
digitized and then put into frames to be sent over a network using Internet Protocol.
The UNIVERGE SV9100 Voice over IP Trunk – H.323 daughter board package
allows communication using standard H.323 (Normal and Fast Start) Protocol and
allows connectivity to any H.323 standards compliant voice gateway and gatekeeper.
This VoIP Trunk daughter board also allows Registration and Authentication Server
(RAS) support to register with an RAS Server and use Gatekeeper for dynamic call
routing.
The GPZ-IPLE daughter board – H.323 is an optional interface that can provide IP
trunks and Tie Lines. It can operate in the following modes:
 COI
 DID
 TLI
Depending on the requirements and resource allocation in the LAN/WAN/Internet,

the GPZ-IPLE – H.323 can be configured to use any of the following voice
compressions:
 G.729 Low bandwidth requirement used on most Wide Area Network links.
 G.711 High bandwidth requirement usually used on Local Area Networks.
 G.722 This CODEC is useful in fixed network, Voice over IP applications, where the
required bandwidth is typically not prohibitive.
 G.723 This CODEC is a ITU-T standard wideband speech CODEC. This is an
extension of Recommendation G.721 adaptive differential pulse code modulation to
24 and 40Kbps for digital circuit multiplication equipment application.

Issue 5.0
SECTION 2 H.323 TRUNK IMPLEMENTATION
The UNIVERGE SV9100 implementation and programming for SIP and H.323 are
very similar. The call routing, call features and speech handling (RTP) are the same –
only the signaling protocol is different. For this reason, the description and examples
below can be used for either implementation (the differences are clearly defined). IP
Trunk is used to describe SIP or H.323 trunks.
The information below relates to basic networking, without the use of external H.323
Gatekeeper. The following steps are required to configure IP Trunks:
2.1 Determine Numbering Plan
When planning the IP Trunk implementation, it is important to determine the

numbering plan for the sites. A properly planned VoIP network allows flexibility
and simplifies additions or changes to the network.
There are two approaches to the numbering scheme;

 Open Numbering
In this case the sites are identified by dialing codes that are appended to
the dialed digits. This allows the extension numbers to be the same at
multiple sites.
Example:
System A System B
Extension Number 200-299 200-299

Site Code 1 2
Extension 200 at System A can dial Extension 200 at System B by dialing

the extension 2200.
This numbering scheme is ideal when there are a large number of sites in
the VoIP network or when many extension numbers are in use at each site.
 Closed Numbering
In this case, the first digit of the extension number is used for routing calls.
This alleviates the need for dialing a site code and makes a more unified
numbering plan. This is feasible only if there is a relatively small number of
sites or the sites have very few extensions.
7-2 H.323 Trunking

Issue 5.0
2.2 Configure VoIPDB Trunk Ports
After installing IPLE (VOIP Daughter Board) in UNIVERGE SV9100 System,

VOIPDB trunk ports need to be defined in 10-68-01.
By default the VOIPDB trunk ports are defined as None. When using H.323, you
need to change the trunk ports to H.323 in 10-68-01.
2.2.1 H.323 Setup
Use the following programs to assign the information for H.323.
Table 7-1 H.323 Setup Assignments
Item No. Selection
IP Trunk Availability – Trunk 0: None Assign the H.323 Trunk Availability

Type 1: SIP
10-68-01
2: H.323
3: CCIS
IP Trunk Availability – Start Range: 0 ~ 400 Assign the start port number of H.323
10-68-02 Port Default is 0 Trunks
IP Trunk Availability – Range: 0 ~ 400 Assign the Number of H.323 Trunks

10-68-03 Number of Ports Default is 0
Trunk Group – Trunk Group 0~100 Default priorities for trunks 1~400 is
Number Priority = 1 ~ 400 1~400.
14-05-01
Assign Trunks to Trunk Groups/
Default is Group 1 Outbound Priority
Incoming Call Trunk Setup 0 = Normal Set the feature type for the trunk you
1 = VRS (second dial tone if no are programming.
VRS installed)
2 = DISA (Second dial tone for option 1 if no
3 = DID VRS is installed)
4 = DIL
22-02-01
5 = E&M Tie Line
6 = Delayed VRS
7 = ANI/DNIS
8 =DID(DDI) Mode Switching
Default is 0

Issue 5.0
2.2.2 Configure VoIPDB Networking Information
Use the following programs to assign VoIPDB network configuration

information.
Table 7-2 VoIPDB Network Configuration Assignments
Item No. Selection
IP Trunk Availability – Trunk 0: None Assign the H.323 Trunk Availability

Type 1: SIP
10-68-01
2: H.323
3: CCIS
IP Trunk Availability – Start Range: 0 ~ 400 Assign the start port number of H.323
10-68-02 Port Default is 0 Trunks
IP Trunk Availability – Range: 0 ~ 400 Assign the Number of H.323 Trunks

10-68-03 Number of Ports Default is 0
2.2.3 VoIPDB (DSP) Basic Setup
Use the following programs to assign VoIPDB (DSP) basic setup

information.
Table 7-3 VoIPDB (DSP) Basic Assignments
Item No. Selection
IPL Basic Setup – IP Address Default = 172.16.0.20 When installing GPZ-IPLE in

84-26-01 UNIVERGE SV9100 system, only 1
IP address is assigned.
IPL Basic Setup – RTP Port 0~65534 Assign the RTP Port Number.
Number
84-26-02
Default = 10020
IPL Basic Setup – RTCP Port 0~65534 Assign the RTCP Port Number.
Number
84-26-03
(RTCP Port Number + 1)
Default = 10021
7-4 H.323 Trunking

Issue 5.0
2.3 Configure Signaling Information (H.323)
H.323 requires a gatekeeper to be configured to manage registration of H.323

endpoints. The UNIVERGE SV9100 has a built-in H.323 gatekeeper and also
supports connection to an external gatekeeper. In most cases, the SV9100
internal Gatekeeper is sufficient. External gatekeepers are used usually only in
very large H.323 networks.
To configure the UNIVERGE SV9100 to use its internal Gatekeeper, set Program
10-17-01 to Manual, and then set the GCD-CP10 IP address (Program10-12-09)
as the gatekeeper address in Program 10-17-02.
The UNIVERGE SV9100 has to register an Alias address with the gatekeeper.
This is used to route incoming calls to the correct destination.
2.3.1 H.323 GateKeeper Setup
Use the following programs to assign H.323 GateKeeper setup information.
Table 7-4 H.323 GateKeeper Setup Assignments
Item No. Selection
H.323 Gatekeeper Setup – 0 = No Gatekeeper Set IP Address either

Gatekeeper Mode 1 = Automatic automatically or manually if using an
2 = Manual external
Gatekeeper.
If Automatic Gatekeeper is selected,

the system searches for the
gatekeeper. If Manual Gatekeeper is
10-17-01
selected, the IP address defined in
Program 10-17-02 is used.
Default is 0
H.323 Gatekeeper Setup – 0.0.0.0 ~ 126.255.255.254 If Program 10-17-01 is set to 2, enter
Gatekeeper IP Address 128.0.0.1 ~ 191.255.255.254 the IP address of the Gatekeeper.
192.0.0.1 ~ 223.255.255.254
This should match the entry made in
10-17-02 0.0.0.0 Program 10-12-09.
Default is 0.0.0.0

Issue 5.0
Table 7-4 H.323 GateKeeper Setup Assignments (Continued)
Item No. Selection
H.323 Gatekeeper Setup – Maximum 124 characters If Program 10-17-01 is set to 1, enter
Preferred Gatekeeper the Gatekeeper ID. When registering
with an external Gatekeeper using
Gatekeeper search, two or more
GRQs (Gate Keeper Request) may
be assigned. In this case, if this ID is
set up, it registers with a Gatekeeper
using the ID set up in this program
10-17-04
(124 characters max).
Default not assigned
2.3.2 H.323 Alias Address Setup
Use the following programs to assign H.323 alias address setup

information.
Table 7-5 H.323 Alias Address Setup Assignments
Item No. Selection
H.323 Alias Address Setup – Dial up to 12 digits Enter the Alias Address of the
Alias Address (0~9, *, #) UNIVERGE SV9100 system
registered into the external
Gatekeeper.
(12 digits max).
10-18-01
This is the System Code of Local
System.

H.323 Alias Address Setup – 0 = E164 Define the type of Alias Address
Alias Address Type registered to the external Gatekeeper.
At this point, only E.164 can be

10-18-02
entered.
Default = 0
7-6 H.323 Trunking

Issue 5.0
2.4 Configure Call Routing for H.323
Enter the remote destination information in Program 10-23. This allows up to 1000
SIP or H.323 destinations (remote systems) to be entered. If more than 1000
destinations are required, it is necessary to use an external H.323 Gatekeeper or
SIP Server.
It may be advantageous to configure F-route to route calls to remote destinations

via the IP trunks (this can simplify dialing for the users). F-route configuration is
discussed fully in the UNIVERGE SV9100 Features and Specifications Manual.
2.4.1 H.323 Call Routing
Use the following programs to assign H.323 call routing.
Table 7-6 H.323 Call Routing Assignments
Item No. Selection
SIP System Interconnection 0 = Disable (No) Enable/Disable the remote

Setup – System Interconnect 1 = Enable (Yes) destination.
10-23-01
Default is 0
SIP System Interconnection 0.0.0.0 ~ 126.255.255.254 Enter the IP address for the remote
Setup – IP Address 128.0.0.1 ~ 191.255.255.254 system.
10-23-02 192.0.0.1 ~ 223.255.255.254
Default = 0.0.0.0
SIP System Interconnection Up to 12 digits (0~9) Enter the alias number for the remote
Setup – Dial Number system.
This is the E.164 number assigned

10-23-04 to the remote destinations.

10-23-05 Keep Alive Mode for SIP 0 = Disable
1 = Enable
(default = 0)
10-23-06 SIP Profile 1 = Profile 1 Only used when Trunk Type is
2 = Profile 2 defined as (SIP) in PRG 10-68-01.
3 = Profile 3
4 = Profile 4
5 = Profile 5
6 = Profile 6
(default = 1)

Issue 5.0
2.4.2 Caller ID (H.323)
Use the following programs to assign H.323 caller ID.
Table 7-7 H.323 Caller ID Assignments
Item No. Selection
IP Trunk (SIP) Calling Party Up to 16 digits Enter the CPN to be used for calls
Number Setup for Trunk (1~0, *, #) made via the selected trunk.
21-17-01

IP Trunk (H.323) Calling Party Up to 16 digits Enter the CPN to be used for calls
Number Setup for Extension (1~0, *, #) made via the selected H.323
Extension.
This Program overrides the

21-18-01
programming in Program 21-17

Class of Service Options 0 = Off Determine if the Calling Line Identity
(Outgoing Call Service) – 1 = On presentation and screening indicators
ISDN CLIP are allowed.
20-08-13
Default is 1
Class of Service Options 0 = Off Turn Off or On the Caller ID display at
(Incoming Call Service) – 1 = On an extension.
Caller ID Display
20-09-02
Default is 1
7-8 H.323 Trunking

Issue 5.0
The CLIP that is actually sent is based on the following priority:
2.5 CODEC Selection for IP Networking
IP Networking can use various CODECs. A CODEC is a standard for converting

an analog signal to digital. This conversion process is handled by the DSP (Digital
Signal Processors) on VoIP Daughter Board.
Each CODEC has different voice quality and compression properties. The correct
choice of CODEC is based on the bandwidth available, the number of calls
required, and the voice quality required.

Issue 5.0
2.5.1 CODEC Selection for IP Networking (H.323)
Use the following programs to assign CODEC selections for H.323.
Table 7-8 IP Networking (H.323) CODEC Assignments
Item No. Selection
84-01-02 H.323 Trunk Basic 1~4 (SV9100)
Information Setup – Number
of G.711 audio frames Default =3
84-01-03 H.323 Trunk Basic 0 = Disable
Information Setup – G.711 1 = Enable
VAD mode
Default is 0
84-01-04 H.323 Trunk Basic 0 = A-law
Information Setup – G.711 1 = -law
Type
Default is 0
84-01-05 H.323 Trunk Basic 1~6
Information Setup – Number 1 = 10ms
of G.729 audio frames 2 = 20ms
3 = 30ms
4 = 40ms
5 = 50ms
6 = 60ms
Default is 3
84-01-06 H.323 Trunk Basic 0 = Disable
Information Setup – G.729 1 = Enable
VAD mode
Default = 0
84-01-07 H.323 Trunk Basic 0~300ms
Information Setup – G.729
Jitter Buffer (min) Default is 30
84-01-08 H.323 Trunk Basic 0~300ms
Jitter Buffer (average) Default is 60
84-01-09 H.323 Trunk Basic 0~300ms
Jitter Buffer (max) Default is 120
84-01-15 H.323 Trunk Basic 1 = Fixed
Information Setup – Jitter 3 = Self adjusting
Buffer Mode
Default is 3
84-01-16 H.323 Trunk Basic 0~300ms
Jitter Buffer(min) Default is 30
7-10 H.323 Trunking

Issue 5.0
Table 7-8 IP Networking (H.323) CODEC Assignments (Continued)
Item No. Selection
84-01-17 H.323 Trunk Basic 0~300ms
84-01-18 H.323 Trunk Basic 0~300ms
84-01-22 Voice Activity Detection 0~30 (-19dB~ +10dB and self
Threshold adjustment)
0 = Self adjustment
1 = -19dB (-49dBm)
:
20 = 0dB (-30dBm)
:
29 = 9dB (-21dBm)
30 = 10dB (-20dBm)
Default is 20
84-01-33 H.323 Trunk Basic 0~3 Priority of voice encoding method.
Information Setup – Priority 0 = G.711
CODEC setting 2 = G.729
Priority of voice encoding 3 = G.722
method.
Default is 0
84-01-34
---Not Used---
84-01-35
---Not Used---
84-01-63 H.323 Trunk Basic 1~4

Information Setup –Number 1 = 10 ms
of G.722 audio frames 2 = 20 ms
3 = 30 ms
4 = 40 ms
Default is 3
84-01-65 H.323 Trunk Basic 0~300ms
Jitter Buffer (min) Default is 30
84-01-66 H.323 Trunk Basic 0~300ms

Issue 5.0
Table 7-8 IP Networking (H.323) CODEC Assignments (Continued)
Item No. Selection
84-01-67 H.323 Trunk Basic 0~300ms
84-01-68 RTP Filter 0 = Disable
1 = Enable
2 = Enable (SSRC)
Default is 1
2.6 H.323 TIE Line Programming Example
Figure 7-1 H.323 TIE Line Programming Example illustrates how to program a
TIE line for H.323.
Figure 7-1 H.323 TIE Line Programming Example
The following programs are assigned in this example.
Program Main Site (Office A) Remote Site (Office B)
PRG 10-12-09 * 192.168.1.10 192.168.2.40

PRG 10-12-10 * 255.255.255.0 255.255.255.0
PRG 11-01-01 Digit 1x 3 Digits, Extension Program 11-01-01 Digit 1x
3 Digits, F-Route
PRG 11-01-01 Digit 2x 3 Digits, F-Route Program 11-01-01 Digit 2x
3 Digits, Extension
PRG 11-02-01 100, 101, etc. 200, 201, etc.
PRG 14-05-01 Set all IP Trunks to Trunk Group 10
7-12 H.323 Trunking

Issue 5.0
Program Main Site (Office A) Remote Site (Office B)
PRG 22-02-01 Set all IP Trunks to TIE

PRG 44-02-01 (Table 1) 4 1
PRG 44-02-02 (Table 1) F-Route F-Route
PRG 44-02-03 (Table 1) 1 1
PRG 44-05-01 (F-Route Trunk Group 10 Trunk Group 10
1)
PRG 44-05-09 (Max 3 3
Digits)
PRG 84-26-01 (DSP) 192.168.1.11 192.168.2.11
H.323 Settings
PRG 10-68-01 Enable H.323 Trunk Enable H.323 Trunk Availability

Availability
PRG 10-68-02 Define the Start Port Define the Start Port Number
Number
PRG 10-68-03 Assign the number of Assign the number of Ports (4
Ports (4 Ports) Ports)
PRG 10-17-01 Manual Manual
PRG 10-17-02 192.168.1.10 192.168.2.10
PRG 10-18-01 50 51
PRG 10-23-01 Enable System Interconnection (Yes)
PRG 10-23-02 Assign the IP Address of Assign the IP Address of the
the Distant System Distant System (192.168.1.10)
(192.168.2.10)
PRG 10-23-04 Starting digit of the Distant Starting digit of the Distant
System Extension Number System Extension Number
* GCD-CP10 reset is required after making changes to this Programs.

Issue 5.0
7-14 H.323 Trunking

UNIVERGE® SV9100
IP Multiline Station (SIP) Chapter 8
SECTION 1 INTRODUCTION1
The UNIVERGE SV9100 system supports IP extensions using a variety of multiline

terminals. These telephones have the same look and functionality as typical multiline
telephones, but they are connected to the CCPU via IP rather than by a hardwired
connection to an ESI port.
The following DT800/DT700 IP Multiline Telephones (ITL/ITZ) support IP extensions.

 ITL-2E-1 (BK) TEL
 ITL-6DE-1 (BK) TEL
 ITL-8LD-1 (BK) TEL / ITL-8LD-1 (WH) TEL
 ITL-12D-1 (BK) TEL / ITL-12D-1 (WH) TEL
 ITL-12PA-1 (BK) TEL
 ITL-320C-1 (BK) TEL
 ITL-12CG-3 (BK) TEL
 ITL-12DG-3 (BK) TEL
 ITZ-12D-3 (BK) TEL/ ITZ-12D-3 (WH) TEL
 ITZ-24D-3 (BK) TEL/ ITZ-24D-3 (WH) TEL
 ITZ-12CG-3 (BK) TEL/ ITZ-12CG-3 (WH) TEL
 ITZ-24CG-3 (BK) TEL
 ITZ-8LD-3 (BK) TEL/ ITZ-8LD-3 (WH) TEL
 ITZ-8LDG-3 (BK) TEL/ ITZ-8LDG-3 (WH) TEL
 ITZ-12DG-3 (BK) TEL/ ITZ-12DG-3 (WH) TEL
1. The voice quality of VoIP is dependent on variables such as available bandwidth, network latency and Quality of Ser-
vice (QoS) initiatives, all of which are controlled by the network and Internet service providers. Because these vari-
ables are not in NEC control, it cannot guarantee the performance of the user’s IP-based remote voice solution.
Therefore, NEC recommends connecting VoIP equipment through a local area network using a Private IP address.

Issue 5.0
SECTION 2 IP TO TDM CONVERSION
When an IP telephone calls a DT300/DT400 multiline telephone, single line

telephone or trunk, the speech must be converted from IP to TDM (Time Division
Multiplexing) technology. The GPZ-IPLE daughter board provides this function.
Each GPZ-IPLE has a number of DSP resources; each can convert a speech
channel from IP to TDM and vice versa.
It is possible for DT800/DT700 IP Phones to talk directly to other DT800/DT700 IP

Phones without using a GPZ-IPLE resource. For more information, refer to Section 6
Peer-to-Peer on page 8-7.
2.1 DT800/DT700 IP Multiline Telephones (ITL/ITZ)
The IP multiline telephone operates in the same way as a DT400 (DTZ)/DT300

(DTL) digital multiline telephone. It has all features and flexibility you expect from
a DT400/DT300 digital multiline telephone. The difference is that the IP telephone
has an RJ-45 for connection to an IP network, rather than an RJ-11 for connection
to a GCD-8DLCA/GCD-16DLCA.
Figure 8-1 DT800/DT700 IP Telephone (ITL)
8-2 IP Multiline Station (SIP)

Issue 5.0
2.2 Conditions
When using DT800/DT700 IP phones, it is not recommended to assign the

following features to a large number of phones (100 or more):
 The same Trunk Line assignment (squared key system)
 The same Virtual Extension assignment
 Paging key with LED ON assignment
 The same location Park key
 The same location CAP key
 The same BLF key assignment
 Day Night Mode Change key assignment
 The same VM Mail Box key assignment
 Trunk Group key
 Trunk Group All Line Busy Indication
SECTION 3 POWER FAIL ADAPTER [PSA-L ( ) UNIT]
The power fail adapter is an add-on module for the IP (DT800/DT700) multiline
telephones and digital (DT400/DT300) multiline telephones. It allows connection to
an analog trunk if the power or system connection fails, or the IP telephone loses
connection to the UNIVERGE SV9100 system.
No programming is required on the UNIVERGE SV9100 to support this adapter.
3.1 Connecting to an IP Telephone
The Power Fail Adapter connects to an analog PSTN (Public Switched Telephone
Network) line. At a small branch office, for example, this may be the same line that
is used for faxes/modems/etc. The handset is also connected to the Power Fail
Adapter. It is necessary to unplug it from the IP telephone and reconnect to the
adapter. This allows the speech path to be redirected to the handset during a
power/network failure.

Issue 5.0
Figure 8-2 Power Fail Adapter Connection
GCD-CP10 PSTN
Public Switched
Telephone Network
Switch
VoIP
k
run
gT
a lo
UNIVERGE SV9100
An
IP Telephone IP Telephone
(with Power Fail Adapter)

Issue 5.0
Figure 8-3 IP Telephone Connection
PSTN DTMF/DP
3.2 Operation During Power Failure
If the telephone becomes disconnected from the power supply (e.g., power loss)
the telephone display is blank.
To make a call, lift the handset to receive dial tone from the analog line. Dial as
normal.
If a call is received on the analog line, the Power Fail Adapter rings. Lift the
handset to answer.
If the telephone is connected to the power supply, but disconnected from the
UNIVERGE SV9100 system (e.g., data network failure), the IP telephone
attempts to reconnect. If this fails, press the button on the top of the adapter.
Refer to Figure 8-2 Power Fail Adapter Connection on page 8-4. This puts the IP
telephone in analog mode. The telephone display shows LINE -> PSTN.
To make a call, lift the handset to receive dial tone from the analog line. Dial as
normal.

Issue 5.0
If a call is received on the analog line, the Power Fail Adapter rings. Lift the
handset to answer.
 Handsfree (Speaker) mode is not supported on calls made to or from the Power Fail
Adapter. The handset must always be used.
 A special dial pad option is supplied with the PSA-L ( ) UNIT.
SECTION 4 LAN CONNECTION
As illustrated in Figure 8-4 Typical Network IP Connection, the IP telephone has two
RJ-45 connections on the back side marked PC and LAN. This allows the IP
telephone and a PC to share one cable run and switch port.
Figure 8-4 Typical Network IP Connection
GCD-CP10
GPZ-IPLE
If installing an IP telephone at a location that already has a PC connected to the data

network, it is possible to use either of the following methods:
 Use a different cable and complete the following steps:
 Leave the PC connected to the LAN.
 Patch a switch port to a new cable run.
 Connect a CAT 5 straight-through cable from the wall outlet to the LAN port on the
IP telephone.
 Share the existing cable and complete the following steps:

 Unplug the cable from the PC network card (NIC).
 Connect that cable to the LAN port on the IP telephone.
 Connect a new straight-through patch lead from the PC NIC to the PC port on the IP
telephone.

Issue 5.0
SECTION 5 PROVIDING POWER
IP telephones require power to function. This can be provided in various ways.
5.1 Local Power
The IP telephone has a connector for external power. This is supplied by an AC

adapter that outputs 27V DC. This means that a power outlet is required in the
vicinity of each IP Phone, and loss of power in the building prevents the
telephones from working.
 Use only the NEC supplied power supply.
5.2 802.3af Power Over Ethernet (PoE)
A 802.3af PoE switch is a data switch that also provides power over the spare
pairs. The switch can be used with any device (not just IP phones) and detects
whether or not power is required. As all phones receive their power from one
device, it is easy to protect the IP phones from loss of power (by connecting the
PoE switch to a UPS).
SECTION 6 PEER-TO-PEER
An IP telephone can send and receive RTP packets to or from another IP telephone
without using DSP resources on a GPZ-IPLE. This operation allows only Intercom
calls between the IP telephones.
If a DT800/DT700 IP multiline telephone or trunk line is required, a DSP resource is

needed and a GPZ-IPLE must be installed. If a conference call is initiated while on a
peer-to-peer call, the peer-to-peer connection is released and a new non peer-to-
peer connection is created using the GPZ-IPLE. If the third party drops out of the
conversation, the call reverts to a peer-to-peer call. There is silence while this
conversion is made by the system.
Although the peer-to-peer feature is supported for IP Station-to-IP Station calls, the
UNIVERGE SV9100 chassis must still have a registered GPZ-IPLE installed in the
system.
With Barge-In, a short silence may occur if the following occurs:

 Peer-To-Peer call receives a Barge-In without a Barge-In tone.
 Peer-To-Peer call receives a Barge-In with Monitor mode.
 Established Barge-In is disconnected.
 The Peer-to-Peer feature is a programmable feature that may be enabled or
disabled by accessing Data Program 15-05-50 - Peer-to-Peer Mode.

Issue 5.0
SECTION 7 MISCELLANEOUS
7.1 System Tones and Ring Tones
IP Phones do not use Program 80-01: Service Tone Setup entries. The tones are
generated locally by the IP telephone. When a Door Box chime rings an IP
telephone, the system activates the chimes using a ring command. Because of
this, if the volume is adjusted while the door chime is sounding, the ringing volume
of the IP Phone is adjusted.
7.2 Music on Hold
In addition, Music on Hold is provided by the IP telephone. The settings in

Program 10-04: Music on Hold Setup are ignored except to determine whether or
not music is to be provided. If 10-04-02 is set to 0, no Music on Hold is heard. If
10-04-02 is set to 1 or 2, music is provided by the IP telephone.
SECTION 8 CONFIGURATION EXAMPLES
8.1 IP Addressing
When using a GPZ-IPLE, only 1 IP Address is needed for all DSP's. It should be
the same subnet of the IP defined in 10-12-09.
The following chart shows the minimum and maximum number of IP addresses
used with different GPZ-IPLE card configurations.
Minimum IP Maximum IP
Card Notes
Addresses Addresses
The number of DSP channels

GPZ-IPLE 1 2 depends on the VOIP license
loaded to GCD-CP10 up to 256.

Issue 5.0
When assigning the IP addresses to the GPZ-IPLE card, the addresses must
be in the same network (subnet). If the CPU is also to be connected to the
network, it requires a separate IP address in a different network (subnet).
IMPORTANT
When a IPL() card is installed it is recommended to not use the CPU NIC and
to change PRG 10-12-01 to 0.0.0.0
When an GPZ-IPLE card is attached to the CPU, using the CPU NIC is no
longer required. All connections that previously terminated to the CPU NIC
card can now be terminated to the GPZ-IPLE NIC. E.g. PCPro, Web Pro,
ACD, etc. all terminate to the GPZ-IPLE NIC card when installed.
The GPZ-IPLE and the CPU NIC share the same gateway assignment. The
default gateway command in Program10-12-03 is used by both NICs allowing
only one device, or CPU, to route outside of its own network.
The following examples show typical scenarios and basic programming required.
These examples assume that the programming steps are performed on a default
system (i.e., no existing configuration).
8.2 Example Configuration 1 - Static IP Addressing, One LAN
This example shows an IP Phone connected to a single LAN (no routers), with
static IP Addresses.
Figure 8-5 Example Configuration 1 - Static IP Addressing, One LAN
GCD-CP10
Switch
VoIP
UNIVERGE SV9100
GCD-CP10 192.168.1.20
VoIP DSP: 192.168.1.21
Subnet Mask: 255.255.255.0
Default Gateway: 192.168.1.254
IP Phone 1 IP Phone 2
192.168.1.200 192.168.1.201
Extension: 100 Extension: 101

Issue 5.0
Programming - GCD-CP10:
10-12-01 : GCD-CP10 Network Setup - IP Address 0.0.0.0

(for GCD-CP10)
10-12-02 : GCD-CP10 Network Setup - Subnet M ask 255.255.255.0
10-12-03 : GCD-CP10 Network Setup - Default Gateway 192.168.1.254
10-12-09 : GCD-CP10 Network Setup - IP Address (This 192.168.1.20
assignment is for GCD-CP10 when the GPZ-IPLE daughter board
is installed.)
Programming - GPZ-IPLE :
84-26-01 : IPL Basic Setup - IP Address (Slot # - DSP) 192.168.1.21
Programming - IP Phones:
DHCP Mode Disabled

IP Address 192.168.1.200
Subnet Mask 255.255.255.0
1st Server Address 192.168.1.20
1st Server Port 5080
8.3 Example Configuration 2 - Dynamic IP Addressing, One LAN
This example shows System IP Phones connected to a single LAN (no routers)
with dynamic IP Addresses. The DHCP server could be:
 Customer supplied (e.g., Windows server)
 inDHCP internal DHCP server
In this case, additional programming would be required. Refer to Chapter 3

General IP Configuration.
Figure 8-6 Example Configuration 2 - Dynamic IP Addressing, One LAN
GCD-CP10
Switch
VoIP
UNIVERGE SV9100
GCD-CP10 192.168.1.20
VoIP DSP: 192.168.1.21
Subnet Mask: 255.255.255.0
E t i 200 E t i 201

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10-12-01 : GCD-CP10 Network Setup - IP Address (for GCD-CP10) 0.0.0.0

10-12-10 : GCD-CP10 Network Setup - Subnet Mask 255.255.255.0
10-12-03 : GCD-CP10 Network Setup - Default Gateway 192.168.1.254
10-12-09 : GCD-CP10 Network Setup - IP Address 192.168.1.20
(This assignment is for GCD-CP10 when the GPZ-IPLE daughter board is
installed.) On
10-13-01 : In-DHCP Server Setup - DHCP Server Mode Enable
10-14-01 : Managed Network Setup 192.168.1.200 Min
10-14-02: Managed Network Setup 192.168.1.250 Max
10-16-16: SIP Server Address 192.168.1.20
10-16-27: SIP Server Port 5080
Programming - GPZ-IPLE:
84-26-01 : IPL Basic Setup - IP Address 192.168.1.21
Programming - IP Phones:
DHCP Mode (Ext. 200): Enabled

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8.4 Example Configuration 3 - Static IP Addressing, Routed WAN
This example shows IP Phones connected to an UNIVERGE SV9100 system

over a Wide Area Network (WAN), with static IP addressing. This is a typical
scenario - a small branch office connecting to the main office.
Figure 8-7 Example Configuration 3 - Static IP Addressing, Routed WAN
GCD-CP10
Switch
VoIP
Router VPN
192.168.1.254
GCD-CP10: 192.168.1.20
VoIP DSP: 192.168.1.21
Subnet Mask: 255.255.255.0
WAN
(Leased Line, Frame
Relay, etc.)
Router VPN
Switch
192.168.2.254
192.168.2.200 192.168.2.201

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10-12-01: GCD-CP10 Network Setup - IP Address 0.0.0.0

(for GCD-CP10)
10-12-10: GCD-CP10 Network Setup - Subnet Mask 255.255.255.0
10-12-03: GCD-CP10 Network Setup - Default Gateway 198.168.1.254
10-12-09: GCD-CP10 Network Setup - IP Address 192.168.1.20
(This assignment is for GCD-CP10 when the GPZ-IPLE daughter
board is installed.)
Programming - GPZ-IPLE:
84-26-01: IPL Basic Setup - IP Address (Slot # - DSP) 192.168.1.21
Programming - IP Phones
DHCP Mode: Disabled

IP Address: 192.168.2.200
Subnet Mask: 255.255.255.0
1st Server Address 198.168.1.20
1st Server Port 5080
SECTION 9 IP PHONE PROGRAMMING INTERFACE
This section describes how to access the programming interface for IP phones. To
access the User Menu follow the steps listed below.
1. Using a DT800/DT700 telephone, press the Menu button to enter program

mode. The IP User Menu is displayed.
2. On the IP User Menu, select Config (0) for the IP Phone. Settings are listed in
Table 8-1 IP Phone Programming Options User Menu.
Table 8-1 IP Phone Programming Options User Menu
Programming Option Default
UserName ADMIN
Password 6633222

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SECTION 10 DHCP SERVER CONFIGURATION
It is possible to use either an external DHCP server (e.g., Windows Server) or the
UNIVERGE SV9100 internal DHCP server. With IP Phones, either of these options
requires the DHCP server to be configured to supply the IP terminal options.
If using the internal DHCP server, enable the DHCP server. Refer to 8.3 Example
Configuration 2 - Dynamic IP Addressing, One LAN on page 8-10.
When using an external DHCP server, you must add a new Option Code to the
DHCP scope for the GPZ-IPLE address. The method for adding this service varies
depending on the DHCP server used.
SECTION 11 AUTO CONFIG FOR IP TERMINALS SV9100
11.1 Required Equipment

 IP Phone Manager and software to create the config file for the IP terminals.
Software is available for download free on the NEC website.
 FTP server and free software available on the web.
 DHCP Server supporting ability to define the following:
 Vendor Class
 Option Codes

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11.2 Building Config File
When building a config file, follow the steps below to launch the phone manager
and create a file.
1. Launch the IP Phone Manager software.
2. Once the software is launched, click Auto Config.
Figure 8-8 IP Phone Manager

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3. Click Terminal.
Figure 8-9 Config Setup APL
4. Click 1st Server Address.

5. Assign the 1st Server Address using the IP address programmed in
command 10-12-09.

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6. Click OK.
Figure 8-10 Assign IP Address
7. Click 1st Server Port. Assign port number 5080.

8. Click OK.
Figure 8-11 Network Settings - 1st Server Port
9. After the changes are made, click Save.

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10. When the Save window opens, click Save as...
Figure 8-12 Save As/Name File Window
11. In the Save As window, name the file xxx.gz.
Example: To name the file test, enter test.gz

12. Place this file in the FTP server.

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11.3 Configuring an FTP Server
The file generated in the IP Phone Manager must be placed in an anonymous

login folder. The FTP server must be configured with an anonymous login
account.
The following procedure is an example using Quick and Easy FTP server.
1. Click Configure Settings.
Figure 8-13 Quick ‘n Easy FTP Server

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2. Click on the Advanced tab to specify the directory to store files.
Figure 8-14 Quick ‘n Easy FTP Server Configuration
3. Place the file (for example: test.gz) in the Default home directory.
4. After the file is loaded to the proper directory, click Start to start the FTP
server.
5. At this point the FTP server can be minimized to run in the background.

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11.4 DHCP Server Setup Windows Server
This section provides instructions for defining vendor classes and setting
Defining Vendor Classes

1. In the DHCP server highlight the server machine on the left side. Right click
on the server and select Define Vendor Classes.
2. Click ADD.
3. Display Name = NECDT800/DT700
4. Description = auto config
5. In the same window there is a section that shows ID, Binary, and ASCII.
Click under ASCII.
6. Enter NECDT800/DT700. This should have also added 4E 45 43 44 54 37
30 30 under the binary section.
7. Click OK and close.
Setting Predefined Options

1. Highlight the server again. Then right click and select Set Predefined
Options.
2. Change the option class to NECDT800/DT700.
3. Click ADD, and provide the following information:
 Name = FTP Address
 Data Type = IP address
 Code = 141
 Click OK, and start the process over again.
4. Click ADD, and provide the following information:

 Name = Auto Config File Name
 Data Type = String
 Code = 151
5. Click on ADD, and give the following information:

 Name = Download Protocol
 Data Type = Byte
 Code = 163
6. Click OK.

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Configuring Options
1. Highlight scope options on the left side. Then right click and choose
Configure Options.
2. Click Advanced and change the vendor class to NECDT800/DT700.
3. Place a check mark next to 141 FTP Address. Down below assign the IP
address of the FTP server. Then click Apply.
4. Place a check mark next to 151 auto config file name. Enter the name of the
config file created using IP Phone Manager. Then click Apply.
5. Place a check mark next to 163 download protocol. Down below change the
HEX address to be 0x1.
6. Click on Apply and OK.
SECTION 12 IP TELEPHONE REGISTRATION AND DELETION
When an IP Phone connects to the UNIVERGE SV9100 system, it is assigned the

first available port, starting from the value set in Program 11-02-01.
The ports are allocated in blocks of two.
For example:
 Insert a GPZ-IPLE
 Program 11-02-01 Extension Numbering
 Configure a System IP Phone and connect to the LAN
When connecting an IP Phone, the MAC address (ID) is automatically registered in

Program 15-05-02. If the registration in Program 15-05-02 is made manually (before
connecting the telephone) it uses the assigned port number when the telephone is
connected. The MAC address is printed on the barcode label on the bottom of the
telephone. It is a 12-digit alphanumeric number, ranging from 0~9 and A~F.
To delete a telephone registration:
Via Telephone Programming:
Enter Program 90-23-01, and enter the extension number of the IP Phone. Press 1
and Transfer to delete the registration.
Via Web Pro:
Enter Program 90-23-01, and place a check next to the extension number of the IP
Phone. Click on Apply to delete the registration.

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SECTION 13 SYSTEM IP PHONES AND ANALOG TRUNKS
Due to the nature of analog-to-digital conversion, considerable echo may be

encountered when using Analog Trunks with IP Phones.
Due to all Analog trunks being different, padding of the Analog Trunks in PRG 81-07
and 14-01 may be necessary. Even after the pad changes are made, echo may still
be present the first few seconds of the call while the echo cancellers are learning the
characteristics of the circuit on this call.
Program 90-68-01 can be used to automatically test the lines and auto assign the
proper values in Program 81-07. It is recommended to use this program whenever
analog trunks are involved.
It is recommended to use digital trunks when using IP phones for best performance.
Digital (ISDN, T-1, and SIP) trunks do not suffer from this problem.
SECTION 14 FIRMWARE UPGRADE PROCEDURE
A new version of NEC firmware for the IP Phones can be applied automatically or
manually.
The upgrade requires using an FTP/TFTP server. This is a software package that
runs on a PC. (These can be downloaded from the Internet, usually as freeware or
shareware.)
14.1 Manually Upgrading Firmware
Manually upgrading the firmware uses an FTP/TFTP server, but requires the
engineer visit each IP Phone individually. This may take longer, but is more
controlled as the downloads can be staggered to avoid excessive bandwidth
utilization.
To manually upgrade the firmware:

1. Install and configure an FTP/TFTP server.
2. Copy the firmware file itlisip(e,s,v).tgz to the default FTP/TFTP directory.
3. To enter Programming Mode, press the Help button on the IP Phone.
4. To enter Maintenance Mode, press 0 (Config) and #3 (Maintenance).
5. To access the Download menu, press 1.
6. Enter the FTP/TFTP server IP address in Option 2 - Download Address.
7. To enter the protocol, press #3 (Protocol FTP or TFTP).

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8. To select download by file, press option 1 (Download File).

9. To boot the program, press 3 (Boot Program).
10. Press the softkey.
The IP Phone downloads the firmware from the FTP/TFTP server and reboots
when the download is complete.
14.2 Checking the Firmware Version
To check the IP Phone firmware version:

1. Press and hold the Help button on the System IP Phone.
2. To access information, press 8 (System Information).
3. To display the firmware version, press the Up softkey.
14.3 Upgrading Automatically
This procedure causes all IP Phones to attempt firmware upgrade the next time
they connect to the GCD-CP10. This can make the upgrade procedure easier, as
it is not necessary to visit every telephone to perform the upgrade.
This can cause problems if, for example, a PoE (Power over Internet) switch is
used. When the PoE switch is powered up, all telephones connect to the FTP/
TFTP server at the same time. This causes a large amount of data for the FTP/
TFTP server to transfer over the data network.
To avoid this, connect the telephones to the PoE switch gradually, to allow time for
each telephone to upgrade before connecting the next.
To enable automatic upgrade:
When the IP phone boots up and connects to the SV9100 it receives download
information from the system that includes firmware information. When the SV9100
reports a version that is different than the version the IP phone is currently
utilizing, the IP phone will initiate the upgrade procedure.

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Below is an example of this setup.
IMPORTANT This is just an example, you must enter your own local information.
Enter the IP Address of the TFTP

server that the IP Phone Firmware
is stored on.
If using FTP, change the server

mode to FTP and enter the proper
IP Address and login name/
password.

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Enter the software version of the

files stored on the TFTP/FTP
server.
Enter the Hardware version of the

IP telephone.
SECTION 15 IP STATION (SIP MULTILINE TELEPHONE)
15.1 IP Phone Registration Modes
The SIP MLT supports three different registration modes.

 Plug and Play
 Automatic Login
 Manual Login
Plug and Play Registration
Plug and play registration mode allows for no authentication. As long as an IP

terminal is configured with the proper IP addressing scheme, and plugged into the
network, the phone comes on-line. In plug and play mode you may assign an
extension number into the IP terminal or allow the system to automatically set an
unused extension number for the station. When the system assigns unused
extension numbers it starts searching for the first available port or starts at a
preassigned port and works its way up from there.
Automatic Login
When set to automatic login the SIP user name and password must be entered in
the configuration in the IP terminal. When the phone tries to register with the CPU
it checks the user name and password against its database. If the user name and
password match, the phone is allowed to complete registration. If the user name
and password do not match, the phone cannot register with the CPU. The IP
terminal displays an error message: Unauthorized Auto Login.

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Manual Login
When set to manual login, no user name, password, or extension number is

entered into the configuration of the telephone. The user is prompted to enter this
user name and password into the IP phone. This information is cross referenced
in the phone system to an associated extension number. If a match is found, the
phone comes online. If there is no match, the phone cannot complete registration
with the CPU. The IP terminal either returns to the login/password screen, or locks
out the user and requires the administrator to unlock the IP terminal. Lockout on
failed attempts is dependent on system programming. Manual mode is good for
an environment where multiple users share the same IP phone at different times.
As one user logs out the next user can login with their credentials and all of their
associated programming follows.
In Manual mode a user can also logoff the IP phone to allow another user to login
with their own login ID and password. To logoff the IP phone use the following
operation:
Press the "Down Arrow" Soft Key, press the "Prog" soft key, and then press the
"LOGOFF" soft key.
Multiple Login
The same user name and password can be assigned to multiple extensions when
using Automatic or Manual Registration. This makes it easier on the user by only
having to remember one password. For example, if a user has an IP Multiline
terminal and uses UC Suite Applications with the Enhancement bundle controlling
the IP Multiline, three different ports are used in the system.
Encryption
The SV9100 Supports AES 128-bit encryption between DT800/DT700 terminals

and the GPZ-IPLE. This feature requires the SV9100 ENCRYPTION LIC which is
a system license. Once installed, any of the DT800/DT700 IP terminals can use
the encryption feature.
Source Destination SRTP Comment
DT800/DT700 SDT SIP (P2P) N

DT800/DT700 STD SIP (Non P2P) DT800/DT700 VoIPDB Encryption
S between DT800/DT700 and
VoIPDB
DT800/DT700 DT800/DT700 S
DT800/DT700 PSTN DT800/DT700 VoIPDB Encryption
S between DT800/DT700 and
VoIPDB
DT800/DT700 IP Network (SIP/H323/ DT800/DT700 VoIPDB Encryption
CCIS) S between DT800/DT700 and
VoIPDB

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Source Destination SRTP Comment
DT800/DT700 DT800/DT700
N
Encryption On Encryption Off
S = Supported N = Not Supported
Figure 8-15 DT800/DT700 Encryption
Conditions
 Encryption is not supported on DT800/DT700 series phones that are connected
via NAPT.
 Encryption is not supported on DT800/DT700 series phones that are registered

to a secondary NetLink system.
 DT800/DT700 series phones that are registered to a primary NetLink can fail
over to a secondary system regardless of encryption settings.
 If the encryption feature is enabled in terminal programming but not licensed,

the terminal display will no longer display "Invalid Info", however the encryption
feature will not work until licensed.
15.2 General IP Configuration
The voice quality of VoIP depends on variables such as available bandwidth,

network latency, and quality of service initiatives (QOS). These variables are
controlled by the network and Internet service providers. Because these variables
are not in NEC control, NEC cannot guarantee the performance of the users IP
based voice solution. NEC recommends connecting the VoIP equipment through
a local area network using private IP addresses.

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To ensure a network meets the specific requirements for VoIP implementation, an

IP ready check and a site survey must be completed at each site before VoIP
implementation.
 One way delay must not exceed 100ms.
 Round trip delay must not exceed 200ms.
 Packet loss must not exceed 1%.
 Data switches must be manageable.
 Routers must provide QOS.
 There must be adequate bandwidth for estimated VoIP traffic. Refer to Section
15.5 Bandwidth on page 8-44.
Depending on how QOS policies are built in the network, assignments may be
needed in both the CPU and IP terminal. The UNIVERGE SV9100 supports the
flagging of packets at layer 2 (VLAN tagging 802.1Q/802.1P) and at layer 3 levels.
15.3 VLANs
A VLAN is used to logically break up the network and minimize broadcast

domains. Without VLANS, the network must be physically segmented to break up
broadcast domains. Each network segment is then connected through a routing
device adding latency and cost. Latency is a delay in the transmission of data and
is caused by routing packets from one LAN to another. In a VoIP environment
latency must be kept to a minimum.
802.1Q allows a change in the Ethernet Type value in the Ethernet header tagging
the Protocol ID 0x8100, identifying this frame as an 802.1Q frame. This inserts
additional bytes into the frame that composes the VLAN ID (valid IDs = 1 ~ 4094).
802.1P allows you to prioritize the VLAN using a 3-bit priority field in the 802.1Q
header. Valid VLAN priority assignments are 0 ~7. A tag of 0 is treated as normal
data traffic giving no priority. Under normal circumstances the higher the tag
numbers, the higher the priority. However this is left up to the network
administrator as they could set the exact opposite where the lower tag numbers
have a higher priority.
Currently the IPLE and CPU do not support the tagging of VLAN packets. These
devices also do not support receiving a frame with a VLAN tag. If either device
receives a packet with a VLAN tag, it is treated as an illegal frame and discarded.
Therefore when the CPU/IPLE is plugged into a data switch supporting VLANS,
the VLAN tag must be removed before passing the frame onto the CPU/IPLE.
Tagging Voice and Data Packets
Built into the IP phones is a 2 port 10/100 manageable data switch allowing for a
PC connection on the back of the IP phone. This built in data switch also supports
802.1Q and 802.1P VLAN tagging capabilities.

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The following procedures describe two methods for tagging the voice packets and
the data packets separately, using the PC, or using the phone keypad.
15.3.1 Logging In on the PC
Follow these steps to log into a PC.
1. Web browse to the IP address of phone.
2. To log in, enter default user name: ADMIN
3. Enter default password: 6633222
4. Click OK.
Figure 8-16 Log In to IP Phone

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15.3.2 Tagging Voice Packets Using IP Phone
Follow these steps to program for voice packet tagging using an IP

telephone.
1. Log in. Refer to Section 15.3.1 Logging In on the PC on page

8-30.
2. To apply a tag to the voice packets only, go to Network
Settings>Advanced Settings>LAN port settings.
3. Access the following three menus to select options for LAN Port
Settings:
 VLAN Mode
 VLAN ID
 VLAN Priority
Figure 8-17 LAN Port Settings Window
4. Select the VLAN Mode to enable or disable this feature.

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5. Select either Enable or Disable (default) and click OK.
Figure 8-18 VLAN Mode
6. VLAN ID allows an entry of 1~4094 for the VLAN ID. VLAN Mode
must be enabled for this entry to be valid.
Enter the VLAN ID and click OK.
Figure 8-19 VLAN ID
7. VLAN Priority allows an entry of 0~7 for the VLAN Priority. VLAN
mode must be enabled for this entry to be valid.
Enter the required priority, and click OK.
Figure 8-20 VLAN Priority

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15.3.3 Tagging Data Packets Using IP Phone
Follow these steps program for data packet tagging using an IP telephone.
1. While logged in to the IP address of the phone on the PC, go to

Network Settings>Advanced Settings>PC Port Settings.
Refer to Section 15.3.1 Logging In on the PC on page 8-30.
2. Access the following three menus to select options for PC Port
Settings:
 Port VLAN Mode
 Port VLAN ID
 Port VLAN Priority.
Figure 8-21 PC Port Settings Window
3. Select the VLAN Mode to enable or disable this feature.

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4. Select either Enable or Disable (default) and click OK.
Figure 8-22 Port VLAN Mode
 The remaining data packets settings for VLAN on the PC Port are
the same as those for the voice packets.
5. VLAN ID allows an entry of 1~4094 for the VLAN ID. VLAN Mode
must be enabled for this entry to be valid.
Enter the VLAN ID, and click OK.
Figure 8-23 Port VLAN ID
6. VLAN Priority allows an entry of 0~7 for the VLAN Priority. VLAN
mode must be enabled for this entry to be valid.
Enter the required priority, and click OK.
Figure 8-24 Port VLAN Priority

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7. Click Save.
Figure 8-25 Save Network Settings
8. After saving settings, click OK to confirm. The telephone reboots and

applies the VLAN settings.
Figure 8-26 Save Confirmation Window

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15.3.4 Entering VLAN Settings by Phone (Voice Packets Only)
Follow these steps to enter VLAN settings using a telephone.

8-30.
2. Press and hold the phone Menu button until the screen changes.
3. Press 0 on the dial pad to access Configuration.
4. Press the OK soft key. No login or password is associated with the IP

terminal in the default state.
5. Press 1 on the dial pad for Network Settings.
6. Press 6 on the dial pad for Advanced Settings
7. Press 1 on the dial pad for LAN Port Settings (VLAN for the voice
packets only)
8. Press 2 on the dial pad for VLAN Mode
9. Press 1 or 2 to disable/enable the VLAN for the voice packets. Press

the OK soft key after the setting is changed.
10. Press 3 on the dial pad for VLAN ID
11. Enter a valid VLAN ID of 1~4094. Press the OK soft key after the
setting is changed.
12. Press 4 on the dial pad for VLAN Priority
13. Enter the VLAN priority of 0~7. Press the OK soft key after the setting
is changed.
14. If no more changes are made, press the Exit soft key three times.
Then press the Save soft key, and the phone reboots.
15.3.5 Entering VLAN Settings for PC Port by Phone (Data Packets

Only)
Follow these steps to enter the VLAN setting for PC port using a telephone.

8-30.
2. Access the main menu.
3. Press 6 on the dial pad for Advanced Settings.
4. Press 2 on the dial pad for PC Port Settings.
5. Press 2 on the dial pad for Port VLAN Mode.
6. Press 1 or 2 to disable/enable the VLAN for the data packets. Press

the OK soft key after the setting is changed.
7. Press 3 on the dial pad for Port VLAN ID.

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8. Enter a valid VLAN ID of 1~4094. Press the OK soft key after the
setting is changed.
9. Press 4 on the dial pad for Port VLAN Priority.
10. Enter the VLAN priority of 0~7. Press the OK soft key after the setting
is changed.
11. If no more changes are made, press the Exit soft key three times.
Then press the Save soft key, and the phone reboots.
15.4 ToS Settings (Layer 3 QoS)
The marking of packets at layer 3 is done by marking the ToS byte in the IP
header of the voice packet. The UNIVERGE SV9100 supports two methods for
marking the ToS byte:
 IP precedence
 DSCP (Diffserv)
IP Precedence
IP Precedence uses the first 3 bits of the ToS field to give eight possible
precedence values (0~7). Under normal circumstances the higher the number the
higher the priority. However this is left to the network administrator for setup. The
administrator may assign this in exactly the opposite manner with the lower
values having a higher priority. Below are the eight common values for IP
precedence.
 000 is an IP precedence value of 0, sometimes referred to as routine or best
effort.
 001 is an IP precedence value of 1, sometimes referred to as priority.
 010 is an IP precedence value of 2, sometimes referred to as immediate.
 011 is an IP precedence value of 3, sometimes referred to as flash.
 100 is an IP precedence value of 4, sometimes referred to as flash override.
 101 is an IP precedence value of 5, sometimes called critical.
 110 is an IP precedence value of 6, sometimes called Internetwork control.
 111 is an IP precedence value of 7, sometimes called network control.

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Working in conjunction with IP precedence, the next 4 bits in the ToS field are
designed to influence the delivery of data based on delay, throughput, reliability,
and cost. However these fields are typically not used.
The following table shows the 8-bit ToS field and the associated IP precedence
bits.
IP IP IP Not
Precedence Precedence Precedence Delay Throughput Reliability Cost Used
1(on) here = 1(on) here = 1(on) here =

value of 4 value of 2 value of 1
Differential Services Code Point (DSCP)
DSCP stands for Differential Services Code Point (or Diffserv for short). It uses
the first 6 bits of the ToS field therefore giving 64 possible values.
The following list shows the most common DSCP code points with their binary
values and their associated names:
DSCP Code
Binary Values Names
Points
000000 0 Best Effort (BE)

100110 38 Assured Forwarding 43 (AF 43)
101110 46 Expedited Forwarding (EF)

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DSCP Code
Binary Values Names
Points

The following table shows the 8 bit TOS field and the associated Diffserv bits.
Diffserv Diffserv Diffserv Diffserv Diffserv Diffserv Not Not
Used Used
1(on) here 1(on) here = 1(on) here 1(on) here 1(on) here 1(on) here
= value of | value of 1 = value of = value of = value of = value of
32 6 8 4 2 1
IP Precedence/Diffserv Values Submitted in Command 84-10
Assignments for the IP Precedence/Diffserv values in the system are submitted in

command 84-10. This setting data affects only the packets sent by the IPLE card.
This does not affect the packets sent from the IP terminals.
Figure 8-27 84-10: ToS Setup
To set the IP Precedence/Diffserv bits for packets leaving the IP terminal there are
the following two options:
 System wide. If all IP phones use the same ToS value, this can be assigned in
commands 84-23-06 and 84-23-12. When an IP phone registers with the CPU,
it looks for settings in these commands. If these are found, they override any
previous individual settings.

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 Individual. If different IP phones require different ToS assignments, due to the

network configuration, these assignments must be set at each individual station
Command 84-23 requires a Hexadecimal representation of the 8 bit ToS field.

For example, to assign the signaling packets an IP precedence value of 4 and the
voice packets an IP precedence value of 5, it would be as follows. Refer to Figure
8-28 SIP DT800/DT700 Basic Setup.
 80 in Hex is 10000000 - This represents the signaling packets leaving the IP
phone
 A0 in Hex is 10100000 - This represents the voice packets leaving the IP phone
Figure 8-28 SIP DT800/DT700 Basic Setup

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The following table shows the common IP Precedence/Diffserv values and their
hexadecimal equivalent.
Table 8-2 Common IP Precedence/Diffserv Values and Hexadecimal Equivalent
IP Precedence
Hex Value
Name
IP Precedence 1 20
IP Precedence 2 40
IP Precedence 3 60
IP Precedence 4 80
IP Precedence 5 A0
IP Precedence 6 C0
IP Precedence 7 E0
DSCP
Hex Value
Name
DSCP 10 28
DSCP 12 30
DSCP 14 38
DSCP 16 40
DSCP 18 48
DSCP 20 50
DSCP 22 58
DSCP 24 60
DSCP 26 68
DSCP 28 70
DSCP 30 78
DSCP 32 80
DSCP 34 88
DSCP 36 90
DSCP 38 98
DSCP 46 B8
DSCP 48 C0
DSCP 56 E0

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Enter Values on a Per Phone Basis

 By the web browser
 By configuration mode through the dial pad
To enter the values per phone, browse to the individual phone or enter the
configuration mode through dial pad.
The following example describes assigning these fields via the web browser.
1. Log in on PC. Refer to Section 15.3.1 Logging In on the PC on page 8-30.
2. Go to Network Settings>Advanced Settings>Type Of Service.
Figure 8-29 Type of Service Window
3. There are two choices: RTP and SIP. RTP = voice packets and SIP =
signaling packets.
Select each field and assign the appropriate value. Then select OK.
These fields are also looking for a Hexadecimal value as with command 84-
23. Refer to Table 8-2 Common IP Precedence/Diffserv Values and
Hexadecimal Equivalent on page 8-41.

Issue 5.0
Access the following menus to select options:

 RTP - Voice Packets
 SIP - Signalling Packets
Figure 8-30 RTP - Voice Packets
Figure 8-31 SIP - Signalling Packets
4. After selecting Save, the following message appears.

5. Select OK and the phone reboots. Once online, the phone tags all packets
with the applied settings.
Assign Values on IP Terminal
The following is an example of assigning fields on the IP terminal.

1. Press and hold the Menu button until display changes.
2. Press 0 on the dial pad for Configuration.
3. At a default state there is no login or password associated to the IP terminal.
Press the OK soft key.
4. Press 1 on the dial pad for Network Settings.
5. Press 6 on the dial pad for Advanced Settings.
6. Press 4 on the dial pad for Type of Service.
7. There are two options (1) is RTP and (2) is SIP.
8. Press 1 on the dial pad for RTP (voice packets), enter the hexadecimal
value, and then press the OK soft key.

Issue 5.0
9. Press 2 on the dial pad for SIP (Signaling packets), enter the hexadecimal
value, and then press the OK soft key.
10. If no more changes are to be made, press the Exit soft key three times, and
then press the Save soft key. The phone reboots.
15.5 Bandwidth
The bandwidth required for VoIP calls depends on the following factors.
 Layer 2 media
 CODEC
 Packet Size
 RTP Header Compression
 Voice Activity Detection (VAD)
 Number of simultaneous calls
 Possibly add encryption after research.
Layer 2 media is concerned with moving data across the physical links in the
network. A few of the most common layer 2 media types are Ethernet, PPP, and
Frame Relay.
CODEC stands for Coder/Decoder and is the conversion of the TDM signal into
an IP signal and vice versa. A CODEC can also compress/decompress the voice
payload to save on bandwidth.
Packet Size is the amount of audio in each PDU (protocol data unit) measured in
milliseconds. The larger the packet the less bandwidth used. This is because
sending larger packets (more milliseconds of voice) requires, overall, less packets
to be sent. The downside of this practice is if a packet is dropped/lost a larger
piece of voice is missing from the conversation as the system waits the additional
delay for the next packet arrival.
RTP Header Compression compacts the RTP header from 40 bytes in size to 2
~ 4 Bytes in size. RTP header compression is used only on low speed links.
Regularly on every voice packet there is an IP/UDP/RTP header that is 40 bytes
in length. Compressing this header, down to 2 ~ 4 bytes, can save a considerable
amount of bandwidth. The following is an example of a VoIP packet without RTP
header compression and one of a packet with RTP header compression.
Notice that the overall packet size, when using RTP header compression, is
considerably smaller.
 VoIP packet without RTP header compression
IP Header UDP Header RTP Header

VOICE PAYLOAD
20 bytes 8 Bytes 12 bytes

Issue 5.0
 VoIP packet with RTP header compression
Compressed
VOICE PAYLOAD
Header 2 ~ 4 bytes
Voice Activity Detection (VAD) is suppression of silence packets from being sent
across the network. In a VoIP network all conversations are packetized and sent,
including silence. On an average a typical conversation contain anywhere from
35% ~ 45% silence. This can be interrupted as 35% ~ 45% transmission of VoIP
packets, as having no audio, using valuable bandwidth. With the VAD option
enabled, the transmitting of packets stops after a threshold is met determining
silence. The receiving side then injects comfort noise into the call so it does not
appear the call has dropped.
Bandwidth Calculations
The first step in calculating the bandwidth of a call is determining how many bytes
the voice payload is going to use. The amount is directly affected by the CODEC
and packet size. Below are the supported default CODEC speeds for SIP
Multiline telephones.
 G.711 = 64000bps
 G.722 = 64000bps
 G.729 = 8000bps
Payload Calculation Voice

 (Packet size * CODEC bandwidth) / 8 = Voice Payload in Bytes
 Example of G.711 with a 20ms packet size
 (.020 * 64000) /8 = 160 Bytes
 Example of G.729 with a 30ms packet size
 (.030 * 8000) /8 = 30 Bytes
Now that you have the voice payload in bytes you can calculate the overall
bandwidth including the layer 2 media. Below are some of the common layer 2
media types and their overhead.
 Ethernet = 18 Bytes
 802.1Q/P Ethernet = up to 32 bytes
 PPP = 9 Bytes
 Frame Relay = 6 Bytes
 Multilink Protocol = 6 Bytes

Issue 5.0
Bandwidth Calculation
( [ Layer 2 overhead + IP/UDP/RTP header + Voice Payload ] /

Voice Payload ) * Default CODEC speed = Total Bandwidth
Example of a G.711 call over Ethernet using a 20ms packet size and not
using RTP header compression
(.020 * 64000) / 8 = 160 Bytes for Voice Payload
([ 18 + 40 + 160] / 160) * 64000 = 87200bps
If VAD is not enabled each side of the conversation would be streaming
87.2kbps in one direction for a total of 174.4kbps.
The following chart shows the supported CODECS for IP phones with different
packet sizes over PPP and Ethernet.
G.711 10 103.2 kbps 110.4 kbps

G.711 20 83.6 kbps 87.2 kbps
G.711 30 77.1 kbps 79.5 kbps
G.711 40 73.9 kbps 75.6 kbps
G.722 10 103.2 kbps 110.4 kbps

G.722 20 83.6 kbps 87.2 kbps
G.722 30 77.1 kbps 79.5 kbps
G.722 40 73.9 kbps 75.6 kbps
G.729 10 47.2 kbps 54.4 kbps

G.729 20 27.6 kbps 31.2 kbps
G.729 30 21.1 kbps 23.5 kbps
G.729 40 17.8 kbps 19.6 kbps
G.729 50 15.9 kbps 17.3 kbps
G.729 60 14.5 kbps 15.7 kbps

Issue 5.0
When using Video (H.263) soft phone, add the bandwidth for the video portion to
the call. The estimated bandwidth per video stream is as follows:
<Receive> Approximately 40k ~ 120k
<Send> Approximately 40k ~ 120k
15.6 Some Network Considerations
When adding the SV9100 to a customers network there are many things to be
aware of. Before implementation a detailed network diagram of the existing
network must be obtained from the customer. This diagram may provide you with
information about possible network conditions that can prevent or hinder the VoIP
equipment from functioning correctly.
Firewalls
Another regular device in customer networks that can hinder VoIP performance is
a firewall. Most corporate LANs connect to the public Internet through a firewall.
A firewall is filtering software built into a router or a stand alone server unit. It is
used to protect a LAN it from unauthorized access, providing the network with a
level of security. Firewalls are used for many things, but in its simplest form, a
firewall can be thought of as a one way gate. It allows outgoing packets from the
local LAN to the Internet but blocks packets from the Internet routing into the local
LAN, unless they are a response to query.
A firewall must be configured to allow specific traffic from the Internet to pass
through onto the LAN. If an IP phone is deployed out over the Internet there is a
very good chance it is passing through a firewall, either at the MAIN , the remote,
or both locations.
The following diagram shows two IP phones on the corporate local LAN and one
IP phone on a Remote network connected via the Internet. The two phones that
are installed on the local LAN are functioning correctly. The IP phone at the
remote site cannot register therefore it is not working.
Figure 8-32 SV9100 Network Example No. 1
Headquarters
Internet
Local LAN
Remote Network
Firewall Firewall

Issue 5.0
The green arrow in the diagram above represents the data packets leaving the IP
phone destined for the SV91000 on the Headquarters LAN. The firewall on the
Headquarters network is not configured to recognize the UDP ports used by the
NEC equipment thus blocking them from resulting in registration failure. To solve
this issue the ports used by the NEC VoIP equipment must be opened in the
firewall allowing the NEC traffic to pass through onto the SV9100.
The ports that are required open on the Headquarters location are 5080 (UDP) for
signaling and the voice ports which depend on how many IPLE ports are installed.
 IPLE UDP Ports : 10020~10531
The ports that need to be opened on the Remote network are 5060 (UDP) for
signaling and ports 3462 and 3463 for voice (UDP).
VPN
Another common feature is the use of the Internet as the WAN between customer
locations. When this is done VPNs are typically used between the locations. A
VPN (Virtual Private Network) is a private data network that maintains privacy
through the use of tunneling protocols and security features over the public
Internet. This allows for remote networks (with private addresses), residing behind
NAT routers and/or firewalls, to communicate freely with each other. When
building the VPN tunnels, throughout the network, they must be assigned as a
fully meshed network. This means that every network is allowed direct connection
to each and every other network in the topology. Network equipment limitations
may sometimes restrict this ability resulting in no voice path on VoIP calls
between sites. When this happens Peer-to-Peer must be disabled in the SV9100.
The downside to disabling Peer-to-Peer is using more DSPs and consumption of
additional bandwidth at the MAIN location.

Issue 5.0
network is not fully meshed due to the lack of a VPN tunnel between Sites B and
C.
Figure 8-33 SV9100 Network Example No. 3
With Peer-to-Peer enabled, the IP phones on site A can communicate with IP

phones on sites B and C. IP Phones on sites B and C cannot communicate
directly with each other though. The IP phone from site B can set up a call to the
IP phone at site C, but there is no speech path. Here are the steps in the call
scenario leading to the failed call.
 Extension 120 goes off hook and dials ext 130.
 An initial invite message is sent from 192.168.2.15 (ext 120) to 192.168.1.10

(IPLE).
 192.168.1.10 (IPLE) forwards that message to 192.168.3.26 (ext 130).
 In the original setup message there is a field labeled SDP (Session Description
Protocol). The SDP portion informs the IP phone where to send the media
(voice) to. The SDP portion of this invite message contains the IP address of
192.168.2.15 (ext 120).
 192.168.3.26 (ext 130) sends a 200 OK message to 192.168.1.10 (IPLE). In the

200 OK message is the SDP field reporting the IP address of 192.168.3.26 (ext
130).
 192.168.1.10 (IPLE) forwards this message to 192.168.2.15 (ext 120).
 192.168.2.15 (ext 120) sends an ACK message to 192.168.1.10 (IPLE).
 192.168.1.10 (IPLE) forwards this message to 192.168.3.26 (130).
 At that point the two IP phones attempt to send voice packets directly to each
other. As there is no VPN connection between these sites the call is set up with
no voice path.

Issue 5.0
To correct this issue another VPN connection between sites B and C is required. If
an additional VPN cannot be implemented, due to network limitations, the Peer-
to-Peer feature can be disabled in the SV8100. With Peer-to-Peer disabled, all
packets (Signaling and Voice) route through the GPZ-IPLE card. This also affects
IP phones at the REMOTE locations calling other IP phones at the same location.
Without Peer-to-Peer enabled the voice path must route to the MAIN location and
then back to the REMOTE instead of directly between the two stations on the
REMOTE network. This forces the use of additional bandwidth on the MAIN, and
REMOTE locations. Peer-to-Peer is disabled in command 15-05-50 per IP Phone.

Issue 5.0
15.7 Guide to Feature Programming
Program
Program Name Description/Comments Assigned Data 1 2 3
Number
0.0.0.0~
126.255.255.254
GCD-CP10 Network 128.0.0.1~
IPLE uses the Default Gateway that
10-12-03 Setup – Default 191.254.255.254 X
is assigned here.
Gateway 192.0.0.1~
223.255.255.254
Default is 0.0.0.0
0.0.0.0~
126.255.255.254
128.0.0.1~
GCD-CP10 Network Assign Layer 3 IP Address to the
10-12-09 191.254.255.254 X
Setup – IP Address IPLE connected to CCPU.
192.0.0.1~
223.255.255.254
GCD-CP10 Network
Assign Subnet Mask to the IPLE
10-12-10 Setup – Subnet Default is 255.255.0.0 X
connected to CCPU.
Mask
Assign an IP Address to IPLE Default Value:
IPL Basic Setup – IP
84-26-01 X
Address
172.16.0.10
Range: 0 ~ 65534
IPL Basic Setup –
84-26-02 Default Values: X
RTP Port Number
RTP Port = 10020
Range: 0 ~ 65534
IPL Basic Setup –
RTCP Port Number
84-26-03 Default Values: X
(RTCP Port Number
+1)
RTCP Port +1 = 10021

Issue 5.0
Program
Number
Define which of the three 0 = Normal

Registration modes you wish the 1 = Automatic
SIP MLTs to use. 2 = Manual
Normal
When the phone boots up it will
report the ext assigned in the phone
or choose the next available
extension in the system. No
password required.
DT800/DT700 Auto
Server Information If set to auto then the SIP user
10-46-01 X
Setup – Register name and password must be
Mode entered into the actual IP phone.
These settings have to match
Programs 84-22/15-05-27 or the
phone does not come on-line.
Manual
When the phone boots up it prompts
you to enter a user ID and password
before logging in. It checks this user
ID/password against Programs 84-
22/15-05-27. If there is not a match,
the phone does not come on-line. Default is 0
USER ID of the SIP URL if Program Up to 32 characters.
10-46-05 is set to domain name.
A SIP URL is made up of three

DT800/DT700
parts. Domain name, host name,
Server Information
and server name.
10-46-04 Setup – Server X
Name
e.g. At default the server name is
sipphd. The URL could look like the
following:
[email protected] Default is sipphd

Port the SIP messages are sent to Range = 0 ~ 65535
DT800/DT700 on the VoIPU card. This same port
Server Information number must be assigned in the SIP
10-46-06 Setup – Register Multiline terminals. X
Port
Changing this command also
requires a CPU reset. Default is 5080
DT800/DT700
0 = Off
Server Information
10-46-07 Enable or disable encryption mode. 1 = On X
Setup – Encryption
Default is 0
Mode

Issue 5.0
Program
Number
DT800/DT700
Server Information 0 = Mode 1
10-46-08 Assign the encryption type. X
Setup – Encryption Default is 0
Type
DT800/DT700 Password used when Program Valid Characters
Server Information 10-46-07 is set to ALL. Assign a (0~9, *, #)
10-46-09 X
Setup – One-Time character string of 10 characters or
Password less. Default Not assigned
DT800/DT700 With Automatic logon the starting Range = (1 ~ 960)
10-46-10 Server Information port number for automatic port X
Setup – Start Port allocation. Default = 1
IP Telephone 1 = H.323
Terminal Basic Data Type of IP terminal registered with 2 = SIP
15-05-01 X
Setup – Terminal the specified extension number. 3 = None
Type 4 = DT800/DT700
Allow association of a MAC Address
IP Telephone
to an extension. When the IP phone 00.00.00.00.00.00~
Terminal Basic Data
sends a register message to the FF.FF.FF.FF.FF.FF
15-05-02 Setup – IP Phone X
CPU the CPU responds back with Default is
Fixed Port
the extension number associated to 00.00.00.00.00.00
Assignment
the MAC address.
IP Telephone
0.0.0.0~
Terminal Basic Data IP address the IP Terminal is using
15-05-07 255.255.255.255. X
Setup – Using IP for the specified extension number.
Default is 0.0.0.0
Address
1 = Type 1
IP Telephone 2 = Type 2
Assign CODEC type for IP Terminal.
Terminal Basic Data 3 = Type 3
15-05-15 If SIP SLT, use Program 84-19. If X
Setup –CODEC 4 = Type 4
SIP MLT, use Program 84-24.
Type 5 = Type 5
1 = Default is 1
0 = No Option
IP Telephone 1 = 8LK Unit
Terminal Basic Data Read Only CM showing type of Line 2 = 16LK Unit
15-05-19 X
Setup – Side Option Key unit installed on the telephone. 3 = 24ADM
Information Default is 0
READ ONLY
0 = No Option
IP Telephone 1 = ADA
Terminal Basic Data Read Only CM showing type of 2 = BHA
15-05-20 X
Setup –Bottom adapter installed on the telephone. 4 = BCA
Option Information Default is 0
READ ONLY

Issue 5.0
Program
Number
0 = Normal Handset
IP Telephone 1 = Handset for Power
Terminal Basic Data Read Only CM showing type of Failure (PSA/PSD)
15-05-21 X
Setup –Handset Handset installed on the telephone. 2 = BCH
Option Information Default is 0
READ ONLY
IP Telephone 0 = No Setting
Terminal Basic Data 1~32 = DSS Console
DSS console number when installed
15-05-22 Setup – Side Option Number X
to a telephone.
Additional Default is 0
Information READ ONLY
0 = No Setting
IP Telephone 1-16 = Terminal
Terminal Basic Data equipment number
15-05-23 Setup –Handset (TEN) of Bluetooth X
Option Additional Cordless Handset
Information (BCH)
Default is 0
When enabled allows SIP Multi-Line 0 = Not Used
IP Telephone
phones to use the Security button 1 = Used
Terminal Basic Data
15-05-24 located at top of the SIP MLT X
Setup –Protection
display. When disabled, the Security
Service
key has no effect. Default is 0
0 = No Setting (default)
1 = ITL-**E-1D/IP-*E-1
2 = ITL-**D-1D/ITL
12BT-1D/
ITL-12PA-1D
(without 8LKI(LCD)-
L)
3 = ITL-**D-1D/ITL-
12BT-1D/ITL-12PA-
1D
(with 8LKI(LCD)-L)
IP Telephone
4 = ITL-320C-1
Terminal Basic Data
Assign type of SIP MLT terminal 5 = SoftPhone
15-05-26 Setup – DT800/ X
connected. 6 = CTI
DT700 Terminal
7 = AGW
Type
8 = IP3NA-8WV
9 = Not Used
10 = ITL-**DG-3
11 = ITL-**CG-3
12 = ITL-2CR-1
13 = ITZ-**D-1D/ITZ-
**PD-1D/ITZ-**pA-1D/
ITZ-**DG/ITZ-**LDG
14 = ITZ-*CG
15 = ITZ-*DE
16 = ITZ-*LDE

Issue 5.0
Program
Number
For SIP Multiline phone using 0 = Not Set

IP Telephone Manual/Auto registration. Assign 1-960 = Set
Terminal Basic Data each phone a unique personal
15-05-27 X
Setup –Personal ID index. When complete go to
Index command 84-22 and set the user
name and password. Default is 0
IP Telephone
0 = Disable
Terminal Basic Data
15-05-28 1 = Enable X
Setup –Additional
Default is 0
Information Setup
IP Telephone
Terminal Basic Data 0.0.0.0 ~
15-05-29 Setup –Terminal Future use with NAT 255.255.255.255 X
WAN Side IP Default is 0.0.0.0
Address
IP Telephone
Terminal Basic Data
0 = Disable
Setup –DTMF Play
15-05-30 1 = Enable X
During Conversation
Default is 0
at Receive
Extension
IP Telephone
Terminal Basic Data
0 = Disable
Setup – Alarm Tone
15-05-31 1 = Enable X
During Conversation
Default is 1
(RTP packet loss
alarm)
IP Telephone
0 = Disable
Terminal Basic Data
15-05-32 1 = Enable X
Setup – Key Pad
Default is 0
Talkie
IP Telephone 0 = Disable
Terminal Basic Data 1 = Enable
15-05-33 X
Setup – LAN Side IP Default is 0.0.0.0
Address of Terminal READ ONLY
IP Telephone Whether the touch screen
0 = Off
Terminal Basic Data used on ITL-320C-( ) (BK)
15-05-34 1 = On X
Setup – Terminal TEL can be used (1) or
Default is 1
Touch Panel On/Off cannot be used (0).

Issue 5.0
Program
Number
Assignments deal with setting of the

Layer 3 IP Header ToS field as it
leaves the VoIPDB unit. Protocol Type
Specify the protocol to assign the 1 = Not Used

ToS field for, and select the 2 = Not Used
populated field to conform to either 3 = Voice Control
IP Precedence or Differentiated 4 = H.323
84-10-XX ToS Setup Services. 5 = RTP/RTCP X
6 = SIP
When setting IP Precedence, assign 7 = CCISoIP
Priority, Delay, Throughput, and 8 = SIP MLT
Reliability in Programs 84-10-01/02/ 9 = SIP Trunk
03/04/05. 10 = NetLink
11 = Video RTP/RTCP
When setting DiffServ, only assign
the DSCP in Program 84-10-07.
DT800/DT700
Assign up to 32 Alpha/
Multiline Logon User ID for Manual or Auto
84-22-01 Numeric Characters X
Information Setup – registration (Program 10-46-01).
User ID
DT800/DT700
Multiline Logon Password for Manual or Auto
Information Setup – registration (Program 10-46-01).
Password
DT800/DT700 When set to manual login mode, the 0 = Off
Multiline Logon user ID is omitted from the display 1 = On
84-22-03 X
Information Setup – during entry by the user.
User ID Omission Default is 0
DT800/DT700
Allow the ability to log off from the IP 0 = Off
Multiline Logon
84-22-04 terminal when using manual 1 = On X
Information Setup –
registration mode. Default is 1
Log Off
DT800/DT700
Multiline Logon
Nickname
DT800/DT700
Multiline Basic At half the value of this timer the IP
Range: 60~65535 Sec.
84-23-01 Information Setup – terminal sends another registration X
Default is 180
Registration Expire message to the CPU.
Timer
DT800/DT700
Multiline Basic At half the value of this timer the IP
Range: 60~65535 Sec.
84-23-02 Information Setup – terminal sends another Subscribe X
Default is 3600
Subscribe Expire message to the CPU.
Timer

Issue 5.0
Program
Number
DT800/DT700
Multiline Basic
At half the value of this timer the IP Range: 60~65535 Sec.
84-23-03 Information Setup – X
terminal sends a re-invite message. Default is 180
Session Expire
Timer
DT800/DT700
Multiline Basic
Minimum time the CPU accepts a Range: 60~65535 Sec.
session timer for a new call. Default is 180
Minimum Session
Expire Timer
When INVITE message received Range: 0~65535 Sec.
from SIP MLT does not contain
DT800/DT700
Expires header, the CPU uses this
Multiline Basic
84-23-05 value for timeout of outgoing call. X
E.g. The SIP MLT hears RBT for
Invite Expire Timer
duration of this timer and then is
disconnected. Default is 180
DT800/DT700
Multiline Basic
Used for updating the IP terminals Range: 0x00 ~ 0xFF
SIGNALING TOS values. Default is 00
Signal Type of
Service
DT800/DT700 The time that an IP terminal holds Range: 0 ~ 65535 Sec.
Multiline Basic an error message in the display.
84-23-07 X
Information Setup – Setting 0 holds the error message
Error Display Timer indefinitely. Default is 0
When Digest Authentication mode is Range: 0 ~ 4294967295
ON, this value is available.
After receiving Initial INVITE without
DT800/DT700 authentication information, CPU will
Multiline Basic send 401 message to the SIP MLT,
Information Setup – then waits for an INVITE message
84-23-08 X
Digest Authorization with the authentication message
Registration Expire from SIP MLT within this timer.
Timer Additionally, after receiving Re-
REGISTER message for Keep Alive
purpose, the CPU sends a 401
message. Default is 0
The number of times an incorrect Range: 0 ~ 255
password can be entered when the
security key is pressed.
DT800/DT700
If set to (1), only one attempt is
Multiline Basic
allowed. When number of password
retries is met an error message
Number of
displays on the phone: Incorrect
Password Retries
security code password entered,
press call key to contact an
administrator Default is 0

Issue 5.0
Program
Number
DT800/DT700
Multiline Basic Time to leave the terminal Locked Range; 0 ~ 120
84-23-11 Information Setup – Out after entering the wrong security Default: 0 X
Password Lock code. Default is 0
Time
DT800/DT700 Assign the network admin telephone
Up to 32 Digits
Multiline Basic number. When the user presses the
84-23-12 (0~9, *, #, P, R, @) X
Information Setup – Call key to contact the network
Reference Number administrator, this number is dialed.
DT800/DT700
Multiline Basic Range: 0x00 ~ 0xFF
Assign the IP terminals MEDIA TOS
84-23-13 Information Setup – (0~9, A~F) X
values.
Media Type of Default is 00
Service
DT800/DT700
Multiline Basic The valid period of the REFER Range: 0 ~ 65535 Sec.
84-23-14 X
Information Setup – subscription. Default is 60
Refer Expire Timer
Range: 1~4
DT800/DT700
Multiline CODEC
1 = 10ms
Basic Information Amount of audio in each RTP
84-24-01 2 = 20ms X
Setup – packet.
3 = 30ms
Number of G.711
4 = 40ms
Audio Frames
Default is 2
DT800/DT700
Multiline CODEC
0 = Disable
Basic Information
84-24-02 Enable/Disable VAD for G.711 1 = Enable X
Setup –
Default is 0
Detection
DT800/DT700
Multiline CODEC 0 = A-law
84-24-03 Basic Information -law used in N.A. 1 = -law X
Setup – Default is 0
G.711 Type
DT800/DT700
Multiline CODEC
Basic Information Minimum value of the dynamic jitter Range: 0 ~ 300ms
84-24-04 X
Setup – buffer. Default is 20
G.711 Jitter Buffer
Minimum

Issue 5.0
Program
Number
DT800/DT700
Multiline CODEC
Basic Information Average value of the dynamic jitter Range: 0 ~ 300ms
84-24-05 X
G.711 Jitter Buffer
Average
DT800/DT700
Multiline CODEC
Basic Information Maximum value of the dynamic jitter Range: 0 ~ 300ms
84-24-06 X
G.711 Jitter Buffer
Maximum
Range: 1~4
DT800/DT700
1 = 10ms
Multiline CODEC
2 = 20ms
84-24-07 3 = 30ms X
Setup – packet.
4 = 40ms
Number of G.729
5 = 50ms
Audio Frames
6 = 60ms
Default is 2
DT800/DT700
Multiline CODEC
0 = Disable
Basic Information
84-24-08 Enable/Disable VAD for G.729 1 = Enable X
Setup –
Default is 0
Detection
DT800/DT700
Multiline CODEC
Basic Information Minimum value of the dynamic jitter 0~300ms
84-24-09 X
G.729 Jitter Buffer
Minimum
DT800/DT700
Multiline CODEC
Basic Information Average value of the dynamic jitter 0~300ms
84-24-10 X
G.729 Jitter Buffer
Average
DT800/DT700
Multiline CODEC
Basic Information Maximum value of the dynamic jitter 0~300ms
84-24-11 X
G.729 Jitter Buffer
Maximum

Issue 5.0
Program
Number
DT800/DT700
1 = Static
Multiline CODEC
2 = Self adjusting
84-24-17 Basic Information X
3 = Adaptive immediate
Setup –
Default is 2
Jitter Buffer Mode
DT800/DT700
Multiline CODEC
Basic Information 0 ~ 30
84-24-18
Setup - Voice Default is 20
Activity Detection
Threshold
DT800/DT700
Multiline CODEC
Basic Information 40 ~ 70 dBm
84-24-23 Setup - Echo X
Canceller Non- Default is 70dBm
linear Processing
Noise
DT800/DT700
Multiline CODEC
0 = Disable
Basic Information
84-24-25 1 = Enable X
Setup –
Default is 1
Echo Canceller 4W
DET
DT800/DT700 0~3
Multiline CODEC 0 = G.711_PT
Basic Information This assigns the CODEC to be 2 = G.729_PT
84-24-28 X
Setup – used. 3 = G.722_PT
Audio Capability Default is 0
Priority
Range: 1~4
DT800/DT700
Multiline CODEC
1 = 10ms
84-24-32 2 = 20ms X
Setup – packet.
3 = 30ms
G.722 Audio Frame
4 = 40ms
Number
Default is 3
84-24-33 --Not Used--
DT800/DT700
Multiline CODEC
Basic Information Minimum value of the dynamic jitter Range: 0 ~ 300ms
84-24-34 X
G.722 Jitter Buffer
Minimum

Issue 5.0
Program
Number
DT800/DT700
Multiline CODEC
Basic Information Average value of the dynamic jitter Range: 0 ~ 300ms
84-24-35 X
G.722 Jitter Buffer
Average
DT800/DT700
Multiline CODEC
Basic Information Maximum value of the dynamic jitter Range: 0 ~ 300ms
84-24-36 X
G.722 Jitter Buffer
Maximum
DT800/DT700
Multiline Firmware
84-28-01 Maximum 64 characters. Default is No Setting X
Name Setup –
Firmware Directory
DT800/DT700
Multiline Firmware
84-28-02 Maximum 30 characters. Default is No Setting X
Name Setup –
Firmware File Name
15.8 SIP MLT Quick Startup Guide
The following guides describe the setup for a SIP MLT from a default state for
these modes:
 Plug and Play
 Automatic Registration
 Manual Registration
15.8.1 Plug and Play

1. Program 10-12
Assign the IPLE registration/signaling IP address, subnet mask, and
default gateway. If no customer provided default gateway is provided,

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leave Gateway IP address at 0.0.0.0.
Figure 8-34 System Data 10-12: GCD-CP10 Network Setup

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2. Program 84-26
Assign IP addresses to the DSPs that are to be used. The IP
addresses assigned must be in the same subnet as the address in
Program 10-12-09.
After these commands are uploaded to the CPU, a system reset must
be applied.
Assign an IP address to the IPLE. An IPLE provides 256 voice paths
from IP to TDM & vice versa with appropriate licenses.
Figure 8-35 System Data 84-26: IPLE Basic Setup (DSP)
3. Program 11-02
SIP MLT Stations are assigned to non-equipped hardware ports.
Physical Station ports are assigned automatically from lowest number
ascending as cards are added to the system.
Because of this you should assign SIP MLT Stations starting with the
higher number ports. By default all Station Ports are assigned
numbers in the SV9100. These are easily changed in Program 11-02
to the required station number as long as the leading digit/digits are
set in Program 11-01 as Extension.
Ports are dedicated to VoIP stations in groups of 2. For example, in
the image to the left, if port 504 (Extension 5203) is used for a SIP
MLT station, that group of 2 ports (Ports 503 and 504) is now
dedicated to VoIP use only.

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After one port in a block of two is used by a VoIP station, the

remaining port can be used only for another VoIP Extension.
Figure 8-36 System Data 11-02: Extension Numbering
4. This step is optional.

To enable Key data and other station feature programming (before IP
Phone is brought online) the extensions must be identified as IP
Phones. Once checked in the IP Phone List in PCPro, the extensions
are available for selection in Program 15 and other station related

Issue 5.0
Programs.
Figure 8-37 PCPro Programming Unregistered Phones
Figure 8-38 IP Phone List
5. The SIP MLT Station requires assignments to be made in the phone

itself. Enter the Program Mode in the station using the following steps.

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❍ The station does not require an Ethernet connection to

NOTE
enter program mode. Only power is required. Power
can be provided by an AC adapter plugged into the
phone or by POE provided by a data switch. If the data
switch is providing POE it must be using the 802.3af
standard.
For Basic bench testing only the following assignments
are required:
1 Press 0 on the dial pad for configuration mode.
You are prompted with a User Name and Password.

The defaults are:
User Name: ADMIN
Password: 6633222
The user name should already be entered in the

terminal.
2 Press Set soft key to step down to the password field.
3 After you enter the password, press OK soft key.
Network Settings
 DHCP Mode – DHCP Disable. Click OK.
 IP Address – Enter the IP Address for the station, and click OK.
 Default Gateway – Enter the Default Gateway Address, and click
OK. If you are testing without a router/gateway, this must be left at
the default 0.0.0.0
 Subnet Mask - Enter the Subnet Mask for the station, and click
OK.
SIP Settings
 SIP User – Intercom Number
Enter the extension number for the IP station, and click OK.
 Server Address & URI – 1st Server Address
Enter the IP address assigned in command 10-12-09, and click
OK.
 SIP Server Port - 1st Server Port
Enter port 5080, and click OK.
 Press the EXIT key until you are back at the Main menu.
 Press the SAVE key and the phone saves the configuration to
memory, reboots itself and registers with the CPU.
15.8.2 Automatic Registration
Follow these steps for automatic registration.
1. Steps 1 ~3 are the same as for Plug and Play mode. Step 4 is
not optional and MUST be assigned when using Automatic
Registration.

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2. Same as Plug and Play mode.
3. Same as Plug and Play mode.
4. To enable key data and other station feature programming before IP

Phone is brought online, the extensions must be identified as IP
Phones. Once checked in the IP Phone List in PC-Pro (see images
below), the extensions are available for selection in Program 15 and
other station related Programs.
Figure 8-39 PCPro Programming Unregistered Phones
Figure 8-40 IP Phone List

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5. Program 10-46
Change Program 10-46-01 to Automatic.
Figure 8-41 Automatic Registration Setting
6. Program 15-05-27
Each IP phone requires a unique personal ID index. Valid settings
are 1 ~ 960.
Figure 8-42 Automatic Registration Basic Setup
Figure 8-43 Automatic Registration Personal ID Index
7. Program 84-22-01
Assign the user ID and password to be associated with the Personal
ID Index assigned in Step 6.

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Figure 8-44 Automatic Registration User Name and Password Assignment


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
NOTE
switch is providing POE it MUST be using the 802.3af
standard.
are required:

The defaults are:
User Name: ADMIN
Password: 6633222

terminal.
3 After you enter the password, press the OK soft key.
Network Settings
 DHCP Mode - DHCP Disable. Click OK.
 IP Address - Enter the IP Address for the station, and click OK.
 Default Gateway - Enter the Default Gateway Address, and click
the default 0.0.0.0.
 Subnet Mask - Enter the Subnet Mask for the station, and click OK.
SIP Settings
 SIP User
 User ID - Enter User ID assigned in command 84-22. Click
OK.
 Password - Enter the password assigned in command 84-22.
Click OK.
 Incom Number - Enter the extension number for the IP
station. Click OK.
 Server Address & URI - 1st Server Address
OK.
Enter port 5080. Click OK.
 Press the SAVE key, and the phone saves the configuration to
15.8.3 Manual Registration

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Follow these steps for manual registration.
Steps 1~4 are the same as for Section 15.8.2 Automatic Registration on
page 8-66.
1. Same as for Automatic Registration Mode.

5. Program10-46 – ChangeProgram10-46-01 to Manual.
Figure 8-45 Manual Registration


Issue 5.0

NOTE
switch is providing POE it MUST be using the 802.3af
standard.
are required:

The defaults are:
User Name: ADMIN
Password: 6633222

terminal.
3 After you enter the password, press the OK soft key.
Network Settings
 DHCP Mode - DHCP Disable. Click OK.
 IP Address - Enter the IP Address for the station, and click OK.
 Default Gateway - Enter the Default Gateway Address, and click
the default 0.0.0.0
 Subnet Mask - Enter the Subnet Mask for the station, and click OK.
SIP Settings
 Do not enter any information in the SIP user field. When the phone
boots up, it requires a user name and password. These are
preassigned in the system. When entered correctly, the phone is
provided an extension number.
 Server Address & URI - 1st Server Address
OK.
Enter port 5080, and click OK.
 Press the SAVE key, and the phone saves the configuration to

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15.9 IP Phone Relocation
The IP Phone Relocation feature gives users access to their IP telephone from
any location by using the override login function. Users have the flexibility of
logging into their IP Station in the office as well as remotely at the home office.
IP Phone Relocation Override
IP Phone Relocation is a feature for overriding the registration of an IP phone

from various locations. To override the registration of an IP phone, you must have
the login ID and Password of that IP phone.
Conditions
 Multiple IP Phones cannot use the same user ID and the same password at the
same time.
 When a user is using multiple IP Phones at the same time, the user ID and
password must be different for each phone.
 When a user is using SoftPhone (CTI mode) and controlling the IP Phone by
this SoftPhone, the user ID and password should be different for the SoftPhone
and IP Phone.
 An IP Phone (IP Phone and Soft phone) with DSS console cannot override
another IP Phone.
 An IP Phone (IP Phone and Soft phone) with DSS console cannot be
overridden from another IP Phone.
 2-button Phones cannot use the IP Phone Relocation feature.
 The login ID and Password are programmed in Program 15-05-27 and

Program 84-22.
 IP Phone Relocation can be used only in Manual Registration Mode.
 The system sees terminal types 1 (Economy), 2 (Value), 3 (Desi-Less), 4

(Sophisticated), and 5 (Softphone) as the same terminal type.
 When using Multiple Login, the same Personal ID index can be assigned to an
ITL/Softphone and a CTI (Desktop).
 Two ports of the same terminal type (Program 15-05-26) cannot be assigned to
the same Personal ID index (Program 15-05-27).
 Program 10-46-01 must be set to 1 (Auto), or 2 (Manual) for Multiple Login to

work.
 When three ports are assigned the same Personal ID index, in Program 15-05-
27, if Program 15-05-26 is not set for those ports, the terminal types will be
assigned based on order of login. If Program 15-05-26 is set, the login order
does not matter and they will assign the correct port.

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 Override is not supported in a SV9100 system that had a 3rd Party CTI
connection to the CPU (i.e., UC Suites Apps Shared Services, UCB), or to a
terminal with a 1st Party CTI connection (i.e., PC Assistant/Attendant and
Softphone or 1st Party TAPI driver), and would show Rejected Override >>>CTI
Link... in the display.
 Override with CTI is supported on a per station basis using Program 15-05-39
with certain restrictions.
 When using Override with an active CTI connection, Program 15-05-39 must be
enabled for the extensions that will be overridden. The overriding terminal must
be of the same type and number of line keys as the terminal to be overridden. If
the types of terminals and number of keys are different between overriding and
overridden phones, the Telephony Service Providers (1st Party and 3rd Party)
may not function properly.
Table 8-3 IP Phone Relocation
Program/Item Description/
No. Selection
0 = Normal
Set up the information of the SIP
DT800/DT700 Server 1 = Auto
Multiline (DT800/DT700 series) Server.
10-46-01 Information Setup – Register 2 = Manual
Mode
This PRG is a system-wide setting.
Default is 0
0~960 Used when the SIP Multiline telephone
is using manual/
0 auto registration. Assign each phone a
IP Telephone Basic Setup –
15-05-27 unique personal index. Then go to
Personal ID Index
command 84-22 to assign
the user name and
Default is 0 password.
DT800/DT700 Multiline Up to 32 characters Input the User ID when using manual or
84-22-01 Logon Information Setup – auto
User ID Default not assigned registration (Program 10-46-01).
DT800/DT700 Multiline Up to 16 characters Input the Password when using manual
84-22-02 Logon Information Setup – or
Password Default not assigned auto registration (program 10-46-01).
0 = Off
Input the Personal ID from terminal
DT800/DT700 Multiline 1 = On
automatically when log on again.
84-22-04 Logon Information Setup –
Log Off
If set to 0, IP Phone Relocation fails.
Default is 1

Issue 5.0
IP Phone Relocation Flow Chart
The following flow chart can be used to enable the IP Phone Relocation feature.
Every user must enter both login ID and Password.
Figure 8-46 IP Phone Relocation Flow Chart

Issue 5.0

UNIVERGE® SV9100
IP Single Line Telephone (SIP) Chapter 9
Session Initiation Protocol (SIP) Station feature provides Voice over Internet Protocol
(VoIP) for IP stations. This feature is defined by the Internet Engineering Task Force
(IETF) RFC3261.
SIP analyzes requests from clients and retrieves responses from servers, then sets
call parameters at either end of the communication, handles call transfer and
terminates. Typically, Voice over IP services are available from an SIP service
provider.
With the GPZ-IPLE up to 256 TDM talk paths are supported. This total may be
shared among SIP stations or SIP trunks. Registered SIP stations and/or SIP trunks
require a one-to-one relation with the GPZ-IPLE DSP Resource. This is a required
component of SIP implementation in the UNIVERGE SV9100. The UNIVERGE
SV9100 GPZ-IPLE contains a regular TCP/RTP/IP stack that can handle real-time
media and supports industry standard SIP (RFC3261) communication on the WAN
side.
For this feature, the GPZ-IPLE is installed and assigned. The GPZ-IPLE supports IP
signaling for up to 256 (SIP Trunks and/or SIP Stations) and reduces the maximum
capacity of system stations and/or Trunks in accordance with the number of
registered SIP Stations.
A maximum of (1) GPZ-IPLE can be installed per UNIVERGE SV9100 System,

supporting the maximum of 896 IP stations.
The UNIVERGE SV9100 supports the following CODECS that are considered to
provide toll-quality equivalent speech path.
The following voice compression methods are supported for the IP Station SIP
feature:
 G.729. Low bandwidth requirement used on most Wide Area Network links.
 G.711. A/Law – High bandwidth requirement usually used on Local Area
Networks.
 G.722 This CODEC is useful in fixed network, Voice over IP applications, where the
required bandwidth is typically not prohibitive.
 G.726 is an ITU-T ADPCM speech-coded standard covering the transmission of

Issue 5.0
voice at rates of 16-, 24-, 32-, and 40Kbps.
The SIP Station feature set supports the HOLD and TRF features based on RFC
draft.
 Draft-Ietf-sipping-service-examples-09.txt.
 Draft-ietf-sipping-service-examples- (Transfer - Attended) 15.txt
 IETF RFC is defined as: Internet Engineering Task Force (RFC) Request for
Comments.
 The SIP Station feature set supports the Message Waiting Indication (MWI) based
on RFC3842.
 When out of band DTMF is used, via RFC2833. The IPLE supports an out of band
DTMF payload of 96 ~ 127.
 If a user on a standard SIP phone is talking to another station via voice announce
(receiving station has not pressed speaker or lifted the handset) and the SIP phone
presses transfer or hold, the call will be terminated. A standard SIP call cannot be
placed on hold or transferred until the other party answers.
 SIP INFO works independent from other DTMF methods such as RFC2833. This
means SIP Terminals should send DTMF information by a single method otherwise
the system will receive both separately causing double digits.
 The system has the ability to receive DTMF information in SIP INFO messages sent
by Standard SIP Terminals. This allows the SIP Terminal to initiate features during a
ringing state such as CAMP ON and Message Waiting. SIP Terminal must support
this feature and have it enabled.
 When PRG 15-05-49 is set to 2: Allowed while RTP is not available, SIP INFO will
be received while RTP is not established. In-band method such as RFC2833 will be
used once voice path is established.
 When PRG 15-05-49 is set to 1: Allowed any time, SIP INFO will be received
whenever they arrive.
 The SV9100 supports NAT for Standard SIP Terminals.
Enhancements
 With Version 2.00 or higher Standard SIP Conference is supported.
9-2 IP Single Line Telephone (SIP)

Issue 5.0
SECTION 2 VOIP AND CPU LAN LINK - SPEED AND DUPLEX MODE
Description
With CPU Version 3.00 or higher, the SV9100 provides the information of CPU and
IPLE LAN Link speed and Duplex mode.
Conditions
 When LAN link is up, this program shows the information of Speed & Duplex mode
of the link.
 PRG 90-77 data can be accessed from PCPro, WebPro & TelPro
 PCPro can show the information once it has downloaded the data from CPU.
System Availability
GPZ-IPLE
Applications
 PCPro
 WebPro
 TelPro

Issue 5.0

feature.
Level
Program
Number
1 2 3
90-77-01 LAN Link Speed Information – LAN Link 0 = No Link 0

Speed of CPU 1 = 1Gbps, Full Duplex
This program shows the speed & duplex mode 2 = 1Gbps, Half Duplex
of CPU LAN Link. 3 = 100Mbps, Full Duplex 
(This is a read only program) 4 = 100Mbps, Half Duplex
90-77-02 LAN Link Speed Information – LAN Link 0 = No Link 0

Speed of VoIP 1 = 1Gbps, Full Duplex
This program shows the speed & duplex mode 2 = 1Gbps, Half Duplex
of VoIP DB LAN Link. 3 = 100Mbps, Full Duplex 
(This is a read only program) 4 = 100Mbps, Half Duplex
Related System Data

None.
Limitation
 LAN Link Speed of the secondary system of NetLink is not displayed by the Primary
System. This information can be accessed from the WebPro of secondary system.

Issue 5.0
SECTION 3 STANDARD SIP-TCP CONNECTION SUPPORT
Description
With CPU Version V3.00 or higher SV9100 supports TCP connection of standard SIP
phones.
Conditions
 If Request Header & Contact Header in REGISTER request contains
“transport=tcp” then SV9100 uses TCP protocol and PRG 15-05-51 is set to
1(TCP).
 If Request Header & Contact Header in REGISTER request contains
“transport=udp” or the Register request doesn’t contain transport information then
SV9100 uses UDP protocol and PRG 15-05-51 is set to 0(UDP).
 SV9100 supports STD SIP Terminal which works without Registration. In that case
Transport protocol should to be set in PRG 15-24-04.
 SV9100 supports SIP Terminals with both transport protocols, UDP & TCP at the
same time.
 If SV9100 doesn’t get the transport protocol information (e.g. system data broken)
then transport protocol is taken as UDP.
License
Version 3 Enhancement License (0413) required
GPZ-IPLE
System Availability
Terminals
Supported Terminals
ST450
Polycom - VVX500
Polycom - VVX600

Issue 5.0
Unsupported Terminals are:

 ST465
 Real Presence
 NEC u-Mobility
 UT880
 G-TEK

Issue 5.0

feature.
Level
Program
Number
1 2 3
15-05-51 IP Phone Basic Setup – Transport Protocol 0=UDP 0

This program shows the transport protocol for 1=TCP
selected SIP terminal.

(This is a read only program.)
90-77-02 Registration setting in the standard SIP 0=UDP 0

terminal where REGISTER isn't used– 1=TCP
Transport Protocol 
This program sets the transport protocol for
selected SIP terminal without register method.
Related System Data

None.
Limitation
 When SV9100 uses SIP terminals with TCP Protocol via NAT the router must be set
to Static NAPT mapping.

Issue 5.0
SECTION 4 STD SIP CONFERENCE
Description
With SV9100 software 2.00 or higher Standard SIP Terminals can initiate a
conference call.
Conditions
 PRG 20-13-08 must be enabled for the Class of Service the Standard SIP Terminal
is in.
 A DSP Resource is needed for each Standard SIP or DT800/700 that is in the
conference.

Issue 5.0
Member 1 Member 2 Member 3 Necessary Number of DSP
STD- SIP DT400 DT400 1

STD- SIP STD SIP DT300 2
STD- SIP DT800 STD-SIP 3
: : : :
: : : :
 Video calls are not supported.

 The following conference features are Not Supported with Standard SIP:
 Meet Me Conference
 Barge in to Conference
 Split between the parties in conference
 Transfer a call into a conference
Default Setting
None
System Availability
Terminals
 Standard SIP Terminal
Required Components
 GCD-CP10
 GPZ-IPLE
 System Port License (0300)
 VoIP Resource License (5103)
 IP Terminal License (5111)
Related Features
 IP Single Line Telephone (SIP)
 Conference

Issue 5.0
Programming
Program
Number
Class of Service Options Turn off or on an extension 0 = Off

20-13-08 (Supplementary Service) user ability to initiate a 1 = On X
Conference Conference. Default = 1
Operation
Polycom VVX 500
To establish a conference:
1. Establish an intercom or trunk call.

2. Press the “Hold” softkey to hold the first call.
3. Establish a second call (intercom or trunk), which is to be added in the conference.
4. When called party answers, press the “Join” softkey to start the conference.
5. Repeat steps 2-4 to add more parties to the conference.
-OR-

2. Press the “Conference” softkey to hold the first call.
3. Establish second call (intercom or trunk), which is to be added in the conference.
4. When called party answers, press the “Conference” softkey to start the conference.
NEC ITX-1DE-1W

2. Press the “Hold key” to hold first call.
3. Establish a second call (intercom or trunk), which is to be added in the conference.
4. When call party answers, press the “CONF” key to start the conference.

Issue 5.0
SECTION 5 STD SIP TRANSFER–UNATTENDED
Description
Any standard SIP terminal can perform an Unattended (Blind/Unsupervised) transfer.
Conditions
 Program 10-05-50 (Peer-to-Peer Mode) must be disabled for the Unattended
Transfer to be performed.
 A SIP terminal must receive the re-Invite message of Session Timer in a state of
Unattended transfer.
 When the transferred destination terminal is busy, unanswered or the extension
number in the Refer-To header is wrong or out of service, the call is sent back to the
original terminal.
 If the standard SIP phone is placed on hold/park, from another extension, this call
cannot be transferred until the station that placed the call on hold/park retrieves the
call. A unattended transfer can only be completed while both parties are in a talking
state.
 An Unattended Transfer can only be performed to the following locations:
 Extension Number
 Department Group Pilot Number
 ACD Group Pilot Number
 Operator Access
 Trunk/Alternate trunk access code
 F-Route Access
 Network Access
 Quick transfer to Voice Mail is not supported when using Unattended Transfer.
Default Setting
None
System Availability
Terminals
 Standard SIP Terminal
Required Components
 GCD-CP10

Issue 5.0
 GPZ-IPLE
Related Features
 Transfer
SECTION 6 NAT MODE FOR STANDARD SIP TELEPHONE
Description
The SV9100 supports NAT for Standard SIP Terminals.
Conditions
 When PRG 10-33-05 NAT mode for SIP phone is set to 1 (Enable), the P2P mode
for SIP Phone becomes always Off, regardless of PRG 15-05-50 setting.
 Standard SIP Video call feature which uses P2P mode cannot be established in the
same system, since the P2P mode is disabled by enabling PRG 10-33-05.
 When connecting multiple SIP Phones via NAT, PRG 15-05-18 has to be set to
admit registration of multiple SIP Phones which are using the same IP address. For
example, if you had a STD SIP Terminal that had two lines registering with the same
IP Address, you would need to flag PRG 15-05-18 for both Extension numbers.
 In the router/firewall that the SV9100 resides behind port forwarding is required.
Port forwarding at the SIP Terminal end is not required as long as PRG 15-05-45
(Plug and Play) is enabled. The ports that must be forwarded to the SV9100 are as
follows:
 UDP Port 5070 MUST be forwarded to the IP Address assigned in PRG 10-12-09.
 UDP Ports 10020 ~10083, UDP Ports 10020~10147 and UDP Ports 10020~10275
(GPZ-IPLE) MUST be forwarded to the IP Address(s) assigned in PRG 84-26-01.
 When PRG 15-05-45 is set to “1” the manual table setting for port forwarding may
not be required on the remote side router, but the router must support the NAT
function itself. If PRG 15-05-45 is set to “0” port forwarding at the Remote side
router is required. This feature requires the installation of GPZ-IPLE.
Default Setting
None
System Availability
Terminals
 Standard SIP Terminal

Issue 5.0
Required Components
 GCD-CP10
 GPZ-IPLE
Related Features
 Transfer
Programming
Program
Number
0.0.0.0~
126.255.255.254
128.0.0.1~
GCD-CP10 Network Assign the default gateway IP
10-12-03 191.255.255.254 X
Setup- Default Gateway Address for the CPU.
192.0.0.1~
223.255.255.254
(default = 0.0.0.0)
0.0.0.0 ~
126.255.255.254
GCD-CP10 Network
128.0.0.1 ~
Setup - NAPT Router IP Define the IP of the WAN side
10-12-07 191.255.255.254 X
Address (Default Of the router.
192.0.0.1 ~
Gateway [WAN]
223.255.255.254
(default = 0.0.0.0)
When receiving ICMP redirect
0= (Enable)
GCD-CP10 Network messages, this determines if
10-12-08 1= (Disable) X
Setup – ICMP Redirect the IP Routing Table updates
(default = 0)
0.0.0.0~
Assign the IP Address for the
126.255.255.254
VoIPDB. If a VoIPDB is
128.0.0.1~
installed in the system it is
GCD-CP10 Network 191.255.255.254
10-12-09 recommended to set PRG 10- X
Setup – IP Address 192.0.0.1~
12-01 to 0.0.0.0 and all
223.255.255.254
connections to the system will
(default =
be made through the VoIPDB.
172.16.0.10)

Issue 5.0
Program
Number
128.0.0.0
192.0.0.0
224.0.0.0 240.0.0.0
248.0.0.0
252.0.0.0
254.0.0.0
255.0.0.0 255.128.0.0
255.192.0.0
255.224.0.0
255.240.0.0
255.248.0.0
255.252.0.0
255.254.0.0
255.255.0.0
255.255.128.0
GCD-CP10 Network Define the Media Gateway 255.255.192.0
10-12-10 X
Setup – Subnet Mask Subnet Mask Address. 255.255.224.0
255.255.240.0
255.255.248.0
255.255.252.0
255.255.254.0
255.255.255.0
255.255.255.128
255.255.255.192
255.255.255.224
255.255.255.240
255.255.255.248
255.255.255.252
255.255.255.254
255.255.255.255
(default =
255.255.0.0)
0 = Auto Detect
1 = 100Mbps, Full
Duplex
Define the LAN interface
GCD-CP10 Network 2= 10Mbps, Full
10-12-11 Speed and Mode of the VoIP X
Setup – NIC Setup Duplex
Application supported.
3 = 1Gbps, Full
Duplex
(default = 0)
Enable or Disable Peer-to-
Peer mode for SIP Phone.
When PRG 10-33-05 NAT
0 = No (Disable)
IP Phone Basic Setup - mode is set 1 = Enable, the
15-05-50 1 = Yes (Enable) X
Peer to Peer Mode P2Pmode for SIP Phone is
(default = 0)
always set (Off) automatically
regardless of this program
setting.

Issue 5.0
Program
Number
When connecting a standard

SIP Registar/Proxy SIP terminal via NAT, this 0 = Disable
10-33-02 Information Basic Setup opion must be enabled to 1 = Enable X
- Authentication Mode prohibit illegal SIP phone (default = 0)
registration.
10-33-05 SIP Registar/Proxy When system controls remote 0 = Disable
Information Basic Setup SIP phone via NAT router, set 1 = Enable X
– NAT Mode this program to 1 = Enable. (default = 0)
10-37-01 UPnP Setup – UPnP Enable/Disable UPnP. 0 = Disable
Mode 1 = Enable X
(default = 0)
10-58-01 Network Address This program sets the IP or 0.0.0.0~
Network address for phones 126.255.255.254
that are not to be routed 128.0.0.1~
through the NAPT 191.255.255.254
translations. 192.0.0.1~
223.255.255.254
For example, if a system had (default = 0.0.0.0)
multiple NAPT phones and
another site, with multiple IP
phones connected via a VPN
X
connection, you would not
want the phones connected
over the VPN to use the
NAPT feature. The network
address (or single IP phone
addresses) of the Remote
location would be entered
here. This is for the IP
Phones at this location to not
use the NAPT feature.

Issue 5.0
Program
Number
10-58-02 Network Address – This program sets the 128.0.0.0 / 192.0.0.0

Subnet Mask netmask for the IP Addresses 224.0.0.0 / 240.0.0.0
assigned in Program 10-58- 248.0.0.0 / 252.0.0.0
01. 254.0.0.0 / 255.0.0.0
255.128.0.0
255.192.0.0
255.224.0.0
255.240.0.0
255.248.0.0
255.252.0.0
255.254.0.0
255.255.0.0
255.255.128.0
255.255.192.0
255.255.224.0
X
255.255.240.0
255.255.248.0
255.255.252.0
255.255.254.0
255.255.255.0
255.255.255.128
255.255.255.192
255.255.255.224
255.255.255.240
255.255.255.248
255.255.255.252
255.255.255.254
255.255.255.255
(default =
255.255.0.0)
15-05-16 IP Telephone Terminal Assign the authentication Up to 24 characters
Basic Data Setup – password for SIP single line (default not assigned)
X
Authentication telephones.
Password

Issue 5.0
Program
Number
15-05-45 IP Telephone Terminal This program is valid when 0 = Disable

Basic Data Setup – NAT Program 10-46-14 is On (NAT 1 = Enable
Plug and Play feature activated). (default = 0)
Select sending RTP port
number to remote Router, use
from negotiation result (0) or
received RTP packet (1).
SV9100 uses this program to
decide a destination port of
RTP transmitting packets from
IPLE to a remote IP terminal.
X
If “0:OFF” is selected, the
destination port of RTP
transmitting packets will be a
SIP/SDP negotiation
result.(same behavior as
before). If you chose “1:ON”,
the destination port of RTP
transmitting packet will be the
same port of a source port of
a receiving RTP packet on
IPLE. IPLE is required.
15-05-47 IP Telephone Terminal On a per station basis, this 0 = Disable
Basic Data Setup – setting defines the SIP 60~65535 (sec)
Registration Expire registration expiry timer. (default = 180)
X
Timer for NAT If this value is set to 0, for a
Program 84-23-01 is applied.
15-05-48 IP Telephone Terminal On a per station basis, this 0 = Disable
Basic Data Setup – setting defines the SIP 60~65535 (sec)
Subscribe Expire Timer Subscribe expiry timer. (default = 180)
X
for NAT If this value is set to 0, for a
Program 84-23-02 is applied.
84-20-01 SIP Extension Basic Define SIP station Proxy Port. 1~65535
Information Setup – (default = 5070) X
Registrar/Proxy Port
84-26-01 IPL Basic Setup – IP Assign the IP address for xxx.xxx.xxx.xxx
Address each DSP on the IPLE. Defaults: X
Slot 1 = 172.16.0.20
84-26-02 IPL Basic Setup – RTP Assign the RTP port number 0~65534
Port Number to be used for each DSP on Defaults:
the IPLE. RTP Port = 10020 X
Only even numbered ports
are supported.
84-26-03 IPL Basic Setup – RTCP Define the TCP port number 0~65534
Port Number (RTP Port for RTCP to use for each Defaults: X
Number + 1) DSP. RTCP Port = 10021

Issue 5.0
SECTION 7 PROGRAMMING
7.1 Configure VoIPDB Networking Information
The VoIPDB should be connected to the same IP Subnet as the GCD-CP10. It

also requires a valid IP address. Use these programs to assign VoIPDB network
related information.
 If any IP Address or NIC setting is changed, the system must be reset for the
changes to take affect.
Program
Number
GCD-CP10 Network Assign the IP Address for 0.0.0.0 ~ 126.255.255.254

Setup –Default the Router. 128.0.0.1 ~ 191.254.255.254
10-12-03 Gateway 192.0.0.1 ~ 223.255.255.254 X
Default is 0.0.0.0
GCD-CP10 Network Assign the IP Address of 0.0.0.0 ~ 126.255.255.254
Setup – IP Address the VoIPDB. 128.0.0.13~191.255.255.254
10-12-09 192.0.0.1~223.255.255.254 X
GCD-CP10 Network Assign the Subnet Mask of 255.255.0.0
10-12-10 Setup –Subnet Mask the VoIPDB. X
IP Trunk Availability Assign the H.323 Trunk 0 = None (Default)
– Trunk Type Availability. 1 = SIP
10-68-01 X
2 = H.323
3 = CCIS
IP Trunk Availability Assign the start port Range: 0~400
10-68-02 X
– Start Port number of H.323 Trunks. Default is 0
IP Trunk Availability Assign the Number of Range: 0~400
10-68-03 X
– Number of Ports H.323 Trunks Default is 0

Issue 5.0
7.1.1 VoIPDB (DSP) Basic Setup
Use these programs to make basic VoIPDB assignments.
Description/
Program Number Program Name Assigned Data 1 2 3
Comments
IPL Basic Setup – IP Default:

84-26-01 X
Address Slot 1 = 172.16.0.20
IPL Basic Setup – RTP Assign the RTP Port 0~65534
84-26-02 Port Number Number. X
Default for is 10020
IPL Basic Setup – Assign the RTCP Port 0~65534
RTCP Port Number Number.
84-26-03 X
(RTCP Port Number + Default for is 10021
1)
7.1.2 VoIP ToS Setup
Use this programs to define ToS.
Program Description/
Program Name Assigned Data 1 2 3
Number Comments
ToS Setup – ToS Mode Use this field to define 0 = Disable (Invalid)
your SIP QoS marking 1 = IP Precedence
for ToS or Diffserve. 2 = Diffserv
When Input Data is set

84-10-01 to 1, Item No. 07 is X
invalid.
When Data is set to 2,

Item No. 02 ~ 06 are
invalid. Default is 0
7.1.3 SIP Peer-to-Peer
Use these programs to make SIP peer-to-peer assignments.
Number Comments
IP Phone Basic Setup - Peer Disabling 15-05-50 0 = Off

to Peer Mode results in SIP MLT 1 = On
15-05-50 Station-to-SIP MLT X
Station calls
using a DSP resource. Default is 1

Issue 5.0
7.1.4 IP Extension Numbering
Use these programs to assign IP extension numbering.
Number Comments
System Numbering Refer to the SV9100 1 = 3 Digit; Intercom

Programming Manual
for all options and
11-01-01 X
default settings.
Default is 3 Digit; Intercom

Extension Numbering Assign up to eight digits Dial (Up to eight digits)
for extension numbering
Default
11-02-01 X
Ports 1~300 = 200~499
Ports 301~960 =
5000~5659
7.1.5 SIP Extension CODEC Information
Use these programs assign SIP extension CODEC information.
Number Comments
SIP Extension CODEC Define the G.711 audio 1~4

Information Basic Setup – Frame Size 1 = 10ms
Number of G.711 Audio 2 = 20ms
84-19-01 Frames 3 = 30ms
4 = 40ms
Default is 2
SIP Extension CODEC Enable/Disable Voice 0 = Disable
Information Basic Setup – Activity Detection for 1 = Enable
84-19-02
G.711 Voice Activity G.711.
SIP Extension CODEC Define the G.711 Type – 0 = A-law
Information Basic Setup – -law is recommended 1 = -law
84-19-03
G.711 Type in USA.
Default is 0
SIP Extension CODEC Define G.711 Jitter 0~300ms
Information Basic Setup – Buffer minimum
84-19-04
G.711 Jitter Buffer (min) accepted value.
Default is 20
Information Basic Setup – Buffer setting.
84-19-05
G.711 Jitter Buffer

Issue 5.0
Number Comments

Information Basic Setup – Buffer maximum
84-19-06
G.711 Jitter Buffer (max) accepted value.
Default is 80
SIP Extension CODEC Define the G.729 audio 1~6
Information Basic Setup – Frame Size. 1 = 10ms
Frames 3 = 30ms
84-19-07 4 = 40ms
5 = 50ms
6 = 60ms
Default is 2
SIP Extension CODEC Enable/Disable Voice 0 = Disable
Information Basic Setup – Activity Detection for 1 = Enable
84-19-08
G.729 Voice Activity G.729.
Information Basic Setup – Buffer minimum
84-19-09
G.729 Jitter Buffer (min) accepted value.
Default is 20
Information Basic Setup – Buffer setting.
84-19-10
G.729 Jitter Buffer
(average) Default is 40
Information Basic Setup – Buffer maximum
84-19-11
G.729 Jitter Buffer (max) accepted value.
Default is 80
SIP Extension CODEC Define the Jitter Buffer 1 = Static
Information Basic Setup – mode – supported 2 = Not Used
84-19-17 Jitter Buffer Mode Static or Immediate. 3 = Self Adjusting
Default is 3
SIP Extension CODEC Define the VAD 0~30
84-19-18 Information Basic Setup –VAD Threshold.
Threshold Default is 20
SIP Extension CODEC Define Audio capability 0 = G.711_PT
Information Basic Setup – priority. 2 = G.729_PT
Audio Capability Priority 3 = G.722
84-19-28 4 = G.726
5 = Not Used
Default is 0

Issue 5.0
Number Comments
SIP Extension CODEC 1~4

Information Basic Setup – 1 = 10ms
84-19-33 Frames 3 = 30ms
4 = 40ms
Default is 3
SIP Extension CODEC 0~300ms
84-19-35 Information Basic Setup –
G.722 Jitter Buffer (min) Default is 30
G.722 Jitter Buffer (Average) Default is 60
G.722 Jitter Buffer (max) Default is 120
SIP Extension CODEC 1~4
Information Basic Setup – 1 = 10ms
84-19-38
Frames 3 = 30ms
4 = 40ms
Default is 3
SIP Extension CODEC 0 = Disable
84-19-39
G.726 Voice Activity Detection
Mode Default is 0
G.726 Jitter Buffer (min) Default is 30
G.726 Jitter Buffer (Average) Default is 60
G.726 Jitter Buffer (max) Default is 120
84-19-43 ---Not Used---
84-19-44 ---Not Used---
84-19-45 ---Not Used---
84-19-46 ---Not Used---
84-19-47 ---Not Used---
84-19-48 ---Not Used---

Issue 5.0
7.1.6 SIP Extension Basic Information Setup
Use these programs to assign basic SIP extension information.
Number Comments
SIP Extension Basic Define SIP station 1~65535

84-20-01 Information Setup – Registrar/ Proxy Port.
Proxy Port Default is 5070
SIP Extension Basic Define the periodic 0~65535
Information Setup – Session refresh time that allows
Timer Value both user agents and
84-20-02 proxies to determine if
the SIP session is still
active.
Default is 180
SIP Extension Basic Define to convey the 0~65535
Information Setup – Minimum minimum allowed value
84-20-03 Session Timer Value for the SIP session
timer.
Default is 180
SIP Extension Basic Define the SIP 0 = Request URI
Information Setup – Called Extension presented 1 = To Header
84-20-04
Party Info Caller ID information.
Default is 0
SIP Extension Basic Define the time out 0~3600 seconds
Information Setup – Expire response value for SIP
84-20-05
Value of Invite invite.
Default is 180
7.1.7 IP Phone Configuration
Use these programs assign IP phone configuration information.
Number Comments
IP Telephone Terminal Basic Review the type 1 = H.323

Data Setup – Terminal Type protocol support by the 2 = SIP
IP Phone. 3 = None
15-05-01 4 = DT800/DT700
Viewing Only – No
changes Default = 3
permitted.

Issue 5.0
Number Comments
IP Telephone Terminal Basic MAC Address of MAC address

Data Setup – IP Phone Fixed registered MLT SIP 00-00-00-00-00-00 to
Port Assignment phone is stored and/or FF-FF-FF-FF-FF-FF
can input the MAC
address of an MLT SIP Default is
15-05-02
phone so when it comes 00-00-00-00-00-00
online it is provided with
the extension in which
the
MAC address matches.
IP Telephone Terminal Basic Informational Only 0.0.0.0~255.255.255.255
15-05-07 Data Setup – Using IP registered IP Phones
Address Default is 0.0.0.0
IP Telephone Terminal Basic Assign the CODEC 1-Type 1
Data Setup – CODEC Type Type of the MLT SIP. 2-Type 2
3-Type 3
15-05-15 4-Type 4
5-Type 5
Default is 1
IP Telephone Terminal Basic Assign the Up to 24 characters
Data Setup – Password authentication
15-05-16 password for SIP single
line
telephones. Default not assigned
IP Telephone Terminal Basic If an adapter has one IP 0 = Disable
Data Setup – IP Duplication address coming into it 1 = Enable
Allowed Group but multiple extensions
off of it, assign all Default is 0
15-05-18 extensions to a group
so the CPU knows that
the one IP address is
assigned to multiple
extensions.
Video Mode This Program is used to 0 = Disable
select the video function 1 = Enable
with the standard SIP
terminal. If the standard Default is 0
15-05-43 SIP terminal supports
the video function, the
SV9100 transfers the
video CODEC in SDP
information.

Issue 5.0
Number Comments
IP Telephone Terminal Basic Determines whether 0 = Disable

Setup - Receiving SIP INFO SIP INFO messages 1 = Allowed any time
are received by the 2 = Allowed while RTP is
15-05-49
system. not available
Default = 1
7.2 SIP Phone Example
The following example show a SIP phone attached to an SV9100 system.
UNIVERGE SV9100
GCD-CP10
Managed Switch
VoIP
GCD-CP10 192.168.1.20
VoIP DSP: 192.168.1.21
Subnet Mask:255.255.255.0
Default Gateway:192.168.1.254
SIP Soft Phone

SIP Phone

Issue 5.0

UNIVERGE® SV9100
SV9100 NetLink Chapter 10
Introduction
NetLink Networking provides a seamless connection, using an IP network, to join

multiple SV9100 communication servers into what would appear to be a single
communications server. With a unified numbering plan, users can access any
extension in the network as if they were in the same location.
With NetLink Networking, multiple locations are linked, allowing only one operator
and a shared voice mail in the network. Up to 50 nodes (locations) can be connected
as part of the NetLink network.
With NetLink, one communication server has the main database and controls other
communication servers. The communication server is called Primary System. All call
control is basically done by the Primary System. All slots, trunks and stations belong
to the Primary System. No need to consider the numbering plan in the other
communication servers. It is easy to consider how to grab a trunk in another
communication server. There is no feature limitation when using NetLink. Any
terminal in any node operates the same as if they were located in the Primary
System.

Issue 5.0
An added feature of NetLink is the Fail-Over feature. If the Primary System is turned off
or disconnected, without the Fail-Over feature, all communication servers would stop
working. With Fail-Over, the server locates one of the Secondary Systems as based on
the SV9100 programming, and it takes over the Primary System database allowing the
other linked nodes to continue functioning.
Figure 10-1 NetLink Fail-Over Example
Primary
System
IP Network
Secondary
Secondary System 2
System 1
10-2 SV9100 NetLink

Issue 5.0
SECTION 2 SV9100 REQUIREMENTS
2.1 Hardware
Each SV9100 Communications Server needs the following items:

 GCD-CP10
 GPZ-IPLE
Install the station/trunk blades as required for your configuration.

 When installing a Secondary System in a NetLink network, and the Secondary
System has GCD-LCA blades, the GCD-LCA blades will come online and get
assigned ports before any GCD-DLCA blades.
Invalid data is displayed in the LCD of the terminal if Program 51-01 is enabled.
The VOIPDB daughter boards in the network do not need to have the same
capacity. However, any chassis which may be used as a Primary must have a
VOIPDB large enough to handle the required network resources. It is
recommended with this feature that the VOIPDB be the same as the largest
installed VOIPDB in the network.
2.2 Capacity
 A maximum of 50 nodes (connected sites).
 960 Extensions/400 Trunk ports (as allowed by hardware and licensing).
 The maximum number of ports, receivers, conference groups, etc., is the same
as stand-alone communication server.
2.3 License
 A NetLink License is required for the Primary System.
 All licenses must be registered into the Primary System.
 Refer to Section 10 Licensing Control for more details.
2.4 Protocol
TCP/IP IPv4 (IPv6 available in a future release)
2.5 Determining System ID
NetLink allows connecting a maximum of 50 communication servers. To identify

the communication server, each communication server must have one unique
number between 1 and 50.

Issue 5.0
2.6 Determining Primary System
To determine the Primary System which should be controlling the NetLink

network, the Primary System is selected based on System ID. The lowest ID
number becomes Primary System.
If you add a new communication server to NetLink once the Primary System is
already determined and functioning, the new communication server becomes a
Secondary System.
2.7 Software Information
All communication servers in the NetLink network should be at the same SV9100
software level. After the NetLink network is set up, a user can press Feature + 5
on their terminal and review the following information:
 The Terminal System ID
 Primary or Secondary Assignment
 IP Address
 System ID of Current Primary System
 Type of VoIP installed: IPLE
SECTION 3 INSTALLATION
3.1 NetLink Connection
When assigning the IP addresses to the IPLE card, the addresses must be in the
same network (subnet). If the CPU is to be connected also to the network, it
requires a separate IP address in a different network (subnet).
When you have an IPLE card attached to the CPU, the CPU NIC is no longer
required. All connections that previously terminated to the CPU NIC card can now
be terminated to the IPLE NIC.
For example, PC Pro, Web Pro, and ACD terminate to the IPLE NIC card, when
installed. Both the IPLE and CPU NIC share the same gateway assignment. The
default gateway command in Program 10-12-03 is used by both NICs, allowing
only one device, IPLE or CPU, to route outside of its own network.
If connecting a Remote site with SV8100 hardware, the hardware must first be
removed to let NetLink establish a connection. The reason for this is the Remote
site must first pull the Licensing for Hardware Migration from the Primary site.
Once NetLink has established, you can slot the SV8100 hardware. You can check
the connection status by logging in the Primary system and checking Blade
configuration.
10-4 SV9100 NetLink

Issue 5.0
General IP Configuration
The voice quality of VoIP depends on variables such as available bandwidth,

network latency, and quality of service initiatives (QoS), all of which are controlled
by the network and internet service providers. Because these variables are not in
NEC control, it cannot guarantee the performance of the users IP based voice
solution. Therefore, NEC recommends connecting the VoIP equipment through a
local area network using private IP addresses.
For a network to be suitable for VoIP it must pass specific requirements. To make
sure that the site meets these requirements, an IP ready check and a site survey
must be completed at each site before VoIP implementation.
 One-way delay must not exceed 100ms.
 Round-trip delay must not exceed 200ms.
 Packet loss must not exceed 1%
 Data switches must be manageable
 Routers must provide QOS
 Adequate bandwidth for estimated VoIP traffic (refer to 1.4.3 Bandwidth)
 Depending on how QoS policies are built in the network, assignments might be
needed in the CPU.
Some Network Considerations
Before adding the UNIVERGE SV9100 to a customer network, a detailed network

diagram of the existing network must be obtained from the customer. This
diagram provides information about any network conditions that can prevent or
hinder the VoIP equipment from functioning correctly.
Firewalls
Another regular device in customer networks that can hinder VoIP performance is
a firewall. Most corporate LANs connect to the public Internet through a firewall. A
firewall is filtering software built into a router or a stand alone server unit. It is used
to protect a LAN it from unauthorized access, to provide the network with a level
of security. Firewalls are used for many things, but in its simplest form, a firewall
can be thought of as a one-way gate. It allows outgoing packets from the local
LAN to the Internet but blocks packets from the Internet routing into the local LAN,
unless they are a response to query.
A firewall must be configured to allow specific traffic from the Internet to pass
through onto the LAN.

Issue 5.0
Figure 10-2 Two SV9100 Systems Connected Via the WAN shows two SV9100
systems. One on the corporate local LAN and one on a Remote network
connected via the WAN. The remote site cannot call the MAIN site, therefore, it is
not working.
Figure 10-2 Two SV9100 Systems Connected Via the WAN
Headquarters
WAN
Local LAN
Remote Network
Firewall Firewall
The green arrow in Figure 10-2 Two SV9100 Systems Connected Via the WAN
represents the data packets leaving the REMOTE IPLE card destined for the
SV91000 on the Headquarters LAN. The firewall on the Headquarters network is
not configured to recognize the TCP/UDP ports utilized by the NEC equipment,
thus blocking them resulting in registration failure. To solve this issue the ports
used by the NEC VoIP equipment must be opened in the firewall allowing the
NEC traffic to pass through onto the SV9100.
The ports, 58000 and 58002 (TCP) for signaling and the voice ports, are required
to be open at each location. This depends on how many IPLE ports are installed.
 IPLE 256 Open UDP Ports 10020~10531
VPN
Another common feature is to use the Internet as the WAN between customer
locations. When this is done VPNs are typically used between the locations. A
VPN (Virtual Private Network) is a private data network that maintains privacy
through the use of tunneling protocols and security features over the public
internet. This allows remote networks (with private addresses), residing behind
NAT routers and/or firewalls, to communicate freely with each other. When
building the VPN tunnels, throughout the network, they must be assigned as a
fully meshed network. This means that every network is allowed direct connection
to each and every other network in the topology.
10-6 SV9100 NetLink

Issue 5.0
network is not fully meshed due to the lack of a VPN tunnel between Sites B and
C.
Figure 10-3 VPN Network
SITE B
VPN Tunnel
Between Sites A and B
SITE A
IPLE
IPLE Signaling Address = 192.168.15.5/
24
Internet IPLE DSP(1) = 192.168.15.6/24
IPLE
IPLE
IPLE Signaling Address = 192.168.15.5/
24
IPLE DSP(1) = 192.168.15.6/24
VPN Tunnel SITE C
Between Sites A and C IPLE Signaling Address = 192.168.20.5/
24
IPLE DSP(1) = 192.168.20.6/24
Site A can communicate with phones on sites B and C. Phones on sites B and C
cannot communicate with each other. To correct this issue, another VPN
connection between sites B and C is required.

Issue 5.0
NetLink with IP Phones connected via NAPT
Figure 10-4 NetLink with IP Phones connected via NAPT
MTU Size
In some network environment the MTU size of CCPU or IPL NIC may need to be
changed. With version 2.00 or lower the MTU size was fixed to 1500 for both
CCPU & IPL NIC. With Version 3.00 or higher now the MTU size can be changed
for both CCPU & IPL NIC.
If data to be send is greater than the defined MTU, the data will be transmitted into two
or more packets as per defined MTU.
Conditions
 MTU size defined in PRG 10-12-18 is applied on the packets transmitted from
IP address define in PRG 10-12-01.
10-8 SV9100 NetLink

Issue 5.0
 MTU size defined in PRG 10-12-19 is applied on the packets transmitted from
IPL IP address define in PRG 10-12-09.
 MTU size value is applied after logging out from WebPro, PcPro or TelPro.
 MTU is not applied on RTP / RTCP packets generated from DSP of VoIPDB.
Related Programming
10-12-18 GCD-CP10 Network Setup-CCPU MTU

 Define MTU for the packets sent from IP address define in 10-12-01.
10-12-19 GCD-CP10 Network Setup-IPL MTU

 Define MTU for the packets sent from IP address define in 10-12-09.
3.2 Limitations
3.2.1 Features
Synchronous Ringing via NetLink is not supported.
3.2.2 IP Terminal
All DT800/DT700 terminals can register to either the primary or a secondary

system. Softphones and wireless LAN phones must still register to the
primary system.
The SV9100 Encryption feature is only supported in the primary system.
3.2.3 IP Trunks
When configuring SIP Trunks on Netlink Secondary system, only SIP Profile
One is supported.
3.2.4 General limitations

 SV8100 CPU is not supported in a Netlink environment with SV9100
CPU.
 The Primary System (Main Site) requires the appropriate NetLink

licenses dependant upon the number of nodes in the NetLink
network.
 Up to 50 Nodes can be supported in a NetLink network.
 When the attendant's telephone exists on a secondary system, alarm

information cannot be displayed on the attendant's telephone.
 When Fail Over occurs between the Primary System and two or more
Secondary Systems, the attendant telephone displays the System ID
of the system that went into Fail-Over last.
 A maximum of 240 Virtual slots are supported.

Issue 5.0
 Port assignment is performed sequentially by the requested order

from the Secondary Systems.
 All nodes in a NetLink network should have the same main

GCD-CP10 software.
 When a Terminal is placed on hold, the Music on Hold comes from

the system where the Terminal resides.
 When a trunk is placed on hold, the Music on Hold comes from the
system where the trunk resides.
 External Paging uses an output on the GCD-CP10 of the Primary

System.
 A PGDAD must be used if External Paging is required in the

Secondary Systems.
 License information in the Primary will be copied to the Secondary

site when doing database duplication.
 Duplicate license information in the Secondary System is available for

only 28 days.
 After 28 days the license expires. To renew the license, a connection

to the original Primary site must be re-established. (Once the
connection to the Primary is recovered, if fail-over occurs again, the
license is once again enabled for the specified duration based on
software version above.)
 If the original Primary site is in the NetLink network as a temporary

secondary, the license information is available.
 If a user wants to enter another additional feature license, it needs to

be entered on the original Primary System.
 When fail-over occurs, the Primary System is changed to another

communication server. The IP applications do not know the new
primary IP Address, so the following features are disabled after fail-
over:
 SMDR
 ACD-MIS
 SIP Terminal
 Soft Phone
 IP K-CCIS
10-10 SV9100 NetLink

Issue 5.0
 The following Programs are not updated by the Primary System

during fail-over:
Table 10-1 The Data Which Isn’t a Target of a Replication
Program Number Name

10-01 Date and Time Setup
10-02 Location Setup
10-12 CCPU Network Setup
10-13 In-DHCP Server Setup
10-14 Managed Network Setup
10-15 Client information Setup
10-16 Option information Setup
10-23 IP System Interconnection Setup
10-28 SIP System Information Setup
10-29 SIP Server Information Setup
10-33 SIP Registrar/Proxy Information Basic Setup
10-36 SIP Trunk Registration Information Setup
10-37 UPnP Setup
10-62 NetBIOS Setting
10-72 Network Security Setup
20-66 STUN/TURN Server Setting
21-19 IP Trunk(SIP) Calling Party Number Setup for Extension
51-01 NetLink Settings
84-09 VLAN Setup
84-13 SIP Trunk CODEC Information Basic Setup
84-14 SIP Trunk Information Basic Setup
90-01 Installation Date
90-08 System Reset
90-09 Automatic System Reset Time Setup
90-68 Side Tone Auto Setup
90-77 LAN Link Speed Information
93-01 Day/Night Mode Information
93-02 Trunk Information
93-03 Extension Information
93-04 Redial List
93-05 Department Group Information
93-06 IP Address list for 1st Party CTI connection

Issue 5.0
 Data in SRAM area is not transferred to the Secondary Systems

during fail-over, therefore when fail-over occurs DND and Caller ID
History may be lost.
 Mobile Extension is supported only when using the PRI installed in

the Primary System.
 When a Secondary System with an ETIA or RTB assigned is added to

a NetLink network, the Primary Systems database does not replicate
the data in Programs 10-55 or 90-61.
 When External MOH is assigned, the AUX1 or AUX2 on the front of

the GCD-CP10 can only be utilized at the Primary Site. All Secondary
sites must provide the External MOH input via an ACI Input [PGDAD].
 A DT800/DT700 may be registered to either the primary or a

secondary NetLink system. When NetLink is turned on, the system
refers to programs 51-17-01 and 51-17-02 for DT800/DT700 registrar
information.
 With both the Primary Site and Secondary Sites can have their own
local MOH source connected to the AUX1 or AUX2 on the front of the
GCD-CP10.
 With the T-1 CCTA blade is supported in the primary system and/or
secondary systems.
 When using Mobile Extension in a NetLink Network, the ISDN/PRI

must be used in the Primary System.
 InMail is supported in a NetLink network; however, replication should

be scheduled for non-peak hours of operation.
 InMail is supported for centralized voice mail in a KTS to KTS CCIS

network.
 It is possible to set Register ID for trunks that belong to that specific

system. For example, a Register ID set in the Primary system cannot
be assigned to a trunk in the Secondary system. The allocation of the
trunk and Register ID of Program14-12 must be in the same system.
 Assigned SIP trunk ports will be cleared after establishing NetLink.
 The following programs no longer replicate and can be set on a per

system basis: PRG10-23, 10-37, 21-19, 84-13, and 84-14.
 When using multiple SIP trunk carriers, programs 10-28, 10-29, 10-
30, and 10-23 will have to be set at each system via WebPro or
PCPro (does not replicate). Program 10-68 can be accessed via
system ID.
 When using multiple SIP trunk carriers, Program 51-19 must be

enabled for extensions on a secondary system to use CPN
information from program 21-19.
 When using multiple SIP trunk carriers, SIP trunk CODEC setup must
be done from each system (does not replicate).

Issue 5.0
 When using multiple SIP trunk carriers, SIP trunk basic setup must be
done from each system (does not replicate).
 Registered SIP trunks can be utilized by any system in the NetLink

network, as long as trunk route programming allows it.
 When using InMail in a CCIS or Netlink network, 8-digit extensions

and mailboxes are not supported.
 When using Set Relocation, and a terminal is re-located from one

physical system to another physical system, route programming must
be made accordingly for 911 calls.
 FoIP (Fax over IP) is supported in a NetLink Network.
 For a network to be suitable for VoIP it must pass specific

requirements. To ensure the site meets these requirements, an IP
ready check and a site survey must be completed at each site before
VoIP implementation.
 One-way delay must not exceed 100ms
 Round Trip delay must not exceed 200ms
 Packet loss must not exceed 1%
 Data switches must be manageable
 Routers must provide QOS
 Adequate bandwidth for estimated VoIP traffic
 Depending on how QoS policies are built in the network,
assignments may be needed in the CPU.
Table 10-1 VoIP Resource Chart
Primary System Secondary 1 Secondary 2
TDM IP Terminal CO IP TDM CO TDM CO

Terminal (DT800/ Analog Trunk Terminal Analog Terminal Analog
DT700) /Digital /Digital /Digital
TDM 0 P:1 0 P:1 P:1 P:1 P:1 P:1

Primary Terminal S1:1 S1:1 S2:1 S2:1
System
IP Terminal P:1 0 P:1 P:1 S1:1 S1:1 S2:1 S2:1
(DT800/
DT700)
Secondary TDM P:1 S1:1 P:1 P:1 0 0 P1:1 S1:1

Terminal S1:1 S1:1 S1:1 S2:1 S2:1
System 1
Secondary TDM P:1 S2:1 P:1 P:1 S1:1 S1:1 0 0

Terminal S2:1 S2:1 S2:1 S2:1 S2:1
System 2
P = Primary
S1 = Secondary System #1
S2 = Secondary System #2

Issue 5.0
 The number of conference blocks in a NetLink network is the same as

a stand-alone system.
 Invalid data is displayed in the LCD of the terminal if Program 51-01

is enabled.
 The SV9100 can recognize what system that each DT800/DT700

extension(s) are connected to and provide an ARS COS based on the
System (System ID) when using NetLink.
 When NetLink is enabled synchronous ringing (PRG 14-02-17)

automatically is disabled. Synchronous ringing is not supported in a
NetLink environment.
Restrictions:
 The number of total ports depends on the Primary System.
 System ID (Program 51-01-02) must be unique for each system in a

NetLink network.
 ACD/MIS can connect to the Primary (Main) site only.
 Only one Voice Mail can be installed in a NetLink network.
 In-Mail and VRS use the VMDB of the Primary (Main) site.
 APSU can be installed in the Secondary (Remote) System, however

the NetLink time zone follows the Primary (Main) System.
 When installing a Secondary System in a Netlink network and the

Secondary System has GCD-LCA blades installed, the GCD-LCA
blades will come online and assign ports before any GCD-DLCA
blades are assigned.
 Secondary systems must follow the primary CPU software settings

for Mu-law/A-law within the country where the primary system is
located. SV9100 NetLink connections are only supported within the
same countries/areas. For example, the SV9100 can be connected
via NetLink between the US and Canada, however the SV9100
cannot be connected via NetLink with systems in other countries
(e.g., Mexico or the U.K.).
 Synchronous Ringing via NetLink is not supported.
 Always connect to the Primary System when using PCPro.
 When using K-CCIS - T1 in a Netlink system, the CCTA blade is only

supported in the primary system.
 The following programs require a reset after making a change using

PCPro, WebPro or Handset programming:
Table 10-2 CPU Reset Programs
Program When Changed Using
10-12-01 Handset, WebPro, PCPro

Issue 5.0
Table 10-2 CPU Reset Programs
Program When Changed Using
51-15 WebPro, Handset Programming
90-04 Handset Programming
10-46-07 PcPro
84-23 Handset, WebPro, PCPro
3.3 DSP
(PB Receiver, Dial Tone Detector, Caller ID Detector, Caller ID Sender, MF

Receiver).
 A DSP is used on the system GCD-CP10 which has the call. For instance, when
a trunk on System ID 2 uses a receiver, a DSP on System ID 2 is used.
 The Primary System must control DSP resources on the Secondary Systems
but it cannot control all of them.
There are five zones to handle DSP resources in NetLink. The zone number
depends on the System ID number.
Zone # System ID
1 1, 6, 11, 16, 21, 26, 31, 36, 41, 46
2 2, 7, 12, 17, 22, 27, 32, 37, 42, 47
3 3, 8, 13, 18, 23, 28, 33, 38, 43, 48
4 4, 9, 14, 19, 24, 29, 34, 39, 44, 49
5 5, 10, 15, 20, 25, 30, 35, 40, 45, 50
In the same Zone, the communication servers must share the DSP resource
number. For instance, if a user in System ID-1 is using DSP resource #1,
resource #1 becomes busy in Zone #1 (System ID 1, 6, 11, 16, 21, 26, 31, 36,
41, 46).
 All communication servers follow the DSP Setting in Program 10-09 in the
Primary System.

Issue 5.0
 In a NetLink environment the Maximum system capacity of analog extensions

changes from 368 to 896. However, the following limitations apply:
Each NetLink Node supports a maximum of 144 DTMF receivers which these
analog stations will use. Analog trunks may also use these same receivers, this
means that a total of 144 analog devices may be used at one time per NetLink
Node.
3.4 E911
A trunk is required for each communication server for 911 calls and the ARS
routing table must be set up to seize the proper trunk. If no DSP resource is
available, one of the active DSPs is dropped and the communication server uses
that DSP for the 911 call.
3.5 Music on Hold

 When External MOH is assigned, the AUX1 or AUX2 on the front of the
GCD-CP10 can only be utilized at the Primary Site. All Secondary sites must
provide the External MOH input via an ACI Input [PGDAD ].
 Both the Primary Site and Secondary Sites can have their own MOH source
connected to the CN8 or CN9 on the front of the GCD-CP10. Same rules below
apply.
 If the external MOH setting is defined in the Primary System, each

communication server needs to have an external MOH source.
MOH MOH
Primary Secondary
Hold
When a terminal is placed on hold, MOH comes from the

communications server in which the terminal resides.

Issue 5.0
MOH MOH
Trunk
Primary
Secondary
Hold
When a terminal is placed on hold, MOH comes from the

communications server in which the terminal resides.
Secondary
Primary
External Paging
 External Paging uses an output on the GCD-CP10 of the Primary System.
 A PGDAD must be used if External Paging is required in the Secondary Systems.
3.6 Bandwidth
The following example shows the approximate bandwidth used for the NetLink
signaling:
 Bandwidth signaling for 1 call = 4kb
 Bandwidth of 1 voice call (G.711, VIF=20ms) = 90kbps
If there are 100 calls (average talk time of five minutes) between the linked
communication servers for one hour:

Issue 5.0
 Signaling data total is 4kb x 1000 calls divided by 3600 seconds = 1.11kbps
 Voice data total is 1000 calls divided by 60/5 x 90Kbps = 7.5Mbps
Using the above calculation, the signaling data is smaller than the voice data. So,
in determining the bandwidth needed, using the voice bandwidth number for both
the voice and data allows adequate bandwidth.
3.7 Conference Block

 The number of conference blocks in NetLink is the same as with a stand-alone
communication server.
3.8 Voice Mail

 The InMail must be installed in the Primary System.
 Only one Voice Mail or InMail can be installed in NetLink network.
 Voice Mail can be installed in the Primary or Secondary System.
 For InMail remote CCIS extensions are not supported in a centralized directory.
3.9 FoIP
 FoIP (Fax over IP) is supported in a NetLink network.
SECTION 4 LIST OF SUPPORTED FEATURES IN SECONDARY SYSTEM
The following table lists supported features.
Table 10-3 Supported Features in Secondary System
Supported in
Feature Name Secondary Comments
System
Account Code – Forced/Verified/ Yes

Unverified
Account Code Entry Yes
Alarm Yes
Alarm Reports Conditions Programs 90-10-80 ~ 90-10-95 are not supported.
Alphanumeric Display Yes

Analog Communications Interface Yes
(ACI)
Ancillary Device Connection Yes
Answer Hold Yes

Issue 5.0
Table 10-3 Supported Features in Secondary System (Continued)
Supported in
System
Answer Key Yes

Attendant Call Queuing Yes
Automatic Call Distribution (ACD) Conditions Fail-Over: VRS card must be installed in Primary
system.
MIS monitor must be connected to the Primary system.
If you are logged in and Fail-over occurs the agent will
not be logged back in when the Secondary system
takes over. The agent will have to manually log back in
to continue taking calls.
Automatic Release Yes
Automatic Route Selection Conditions Fail-Over: ARS Class Of Service must be
programmed for each System in a NetLink Network to
ensure proper call routing during Fail-Over.
Background Music Conditions If using Background Music, each site must provide its
own music source. Primary and Secondary Systems
cannot share a source. If the Primary System is set to
Music off of the GCD-CP10, then all Secondary
Systems also must get their music off of the
GCD-CP10.
Barge-In Yes
Battery Backup – System Memory N/A
Battery Backup – System Power N/A
Call Appearance (CAP) Keys Yes
Call Arrival (CAR) Keys Yes
Call Duration Timer Yes
Call Forwarding Yes
Call Forwarding – Centrex Yes
Call Forwarding – Park and Page Yes
Call Forwarding with Follow Me Yes
Call Forwarding, Off-Premise Yes
Call Forwarding/Do Not Disturb Yes
Override
Call Monitoring Yes
Call Redirect Yes
Call Waiting/Camp-On Yes

Issue 5.0
Supported in
System
Callback Yes
Caller ID Yes
Caller ID Call Return Yes
Central Office Calls, Answering Yes
Central Office Calls, Placing Yes
Class of Service Yes
Clock/Calendar Display Yes
CO Message Waiting Indication Yes
Code Restriction Yes
Code Restriction Override Yes
Code Restriction, Dial Block Yes
Conference Yes
Conference, Voice Call/Privacy Yes
Release
Continued Dialing Yes
Cordless DECT Terminals Yes
Cordless Telephone Connection Yes
Data Line Security Yes
Delayed Ringing Yes
Department Calling Yes
Department Step Calling Yes
Dial Pad Confirmation Tone Yes
Dial Tone Detection Yes
Dialing Number Preview Yes
Digital Trunk Clocking Yes
Direct Inward Dialing (DID) Yes
Direct Inward Line (DIL) Yes
Direct Inward System Access (DISA) Yes
Direct Station Selection (DSS) Yes
Console

Issue 5.0
Supported in
System
Directed Call Pickup Yes

Directory Dialing Yes
Distinctive Ringing, Tones and Flash Yes
Patterns
Do Not Disturb Yes
Door Box Yes
Drop Key Yes
Dterm Cordless II Terminal Yes
Dterm Cordless Lite II Terminal Yes
Facsimile CO Branch Connection Yes

Flash Yes
Flexible System Numbering Yes
Flexible Timeouts Yes
Forced Trunk Disconnect No
Group Call Pickup Yes
Group Listen Yes
Handset Mute Yes
Handsfree and Monitor Yes
Handsfree Answerback/Forced Yes
Intercom Ringing
Headset Operation Yes
Hold Yes
Hot Key-Pad Yes
Hotel/Motel Yes
Hotline Yes
Howler Tone Service Yes
Intercom Yes
IP Multiline Station (SIP)
IP Single Line Telephone (SIP) Yes All IP Terminals connect to the Primary System.
IP Trunk - (SIP) Session Initiation Conditions Only SIP Profile One is supported in Secondary
Protocol systems.

Issue 5.0
Supported in
System
IP Trunk - H.323 Conditions All IP Trunks connect to the Primary System.
ISDN Compatibility Yes

K-CCIS - IP Conditions All IP Trunks connect to the Primary System.
Last Number Redial Yes

Licensing Conditions All licenses must be registered into the Primary
System. During Fail-over, the Secondary System will
contain all licenses for seven days.
Line Preference Yes
Long Conversation Cutoff Yes
Maintenance Conditions Web Pro and PC Pro can be used to view the individual
Card Configuration screens of all Systems in a NetLink
Network, however the System Data is for all Systems
combined.
Meet Me Conference Yes
Meet Me Paging Yes
Meet Me Paging Transfer Yes
Memo Dial Yes
Message Waiting Yes
Microphone Cutoff Yes
Mobile Extension Conditions Mobile Extension is only supported when using the
ISDN/PRI installed in the Primary System.
Multiple Trunk Types Yes
Music on Hold Conditions Refer to Section 3.5, Music on Hold.
Name Storing Yes

Night Service Yes
Off-Hook Signaling Yes
One-Touch Calling Yes
Operator Yes
(OPX) Off-Premise Extension Yes
Paging, External Conditions Refer to Section 3.5, Music on Hold.
Paging, Internal Yes

Park Yes

Issue 5.0
Supported in
System
PBX Compatibility Yes

PC Programming Conditions WebPro and PCPro can be used to view the individual
Card Configuration screens of all Systems in a NetLink
Network, however the System Data is for all Systems
combined. You must be connected to the Primary
System.
Power Failure Transfer Yes
Prime Line Selection Yes
Private Line Yes
Programmable Function Keys Yes
Programming from a Multiline Yes
Terminal
Pulse to Tone Conversion Yes
Redial Function Yes
Remote (System) Upgrade Yes
Repeat Redial Yes
Resident System Program Yes
Reverse Voice Over Yes
Ring Groups Yes
Ringdown Extension, Internal/ Yes
External
Room Monitor Yes
Save Number Dialed Yes
Secondary Incoming Extension Yes
Secretary Call (Buzzer) Yes
Secretary Call Pickup Yes
Selectable Display Messaging Yes
Selectable Ring Tones Yes
Serial Call Yes
Single Line Telephones Yes
Softkeys Yes
Speed Dial – System/Group/Station Yes

Issue 5.0
Supported in
System
Station Hunt Yes

Station Message Detail Recording Yes
Station Name Assignment – User Yes
Programmable
Station Relocation Conditions When using Set Relocation, and a terminal is relocated
from one physical system to another, route
programming must be made accordingly for 911 calls.
SV9100 Communications Analyst Conditions The SV9100 Communications Analyst Server must
Enterprise connect to the Primary System.
SV9100 UC Suite Applications Conditions The SV9100 UC Suite must connect to the Primary
System.
SV9100 Internal Router Yes
SV9100 PoE Gigabit Switch Yes
SV9100/SV9300 Terminals Yes
Synchronous Ringing No
Tandem Ringing Yes
Tandem Trunking (Unsupervised Yes
Conference)
TAPI Compatibility Yes
Tone Override Yes
Traffic Reports No
Transfer Yes
Trunk Group Routing Yes
Trunk Groups Yes
Trunk Queuing/Camp-On Yes
UCB (Unified Communications for No UCB is not supported with the SV9100.
Business)
UM8000 Mail Yes
Uniform Call Distribution (UCD) Yes
Uniform Numbering Network Yes
UNIVERGE Multimedia Conference Yes
Bridge
Universal Slots Yes

Issue 5.0
Supported in
System
User Programming Ability Yes

Virtual Extensions Yes
InMail Yes
Voice Mail Integration (Analog)
Voice Mail Message Indication on Yes

Line Keys
Voice Over Yes
Voice Response System (VRS) Yes
Volume Controls Yes
Warning Tone For Long Conversation Yes

Issue 5.0
SECTION 5 NETLINK FUNCTIONS
5.1 Programming
The following programs require a GCD-CP10 reset after making a change via
PCPro, Web Pro and handset programming.
Program When Changed via
10-12-01 Handset, Web Pro, PCPro

51-15 Handset, Web Pro
10-46-07 PCPro
84-23 Handset, Web Pro, PCPro
 When Program 90-08-01 System Reset is performed from any site in a NetLink
Network, the Primary System will be reset.
Minimum Required Programming

 10-12-09 : GCD-CP10 Network Setup – VoIP Daughter Board IP Address
Set the LAN IP address VoIP daughter board. The IP number increases by one
for each increase in the slot number.
 10-12-10 : GCD-CP10 Network Setup – VoIP Daughter Board Subnet Mask
Define the subnet mask for the channel VoIP daughter board. The setting of
Subnet Mask errors when all Host Addresses are 0.
 51-01-01 : NetLink Settings – NetLink System ID
This is the ID number (0 = Disabled, 1-50) that identifies each SV9100 in the
NetLink network. Each SV9100 must have a unique number in the network.
When this option is set to 0, NetLink is disabled.
 The SV9100 must be reset when changing this option.
 51-03-01 : NetLink IP Address List – IP Address
The communications server sees the Primary SV9100 from in this list. When no
Primary SV9100 is seen or Fail-Over occurs, the Node List is referred to
establish a new link. This setting is necessary when Program 51-01-03 is set to
0 or Program 51-05-02 is other than 0.
Enter IP addresses for any communications server which is included in the

Issue 5.0
network (especially the Primary SV9100). Once the communications server

connects to the Primary SV9100, this setting is updated by the Primary SV9100
when Program 51-13-01 is enabled. This allows any new or changed SV9100s
to be added automatically.
An IP address cannot be defined more than once.
Required Programming for the Secondary System Setup

 51-04-01 : NetLink Top Priority Primary System IP Address – IP Address
Enter the IP address of the Primary SV9100. This setting is needed to use the
Primary SV9100 Automatic Integration Feature (Program 51-06-01). If the
secondary flag is set in Program 51-01-03 secondary SV9100s connect with
this IP address.
 51-05-01 : NetLink Timer Settings – Keep Alive Forwarding Interval
Set the interval (1~3600) for the secondary SV9100 to send the Keep Alive
packet to the Primary SV9100 to confirm communication.
 51-05-02 : NetLink Timer Settings – Keep Alive Response Waiting Time
The secondary SV9100 waits this time (0, 5~10800) for response to the Keep
Alive packet to confirm communication with Primary SV9100. If no response is
received, Fail-Over occurs. A setting of 0 disables this and the SV9100 waits
indefinitely for a response from the Primary SV9100. For Fail-Over to occur, this
time must be set to an entry other than 0.
 51-05-03 : NetLink Timer Settings – Primary System Seeking Interval
This timer determines the time between packets the SV9100 waits when
searching for a new Primary SV9100 with the Fail-Over feature (1~3600).
 51-05-04 : NetLink Timer Settings – Primary System Seeking Time
Set the time (5~10800) between packet sending when the SV9100 is reviewing
priority levels for a new higher-priority Primary SV9100.
 51-05-05 : NetLink Timer Settings – Top Priority Primary System Seeking
Interval
When the current Primary SV9100 is not the Top Priority Primary SV9100, the
SV9100 sends packets at this interval (1-3600) to check if a higher priority
SV9100 exists.
 51-05-06 : NetLink Timer Settings – Enforced Primary System Seeking
Time
When the Forced Change Primary command is executed, the SV9100 searches
for the new Primary SV9100 for this time (1~10800).
 51-05-07 : NetLink Timer Settings – Socket Refresh Time
If the IP connection becomes unstable and communication is lost (keep-alive
function does not work), the SV9100 retries the connection at this time
(20~3600).
 51-06-01 : NetLink Automated Primary System Integration – Automated
Primary System Integration
When Fail-Over occurs, multiple Primary SV9100s may be established. When
the connection is recovered, with this option enabled (1), the NetLink feature is
automatically reestablished around the top priority Primary SV9100.
 51-06-02 : NetLink Automated Primary System Integration – System
Package Reset Method
When the Primary Automatic Integration reestablishes the NetLink network, the

Issue 5.0
blades in the secondary SV9100s are reset. This option determines if the
secondary SV9100 blades are reset only when all blades are idle (0) or anytime
(1).
Other Feature Programming
Refer to the specific NetLink functions which follow for additional programs.
 10-01-01 ~ 10-01-07 : Time and Date
Change the SV9100 Time and Date through SV9100 programming. Extension
users can also dial Service Code 828 to change the Time if allowed by an
extension Class of Service.
 51-01-02 : NetLink Settings – Primary Candidate Order
When the Primary SV9100 is turned off or disconnected from the network, this
value (1~50) is used to select the new Primary SV9100 with the Fail-Over
feature. The smaller the number, the higher the priority for the SV9100. When
the value for two SV9100s is the same, lower NetLink System ID (51-01-01) is
selected as the Primary SV9100.
 51-01-03 : NetLink Settings – Primary System Selection Method
The link between SV9100s is established based on this setting (0, 1). When
enabled, the SV9100 connects with the top priority Primary SV9100 (address
set in Program 51-04-01). If the SV9100 is not found within the defined time
(Program 51-05-02), the Fail-Over feature of the SV9100 searches for the
Primary SV9100 as if this option were set to 0.
The link between SV9100s is dynamically established based on the node list set
in Program 51-03-01. The Primary SV9100 is selected in the order in which the
SV9100s wake up.
 51-02-01 : NetLink System Basic Setup – System Name
Set the desired site name (up to 20 characters) for ease of maintaining
information. Once the SV9100 is connected to the Primary SV9100, this setting
is updated by the Primary System data.
 51-02-02 : NetLink System Basic Setup – Time Zone – Hour
Determine the offset hours from the Primary SV9100. This setting affects the
time display on display telephones (0~24 = -12 ~ +12 hours).
 51-02-03 : NetLink System Basic Setup – Time Zone – Minute
Determine the offset minutes from the Primary SV9100. This setting affects the
time display on display telephones (0 ~ 120 = -60 ~ +60 minutes).
 51-02-04 : NetLink System Basic Setup – System Authentication MAC
Address
When Program 51-13-03 is enabled, the SV9100 checks this MAC address
against the MAC address of the connecting CCPU. If the value is different, the
connection is refused.
 51-07-01 : NetLink Enforced Primary System Integration – System
When Fail-Over occurs, you can manually change the Primary SV9100 using
Program 51-08, if this option is enabled. Program 51-06-01 must be set to 0.
When the Forced Change of Primary Settings is performed, the blades in the
secondary communication servers are reset. This option determines if the

Issue 5.0
(1).
 51-08-01 : Primary NetLink Setting – IP Address of New Primary System
When forcing the communications server to update to a new Primary SV9100,
the communications server using the IP address defined here as the new
Primary.
 After a Forced Change of Primary SV9100 is done, this entry is erased.
 51-08-02 : Primary NetLink Setting – System ID of New Primary System
If you already have an IP address registered in Program 51-11-03, you can
execute a Forced Change Primary SV9100 by entering the system ID. If this ID
is set to 0, the Top Priority SV9100 is selected as the new Primary.
 51-09-01 : NetLink TCP Port Settings – Primary to Secondary System
Communication Port
Define the port the Primary SV9100 uses to communicate with the Secondary
SV9100 (0~65535).
 51-09-02 : NetLink TCP Port Settings – Secondary to Secondary System
Communication Port
Define the port used to communicate between other networked communications
servers (0~65535).
 51-09-03 : NetLink TCP Port Settings – Secondary to Primary System
Communication Port
This setting defines the port used by the Secondary SV9100 to communicates
to the Primary SV9100 (0~65535). If 0 is entered, the port is selected
dynamically.
 51-09-04 : NetLink TCP Port Settings –Fail-Over Communication Port
When Fail-Over occurs, each SV9100 communicates with the other
communication servers using the port number specified in this entry (0~65535).
If 0 is entered, the port is selected dynamically.
If an entry other than 0 is made, up to 50 ports (depending in the number of
networked systems) are continuously used from the specified port number. (Ex:
If 5000 is entered, 5001~5049 are used.)
 51-09-05 : NetLink TCP Port Settings – Top Priority Primary System
Seeking Port
Enter the port number to search for the Top Priority Primary SV9100. If 0 is
entered, the port is selected dynamically (0~65535).
 51-09-06 : NetLink TCP Port Settings – System Data Replication
Communication Listening Port
Define the listening port used so that the Secondary SV9100 can replicate the
Primary SV9100 database (0~65535).

Issue 5.0
 51-09-07 : NetLink TCP Port Settings – System Data Replication Primary

Detection Port
Define the port used for communication so that the Primary SV9100 may
synchronize the Secondary SV9100 with the program data (0~65535). If 0 is
entered, the port is selected dynamically.
 51-10-01 : NetLink Virtual Slot Settings
View the remaining number of slots which can be controlled by NetLink. The
NetLink feature can control up to 240 virtual slots maximum. (The physical slots
in the NetLink network are maintained as virtual slots by the SV9100.) This
option is not user-definable and is view-only.
 51-11-01 - 51-11-06 : NetLink System Information
View the SV9100 information for communication servers connecting with the
Primary SV9100. These options are not user-definable and are view-only.
 51-12-01 - 51-12-06 : NetLink Primary System Information
View the information for the Primary SV9100 in a NetLink network. These
options are not user-definable and are view-only.
 51-13-01 : NetLink Options – Automatic IP Address List Update
If this option is enabled (1), the Internet Protocol address list is updated to
include the IP address of a Secondary SV9100 upon connection.
 51-13-02 : NetLink Options – Time Zone Enhancing
When enabled (1), the time zone is applied to the following items: LCD Clock
Display, Caller ID History, VRS Time Announce, Time and Date Set by Service
Code, Alarm Clock, Hotel Mode Wake-Up Call (time announce included).
When disabled (0), the time zone is only applied to the LCD Clock Display and
Caller ID
History.
Program 51-02-02 must also be set for this option.
 51-13-03 : NetLink Options – MAC Address Authentication
When enabled (1), connection authentication of the SV9100 is done with the
MAC address set in Program 51-02-04. The system compares its own MAC
address and if the address does not match, the Primary System rejects the
connection.
 51-14-01 : NetLink System Control – Delete System Information
Delete the SV9100 and slot information for a particular communications server
using the NetLink feature. To use this program, the communications server must
not be connected to the network.
 51-15-01 : Easy Set Command
Automatically set the minimum settings for the NetLink feature. This program
sets up to four communication servers and applies the minimum settings
required to Programs 10-12-09, 51-01-01, 51-03-01, 51-04-01, and 51-05-02.
 51-16-01 : NetLink System Data Replication Mode Setting – System Data
Replication Mode
Set the replication mode (0 = Disabled, 1 = Setting Time Mode, 2 = Interval
Mode). An entry of‘1 replicates the data at the time set in 51-16-02. If this option
is set to 2, replication occurs at the interval set in 51-16-03.
Replication Time Setting

Issue 5.0
If Program 51-16-01 is set to 1, set the time to replicate the program data
(0000~2359).
Replication Interval Setting
If Program 51-16-01 is set to 2, set the interval time between replicating the
program data (15-1440 minutes).
 51-16-04 : NetLink System Data Replication Mode Setting – Replication
Time Stamp
This program displays the last time the program data was replicated. This is
automatically updated when the replication occurs. This option is view-only.
Replication Wait Time
When a NetLink network is created, this option determines the time the SV9100
waits until replication is started (1~86400 seconds).
Replication Interval
Set the time to start replication to the next node after replication has completed
to one node (1~86400 seconds).
 51-17-01 : NetLink DT800/DT700 Server Individual Information Setup –
Registrar Port
This is the SIP registrar port of each system when NetLink is used. Open range
is 0-65535, default is 5080.
 51-17-02 : NetLink DT800/DT700 Server Individual Information Setup –
Subscribe Session Port
This is the SIP subscribe session port number for each system when NetLink is
used. Open range is 0-65535, default is 5081.
 51-18-01 : NetLink Config Setting – NetLink Fail-Over Limit
When tear-down of network is repeated more than the specified times, NetLink
is operated in stand-alone. Input range is 0, 2~10 (0 = No limit). Default is 0.
 51-19 : NetLink IP Trunk(SIP) Calling Party Number Setup for Extension
This program assigns transmission of Calling Party Number from PRG21-19 for
each secondary system. The transmission applies for every extension.
SECTION 6 FAIL-OVER
With NetLink, a connection to the Primary System may be lost (disconnected or

powered down) from the TCP/IP connection. When this occurs, there could be a loss
of control at all Secondary Systems.
The Fail-Over feature works when NetLink has lost connection to the Primary System
for longer than the defined keep-alive timer. It finds one of the Secondary Systems to
take over the Primary Systems database so that NetLink can still work. The
networked communication servers reboot (this takes approximately 30 seconds), and
when they come back up are linked to the new Primary System.

Issue 5.0
The Primary System is selected based on the Primary Candidate Order and System
ID. The decision for the new Primary System is based first on the value of the
Primary Candidate Order. The lower the number, the higher the priority. If the value is
the same number, the System ID is then checked.
Netlink has been enhanced with PRG 51-18-01 (Netlink Fail Over Limit). This setting
allows the system to be programmed for the maximum number of specified times that
Fail Over occurs before the Secondary Systems remain in Stand Alone mode (Own
Primary). Settings are 0: Infinity and 2~10. The following is an example when PRG
51-18-01 has a setting of “4”.
Figure 10-5 PRG 51-18-01 with setting of “4”
Once the specified number of Fail-Over times have been reached, the Secondary
Systems will remain in stand alone mode until the Forced Change Primary System
Enabling (PRG-51-07-01) and IP Address of New Primary System (PRG-51-08-01)
are performed.
If the communication server database is set up as follows, System ID 2 becomes the

new Primary System.
System ID Primary Candidate Order

1 30
2 30
3 30
4 30

Issue 5.0
If communication server database is set up as follows, System ID 4 becomes the

new Primary System.
System ID Primary Candidate Order

1 30
2 30
3 30
4 1
Required Programs:
 51-01-01 : NetLink Settings - NetLink System ID
This is the ID number (0 = Disabled, 1-50) that identifies each SV9100 in the
NetLink network. Each SV9100 must be a unique number in the network. When
this option is set to 0, NetLink is disabled.
 The SV9100 must be reset when changing this option.
 51-01-02 : NetLink Settings – Primary Candidate Order
When the Primary SV9100 is turned off or disconnected from the network, this
value (1~50) is used to select the new Primary SV9100 with the Fail-Over
feature. The smaller the number, the higher the priority for the SV9100. When
the value for two SV9100s is the same, lower NetLink System ID (51-01-01) is
selected as the Primary SV9100.
 51-03-01 : NetLink IP Address List – IP Address
The communications server sets the Primary SV9100 using this list. When no
Primary SV9100 is seen or Fail-Over occurs, the Node List is used to establish
a new link. This setting is necessary when Program 51-01-03 is set to 0 or
Program 51-05-02 is other than 0.
Enter IP address for any communications server included in the network
(especially the Primary SV9100). After the communications server connects to
the Primary SV9100, this setting is updated by the Primary SV9100 when
Program 51-13-01 is enabled. This allows any new or changed SV9100s to be
added automatically.
An IP address cannot be defined more than once.
- OR -
 51-01-03 : NetLink Settings – Primary System Selection Method
The link between SV9100s is established based on this setting (0, 1). When
enabled, the SV9100 connects with the top priority Primary SV9100 (address
set in Program 51-04-01). If the SV9100 is not found during the defined time
(Program 51-05-02), the Fail-Over feature of the SV9100 searches for the
Primary SV9100 as if this option were set to 0.
The link between SV9100s is dynamically established based on the node list set
in Program 51-03-01. The Primary SV9100 is selected in the order in which the
SV9100s wake up.

Issue 5.0

this IP address.
 10-12-09 : GCD-CP10 Network Setup – VoIP Daughter Board IP Address
Set the LAN IP address for the VoIP daughter board. The IP number increases
by one for each increase in the slot number.
Related Programs:
 51-09-01 : NetLink TCP Port Settings – Primary to Secondary System
Communication Port
Define the port the Primary SV9100 uses to communicate with the Secondary
SV9100 (0~65535).
 51-09-02 : NetLink TCP Port Settings – Secondary to Secondary System
Communication Port
Define the port used to communicate between other networked communications
servers (0~65535).
 51-09-03 : NetLink TCP Port Settings – Secondary to Primary System
Communication Port
This setting defines the port used by the Secondary SV9100 to communicates
to the Primary SV9100 (0~65535). If 0 is entered, the port is selected
dynamically.
 51-09-04 : NetLink TCP Port Settings – Fail-Over Communication Port
When Fail-Over occurs, each SV9100 communicates with the other
communication servers using the port number specified in this entry (0~65535).
If 0 is entered, the port is selected dynamically.
If an entry other than 0 is made, up to 50 ports (depending in the number of
networked systems) are continuously used from the specified port number. (Ex:
If 5000 is entered, 5001~5049 are used.)
The system ID for each communication server must be unique. Otherwise, the
connection is refused. The information for the refused connection is included in
the communication server alarm report.
If a communication server loses connection and another communication server
with the same ID then tries to connect to the NetLink, it is recognized as the first
communication server (which lost the connection) and is allowed to connect.
If a Secondary System loses connection, it does not affect the operation of the
Primary System.
 51-18-01 : NetLink Config Setting – NetLink Fail-Over Limit
When tear-down of network is repeated more than the specified times, NetLink
is operated in stand-alone. Input range is 0, 2~10 (0 = Infinity). Default is 0.
Conditions
 When the Primary System is switched, all terminals and blades are reset so the
change of the Primary System can be made. This process takes approximately
30 seconds.
 When Fail Over occurs between the Primary System and two or more
Secondary Systems, the attendant telephone displays the System ID of the
system that went into Fail-Over last.

Issue 5.0
 When the attendant's telephone exists on a secondary system, alarm

information cannot be displayed on the attendant's telephone.
 Only the voice mail connected with the Primary System can be used. Since the
recording data of the voice mail is not synchronized, the content originally
recorded by the Primary cannot be succeeded, though the voice mail of the
Secondary System can be used after it becomes the Primary after Fail-Over.
 Synchronous Ring and NetLink cannot be used together.
 The system IP address set can be connected as Primary by turning on Program

51-01-03 instead of Program 51-03-01 and specifying the IP address of the
Primary using Program 51-04-01.
 During Fail-Over, the IP Multiline terminals can connect to the Secondary

System, however when the Primary System comes back online, a 60-second
reset of the Secondary System is required. IP Terminal Programming is
required.
 Only the voice mail connected to the Primary System can be used. During Fail-
Over if a InMail Flash is installed in the new Primary it will now function. This is
done automatically, and no programming is required. During the Fail-Over the
messages do not synchronize to the new InMail Flash. After Recovery the
Original InMail Flash will now function, and any messages stored on the other
Inmail flash will not synchronize.

Issue 5.0
SECTION 7 TOP PRIORITY PRIMARY SYSTEM (PRIMARY SYSTEM AUTOMATIC

INTEGRATION)
When the Primary System is temporarily disconnected/powered down, another

communications server can become the new Primary System. To have the original
Primary System restored as the Primary when it is reconnected/powered up, enable
this function.
 The IP address of the Primary communication server must be defined.
 By default, this function is disabled.
Old Primary
Define the following programs: New Primary
(old Secondary A)
51-04-01 = 192.168.1.20
51-06-01 = 1 (On)
51-06-02 = option 192.168.1.20
Control reverts back to the old
Primary when it becomes available.
Secondary B Check the system availability

51-05-04: interval timer
Secondary E
Secondary C
Secondary D
Related Programming:
this IP address.
 51-06-01 : NetLink Automated Primary System Integration – Automated
When Fail-Over occurs, multiple Primary SV9100s may be established. When
the connection is recovered, with this option enabled (1), the NetLink feature
automatically reestablishes around the top priority Primary SV9100.

Issue 5.0
(1).
 51-09-05 : NetLink TCP Port Settings – Top Priority Primary System
Seeking Port
Enter the port number to search for the Top Priority Primary SV9100. If 0 is
entered, the port is selected dynamically (0~65535).
7.1 Forced Change Primary System
If you need to change the Primary System, you can specify the communication
server manually. While the Top Priority Primary System is running, this feature is
not available. To use this feature, define the IP address or System ID of the target
communication server. From telephone programming, the user can change the
Primary System. When a new Primary System is manually selected, for the IP
terminals to reregister to the new Primary, the existing Primary System must be
powered down and then powered up.
 51-05-06 : NetLink Timer Settings – Enforced Primary System Seeking
Time
When the Forced Change Primary command is executed, the SV9100 searches
for the new Primary SV9100 for this time (1~10800).
(1).
 51-07-01 : NetLink Enforced Primary System Integration - Enforced
When Fail-Over occurs, you can manually change the Primary SV9100 using
Program 51-08, if this option is enabled. Program 51-06-01 must be set to “0”.
When the Forced Change of Primary Settings is performed, the blades in the
secondary communication servers are reset. This option determines if the
(1).
 51-08-01 : Primary NetLink Setting – IP Address of New Primary System
When forcing the communications server to update to a new Primary SV9100,
the communications server uses the IP address defined here as the new
Primary.
 After a Forced Change of Primary SV9100 is done, this entry is erased.

Issue 5.0
 51-08-02 : Primary NetLink Setting - System ID of New Primary System

If you already have an IP address registered in Program 51-11-03, you can
execute a Forced Change Primary SV9100 by entering the system ID. If this ID
is set to 0, the Top Priority SV9100 is selected as the new Primary.
 Blade reset is required.
When the Primary System is changed by Automatic Integration or Forced
Change Primary System, all blades are reset automatically.
7.2 IP Settings
When Fail-Over occurs, the Primary System is changed to another

communication server. The IP applications do not know the new primary IP
Address, so the following features are disabled after Fail-Over.
 SMDR
 ACD-MIS
 SIP Terminal
 Soft Phone
 K-CCIS-IP
SECTION 8 COMMUNICATION SERVER DATABASE SYNCHRONIZATION
When a communication server is connected to the NetLink, the databases of the

Secondary Systems are updated by the Primary System. Should the Primary System
be taken off line, it is possible for any other communication server to take over as the
Primary System position at Fail-Over.
By default, synchronization starts three minutes after the Secondary System is

connected to the Primary System. Each Secondary System is updated one by one.
The databases are synchronized based on the schedule defined in the database
programming.
All systems come up Start database synchronization
3 minutes 30 minutes
}
}
Time
Start database synchronization

NetLink is established
Synchronization completed

Issue 5.0
Be careful when changing the communication server database. If the Primary

System is changed before synchronization, the edited data is lost.
8.1 Asynchronous Settings
The following program settings are not updated by the Primary System:
Program Number Name

10-01 Date and Time Setup
10-02 Location Setup
10-12 CCPU Network Setup
10-13 In-DHCP Server Setup
10-14 Managed Network Setup
10-15 Client information Setup
10-16 Option information Setup
10-23 IP System Interconnection Setup
10-28 SIP System Information Setup
10-29 SIP Server Information Setup
10-33 SIP Registrar/Proxy Information Basic Setup
10-36 SIP Trunk Registration Information Setup
10-37 UPnP Setup
10-62 NetBIOS Setting
10-72 Network Security Setup
20-66 STUN/TURN Server Setting
21-19 IP Trunk(SIP) Calling Party Number Setup for Extension
51-01 NetLink Settings
84-09 VLAN Setup
84-13 SIP Trunk CODEC Information Basic Setup
84-14 SIP Trunk Information Basic Setup
90-01 Installation Date
90-08 System Reset
90-09 Automatic System Reset Time Setup
90-68 Side Tone Auto Setup
90-77 LAN Link Speed Information
93-01 Day/Night Mode Information
93-02 Trunk Information

Issue 5.0
Program Number Name

93-03 Extension Information
93-04 Redial List
93-05 Department Group Information
93-06 IP Address list for 1st Party CTI connection
8.2 SRAM Database
Data in SRAM area is not transferred to Secondary Systems. When Fail-Over

occurs, a terminal may lose DND, Caller ID History, etc.
 51-09-06 : NetLink TCP Port Settings – System Data Replication
Communication Listenning Port
Define the listening port used so the Secondary SV9100 can replicate the
Primary SV9100 database (0~65535).
 51-09-07 : NetLink TCP Port Settings – System Data Replication Primary
Detectiong Port
Define the port used for communication so that the Primary SV9100 may
synchronize the Secondary SV9100 with the program data (0~65535). If 0 is
entered, the port is selected dynamically.
Replication Mode
Set the replication mode (0 = Disabled, 1 = Setting Time Mode, 2 = Interval
Mode). An entry of 1 replicates the data at the time set in Program 51-16-02. If
this option is set to 2, replication occurs at the time set in Program 51-16-03.
Replication Time Setting
If Program 51-16-01 is set to 1, set the time to replicate the program data
(0000~2359).
Replication Interval Setting
If Program 51-16-01 is set to 2, set the time between replicating the program
data (15-1440 minutes).
 51-16-04 : NetLink System Data Replication Mode Setting – Replication
Time Stamp
This program displays the last time the program data was replicated. This is
automatically updated when the replication occurs. This option is view-only.
Replication Wait Time
When a NetLink network is created, this option determines the time the SV9100
waits before replication is started (1~86400 seconds).

Issue 5.0

Replication Interval
Set the time to start replication to the next node after replication has completed
to one node (1~86400 seconds).
SECTION 9 CREATE RECOVERY DATABASE
Before assigning a System ID, you should create a Recovery Database. The
Recovery Database can be used when you want to disconnect the communication
server from the NetLink.
Communication server reset is required.

IMPORTANT
 90-57-01 : Backup Recovery Data
Back up the SV9100 data file preserved in the flash memory on the CCPU. This
data is used for recovering data if required. Up to five recovery files can be
preserved in the flash memory on the CCPU.
 90-58-01 : Restore Recovery Data
If required, use this option to restore one Backup Recovery Data file (files 1-5,
saved in Program 90-57-01) to restore the previous SV9100 data. After executing
this option, the SV9100 restarts automatically.
 90-59-01 : Delete Recovery Data
Delete the Backup Recovery Data files (files 1~5, saved in Program 90-57-01), if
needed.
SECTION 10 LICENSING CONTROL
For the NetLink feature, a license with an activation code is required. This license
determines how many communication servers can be connected.
All licenses for all options (applications, adapters, ports, etc.) must be registered in
the Primary System.
The VoIP Channel for each Secondary must be licensed in PRG 10-54 in the Primary
System.
When Fail-Over occurs:
The Primary System gives a copy of the activation codes to the Secondary Systems.
This allows the features to continue even if the Primary System fails and a
Secondary System becomes the new Primary System.

Issue 5.0
However, the license of Secondary System is temporary – lasting 28 days. After the
specified duration, the communication server is reset and the licenses are removed.
If the link is recovered and original Primary System is back in service, the licenses
are returned to use without any expiration time.
 License information in the Primary is copied to the Secondary site when doing
database duplication.
 After 28 days, the license(s) expire. To renew the license(s), a connection to the
original Primary site must be reestablished. (After the connection to the Primary
is recovered, if Fail-Over occurs again, the license(s) are once again enabled
for the specified duration based on the software version listed above)
 When the Forced Change of the Primary System (Program 50-07-01) is used to
manually force the original Primary System to become one of the Secondary
Systems in a Netlink Network, the license information will still be available.
 If a user wants to enter another additional feature license, it must be entered on

the original Primary System.
SECTION 11 MAC ADDRESS AUTHORIZATION
NetLink can reject the connection from unauthenticated communication servers.
The Primary System finds the MAC address of the communication server which is
trying to connect, and if the Primary System does not know the MAC address, the
connection is refused.
 51-02-04 : NetLink System Individual Setting – System Authentication MAC
Address
When Program 51-13-03 is enabled, the SV9100 checks this MAC address against
the MAC address of the connecting CCPU. If the value is different, the connection is
refused.
 51-13-03 : NetLink Option Settings – MAC Address Authentication
When enabled (1), connection authentication of the SV9100 is done with the MAC
address set in Program 51-02-04. The system compares its own MAC address and
if the address does not match, the Primary System rejects the connection.

Issue 5.0
SECTION 12 MAINTENANCE FEATURES
Confirming NetLink Basic Information
From a multiline terminal, press Feature + 5. The LCD display shows the System ID,
Primary/Secondary, IP Address of the VOIPDB (not GCD-CP10), and the Primary
System ID.
Confirming Communication Server Status
If an installer needs to know the status of each communication server, on the Primary
System, access Program 51-11 from telephone programming. The following is
displayed:
 IP Address
 MAC Address (GCD-CP10)
 Communication Server Name
 Software Version
 Primary Priority
 Connect Status
– OFF: The communication server is disconnected.
– ON: The communication server is connected as secondary.
– Primary: The communication server is connected as primary.
SECTION 13 PORT ASSIGNMENT
Port assignment starts from the Primary System lowest slot number. The port
numbers for the Secondary Systems are assigned in the order in which the
communication servers are connected. The next port number is assigned to the first
Secondary System which connects to the NetLink SV9100.
(See the example)

Issue 5.0
Primary System
Slot GCD-CP10
Slot GCD-DLCA Ext port: 1 - 16
Slot GCD-4COT Trunk port: 1 - 4
Slot GCD-4PRTA Trunk port: 5 - 12
Secondary System 1 Connected First
Slot GCD-CP10
Slot GCD-16DLCA Ext port: 17 - 32
Slot GCD-8LCA Ext port: 33 - 40
Slot GCD-4PRTA Trunk port: 13 - 20
Slot 5
Secondary System 2 Connected Second
Slot GCD-CP10
Slot GCD-16DLCA Ext port: 41- 56
Slot GCD-2BRIA Trunk port: 21 - 24
As described above, port assignment is performed in the order which the

communication server connects. With this condition, if a GCD-4COT is installed to
the Communication Server 1 – slot 5, the trunk ports are from 25 to 28. To predefine
the slots (and not have them automatically assigned by the communication server),
use PCPro.
SECTION 14 SLOT CONTROL
When performing a Remote Upgrade, you must connect to each individual system in
a NetLink Network. A Remote Upgrade to the Primary System does not upgrade the
Secondary Systems.
In the NetLink, the Primary System needs to control all the slots of the Secondary
Systems. For the Primary System to handle more slots, there is a Virtual Slot
Function in NetLink.

Issue 5.0
This function allows the Primary System to control 240 slots and, users can check
what kind of blade is installed in each communication server and how many blades
are available in the link.
Virtual Slot Image
Virtual Slot 1
Virtual Slot 2
Virtual Slot 3
Virtual Slot 4
Virtual Slot 240

~
Up to 240 Virtual Slots

Virtual Slot 1
Virtual Slot 2
Virtual Slot 3
Virtual Slot 4
Physical Slot 1
Physical Slot 2
System ID 1 (Primary) System ID 2 (Secondary)
 51-10-01 : NetLink Virtual Slot Settings – Next Available Slot
View the remaining number of slots which can be controlled by NetLink. The
NetLink feature can control up to 240 virtual slots maximum. (The physical slots in
the NetLink network are maintained as virtual slots by the SV9100.) This option is not
user-definable.
Slot Deletion:
1. Unplug the blade from the communications server chassis.
2. Use Program 90-05 : Slot Control and select menu #1 (delete command).
3. Specify the System ID and Slot Number.
The information in the virtual slot is deleted as well.

IMPORTANT
Communication Server Information Deletion:

 Using Program 51-14-01 : NetLink System Control – Delete System
Information, Specify the System ID
To use this program, the communications server must not be connected to the
network.
All virtual slot information of the communication server is deleted. The

Issue 5.0
communication server no longer appears in Program 51-11.
SECTION 15 TIME ZONE SETUP
If a communication server in the network is in a different location, the time zone may
be different. To adjust the clock display on multiline terminals, the user can define the
offset from the Primary System.
The following features are adjusted when the offset is enabled in Program 51-13-02:
 Terminal Time Display
 Incoming History/Redial History
 VRS Recording Announce
 Date and Time Setting by Service Code
 Calendar Setting by Service Code
 Alarm Clock Setting
 Hotel/Motel Wake-Up Call
 10-01-01 - 10-01-07 : Time and Date
Change the SV9100 Time and Date through SV9100 programming. Extension
users can also dial Service Code 828 to change the Time, if allowed by an
extension Class of Service.
 51-02-02 : NetLink System Individual Setting – Time Zone – Hour
Determine the offset hours from the Primary SV9100. This setting affects the time
display on display telephones (0~24 = -12 ~ +12 hours).
 51-02-03 : NetLink System Individual Setting – Time Zone – Minute
Determine the offset minutes from the Primary SV9100. This setting affects the time
display on display telephones (0 ~ 120 = -60 ~ +60 minutes).
 51-13-02 : NetLink Option Settings – Time Zone Enhancing
When enabled (1), the time zone is applied to the following items:
LCD Clock Display, Caller ID History, VRS Time Announce, Time and Date Set by
Service Code, Alarm Clock, Hotel Mode Wake-Up Call (time announce included).
When disabled (0), the time zone is applied only to the LCD Clock Display and
Caller ID History.
Program 51-02-02 must also be set for this option.

Issue 5.0
SECTION 16 VOIP HANDLING
Using VoIP equipment is permitted with NetLink. A maximum of VoIP 256 channels
can be controlled.
When an analog/digital terminal talks via NetLink, a VoIP resource in the

communications server to which the terminal is connecting is used.
When an IP terminal talks to an analog/digital terminal via NetLink:

 For an IP terminal, a VoIP resource is not required.
 For an analog/digital terminal, one VoIP resource is required.
When IP terminals talk to an IP terminal, they follow Peer-to-Peer setting:

 Peer to Peer: VoIP Resource is not required.
 Not Peer to Peer: Two VoIP resources are required.
When an analog/digital terminal uses a CO line via NetLink:

 The chassis which has the analog/digital terminal needs one VoIP resource,
and the chassis which has the CO line needs one VoIP resource.
When an IP terminal uses a CO line via NetLink:

 The chassis which has the CO line needs one VoIP resource.
When an IP terminal talks to an IP trunk:

 The Primary System requires two VoIP resources.
When an analog/digital terminal talks to an IP trunk via NetLink:

 The chassis which has the analog/digital terminal needs one VoIP resource.
 The Primary System requires two VoIP resources.

Issue 5.0
The number in each box indicates how many VoIP resources are required to talk.
Primary System Secondary 1 Secondary 2
CO CO CO
TDM IP IP TDM TDM
Analog/ Analog/ IP Trunk Analog/ IP Trunk
Terminal Terminal Trunk Terminal Terminal
Digital Digital Digital
TDM P:1 P:1 P:1 P:1 P:1 P:1

0 P:1 0 P:1
Terminal S1:1 S1:1 S1:2 S2:1 S2:1 S2:2
Primary
System
IP
P:1 0 P:1 P:2 S1:1 S1:1 S1:2 S2:1 S2:1 S2:2
Terminal
Secondary TDM P:1 S1:1 P:1 P:2 P1:1 S1:1 S1:1

0 0 S1:1
System 1 Terminal S1:1 S1:1 S1:1 S2:1 S2:1 S2:2
Secondary TDM P:1 S2:1 P:1 P:2 S1:1 S1:1 S2:1

0 0 S2:1
System 2 Terminal S2:1 S2:1 S2:1 S2:1 S2:1 S1:2
 P: Primary System
S1: Secondary System #1
S2: Secondary System #2
About VOIP DSP Settings
Program 84-26 determines the VoIPDB IP address and port number. The settings are
programmed by the Primary System.
If you assign a VoIPDB IP address in the Secondary System before connecting the
Primary System, the Primary System reflects the data for the VoIPDB settings.

Issue 5.0
Sample Configurations
NetLink
NetLink
NetLink
LAN
System ID = 1 System ID = 2 System ID = 3
LAN
Primary Secondary Secondary

System ID = 1 System ID = 2 System ID = 3

Issue 5.0
Configuration Flow (Case 1) – Primary Based on System ID
1. Create a Recovery Database – Recommended.

Because each communication server database and communication server
configuration are updated by the Primary System, creating a recovery database is
recommended. This may be needed if the communication server goes back to a
stand-alone server.
Refer to the Section 8 Communication Server Database Synchronization on page
10-38 and Section 9 Create Recovery Database on page 10-41 sections.
Before programming the NetLink feature, general IP configuration

must be performed.
CAUTION ❍ Program 10-12-09 : VoIPDB IP Address
❍ Program 10-12-10 : Subnet Mask
❍ Program 84-26-01 : VoIPDB DSP IP Address
❍ Program 10-12-03 : Default Gateway
❍ Program 10-12-09 : VoIPDB IP Address
❍ Program 10-12-03 : Default GatewayProgram 10-12-09 : VoIPDB
IP Address
2. For NetLink, the minimum programs must be set:

 System IP Address List (Program 51-03-01)
 System ID (Program 51-01-01)
3. After entering the System ID, the communication server asks to reboot the
communication server. Reboot.
4. When the communication server comes up, the Primary System and Secondary
Systems are determined automatically.
If you need to change the Primary System to another server, use 7.1 Forced
Change Primary System on page 10-37.
When connection cannot be made to the original Primary System, and a new
Primary is selected, it can automatically return the original server as the Primary
System after connection/power is restored. Refer to the Section 7 Top Priority
Primary System (Primary System Automatic Integration) on page 10-36 function.

Issue 5.0
 For centralized voice mail the voice mail must be installed in the Primary System
(Site A). Assigning Top Priority Primary System and Primary System Automatic
Integration are recommended.
Specify the communication server in Site-A for the Top Priority Primary
System
Site-A becomes the Primary System. Without this function, either Site-A or Site-B
can become the Primary System.
Enable Primary System Automatic Integration
When Site-A is disconnected from the LAN, Fail-Over works and Site-B becomes
the Primary.
If you would rather stop the Fail-Over function, you can disable it by setting Program
51-05-02 to 0. With it disabled, Site-B is locked while the Primary System is down.
If Site-A connects to the LAN again, the Automatic Integration function reestablishes
the NetLink and Site-A becomes the Primary. Without this function, Site-B remains
the Primary.
Configuration Flow (Case 2) – Primary Defined Based on Programming
After you decide which communication server should become the Primary:
1. Create a Recovery Database – Recommended.

Because each communication server database and communication server
configuration is updated by the Primary System, creating a recovery database is
recommended. This may be needed if the communication server goes back to a
stand-alone server.
Refer to the Section 8 Communication Server Database Synchronization on page
10-38 and Section 9 Create Recovery Database on page 10-41 sections.
2. Before programming the NetLink feature, general IP configuration must be
performed.
 Program 10-12-09 : VOIPDB IP Address
 Program 10-12-10 : Subnet Mask
 Program 84-26-01 : VOIPDB DSP IP Address
 Program 10-12-03 : Default Gateway, etc.
3. Program the following NetLink settings:
Primary - Site A Secondary - Site B
51-01-01 System ID 51-01-01 System ID

51-04-01 Top Priority System 51-04-01 Top Priority System
Enter own IP address Enter site A (Primary) IP Address
51-06-01 Enable Automatic Integration 51-01-03 Set Secondary Flag
4. After entering the System ID, the communication server asks to reboot the
communication server. Reboot.

Issue 5.0
5. When the communication servers come up, the Primary System and Secondary
Systems are connected and start working.
6. For database synchronization, wait three minutes (or more).
SECTION 17 NETLINK MULTI-SIP CARRIER
 When the Secondary NetLink system calls out using its own SIP Trunk, no
DSPs are used from the Primary system.
 The NetLink Nodes which have their own SIP trunks can use Register IDs
independently of each other. A Secondary NetLink system is able to utilize its
own SIP trunks independently to the Primary system.
Conditions
 When Programming SIP Trunks on a secondary system only SIP Profile
One can be utilized.
 It is possible to set Register ID for trunks that belong to that specific
system. For example, a Register ID set in the Primary system cannot be
assigned to a trunk in the Secondary system. The allocation of the trunk
and Register ID of Program14-12 must be in the same system.
 In order to use CPN in a Secondary system, Program 51-19 must be
turned on for those extensions. Once enabled, CPN may be sent on a
per station basis using Program 21-19.
 Each NetLink system can use either SIP trunks to a provider or SIP
trunk TIE line mode, but not both.
 Once NetLink is established, PCPro or WebPro must be used to change
the system data related to the SIP trunks.
 Any SIP trunks that are built in a system before establishing NetLink will
be deleted after establishing NetLink.
 SIP trunks are assigned in the order of system set up. System ID's are
needed to assign Program 10-40 data.
 Registered SIP trunks can be utilized by any system in the NetLink
network, as long as trunk route programming allows it.
 When a Secondary system becomes the primary after fail over, the SIP
trunks will work for the effective license time.

Issue 5.0
Default Setting
No SIP Trunks are defined
Programming
This section lists each program in numerical order. For example, Program 10-01 is at
the beginning of the section and Program 92-01 is at the end. The information on
each program is subdivided into the following headings:
Description describes what the program options control. The Default Settings for
each program are also included. When you first install the system, it uses the Default
Setting for all programs. Along with the Description are the Conditions which
describe any limits or special considerations that may apply to the program.
The reverse type (white on black) just beneath the Description heading is the
program access level. You can use the program only if your access level meets or
exceeds the level the program requires. Refer to Section 18 How to Enter
Programming Mode on page 10-54 for a list of the system access levels and
passwords.
Feature Cross Reference provides you with a table of all the features affected by
the program. You should keep the referenced features in mind when you change a
program. Customizing a feature may have an effect on another feature that you did
not intend.
Telephone Programming Instructions shows how to enter the program data into
system memory. For example:
1. Enter the programming mode.

2. Dial 150701 from the telephone dial pad. The message 15-07-01 TEL is displayed
on the first line of the telephone display.
15-07-01 TEL
KY01 = *01

This indicates the program number (15-07), item number (01), and that the options
are being set for the extension. The second row of the display KY01 = 01 indicates
that Key 01 is being programmed with the entry of 01. The third row allows you to
move the cursor to the left or right, depending on which arrow is pressed. To learn
how to enter the programming mode, refer to Section 18 How to Enter Programming
Mode on page 10-54.

Issue 5.0
SECTION 18 HOW TO ENTER PROGRAMMING MODE

 In a newly installed system, use extension (port 1).
3. Press Speaker.
4. ## .
Password

Refer to the following table for the default system passwords. To change the
passwords, use Program 90-02: Programming Password Setup.
User
Name
----- ----- 1 (MF) Manufacturer (MF):

All programs
All programs in this section not listed below for SA
and SB
0000 ADMIN1 3 (SA) System Administrator – Level 1 (SA):
10-01, 10-02, 10-12, 10-13, 10-14, 10-15, 10-16,
10-17, 10-18, 10-22, 12-02, 12-03, 12-04, 15-01,
15-07, 15-09, 15-10, 15-11, 20-16, 21-07, 21-14,
22-04, 22-11, 25-08, 30-03, 32-02, 40-02, 41-02,
41-03, 41-04, 41-05, 41-06, 41-07, 41-08, 41-09,
41-10, 41-11, 41-12, 41-13, 41-14, 41-15, 41-16,
41-17, 41-18, 90-03, 90-04, 90-06, 90-07, 90-18,
90-19
9999 ADMIN2 4 (SB) System Administrator – Level 2 (SB):
13-04, 13-05, 13-06

Issue 5.0
SECTION 19 HOW TO EXIT PROGRAMMING MODE
 When you are done programming, you must be out of a program option to exit (press
Answer to exit the program option).
1. Press Answer to exit the program options, if needed.
Program Mode
2. Press Speaker. If changes were made to system programming, Saving System

Data is displayed.
3. The display shows Complete Data Save when completed and exits the
telephone to idle.
 To save a customer database, a blank USB Drive is required. Insert the USB Drive
into the GCD-CP10 and, use Program 90-03, to save the software to the USB Drive.
(Use Program 90-04 to reload the customer data if necessary). A USB Drive can
hold only one customer database. Each database to be saved requires a separate
drive.
SECTION 20 USING KEYS TO MOVE AROUND IN THE PROGRAMS
After you enter the programming mode, use the keys in the following chart to enter
data, edit data and move around in the menus.
0~9 and  Enter data into a program.
Transfer Complete the programming step you just made (e.g., pressing Enter
on a PC keyboard). When a program entry displays, press Transfer to
bypass the entry without changing it.
Recall Delete the entry to the left (e.g., pressing Backspace on a PC
keyboard).
For example, if programming item 5 in 15-03, press Answer to enter a
new option in program 15-03. Press Answer again to select a new
program in the 15-XX series. Press Answer a third time to enter a new
program beginning with 1. Press Answer one last time to bring you to the
beginning program display to allow you to enter any program number.

Issue 5.0
MIC Press MIC to switch between the different input data fields. The cursor
moves up to the top row of the display. Press MIC again to move the
cursor back to the middle row.
LINE KEYS Use programmed settings to help with the program entry. These settings
vary between programs from LINE 1 = 0 (off) and LINE 2 = 1 (on) to preset
values for timers where LINE 1 = 5, LINE 2 = 10, LINE 3 = 15, etc.
settings.)
LINE KEY 1 Program a pause into a Speed Dialing bin.
LINE KEY 2 Program a recall/flash into a Speed Dialing bin.
LINE KEY 3 Program an @ into a Speed Dialing bin.
VOL  Scroll backward through a list of entry numbers (e.g., from extension
etc.) or through entries in a table (e.g., Common Permit Table).
VOL  Scroll forward through a list of entry numbers (e.g., from extension

Issue 5.0
SECTION 21 PROGRAMMING NAMES AND TEXT MESSAGES
Several programs (e.g., Program 20-16 : Selectable Display Messages) require you
to enter text. Use the following chart when entering and editing text. When using the
keypad digits, press the key once for the first character, twice for the second
character, etc. For example, to enter a C, press 2 three times. Press the key six times
to display the lower case letter. The name can have up to 12 digits.
1 Enter characters:
1 @ [ ¥ ] ^ _ ` { | }  Á À Â Ã Ç É Ê ì ó

0 Enter characters:
0 ! “ # $ % & ’ ( ) ô Õ ú ä ö ü 
 Enter characters:
 + , - . / : ; < = > ? B E  S ¢ £
# # = Accepts an entry (only required if two letters on the same key
are needed – ex: TOM). Pressing # again = Space. (In system
programming mode, use the right arrow Softkey instead to accept
and/or add a space.)
Conf Clear the character entry one character at a time.
Hold Clear all the entries from the point of the flashing cursor and to the
right.

Issue 5.0
SECTION 22 USING SOFTKEYS FOR PROGRAMMING
Each UNIVERGE SV9100 display telephone provides interactive Softkeys for

intuitive feature access. The options for these keys automatically change depending
on where you are in the system programming. Press the Softkey located below the
option you want, and the display changes accordingly.
_
Program Mode

Press the VOLUME  or VOLUME  to scroll between the menus.

_
Program Mode
CCIS Hard Mtnance

SECTION 23 WHAT THE SOFTKEY DISPLAY PROMPTS MEAN
When using a display telephone in programming mode, various Softkey options are
displayed. These keys allow you to easily select, scan, or move through the
programs.
If you press this

The system will. . .
Softkey . . .

You can press VOLUME  or VOLUME  to scroll forward or

Issue 5.0

Level:
SA
Description
Use Program 10-12 : GCD-CP10 Network Setup to setup the IP Address,
Subnet-Mask, and Default Gateway addresses.
Caution! If any IP Address or NIC settings are changed, the system must be
reset for the changes to take affect.
Input Data
Item
No.

128.0.0.1 ~ 191.254.255.254
192.0.0.1 ~ 223.255.255.254

all Host Addresses
254.0.0.0 255.0.0.0 255.128.0.0
are 0.
255.192.0.0 255.224.0.0 255.240.0.0
255.248.0.0 255.252.0.0 255.254.0.0
is:
255.255.0.0 255.255.128.0 255.255.192.0
0,
255.255.224.0 255.255.240.0 255.255.248.0
127,
255.255.252.0 255.255.254.0 255.255.255.0
255.255.255.128 255.255.255.192 255.255.255.224 128.0,
255.255.255.240 255.255.255.248 255.255.255.252 191.255,
255.255.255.254 255.255.255.255 192.0.0,
223.255.255
Mask is invalid.

Gateway 128.0.0.1 ~ 191.254.255.254
192.0.0.1 ~ 223.255.255.254

Issue 5.0
Item
No.
04 Time Zone 0~24 (0 = -12 Hours and 24 = +12 Hours) 12 Determine the offset
(0 hours) from Greenwich Mean
Time (GMT) time.
Then enter its
example, Eastern
Time (US and
Canada) has a GMT
offset of -5. The
program data would
then be 7 (0= -12, 1=
-11, 2= -10, 3= -9, 4= -
8, 5= -7, 6= -6, 7= -5,
……24= +12)

Interface 1 = 100Mbps, Full-Duplex (GCD-CP10)
07 NAPT 0.0.0.0 ~ 126.255.255.254 0.0.0.0 Set the IP address on

Router IP 128.0.0.1 ~ 191.255.255.254 the WAN side of
Address router.
(Default 192.0.0.1 ~ 223.255.255.254
Gateway
[WAN])

Redirect 1= (Disable) redirect message,
determine if the IP
Routing Table
or not.
09 IP Address 0.0.0.0 ~ 126.255.255.254 172.16.0.10 Set for IPLE.

128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
10 Subnet 128.0.0.0 192.0.0.0 224.0.0.0 255.255.0.0 Set for IPLE.

Mask 240.0.0.0 248.0.0.0 252.0.0.0
254.0.0.0 255.0.0.0 255.128.0.0
255.192.0.0 255.224.0.0 255.240.0.0
255.248.0.0 255.252.0.0 255.254.0.0
255.255.0.0 255.255.128.0 255.255.192.0
255.255.224.0 255.255.240.0 255.255.248.0
255.255.252.0 255.255.254.0 255.255.255.0
255.255.255.128 255.255.255.192 255.255.255.224
255.255.255.240 255.255.255.248 255.255.255.252
255.255.255.254 255.255.255.255

Issue 5.0
Item
No.
11 NIC Setup 0 = Auto Detect 0 Set for IPLE.

5 = 1 Gbps, Full-Duplex
Conditions
 The system must be reset for these changes to take affect.


Issue 5.0

10-54 : License Configuration for Each Package
Level:
IN
Description
Use Program 10-54 : License Configuration for Each Package to set the license
information for each unit.
Input Data
Slot Number 1~24
License Index Number 1~32
Item
Item Read Data
No.
01 License Code 0000~9999
02 License Quantity 0~255
Conditions
 If applying more than 255 licenses to a slot the licenses must be applied across
multiple indexes. For example assigning 256 VoIP resource licenses (5103) to
the CPU slot could be assigned using different methods as long as the total for
the CPU slot is 256:
1. Index 1 has 128 of feature code 5103 and index 2 also has 128 of feature
2. Index 1 has 255 of feature code 5103 and index 2 has 1 of feature code
5103 for a total of 256.
 When using IP devices IP Resource licenses (5103) must be assigned to the
CPU Slot (1) for them to be available for use. If this is not done, IP related
features will not work.

None

Issue 5.0
Program 51 : NetLink Service

51-01 : NetLink Settings
Level:
IN
Description
Use Program 51-01 : NetLink Settings to define the parameters of the NetLink
feature.
❍ Each system must be set with its own information.

NOTE
❍ When the NetLink System ID is changed (Item 01), the system must be reset.
Input Data
Item
No.
01 NetLink System ID 0~50 0

This is the ID of each NetLink system. Setting (0 = No operation)
should insure that no overlap occurs between
nodes.
02 Primary Candidate Order 1~50 30

When the Primary system is turned off or
disconnect from network, this value is used to
select a new Primary system. Smaller number
is higher priority.
If this value is the same number, the System ID
(Program 51-01-01) is referred, and the
system which has the smaller number is
selected as Primary system.
03 Primary System Selection Method 0 = Dynamic 0

0: NetLink is dynamically established based on 1 = Top Priority Primary
Node List in Program 51-03-01. System
Primary System is selected in the order which
the system wakes up.
1: The system connects with Top Priority
Primary System.
If Top Priority Primary System is not found, the
system searches Primary System as if this
setting is 0.

Issue 5.0
04 Signal Transmit Method 0 = Immediate 0

0 = Immediate 1 = Buffering
This is the default setting which does not use
Nagle Algorithm. When this is enabled data
packets are immediately sent across the
network with no buffering delay.
1 = Buffering
Nagle Algorithm enabled. This means that
small data packets will not be transmitted
immediately across the network. The smaller
data packets will be buffered and then sent
across as larger data packets therefore
decreasing the number of packets sent across
the network. When the number of packets
sent across the network decreases, the
amount of bandwidth also decreases.
Conditions
None

None

Issue 5.0

51-02 : NetLink System Basic Setup
Level:
IN
Description
Use Program 51-02 : NetLink System Basic Setup to set system data for each
NetLink system.

NOTE
Program 51-02-03 is not used in US, but is used in other countries.
Input Data
System ID 1~50
Item
No.
01 System Name Up to 20 characters. blank

This name is given to each system.
02 Time Zone (Hour) 0~24 12

Determine the time offset from the Primary
system.
(0 = -12, 1 = -11, 2 = -10.... 12 = 0
13 = +1, 14 = +2, 24 = +12)
This setting affects Time Display on MLT
(see Program 51-13-02).
03 Time Zone (Minute) 0~120 60

Determine the time offset from the Primary
system.
(0 = -60, 1 = -59, 2 = -58.... 120 = +60)
This setting affects Time Display on MLT
(see Program 51-13-02).
This program is not used in the US, but is
used in other countries.
04 Authenticate System MAC Address 00-00-00-00-00-00~ 00-00-00-00-00-00

To use this function, set Program 51-13-03 FF-FF-FF-FF-FF-FF
to 1 (enable), NetLink systems reject the
connection from unauthenticated system
access.

Issue 5.0
Conditions
None

None

Issue 5.0

51-03 : NetLink IP Address List
Level:
IN
Description
Use Program 51-03 : NetLink IP Address List to set the IP address of the NetLink
system.
Input Data
List ID 1~50
Item
No.
01 Internet Protocol Address List 0.0.0.0 ~ 126.255.255.254 0.0.0.0

The system seeks the Primary system based on 128.0.0.1 ~ 191.255.255.254
this list. 192.0.0.1 ~ 223.255.255.254
When there is no Primary system yet, or
Fail-Over occurs, Node List is referred to
establish new link. This setting is necessary
when Program 51-01-03 is 0, or Program
51-05-02 is other than 0.
Once the system connects to the Primary
System, this setting is updated by the Primary
system when Program 51-13-01 is On. So,
enter IP address of the systems which may
become Primary at least.
Conditions
 The system seeks Primary system based on this list.
 When there is no Primary System yet, or Fail-Over occurs, Node List is referred
to establish new link.
 This setting is necessary when Program 51-01-03 is 0, or Program 51-05-02 is

other than 0. Once the system connects to the Primary System, this setting is
updated by the Primary system when Program 51-13-01 is on. So, enter IP
address of the systems which may become Primary.

None

Issue 5.0

51-04 : NetLink Top Priority Primary System IP Address
Level:
IN
Description
Use Program 51-04 : NetLink Top Priority Primary System IP Address to set the
IP address of the new Primary System.
Input Data
List ID 1~50
Item
No.
01 Internet Protocol Address of Top Priority 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Primary 128.0.0.1 ~ 191.255.255.254
Enter the IP address of the Top Priority Primary 192.0.0.1 ~ 223.255.255.254
System.
To use this feature, set Program 51-06-01 to
1(On).
Conditions
None

None

Issue 5.0

51-05 : NetLink Timer Settings
Level:
IN
Description
Use Program 51-05 : NetLink Timer Settings to set the various timers within the
NetLink system.
Item
No.
01 Keep Alive Sending Interval 1~3600 5

This is the Keep Alive timer sending time from
the Secondary system to confirm
communication with the Primary system.
02 Keep Alive Response Waiting Time 0, 5~10800 0

This is the time the Secondary system waits for (0 = infinity)
a response from the Primary system before
cutting off communication.
03 Primary Search Packet Sending Interval 1~3600 5

While searching the Primary system, the system
sends a packet at this interval.
04 Primary Search Time Maximum Value 5~10800 20

Total time of Primary system seek time.
05 Top Priority Primary Detection Packet 1~3600 10

Sending Interval
When current Primary system is not Top Priority
Primary System, the system sends packet to
check if Top Priority System exists.
06 Primary Compulsion Specification Trial 1~10800 30

Maximum Time
When the forced change Primary command is
executed, the system searches the new Primary
system for this time.
07 Socket Refresh Time 20~3600 40

For some reason, the IP connection may
become unstable. Then keep-alive function
does not work. To avoid this, if there is no data
traffic for this time, the socket is refreshed.
Conditions
None

None

Issue 5.0

51-06 : NetLink Automated Primary System Integration
Level:
IN
Description
Use Program 51-06 : NetLink Automated Primary System Integration to set the
automatic integration of the Primary system.
Input Data
Item
No.
01 Primary Integration Right or Wrong 0 = Off 0

When LAN cable was divided, multiple Primary 1 = On
systems may appear. If the LAN connection is
recovered, multiple Net-Links exist in the
network. When this option is enabling, NetLink is
composed around Top priority Primary System.
02 Package Reset Timing Option 0 = Reset when all packages 0

When Primary System Automatic Integration is are idle.
done, all packages of secondary systems reset. 1 = Anytime
Select the timing of package reset.
Conditions
None

None

Issue 5.0

51-07 : NetLink Enforced Primary System Integration
Level:
IN
Description
Use Program 51-07 : NetLink Enforced Primary System Integration to set
compulsion specification of the Primary system.
Input Data
Item
No.
01 Forced Change Primary System Enabling 0 = Disable 0

Set this item whether the Forced Change 1 = Enable
Primary is available or not.
02 Package Reset Timing Option 0 = On 0

When Forced Change Primary System is done, 1 = Off
all packages reset. Select the timing of package
reset.
0 = Reset when all packages are idle, otherwise
reject Primary System Integration.
1 = Anytime
Conditions
None

None

Issue 5.0

51-08 : Primary NetLink Setting
Level:
IN
Description
Use Program 51-08 : Primary NetLink Setting to set the IP address and system ID
of the compulsory specification of the Primary system..
This program is available only via telephone programming and not through PC

NOTE
Programming.
Input Data
Item
No.
01 IP Address of New Primary System 0.0.0.0 ~ 126.255.255.254 0.0.0.0

Enter target IP address for New Primary system. 128.0.0.1 ~ 191.255.255.254
When the Forced Change Primary system is 192.0.0.1 ~ 223.255.255.254
done, this setting is erased.
02 System ID of New Primary System 0~50 No

When set to 0, top priority Primary system is setting
assumed to be the new Primary system.
Conditions
None

None

Issue 5.0

51-09 : NetLink TCP Port Settings
Level:
IN
Description
Use Program 51-09 : NetLink TCP Port Settings to set the various communication
ports used on the system.
Input Data
Item
No.
01 Primary to Secondary System 0~65535 58000

Communication Port
Set the communication port that the Primary
system uses to communicate with the
Secondary system.
02 Secondary to Secondary System 0~65535 58001

Communication Port
Port used to communicate between nodes. It is
always opened by all nodes.
03 Secondary to Primary System 0~65535 0

Communication Port
Secondary system communicates with Primary
system at this port number. If 0 is specified,
temporary port is selected dynamically.
04 Fail-Over Communication Port 0~65535 0

When Fail-Over occurred, each system
communicates with other system at this port
number.
If 0 is set, temporary port is selected
dynamically.
If 0 is not specified, the number and continuous
maximum 50 number is used.
(Ex. 5000 is specified 5001, 5002…5049 are
used).
05 Top Priority Primary System Seeking Port 0~65535 0

Enter port number to seek the Top Priority
Primary system.
If 0 is specified, temporary port is selected
dynamically.
06 System Data Replication Communication 0~65535 58002

Listening Port
Replicate the database.

Issue 5.0
Item
No.
07 System Data Replication Primary 0~65535 0

Detection Port
Replicate database.
If 0 is specified, temporary port is selected
dynamically.
Conditions
None

None

Issue 5.0

51-10 : NetLink Virtual Slot Settings
Level:
IN
Description
Use Program 51-10: NetLink Virtual Slot Setting to view the number of Virtual
slots that are remaining in a NetLink network. There can be up to 240 virtual slots
available in NetLink.
Item
No.
01 Next Available Slot

240 slots can be controlled in NetLink. This
command can check how many slots are
available.
Conditions
 This Program is Read Only.

None

Issue 5.0

51-11 : NetLink System Information
Level:
IN
Description
Use Program 51-11: NetLink System Information to reference information about
other systems in the NetLink network.
Input Data
System ID 1~50
Item
No.
01 System Name For reference only. Blank
02 Connected State For reference only. 0
03 IP Address For reference only. 000.000.000.000
04 MAC Address For reference only. 00:00:00:00:00:00
05 Primary Priority Level For reference only. 0
06 Main Software Version For reference only. XX.XX
Conditions

None

Issue 5.0

51-12 : NetLink Primary System Information
Level:
IN
Description
Use Program 51-12: NetLink Primary System Information to reference information
about the Primary System in the NetLink network.
Data Input
Item
No.
01 System ID For reference only. 0
02 System Name For reference only. blank
03 IP Address For reference only. 000.000.000.000
04 MAC Address For reference only. 00:00:00:00:00:00
05 Primary Priority Level For reference only. 0
06 Main Software Version For reference only. XX.XX
Conditions

None

Issue 5.0

51-13 : NetLink Options
Level:
IN
Description
Use Program 51-13: NetLink Options to enable automatic IP address List
Operation updates, time zone information, and MAC address authorization.
Data Input
Item
No.
01 Automatic IP Address List Update 0 = Disable (Off) 1

When set to 1, the list in Program 51-03-01 is 1 = Enable (On)
automatically updated.
02 Time Zone Method 0 = Disable (Off) 0

When set to 0, the following features are 1 = Enable (On)
affected: Clock Display, Incoming/Outgoing
History List.
When set to 1, the following features are
affected: VRS Time Announce, Date and Time
Setting Service Code, Alarm Clock setting, and
Hotel mode wake-up call.
03 MAC Address Authorization 0 = Disable (Off) 0

Refer to Program 51-02-04 to set MAC address. 1 = Enable (On)
Conditions
None

None

Issue 5.0

51-14 : NetLink System Control
Level:
IN
Description
Use Program 51-14: NetLink System Control to delete system and slot
information.

NOTE
Programming.
Input Data
System ID 1~50
Input Data
Menu Number 1 = System information deletion
Item
No.
01 Delete System Information 1 blank

Delete system information and the slot
information. The system must be disconnected.
Conditions
None

None

Issue 5.0

51-15 : Demonstration Setting
Level:
IN
Description
Use Program 51-15: Demonstration Setting to automatically set the minimum
setting values in NetLink. A system reset occurs after this command is executed..

NOTE
Programming.
Input Data
Menu Number 1 = Primary automatic setting

2 = Secondary 1 - automatic operation setting
Conditions
None

None

Issue 5.0

51-16 : NetLink System Data Replication Settings
Level:
IN
Description
Use Program 51-16: NetLink System Data Replication Settings to set the system
data replication between the Primary and Secondary systems.
Input Data
Item
No.
01 System Data Replication Mode 0 = Disable 0

Set the synchronous mode of the system data. 1 = Setting Time Mode
When set to 1, the systems are synchronized at 2 = Interval Mode
the time set in Item 02 below.
When set to 2, the systems are synchronized at
regular time intervals set in Item 03 below.
02 System Data Replication Time Setting 0000~2359 0200

Set the time of day that both systems
synchronize database (when Item 01 is set to 1.)
03 System Data Replication Interval Setting 15~1440 (minutes) 30 (min)

Set the time interval that both systems
synchronize database (when Item 01 is set to 2).
Replication Time Stamp Month: 0~12 –

Show next replication time. (Read-Only)
Day: 0~31 –
04
Hour: 00~23 –
Minute: 00~59 –
05 System Data Replication Wait Time 1~86400 (seconds) 180 sec

Set the wait time until replication starts when
NetLink is created.
06 System Data Replication Interval 0~86400 (seconds) 1 sec

Set time to start replication to the next node after
replication to one node is completed.
.
In a NetLink system with In-Mail, replication should be scheduled during
non-peak hours of operation
IMPORTANT

Issue 5.0
Conditions
None

None

Issue 5.0

51-17 : NetLink DT800/DT700 Server Individual Information Setup
Level:
IN
Description
Use Program 51-17: NetLink DT800/DT700 Server Individual Information Setup
to set the NetLink port information.
Input Data
System ID 1~50
Input Data
Item
No.
01 Register Port 0 ~ 65535 5080

Use to set the SIP Register Port of each system.
02 Subscribe Session Port 0 ~ 65535 5081

Use to set the SIP Subscribe Session Port
number of each system when NetLink is used.
Conditions
None

None

Issue 5.0

51-18 : NetLink Config Setting
Level:
IN
Description
Use Program 51-18: NetLink Config Setting to set the NetLink Fail-Over limits.
Input Data
Item
No.
01 NetLink Fail-Over Limit 0, 2~10 0

When tear-down of network was repeated more (0 = Infinity)
than the specified times, NetLink is operated
stand-alone.
Conditions
None

None

Issue 5.0

51-19 : NetLink IP Trunk (SIP) Calling Party Number Setup for Extension
Level:
IN
Description
Use Program 51-19: NetLink IP Trunk (SIP) Calling Party Number Setup for
Extension to set CPN transmission for each secondary system.
Input Data
Item
No.
01 NetLink CPN Transmission 0 = Disable 1

This program assigns transmission of Calling 1 = Enable
Party Number (CPN) from PRG 21-19 for each
secondary system. The transmission applies for
every extension.
Conditions
None

None

Issue 5.0

90-57 : Backup Recovery Data
Level:
IN
Description
Use Program 90-57 : Backup Recovery Data to backup the system data in the
flash memory on the GCD-CP10 and to make the recovery data.
Input Data
Data ID 1~5
Item
Item Input Data
No.
01 Backup Recovery Data [Backup?] : Dial 1+ press Transfer

(Press Transfer to cancel.)
Conditions
None

None

Issue 5.0

90-58 : Restore Recovery Data
Level:
IN
Description
Use Program 90-58 : Restore Recovery Data to select the recovery data stored in
the flash memory of the GCD-CP10. After this command is executed, the system
restarts automatically.
Input Data
Data ID 1~5
Item
Item Input Data
No.
01 Restore Recovery Data [Restore & Reset?] : Dial 1+ press Transfer

Conditions
None

None

Issue 5.0

90-59 : Delete Recovery Data
Level:
SA
Description
Use Program 90-59 : Delete Recovery Data to select and delete the recovery data
stored in the flash memory of the GCD-CP10.
Input Data
Data ID 1~5
Item
Item Input Data
No.
01 Delete Recovery Data [Delete?] : Dial 1+ press Transfer

Conditions
None

None

UNIVERGE® SV9100
NAPT Chapter 11
SECTION 1 NAPT
1.1 Introduction
NAPT, or Network Address Port Translation, is a method by which a private

address, or addresses, and their TCP/UDP ports are translated into a single
public address and its TCP/UDP ports. In the case of DT800/DT700 IP phones
with the SV9100, it allows their connection to a public (Internet) IP address which
is then converted back to the private (non-Internet) IP address on the customers
network. The translation is available at the SV9100 end as well as at the remote
IP Phone end of the connection if required. The feature is NOT available for IP-
CCIS and Netlink connections.
The router that the SV9100 resides behind still requires Port Forwarding
statements. However the router that the IP terminal/terminals reside behind may
not require any port forwarding.
Due to the fact that there are many manufacturers producing routers there may
still be times when port forwarding is required.
This feature is only effective when PRG 15-05-45 set to a "1" (ON).
In all software versions, SIP ALG’s (Application Level Gateway) or other SIP
Applications MUST be disabled in all routers. If a SIP ALG or a similar SIP
application is enabled, IP phone service WILL be interrupted.

Issue 5.0
The example below is with 256 DSP Resources licensed to the GCD-CP10. Your
Port forwarding range could change depending on number of DSP Resource
Licenses.
Figure 11-1 NAPT Configuration Example
1.2 Requirements
The following information provides requirements for NAPT.
1.2.1 Main Software
NAPT is supported with the V1000/V1.00 or higher release of the SV9100.
1.2.2 Hardware
The SV9100 requires the following hardware:
 GCD-CP10
 GPZ-IPLE
1.2.3 Capacity
896 Extensions (as allowed by hardware and licensing).
11-2 NAPT
Issue 5.0
1.2.4 License
 The system must be licensed for this feature (License 0031).
 The system must be licensed for DT800/DT700 ITL Terminals.
 IP Terminal License (5111)
 System Port License (0300)
 Version 1 License (0411)
1.3 Installation
The following settings have been added for NAT traversal in the DT800/DT700
ITL Terminal.
 Terminals must be V3.0.0.0 or higher to support the NAPT feature.
1.3.1 Settings for Terminals
NAT Traversal 2. SIP Settings/ 8. NAT Traversal [Eco, Value, Sophisticated]
Number
Setting Default Factory Auto
and Name Description
Value Value Value Config
of Setting
1. NAT 1. Disable 1. Available Available Settings for NAT Traversal:

Traversal 2. Dynamic Disable Disable: Disables NAT Traversal
Mode 3. Static Dynamic: Performs NAT
Traversal using a dynamic
conversion table.
Static: Performs NAT Traversal
using a static conversion table and
requires the WAN IP Address to
be entered into the IP Terminal.
2. Network A character No value Available Available The name of the network area to
Area Name string which terminals belong. Specify
up to 32 alphanumeric characters.
(Optional step)
3. WAN See table below.
Settings

Issue 5.0
WAN Settings: 0. Config/ 2. SIP Settings/ 8. NAT Traversal/ 3. WAN

Settings [Eco, Value, Sophisticated]
Number and
Setting Default Factory Auto
Name of Description
Value Value Value Config
Setting
1. WAN Mate IP Address 0.0.0.0 Available Available WAN Address of the router that
IP Address the SV9100 resides behind. This
setting must match what is
programmed in PRG 10-12-07.
2. WAN SIP 1024~6553 5060 Available Available Port number the SV9100 uses for
Mate Port 5 SIP registration in PRG 10-46-
06.
3. WAN Self IP Address 0.0.0.0 Available Available Only used when Static NAT is
IP Address enabled.
This setting is the WAN address
of the router that the NAPT
Terminal resides behind.
Self Port Settings: 0. Config/ 1. Network Settings/ 6. Advanced Settings/ 5.

Self Port Settings [Eco, Value, Sophisticated]
Number
Default Factory
and Name Setting Value Auto Config Description
Value Value
of Setting
1. RTP Self 1024~65528 3462 Available Not available The number of the port
Port (Even receiving RTP data.
numbers Note: At default this is
only) assigned to port 3462. The
first IP phone on this local
LAN can use this port. The
second IP phone would need
to be changed to port 3464.
2. SIP Self 1024~65535 5060 Available Not available The number of the port
Port receiving SIP data.
Note:At default this is
assigned to port 5060. The
first IP phone on this local
LAN can use this port. The
second IP phone would need
to be changed to port 5062.
11-4 NAPT
Issue 5.0
1.4 Programming Example
NAPT, or Network Address Port Translation, is a method by which a private address

or addresses and their TCP/UDP ports are translated into a single public address and
its TCP/UDP ports. In the case of IP phones with the SV9100 it allows their
connection to a public (internet) IP address which is then converted back to the
private (non-internet) IP address on the customers network. The translation is
available at the SV9100 end as well as at the remote IP Phone end of the connection
if required. The feature is NOT available for IP-CCIS and Netlink connections.
Note 1: The NAPT feature requires the following licenses to be loaded to the CPU: IP
Terminal (5111), Version 1000 (0411), VoIP Channel (5103), and System Port (0300).

Issue 5.0
11-6 NAPT
Issue 5.0

Issue 5.0
11-8 NAPT
Issue 5.0

Issue 5.0
11-10 NAPT
Issue 5.0

Issue 5.0
SECTION 2 CONDITIONS
 NAPT is supported for DT800/DT700, and Standard SIP Terminals.

 “License Exceeded” will display on the DT800/DT700 terminal when trying to
register a NAT phone, if the feature is not licensed.
 Terminals using NAPT must be at firmware V3.0.0.0 or higher.
 IP terminals can be connected via NAT router or WAN (direct connection).
 The NAT router on the SV9100 side will require port forwarding.
 K-CCIS IP does not use NAPT translations.Softphone (SP310) is not supported.
 If NAPT enabled phones become unresponsive after being idle, the timer in PRG
15-05-47 and 15-05-48may need to be changed to a shorter interval. NAPT
terminals will ignore the timers in PRG 84-23-01 and 84-23-02 and use the timers in
PRG 15-05-47 and 15-05-48 instead.
 It is necessary to set Program 10-46-14 to OFF when the IPLE is assigned a global
(public) IP address.
 When Program 10-46-14 is set to ON, it references programs 10-58-01 and 10-58-
02. These programs are used to define any destination networks that do not get
sent through the NAPT translations.
 UDP ports in the remote routers may be required to be forwarded to the IP
Terminals.
 NAPT can be used for SIP trunks and terminals on the same system, however they
require the same WAN Address in PRG 10-12-07.
 When Program 15-05-45 is set to “1” the manual table setting for port forwarding
may not be required on the remote side Router B, but the router must support the
NAT function setting itself. If Program 15-05-45 is set to “0” port forwarding at the
Remote side router is required.
 The router may close the port being used if packet exchange is not performed
during a certain time frame. In this case, change Programs 15-05-47 and 15-05-48
to a shorter interval. These programs are changed on a per station basis. No- NAPT
phones will still use Programs 84-23-01 and 84-23-02 while only NAPT phones will
use Programs 15-05-47 and 15-05-48.
 The SIP-ALG or similar SIP application function of all routers/firewalls in the network
must be disabled.
 After starting SIP negotiation for the call, if any RTP packet can’t be received from
the terminal within 10 seconds, the call is dropped and the IP Terminal displays the
following:
11-12 NAPT
Issue 5.0
 Each NAPT terminal can have a separate Register and Subscribe expire timer. The
load of the CPU will increase with each NAPT terminal using a short timer. The
following chart shows the minimum timer settings based on the number of NAPT
terminals using PRG 15-05-47 and 15-05-48.
SECTION 3 RESTRICTIONS – STATIC NAT
 With static NAT, the terminal needs a static IP Address assigned to it, or entries in
the DHCP must be made to provide the same IP Address to the terminal.
 The NAT router on the terminal side must have the function for setting up static NAT.
 A conversion table must be manually set up for the NAT router on the terminal side.
 If installing multiple terminals in the domain of the NAT router on the terminal side,
the SIP port number and RTP/RTCP port number for each terminal must be
specified so as to avoid overlapping.
 The SIP server cannot be switched. (Only one address can be registered as the SIP
server.)
 Encryption with the SV9100 IPLE is not supported with IP terminals connected via
NAPT.
 NetLink Failover is not supported.
Dynamic NAT
 The NAT router on the terminal side must have the function for setting up dynamic
NAT.

Issue 5.0
 It is assumed that port numbers are not changed by the NAT router on the terminal
side. If a port number is changed by NAT router, NEC does not guarantee proper
operation.
 If installing multiple terminals in the domain of the NAT router on the terminal side,
the SIP port number and RTP/RTCP port number for each terminal must be
specified so as to avoid overlapping.
 The SIP server cannot be switched. (Only one address can be registered as the SIP
server.).
 NetLink Failover is not supported.
SECTION 4 MINIMUM REQUIRED PROGRAMMING
 Level 2 – these are the next most commonly assigned programs for this feature.
Level
Program
Program Name Description/Comments Assigned Data
Number
1 2 3
10-12-01 GCD-CP10 Network Setup – Assign the IP Address for the CPU 0.0.0.0 ~
IP Address NIC card. When a IPLE card is 126.255.255.254
installed in the system it is 128.0.0.1 ~
recommended to set this PRG to 191.254.255.254 
0.0.0.0 192.0.0.1 ~
223.255.255.254
(default = 172.16.0.10
10-12-03 GCD-CP10 Network Setup – Assign the default gateway IP 0.0.0.0 ~

Default Gateway address for the GCD-CP10. 126.255.255.254
128.0.0.1 ~
191.254.255.254 
192.0.0.1 ~
223.255.255.254
(default = 0.0.0.0)
10-12-07 GCD-CP10 Network Setup – Define the IP Address of the WAN 0.0.0.0 ~
NAPT Router IP Address side of the router. 126.255.255.254
(Default Gateway [WAN]) 128.0.0.1 ~
191.255.255.254 
192.0.0.1 ~
223.255.255.254
(default = 0.0.0.0)
11-14 NAPT
Issue 5.0
Level
Program
Number
1 2 3
10-12-09 GCD-CP10 Network Setup – IP Set for IPLE 0.0.0.0 ~

Address The IP Address assigned in 126.255.255.254
Program 10-12-01 cannot start 128.0.0.1 ~
with the same leading digits as the 191.255.255.254 
IP Address assigned here. 192.0.0.1 ~
223.255.255.254
(default = 172.16.0.10)
10-12-10 GCD-CP10 Network Setup – Define the Media Gateway Subnet 128.0.0.0
Subnet Mask Mask Address. 192.0.0.0
224.0.0.0
240.0.0.0
248.0.0.0
252.0.0.0
254.0.0.0
255.0.0.0
255.128.0.0
255.192.0.0
255.224.0.0
255.240.0.0
255.248.0.0
255.252.0.0
255.254.0.0
255.255.0.0
255.255.128.0 
255.255.192.0
255.255.224.0
255.255.240.0
255.255.248.0
255.255.252.0
255.255.254.0
255.255.255.0
255.255.255.128
255.255.255.192
255.255.255.224
255.255.255.240
255.255.255.248
255.255.255.252
255.255.255.254
255.255.255.255
(default = 255.255.0.0)
15-05-50 IP Phone Basic Setup - Peer-to- Enable/Disable the Peer-to-Peer 0 = Off

Peer Mode feature for the IP station. 1 = On (default)


Issue 5.0
Level
Program
Number
1 2 3
10-46-01 DT800/DT700 Server Information Normal mode: 0 = Normal (default)

Setup - Register Mode 1 = Auto
When the phone boots up, it 2 = Manual
reports the ext. assigned in the
phone or chooses the next
available extension in the system.
No password is required.
Auto mode:
If set to Auto, the SIP user name

and password must be entered in
the actual IP phone. These
settings must match 84-22/ 
15-05-27, for the phone to come
on-line.
Manual mode:
When the phone boots up it
prompts a user to enter a user ID
and password before logging in. It
checks the user ID/password
against 84-22/15-05-27. If there is
no match, the phone does not
come on-line.
10-46-06 Register Port 0~65535 5080

Assign the port number to which
the SIP messages are sent to on
the IPLE. This same port number
must be assigned in the SIP

Multiline terminals. If this command
is changed, it requires a CPU
reset.
10-46-13 Subscribe Session Port 0~65535 5081 

10-46-14 NAT Mode Turns On/Off the NAT mode of the 0 = Off
system. 1 = On 
Default = Off
10-58-01 Network Address This program sets the local area 0.0.0.0 ~
network of the DT800/DT700 126.255.255.254
terminal when the system has a 128.0.0.1 ~
NAT router and a local router. 191.255.255.254

192.0.0.1 ~
223.255.255.254
11-16 NAPT
Issue 5.0
Level
Program
Number
1 2 3
10-58-02 Network Address - Subnet Mask This program sets the local area 248.0.0.0 / 252.0.0.0 /
network of the DT800/DT700 254.0.0.0 / 255.0.0.0
terminal 255.128.0.0 /
when the system has a NAT router 255.192.0.0 /
and a local router. 255.224.0.0
255.240.0.0 /
255.248.0.0 /
255.252.0.0
255.254.0.0 /
255.255.0.0 /
255.255.128.0
255.255.192.0 /
255.255.224.0 
255.255.240.0 /
255.255.248.0
255.255.252.0 /
255.255.254.0
255.255.255.0 /
255.255.255.128
255.255.255.192 /
255.255.255.224
255.255.255.240 /
255.255.255.248
255.255.255.252 /
255.255.255.254
15-05-45 IP Telephone Terminal Basic This Program is valid when 0 = Disable (default)
Data Setup - NAT Plug and Play Program 10-46-14 is On (NAT 1 = Enable
feature activated). Select sending
RTP port number to remote
Router, use from negotiation result
(0) or received RTP packet (1).
SV9100 uses this program to
decide a destination port of RTP
transmitting packets from IPLE to
a remote IP terminal. If “0:OFF” is

selected, the destination port of
RTP transmitting packets will be a
SIP/SDP negotiation result.(same
behavior as before). If you chose
“1:ON”, the destination port of RTP
transmitting packet will be the
same port of a source port of a
receiving RTP packet on IPLE.

Issue 5.0
Level
Program
Number
1 2 3
15-05-47 Registration Expire Timer for On a per station basis, this setting 0 = Disable
NAPT defines the SIP registration expiry 1 = 60 - 65535 (sec)
timer. This setting only applies to (default - 180)
DT800/DT700 stations connected
via NAPT. If this value is set to 0,

for a NAPT terminal, the value in
PRG 84-23-01 is applied.
15-05-48 Subscribe Expire Timer for On a per station basis, this setting 0 = Disable
NAPT defines the SIP subscribe expiry 1 = 60 - 65535 (sec)
timer. This setting only applies to (default - 180)
DT800/DT700 stations connected
via NAPT. If this value is set to 0,

for a NAPT terminal, the value in
PRG 84-23-02 is applied.
84-26-01 IPL Basic Setup – IP Address When using a GPZ-IPLE assign Slot 1 = 172.16.0.20
only one IP address.

84-26-02 IPL Basic Setup – RTP Port Assign the RTP port number to be VoIP GW1 = 10020
Number used for each DSP on the IPLE.
 Only even numbered ports

are supported.
84-26-03 IPL Basic Setup – RTCP Port Define the TCP port number for VoIP GW1 = 10021
Number (RTP Port Number + 1) RTCP to use for each DSP.

11-18 NAPT
UNIVERGE® SV9100
All DSP Busy Indication Chapter 12
The All DSP Busy feature is used to alert users via telephone displays and/or Alarm
reports when all DSP resources in the system are being used. This can be used to
trouble shoot issues or to alert when the current hardware might need to be
upgraded to a higher capacity.
Alarm Message Format:
The Alarm message for will vary depending on what type of resource is unavailable
and whether the system is stand alone, Netlink or CCISoIP. This will be displayed on
telephones and included in printed or emailed Alarm Reports.
Table 12-1 Alarm Types
Parameters Description
STA DSP for IP Station Calls Were All Busy.

TRK DSP for Trunk Calls Were All Busy, includes SIP and
CCISoIP Trunks.
NET DSP for CCISoIP Networking Calls Were All Busy.
LNK SP for NetLink Calls Were All Busy.

Issue 5.0
Alarm Report Example:
The report example below shows an alarm for all busy Station and Trunk DSPs.
<< ALARM REPORT >> 01/22/2009 23:37

LVL NO STAT DATE TIME ITEM UNIT SLT PRT PARAMETER
-------------------------------------------------------------------------------
MAJ 0068 ERR 01/22/09 09:30 VoIP All DSP Busy VoIPDB 01 00 STA
MAJ 0068 ERR 01/22/09 09:31 VoIP All DSP Busy VoIPDB 01 00 TRK
MIN 0002 REC 01/22/09 09:32 PKG Installation PRIU 02 00
LCD Display
LCD Display Indication Note
Clock/Calendar Non-NetLink System

VoIP DSP All Busy XX
Soft Key XX = Slot number of CCPU with IPLE blade.
Clock/Calendar NetLink System

VoIP DSP All Busy YY-XX YY = System ID
Soft Key XX = Slot number of CCPU with IPLE blade.
 When using IP Phones, the alarm is shown on both terminals involved in that call if
they are both on the same system, this includes NetLink systems.
 The alarm cannot be displayed on Standard SIP Phones, Soft Phones or Single
Line phones.
 If a call from a Standard SIP telephone to a Multiline telephone cannot be
established due to an All DSP Busy condition, the Multiline telephone will not
display the “All DSP” busy message.
 If a SIP trunk call is sent to the SV9100 when all DSP resources are busy, the call is
rejected but the alarm is not displayed on any system telephone.
 The default alarm setting is Minor.
 If an all DSP condition is encountered when making a call across a CCISoIP
Network, only the calling stations receives the alarm indication.
12-2 All DSP Busy Indication

Issue 5.0
SECTION 3 RELATED FEATURES
 IP Multiline Station (SIP)

 IP Trunk – (SIP) Session Initiation Protocol
 K-CCIS – IP
 SV9100 NetLink
Level
Program
Number
1 2 3
20-13-52 VoIP All DSP Busy Display Set on a per station class of 0 = Off
service basis, whether the “All 1 = On
DSP Busy” alarm displays on the (default = 1)
LCD when the caller makes an IP 
call and there is no VoIP DSP
resource.
90-10-01 System Alarm Setup Alarm Number 68 is used for All 0 = Not set
DSP Busy. 1 = Major
2 = Minor

(default = 2)

Issue 5.0
12-4 All DSP Busy Indication

UNIVERGE® SV9100
AspireNet
Chapter 13
Introduction
1.1 What is AspireNet?
The AspireNet package provides a seamless connection of multiple systems into a single “virtual”
communications system using ISDN (PRI/BRI) and VoIP lines with a unified numbering plan.AspireNet
will allow many companies to connect their telephone systems so they appear as one. This will give
them the ability to have only one operator to manage the system and share one voice mail within the
network. An extension user in the network can easily dial another extension or transfer a call within the
AspireNet System. Calls are passed from network node to network node using a protocol that contains
information about the source of the call, the type of call and the destination of the call.
Centralized Voice Mail

Centralized Voice Mail allows multiple networked systems to share a single voice mail system. This
centralized voice mail can receive calls from and transfer calls to any destination in any network node.
Unanswered calls recall and route as if they were part of a single, much larger system.
Flexible Network Routing

Use network routes to set up “single channel” networking between many separate systems - or use
multiple networking channels per system for greater network performance. Data tables in the system
program define the routing for each extension in each network node. These tables are easily
customized to meet the requirements of each networking configuration.
Users may place an intercom call or transfer a call to any extension at any location by simply dialing
an extension number. The system analyses each extension number received and determines how to
route the call to its final destination. The feature which handles this route selection is called Flexible
Routing (F-Routing). Once an extension number is dialed, the system checks the routing, accesses
the assigned trunk group and places the call. Each extension is assigned a route or routes that
decides which trunk group to access and any modified dialed data if required.
Busy Lamp Indication

The status of an extension will be shown at a Hotline key/DSS Console on another networked system.
This allows a Centralized Operator to have lamp indication of extensions in the network or an
extension user to have a Hotline key for a co-worker on another system.
Centralized Operation
Centralized Operator allows multiple networked systems to share a single operator. The operator can
be accessed by a single digit code, if the operator is busy your call will automatically queue until the
operator becomes free. The operator can have a DSS console to show the status of users anywhere
in the network.

Issue 5.0
Paging
AspireNet allows a user to place Paging call to a networked system. If you need to get through to a co-
worker who is not at their desk then place a paging call to their system.
Call Forwarding
You can forward your calls to an extension at another networked extension. If you visit another site
within your network but forgot to set a call forward then use Follow Me to have your calls forwarded to
you. Follow Me is also useful if you have an DECT handset. If your handset is subscribed at the site
you are visiting then you can use Follow Me to have your calls forwarded to your DECT handset.
Direct Dial In(DDI)

A DDI call can be routed to any extension within the network. This allows an extension to receive a
DDI call from any other system in the network. Along with trunk access it is possible for an extension
on a system that has no trunk lines to use the trunks of another system in the network.
Trunk Access
An extension can access a trunk line at another system in the network. The user dials the standard
trunk access code and the system will automatically route the call to the system that has trunks
connected.
Conference
An extension can have a conference call that includes co-workers at another system within the
network.
1.2 SV9100 PCBs
There are two methods available for AspireNet connection as shown in the following table.
Interface Description Note
ISDN Using Q.931 based proprietary protocol, A PRIA or BRIA Blade is

Basic rate interface and Primary rate required for connection.
interfaces are available.
VoIP Using H.323 protocol for voice transmit An IPLE daughter board is
protocol. required.
ISDN AspireNet
Using ISDN AspireNet the system provides up to 256 B-channel ports which can be used for
Networking.
A PRIA circuit will take 30 ports and each BRI circuit will take 2 ports.
These ISDN AspireNet ports are independent of the trunk and station ports available on the system.
IP AspireNet
Using IP AspireNet the maximum quantity of simultaneous calls is limited by the availability of
resources on the IPLE PCB’s installed. A GPZ-IPLE PCB’s giving a maximum of 256 speech
channels.
The maximum quantity of calls may also be reduced by the compression mode (CODEC type) of the
IPLE PCB’s, this is selectable by the installer in Program 84-12-28.
13-2 AspireNet
Issue 5.0
SV9100 System Requirements

AspireNet is supported on the GCD-CP10 blade
1.3 Available Features

Feature Name
ARS/F-Route
Barge In
BLF Indication (at Hotline / DSS Console key)
Call Forwarding Override
Caller ID Display
Call Forwarding:
Immediately
Busy
No Answer
Both Ring
Call Forwarding, Follow Me
Call Forwarding, Off-Premise
Call Waiting / Camp On
Callback
Central Office Calls, Placing: Seizing a trunk in networked system
Conference
Department Step Calling
Direct Dial In (DDI)
Direct Inward Line (DIL)
Direct Inward System Access (DISA)
Department Calling
Hold
Hotline / DSS Console
Intercom: Change Voice/Signal Ring
Keep Alive Operation
Last Number Redial
Message Waiting
Operator, Centralized
Paging
Park
Ringdown Extension, Internal/External
Selectable Display Messaging
Toll Restriction
Transfer
Voice Mail, Centralized

Issue 5.0
1.4 Using This Manual
This Chapter is in four sections:

Section 1: Introduction
This section provides an Introduction for the AspireNet.
Section 2: Setting Up AspireNet

This section guides you step by step in setting up a basic AspireNet system.
Section 3: AspireNet Features

This section provides a description of each of the AspireNet Features
Section 4: Programming
This section describes all of the programming commands required to install an AspireNet system.
Section 5: Examples of AspireNet Configurations

This section shows diagrams and simple programming instructions for AspireNet.
Telephone Programming Instructions shows you how to enter the program’s data into system
memory. For example:
2. 15-07-01
15-07-01 TEL301
KY01 = *01

tells you to enter the programming mode, dial 150701 from the telephone dial pad. After you do, you’ll
see the message “15-07-01 TEL301” on the first line of the telephone display. This indicates the program
number (15-07), item number (01), and that the options are being set for extension 301. The second row
of the display “KY01 = *01” indicates that Key 01 is being programmed with the entry of *01. The third
row allows you to move the cursor to the left or right, depending on which arrow is pressed. To learn how
to enter the programming mode, see 4.3 How to Enter Programming Mode on page 13-55.
1.5 Unique Considerations

Read These Notes
Simplifying Keyset Operation with One-Touch Keys...
A system phone user can access many features through Service Codes (e.g., Service Code 841
answers a Message Waiting from a co-worker). To streamline the operation of their phone, a system
phone user can store these codes under One-Touch Keys. This provides one-button operation for almost
any feature. To find out more, read the One-Touch Calling feature in your Feature & Specifications
Manual.
Programmable Keys...
13-4 AspireNet
Issue 5.0
When reading an instruction using programmable keys, you will see a notation similar to (PGM 15-07 or
SC 851: 06). This means that the key requires function code 06, and you can program this code through
Program 15-07 or by dialing Service Code 851. Refer to the Programmable Function Keys feature in
your Features and Specifications Manual if you need more information.
Using Handsfree...
The manual assumes each extension has Automatic Handsfree. This lets a user just press a line key or
Speaker key to answer or place a call. For extensions without Automatic Handsfree, the user must:
Lift the handset or press SPK for Intercom dial tone
Lift the handset or press SPK, then press a line key for trunk dial tone
SECTION 2 SETTING UP ASPIRENET
2.1 Required System Programming
The selection of an ISDN PRIA, BRIA, or IPLE Blade determines the type of programming you must
do on the SV9100.
Refer to either the AspireNet ISDN or AspireNet IP section below.
Also refer to the examples in the Section 5.
2.2 AspireNet ISDN

° 10-03-01 : PCB Setup - ISDN Line Mode
Determine the line mode of the ISDN. If Basic Rate Interface (BRI) is chosen, the setting must be done for each
line. The settings must match at each end of the networked line. The following entries are acceptable for
Networking.
ISDN Line Number 1-4
Item
No.
01 ISDN Line Mode 1 = T-Point 1

2 = S-Point
3 = Network Mode (Leased Line)
4 = Network Mode (Interconnected Line)
5 = Interconnection (Interconnected Line,
Fixed Layer 1 Forced NT Mode)
A PRIU interface will provide up to 30 channels, the BRIU interface will provide 2 channels. Program 10-32-01
can limit the quantity of channels available for PRIU interfaces.

Issue 5.0
The installation of each mode is shown in the following diagrams.
Mode 3 BRI/PRI Network Mode (Leased Line)
SV9100 SV9100
Mode 3 NT1 BRI Leased Line NT1 Mode 3
Master side Slave side
T<-S S->T
Slave Clock signal Slave

Generator
Mode 4 BRI/PRI Network Mode (Interconnected Line).
SV9100 SV9100
Mode 4 Mode 4
Master side Slave side
S->T
Clock signal Slave
Mode 5 BRI/PRI Interconnection Mode (Interconnected Line, Layer 1 = NT).
SV9100 SV9100
Mode 5 GW Ethernet GW Mode 4
Slave side Master side
S<-T T->S
Clock signal Slave Slave Clock signal

Generator Generator
° 10-03-03: PCB Setup - Connection Type

The connection type should be changed if Basic Rate Interface (BRI) is used. Only Point-to-Point connection
(1) is available for system interconnection.
Item
No.
13-6 AspireNet
Issue 5.0
03 Connection Type 0 = Point-to-Multipoint (not available for 0

Networking)
1 = Point-to-Point
Example:
System – A System – B
1: Point-to-Point 1: Point-to-Point
° 10-03-10 : PCB Setup - Master/Slave System

Determine which system will be the master system and which one(s) will be the slave system(s). If one end of
the line is set as the Master, the other end of the line must be set as the Slave. The choice of Master/Slave is
determined by the ISDN clock available at the SV9100 System, see the Section 5 for further detail.
Item
No.
10 Master/Slave System 0- Slave System 0
(Network Mode Only) 1- Master System
Example:
1: Master 2: Slave
° 10-03-11 : PCB Setup - Networking System Number

The Networking ID is used to select the access route for dialed digits. You can choose any number 0 to 50 (0
equals no operation). This ID is used when setting the numbering plan for the networked systems. The same ID
number must be set in both 10-03-11 and 11-01. Refer to 2.5 Numbering Plan in on page -12 for more on
the numbering plan settings.
Item
No.
10 Networking System Number (Net- 0-50 0
work Mode Only)
Example:
Networking ID: 1 Networking ID: 1
2.3 AspireNet IP
As with any Voice Over IP (VoIP) implementation, there are several

issues that should be considered when setting up AspireNet IP
CAUTION

Issue 5.0
2.3.1 System IP Address

° 10-12-09 : GCD-CP10 Network Setup - IPL IP Address (IPLE)
Select the IP address of the IPLE Blade (default: 172.16.0.10). A static IP address is required. Each SV9100
system within the network must have a unique IP address.
The system must be reset in order for the change to take effect.
° 10-12-10 : GCD-CP00 Network Setup - Subnet Mask (IPLE)
Select the Subnet Mask to be used by the IPLE Blade (default: 255.255.0.0).
Item
Item Input Data Default Conditions
No.
09 IP 1.0.0.1 - 126.255.255.254 172.16.0.1 Set for IPLE.

Address 128.1.0.1 -191.254.255.254 0
(IPLE) 192.0.1.1 - 223.255.254.254
10 Subnet 128.0.0.0 192.0.0.0 224.0.0.0 255.255.0. Set for IPLE.

Mask 240.0.0.0 248.0.0.0 252.0.0.0 0
(IPLE) 254.0.0.0 255.0.0.0 255.128.0.0
255.192.0.0 255.224.0.0 255.248.0.0
255.252.0.0 255.254.0.0 255.255.0.0
255.255.128.0 255.255.192.0 255.255.224.0
255.255.248.0 255.255.252.0 255.255.254.0
255.255.255.0 255.255.255.128 255.255.255.192
255.255.255.224 255.255.255.240 255.255.255.248
255.255.255.252 255.255.255.254 255.255.255.255
Example:
Subnet Mask: 255.255.0.0 Subnet Mask: 255.255.0.0
2.3.2 IPLE Card IP Address

° 84-26-01 : IPLE Basic Setup - IP Address (DSP)
.
Slot Number 1
Item Input Data Default Description Related Program
IP 1.0.0.1–126.255.255.254 Slot 1: 172.16.0.20 Assign the IP address for

Address 128.1.0.1–191.254255.254 each DSP on the GPZ-
192.0.1.1–223.255.254.254 IPLE.
Example:
Subnet Mask: 255.255.0.0 Subnet Mask: 255.255.0.0
13-8 AspireNet
Issue 5.0
2.3.3 IP Address of other systems within the network

° 10-27 : IP System ID - IP Address
The IP Addresses of all other systems within the network must be defined in this program. Each system is
identified by a 'System Number' (1 to 50), this is used within the numbering plan to route calls to another
system.
System ID 01-50
Item Related
No. Program
01 IP Address 1.0.0.1 - 126.255.255.254 0.0.0.0 11-01-03

System ID is related 128.1.0.1 - 191.254.255.254 10-12-01
with the System ID in 192.0.1.1 - 223.255.254.254
the Numbering Plan
(Program 11-01-03).
When the digits are ana-
lyzed and the system ID
is determined from the
system data set in the
Numbering Plan, the
Networking call will be
sent to the IP Address
set in this program.
The IP Address should

be the IP Address of the
peer GCD-CP10 (Pro-
gram 10-12-01).
02 Call Procedure Port 1-65535 1730 84-02-33

The Port Number should
be set with the same
value as the H.225 setup
port in Program 84-02-
33.
Example:
System ID: 1 System ID: 1
Port: 1730 Port: 1730
2.3.4 Networking TCP Port

° 10-20-01 : LAN Setup for External Equipment - TCP Port
Define the TCP port number for communicating to external equipment. The port number defined should be the
same in each networked system.

Issue 5.0
Type of external equipment 1- CTI Server

2- ACD MIS
3- - Reserve -
4- Network System
5- - Reserve -
Item
No.
01 TCP Port 0-65535 External Device 1 = 0

External Device 2 = 0
:
Example:
External Equipment: 4 External Equipment: 4
Port: 30000 Port: 30000
2.3.5 Networking TCP Port

° 84-02-35 : H.225, H.245 Information Basic Setup - Fast Start Mode
If IPLA is used for networking, the Fast Start option must be enabled.
Item
No.
35 Fast Start 0: Disable 1

1: Enable
13-10 AspireNet
Issue 5.0
2.4 AspireNet Multi Site
AspireNet Multi Site is a network of three or more systems connected by either AspireNet ISDN
or AspireNet IP. The configuration for three or more sites is the same as for two sites, refer to
AspireNet ISDN or AspireNet IP for details of setting up the SV9100 systems.
The example below shows three systems, each has a IPLE Blade installed.
The IPLE Blade in each system are connected to an Ethernet hub.
System A has extension numbers in the range 200-299, system B has 300-399 and system C
has 400-499.
Ethernet Hub
System Configuration
IPLA IP Address IPLA IP Address IPLA IP Address

172.16.0.10 172.16.0.11 172.16.0.12
Address for Address for Address for

Remote System Remote System Remote System
System 01 = System 01 = System 01 =
172.16.0.11 172.16.0.10 172.16.0.10
System 02 = System 02 = System 02 =
172.16.0.12 172.16.0.12 172.16.0.11
System System System

Numbering Plan Numbering Plan Numbering Plan
Dial 2x = Dial 2x = Dial 2x =
Intercom Networking Intercom
Networking Intercom Intercom
Networking Networking Networking
VoIP Gateway IP VoIP Gateway IP VoIP Gateway IP

Address Address Address
172.16.0.21 172.16.0.22 172.16.0.23
• All systems within the network must have a direct connection to all other systems in the
network.This makes AspireNet IP the more practical type for Multi Site networks.
• IPLE Resource will be used at each system when a call is transferred. For example a call that
originates from site A and calls site B is held and transferred to site C will use IPLE resources at
all three sites.

Issue 5.0
The same also applies for AspireNet ISDN, an ISDN channel will be used from site A to B and
also from site B to C.
2.5 Numbering Plan

° 11-01-01 : System Numbering
Set the system’s internal (Intercom) numbering plan and system ID to route to networked systems. The
numbering plan assigns the first and second digits dialed and affects the digits an extension user must dial to
access other extensions and features, such as service codes and trunk codes, within a networking node or to
reach another node.
Consider using a "Unified Numbering Plan" for extensions. This gives every extension in the network a unique
extension number. The extension number can then be used to route a call to the correct node. This also allows
the same extension number to be dialed at any node to reach a given extension.
Improperly programming this option can adversely affect system operation. Make sure
you thoroughly understand the default numbering plan before proceeding. If you
CAUTION
k must change the standard numbering, use the chart for Table 13-5 System Numbering
Default Settings on page 13-83 to keep careful and accurate records of your changes.
Before changing your numbering plan, use PC Pro to make a backup copy of your
system data.
Changing the numbering plan consists of three steps:
1. Enter the digits you want to change.

2. Specify the length of the code you select to change.
3. Assign a function to the code selected.
Step 1: Enter the digit(s) you want to change

You can make either single or two digit entries. In the dialed Number column in the table, the nX rows (e.g., 1X)
are for single digit codes. The remaining rows (e.g., 11, 12, etc.) are for two digit codes.
--Entering a single digit affects all the dialed Number entries beginning with that digit. For example, entering 6
affects all number plan entries beginning with 6. The entries you make in step 2 and step 3 below affect the
entire range of numbers beginning with 6. (For example, if you enter 3 in step 2 the entries affected would be
600-699. If you enter 4 in step 2 below, the entries affected would be 6000-6999.)
--Entering two digits lets you define codes based on the first two digits a user dials. For example, entering 60
allows you to define the function of all codes beginning with 60. In the default program, only * and # use two-
digit codes. All the other codes are single digit. If you enter a two digit code between 0 and 9, be sure to make
separate entries for all the other two digit codes within the range as well. This is because in the default program
all the two digit codes between 0 and 9 are undefined.
Step 2: Specify the length of the code you want to change

After you specify a single or two digit code, you must tell the system how many digits comprise the code. This is
the Number of Digits Required column in the table. In the default program, all codes from 100-999 are three
digits long. Codes beginning with 0 are one digit long. Codes beginning with * are 3 digits long and codes
beginning with # are 4 digits long.
13-12 AspireNet
Issue 5.0
Step 3: Assign a function to the code selected

After entering a code and specifying its length, you must assign its function. This is the Dial Type column in the
table. The choices are:
Dial
Dial Type Description Related Program
Types
0 - Not Used -
1 Service Code 11-10: Service Code Setup (for System Administrator)

11-11: Service Code Setup (for Registration)
11-12: Service Code Setup (for Service Access)
11-13: Service Code Setup (for ACD)
11-14: Service Code Setup (for HOTEL)
11-15: Service Code Setup (Special access)
2 Extension Number 11-02: Extension Numbers

11-04: Virtual Extension Numbers
11-06: 2PGDAD (ACI) Extension Numbers
11-07: Department Calling Group Numbers
11-08: 2PGDAD (ACI) Group Pilot Numbers
3 Trunk Access Code 11-09: Trunk Access Code.
4 Special Trunk Access 11-09: Trunk Access Code.
5 Operator Access 20-17: Operator’s Extension
6 ARS/F-Route Access 44-xx
8 Networking 10-03: PCB Setup

10-12: GCD-CP10 Network Setup
10-20: LAN Setup for External Equipment
10-27: IP System ID
--Changing the Dial Type for a range of codes can have a dramatic affect on how your system operates.
Assume, for example, the site is a hotel that has room numbers from 100-399. In order to make extension
numbers correspond to room numbers, you should:
- In Program 11-02, reassign extension numbers on each floor from 100 to 399.
(Other applications might also require you to change entries in Program 11-10 through 11-16.)

Issue 5.0
Example:
This example shows two separate extension numbers assigned for the networked systems. System A dials 4xx
to reach System B, while system B dials 3xx to reach System A.
Dial “3x”: Dial “3x”:

Digit “3” Digit “3”
Type 2 (Intercom) Type 8 (Networking)
System ID “1”
Dial “4x”: Dial “4x”:

Digit “3” Digit “3”
Type 8 (Networking) Type 2 (Intercom)
System ID “1”
The following example shows a unified extension number assignment. All users dial a 4-digit extension number
(2xxx) to reach anyone within the network, regardless of which system they are connected. System A users
have extension numbers 20xx, while system B users have extension numbers 23xx.
Programming System – A System – B
Program 11-01 Dial “2”: Dial “2”:

2x = Digit “0”, Type “0” 2x = Digit “0”, Type “0”
20 = Digit “4”, Type 2 (Intercom) 20 = Digit “4”, Type 8 (Network),
23 = Digit “4”, Type 8 (Network), System ID “1”
System ID “1” 23 = Digit “4”, Type 2 (Intercom)
Program 11-02 Port 1 = extension number 2001 Port 1 = extension number 2301
Port 2 = extension number 2002 Port 2 = extension number 2302
Port 3 = extension number 2003, etc. Port 3 = extension number 2303, etc.
It is also possible to use F-Route to select the correct node for the destination extension number.
The example below shows a numbering scheme where the user must dial an additional digit 7 before the
extension number, this is routed by F-Route to the correct node and analyzed again in the F-Route tables at the
remote SV9100.
When using F-Route you must translate the dialed number (e.g 72301 translates to 2301) otherwise the call will
not ‘exit’ from the F-Route tables.
Programming System – A System – B
Program 11-01 Dial “7”: Dial “7”:

7x = Digit “5”, Type 6 (F-Route) 7x = Digit “5”, Type 6 (F-Route)
Program 11-02 Port 1 = extension number 2001 Port 1 = extension number 2301
Port 2 = extension number 2002 Port 2 = extension number 2302
Port 3 = extension number 2003, etc. Port 3 = extension number 2303, etc.
13-14 AspireNet
Issue 5.0
Program 44-01-01 0 (Not Used) 0 (Not Used)
Program 44-02 Table 1 Table 2 Table 1 Table 2
Program 44-02-01 720@@ 723@@ 720@@ 723@@
Program 44-02-02 2 2 2 2
Program 44-02-03 1 2 1 2
Program 44-05 Table 1 Table 2 Table 1 Table 2
Program 44-05-01 255 101 101 255
Program 44-05-02 1 0 0 1
° Program 11-02-01 : Extension Numbering

Assign the extension numbers to the ports. The extension number can be up to eight digits long. The first/
second digit(s) of the number should be assigned in Program 11-01. This lets an employee move to a new
location (port) and retain the same extension number.
° 44-01-01 : System Options for ARS/F-Route - ARS/F-Route Time Schedule
Set this option to ‘0’ so that the F-Route table selected is determined only by the digits dialed without any
relation to the day or time of the call.
° 44-02-01 : Dial Analysis Table for ARS/F-Route Access - Dial
Set the number of digits to be analyzed by the system for ARS routing. Using the 4-digit extension number
example in Program 11-01-01, the entry would be: Analysis Table 1: 720@@; Analysis Table 2: 723@@.
° 44-02-02 : Dial Analysis Table for ARS/F-Route Access - Service Type
Select the Service Type (0=no setting, 1=extension call, 2=ARS/F-Route table, 3=Dial extension analyze table).
Using the 4-digit extension number example in Program 11-01-01, the entry would be: Analysis Table 1: 2 (ARS/
F-Route table); Analysis Table 2: 2 (ARS/F-Route table).
° 44-02-03 : Dial Analysis Table for ARS/F-Route Access - Additional Data
Enter the additional data required for the service type selected in Program 44-02-02, either the number of digits
to be deleted or the table number to be used. Using the 4-digit extension number example in Program 11-01-01,
the entry would be: 1 (delete 1 digit).
° 44-02-04 : Dial Analysis Table for ARS/F-Route Access - Dial Tone Simulation
If enabled (1), this option sends dial tone to the calling party once the routing is determined. This may be
required if the central office at the destination does not send dial tone. Using the 4-digit extension number
example in Program 11-01-01, the entry would be: 0 (disabled).
° 44-05-01 : ARS/F-Route Table - Trunk Group Number
Select the trunk group number to be used for the outgoing ARS call (0-100, 101-150, 255 [0 = No setting, 101-
150 = Networking, 255 = Extension Call]). Using the 4-digit extension number example in Program 11-01-01,
the entry would be: ARS/F-Route Table Number 1: Priority Number 1, Trunk Group = 255; ARS/F-Route Table
Number 2: Priority Number 1, Trunk Group = 101.
° 44-05-02 : ARS/F-Route Table - Delete Digits
Enter the number of digits to be deleted from the dialed number (0-255 [255 = Delete all]). Using the 4-digit
extension number example in Program 11-01-01, the entry would be: ARS/F-Route Table Number 1: Priority
Number 1, Delete Digits = 1; ARS/F-Route Table Number 2: Priority Number 1, Delete Digits = 0.

Issue 5.0
° 44-05-03 : ARS/F-Route Table - Additional Dial Number Table

Enter the table number (defined in Program 44-06) for additional digits to be dialed (0-1000). Using the 4-digit
extension number example in Program 11-01-01, the entry would be: ARS/F-Route Table Number 1: Priority
Number 1, Additional Dial Number Table = 0; ARS/F-Route Table Number 1: Priority Number 1, Additional Dial
Number Table = 0.
13-16 AspireNet
Issue 5.0
SECTION 3 ASPIRENET FEATURES

Refer to the SV9100 Feature & Specifications Manual for complete description and programming information for
the following features. The information detailed here applies only to the feature when used in a Networked
system
3.1 ARS/F-Route
Digits dialed by a user can be sent to the F-Route tables and specified as an AspireNet number by enter-
ing the Networked node ID (Trunk Group 101-150 correspond to Network ID’s 1-50) as the target trunk
group number, calls will be routed to the target system via the node ID specified. The dialed digits will then
be analyzed by the F-Route tables in the target system. At the target system the call will be analyzed within
F-Route for the following call types:
• Outgoing call to a trunk
• Extension access call (you must translate the dialed digits)
• Access to the other system via AspireNet
• No defined dial
Alternate route selection is not available when the primary AspireNet route is busy. When all channels are
busy the call will return busy tone.
Compared With Single System Configuration

In a single system with F-Route used, the dialing is analyzed when the call is initially dialed.
Operation
With the sample programming shown below, dialing the F-Route access number, which is defined in the F-
Route table (7300), the system calls extension number 300 within System 1 (System ID 1). The tele-
phone’s display initially indicates the F-Route number in progress, then changes to appear as a normal
intercom call.
(Sample Programming for calls from System 1 to System 2, to simplify the example the programming for
calls from System 2 to System 1 has been omitted)
Kind of setting System 1 System 2
PRG11-01-01 Dial “7” Dial “7”

System Numbering Digit “4” (4 digit) Digit “4” (4 digit)
Type “6” (F-Route) Type “6” (F-Route)
PRG44-02- 01 “7300” “7300”

Dial analysis table for “73@@” (using wildcard) is “73@@” (using wildcard) is
F-Route access available. available.
02 “2” (F-Routing) “2” (F-Routing)
03 Add Code = “1” Add Code = “1”
PRG44-05- 01 F-Route table-1 F-Route table-1

System Numbering Trunk Group No. “101” Trunk Group “255”
(Route to System ID 1) (Route to Intercom)
02 Delete Digit: 0 Delete Digit: 1
Notice: In the example above Dial 7 is set to 4 digit number length, this is important for AspireNet as it is possible to

Issue 5.0
set call forward to an F-Route number. The full digit length must be set in Program 11-01 to allow the user to
enter the full F-Route destination number.
For example if 11-01 was set to 1 digit number length in the example above a user who sets call forward can only
enter 1 digit as the destination (e.g 848+1+7). This would cause all calls to the extension to fail whilst the call
forward is set to an invalid destination number.
Notice : When a call is routed via AspireNet by using F-Route the dialed digits MUST be translated otherwise the call
will not ‘exit’ from the F-Route tables.
Notice : When an AspireNet call is routed via F-Route it is possible to translate the dialed digits at either (or both)
the originating or destination system. Programs 44-05-02 (Delete digits) and 44-05-03 (Additional Dial Table)
are available for digit translation.
Related Programs
Program Number Title
11-01 Numbering Plan
44-01 System Options for F-Route Service
44-02 Dial Analysis Table for F-Route Access
44-04 F-Route Selection for time schedule
44-05 F-Route Table
44-06 Additional Dial Table
44-07 Gain Table for F-Route Access
44-08 Time Schedule for F-Route Service
44-09 Weekly Schedule for F-Route Service
44-10 Holiday Schedule for F-Route Service
13-18 AspireNet
Issue 5.0
3.2 Barge In
Barge In is available in the Networking feature with the following options:
• Barge into a conversation between an extension’s own system and a networked system
• Barge into a conversation between callers in a networked system
• Barge into a call between two networked systems
Barge In can be used in either Monitor Mode (Silent Monitor) or Speech Mode (determined in Program 20-
13-10).
Barge In cannot barge into calls across the network in the following instances:
• Conference calls
• Off hook signaling a telephone in the other system
• Barge into an extension’s call without first calling the busy extension in the other system
Operation
To Barge In after calling a busy extension:
An analog trunk call must be set up for 10 seconds before you can Barge In.
Listen for busy/ring or busy tone.
1. Call busy extension.
2. Dial Barge In service code 810
OR
3. Press Barge In key (PGM 15-07 or SC 851: 34).
Related Programs
11-12-08 Service Code Setup (for Service Access) - Barge In
11-16-02 One Digit Service Code Setup - Barge In
20-13-15 Class of Service Options (Supplementary Service) - Barge In, Initiate
20-13-16 Class of Service Options (Supplementary Service) - Barge In, Receive
20-13-17 Class of Service Options (Supplementary Service) - Barge In Tone/

Display

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3.3 Call Forwarding

Call Forwarding Immediate / Busy / No Answer / Busy-No Answer / Both Ring options are available with
the Networking feature.
With a networked system, when Call Forwarding enabled, there is a slight difference in the telephone’s dis-
play. With a single system, the extension name is displayed on the extension which has Call Forwarding.
With a networked system, the extension number is displayed.
Operation
To activate or cancel Call Forwarding:

1. Press idle SPK key (or lift handset).
2. Dial Call Forwarding condition :
842 = Call Forward Both Ring
843 = Call Forward Busy
844 = Call Forward Busy or not answered
845 = Call Forward No answer
848 = Immediate
3. Dial 1 to set Call Forwarding, Dial 0 to cancel Call Forwarding.
4. Dial destination extension, Voice Mail Master Number.
When you enable Call Forwarding, your Call Forwarding key lights.
Your Call Forwarding (Station) Programmable Function Key flashes when Call Forwarding is activated.
OR
1. Press Call Forwarding key.
PGM 15-07 or SC 851: code 10 for Forward All Calls Immediately
PGM 15-07 or SC 851: code 11 for Forward when Busy
PGM 15-07 or SC 851: code 12 for Forward when Unanswered
PGM 15-07 or SC 851: code 13 for Forward Busy/No Answer
PGM 15-07 or SC 851: code 14 for Forward with Both Ringing
PGM 15-07 or SC 851: code 15 for Follow Me
2. Dial 1 plus extension to enable; dial 0 to disable.
Once you activate Call Forwarding, only your Call Forwarding destination can place an Intercom call to you.
3. Dial destination extension, Voice Mail master number.
You’ll hear stutter dial tone when placing a new call.
Your Call Forwarding Programmable Function Key flashes when Call Forwarding is activated.
13-20 AspireNet
Issue 5.0
Related Programs
11-11-01 Service Code: Call Forward – Immediate
11-11-02 Service Code: Call Forward – Busy
11-11-03 Service Code: Call Forward – No Answer
11-11-04 Service Code: Call Forward – Busy/No Answer
11-11-05 Service Code: Call Forward – Both Ring
11-11-06 Service Code: Call Forwarding (Select Option)
20-11-01 Call Forward – Immediate Class of Service
20-11-02 Call Forward – Busy Class of Service
20-11-03 Call Forward – No Answer Class of Service
20-11-04 Call Forward – Both Ring Class of Service
3.4 Call Forwarding/ Do Not Disturb Override

The extension user may be able to call an extension which has Call Forwarding or Do Not Disturb set.
Operation
To override an extension’s Call Forwarding or Do Not Disturb:

1. Call the forwarded or DND extension.
2. Dial Call Forward / DND Override service code 807
OR
2. Press Override key (PGM 15-07 or SC 851: 37).
Related Programs
11-12-01 Service Code Setup (for Service Access) : Call Forwarding / Do Not
Disturb Ovveride
11-16-06 Single Digit Service Code Setup : DND/Call Forward Override
15-07-01 Programming Function Keys
20-06-01 Class of Service for Extensions
20-13-04 Class of Service Options (Supplementary Service) - Call Forwarding/

DND Overrride
3.5 Call Forward/ Off-Premise

Off-Premise Call Forwarding allows an extension user to forward their calls to an off-site location. This fea-

Issue 5.0
ture works the same in a networked system as it does in a single system.

A call to an extension at a remote system will forward to the Abbreviated Dial bin using a trunk at the
remote system.
Operation
To activate Call Forwarding Off-Premise

1. At system phone, press SPK key + Dial Call Forward Service Code (848, 843, 844, 845).
OR
Press Call Forward key (PGM 15-07 or SC 851: 10, 11, 13 or 12)
OR
At an SLT, lift handset Dial 848, 843, 844 or 845.
2. Dial 1.
3. Dial the Abbreviated Dial Bin number (000 to 999) which your calls should be for warded.
4. Press SPK (or hang up at SLT) to hang up.
Your DND and Call Forwarding Programmable Function Key flashes.
To cancel Call Forwarding Off-Premise

1. At system phone, press SPK key + Dial 848, 843, 844 or 845.
OR
Press Call Forward key (PGM 15-07 or SC 851: 10, 11, 13 or 12)
OR
At an SLT, lift handset and dial 848, 843, 844 or 845.
2. Dial 0.
3. Press SPK (or hang up at SLT) to hang up.
Your DND or Call Forwarding Programmable Function Key stops flashing.
Related Programs
13-01 Abbreviated Dial Function Setup
14-05 Trunk Group
14-06 Trunk Group Routing
20-11-12 Enable call forward to Abbreviated Dial bin
21-02 Trunk Group Routing for Extensions
21-16 Trunk Group Routing for Networking
3.6 Call Forwarding with Follow Me

The extension user can program Call Forward Follow-Me to extension in a networked system. When the
extension with the Follow Me setting receives an incoming call, both the original extension and the pro-
grammed destination extension starts ringing.
With a networked system, when Call Forward Follow-Me is enabled, there is a slight difference in the tele-
13-22 AspireNet
Issue 5.0
phone’s display. With a single system, the destination extension displays the extension name for the phone
with Follow-Me enabled. With a networked system, the extension number is displayed.
Operation
To activate Call Forward Follow Me:

1. At a system phone other than your own, press idle SPK key and dial 846..
OR
Press Call Forward (Station) key (PGM 15-07 or SC 851: 15).
OR
At an SLT other than your own, lift handset and dial 846.
2. Dial 1 to set.
3. Dial the Extension to forward.
4. SPK (or hang up at SLT).
Your Call Forwarding (Station) Programmable Function Key flashes when Call Forwarding is activated.
To cancel Call Forward Follow Me:

1. At system phone, press SPK key.
OR
Press Call Forward (Station) key (PGM 15-07 or SC 851: 15).
OR
At an SLT, lift handset and dial 846..
3. Dial 0 (Cancel All Forward Follow Me).
Your Call Forwarding (Station) Programmable Function Key goes out.
Related Programs
11-11-07 Service Code of Follow Me
11-11-06 Call Forwarding (Select Option)

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3.7 Camp On
With Camp On, an extension user may call a busy extension and wait in line (Camp-On) without hanging
up. The call goes through when the busy extension becomes free. Camp On helps extension users to get
through as soon as a busy extension becomes free. It also lets callers wait in line for a busy extension
without being forgotten.
When you have set Camp-On you can choose to wait off hook or go on hook. If you go on hook your phone
will ring when the extension becomes free.
With a networked system, camping on to an idle extension and Trunk Queuing/Camp On for a trunk port
are not supported.
With a networked system, when Camp On is enabled, there is a slight difference in the telephone’s display.
With a single system, the target extension’s name is displayed on the phone which has enabled Call Wait-
ing. With a networked system, the extension number is displayed.
Operation
To Camp-On to a busy extension:

1. Call busy extension.
2. Dial Camp-On service code 850
OR
Press Camp-On key (PGM 15-07 or SC 851: 35).
3. You can choose to hang up or not.
To cancel a Camp-On request:

1. Hang up or SPK key.
2. At system phone, press SPK key and dial 870.
OR
At system phone, press Camp-On key (PGM 15-07 or SC 851: 35).
OR
At single line set, lift handset and dial 870.
Related Programs
11-12-05 Service Code: Setting Camp On
11-12-06 Service Code: Cancelling Camp On
11-16-05 One-Digit Service Code: Camp On
13-24 AspireNet
Issue 5.0
3.8 Caller ID Display

Caller ID information can be sent to the target extension in a networked system and show the Caller ID on
the phone’s display. The ABB Name is also shown on LCD by searching ABB table at the target system.
Operation
A DDI call routed directly to a remote extension will send the Caller ID information to the remote system.
The DDI name set in Program 22-11-03 is also sent to the remote system.
At the remote system the Caller ID information will be displayed or, if the Caller ID number matches an
Abbreviated Dial entry the Abbreviated Dial name will be displayed.
A trunk call that is first answered and then transferred to a remote extension will display the Caller ID num-
ber and the DDI name. The Abbreviated Dial name will NOT be displayed.
Related Programs
° 20-09-02 : Class of Service Options (Incoming Call Service) - Caller ID Display
Define the option whether display Caller ID or not in each station port.
Default
Item Input
Item
No. data COS
COS 15
01-14
02 Caller ID Display 0-Off 1 1

Enables/disables the Caller ID display at an 1-On
extension.
° 20-09-03 : Class of Service Options (Incoming Call Service) - Sub-Address Identification
Define whether an extension displays the Caller Sub-Address.
Default
Item Input
No. Item data COS
COS 15
01-14
03 Sub Address Identification 0-Off 0 0

Define whether an extension displays the Caller 1-On
Sub-Address.
° 13-04 : Abbreviated Dialing Number and Name
Define the abbreviated dialing number and name. The ABB name will be shown on a phone’s LCD when
the ABB table has a matching number with the incoming Caller ID. The common abbreviated number table
is used to search for a match.
Item
No.
01 Abbreviated Dialing Data 1-9, 0, *, #, No Setting

Pause (Press line key 1),
Recall/Flash (Press line key 2),
@ for Additional Digit for ISDN
Functionality (Press line key 3)
(max. 24 digits)
02 Name Max. 12 Characters No Setting

Issue 5.0
3.9 Central Office Calls, Placing: Seizing a trunk in a networked system

The system allows a user to seize a trunk in a networked system using the following methods:
Method of Outgoing Available Note
Specified Trunk Access (805 + the trunk number) No
Specified Trunk Group Access (804 + group number) No
Trunk Route Access (9) Yes
ARS/F-Route Yes See ARS/F-Route
Operation
The operation is automatic, the user dials the trunk access code in the normal way.
Abbreviated Dial numbers will follow the trunk routing if set to TRG 0 in Program 13-05.
For IP AspireNet ensure that the VOIPU ‘trunk’ ports are in their own trunk group in Program 14-05, do not
create a trunk group with a mix of VOIP trunk ports and any other trunk port type. VOIP trunk ports should
not be seized directly via line keys or trunk access (SC 9, 805 or 804).
Related Programs
The following example indicates the setting required to seize the trunk in a networked system (Extension in
System A tries to make an external call using a trunk in System B).
Program System – A System – B
Program 14-06-01 Route 1 Route 1

Trunk Group Routing 101 (Route to System ID 1) 1 (Route to Trunk Group 1)
Program 21-02-01 Extension 301 (which make a call

Trunk Group Routing for via networking)
Extensions 1 (Route 1)
This setting is referenced in
Program 14-06-01
Program 21-16-01 System ID 1

Trunk Group Routing for Route 1
Networking This setting is referenced in
Program 14-06-01
13-26 AspireNet
Issue 5.0
3.10 Conference
ference
The user can create a Conference call to include a user in a networked system.
Operation
To establish a Conference:
Keyset
1. Establish Intercom or trunk call.
2. Press Conf or Conference key (PGM 15-07 or SC 851: 07) or press HOLD and dial 826 .
3. Dial extension you want to add.
OR
Access outside call
OR
Retrieve call from Park orbit.
To get the outside call, you can either press a line key or dial a trunk/trunk group code.
You can optionally go back to step 2 to add more parties to your Conference.
4. When called party answers, press CONF or Conference key twice or HOLD key twice.
If you cannot add additional parties to your Conference, you have exceeded the system’s Conference limit.
5. Repeat steps 2 - 4 to add more parties.
Single Line Set / 2-Button Telephone

1. Establish Intercom or trunk call.
2. Single Line Set
Hookflash and dial 826.
2-Button Telephone
Press HOLD and dial 826
3. Dial extension you want to add.
OR
Access trunk call.
OR
Retrieve call from Park orbit.
4. Single Line Set
Hookflash and repeat step 3 to add more parties.
OR
Hookflash twice to set up the Conference.
2-Button Telephone
Press HOLD and repeat step 3 to add more parties.
OR
Press HOLD twice to set up the Conference.
If you cannot add additional parties to your Conference, you have exceeded the system’s Conference
limit.

Issue 5.0
Related Programs
15-02-24 CONF key operation mode (for system phones)
15-07-01 Programming Function Keys - Conference (code 07)
20-06-01 Calls of Service for Extensions
20-13-08 Class of Service Options (Supplementary Service) - Conference
13-28 AspireNet
Issue 5.0
3.11 Department Calling

epartment Calling
Department Group access is available via Networking. When the extension at System A tries to make a
Department Call to System B, System A should have a numbering plan which defines the Department
Access code at System-B (must be defined as dial type 8, Networking in Program 11-01: System Number-
ing).
The following Department Calling options are supported with the Networking feature.
Program Number Mode Same as Single System
16-01-02 Department Calling Cycle Yes
16-01-03 Department Routing When Busy Yes
16-01-04 Hunting Mode No
16-01-05 All Ring Mode When a call is transferred to a Depart-

ment Group with All Ring, there is a
difference in operation. In a single sys-
tem, an extension within the same sys-
tem can transfer a call to a Department
Group and the call will ring an exten-
sion within the Department Group once
the transferring user hangs up. In a net-
worked system, the transfer will not go
through and the call will recall the
extension performing the transfer.
16-01-06 STG Withdraw Mode No
16-01-07 Call Recall Restriction for STG No
16-01-08 Maximum Queuing Number No
16-01-10 Enhanced Hunting Mode No
Operation
1. When dialing the Department access code for the networked system, the call is dialed in the same way
as normal.
2. A Department Call from the outgoing system will be routed to an available extension in the Department
Group.
Related Programs
11-07-01 Department Group Pilot Numbers
16-01 Department Group Basic Data Setup
16-02-01 Department Group Assignment for Extensions
22-07-01 DIL Assignment
22-11-02 DDI

Issue 5.0
3.12 Department Step Call

Department Step Call
After calling a busy Department Calling Group member in a networked system, an extension user can
have Department Step Calling Quickly call another member in the group.
Operation
To make a Step Call:

You step through Extension Groups set in Program 16-02.
1. Place call to busy Department Group member.
OR
Place call to Department Group pilot number.
2. 851Repeat step 2 to call other Department Group members.
Related Programs
11-12-07 Service Code Setup (for Service Access) - Step Call
11-16-01 Single Digit Service Code Setup - Step Call
20-08-12 Class of Service Options (Outgoing Call Service) - Department Step Calling
13-30 AspireNet
Issue 5.0
3.13 Direct Inward Dialing (DDI)

Direct Inward Dialling (DDI)
An incoming DDI call can be routed to an extension in a networked system.
Operation
For incoming DDI calls, the system refers to Program 22-11: DDI Translation Number Conversion to deter-
mine how to route the call. If the extension number is determined to be in the networked system, the call
will be routed to the appropriate system node.
It is possible to route to a Department Group Pilot number at a remote system, the group can be set to all
ring mode.
It is not possible to route a DDI call to an IRG at the remote system.
The timer value is determined by the system data at the incoming trunk side if the incoming DDI call is
transferred to a ring group due to the no-answer timer expiring.
Related Programs
22-02 Incoming Service Type
22-09 DDI Basic Data Setup
22-10 DDI Conversion Area Setup
22-11 DDI Conversion Table Data Setup
22-12 DDI Transferred Destination Setup

Issue 5.0
3.14 Direct Inward Line (DIL)

Direct Inward Line (DIL)
A Direct Inward Line (DIL) is a trunk that rings an extension, virtual extension or Department Group
directly. Since DILs only ring one extension or group (i.e., the DIL destination), employees always know
which calls are for them. For example, a company operator can have a Direct Inward Line for International
Sales Information. When outside callers dial the DIL’s phone number, the call rings the operator on the
International Sales line key. The DIL does not ring other extensions.
The outside party can call an extension at a networked system, if the DIL trunk is set to route to the other
system.
Operation
1. An outside caller places a call to a DIL trunk.
2. The call will be routed to the networked system if the DIL target is defined as an extension in the net-
worked system in Program 22-07: DIL Assignment.
Related Programs
22-02 Incoming Service Type Setup
22-07 DIL Assignment
22-08 Second IRG Setup for Unanswered DIL/IRG
13-32 AspireNet
Issue 5.0
3.15 Direct Inward System Access (DISA)

Direct Inward System Access (DISA)
Networking allows DISA callers to place a call to an extension in a networked system. Some system fea-
tures can also be accessed from the networked system. The Class of Service is determined by the pass-
word entered by the DISA caller. The password table is referred to by the system on the incoming trunk
side.
The Networking feature allows the following DISA services as allowed in Program 20-14: Class of Service
Options for DISA/E&M.
Program Number Service Name Available
20-14-02 Trunk Route Access Yes
20-14-03 Trunk Group Access No
20-14-04 Common Abbreviated dialing No
20-14-05 Operator access Yes
20-14-06 Internal Paging No
20-14-07 External Paging Yes
20-14-08 Specified trunk access No
20-14-09 Forced trunk disconnect No
20-14-10 Call Forward setting No
20-14-11 Break In No
Operation
To place a DISA call into the system:

1. Dial the telephone number that rings the DISA trunk.
2. Wait for the DISA trunk to automatically answer with a unique dial tone.
3. Dial the 6-digit DISA password (access code).
4. Wait for a second unique dial tone.
5. Dial an extension number.
OR
Dial 9 for Trunk Group Routing or ARS.
OR
Dial Alternate Trunk Route Access Code (if enabled).
OR
Dial 805 + a trunk number (1-400) for an outside call.
OR
Dial 0 for the operator.
OR
Dial 803 + an External Paging Zone number (1-8 or 0 for All Call).
If the received digits are analysed as a networked extension number, the call is routed to the proper net-
work node.

Issue 5.0
Related Programs
20-14 Class of Service Options for DISA/E&M
22-02 Incoming Service Type Setup
25-01 DDI/DISA Line Basic Data Setup
25-02 DDI/DISA VRS Error Message
25-03 DDI/DISA Transfer Ring Group with Incorrect Dialing
25-04 DDI/DISA Transfer Ring Group with No Answer/Busy
25-05 DDI/DISA Error Message Setup
25-06 DDI/DISA One-Digit Code Attendant Setup
25-07 System Timers for DDI/DISA
25-08 System timer for DDI/DISA Service
25-09 Class of Service for DISA User
25-10 Trunk Group Routing for DISA
25-11 Toll Restriction Class for DISA
25-12 Individual Trunk Group Routing for DISA
25-13 System Option DISA Service
3.16 Hold
This feature is available with no changes in programming or operation.
The MOH tone is sourced at the local system where the caller is hearing the hold tone. For example a user
at System A places a call to system B and puts the call on hold, the MOH source at system B will be heard
by the held user.
Whilst the caller is on hold the AspireNet speech path will be reserved, waiting for the call to be taken off
hold.
13-34 AspireNet
Issue 5.0
3.17 Hotline / Direct Station Selection (DSS)

HoldHotline / Direct Station Selection (DSS)
An extension user can have a Hotline key to a networked extension. The Hotline or DSS console keys can
display the status lamp indication of an extension in a networked system.
The key will show the status for idle, busy, DND set and Call Forward Immediate set.
Lamp status may not be updated immediately. Status will be updated in the time interval specified in Pro-
gram 20-01-04.
The status for ACD extensions or Virtual extensions will not be sent via AspireNet . It is still possible to
have a Hotline / DSS key for an ACD or Virtual extension but it will not show any BLF information, the key
can only be used to call the extension.
The basic status for a Hotel extension is sent via AspireNet e.g idle, busy, DND set, Call forward set. The
Hotel room status is not sent e.g. check in/out, room status etc.
Operation
To Set a Function Key to your Hotline partner:

1. Press SPK to go off hook.
2. Dial service code 851.
3. Press Hotline key + 01 +partner’s extension number + HOLD.
To place a call to your Hotline partner:

1. Press Hotline key
You can optionally lift handset after this step for privacy.
To transfer your outside call to your Hotline partner:

1. Press Hotline key.
2. Announce call and press the Transfer button.
OR
Press Transfer to have the call wait at your Hotline partner unannounced.
If unanswered, the call recalls like a regular transferred call.
To answer a call from your Hotline partner:

1. If you hear two beeps, speak toward phone.
OR
2. If your telephone rings, lift handset.
Calling an extension from your 60 button DSS Console:

1. Press DSS Console key.
If the call voice-announces, you can make it ring by dialing 1.
If you don’t have Handsfree, you must lift handset to speak.

Issue 5.0
Related Programs
20-01-04 System Options - Network BLF Indication
20-02-03 System Options for Multi-Line Telephones - BLF Control
20-13-06 Class of Service Options (Supplementary Service) - Automatic Off Hook

Signaling
30-02-01 110 button DSS Console Key Assignment
30-05-01 DSS Console Lamp Table
3.18 Intercom
An extension user can make an intercom call to a networked system if the networked extensions are
defined with the Network Access Code (Program 11-01, Dial Type = 8)
A user can change the signaling type for the intercom call they place to either a voice announce or ringing
call to extension in a networked system.
Operation
To place an Intercom call:

OR
At single line telephone, lift handset.
2. Dial extension number (or 0 for your operator).
Your call may voice-announce or ring the called extension. Dial 1 to change the way your call alerts the
called extension.
If the extension you call is busy or doesn’t answer, you can dial another extension without hanging up.
Related Programs
11-12-06 Service code of Voice/Signal Change
11-16-03 One digit service code of Voice/Signal Change
20-08-01 Class of Service Options (Outgoing Call Service) - Intercom Calls
13-36 AspireNet
Issue 5.0
3.19 Keep Alive Operation

Keep Alive Operation
The Keep Alive operation will check that the distant end is available. A dummy message is sent that the
distant end must respond to, if no response is received the line will be taken out of service.
Keep alive operation is effective on ISDN BRI/PRI and IP AspireNet connections.
Operation
The response to the keep alive message is automatic.
The generation of the keep alive message is set by Program 10-31-01, when the timer is set. If the timer is
set to 0 the keep alive generation is turned off.
The retry count for a keep alive message that is not responded to is also set at the originating system.
The line will be placed back in service when the line is active and a keep alive message is responded to a
keep alive message.
The keep alive operation will only take place if the message is sent and not responded to by the distant
end, if the message is not sent (for example if ISDN layer 1 is not active) then the keep alive operation will
not take place.
When the keep alive operation occurs the link will be taken out of service:
• Any calls that are in progress will be released.
• Park Hold orbits will be released.
• No further Park Hold information will be sent until the link is active.
Related Programs
10-31-01 The interval of the keep alive time. The interval can be set from 0 to 65535
Seconds. When set to 0 the keep alive is disabled.
10-31-02 Retry Count. The system will retry the keep alive message, after the distant
end does not respond after this count the line will be taken out of service.

Issue 5.0
3.20 Last Number Redial

Last Number Redial
Last Number Redial allows an extension user to quickly redial the last number dialed. When used with a
networked system, the system can use the same trunk route on which the call was originally placed, even
if the trunk is a trunk in another system.
Operation
To redial your last call:
1. Without lifting the handset, press Redial key.
The last dialed number is displayed.
2. To redial the last number, press #.
OR
Search for the desired number from the Redial List by pressing the Redial or VOL  or VOL  keys.
3. Lift the handset or press SPK to place the call.
The system automatically selects a trunk from the same group as your original call and dials the last num-
ber dialed.
OR
1. At system phone, press idle line key (optional).
The system automatically selects a trunk from the same group as your original call.
2. Press Redial.
OR
OR
2. Dial 816.
The system automatically selects a trunk from the same group as your original call and dials the last num-
ber dialed.
Related Programs
11-12-12 Service Code Setup (for Service Access) - Last Number Dial
15-02-13 Multi-Line Telephone Basic Data Setup - Redial List Mode
13-38 AspireNet
Issue 5.0
3.21 Message Waiting

Message Waiting
This feature can be used when placing an intercom call to a networked extension and receives either no
answer or hears a busy tone.
With a networked system, when a Message Waiting has been left, there is a slight difference in the telephone’s
display. With a single system, the extension’s name which left the message is displayed. With a networked sys-
tem, the extension number is displayed.
Operation
To leave a Message Waiting:

1. Call busy or unanswered extension.
2. Dial the Message Waiting service code 841
OR
3. ress Message Waiting key (PGM 15-07 or SC 851: 38)
4. Hang up.
With system phones, the MW LED lights.
To answer a Message Waiting:

When you have a message, your MW LED flashes fast for system phones.
1. At a system phone, press SPK key and dial 841.
OR
Press Message Waiting key (PGM 15-07 or SC 851: 38).
OR
At single line telephones, lift the handset and dial 841.
If the called extension doesn’t answer, dial 0 or press your Message Waiting key to automatically leave
them a message.
Normally, your MW LED goes out. If it continues to flash, you have new messages in your “Voice Mail” mailbox
or a new “General Message”. Go to “To check your messages” below.
Related Programs
11-11-09 Service Code Setup (for Setup/Entry Operation) - Answer Message

Waiting
11-11-10 Service Code Setup (for Setup/Entry Operation) - Cancel All Messages Wait-
ing
11-11-11 Service Code Setup (for Setup/Entry Operation) - Cancel Message

Waiting
11-16-07 Single Digit Service Code Setup: Message Waiting
20-13-07 Class of Service Options (Supplementary Service) - Message Waiting

Issue 5.0
3.22 Operator, Centralized

Operator, Centralised
It is possible to have a centralized network operator extension that can be dialed with the operator access
code (0).
Calls to the operator will be queued and answered in order. Up to 32 calls can be queued at the operator
extension. The quantity of network calls that can queue at the operator may be limited by the quantity of
AspireNet channels available.
Operation
The network numbering plan must be set up to route the operator access code (0) to the system that has
the operator extension.
The operator extension must be set in Program 20-17-01.
Centralized Operator
SYSTEM A SYSTEM B
CENTRALIZED
OPERATOR
(EXTN 2200)
• At System-A extension 2200 must be set as the operator extension in Program 20-17-01.
• At system-B dial 0 must be set as networking in Program 11-01 and routed to the node ID of System-A.
• Users at System A and B can access the operator by dialing 0.
13-40 AspireNet
Issue 5.0
3.23 Paging
An extension user can make internal or external pages to a networked system. Paging to a networked sys-
tem can only be activated by dialing a service code and the target network’s system ID.
Operation
To Make an Internal Page
1. Dial 801.
2. Dial # and the system ID.
The system ID must be dialed as 2 digits (ex: #02).
3. Dial the Paging Zone number (00-64).
Dialing 00 calls All Call Internal Paging.
4. Make announcement to the networked system.
5. Press SPK to hang up.
To Make an External Page

1. Dial 803.
Dialing 0 calls All Call External Paging.
To Make a Combined Page

1. Dial 751.
Dialing 0 calls All Call Combined Paging.

Issue 5.0
Related Programs
11-12-19 Service Code Setup (for Service Access) - Internal Group Paging
11-12-20 Service Code Setup (for Service Access) - External Paging
11-12-24 Service Code Setup (for Service Access) - Combined Paging
31-01 System Options for Internal/External Paging
31-02 Internal Paging Group Assignment
31-03 Internal Paging Group Settings
31-04 External Paging Zone Group
31-07 Combined Paging Assignment
13-42 AspireNet
Issue 5.0
3.24 Park
Park
Park places a call in a waiting state (called a Park Orbit) so that an extension user may pick it up. Any
extension user who is in the same Park Group as the extension which placed the call in Park can answer
the call. This includes extension users in a networked system. For example, when an extension user in
Park Group 3 within System A places a call in Park, the extension users in Park Group 3 at any connected
system can retrieve the call by pressing the flashing park key or dialing a service code.
If you do not want the park hold orbits to be available to other users within the AspireNet network, then
place the extension at each site in a different park hold group in Program 24-03.
With a single system, when two users within the same Park Group try to place a call in the same park orbit
at the same time, one user will get the orbit while the other user’s call will either ring back or it will remain
an active call, depending on how the park orbit was accessed. With Networking, if both users try to access
the same orbit, one user will get the orbit, while the other will hear ringback, at which time they can park the
call in a different orbit.
Operation
To Park a call in a system orbit:

You can Park Intercom or trunk calls.
1. Press Park key (PGM 15-07 or SC 852: *04 + orbit).
The Park key LED lights.
If you hear busy tone, the orbit is busy. Try another orbit.
2. Use Paging to announce call.
If not picked up, the call will recall to you.
OR
1. At system phone or 2-Button telephone, press HOLD.
OR
At a SLT single line telephone, hookflash.
2. Dial 861 and the Park orbit (01-64).
If you hear busy tone, the orbit is busy. Try another orbit.
3. Use Paging to announce call.
If not picked up, the call will recall to you.
Note: The parked call recalls after the Park Hold Time (Program 24-01-06). The call rings the extension to
which it recalled for the Hold Recall Callback Time (Program 24-01-02). The call then goes on Hold for the
Park Hold Time - then recalls again for the Hold Recall Callback Time. The call continues to cycle between
Hold and recall until the extension user answers the call or the outside party hangs up.
To pick up a parked call:

1. Lift handset.
2. Press Park key (PGM 15-07 or SC 852: *04 + orbit).
OR

1. Dial 861 and the Park orbit (1-64).

Issue 5.0
Related Programs
11-12-31 Service Code Setup (for Service Access) - Placing a Call in Park
11-12-32 Service Code Setup (for Service Access) - Retrieving Call from Park
20-11-19 Class of Service Options (Hold/Transfer Service) – Normal/Extended Park

If enabled, the recall timer set in Program 24-01-07 is used. If this option is
disabled, the timer in Program 24-01-06 is used.
24-01-06 System Options for Hold - Park Hold Recall Timer – Normal
24-01-07 System Options for Hold - Park Hold Recall Timer – Extended
24-03-01 Extension Park Group assignment
3.25 Ringdown Extension, Internal/External

A networked system can have a phone defined as a Ringdown Extension to dial either an internal or exter-
nal number.
Operation
To place a call if your extension has ringdown programmed:

1. Lift handset.
If you want to place a trunk call, press a line key before lifting the handset.
Depending on the setting of your ringdown timer, you may be able to dial an Intercom call before your ring-
down goes through.
If the destination has Handsfree Answerback enabled, your call will voice announce. If the destination has
Forced Intercom Ringing enabled, your call will ring.
Related Programs
20-08-09 Class of Service Options (Outgoing Call Service) - Hotline/Extension Ring-

down
21-01-09 System Options for Outgoing Calls - Ringdown Extension Timer
21-11-01 Extension Ringdown (Hotline) Assignments
13-44 AspireNet
Issue 5.0
3.26 Selectable Display Messaging

Selectable Display Messaging
An extension user can select a preprogrammed Selectable Display Message for their extension. This mes-
sage will be displayed on an incoming intercom caller’s LCD when they call the extension in a networked
system.
Operation
To select a message:
1. Press SPK key + dial 836.
OR
Press SPK key + press Text Message key (PGM 15-07 or SC 851: 18 + Message Number).
2. Enter Message number (01-20) if needed.
Use VOL  or VOL  to scroll through the messages.
3. (Optional for messages 1-8 and 10)
Dial the digits you want to append to the message.
You can append messages 1-8 and 10 with digits (e.g., the time when you will be back). You enter the time in 24-
hour format, but it displays in 12-hour format.
To cancel a message:
1. Press SPK key + dial 0.
OR
Press SPK key + press Text Message key (PGM 15-07 or SC 851: 18)
Related Programs
20-13-19 Class of Service Options (Supplementary Service) - Selectable Display Mes-

saging
20-16 Selectable Display Messages

Issue 5.0
3.27 Toll Restriction

Toll Restriction
Toll Restriction limits the numbers an extension user may dial. When accessing a trunk at a remote system
the Toll Restriction Class of Service is defined by the calling extension’s system, but the Toll Restriction
tables will be used from the system which has the outgoing trunk. The Toll restriction class number is sent
to the remote system, the remote system will use the class number to define the Toll Restriction tables to
use.
Since the restriction table is used for the system which has the outgoing trunk, the definition of the Class of
Service may be different, unless all Toll Restriction Classes of Service and Toll Restriction Tables are
defined the same between systems.
Operation
Example:
The extension user in System A, which has a Toll Restriction Class 2, dials an outside party after seizing a
trunk from a networked system (System B). The received digits are compared to the Class 2 Restriction
Table in System B. The call is then either allowed or rejected based on this table.
Related Programs
21-04 Toll Restriction Class Assignment for Extensions
21-05 Toll Restriction Class Setup
21-06 Toll Restriction Table Setup
13-46 AspireNet
Issue 5.0
3.28 Transfer
Transfer
The following types of Transfer are available with Networking:

• Screened Transfer
• Unscreened Transfer
• Transfer to busy extension
Operation
Transferring Trunk Calls

To Transfer a trunk call to a co-worker’s extension:
1. At system phone, press HOLD.
OR
At a single line telephone, hookflash.
You hear Transfer dial tone.
2. Dial co-worker’s extension number.
If the extension is busy or doesn’t answer, you can dial another extension number or press the line key to return
to the call.
SLT users can retrieve the call by pressing hookflash. If a call has been transferred and the SLT user has hung
up the handset, the call be can retrieved by dialing 715 and the extension number to which it had been trans-
ferred.
3. Announce call and hang up.
If you don’t have Automatic On Hook Transfer, you must press Conf or your Transfer Programmable Function
Key to Transfer the call.
If your co-worker doesn’t want the call, press the flashing line key to return to the call.
ferred.
If you don’t want to screen the call, hang up without making an announcement.
Transferring to a busy extension

Will require setting in Program number 24-01-01, must be enabled at the originating system and the target
system.
1. Dial the co-worker’s extension number.
Busy tone is heard.
2. Press the Transfer button.
The call will wait for the busy co-worker to become free and then will ring.

Issue 5.0
Transferring Intercom Calls

To Transfer your Intercom call:
1. At system phone, press HOLD.
OR
At single line telephone, hookflash.
2. Dial extension to receive your call.
If the extension is busy, doesn’t answer or does not want the call, you can dial another extension number. In
addition, you may be able to transfer the call to the busy extension.
ferred.
3. Announce your call and press Transfer (Program 15-07 or SC 851: 06) or hang up.
With Automatic On Hook Transfer
When you hang up, the call is automatically transferred.
Without Automatic On Hook Transfer
You must press your Transfer Programmable Function Key to Transfer the call.
To Transfer the call unscreened, press your Transfer Programmable Function Key and hang up without making
an announcement.
Related Programs
20-11-06 Class of Service Options (Hold/Transfer Service) - Unscreened Transfer
20-11-07 Class of Service Options (Hold/Transfer Service) - Transfer Without Holding
20-11-08 Class of Service Options (Hold/Transfer Service) - Transfer Display
20-11-18 Class of Service Options (Hold/Transfer Service) - No Recall
24-02-01 System Options for Transfer - Busy Transfer
13-48 AspireNet
Issue 5.0
3.29 Voice Mail, Centralized

Voice Mail, Centralised
Networking will support the use of a single voice mail for the entire network. A user may call into the voice
mail from anywhere in the network and perform most functions as if the voice mail were located on their
premises.
With a networked system, when voice mail is busy, there is a slight difference in the telephone’s display.
With a single system, the extension calling a busy voice mail will see WAITING VOICE MAIL on their dis-
play. With a networked system, the extension will display CALLING XXX (XXX = extension number).
Operation
Pilot Call
When the extension in System B dials the centralized voice mail access number (Program 45-01-07), then
the voice mail in System A is accessed as the centralized voice mail.
Service Codes
When the voice mail service code (Program 11-12-51, SC 717) is dialed, the system calls either the voice
mail at the same site as the user, or if the centralized voice mail access number is defined in Program 45-
01-07, then the centralized voice mail is called.
One-Digit Service Code

If an extension user hears either a busy signal or a ring back tone when placing an intercom call and dials
the one-digit service code (or SC717) for voice mail, the call is connected to the voice mail at the same
site, or if the centralized voice mail access number is defined in Program 45-01-07, then the centralized
voice mail is called.
Voice Mail Message Key

If an extension user presses their Voice Mail Message key (Program 15-07 or SC 851: 77), the voice mail
at the same site as the user is called, or if the centralized voice mail access number is defined in Program
45-01-07, then the centralized voice mail is called.
Conversation Recording
If an extension user presses their Conversation Recording key (Program 15-07 or SC 851: 78), the voice
mail at the same site as the user is called, or if the centralized voice mail access number is defined in Pro-
gram 45-01-07, then the centralized voice mail is called and the conversation is recorded to that voice mail.
Automatic Attendant
If an extension user presses their Automatic Attendant Message key (Program 15-07 or SC 851: 79), the
voice mail at the same site as the user is called, or if the centralized voice mail access number is defined in
Program 45-01-07, then the centralized voice mail is called.
Incoming Calls - Normal Trunks

If centralized voice mail is set in Program 22-05:103 as the destination of an incoming call, this call is auto-
matically transferred to the centralized voice mail.
If 102 (in-skin/external voice mail) is selected for the ring group instead of 103, the voice mail within the
extension’s own system will be called.
No Answer at Incoming Ring

If centralized voice mail is set in Program 22-08:103 as the destination, normal incoming calls or DIL calls
which receive no answer are transferred to the centralized voice mail
Transferred Destination for Each DDI Translation Table

If centralized voice mail is set in Program 22-11-05: and 22-11-06:103 for the transferred destination for
each DDI translation table, then the call will be transferred to the system which has the centralized voice
mail.

Issue 5.0
DDI, DISA: Mis-Dial Calls

If the centralized voice mail is set in Program 25-03:103 as the transferred destination for DDI/DISA mis-
dial calls, then the call will be transferred to the centralized voice mail.
DDI, DISA: No Answer and Busy Calls

If the centralized voice mail is set in Program 25-04:103 as the transferred destination for DDI/DISA no
answer and mis-dial calls, then the call will be transferred to the centralized voice mail.
System Configuration Examples

Only Local Voice Mail
SYSTEM A SYSTEM B
LOCAL VOICE
System A System B
16-02-01 VM ports = Group 64 16-02-01 None

45-01-01 64 45-01-01 None
11-07-01 Group 64 = 600 11-07-01 None
45-01-07 None 45-01-07 None
11-01-01 Dial 6x = Type 2 (Inter- 11-01-01 Dial 6x = Type 8 (Net-
com) working)
• The inbound and outbound calls in System-A can access the local voice mail (600), but the inbound
and outbound calls in System-B can not reach the local voice mail (600). Access from System-B to the
voice mail is available only when a called telephone (at system-A) has Call Forward set to the local voice
mail (600).
• Users at System-B can not use any voice mail services (SC 717, voice mail function key 77, conversa-
tion record key 78 or auto attendant key 79).
• The voice mail system should not have any mail boxes set for extension numbers at System-B.
13-50 AspireNet
Issue 5.0
Only Local Voice Maill

SYSTEM A SYSTEM B
CENTRALIZED VOICE
MAIL (600)
System A System B
16-02-01 VM ports = Group 64 16-02-01 None
45-01-01 64 45-01-01 None
11-07-01 Group 64 = None 11-07-01 None
45-01-07 600 45-01-07 600
45-01-08 64 45-01-08 None
11-01-01 Dial 6x = Type 2 (Inter- 11-01-01 Dial 6x = Type 8 (Net-

com) working)
• The inbound and outbound calls in Systems A and B can access the centralized voice mail (600).
• Program 11-07-01 must be set to none (for the VM group) in System-A otherwise the operation of cen-
tralized voice mail will be disabled.
• Trunk calls must be routed to IRG 103 (Centralized Voice Mail) in System A or B. A trunk routed to IRG
102 (Local Voice Mail) will fail.
• The DDI voice mail tag (@xxx) can be used in Program 22-11-02 at System A and B.
• The voice mail function key (SC 851 +77), conversation record key (SC 851+78) and auto attendant
key (SC 851+79) can be used at System A and B.

Issue 5.0
Local Voice Mail at each SV9100 System
SYSTEM A SYSTEM B
LOCAL
VOICE MAIL (600) LOCAL
System A System B
16-02-01 VM ports = Group 64 16-02-01 VM ports = Group 64

45-01-01 64 45-01-01 64
11-07-01 Group 64 = 600 11-07-01 Group 64 = 500
45-01-07 None 45-01-07 None
11-01-01 Dial 5x = Type 8 (Net- 11-01-01 Dial 5x = Type 2 (Inter-
working) com)
Dial 6x = Type 2 (Inter- Dial 6x = Type 8 (Net-
com) working)
• The inbound and outbound calls in System-A can access the local voice mail (600),
• The inbound and outbound calls in System-B can access the local voice mail (500).
• Access to the voice mail of the remote system is available only when a called telephone has Call For-
ward set to their local voice mail.
• Conversation record key (SC 851+78) and auto attendant key (SC851+79) can be used.
• The voice mail function key (SC 851+77) can be used for access to the mail box and message waiting.
• Users at System A or B can not use the voice mail function key (SC 851+77) or SC 717 to access to
called party’s mail box when placing calls to a remote extension. The voice mail function key should only
be used when calling a local extension.
• The voice mail systems should not have any mail boxes set for extension numbers at the remote sys-
tem.
13-52 AspireNet
Issue 5.0
Local and Centralized Voice Mail
SYSTEM A SYSTEM B
CENTRALISED
VOICE MAIL (600) LOCAL
This configuration is not supported, the local voice mail should be removed and only the centralized voice
mail used.
Related Programs
11-01 System numbering plan
11-07 ICM Pilot Calling Number
11-12-51 Service Code (Call Own Mailbox)
16-02 ICM Group
22-05 Destination for normal incoming ringing
22-08 Destination for no answering
22-11-05 Transferred destination-1 for each DDI translation table
22-11-06 Transferred destination-2 for each DDI translation table
22-12 Transferred destination for each DDI translation table area
25-03 Destination for DDI/DISA: Mis-Dial
25-04 Destination for DDI/DISA: No Answer/Busy Status
45-01 Voice Mail Integration Options

Issue 5.0
SECTION 4 PROGRAMMING
4.1 Before Reading This Section

This section provides you with detailed information about the system programs. By changing a
program, you change the way the feature associated with that program works. In this section,
you find out about each program, the features that the program affects and how to enter the pro-
gram data into system memory.
Do not start customizing your system without first reading Section 1, Setting Up the
Networking Feature.
CAUTION
k
When you want to customize a feature, find it in Section 1 and learn about it. (If you have trouble
finding the feature, try cross-referencing it in the Index at the back of this book.) Section 1 will tell
you what programs you have to change to get the operation you want. Then, look the program up
in this section if you have any questions about how to enter the data.
4.2 How to Use This Section
This section lists each program in numerical order. For example, Program 10-01 is at the
beginning of the section and Program 92-01 is at the end. The information on each program is
subdivided into the following headings:
Description describes what the program options control. The Default Settings for each program
are also included. When you first install the system, it uses the Default Setting for all programs.
Along with the Description are the Conditions which describe any limits or special
considerations that may apply to the program.
The reverse type (white on black) just beneath the Description heading is the program access
level. You can only use the program if your access level meets or exceeds the level the program
requires. Refer to 4.3 How to Enter Programming Mode on page 13-55 for a list of the system
access levels and passwords.
Feature Cross Reference provides you with a table of all the features affected by the program.
You will want to keep the referenced features in mind when you change a program. Customizing
a feature may have an effect on another feature that you did not intend.
Telephone Programming Instructions shows how to enter the program data into system
memory. For example:
2. 15-07-01
15-07-01 TEL
KY01 = *01

tells you to enter the programming mode, dial 150701 from the telephone dial pad. After you do,
you will see the message “15-07-01 TEL” on the first line of the telephone display. This indicates
the program number (15-07), item number (01), and that the options are being set for the exten-
13-54 AspireNet
Issue 5.0
sion . The second row of the display “KY01 = *01” indicates that Key 01 is being programmed
with the entry of *01. The third row allows you to move the cursor to the left or right, depending on
which arrow is pressed. To learn how to enter the programming mode, refer to 4.3) How To Enter
Programming Mode below.
4.3 How to Enter Programming Mode

 In a newly installed system, use extension 200 (port 1).
3. Press Speaker.
4. # *#*.
Password

Refer to the following table for the default system passwords. To change the passwords, use 90-
02 : Programming Password Setup.
User
Name
------ ------ 1 (MF) Manufacturer (MF):

All programs

All programs in this section not listed below for SA and
SB
0000 ADMIN1 3 (SA) System Administrator – Level 1 (SA):
10-01, 10-02, 10-12, 10-13, 10-14, 10-15, 10-16, 10-17,
10-18, 10-22, 12-02, 12-03, 12-04, 15-01, 15-07, 15-09,
15-10, 15-11, 20-16, 21-07, 21-14, 22-04, 22-11, 25-08,
30-03, 32-02, 40-02, 41-02, 41-03, 41-04, 41-05, 41-06,
41-07, 41-08, 41-09, 41-10, 41-11, 41-12, 41-13, 41-14,
41-15, 41-16, 41-17, 41-18, 90-03, 90-04, 90-06, 90-07,
90-18, 90-19
9999 ADMIN2 4 (SB) System Administrator – Level 2 (SB):
13-04, 13-05, 13-06

Issue 5.0
4.4 How to Exit Programming Mode
When you are done programming, you must be out of a program option to exit (pressing the
Answer key will exit the program option).
1. Press Answer key to exit the program options, if needed.
Program Mode
2. Press Speaker. If changes were to the system programming, "Saving System Data" is
displayed.
3. The display shows "Complete Data Save" when completed and exits the telephone to an idle
mode.
 To save a customer’s database, a blank USB Drive is required. Insert the USB Drive into the GCD-CP10 and,
using Program 90-03, save the software to the USB Drive. (Program 90-04 is used to reload the customer data if
necessary.) Note that a USB Drive can only hold one customer database. Each database to be saved requires a
separate drive.
4.5 Using Keys to Move Around in the Programs
Once you enter the programming mode, use the keys in the following chart to enter data, edit
data and move around in the menus.
0~9 and * Enter data into a program.

Transfer Complete the programming step you just made (e.g., pressing Enter
on a PC keyboard). When a program entry displays, press Transfer
to bypass the entry without changing it.
Recall Delete the entry to the left (e.g., pressing Backspace on a PC
keyboard).
For example, if programming item 5 in 15-03, pressing Answer allows
you to enter a new option in program 15-03. Pressing Answer again
allows you to select a new program in the 15-XX series. Pressing
Answer a third time allows you to enter a new program beginning with 1.
Pressing Answer one last time brings you to the beginning program
display, allowing you to enter any program number.
13-56 AspireNet
Issue 5.0
MIC Switch between the different input data fields by pressing MIC. The
cursor moves up to the top row of the display. Pressing MIC again
moves the cursor back to the middle row.
LINE KEYS Use pre-programmed settings to help with the program entry. These
settings vary between programs from LINE 1 = 0 (off) and LINE 2 = 1
(on) to preset values for timers where LINE 1 = 5, LINE 2 = 10, LINE 3 =
15, etc.
settings.)
LINE KEY 1 Program a pause into a Speed Dialing bin.
LINE KEY 2 Program a recall/flash into a Speed Dialing bin.
LINE KEY 3 Program an @ into a Speed Dialing bin.
VOL  Scroll backward through a list of entry numbers (e.g., from extension
VOL  Scroll forward through a list of entry numbers (e.g., from extension
4.6 Programming Names and Text Messages
Several programs (e.g., Program 20-16 : Selectable Display Messages) require you to enter text.
Use the following chart when entering and editing text. When using the keypad digits, press the
key once for the first character, twice for the second character, etc. For example, to enter a C,
press the key 2 three times. Press the key six times to display the lower case letter. The name
can be up to 12 digits long.
Use this keypad digit . .

When you want to. . .
.
1 Enter characters:
1 @ [ ¥ ] ^ _ ` { | } Æ ¨ Á À Â Ã Ç É Ê ì ó

Issue 5.0
Use this keypad digit . .

When you want to. . .
.

0 Enter characters:
0 ! “ # $ % & ’ ( ) ô Õ ú ä ö ü 
* Enter characters:
* + , - . / : ; < = > ? π Σ  Ω ¢ £
# # = Accepts an entry (only required if two letters on the same
key are needed – ex: TOM). Pressing # again = Space. (In
system programming mode, use the right arrow Softkey
instead to accept and/or add a space.)
Recall Delete the entry to the left.
Hold Clear all the entries from the point of the flashing cursor and to
the right.
4.7 Using Softkeys For Programming
Each UNIVERGE SV9100 display telephone provides interactive Softkeys for intuitive feature
access. The options for these keys will automatically change depending on where you are in the
system programming. Simply press the Softkey located below the option you wish and the
display will change accordingly.
_
Program Mode

Pressing the VOLUME  or VOLUME  will scroll between the menus.
_
Program Mode
CCIS Hard Mtnance

13-58 AspireNet
Issue 5.0
4.8 What the Softkey Display Prompts Mean
When using a display telephone in programming mode, various Softkey options are displayed.
These keys will allow you to easily select, scan, or move through the programs.
Table 13-4 Softkey Display Prompts
If you press this

The system will. . .
Softkey . . .

You can press VOLUME  or VOLUME  to scroll forward or

Issue 5.0

10-03 : ETU Setup
Level:
IN
Description
Use Program 10-03 : ETU Setup to setup and confirm the Basic Configuration data for each blade.
When changing a defined terminal type, first set the type to 0 and then plug the new device in to have
the system automatically define it or you may have to reset the blade.
 The items highlighted in gray are read only and cannot be changed.
Input Data
For CNF PKG Setup

Item
No.
For DLCA PKG Setup

Item
No.
01 Terminal Type (B1) 0 =Not set 0

1 =Multiline Terminal
2 =SLT Adapter
3 =Bluetooth Cordless Handset
6 =PGDAD (Paging)
7 =PGDAD (Tone Ringer)
8 =PGDAD (Door Box)
9 =PGDAD (ACI)
10 =DSS Console
11 =--- Not Used ---
13-60 AspireNet
Issue 5.0
Item
No.
02 Logical Port Number (B1) 0 = Not set 0

1 = Multiline Terminal (1~960)
2 = SLT Adapter (1~960)
3 = Bluetooth Cordless Handset (BCH) (1~960)
6 = PGDAD (Paging) (1~8)
7 = PGDAD (for Tone Ringer) (1~8)
8 = PGDAD (for Door Box) (1~8)
9 = PGDAD (for ACI) (1~96)
10 = DSS (1~32)
11 = --- Not Used ---
03 --- Not Used ---
04 --- Not Used ---
05 --- Not Used ---
B-Channel 2
Item
No.
06 Terminal Type (B2) 0 =Not set 0

6 =PGDAD (Paging)
7 =PGDAD (Tone Ringer)
8 =PGDAD (Door Box)
9 =PGDAD (ACI)
12 =APR (B2 Mode)
07 Logical Port Number (B2) 0 =Not set 0

6 =PGDAD (Ext. Speaker)
7 =PGDAD (Paging/Tone Ringer) = (1~8)
8 =PGDAD (for Door Box) = (1~8)
9 =PGDAD (ACI) = (1~96)
12 =APR (for B2 mode) (193~512)
08 Multiline Telephone Type 0 =DT3** 0

1 =DT4**
09 Side Option Information 0 =No option 0

1 =8LK Unit
2 =16LK Unit
3 =24ADM

Issue 5.0
10 Bottom Option Information 0 =No option 0

(Only applies to DTL–style
telephones) 1 =APR
2 =ADA
3 =BHA
4 =No Used
5 =BCA
11 Handset Option 0 =No option 0

Information
1 =PSA/PSD
2 =Bluetooth Cordless Handset
For LCA PKG Setup

Item
No.
03 Transmit Gain Level (S-Level) 1~57 (-15.5 +12.5dB) 32 (0dB)
04 Receive Gain Level (R-Level) 1~57 (-15.5 +12.5dB) 32 (0dB)
For COTA/C Unit Setup

Item
No.
13-62 AspireNet
Issue 5.0
For ODTB PKG Setup

Item
No.
02 2/4 Wire 0 = 2 Wire 1

1 = 4 Wire
03 E&M Line Control Method 0 = TYPE I 1

1 = TYPE V
For DIOP PKG Setup

Item
No.
01 LD/OPX Specification 0 =LD Trunk 0

1 =OPX
02 Logical Port Number 0 =1~400 (LD Trunk) 0

1 =1~960 (OPX)

Issue 5.0
For BRIA PKG Setup

Item
No
01 ISDN Line Mode 0 =No setting 1

1 =T-Point
2 =S-Point
3 =NW Mode (Leased Line)
4 =NW Mode (Interconnected Line)
5 =NW Mode (Interconnected Line,
Fixed Layer1=NT)
6 =S-Point (Leased Line)
02 Logical Port Number [0:No setting] = 0 0

 The starting port number of a BRI [1:T-Point] = 1-400
line is displayed. Two logic ports are [2:S-Point] = 1-960
automatically assigned to a BRI line. [3:NW Mode (Leased Line)] = 1-256
[4:NW Mode (Interconnected Line)]
= 1-256
[5:NW Mode (Interconnected Line,
Fixed Layer1=NT)] = 1-256
[6:S-Point(Leased Line)] = 1-960
03 Connection Type 0 =Point-to-Multipoint 0

1 =Point-to-Point

 Each timer value of Layer 3 is set up
for every type using Program 81-06
(T-Bus).
05 CLIP Information Announcement 0 =Disable 1

Based on this setting, the system 1 =Enable
06 Connection Bus Mode 0 =Extended passive bus 0

1 =Short passive bus
07 S-point DDI digits 0-4 0
08 Dial Sending Mode 0 =Enblock Sending 1

ISDN Protocol definition 1 =Overlap Sending
09 Dial Information Element 0 =Keypad Facility 0

ISDN Protocol definition 1 =Called Party Number
[Only when Dialing Sending Mode
(10-03-08) is set for 1 (Overlap Sending]
13-64 AspireNet
Issue 5.0
Item
No
10 Master/Slave System 0 =Slave System 0

If set to 0, system is synchronized to 1 =Master System
network clock. If set to 1, system is not
synchronized to the network clock.
11 Networking System No. 0 - 50 0
14 Service Protocol for S-point 0 =Keypad facility 0

1 =Specified Protocol for Aspire
system
15 Call Busy Mode for S-point 0 =Alerting 0

1 =Disconnect
17 ISDN Line Ringback Tone 0 =Disable 0

If Telco does not provide ringback tone, 1 =Enable
SV9100 can if set to 1:Enable.
18 Type of Number 0 =Unknown 0

ISDN Protocol definition 1 =International number
2 =National number
3 =Network specific number
4 =Subscriber number
5 =Abbreviated number
19 Numbering Plan Identification 0 =Unknown 0

ISDN Protocol definition 1 =ISDN numbering plan
2 =Data numbering plan
3 =Telex numbering plan
4 =National standard numbering
plan
5 =Private numbering plan
22 QSIG Operation Mode 0 =Disable 0

1 =Enable
23 Straight/Cross Wiring 0 =Auto 0

1 =Manual (Cross)
2 =Manual (Straight)
24 Power feeding for S-point 0 =Disable 0

1 =Enable

Issue 5.0
For PRTA PKG Setup

Item
No.
01 ISDN Line Mode 0 =No setting 1

1 =T-Point
2 =S-Point
3 =NW Mode (Leased Line)
4 =NW Mode (Interconnected Line)
5 =NW Mode (Interconnected Line,
Fixed Layer1=NT)
6 =S-Point (Leased Line)
02 Logical Port Number 1 =for T-Bus 1~400 1

 The start port number of a PRI line is
displayed.
03 CRC Multi-frame(CRC4) 0 =off 1

(Only for E1[30B+D] Mode) 1 =on

 Each timer value of Layer 3 is set up
for each type in
Program 81-06 (T-Bus)
05 CLIP Information 0 =Disable 1

Based on this setting, the system 1 =Enable
06 Length of Cable 0 =Level 1 2

1 =Level 2
2 =Level 3
3 =Level 4
4 =Level 5
07 S-point DDI digits 0-4 0
08 Dial Sending Mode 0 =Enbloc Sending 0

ISDN Protocol definition 1 =Overlap Sending
09 Dial Information Element 0 =Keypad Facility 0

ISDN Protocol definition 1 =Called Party Number
Only when Dial Sending Mode
(10-03-08) is set for 1 (Overlap
Sending).
10 Master/Slave System (Network Mode 0 =Slave System 0

only)
1 =Master System
13-66 AspireNet
Issue 5.0
Item
No.
11 Networking System Number 0 - 50 0

(Network Mode only)
12 --- Not Used ---
13 Loss-Of-Signal Detection Limit 0 =Level 0 (lowest sensitivity) 2

If the transmit/receive voltage is less 1 =Level 1
than the setting in 10-03-13, the system 2 =Level 2
considers this as Loss-Of-Signal and 3 =Level 3
the PRTA does not come up. Note that 4 =Level 4
there are different values based on the
5 =Level 5
setting in 10-03-12 for the PRI.
6 =Level 6
7 =Level 7 (highest sensitivity)
14 Service Protocol for S-point 0 =Keypad facility 0

1 =Specified Protocol for Aspire
system
15 Call Busy Mode for S-point 0 =Alerting 0

1 =Disconnect
16 Two B-Channel Transfer for PRI 0 =off 0

Service
1 =on
17 ISDN Ringback Tone 0 =Disable 0

If Telco does not provide ringback tone, 1 =Enable
SV9100 can if 10-03-17 is set to
1:Enable.
18 Type of Number 0 =Unknown 0

ISDN Protocol definition. 1 =International number
Select the number type for the ISDN 2 =National number
circuit. 3 =Network Specific number
4 =Subscriber number
5 =Abbreviated number
19 Numbering Plan Identification 0 =Unknown 0

ISDN Protocol definition. 1 =ISDN numbering plan
Select the Numbering Plan used for the 2 =Data numbering plan
ISDN circuit. 3 =Telex numbering plan
4 =National standard numbering
plan
5 =Private numbering plan
20 Network Exchange Selection 0 =Standard (same as NI-2) 0

Select the ISDN protocol for the ISDN 1 =reserved
circuit. 2 =reserved
3 =DMS (A211)
4 =5ESS
5 =DMS (A233)
6 =4ESS
7 =NI-2

Issue 5.0
Item
No.
21 Number of Ports 0 =Auto 0

1 =4 Ports
2 =8 Ports
3 =12 Ports
4 =16 Ports
5 =20 Ports
6 =24 Ports
7 =28 Ports
22 QSIG Operation Mode 0 =Disable 0

1 =Enable

1 = Manual (Cross)
For DTI (T1) PKG Setup

Item
No.

line.
02 T1 Signal Format Selection 0 =D4 (12 Multi Frame) 0

1 =ESF (24 Multi Frame)
03 Zero Code Suppression 0 =B8ZS 0

1 =AMI/ZCS
04 Line Length Selection 0 =0 feet ~ 133 feet 0

1 =133 feet ~ 266 feet
2 =266 feet ~ 399 feet
3 =399 feet ~ 533 feet
4 =533 feet ~ 655 feet
05 T1 Clock Source 0 =Internal 1

1 =External

1 =4 Ports
2 =8 Ports
3 =12 Ports
4 =16 Ports
5 =20 Ports
13-68 AspireNet
Issue 5.0

1 = Manual (Cross)
For IPLE PKG Setup
Item
Item Input Data
No.
01 VoIP Type IPLE
02 Number of Channel 256
03 Number of Voice Channels 256
For VM00 PKG Setup

Item
No.
For CCTA PKG Setup

Item
No.

line.
02 T1 Signal Format Selection 0 =D4 (12 Multi Frame) 0

1 =ESF (24 Multi Frame)
03 Zero Code Suppression 0 =B8ZS 0

1 =AMI/ZCS
04 Line Length Selection 0 =0 feet ~ 133 feet 0

1 =133 feet ~ 266 feet
2 =266 feet ~ 399 feet
3 =399 feet ~ 533 feet
4 =533 feet ~ 655 feet
05 T1 Clock Source 0 =Internal 1

1 =External

Issue 5.0

1 =4 Ports
2 =8 Ports
3 =12 Ports
4 =16 Ports
5 =20 Ports

1 = Manual (Cross)
Conditions
When changing a defined terminal type, first set the type to 0 and then plug the new device in to
have the system automatically define it, or redefine the type manually.
 The system must have a blade installed to view/change the options for that type of blade.

 Universal Slots
13-70 AspireNet
Issue 5.0

Level:
SA
Description
Use Program 10-12 : GCD-CP10 Network Setup to setup the IP Address, Subnet-Mask, and Default
Gateway addresses.
If any IP Address or NIC setting is changed, the system must be reset for the changes
to take affect.
CAUTION
Input Data
Item
No.

128.0.0.1 ~ 191.254.255.254
192.0.0.1 ~ 223.255.255.254

all Host Addresses
254.0.0.0 255.0.0.0 255.128.0.0
are 0.
255.192.0.0 255.224.0.0 255.240.0.0
255.248.0.0 255.252.0.0 255.254.0.0
is:
255.255.0.0 255.255.128.0 255.255.192.0
0,
255.255.224.0 255.255.240.0 255.255.248.0
127,
255.255.252.0 255.255.254.0 255.255.255.0
255.255.255.128 255.255.255.192 255.255.255.224 128.0,
255.255.255.240 255.255.255.248 255.255.255.252 191.255,
255.255.255.254 255.255.255.255 192.0.0,
223.255.255
Mask is invalid.

Gateway 128.0.0.1 ~ 191.254.255.254
192.0.0.1 ~ 223.255.255.254

Issue 5.0
Item
No.
04 Time Zone 0~24 (0 = -12 Hours and 24 = +12 Hours) 12 Determine the offset
from Greenwich Mean
Time (GMT) time.
Then enter its
example, Eastern
Time (US and
Canada) has a GMT
offset of -5. The
program data would
then be 7 (0= -12, 1=
-11, 2= -10, 3= -9, 4= -
8, 5= -7, 6= -6, 7= -5,
……24= +12)

Interface
1 = 100Mbps, Full Duplex (GCD-CP10)
06 Network 0 = No (Disable) 0 If using an external

Address NAPT Router or not.
Port 1 = Yes (Enable)
Translation(
NAPT)
Router
Setup
07 NAPT 0.0.0.0 ~ 126.255.255.254 0.0.0.0 Sets the IP address

Router IP on the WAN side of
Address 128.0.0.1 ~ 191.255.255.254
router.
192.0.0.1 ~ 223.255.255.254
(Default
Gateway
[WAN])

Redirect redirect message, this
1= (Disable)
determines if the IP
Routing Table
or not.
09 IP Address 0.0.0.0 ~ 126.255.255.254 172.16.0.10

128.0.0.1 ~ 191.255.255.254
192.0.0.1 ~ 223.255.255.254
13-72 AspireNet
Issue 5.0
Item
No.
10 Subnet 128.0.0.0 192.0.0.0 224.0.0.0 255.255.0.0

Mask
240.0.0.0 248.0.0.0 252.0.0.0
254.0.0.0 255.0.0.0 255.128.0.0
255.192.0.0 255.224.0.0 255.240.0.0
255.248.0.0 255.252.0.0 255.254.0.0
255.255.0.0 255.255.128.0 255.255.192.0
255.255.224.0 255.255.240.0 255.255.248.0
255.255.252.0 255.255.254.0 255.255.255.0
255.255.255.128 255.255.255.192 255.255.255.224
255.255.255.240 255.255.255.248 255.255.255.252
255.255.255.254 255.255.255.255
11 NIC Setup 0 = Auto Detect 0 Set for GPZ-IPLE.

5 = 1 Gbps, Full Duplex
13 DNS 0.0.0.0
Primary 0.0.0.0 ~ 126.255.255.254
Address Set for adding a
128.0.0.1 ~ 191.255.255.254
14 DNS function for DNS.
192.0.0.1 ~ 223.255.255.254
Secondary
Address
15 DNS Port 0~65535 53
17 IPL NIC Port 0 = MDI 0

Setting
1 = MDI-X
Conditions
The system must be reset for these changes to take affect.


Issue 5.0

10-19 : VoIP DSP Resource Selection
Level:
SA
Description
Use Program 10-19 : VoIP DSP Resource Selection to define the criteria for each DSP resource on
the VoIP blade.
Input Data
Slot Number 1
Input Data
DSP Resource Number 01~256
Input Data
Item
No.
01 VoIP DSP Resource Selection 0 = Common use for both IP Resource 1~256 = 0
extensions and trunks
1 = IP Extension
2 = SIP Trunk
3 = Networking(CCIS)
4 = Use for NetLink
5 = Blocked
Paging
8 = Unicast Paging
Conditions
None

None
13-74 AspireNet
Issue 5.0

10-20 : LAN Setup for External Equipment
Level:
IN
Description
Use Program 10-20 : LAN Setup for External Equipment to define the TCP port/address/etc. for
communicating to external equipment.
Input Data
Type of External Equipment 1 = CTI Server
2 = ACD MIS
3 = Not Used
4 = Networking System
5 = SMDR Output
6 = DIM Output
7 = Reserved
8 = Reserved
9 = 1st Party CTI
10 = ACD Agent Control
11 = O&M Server
12 = Traffic Report Output
13 = Room Data Output for Hotel Service
14 = IP-DECT Directory Access
15 = Presence
Item
No.
01 TCP Port 0~65535 External Device 1 (CTI Server) = 0
External Device 2 (ACD MIS) = 0
External Device 4 (Networking System)
= 30000
External Device 5 (SMDR Output) = 0
External Device 6 (DIM Output) = 0
External Device 11 (O&M Server) = 8010
External Device 12 (Traffic Report Output) = 0
External Device 13 (Room Data Output for
Hotel Service) = 0
03 Keep Alive Time 1~255 (sec) 30

Issue 5.0
Conditions
None

None
13-76 AspireNet
Issue 5.0

10-27 : H.323 System Interconnection with Application Setup
Level:
SA
Description
Use Program 10-27 : H.323 Interconnection with Application Setup to set the IP address of the
networked IP systems.
Input Data
System ID 01-50
Input Data
Item
Item Input Data Default Default
No.
01 IP Address 1.0.0.1_126.255.255.254 0.0.0.0 11-01-01

System ID is related with the 128.1.0.1 _191.254.255.254 10-12-01
System ID in the Numbering Plan 192.0.1.1 _223.255.254.254
(Program 11-01-03). When the
digits are analyzed and the
system ID is determined from the
SV9100 data set in the
Numbering Plan, the networking
call will be sent to the IP Address
set in this program.
The IP Address should be the IP
Address of the peer CPU
(Program 10-12-01)
02 Call Procedure Port 1-65535 1730 84-02-33

The Port Number should be set
with the same value as the H.225
setup port in Program 84-02-33.
Conditions
None

None

Issue 5.0

10-31 : Networking Keep Alive Setup
Level:
IN
Description
Use Program 10-31 : Networking Keep Alive Setup to set the interval and retry count of the AspireNet
networking keep alive message. The keep alive is used for ISDN and IP networking.
The keep alive message is automatically responded to by the destination SV9100, if the response is not
received the retry count will start. If a response is not received within the number of retries the
networking link will be taken out of service. When the link is taken out of service:
 Any calls that are in progress will be released.
 Park Hold orbits will be released.
 No further Park Hold information will be sent until the link is active.
The link will automatically become active when the next keep alive response is received.
Input Data
Item
No.
01 Keep Alive Interval 0-65535 0

This program is used to set the interval of
Keep Alive. The SV9100 does not send
Keep alive when this item is set to "0".
If this entry is greater than "0", networked
PRI spans which are using Kentrox DSUs
will not re-sync when removed from
service then returned to service.
02 Keep Alive Retry Timer 1-255 5

Set how many times the SV9100 resends
Keep Alive.
13-78 AspireNet
Issue 5.0
Conditions
The keep alive message must be sent and a response not received for the retry count, for the link to be taken
out of service and the calls in progress and Park Hold orbits to be released.
For example: If an ISDN Net Link connection is disconnected at Layer 1 then the keep alive message can not
be sent, therefore the keep alive operation will not occur.

 Networking - AspireNet

Issue 5.0

10-32 : Networking Maximum PRI Channel Setup
Level:
IN
Description
Use Program 10-32 : PRI Networking Maximum PRI Channel Setup to assign the number of B-
channels to be used for each ISDN blade. This allows for fractional PRIs when used with multiple site
networking.
If this program is limited to less than "30" on one side of the network, then it also limits both inbound and
outbound network calls. For example, when you select 10 channels then only channels 1 to 10 will be
available. If a call is attempted on channels 11 to 30 the caller will receive busy tone. This also applies on
the other side of the network as well.
The setting is for each slot within the SV9100; ensure that you select the correct slot before making any
changes.
This program will not affect a PRI card set as Trunk or Station mode.
Input Data
Slot Number 1-24
Item
No.
01 Maximum Channels 1 - 30 30
Set the maximum number of channels
which can be used with PRI NetLink.
Conditions
None

13-80 AspireNet
Issue 5.0
Program 11 : System Numbering

11-01 : System Numbering
Level:
IN
Description
Use Program 11-01 : System Numbering to set the system numbering plan. The numbering plan assigns
the first and second digits dialed and affects the digits an extension user must dial to access other
extensions and features, such as service codes and trunk codes. If the default numbering plan does not
meet the site requirements, use this program to tailor the system numbering to the site.
Improperly programming this option can adversely affect system operation. Make sure
you thoroughly understand the default numbering plan before proceeding. If you
CAUTION
k must change the standard numbering, use the chart for Table 13-5 System Numbering
Default Settings on page 13-83 to keep careful and accurate records of your changes.
Before changing your numbering plan, use PC Pro to make a backup copy of your
system data.
Changing the numbering plan consists of three steps:
Step 1: Enter the digit(s) you want to change
You can make either single or two digit entries. In the Dialed Number column in the Table 1-4 System
Numbering Default Settings on page 1-91 table, the nX rows (e.g., 1X) are for single digit codes. The
remaining rows (e.g., 11, 12, etc.) are for two digit codes.
 Entering a single digit affects all the Dialed Number entries beginning with that digit. For example,
entering 6 affects all number plan entries beginning with 6. The entries you make in step 2 and step 3
below affect the entire range of numbers beginning with 6. (For example, if you enter 3 in step 2 the
entries affected are 600~699. If you enter 4 in step 2 below, the entries affected are 6000~6999.)

Issue 5.0
 Entering two digits lets you define codes based on the first two digits a user dials. For example, entering
60 allows you to define the function of all codes beginning with 60. In the default program, only  and #
use 2-digit codes. All the other codes are single digit. If you enter a two digit code between 0 and 9, be
sure to make separate entries for all the other two digit codes within the range as well. This is because
in the default program all the two digit codes between 0 and 9 are undefined.
 Defining codes based on more than 2 digits require a secondary program (PRG 11-20) to define the codes.
Step 2: Specify the length of the code you want to change
After you specify a single or two digit code, you must tell the system how many digits comprise the code.
This is the Number of Digits Required column in the Table 1-4 System Numbering Default Settings on
page 1-91 table.
Step 3: Assign a function to the code selected
After entering a code and specifying its length, you must assign its function. This is the Dial Type column
in the Table 1-4 System Numbering Default Settings on page 1-91 table. The choices are:
Dial
Types
0 --- Not Used ---
1 Service Code 11-10 : Service Code Setup (for System Administrator)

11-11 : Service Code Setup (for Setup/Entry Operation)
11-12 : Service Code Setup (for Service Access)
11-13 : Service Code Setup (for ACD)
11-14 : Service Code Setup (for Hotel)
11-15 : Service Code Setup, Administrative (for Special
Access)
11-16 : Single Digit Service Code Setup
2 Extension Number 11-02 : Extension Numbering
11-04 : Virtual Extension Numbering
11-06 : ACI Extension Numbering
11-07 : Department Group Pilot Numbers
11-08 : ACI Group Pilot Number
11-17 : ACD Group Pilot Number
3 Trunk Access Code 11-09-01 : Trunk Access Code
4 Special Trunk Access 11-09-02 : Trunk Access Code
5 Operator Access 20-17 : Operator Extension
6 ARS/F-Route Access 44-xx
8 Networking System
Access
13-82 AspireNet
Issue 5.0
Dial
Types
9 Dial Extension Analyze 11-20 : Dial Extension Analyze Table
 Changing the Dial Type for a range of codes can have a dramatic affect on how your system operates.
Assume, for example, the site is a hotel that has room numbers from 100-399. To make extension
numbers correspond to room numbers, you should use Program 11-02 to reassign extension numbers on
each floor from 100 to 399. (Other applications might also require you to change entries in Program
11-10 ~ 11-16.)
Default
See the following tables for default settings.
Table 13-5 System Numbering Default Settings
Dial Types: 1=Service Code, 2=Extension Number, 3=Trunk Access, 4=Special Trunk Access,
5=Operator Access, 6=Flexible Routing, 8 = Networking 9 = Dial Extension Analyze, 0=None
Number of Digits Network System

Dial Type
Required ID [if type 8] 0~50
Dialed
Default New Default New
1X 3 2
11 0 0
12 0 0
13 0 0
14 0 0
15 0 0
16 0 0
17 0 0
18 0 0
19 0 0
10 0 0
1 0 0
1# 0 0
2X 3 2
21 0 0
22 0 0
23 0 0
24 0 0
25 0 0
26 0 0

Issue 5.0
Table 13-5 System Numbering Default Settings (Continued)

Dial Type
Dialed
27 0 0
28 0 0
29 0 0
20 0 0
2 0 0
2# 0 0
3X 3 2
31 0 0
32 0 0
33 0 0
34 0 0
35 0 0
36 0 0
37 0 0
38 0 0
39 0 0
30 0 0
3 0 0
3# 0 0
4X 3 2
41 0 0
42 0 0
43 0 0
44 0 0
45 0 0
46 0 0
47 0 0
48 0 0
49 0 0
13-84 AspireNet
Issue 5.0

Dial Type
Dialed
40 0 0
4 0 0
4# 0 0
5X 4 2
51 0 0
52 0 0
53 0 0
54 0 0
55 0 0
56 0 0
57 0 0
58 0 0
59 0 0
50 0 0
5 0 0
5# 0 0
6X 3 2
61 0 0
62 0 0
63 0 0
64 0 0
65 0 0
66 0 0
67 0 0
68 0 0
69 0 0
60 0 0
6 0 0
6# 0 0

Issue 5.0

Dial Type
Dialed
7X 3 1
71 0 0
72 0 0
73 0 0
74 0 0
75 0 0
76 0 0
77 0 0
78 0 0
79 0 0
70 0 0
7 0 0
7# 0 0
8X 3 1
81 0 0
82 0 0
83 0 0
84 0 0
85 0 0
86 0 0
87 0 0
88 0 0
89 0 0
80 0 0
8 0 0
8# 0 0
9X 1 3
91 0 0
92 0 0
13-86 AspireNet
Issue 5.0

Dial Type
Dialed
93 0 0
94 0 0
95 0 0
96 0 0
97 0 0
98 0 0
99 0 0
90 0 0
9 0 0
9# 0 0
0X 1 5
01 0 0
02 0 0
03 0 0
04 0 0
05 0 0
06 0 0
07 0 0
08 0 0
09 0 0
00 0 0
0 0 0
0# 0 0
X 4 1
1 0 0
2 0 0
3 0 0
4 0 0
5 0 0

Issue 5.0

Dial Type
Dialed
6 0 0
7 0 0
8 0 0
9 0 0
0 0 0
 0 0
# 0 0
#X 4 1
#1 0 0
#2 0 0
#3 0 0
#4 0 0
#5 0 0
#6 0 0
#7 0 0
#8 0 0
#9 0 0
#0 0 0
# 0 0
## 0 0
Conditions
None

 Flexible System Numbering
13-88 AspireNet
Issue 5.0

11-02 : Extension Numbering
Level:
IN
Description
Use Program 11-02 : Extension Numbering to set the extension number. The extension number can have
up to eight digits. The first/second digit(s) of the number should be assigned in Program 11-01 or
Program 11-20. This allows an employee to move to a new location (port) and retain the same extension
number.
Input Data
Extension Port Number 001 ~ 960
Item Extension
Description
No. Number
01 Dial (Up to 8 digits) Set up extension numbers for multiline telephones, single line
telephones (including SLT Adapter, APR), and IP telephones.
Extension number assignments cannot be duplicated in Programs
11-02, 11-06, 11-07, 11-08, and 11-17.
Default
Extension Port Extension
Number Number
1 200
2 201
3 202
~
300 499
301 5000
~
960 5659
Conditions
None

Issue 5.0
 Department Calling
 Flexible System Numbering
 Intercom
13-90 AspireNet
Issue 5.0

11-07 : Department Group Pilot Numbers
Level:
IN
Description
Use Program 11-07 : Department Group Pilot Numbers to assign pilot numbers to each Department Group
set up in Program 16-02. The pilot number is the number users dial for Department Calling and
Department Step Calling. The pilot number can have up to eight digits. The first and second digits of the
number should be assigned in Program 11-01 or Program 11-20 as type 2.
Input Data
Department (Extension) Group Number 01~64
Extension
Item
Group Pilot Description Related Program
No. Number
01 Dial Use this program to assign  16-01 : Department (Extension)

(Up to 8 digits) department group pilot Group Basic Data Setup
numbers.  16-02 : Department Group
The number set up by Program Assignment for Extensions
11-02 (Extension Numbering)  16-03 : Secondary Department
cannot be used. Group
The extension number cannot
be duplicated in Programs
11-02, 11-04, 11-06, 11-08, and
11-17.
Default
No Setting
Conditions
None

 Department Step Calling

Issue 5.0

11-10 : Service Code Setup (for System Administrator)
Level:
IN
Description
Use Program 11-10 : Service Code Setup (for System Administrator) to customize the Service Codes for
the System Administrator. You can customize additional Service Codes in Programs 11-11~11-16. The
following chart shows:
 The number of each code (01~42).
 The function of the Service Code.
 The type of telephones that can use the Service Code.
 The default entry. For example, dialing (item 26) allows users to force a trunk line to disconnect.
Input Data
Item Related
Item Terminals Default
No. Program
01 Night Mode Switching MLT, SLT 818 12-xx

20-07-01
02 Change of music on hold tone MLT 881
03 Setting the System Time MLT 828
04 Storing Common Speed Dialing MLT 853

Numbers
05 Storing Group Speed Dialing MLT 854

Numbers
06 Setting the Automatic Transfer for MLT 833 24-04-01

Each Trunk Line
07 Canceling the Automatic Transfer MLT 834 24-04-01

for Each Trunk Line
08 Setting the Destination for MLT 835 24-04-01

Automatic Trunk Transfer
09 Charging Cost Display by the MLT 771

Supervisor
10 --- Not Used ---
11 Entry Credit for Toll Restriction MLT 774
12 Night Mode Switching for Other MLT 718 12-xx

Group 20-07-01
13-92 AspireNet
Issue 5.0
Item Related
No. Program
13 --- Not Used ---
14 --- Not Used ---
15 --- Not Used ---
16 Leaving Message Waiting MLT 726 11-11-09

(Requires CPU to be licensed for
Hotel/Motel)
17 Dial Block by Supervisor MLT 701 90-19
18 Off-Premise Call Forward by Door MLT 822 13-05

Box
19 --- Not Used ---
20 VRS - Record/Erase Message MLT, SLT 716 20-07-13
21 VRS - General Message Playback MLT, SLT 711 20-07-14
22 VRS - Record or Erase General MLT, SLT 712 20-07-15

Message
23 SMDR - Extension Accumulated MLT 721 20-07-18

Printout Code
24 SMDR - Group Accumulated MLT 722 20-07-19

Printout Code
25 Account Code Accumulated MLT 723 20-07-20

Printout Code
26 Forced Trunk Disconnect MLT, SLT 724 20-07-11
27 Trunk Port Disable for Outgoing MLT, SLT 745 20-07-12

Calls
28 --- Not Used ---
29 --- Not Used ---
30 Register DECTPP MLT Not Set
31 Delete DECTPP MLT Not Set
32 Set Private Call Refuse MLT, SLT Not Set
33 Entry Caller ID Refuse MLT Not Set
34 Set Caller ID Refuse MLT, SLT Not Set
35 Dial-In Mode Switching MLT, SLT Not Set
36 Change the Guidance Message MLT, SLT Not Set

Number on Voice Mail Auto
Attendant
37 --- Not Used ---
38 --- Not Used ---

Issue 5.0
Item Related
No. Program
39 --- Not Used ---
40 --- Not Used ---
41 Date Setting MLT Not Set 20-07-30
42 Maintenance Service MLT Not Set
43 --- Not Used ---
44 --- Not Used ---
45 --- Not Used ---
46 Watch Message Setting MLT, SLT 714
47 Warning Message Setting MLT 719
48 Auto Dial Setting for Sensor MLT 748
49 Auto Dial Setting for Remote Watch MLT 749
51 Power Saving for Power Save MLT, SLT 746

Group
 MLT = Multiline Terminal

 SLT = Single Line Telephone
 Conditions
None

 Refer to Input Data chart on the previous pages.
13-94 AspireNet
Issue 5.0

11-11 : Service Code Setup (for Setup/Entry Operation)
Level:
IN
Description
Use Program 11-11 : Service Code Setup (for Setup/Entry Operation) to customize the Service
Codes which are used for registration and setup. You can customize additional Service Codes in
Programs 11-10, and 11-12 ~ 11-16.
The following chart shows:

 What type of telephones can use the Service Code.
 The default entry. For example, dialing 725 (Item 18 ) allows users to turn on or turn off Background
Music .
Input Data
Item Terminal Related

Item Default
No. s Program
01 Call Forward – All MLT, SLT 848
02 Call Forward – Busy MLT, SLT 843
03 Call Forward – No Answer MLT, SLT 845
04 Call Forward – Busy/No Answer MLT, SLT 844
05 Call Forward – Both Ring MLT, SLT 842
06 --- Not Used ---
07 Call Forwarding – Follow-Me MLT, SLT 846
08 Do Not Disturb MLT, SLT 847
09 Answer Message Waiting MLT, SLT 841 11-10-16
10 Cancel All Messages Waiting MLT, SLT 873
11 Cancel Message Waiting MLT, SLT 871
12 Alarm Clock MLT, SLT 827 20-01-06
13 Display Language Selection for Multiline MLT 778 15-02

Terminal

Issue 5.0

Item Default
No. s Program
14 Text Message Setting MLT 836
15 Enable Handsfree Incoming Intercom Calls MLT 821 20-09-05

20-02-12
16 Force Ringing of Incoming Intercom Calls MLT 823 20-09-05

20-02-12
17 Programmable Function Key Programming MLT 851 15-07

(2-Digit Service Codes) 11-11-38
18 BGM On/Off MLT 825
19 Key Touch Tone On/Off MLT 824
20 Change Incoming CO and ICM Ring Tones MLT 820 15-02
21 Check Incoming Ring Tones MLT 811
22 Extension Name Programming MLT 800 15-01
23 Second Call for DID/DISA/DIL MLT 779
24 Change Station Class of Service MLT 777 20-13-28

Allow an extension user to change the COS of
another extension. Must be allowed in Program
20-13-28.
25 Automatic Transfer Setup for Each Extension MLT, SLT 702 20-11-17
Group 24-05
26 Automatic Transfer Cancellation for Each MLT, SLT 703

Extension Group
27 Destination of Automatic Transfer Each MLT 704 20-11-17

Extension Group 24-05
28 Delayed Transfer for Every Extension Group MLT, SLT 705 20-11-17
24-05
24-02-08
29 Delayed Transfer Cancellation for Each MLT, SLT 706 20-11-17

Extension Group
30 DND Setup for Each Extension Group MLT, SLT 707
31 DND Cancellation for Each Extension Group MLT, SLT 708
32 --- Not Used ---
33 Dial Block MLT, SLT 700
34 Temporary Toll Restriction Override MLT, SLT 875 21-07
35 Pilot Group Withdrawing MLT, SLT 750
36 Toll Restriction Override MLT, SLT 763 21-14
37 Ring Volume Set MLT 829
13-96 AspireNet
Issue 5.0

Item Default
No. s Program
38 Programmable Function Key Programming MLT 852 15-07

(3-Digit Service Codes) 11-11-17
39 Station Speed Dial Number Entry MLT, SLT 855
40 --- Not Used ---
41 Tandem Ringing MLT, SLT 744 15-07

30-03
42 --- Not Used ---
43 Headset Mode Switching MLT, SLT 788
44 Auto Attendant MLT, SLT 790
45 Set/Cancel Call Forward All (Split) MLT, SLT 782
46 Set/Cancel Call Forward Busy (Split) MLT, SLT 783
47 Set/Cancel Call Forward No Answer (Split) MLT, SLT 784
48 Set/Cancel Call Forward Busy No Answer (Split) MLT, SLT 785
49 Set/Cancel Call Forward Both Ring (Split) MLT, SLT 786
50 Set Message Waiting Indication SLT No 15-03-03

Setting 45-01-01
51 Cancel Message Waiting Indication SLT No 15-03-03

Setting 45-01-01
52 Set/Cancel Call Forward All Destination (No MLT, SLT 791

Split)
53 Set/Cancel Call Forward Busy Destination (No MLT, SLT 792

Split)
54 Set/Cancel Call Forward No Answer Destination MLT, SLT 793

(No Split)
55 Call Forward Busy No Answer Destination (No MLT, SLT 794

Split)
56 Telephone Book Lock Service MLT No

Setting
57 Set Do Not Call Table MLT, SLT No

Setting
58 Call Forward with Personal Greeting MLT, SLT 795
59 Call Forward to Attendant except Busy MLT, SLT 796 15-01-08
60 Call Forward to Attendant/No Answer MLT, SLT 797 15-01-09
62 Headset Ring Volume Adjustment MLT 874 11-11-37

15-02-12
15-02-41
15-02-42

Issue 5.0

Item Default
No. s Program
63 Double Height Character Indication MLT No 15-02-45

Setting
64 Reverse Display Indication MLT No 15-02-44

Setting
65 Headset Mode Switching MLT No

Setting
68 IntraMail Language Selection for own Extension MLT, SLT No 47-02-16

Setting
69 IntraMail Language Selection for Specific MLT, SLT No 20-13-53

Extension Setting 47-02-16

 Conditions
None

 Refer to the Input Data chart above.
13-98 AspireNet
Issue 5.0

11-12 : Service Code Setup (for Service Access)
Level:
IN
Description
Use Program 11-12 : Service Code Setup (for Service Access) to customize the Service Codes
which are used for service access. You can customize additional Service Codes in Programs 11-10, 11-
11, and 11-13 through 11-16.

 The type of telephones that can use the Service Code.
 The default entry. For example, dialing 870 (Item 05) cancels a previously set Camp-On.
 Programs that may be affected with the changing the code.
Input Data
Item Related
No. Program
01 Bypass Call MLT, SLT 807 11-16-09

Activate Call Forwarding/Do Not Disturb Override.
This code is available only if you disable the voice
mail Single Digit dialing code in Program 11-16-09.
02 Conference MLT, SLT 826
03 Override (Off-Hook Signaling) MLT, SLT 809
04 Set Camp-On MLT, SLT 850
05 Cancel Camp-On MLT, SLT 870
06 Switching of Voice Call and Signal Call MLT, SLT 812
07 Step Call MLT, SLT 808
08 Barge-In MLT, SLT 810
09 Change to STG (Department Group) All Ring MLT, SLT 780 16-02
10 Station Speed Dialing MLT, SLT 813
11 Group Speed Dialing MLT, SLT 814

Issue 5.0
Item Related
No. Program
12 Last Number Dial MLT, SLT 816
13 Saved Number Dial MLT, SLT 815
14 Trunk Group Access MLT, SLT 804
15 Specified Trunk Access MLT, SLT 805
16 Trunk Access Via Networking MLT, SLT 866
17 Clear Last Number Dialing Data MLT, SLT 876
18 Clear Saved Number Dialing Data MLT, SLT 885
19 Internal Group Paging MLT, SLT 801 31-01-01
20 External Paging MLT, SLT 803
21 Meet-Me Answer to Specified Internal Paging MLT, SLT 864 31-02-01

Group
22 Meet-Me Answer to External Paging MLT, SLT 865
23 Meet-Me Answer in Same Paging Group MLT, SLT 863 31-02-01
24 Combined Paging MLT, SLT 751 31-02-01

31-07
25 Direct Call Pickup - Own Group MLT, SLT 856
26 Call Pickup for Specified Group MLT, SLT 868 23-02
27 Call Pickup MLT, SLT 867 23-02
28 Call Pickup for Another Group MLT, SLT 869 23-02
29 Direct Extension Call Pickup MLT, SLT 715
30 Specified Trunk Answer MLT, SLT 772
31 Park Hold MLT, SLT 831 24-03
32 Answer for Park Hold MLT, SLT 861 24-03
33 Group Hold MLT, SLT 832
34 Answer for Group Hold MLT, SLT 862
35 Station Park Hold MLT, SLT 773
36 Door Box Access MLT, SLT 802
37 Common Canceling Service Code MLT, SLT 720
38 General Purpose Indication MLT 883 15-07-56

15-07-57
39 --- Not Used ---
13-100 AspireNet
Issue 5.0
Item Related
No. Program
40 Station Speed Dialing MLT, SLT 761
41 Voice Over MLT, SLT 890 11-16-08
42 Flash on Trunk lines SLT 806
43 Answer No-Ring Line (Universal Answer) MLT, SLT 872 14-05

14-06
44 Callback Test for SLT SLT 899
45 Enabled On Hook When Holding (SLT) SLT 849 15-03-07
46 Answer On Hook When Holding (SLT) SLT 859 15-03-08
47 Call Waiting Answer/Split Answer SLT 894 11-12-03

Splitting (switching) between calls
48 Account Code SLT 891
49 --- Not Used ---
50 General Purpose Relay MLT, SLT 880
51 VM Access (InMail and VMS) MLT, SLT 717
52 Live Monitoring (InMail) MLT 725
53 Live Recording at SLT MLT, SLT 754
54 VRS Routing for ANI/DNIS MLT, SLT 882

Use when setting up ANI/DNIS Routing to the VRS
Automated Attendant. Using the Transfer feature,
this also allows a call to be transferred to the VRS.
55 --- Not Used ---
56 E911 Alarm Shut Off MLT No 21-01-13

Enter the Service Code that an extension user can
Setting 21-01-14
dial to shut off the E911 Alarm Ring.
57 Tandem Trunking MLT, SLT 753
58 Transfer Into Conference MLT, SLT No 20-13-10

Assign the Service Code a user dials to Transfer a
Setting 20-13-15
20-13-16
call to a Conference call.
59 Trunk Drop Operation for SLT SLT No

Setting
60 --- Not Used ---
61 --- Not Used ---
62 Security Sensor Rest MLT, SLT 837
63 Watch Mode Start MLT, SLT 817

Issue 5.0
Item Related
No. Program
64 Security Sensor Mode Start MLT, SLT 819

 Conditions
None

Refer to the Input Data chart on the previous pages.
13-102 AspireNet
Issue 5.0

11-16 : Single Digit Service Code Setup
Level:
IN
Description
Use Program 11-16 : Single Digit Service Code Setup to customize the one-digit Service Codes used
when a busy or ring back signal is heard. You can customize additional Service Codes in Programs 11-
10 ~ 11-15.

 The default entry. For example, dialing 1 (code 03) when calling an extension switches the call from
either a voice or signal call (depending on how it is currently defined).
 Programs that may be affected by changing these codes.
Input Data
Item Related
Item Default
No. Program
01 Step Call No Setting 11-12-07
02 Barge-In No Setting 11-12-08
03 Switching of Voice/Signal Call 1 11-12-06
04 Intercom Off-Hook Signaling No Setting 11-12-03
05 Camp-On No Setting 11-12-04
06 DND/Call Forward Override Bypass No Setting 11-12-01
07 Message Waiting No Setting 11-12-09
08 Voice Over No Setting 11-12-41
09 Access to Voice Mail No Setting 11-12-51
10 (Department) STG All Ring Mode No Setting 11-12-09

16-01-05
11 Station Park Hold No Setting 11-12-35

Issue 5.0
Conditions
None

Refer to the Input Data chart on previous pages.
13-104 AspireNet
Issue 5.0
Program 14 : Trunk, Basic Setup

14-01 : Basic Trunk Data Setup
Level:
IN
Description
Use Program 14-01 : Basic Trunk Data Setup to set the basic options for each trunk port. Refer to the
chart below for a description of each option, its range and default setting.
Input Data
Trunk Port Number 001~400
Item Related
No. Program
01 Trunk Name Up to 12 Characters Line 001

Set the names for
Line 002
Line 003
trunks. The trunk :
name displays on a Line 400
multiline terminal for
incoming and
outgoing calls.
02 Transmit Level 1~63 32 (0dB)

Use this option to (-15.5dB~
select the CODEC +15.5dB in 0.5dB
gain for the trunk. The intervals)
option sets the gain
(signal amplification)
for the trunk you are
programming.
03 Receive Level 1~63 32 (0dB)

Use this option to (-15.5dB ~ +15.5dB
select the CODEC in 0.5dB intervals)
gain for the trunk. The
option sets the gain
(signal amplification)
for the trunk you are
programming.

Issue 5.0
Item Related
No. Program
04 Transmit Gain Level 1~63 32 (0dB)

for Conference and
Transfer Calls
(-15.5dB ~ +15.5dB
in 0.5dB intervals)
Use this option to
select the CODEC
gain type used by the
trunk when it is part of
an Unsupervised
Conference.
05 Receive Gain Level 1~63 16(-8dB)

for Conference and
Transfer Calls
(-15.5dB ~ +15.5dB
in 0.5dB intervals)
Use this option to
select the CODEC
gain type used by the
trunk when it is part of
an Unsupervised
Conference.
06 SMDR Printout 0 =No Print Out 0 35-01

Use this option to 1 =Prints Out 35-02
have the system
include/exclude the
trunk you are
programming from the
SMDR printout. Refer
to Program 35-01 and
35-02 for SMDR
printout options.
07 Outgoing Calls 0 =Deny (No) 1

Use this option to 1 =Allow (Yes)
allow/prevent
outgoing calls on the
trunk you are
programming.
08 Toll Restriction 0 =Restriction 1 21-04

Use this option to Disabled (No) 21-05
enable/disabled Toll 1 =Restriction 21-06
Restriction for the Enabled (Yes)
trunk. If enabled, the
trunk follows Toll
Restriction
programming
(example: Programs
21-05, 21-06). If
disabled, the trunk is
a toll free line.
09 Private Line 0 =Disable Private 0

Line (Normal)
1 =Enable Private
Line (Private
Line)
13-106 AspireNet
Issue 5.0
Item Related
No. Program
10 DTMF Tones for 0 =Disable (No) 1

Outgoing Calls
1 =Enable (Yes)
Use this option to
enable (1) or disable
(0) DTMF tones for
outgoing trunk calls.
11 Account Code 0 =Disable (No) 1

Required
1 =Enable (Yes)
12 --- Not Used ---
13 Trunk-to-Trunk 0 =Disable (No) 0

Transfer 1 =Enable (Yes)
Use this option to
enable (1) or disable
(0) loop supervision
for the trunk. This
option is required for
Call Forwarding
Off-Premise and
Tandem Trunking
only.
14 Long Conversation 0 =Disable (No) 0 20-21-03

Cutoff
1 =Enable (Yes) 20-21-04
Use this option to
enable or disable the
Long Conversation
Cutoff feature for
each trunk.
15 Long Conversation 0 =Disable (No) 0 20-21-01

Alarm Before Cutoff
1 =Enable (Yes) 20-21-02
Use this option to
Long Conversation
Alarm for each trunk.
16 Forced Release of 0 =Disable (No) 0 24-01-01

Held Call
1 =Enable (Yes) 24-01-05
Use this option to
enable/disable forced
release for calls on
Hold. If enabled, the
system disconnects a
call if it is on Hold
longer than a
programmed interval
(Program 24-01-05).
If disabled, forced
disconnection does
not occur. Program
24-01-01 also affects
this option.

Issue 5.0
Item Related
No. Program
17 Trunk to Trunk 0 =Disable (No) 0

Warning Tone for
Long Conversation
1 =Enable (Yes)
Alarm
Use this option to
Warning Tone for
Long Conversation
feature for DISA
callers.
18 Warning Beep Tone 0 =Disable (No) 0

Signaling 1 =Enable (Yes)
19 Privacy Mode 0 =Disable (No) 0

Toggle Option 1 =Enable (Yes)
Use this option to
enable or disable a
trunk ability to be
switched from private
to non-private mode
by pressing the line
key or Privacy
Release function key.
20 Block Outgoing 0 =Disable (No) 0 14-01-21

Caller ID
1 =Enable (Yes)
Allow (1) or prevent
(0) the system from
automatically blocking
outgoing Caller ID
information when a
user places a call. If
allowed (i.e. block,
enabled), the system
automatically inserts
the Caller ID block
code (defined in
14-01-21) before the
user dialed digits.
21 Caller ID Block Code Dial (up to eight No Setting 14-01-20

Enter the code, up to digits)
8 digits, that should
be used as the Caller
ID Block Code. This
code is automatically
inserted before dialed
digits if Program
14-01-20 is set to 1.
13-108 AspireNet
Issue 5.0
Item Related
No. Program
22 Caller ID to Voice 0 =Disable (No) 0

Mail
1 =Enable (Yes)
Enable or disable the
system ability to send
the Caller ID digits
(Remote Log-On
Protocol) to voice
mail.
23 Least Cost Routing 0 =LCR Off 0

1 =LCR On
2 =LCR On (Cost
Center Code
only)
24 Trunk-to-Trunk 0 =Disable (No) 0

Outgoing Caller ID
through Mode 1 =Enable (Yes)
Enable (1) or Disable

(0) the ability to send
the original Caller ID
through when the call
is Forward Off-
Premise.
25 Continued/ 0 =Disable (No) 0 20-28-01

Discontinued
Trunk-to-Trunk
1 =Enable (Yes) 20-28-02
Conversation 20-28-03
Enable (1) or Disable 24-02-07
(0) the ability to dial a 24-02-10
service code code to
continue or 25-07-07
disconnect the 25-07-08
Trunk-to-Trunk
conversation after the
alert tone is heard.
26 Automatic 0 = Normal Transfer 0 24-02-11

Trunk-to-Trunk (Normal)
Transfer Mode 24-02-12
1 = Step Transfer
(Step)
27 Caller ID Refuse 0 =Disable (No) 0

Setup
1 =Enable (Yes)
28 Effectivity of 0 = No Effect (No) 1 15-12

“Conversation
Recording
1 = Available (Yes)
Destination for
Extension”
30 Flexible Ringing by 0 =Disable (No) 1 13-04

Caller ID
1 =Enable (Yes)
32 Anti-trombone 0 = No Effect (No) 0

Function
1 = Available (Yes)

Issue 5.0
Item Related
No. Program
33 APSU Trunk Receive 1~57 32 (0dB)

Gain
(-15.5dB ~ +12.5dB
Additional PAD when in 0.5dB
a trunk call connects intervals)
to APSU Voice Mail.
35 DT800/DT700 Large 2 = Red 2

LED Illumination
Setup
3 = Green
4 = Blue
Sets LED color for
5 = Yellow
incoming trunk call. In
DT800/DT700 local 6 = Purple
terminal setting menu, 7 = Light Blue
illumination setting 8 = White
must be ‘Automatic’, 9 = Rotation
otherwise the terminal
will ignore PRG 14-
01-35, PRG 15-05-37
and PRG 15-23
settings.
36 Calling Party Name 0 = Disable 0

Indication (ISDN 1 = Enable
Trunk)
Shows sending caller
name on outgoing
ISDN calls.
13-110 AspireNet
Issue 5.0
Item Related
No. Program
38 Outgoing CLI 0 = Contract Number 0 21-13-01

Selection 21-18-01
1 = Extension
Select CLI (Calling Number 21-19-01
Party Number) 2 = Extended Table 21-25-01
sending way to trunk.
3 = PRG 21-13
When set to 0,
extension CLI number 4 = PRG 21-18
set in PRG21-13-01, 5 = PRG 21-19
PRG21-18-01, or 6 = No digits
PRG21-19-01is sent
according to seized
trunk type (ISDN/
H.323/SIP)
automatically.
When set to 1, calling
extension number is
sent as CLI.
When set to 2,
extension table
number set in
PRG21-25-01 is sent
as CLI.
When set to 3, 4, or 5,
extension CLI number
set in PRG21-13-01,
PRG21-18-01, or
PRG21-19-01is sent
to seized trunk
regardless of trunk
type.
39 CLI Composition 0 = Prefer Extension 0

If select default value 1 = Combine Trunk +
0:”prefer extension”, Extension
the extension's CLI is
sent out, if it is not
empty. If it is empty,
the trunk's CLI is sent
instead.
If select value
1:”combine trunk +
extension”, the trunk's
CLI is stored in the
sending buffer,
padded with the
extension's CLI.
40 ISDN Queue 0 = Send CONNECT 0 22-14

Announcement
Connect Mode 1 = Send 22-15
PROGRESS 41-11
#8 41-19

Issue 5.0
Default
Trunk Port
Name
Number
1 Line 001
2 Line 002
: :
400 Line 400
13-112 AspireNet
Issue 5.0
Conditions
None

Refer to features in the Input Data table.

Issue 5.0
Program 14 : Trunk, Basic Setup

14-06 : Trunk Group Routing
Level:
IN
Description
Use Program 14-06 : Trunk Group Routing to set up an outbound routing table for the trunk groups
you assigned in Program 14-05. When users dial 9, the system routes their calls in the order (priority)
specified. For example, if a user dials 9 and all calls in the first group are busy, the system may route the
call to another group. Trunk Access Map programming (Programs 14-07) may limit this option. The
system contains 100 routing tables for trunk access. Each table has four priority orders for trunk access.
There are 100 available Trunk Group Numbers.
Example for setting:
With less than four trunk groups,

Route Number 1 : Order 1 – Trunk Group 1
: Order 2 – Trunk Group 2
For the above setting, if all the lines in trunk group 1 are busy, the system searches
for an idle line in trunk group 2.
With more than four trunk groups,

: Order 4 – 1002 (Jump To Route Number 2)
For the above setting, if all the lines in the trunk groups 1, 2 and 3 are busy, the
system searches for an idle line in trunk groups 4 and 5.
13-114 AspireNet
Issue 5.0
Input Data
Route Table Number 001~100
Priority
Item Related
Order Input Data
No. Program
Number
01 1~4 0 = Not set 14-01-07

001-100 = Trunk group No. 14-05
101-150 = 100+Networking System No. 15-01-02
1001-1100 = 1000+Route Table No. 21-02
Default
- Route 1, Order Number 1 = 1 (Trunk Group 1).
- Order Numbers 2, 3, 4 = 0 (Not Specified).
- All Other Routes (2~100) and Order Numbers (1~4) = 0 (Not Specified).
Conditions
None

None

Issue 5.0
Program 16 : Department Group Setup

16-01 : Department Group Basic Data Setup
Level:
IN
Description
Use Program 16-01 : Department Group Basic Data Setup to set the function mode for each
department group. There are 64 available Department Groups.
Input Data
Department Group Number 1~64
Related
Item
Item Input Data Default Progra
No. m
01 Department Name Maximum 12 characters No 11-07

setting
02 Department Calling Cycle 0 =Normal Routing 0 16-02

Use this option to set the call (Priority)
routing for Department Calling. 1 =Easy – UCD
Routing can be either circular Routing (Circular)
(cycles to all phones in group)
or priority (cycles to highest
priority extensions first).
03 Department Routing when 0 =Normal (Intercom 0 16-02

Busy (Auto Step Call) caller to busy
Use this option to set how the department
system routes an Intercom call member hears
to a busy Department Group busy)
member. Intercom callers to the 1 =Circular (Intercom
extension can either hear busy callers to busy
or route to the first available
department
department number. This only
member routes to
occurs for calls to the extension
directly, not the department
idle member)
number assigned in Program
11-07.
04 Hunting Mode 0 =Last extension is 0

Use this option to set the action called and hunting
taken when a call reaches the is stopped
last extension in the Department 1 =Circular
Group (0=hunting stopped,
1 =hunting repeats with circular
routing through the Department
Group).
13-116 AspireNet
Issue 5.0
Related
Item
Item Input Data Default Progra
No.
m
05 Extension Group All Ring 0 =Manual 0 11-16-10

Mode Operation
1 =Automatic
Determine whether calls ringing
a Department Group should ring
all extensions in the group
simultaneously automatically or
manually when using the
service code defined in Program
11-12-09.
 When set to (1)
Automatic, only ICM
Calls and DID Calls will
ring all the stations in the
Department Group.
06 STG Withdraw Mode 0 = Disable (Camp On) 0

1 = Enable (Overflow
Mode)
07 Call Recall Restriction for 0 =Disable (Recall) 0

STG
1 =Enable (No Recall)
Determine whether or not an
unanswered call transferred to a
Department Group should recall
the extension from which it was
transferred.
08 Maximum Queuing number 0 =No Queuing 0

for Department Group Call
1 =Queuing
To have Department Group calls
queue when busy, set this entry
to maximum queuing number.
09 Department Hunting No 0~64800 seconds 15

Answer Time
Set how long a call rings a
Department group extension
before hunting occurs.
10 Enhanced Hunt Type 0 =No queuing 0

Set the type of hunting for each 1 =Hunting When Busy
Extension (Department) Group. 2 =Hunting When Not
Answered
3 =Hunting When Busy
or No Answer
Conditions
None

Issue 5.0
13-118 AspireNet
Issue 5.0
Program 16 : Department Group Setup

16-02 : Department Group Assignment for Extensions
Level:
IN
Description
Use Program 16-02 : Department Group Assignment for Extensions to set the Department Groups.
The system uses these groups (64 Department Groups) for Department Calling. Assign pilot numbers to
Department Groups you set up in Program 11-07. This lets system users place calls to the departments.
Use Program 16-01 to set the priority of each extension in each Department Group. When a call comes
to the group, the extensions ring in order of their priority.
Input Data
Extension Number Maximum eight digits
Item Group Related

Priority Default Description
No. Number Program
01 1~64 1~999 1 – xxx Set up the Department Group called 11-07

(See by the pilot number and the extension 16-01
Note) priority when a group is called.
Call Pickup Groups are set up in
23-02.
 The initial value of a priority becomes the ports numerical order assigned in Program 11-02 and 11-04.
(Extension ports are 1~ 960. Virtual extension ports are 961~1472.)
Conditions
None


Issue 5.0
Program 20 : System Option Setup

20-01 : System Options
Level:
IN
Description
Use Program 20-01 : System Options to set various system options.
Input Data
Ite
Defaul Related
m Item Input Data t Description Program
No.
01 Operator Access 0 =Step Call 0 Use this program to set up priority of 20-17
Mode a call when calling an operator
1 =Circular
telephone.
02 Text Message Mode 0 =Call mode 0 Use this program to select the mode 11-11-14
1 =No Answer/ when calling the telephone which set 15-07-08
Busy mode up the text message.
04 Network BLF 0-64800 in 100ms 0 Used to determine how often the 30-05
Indication increments SV9100 updates the DSS key BLF
indications.
For NetLink, the entry should be “30”
in all SV9100s.
05 DTMF Receive 0~64800 seconds 10 For OPXs, analog telephones and 25-07-01
Active Time certain analog trunks (like DISA), the
system attaches a DTMF receiver to
the port for this interval. The system
releases the receiver after the
interval expires.
06 Alarm Duration 0~64800 seconds 30 This time sets the duration of the 11-12-05
alarm signal.
07 Callback Ring 0~64800 seconds 15 Callback rings an extension for this 11-12-05
Duration Time time. 15-07-35
08 Trunk Queuing 0~64800 seconds 15 Trunk Queuing callback rings an 11-12-05
Callback Time extension for this time. 15-07-35
09 Callback/Trunk 0~64800 seconds 64800 The system cancels an extension 11-12-05
Queuing Cancel Callback or Trunk Queuing request 15-07-35
Time after this time.
10 Trunk Guard Timer 0~64800 seconds 1 The amount of time the system waits
to seize the next outside line after
the system releases an outside line.
12 Telephone/Web Pro 1~84600 seconds 900 The system automatically logs out of
Logout Time (84600sec = a Telephone/Web Pro session after
1 day) inactivity lasting this time.
13-120 AspireNet
Issue 5.0
Conditions
None

Refer to the Input Data table at the beginning of this section.

Issue 5.0
Program 21 : Outgoing Call Setup

21-16 : Trunk Group Routing for Networks
Level:
IN
Description
Use Program 21-16 : Trunk Group Routing for Networks to assign Program 14-06 routes for a
networked system. This is required to seize the trunk in a networked system (Extension in System A
tries to make an external call using a trunk in System B).
The route number is specified for each system ID (01-50).
Input Data
System ID 01-50
Related
Item Day/Night Route Table
No. Mode Number Default Progra
m
01 1~8 0-100 1 14-06

(0=No setting)
Conditions
None

 Central Office Calls, Placing
 Networking - NetLink
13-122 AspireNet
Issue 5.0
Program 22 : Incoming Call Setup

22-05 : Incoming Trunk Ring Group Assignment
Level:
IN
Description
Use Program 22-05 : Incoming Trunk Ring Group Assignment to assign trunks to incoming Ring
Groups.There are 100 available Ring Groups.
Input Data
Item Day/Night Incoming Group Defaul Related

Description
No. Mode Number t Program
01 1~8 0 (No Setting) 1 Use this program to assign 22-04

001~100 Normal Ring Trunks (22-02) to 22-06
Incoming Ring Groups
(Incoming Group) (22-04).
102 (In-Skin/
External Voice Mail
or InMail)
103 Centralized VM
Conditions
None

 Ring Groups

Issue 5.0

22-08 : DIL/IRG No Answer Destination
Level:
IN
Description
Uor DIL Delayed Ringing, use Program 22-08 : DIL/IRG No Answer Destination to assign the DIL No
Answer Ring Group. An unanswered DIL rings this group after the DIL No Answer Time expires
(Program 22-01-04). DIL Delayed Ringing can also reroute outside calls ringing a Ring Group.
Make eight assignments, one for each Night Service mode.
Input Data
Item Day/Night
No. Mode Incoming Group Number Default
01 1~8 0 (No Setting) 0

001~100 (Incoming Ring Group)
102 (In-Skin/External Voice Mail
or InMail)
Conditions
None

 Direct Inward Line (DIL)
 Ring Group
13-124 AspireNet
Issue 5.0

22-10 : DID Translation Table Setup
Level:
IN
Description
Use Program 22-10 : DID Translation Table Setup to specify the size of the DID Translation Tables.
There are 4000 Translation Table entries that you can allocate among 20 Translation Tables.
Input Data
Conversion Table Area Number 01~20
Item
Item Input Data
No.
01 1st Area Setup (Start Address) 0~2000

(0 = No Setting)
1st Area Setup (End Address) Default Table
2nd Area Setup (Start Address) Default Table
2nd Area Setup (End Address) Default Table

Issue 5.0
Default Table
1st 2nd
Conversion
Table Area
Start Table End Table Start Table End Table
1 1 200 0 0
2 201 400 0 0
3 401 600 0 0
4 601 800 0 0
5 801 1000 0 0
6 1001 1200 0 0
7 1201 1400 0 0
8 1401 1600 0 0
9 1601 1800 0 0
10 1801 2000 0 0
: : : : :
20 0 0 0 0
Conditions
None

 Direct Inward Dialing (DID)
13-126 AspireNet
Issue 5.0

22-11 : DID Translation Number Conversion
Level:
SA
Description
Use Program 22-11 : DID Translation Table Number Conversion to specify for each Translation Table
entry (2000).
 The digits received by the system (eight maximum)
 The extension the system dials after translation (24 digits maximum)
 The name that should show on the dialed extension display when it rings (12 characters maximum)
 The Transfer Target – 1 and 2
 If the Transfer Targets are busy or receive no answer, those calls are transferred to the final transfer
destination (Program 22-10).
 Operation Mode
Use the following chart when entering and editing text for names. Press the key once for the first
character, twice for the second character, etc. For example, to enter a C, press 2 three times.
Key for Entering Names
When entering names in the procedures below, refer to this chart. Names can have up to 12 digits.
1 Enter characters:
1 @ [ ¥ ] ^ _ ` { | }  Á À Â Ã Ç É Ê ì ó
0 Enter characters:
0 ! “ # $ % & ’ ( ) ô Õ ú ä ö ü 

Issue 5.0
Key for Entering Names
When entering names in the procedures below, refer to this chart. Names can have up to 12 digits.
* Enter characters:
 + , - . / : ; < = > ? B E  S ¢ £
# # = Accepts an entry (only required if two letters on the same key are
needed - ex: TOM). Pressing # again = Space.
(In system programming mode, use the right arrow soft key instead to
accept and/or add a space.)
Recall Clear the character entry one character at a time.
HOLD Clear all the entries from the point of the flashing cursor and to the
right.
Input Data
Conversion Table Number 1~4000
Item
No.
01 Received Number Maximum eight digits See

This is the received DID digits.
Default
Value
02 Target Number Maximum 24 digits See

Enter the destination number to which the
Default
Value
DID number is sent.
03 DID Name Maximum 12 characters No Setting

This is the name that is assigned to the
DID digits when it rings the extension.
04 Transfer Operation Mode 0 =No Transfer 0

1 =Busy
2 =No Answer
3 =Busy/No Answer
13-128 AspireNet
Issue 5.0
Item
No.
05 Transfer Destination Number 1 0 = No Setting 0

1~100 = Incoming Group
06 Transfer Destination Number 2 0
102 = In-Skin/External Voice Mail
400 = Allows the outside party to dial a or InMail
different extension number in the 103 = Centralized VM
translation table (for example, ring no
201~264 = Extension Group
answer to a dialed number, the caller then
hears a dial tone, allowing them to enter 400 = Valid Extension Number
another Valid Extension Number). 401 = DISA
401 = Provides the caller with DISA 501~599 = DISA/VRS Message
dialing options (requires using the DISA 1000~1999 = Speed Dial Number
password). (000~999)
 This applies to 22-11-05 and
22-11-06.
07 Call Waiting 0 = Disable (No) 0

PRG 20-09-07 overrides this setting. 1 = Enable (Yes)
08 Maximum Number of DID Calls 0~200 (0 = No Limit) 0
09 Music on Hold Source 0 = IC/MOH Port 0

1 = BGM Port
2 = ACI Port
10 ACI Music Source Port When a sound source type is 2 in 0

above : (0~96)
11 Ring Group Transfer 0 = Disable (Caller will hear 1

Enable (1) or disable (0) each Ringback)
conversation tables ability to follow the 1 = Enabled (Go to normal ring)
Ring Group programming defined in
Program 22-12-01 : DID Intercept Ring
Group.
If Program 22-11-05 : DID Translation
Number Conversion, Transfer Destination
Number 1 and Program 22-11-06 : DID
Translation Number Conversion, Transfer
Destination Number 2 are set, the priority
of transferring is in this order: Program
22-11-05 then Program 22-11-06 then if
Program 22-11-11 is enabled, Program
22-12-01.

Issue 5.0
Default
The default value of PRG22-11-01/PRG22-11-02 is shown as below.

Conversion
Received Number Target Number
Table
1 00 200
: : :
99 98 298
100 99 299
101 No Setting No Setting
: : :
: : :
300 No setting No setting
: : :
: : :
500 No setting No setting
:
:
: : :
13-130 AspireNet
Issue 5.0
Conditions
- When the trunk type is set to 3 (DID) in 22-02-01, the DID Transfer Destination for each DID
feature is not supported. This feature is supported only for DID trunks when assigned as
VRS.


Issue 5.0

22-12 : DID Intercept Ring Group
Level:
IN
Description
For each DID Translation Table, use Program 22-12 : DID Intercept Ring Group to define the first
destination group for DID calls.
Depending on the entry in Program 22-09-02 and 22-11-04, the incoming calls route to the first
destination group by the following:
 Vacant number intercept (vacant number means that no phone is connected, no station blade is
installed, or the extension number is not defined in Program 11-02)
 Busy intercept
 Ring-no-answer intercept
If the destination is 0, the calls are forwarded to the trunk ring group defined in Program 22-11 based on
the table assigned to the DID trunk.
 If Programs 22-11-05 and 22-11-06 are set, the priority of transferring is in this order:
Program 22-11-05 + Program 22-11-06 + Program 22-12.
For busy and no-answer calls, if the first and third destinations are programmed, but the second
destination is not, the incoming call goes to the third destination after the first destination. If the first
and second destinations are not defined, but the third destination is, the call goes directly to the
third destination.
Input Data
Conversion Table Area Number 01~20
Item
Day/Night Mode Incoming Group Number Default
No.
01 1~8 0 (No Setting) 0

or InMail)
103 = Centralized VM
13-132 AspireNet
Issue 5.0
Conditions
None


Issue 5.0
Program 25 : VRS/DISA Setup

25-03 : VRS/DISA Transfer Ring Group With Incorrect Dialing
Level:
IN
Description
Use Program 25-03 : VRS/DISA Transfer Ring Group With Incorrect Dialing to set what happens to
a call when the DISA or Automated Attendant caller dials incorrectly or waits too long to dial. The call can
either disconnect (0) or Transfer to an alternate destination (a ring group or voice mail). When setting the
DISA and DID Operating Mode, make an entry for each Night Service mode.
Input Data
Item Day/Night Related

Incoming Group Number Default
No. Mode Program
01 1~8 0 (Disconnect) 0 22-04

or InMail)
103 = Centralized VM
Conditions
None

 Direct Inward System Access (DISA)
13-134 AspireNet
Issue 5.0

25-04 : VRS/DISA Transfer Ring Group With No Answer/Busy
Level:
IN
Description
Use Program 25-04 : VRS/DISA Transfer Ring Group With No Answer/Busy to set the operating
mode of each DISA trunk. This sets what happens to the call when the DISA or Automated Attendant
caller calls a busy or unanswered extension. The call can either disconnect (0) or Transfer to an
alternate destination (a ring group or voice mail). When setting the DISA and DID Operating Mode, make
an entry for each Night Service mode.
Input Data
Day/
Item Related
No. Night Incoming Group Number Default Program
Mode
01 1~8 0 (Disconnect) 0 22-04

102 (In-Skin/External Voice Mail or
InMail)
Conditions
None


Issue 5.0

25-08 : DISA User ID Setup
Level:
SA
Description
Use Program 25-08 : DISA User ID Setup to set the 6-digit DISA password for each user. There are 15
users each with one 6-digit password.
Input Data
DISA User Number 1~15
Item
Password Default Related PRG
No.
01 Dial DISA User 49-10-11

(Fixed – six digits) No.1-15: No
0~9, , # Setting
Conditions
None

13-136 AspireNet
Issue 5.0
Program 44 : ARS/F-Route Setup

44-01 : System Options for ARS/F-Route
Level:
IN
Description
Use Program 44-01 : System Options for ARS/F-Route to define the system options for the ARS/F-Route
feature.
Input Data
Ite
m Item Input Data Default
No.
01 ARS/F-Route Time Schedule 0 = Not Used 0

If this option is set to 0, the F-Route table selected is 1 = Used
determined only by the digits dialed without any relation to
the day or time of the call.
If this option is set to 1, the system first refers to Program
44-10. If there is a match, the pattern defined in that
program is used. If not, the F-Route pattern in Program
44-09 and time setting in 44-08 are used.
Conditions
None

 Automatic Route Selection (ARS)
 Uniform Numbering Network

Issue 5.0

44-02 : Dial Analysis Table for ARS/F-Route Access
Level:
IN
Description
Use Program 44-02 : Dial Analysis Table for ARS/F-Route Access to set the Pre-Transaction Table for
selecting ARS/F-Route.
Input Data
Dial Analysis Table Number 1~120
Item Defaul
Item Input Data
No. t
01 Dial Up to eight digits No

Set the number of digits to be analyzed by the (Use line key 1 for a Don’t Setting
system for ARS routing. Care digit, @)
02 Service Type 0 =No setting (None) 0

- Service Type 1 (Extension Number) 1 =Extension Call (Own)
The number goes to an extension after 2 =ARS/F-Route Table
deleting the front digit(s). (F-Route)
3 =Dial Extension Analyze
Additional Data
Table (Option)
Assign the digit(s) to be deleted on top of the
number for extension number usage. At least
one digit must be deleted.
- Service Type 2 (ARS/F-Route)
The number is controlled by ARS/F-Route
table.
Additional Data:
If the ARS/F-Route Time Schedule is not
used, assign the ARS/F-Route table number
for Program 44-05.
If the ARS/F-Route Time Schedule is used,

assign the ARS/F-Route selection number for
Program 44-04.
- Service Type 3 (Dial Extension Analyze Table)
The total length of the number exceeds more
than 8 digits.
Additional Data:
Assign the Dial Extension Analysis Table
number to be used in Program 44-03.
13-138 AspireNet
Issue 5.0
Item Defaul
Item Input Data
No. t
03 Additional Data 1 =Delete Digit = 0~255 0

For the Service Type selected in 44-02-02, enter (255 : Delete All Digits)
the additional data required. 2 =0~500
1: Delete Digit = 0~255 (0 = No Setting)
(255 = Delete All Digits) 3 =Dial Extension Analyze
2: [Program 44-01 : 0] Table Number = 0~4
ARS/F-Route Table Number = 0~500 (0 = No Setting)
(0 = No Setting)
Refer to Program 44-05.
[Program 44-01 : 1]
ARS/F-Route Select Table Number = 0~500

(0 = No Setting)
3: Dial Extension Analyze Table Number = 0~4 (0
= No Setting)
04 Dial Tone Simulation 0 =Off 0

If enabled, this option sends dial tone to the calling 1 =On
party after the routing is determined. This may be
required if the central office at the destination does
not send dial tone.
Conditions
None


Issue 5.0

44-03 : Dial Analysis Extension Table
Level:
IN
Description
When Program 44-02-02 is set to type 3, use Program 44-03 : Dial Analysis Extension Table to set
the dial extension analysis table. These tables are used when the analyzed digits must be more than
eight digits. If the received digits do not match the digits set in tables 1~250, table number 252 is used to
refer to the next Extension Table Area (1~4) to be searched. If the received digits are not identified in
tables 1~250, the F-Route selection table number defined in table 251 is used.
Input Data
Extension Table Area Number 1~4
Dial Analysis Table Number 1~252
Dial Analysis Table Number : 1~250
Item
No. Item Input Data Default
01 Dial Up to 24 digits No Setting

Digits = 1~9, 0, *, #, @
(Press Line Key 1 for wild character @)
02 ARS/F-Route Select 0~500 (ARS/F-Route Table Number) 0

Table Number
With Program 44-01 set to 0, Program 44-05 is
checked.
checked.
13-140 AspireNet
Issue 5.0
Dial Analysis Table Number : 251
Item
No.
03 ARS/F-Route Select 0~500 (ARS/F-Route Table Number) 0

Table Number
checked.
checked.
Dial Analysis Table Number : 252
Item
No. Item Input Data Default
04 Next Table Area Number 0~4 0
Conditions
None


Issue 5.0

44-04 : ARS/F-Route Selection for Time Schedule
Level:
IN
Description
Use Program 44-04 : ARS/F-Route Selection for Time Schedule to assign each ARS/F-Route Selection
number to an ARS/F-Route table number for each ARS/F-Route time mode. There are eight time modes
for ARS/F-Route Access.
Input Data
ARS/F-Route Selection Number 1~500
Ite ARS/F-
m Route Time ARS/F-Route Default
Table Number
No. Mode
01 1~8 0~500 0
Conditions
None

13-142 AspireNet
Issue 5.0

44-05 : ARS/F-Route Table
Level:
IN
Description
Use Program 44-05 : ARS/F-Route Table to set the ARS/F-Route table. There are four kinds of order. If the
higher priority trunk groups are busy, the next order group is used. If a lower priority route is selected, the
caller may be notified with a beep tone.
Input Data
ARS/F-Route Table Number 1~500
Priority Number 1~4
Ite
No.
01 Trunk Group Number 0~100, 255 0

Select the trunk group number to be used for the 0 = No Setting
outgoing ARS call. 255 = Extension Call
02 Delete Digits 0~255 0

Enter the number of digits to be deleted from the (255 = Delete All)
dialed number.
03 Additional Dial Number Table 0~1000 0

Enter the table number (defined in Program 44-06)
for additional digits to be dialed.
04 Beep Tone 0 = Off 0

Select whether or not a beep is heard if a lower 1 = On
priority trunk group is used to dial out.
05 Gain Table Number for Internal Calls 0~500 0

Select the gain table number to be used for the 0 = No Setting
internal call (defined in Program 44-07).
06 Gain Table Number for Tandem Connections 0~500 0

Select the gain table number to be used for the 0 = No Setting
tandem call (defined in Program 44-07).
07 ARS Class of Service 0~16 0

Select the ARS Class of Service to be used for the
table. An extension ARS COS is determined in
Program 26-04-01.

Issue 5.0
Ite
No.
08 Dial Treatment 0~15 0

Select the Dial Treatment to be used for the table. If
a Dial Treatment is selected, Programs 44-05-02
and 44-05-03 are ignored and the Dial Treatment
defined in Program 26-03-01 is used instead.
09 Maximum Digit 0~24 0

Input the maximum number of digits to send when
using the F-Route.
10 CCIS over IP Destination Point Code 0~16367 0

Input the Destination Point Code to send when using
this F-Route.
11 Network Specified Parameter Table 0~16 0

Enter a table number from Program 26-12.
Conditions
None

13-144 AspireNet
Issue 5.0

44-06 : Additional Dial Table
Level:
IN
Description
Use Program 44-06 : Additional Dial Table to set the additional dial table to add prior to the dialed ARS/F-
Route number. The Additional Dial Table used is determined in Program 44-05-03.
Input Data
Additional Dial Table Number 1~1000
Item
Additional Dial Default
No.
01 Up to 24 digits No Setting
Enter: 1~9, 0, *, #, Pause (press LK 1 to enter a pause)
Conditions
None


Issue 5.0

44-07 : Gain Table for ARS/F-Route Access
Level:
IN
Description
Use Program 44-07 : Gain Table for ARS/F-Route Access to set the gain/PAD table. If an extension dials
ARS/F-Route number:
 The Extension Dial Gain Table, assigned in Program 44-05, is activated.
 The Extension Dial Gain Table follows Outgoing transmit and Outgoing receive settings.
If the incoming call is transferred to another line using ARS/F-Route:

 The Tandem Gain Table, assigned in Program 44-05, is activated.
 The Tandem Gain Table follows the Incoming transmit and Incoming receive settings for incoming line,
and Outgoing transmit and Outgoing receive settings for the outgoing line.
 For ARS/F-Route calls, the CODEC gains defined in Program 14-01-02 and 14-01-03 are not activated.
Input Data
Gain Table Number 1~500
Item
No.
01 Incoming Transmit 1~57 32

(-15.5 ~ +12.5dB) (0dB)
02 Incoming Receive 1~57 32

(-15.5 ~ +12.5dB) (0dB)
03 Outgoing Transmit 1~57 32

(-15.5 ~ +12.5dB) (0dB)
04 Outgoing Receive 1~57 32

(-15.5 ~ +12.5dB) (0dB)
13-146 AspireNet
Issue 5.0
Conditions
None


Issue 5.0

44-08 : Time Schedule for ARS/F-Route
Level:
IN
Description
Use Program 44-08 : Time Schedule for ARS/F-Route to define the daily pattern of the ARS/F-Route
feature. ARS/F-Route has 10 time patterns. These patterns are used in Program 44-09 and 44-10. The
daily pattern consists of 20 time settings.
Input Data
Schedule Pattern Number 01~10
Item Time
No. Number Start Time End Time Mode
01 01~20 0000~2359 0000~2359 1~8
Default
All Schedule Patterns : 0:00 – 0:00, Mode 1
Example:
Pattern 1
0:00 8:00 18:00 22:00 0:00

Mode 3 Mode 1 Mode 2 Mode 3
Time Number 01 : 00:00 – 08:00 Mode 3

Time Number 02 : 08:00 – 18:00 Mode 1
Time Number 03 : 18:00 – 22:00 Mode 2
Time Number 04 : 22:00 – 00:00 Mode 3
Pattern 2
0:00 0:00
Mode 2
Time Number 01 : 0:00 – 0:00 Mode 2
13-148 AspireNet
Issue 5.0
Conditions
None


Issue 5.0

44-09 : Weekly Schedule for ARS/F-Route
Level:
IN
Description
Use Program 44-09 : Weekly Schedule for ARS/F-Route to define a weekly schedule for using ARS/F-
Route. The pattern number is defined in Program 44-08-01.
Input Data
Schedule
Item
No. Day Number Pattern Default
Number
1 =Sunday 1~10 Pattern 1
2 =Monday 1~10 Pattern 1
3 =Tuesday 1~10 Pattern 1
01 4 =Wednesday 1~10 Pattern 1
5 =Thursday 1~10 Pattern 1
6 =Friday 1~10 Pattern 1
7 =Saturday 1~10 Pattern 1
Conditions
None

13-150 AspireNet
Issue 5.0

44-10 : Holiday Schedule for ARS/F-Route
Level:
IN
Description
Use Program 44-10 : Holiday Schedule for ARS/F-Route to define a yearly schedule for ARS/F-Route.
This schedule is used for setting special days such as national holidays. The pattern number is defined
in Program 44-08-01.
Input Data
Item
Date Schedule Pattern Number Default
No.
01 0101~1231 0~10 (0 = No Setting) 0
Conditions
None


Issue 5.0
Program 45 : Voice Mail Integration

45-01 : Voice Mail Integration Options
Level:
IN
Description
Use Program 45-01 : Voice Mail Integration Options to customize certain voice mail integration options.
Input Data
Item
No.
01 Voice Mail Department Group Number 0~64 0

Assign which Extension (Department) Group 0 = No Voice Mail
number is to be assigned as the voice mail
group.
02 Voice Mail Master Name Up to 12 Characters VOICE

Enter the Voice Mail Master Name.
MAIL
04 Park and Page 0 = Off 1

Enable/disable the system ability to process the 1 = On
Voice Mail Park and Page () commands. You
should normally enable this option.
05 Message Wait 0 = Off 1

Enable/disable the system ability to process the 1 = On
Voice Mail Message Wait (#) commands. You
should normally enable this option. If enabled,
be sure that the programmed Message
Notification strings don’t contain the code for
trunk access.
06 Record Alert Tone Interval Time 0~64800 (sec) 30

This time sets the interval between Voice Mail
Conversation Record alerts
07 Centralized Voice Mail Pilot Number Dial (up to eight No Setting

This number is the same as the extension digits)
number or pilot number.
08 Centralized Voice Mail Department Group 0~64 0

Number
Assigns which Extension (Department) Group
Number is to be assigned as the Centralized
Voice Mail group.
13-152 AspireNet
Issue 5.0
Item
No.
09 Centralized Voice Mail Master Name Up to 12 characters “C.V.M.”

Assigns the Centralized Voice Mail Master
Name.
10 New NSL Protocol support 0 = Off 0

1 = On
11 Prefix for Call Screening Dial (One digit) 1
12 Prefix for Park and Page Dial (One digit) 
13 Prefix for Message Wait Dial (One digit) #
14 CCIS Centralized Voice Mail Number Dial (up to eight No Setting

Assign the pilot number to Centralized Voice digits)
Mail over CCIS Link. This is assigned only in
the remote switches.
15 Analog Voice Mail Protocol Selection 0: Fixed 0

Assigns whether fixed codes are used or the 1: Program
codes used in PRG 45-04 are used for analog
voice mail protocol.
16 Voice Mail Fax Digit Add Assignment Up to four digits None

Assign up to four digits in front of the station
number sent to the SLT port when a call is
forwarded.
17 Reply Mailbox Number 0: No 1

Whether or not to include the mailbox number 1: Yes
in the analog voice mail protocol.
18 Trunk Number Mapping 2~3 2

Assign the digits of trunk number mapping.
19 Centralized Voice Mail Type 0 = Retro 1

Assign which Centralized Voice Mail types to 1 = Enhanced
use, Retro (Aspire) or Enhanced (Cygnus).
Conditions
None

 Voice Mail Integration (Analog)

Issue 5.0

84-01 : H.323 Trunk Basic Information Setup
Level:
IN
Description
Use Program 84-01 : H.323 Trunk Basic Information Setup to set the basic information of the H.323
Trunk.
Input Data
Item
No.
02 Number of G.711 audio frames 1~4 3
03 G.711 VAD mode 0 = Disable 0

1 = Enable
04 G.711 Type 0 = A-law 0

1 = -law

1 = 10ms
2 = 20ms
3 = 30ms
4 = 40ms
5 = 50ms
6 = 60ms
06 G.729 VAD mode 0 = Disable 0

1 = Enable
07 G.729 Jitter Buffer( min) 0~300ms 30

Setting 2 (Self adjusting-silent period-is not valid by 3 = Self adjusting
IPLB. If this value is set, the system will operate as
Setting 3.
16 G.711 Jitter Buffer( min) 0~300ms 30
13-154 AspireNet
Issue 5.0
Item
No.
22 VAD Threshold 0~30 (-19dB~ +10dB and self 20

adjustment)
0 = Self adjustment
1 = -19dB (-49dBm)
:
20 = 0dB (-30dBm)
:
29 = 9dB (-21dBm)
30 = 10dB (-20dBm)
33 Priority CODEC setting 0~3 0

Priority of voice encoding method. 0 = G.711
2 = G.729
3 = G.722

1 = 10ms
2 = 20ms
3 = 30ms
4 = 40ms
68 RTP Filter 0 = Disable 1

To avoid incorrect voice pass connection, this Program 1 = Enable
checks the sending side address from received RTP
packet at VoIPDB.
Conditions
None


Issue 5.0

84-02 : H.225 and H.245 Information Basic Setup
Level:
IN
Description
Use Program 84-02 : H.225 and H.245 Information Basic Setup to define the basic setup information
of H.225 and H.245.
Input Data
Item
No.
01 H.225 0~255sec 180
02 H.225 Setup Acknowledge Timer 0~255sec 9
03 H.225 Setup Timer 0~255sec 4
04 H.225 Info Ack Timer 0~255sec 9
05 H.225 Call Proceeding Timer 0~255sec 10
07 H.245 Master Slave Determination Timer 0~255sec 5
08 H.245 Master Slave Determination Retry Count 0~255sec 3
09 H.245 Capability Exchange Timer 0~255sec 5
10 H.245 Logical Channel Establishment Timer 0~255sec 50
11 H.245 Mode Request Procedures Timer 0~255sec 50
12 H.245 Close Logical Channel Timer 0~255sec 50
13 H.245 Round Trip Delay Timer 0~255sec 50
14 H.245 Maintenance Loop 0~255sec 50
15 RAS GRQ Timer 0~255sec 5
16 GRQ Retry Count 0~255 2
17 RAS RRQ Timer 0~255sec 5
18 RRQ Retry Count 0~255 3
19 RAS URQ Timer 0~255sec 3
20 URQ Retry Count 0~255 1
21 RAS ARQ Timer 0~255sec 5
13-156 AspireNet
Issue 5.0
Input Data
Item
No.
22 ARQ Retry Count 0~255 2
23 RAS BRQ Timer 0~255sec 5
24 BRQ Retry Count 0~255 2
25 RAS IRR Timer 0~255sec 5
26 IRR Retry Count 0~255 2
27 RAS DRQ Timer 0~255sec 8
28 DRQ Retry Count 0~255 2
29 RAS LRQ Timer 0~255sec 5
30 LRQ Retry Count 0~255 2
31 RAS RAI Timer 0~255sec 3
32 RAI Retry Count 0~255 2
33 Call Signaling Port Number 0~65535: 1730

0~1719,
1721~65535
35 Fast Start Mode 0 = Disable 1

1 = Enable0
36 RAS Unicast Port Number 0~65535 20001
37 Terminal Type setting 0~255 60
Conditions
None


Issue 5.0

84-12 : Networking CODEC Information Basic Setup
Level:
IN
Description
Use Program 84-12: Networking CODEC Information Basic Setup to define the CODEC Information
for Networking.
Input Data
Item
No.
01 Number of G.711 Audio Frame 1 =10 ms 3

Maximum number of G711 Audio 2 =20 ms
Frames. When the voice is 3 =30 ms
encoded using the PCM (Pulse 4 =40 ms
Code Modulation) method, a unit is
a frame of 10ms.
02 G.711 Silence Detection (VAD) 0 =Disable 0

Mode
1 =Enable
Select whether to compress silence
with G.711. When there is silence,
the RTP packet is not sent.
03 G.711 Type 0 =A-law 0

Set the type of G.711. 1 =µ-law
04 G.711 Jitter Buffer - Minimum 0~160 ms 30

Set the minimum value of the G.711
Jitter Buffer.
05 G.711 Jitter Buffer - Standard 0~160 ms 60

Set the average value of the G.711
Jitter Buffer.
06 G.711 Jitter Buffer - Maximum 0~160 ms 120

Set the maximum value of the
G.711 Jitter Buffer.
07 G.729 Audio Frame 1-6 3

Maximum number of G729 Audio (1 = 10ms, 2 = 20ms, etc.)
Frames. G.729 assumes the audio
signal made by a specimen by
8kHz and the frame of 10ms is
assumed to be a unit to 8kbps by
the encoding compressed method.
13-158 AspireNet
Issue 5.0
Input Data
Item
No.
08 G.729 Silence Compression 0 =Disable 0

(VAD) Mode
1 =Enable
09 G.729 Jitter Buffer - Minimum 0-270 ms 30

Set the minimum value of the Jitter
Buffer of G.729 is set. Jitter is the
variation in the time between
packets arriving and the buffer
allows this variation to be
absorbed.
10 G.729 Jitter Buffer - Standard 0-270 ms 60

Set the average G.729 Jitter Buffer.
11 G.729 Jitter Buffer - Maximum 0-270 ms 120

Set the maximum G.729 Jitter
Buffer.
12 Number of G.723 Audio Frame 1 =30 msec 1

Maximum number of the G.723 2 =60 msec
Audio Frame.

(VAD) Mode
1 =Enable
If enabled, RTP packets are not
sent for the compressed silence.
14 G.723 Jitter Buffer - Minimum 0~270 ms 30

Set the minimum value of the
15 G.723 Jitter Buffer - Standard 0~270 ms 60

Set the average value of the G.723
Jitter Buffer.
16 G.723 Jitter Buffer - Maximum 0~270 ms 120

Set the maximum value of the
17 Jitter Buffer Mode 1 =static 3

Set the mode of the Jitter Buffer. 2 =adaptive during silence
1 = Size set to the fixed amount 3 =adaptive immediately
for the codec.
2 = The minimum/maximum
range for the codec is used.
3 = The minimum/maximum
range for the codec is used
and adjusts at any time,
regardless of silence.

Issue 5.0
Input Data
Item
No.
18 Silence Compression (VAD) 0-30 20

Threshold
(self-adjustment and
Set the voice level judged to be -19db ~ +10db)
silence. 0 = self-adjustment
Change value based .30
1:-19db (-49dbm)
This entry is ignored if silence
compression is disabled in 84-01- :
03 with G.711, or 84-01-06 with 20 = 0db (-30dbm)
G.729. :
29 = 9dbm (-21dbm)
30:10dbm (-20dbm)
19 Idle Noise Level 5000-7000 7000

Set the noise level which is (-5000 ~ -7000dbm)
generated when silent. 5000 = -5000dbm
:
7000 = -7000dbm
20 Echo Canceller Mode 0 =Disable 1

Determine whether or not to use 1 =Enable
Echo canceller.
21 Signal limiter 1 =Mode0 6

Set the Signal Limiter Mode. 2 =Mode1
3 =Mode2
4 =Mode3
5 =Mode4
6 =Mode5
22 Echo Canceller NLP Mode 0 =2 wire & 4 wire 1

Non-linear processing mode. 1 =2 wire only
Enable this option to decrease the
low level echo. When NLP is
enabled, the voice with low level is
replacecd with NLP noise. As a
result, a low echo of the level is
usually removed compared with the
conversation level.
23 --- Not Used ---
24 Echo Canceller NLP Noise 0 =adaptive 0

Setting
1 =fixed
Becomes invalid item if 84-12-22 is
set to Disabled.
Set the noise level adjusting
method added with NLP.
When “0” is set, the level is self-
adjusted - when “1” is set, Program
84-13-23 is used.
25 - Not Used - - -
13-160 AspireNet
Issue 5.0
Input Data
Item
No.
26 TX (Transmit) Gain 0-40 (-20 ~ +20) 20 (0 dbm)

Define the setting to amplify and to 0 = -20 dbm
attenuate the size of the 1 = -19 dbm
transmission voice. The gain given :
when the voice packet is sent from 20 = 0 dbm
the VOIPDB is set.
:
39 = 19 dbm
40 = 20 dbm
27 RX (Receive) Gain 0-40 (-20~+20) 20 (0 dbm)

Define the setting to amplify and to 0 = -20 dbm
attenuate the size of the received 1 = -19 dbm
voice. The gain given when the :
voice packet is received from the 20 = 0 dbm
VOIPDB is set.
:
39 = 19 dbm
40 = 20 dbm
28 Priority Codec Setting 0 =G711 PT 0

The option selected here 1 =G723 PT
determines what other codec 2 =G729 PT
options are applied by priority. 3 =G.722 PT
30 EchoAuto Gain Control 0-5 0

Define the Auto Gain Control.
31 DTMF Relay Mode 0 =Disable 2

The initial setup information for the 1 =RFC2833
VOIPU is set in Program 84-27-02. 2 =VoIPU
If this option is set to either 0 or 1,
priority is given.
32 FAX Relay Mode 0 =Disable 0

Select "2" for FAX Relay to SLT 1 =Enable
(Program 15-03-03:special), Trunk 2 =Each Port Mode
and NetLink.
Refer to Program 84-01-36 through
84-01-58 for FAX Relay options.
33 G.722 Audio Frame 1 =10 ms 3

Maximum number of G.722 Audio 2 =20 ms
Frames. G.722 assumes the audio 3 =30 ms
signal made by a specimen by 4 =40 ms
16kHz and the frame of 10ms is
assumed to be a unit to 64kbps by
the encoding compressed method.

Mode
1 =Enable

Issue 5.0
Input Data
Item
No.
35 G.722 Jitter Buffer - Minimum 0-160 ms 30

Set the minimum value of the Jitter
Buffer of G.722 is set. Jitter is the
variation in the time between
packets arriving and the buffer
allows this variation to be
absorbed.
36 G.722 Jitter Buffer - Standard 0-160 ms 60

Set the average G.722 Jitter Buffer.
37 G.722 Jitter Buffer - Maximum 0-160 ms 120

Set the maximum G.722 Jitter
Buffer.
Conditions
None

13-162 AspireNet
Issue 5.0

84-26 : IPL Basic Setup
Level:
IN
Description
Use Program 84-26 : IPL Basic Setup to set the IP address of IPL and the port.
Input Data
Slot Number 1
Item
No.
01 IP Address xxx.xxx.xxx.xxx Slot 1 = 172.16.0.20
02 RTP Port Number 0~65534 VoIP GW1 = 10020
03 RTCP Port Number 0~65534 VoIP GW1 = 10021

(RTP Port Number +1)
12 Video RTP Port 0 ~ 65534 20020

Sets the starting RTP port used by
standard SIP terminal video.
13 Video RTCP Port 0 ~ 65534 20021

Sets the starting RTCP port used by
standard SIP terminal video.
Conditions
None


Issue 5.0
SECTION 5 ASPIRENET EXAMPLE

The examples in this section are only a guide to the configuration required on the SV9100 to install AspireNet
Operation.
5.1 AspireNet - ISDN BRI
AspireNet Basic Rate ISDN is supported on all BRI PCB’s (2, 4 or 8 BRIU). Each circuit of the BRIU PCB
can be set independently to Trunk / S0 / AspireNet mode.
Each circuit set as AspireNet mode will require 2 of the AspireNet ports on the system, there are a total of
256 AspireNet ports available.
The example below shows two SV9100 systems, each has a BRIU card installed with one circuit set to
Mode 4 for AspireNet . The two BRIU circuits are cross connected via a 2-pair twisted cable.
SV9100 A has extension numbers in the range 200-299, SV9100 B has 300-399.
2-pair twisted cable
SV9100 A SV9100 B
SV9100 A SV9100 B
Program 10-03 BRI PCB Setup Program 10-03 BRI PCB Setup
10-03-01 = Mode 4 10-03-01 = Mode 4
10-03-03 = P-P 10-03-03 = P-P
10-03-10 = Master 10-03-10 = Slave
10-03-11 = 1 10-03-11 = 1
Program 11-01-01 System Numbering Program 11-01-01 System Numbering
Dial 2x = 3 digit, Type 2 (Intercom) Dial 2x = 3 digit, Type 8 (Networking),
Dial 3x = 3 digit, Type 8 (Networking), ID=1
ID=1 Dial 3x = 3 digit, Type 2 (Intercom)
More than one circuit required.

If more BRI circuits were added then Program 10-03-xx would copied for each new circuit.
Master/Slave setting
The choice of Master/Slave of each circuit is not important in the example above as there is no ISDN Net-
work connection to either of the systems.
If System A had an ISDN Network connection (BRI trunk or PRI trunk) then System A would be the Master
for AspireNet circuits.
If both system A and System B have an ISDN Network connection then the setting of Master/Slave is not
important.
When the AspireNet BRI circuit is connected to external equipment then ensure that the equipment is set
to Master (Clock generator) or Slave (Clock receiver) accordingly.
13-164 AspireNet
Issue 5.0
AspireNet - ISDN PRI
5.2 AspireNet - ISDN PRI

AspireNet Primary Rate ISDN is supported on the standard PRTA Blade’s. Each PRTA Blade can be set
independently to Trunk / S0 / AspireNet mode.
When set to AspireNet a PRTA Blade will require 30 of the AspireNet ports on the system, there are a total
of 256 AspireNet ports available.
The example below shows two SV9100 systems, each has a PRTA Blade installed which are set to Mode
4 for AspireNet . The two PRTA Blade’s are cross connected via a 2-pair twisted cable.
2-pair twisted cable
SV9100 A SV9100 B
SV9100 A SV9100 B
Program 10-03 PRI PCB Setup Program 10-03 PRI PCB Setup
10-03-01 = Mode 4 10-03-01 = Mode 4
10-03-03 = CRC-4 ON 10-03-03 = CRC-4 ON
10-03-10 = Master 10-03-10 = Slave
10-03-11 = 1 10-03-11 = 1
More than one circuit required.

If more PRI PCBs were added then Program 10-03-xx would copied for each new PCB.
CRC-4 Setting
The choice of CRC-4 on/off is not important in the example. When connected to external PRI equipment
ensure that this option is set the same as the external equipment.
Master/Slave setting
The choice of Master/Slave of each circuit is not important in the example above as there is no ISDN Net-
work connection to either of the systems.
If System A had an ISDN Network connection (BRI trunk or PRI trunk) then System A would be the Master
for AspireNet circuits.
If both system A and System B have an ISDN Network connection then the setting of Master/Slave is not
important.
When the AspireNet PRI circuit is connected to external equipment then ensure that the equipment is set
to Master (Clock generator) or Slave (Clock receiver) accordingly.
Limit the Quantity of B-Channels available

Use Program 10-32-01 to limit the quantity of channels available.

Issue 5.0
AspireNet - IP
5.3 AspireNet - IP
5.3.1 VoIP Networking Example

AspireNet IP requires the VOIPDB Blade and SV9100 Networking License. The VOIPDB Blade provides
the speech resources.
The example below shows two SV9100 systems, each has a VOIPDB Blade installed.
The VOIPDB Blades are connected to a hub via a standard LAN patch cable.
Hub
IPLE IPLE
SV9100 A SV9100 B
SV9100 A SV9100 B
IP Address of the IPLE = 172.16.0.10 IP Address of the IPLE= 172.16.5.10
Subnet Mask = 255.255.0.0 Subnet Mask = 255.255.0.0
DSP Resource is assigned for the Net- DSP Resource is assigned for the Net-
work. work.
=3 =3
Program 10-20-01 TCP Port Setup Program 10-20-01 TCP Port Setup
Device 4 = 30000 Device 4 = 30000
Program 10-27-01 Destination IP Address Program 10-27-01 Destination IP Address
and port number for each ID and port number for each ID
ID 1 = IP Address 172.16.5.10 ID 1 = IP Address 172.16.0.10
Procedure port must be set to 1730 Procedure port must be set to 1730
Enable Fast Start Enable Fast Start
Program 84-12-31 DTMF Relay Mode Program 84-12-31 DTMF Relay Mode
1 = RFC2833 (default: VoIPDB) 1 = RFC2833 (default: VoIPDB)
Program 84-26-01 IP address for each Program 84-26-01 IP address for each
DSP DSP
172.16.0.20 172.16.5.20
13-166 AspireNet
Issue 5.0
Explanation of routing for IP AspireNet .

When a user places a call to a remote extension number the dialed digits are checked against the Number-
ing Scheme in Program 11-01. This will provide the Node ID number of the route to the remote system.
The SV9100 will then find the destination IP Address by searching Program 10-27 for the destination IP
Address for the given Node ID number.
Quantity of Voice Paths Available.

The quantity of voice calls that can be made at any one time are dependent on various factors.
IPLA Resource - A free IPLA resource must be available at each SV9100 system.
Program 10-19 - The mode of each VOIPDB resource can be configured, modes 0 (ICM/Trunk) or 3 (NTW)
can be used by AspireNet calls.
IP Network Bandwidth Restrictions - These limitations are beyond the scope of the SV9100 . If there is not
enough bandwidth available the call can not be maintained.

Issue 5.0
5.3.2 Multi-Site Networking Example

Multi-Site Networking is a network of three or more SV9100 systems. The network can consist of either
ISDN or IP network connections.
The example below shows three SV9100 systems, each has a VOIPDB Blade installed. The VOIPDB
Blades are connected to a hub via a standard LAN patch cable.
SV9100 A has extension numbers in the range 200-299, SV9100 B has 300-399, and SV9100 C has 400-
499.
Hub
IPLE IPLE
IPLE
SV9100 A SV9100 B
SV9100 C
SV9100 A SV9100 B SV9100 C

IP Address of the IPLE = 172.16.0.10 IP Address of the IPLE = 172.16.5.10 IP Address of the IPLE
Program 10-12-10 Program 10-12-10 =172.16.10.10
Subnet Mask = 255.255.0.0 Subnet Mask = 255.255.0.0 Program 10-12-10
Subnet Mask = 255.255.0.0
DSP Resource is assigned for the DSP Resource is assigned for the DSP Resource is assigned for the
Network. Network. Network.
=3 =3 =3
Program 10-20-01 TCP Port Setup Program 10-20-01 TCP Port Setup Program 10-20-01 TCP Port Setup
Device 4 = 30000 Device 4 = 30000 Device 4 = 30000
Program 10-27-01 Destination IP Program 10-27-01 Destination IP Program 10-27-01 Destination IP
Address and port number for each Address and port number for each ID Address and port number for each ID
ID ID 1 = IP Address 172.16.0.10 ID 1 = IP Address 172.16.0.10
ID 1 = IP Address 172.16.5.10 Procedure port must be set to 1730 Procedure port must be set to 1730
Procedure port must be set to 1730 ID 2 = IP Address 172.16.10.10 ID 2 = IP Address 172.16.5.10
ID 2 = IP Address 172.16.10.10 Procedure port must be set to 1730 Procedure port must be set to 1730
Procedure port must be set to 1730
Program 11-01-01 System Number- Program 11-01-01 System Number- Program 11-01-01 System Number-
ing ing ing
Dial 2x = 3 digit, Type 2 (Intercom) Dial 2x = 3 digit, Type 8 (Network- Dial 2x = 3 digit, Type 8 (Network-
Dial 3x = 3 digit, Type 8 (Network- ing), ID=1 ing), ID=1
ing), ID=1 Dial 3x = 3 digit, Type 2 (Intercom) Dial 3x = 3 digit, Type 8 (Network-
Dial 4x = 3 digit, Type 8 (Network- Dial 4x = 3 digit, Type 8 (Network- ing), ID=1
ing), ID=2 ing), ID=2 Dial 4x = 3 digit, Type 2 (Intercom)
Enable Fast Start Enable Fast Start Enable Fast Start
13-168 AspireNet
Issue 5.0
Program 84-12-31 DTMF Relay Program 84-12-31 DTMF Relay Program 84-12-31 DTMF Relay
Mode Mode Mode
1 = RFC2833 (default: VoIPDB) 1 = RFC2833 (default: VoIPDB) 1 = RFC2833 (default: VoIPDB)
Program 84-26-01 IP address for Program 84-26-01 IP address for Program 84-26-01 IP address for
each DSP each DSP each DSP
172.16.0.20 172.16.5.20 172.16.10.20
 After programming, each system must be reset in order for the IP address changes to take affect.

Networking Manual
NEC Corporation
Issue 5.0

SV9100 Networking Manual GE 5 0

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What are the main components of the SV9100 system?

What are the main components of the SV9100 system?

How is the network structured between multiple SV9100 systems?

How is the network structured between multiple SV9100 systems?

Networking Manual

UNIVERGE is a registered trademark of NEC Corporation. Windows is a registered trademark

Copyright 2014 - 2016

Section 2 Manual Organization ............................................................................... 1-1

Section 3 Common Terms ....................................................................................... 1-1

Book 1 – SV9100 K-CCIS

1.1 Common Channel Handler (GCD-CCTA) ............................................ 1-2

Section 2 DTI Specifications ................................................................................... 1-3

2.1 Characteristics .................................................................................... 1-3

Section 3 Network Structure Considerations ........................................................ 1-7

3.1 Determining System Configurations .................................................... 1-7

Chapter 2 Hardware Installation <US Only>

1.1 Busying Out Extension/Line Blades ..................................................... 2-2

Section 2 Installing an Extension or Trunk Blade ................................................. 2-2

2.1 Installing the Blades ............................................................................. 2-2

2.2 Order of Installing Extension Blades .................................................... 2-3

2.3 Order of Installing Trunk Blades .......................................................... 2-4

SV9100 Networking Manual i

2.3.1 Installing COIU-LS1/LG1, GCD-4ODTA, GCD-4DIOPA, or GCD-2BRIA

Section 3 Installing the GCD-CCTA (CCIS Trunk Interface) ................................. 2-5

3.1 Description ........................................................................................... 2-5

3.2 Installation ........................................................................................... 2-6

3.3 LED Indications ................................................................................... 2-6

3.4 Connectors .......................................................................................... 2-7

Chapter 3 System Data Programming <US Only>

1.1 Digital Trunk Data Assignment ............................................................ 3-1

1.2 CCIS Assignment ................................................................................ 3-3

1.3 Numbering Plan Assignment ............................................................... 3-4

1.5 Programming for Open Numbering Plan ............................................. 3-5

1.6 Closed Number Programming Example .............................................. 3-7

1.7 Open Number Programming Example .............................................. 3-11

1.8 Dedicated Tandem CO Trunk Calls ................................................... 3-15

1.9 Shared Tandem CO Trunk Calls ....................................................... 3-17

Chapter 4 Features and Specifications

Section 2 System Availability ................................................................................. 4-2

2.1 Terminal Type ...................................................................................... 4-2

2.2 Required Components ......................................................................... 4-2

2.3 Operating Procedures ......................................................................... 4-2

Section 3 Service Conditions ................................................................................. 4-3

Section 4 Related Feature List ................................................................................ 4-5

Section 5 K-CCIS Features ...................................................................................... 4-5

SV9100 Networking Manual iii

Chapter 5 SV9100 IP K-CCIS

1.1 IP K-CCIS Application using the GPZ-IPLE ........................................ 5-1

1.2 Description ........................................................................................... 5-2

1.3 Systems Requirements ....................................................................... 5-3

1.3.1 Hot Swap ......................................................................................... 5-4

1.3.2 Connectors ...................................................................................... 5-4

1.3.3 Ethernet Status (IPLE) ..................................................................... 5-4

1.3.4 Environmental Conditions ................................................................ 5-4

1.3.5 Installation Procedure ...................................................................... 5-4

1.3.6 Switch Settings ................................................................................ 5-4

1.3.7 LED Indications ............................................................................... 5-5

1.3.8 Connectors ...................................................................................... 5-8

1.4 IP Addressing ..................................................................................... 5-9

1.4.1 General IP Configuration ............................................................... 5-10