Spectralink 84-Series Wireless Telephones Administration Guide

Spectralink 84-Series Wireless Telephone
Administration Guide
1725-86984-000 Rev: R
March 2017
Spectralink 84-Series Wireless Telephones Administration Guide
Copyright Notice
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express written permission of Spectralink Corporation.
Do not remove (or allow any third party to remove) any product identification, copyright or other
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Notice
Spectralink Corporation has prepared this document for use by Spectralink personnel and
customers. The drawings and specifications contained herein are the property of Spectralink and
shall be neither reproduced in whole or in part without the prior written approval of Spectralink,
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The Product Warranty and Software License and Warranty and other support documents are
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Contact Information
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+1 800-775-5330
Denmark Location
+45 7560 2850
UK Location
+44 (0) 20 3284 1536
Spectralink Corporation
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United Kingdom
info@spectralink.com
infoemea@spectralink.com
infoemea@spectralink.com
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Contents
About This Guide .............................................................. 11
Who Should Read This Guide? .......................................................................................11
What’s New in This Guide ...............................................................................................11
Recommended Software Tools .......................................................................................12
Reading the Feature Parameter Tables ..........................................................................12
Product Support ..............................................................................................................12
Spectralink References ...................................................................................................13
Specific Documents ....................................................................................................13
White Papers ..............................................................................................................14
Conventions Used In This Document .............................................................................15
Icons ...........................................................................................................................15
Writing Conventions ....................................................................................................16
Part I: Getting Started ...................................... 17
Chapter 1: Welcome to the Spectralink 84-Series Handsets ......... 18
Key Features of your Spectralink Handsets ..................................................................20
Chapter 2: System Overview ................................................ 21
What is SIP? .....................................................................................................................21
Network Requirements ....................................................................................................21
Network Configuration ....................................................................................................22
Understanding Spectralink Phone Software Architecture ............................................23
What is the Updater? ..................................................................................................23
What is the Spectralink Software? ...............................................................................24
What are the configuration files? .................................................................................24
What are the resource files? .......................................................................................25
Part II: Setting Up Your Environment ................... 26
Chapter 3: Setting Up Your Device Network ............................. 27
Wireless Device Settings.................................................................................................27
IP Communication Settings ............................................................................................27
Provisioning Server Discovery .......................................................................................28
Supported provisioning protocols ................................................................................29
Network Configuration Menus ........................................................................................30
Network configuration menu ........................................................................................31
Provisioning server menu ............................................................................................32
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Spectralink 84-Series Series Wireless Telephones Administration Guide
Network Interfaces Menu ............................................................................................34
Syslog Menu ...............................................................................................................39
Login Credentials Menu ..............................................................................................40
TLS Security Menu...........................................................................................................41
TLS Profile Menu ........................................................................................................41
TLS Applications Menu ...............................................................................................41
Chapter 4: Setting Up the Provisioning Server.......................... 43
Why Use a Provisioning Server? ....................................................................................43
Provisioning Server Redundancy ................................................................................43
Provisioning Server Security Notes ...............................................................................44
Setting up an FTP Server as Your Provisioning Server ................................................44
Downloading Spectralink Software Files to the Provisioning Server ..........................46
Microsoft® Skype for Business compatibility ................................................................46
Spectralink 84-Series Hardware IDs............................................................................47
Deploying and Updating Spectralink Handsets with a Provisioning Server ................48
Shortcut Method to Deploy Spectralink Handsets with a Provisioning Server ..............49
Upgrading Spectralink Software ..................................................................................51
Chapter 5: Understanding the Files Written by the Handsets ........ 52
Log Files ...........................................................................................................................53
Overrides ..........................................................................................................................53
Deleting the override file..............................................................................................54
Contacts ...........................................................................................................................54
Call List.............................................................................................................................55
Part III: Configuring Features ............................. 58
Chapter 6: Features that Cannot be Configured ........................ 59
Audio Processing Features .............................................................................................59
Automatic Gain Control ...............................................................................................59
Background Noise Suppression ..................................................................................59
Comfort Noise Fill........................................................................................................59
Dynamic Noise Reduction ...........................................................................................59
Jitter Buffer and Packet Error Concealment ................................................................59
Low-Delay Audio Packet Transmission .......................................................................60
Call Timer .........................................................................................................................60
Called Party Identification ...............................................................................................60
Connected Party Identification .......................................................................................60
Microphone Mute .............................................................................................................60
Synthesized Call Progress Tones ...................................................................................61
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Spectralink 84-Series Series Wireless Telephones Administration Guide
Chapter 7: Configurable Features on the User Menus ................ 62
Call Forwarding................................................................................................................62
Keypad Lock ....................................................................................................................66
Multi Key Answer .............................................................................................................67
Notification Profiles .........................................................................................................67
Time and Date Display .....................................................................................................77
Synchronizing with SNTP ............................................................................................79
User Preferences Parameters .........................................................................................80
Chapter 8: Features Configured by the Administrator ................ 82
AutoComplete List ...........................................................................................................82
Audio Settings .................................................................................................................83
Context Sensitive Volume Control ...............................................................................83
<voice.volume/> ..........................................................................................................83
<voice/> ......................................................................................................................84
<rxQoS/>.....................................................................................................................84
Automatic Off-Hook Call Placement ...............................................................................85
Background Images.........................................................................................................86
Configuring Background Images .................................................................................86
Backlight Off while Phone is Charging ..........................................................................87
Battery End-of-life Alert ...................................................................................................87
Feature and Basic Settings Menu Password .................................................................88
Call Hold ...........................................................................................................................89
Call Handling Features ....................................................................................................91
Call Park and Retrieve ................................................................................................92
Call Waiting Alerts .......................................................................................................92
Calling Party Identification ...........................................................................................92
Missed Call Notification ...............................................................................................93
Call Transfer ...............................................................................................................93
Call Lists .....................................................................................................................95
Miscellaneous Call Handling Parameters ....................................................................95
CMS 2.0.............................................................................................................................97
Conference Calls .............................................................................................................97
Corporate Directory .........................................................................................................98
Default Ring Tones and Alert Tones.............................................................................101
Call Progress Patterns ..............................................................................................104
Do Not Disturb ...............................................................................................................106
Dual Tone Multi-Frequency (DTMF) Tones...................................................................106
DTMF Event RTP Payload ........................................................................................107
Emergency Calls ............................................................................................................108
Emergency Dial via Authorized Call menu.................................................................108
Emergency Dial via Duress Button ............................................................................109
Enhanced Feature Keys ................................................................................................111
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Guidelines for Configuring Enhanced Feature Keys ..................................................112
Understanding Macro Definitions ..............................................................................115
Macro Action .............................................................................................................115
Prompt Macro Substitution ........................................................................................117
Expanded Macros .....................................................................................................118
Special Characters ....................................................................................................118
Features Softkey Menu Options Customization ..........................................................120
Example Softkey Configurations ...............................................................................122
Handsfree Settings ........................................................................................................125
Bluetooth Headset Support .......................................................................................126
Language Support .........................................................................................................126
Local Contact Directory.................................................................................................129
Provisioning the Seed Directory ................................................................................130
Configuring the Contact Directory .............................................................................132
Editing the Users’ MACaddress-directory.xml File .....................................................133
Specialized Caller Treatments...................................................................................133
Location Services (Ekahau) ..........................................................................................135
Microsoft Exchange Calendar Integration....................................................................136
Open Application Interface ...........................................................................................138
Passwords – User and Administrator...........................................................................139
Personal Alarms ............................................................................................................141
Administrator Configurable Options ...........................................................................142
User Experience........................................................................................................144
Integration with Third Party Applications ...................................................................147
XML API Detail ..........................................................................................................151
Viewing an Alarm Event ............................................................................................151
Configuration Template .............................................................................................152
Phone Lock ....................................................................................................................153
Provisional Polling of Spectralink Handsets ...............................................................155
Push-to-talk and Group Paging ....................................................................................156
Push-to-talk ...............................................................................................................157
Group Paging ............................................................................................................159
Quick Barcode Connector Application.........................................................................161
Registrations ..................................................................................................................161
Multiple Registrations ................................................................................................164
Multiple Concurrent Calls ..........................................................................................165
Flexible Call Appearances .........................................................................................167
User Profiles...................................................................................................................167
Placing Authorized (Emergency) Calls without Logging In.........................................169
Voicemail Integration.....................................................................................................170
<voIpProt/> .....................................................................................................................172
Web Browser ..................................................................................................................178
<mb/>........................................................................................................................181
<oai/> ........................................................................................................................182
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Chapter 9: Web Application Parameters................................ 183
Application menu configuration <apps.> .....................................................................183
Web browser parameters <mb.> ...................................................................................184
State Polling Parameters <apps.statePolling.> ...........................................................185
Push Request Parameters <apps.push.> .....................................................................185
Telephony Notification Parameters (apps.telNotification.> ........................................187
Open Application Interface parameters <oai.> ............................................................189
Sample Configurations ..................................................................................................189
Push..........................................................................................................................189
Telephony Notifications .............................................................................................190
State Polling ..............................................................................................................190
Personal Alarms ........................................................................................................191
Chapter 10: System-Level Parameters .................................. 192
Configuration File Encryption .......................................................................................192
Understanding Digital Certificates ...............................................................................193
About Digital Certificates ...........................................................................................195
Types of certificates ..................................................................................................195
Configuring certificates ..............................................................................................196
Generating a Certificate Signing Request .................................................................207
Downloading Certificates to a Spectralink Phone ......................................................208
DNS SIP Server Name Resolution ................................................................................208
Behavior When the Primary Server Connection Fails ................................................209
Incoming Signaling Validation ......................................................................................211
Instant Messaging..........................................................................................................212
IP Type-of-Service..........................................................................................................215
<qos/> .......................................................................................................................215
Logging Parameters ......................................................................................................216
<level/> <change/>and<render/> ..............................................................................217
<sched/> ...................................................................................................................218
Microsoft Skype for Business Integration....................................................................219
Network Address Translation (NAT) .............................................................................219
Provisioning Server System Settings...........................................................................220
<request/> ......................................................................................................................221
Security <sec/> ..............................................................................................................221
<srtp/>.......................................................................................................................221
<dot1x><eapollogoff/>...............................................................................................222
Secure Real-Time Transport Protocol ..........................................................................222
Server Redundancy .......................................................................................................225
Terminology ..............................................................................................................225
About the Optional Failover Behaviors ......................................................................226
Fallback Deployments ...............................................................................................229
Failover Deployments ...............................................................................................230
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DNS Server Unavailability .........................................................................................230
Redundancy Parameters...........................................................................................230
Supporting 802.1X Authentication ................................................................................233
<tcpIpApp/> ....................................................................................................................235
<dhcp/> .....................................................................................................................235
<dns/> .......................................................................................................................235
<ice/> ........................................................................................................................235
<keepalive/>..............................................................................................................236
Tones <tones/> ..............................................................................................................237
<chord/> ....................................................................................................................237
Web Configuration Utility ..............................................................................................238
<httpd/> .....................................................................................................................238
Chapter 11: Special Use Cases ........................................... 239
Acoustic Echo Cancellation ..........................................................................................239
Audio Codecs.................................................................................................................239
Band Steering ................................................................................................................240
Bridged Line Appearance..............................................................................................242
Local Digit Map ..............................................................................................................243
Understanding Digit Map Rules .................................................................................244
Location Values for E.911 Services ..............................................................................249
Real-Time Transport Protocol Ports .............................................................................250
Shared Line Appearances .............................................................................................251
Shared Call Appearance Signaling ............................................................................252
Static DNS Cache ...........................................................................................................253
Using Static DNS Cache for Redundancy .................................................................258
DNS Cache <dns/> .........................................................................................................259
NAPTR <NAPTR/>....................................................................................................259
SRV <SRV/> .............................................................................................................260
A <A/> .......................................................................................................................260
Voice Activity Detection ................................................................................................264
Part IV: Troubleshooting and Maintaining your
Deployment ................................................. 265
Chapter 12: Troubleshooting Your Spectralink Handsets .......... 266
Troubleshooting Flow Diagram ....................................................................................267
Understanding Error Message Types ...........................................................................268
Updater Error Messages ...........................................................................................268
Spectralink Software Error Messages .......................................................................270
Status Menu ...................................................................................................................275
Log Files .........................................................................................................................276
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Logging Modules .......................................................................................................278
Major categories of WLAN entries .............................................................................278
Managing the Phone’s Memory Resources .................................................................280
Identifying Symptoms ................................................................................................280
Checking the Phone’s Available Memory ..................................................................281
Managing the Phone Features ..................................................................................282
Testing Phone Hardware ...............................................................................................283
Uploading a Phone’s Configuration .............................................................................284
Network Diagnostics .....................................................................................................284
Network Protocols and Ports Used on Spectralink Handsets ....................................285
Power and Startup Issues .............................................................................................286
Key Pad Issues ..............................................................................................................286
Screen and System Access Issues ..............................................................................287
Calling Issues.................................................................................................................287
Display Issues ................................................................................................................288
Audio Issues ..................................................................................................................288
Upgrading Issues ...........................................................................................................289
Chapter 13: Miscellaneous Maintenance Tasks ....................... 291
Encrypting Configuration Files .....................................................................................291
Comparing encrypted and unencrypted files .............................................................295
Decrypting existing configuration files .......................................................................295
Changing an existing key ..........................................................................................296
Log messages ...........................................................................................................296
Multiple Key Combinations ...........................................................................................297
Rebooting the Phone ................................................................................................297
Resetting to factory defaults ......................................................................................298
Updating log files.......................................................................................................298
Setting base profile ...................................................................................................298
Default Feature Key Layouts .........................................................................................299
Parsing Vendor ID Information .....................................................................................300
Product Model Number and Hardware ID Mapping .....................................................301
Capturing the Phone’s Current Screen ........................................................................302
Part V: Appendices ........................................ 303
Appendix A: Ringtone Pattern Names and Sound Effects Parameters
.................................................................................. 304
Ringer Patterns ..............................................................................................................304
Ring Tones <rt/> ............................................................................................................305
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Appendix B: Session Initiation Protocol (SIP) Information ......... 307
RFC and Internet Draft Support ....................................................................................307
Request Support .......................................................................................................309
Header Support .........................................................................................................309
Response Support ....................................................................................................312
Hold Implementation .................................................................................................314
Reliability of Provisional Responses ..........................................................................315
Transfer ....................................................................................................................315
Third party call control ...............................................................................................315
SIP for Instant Messaging and Presence ..................................................................315
Appendix C: Open Source Information ................................. 316
OFFER for Source for GPL and LGPL Software ..........................................................316
Contact Information for Requesting Source Code .....................................................316
Appendix D: Library of <device/> Settings ............................. 317
Appendix E: Trusted Certificate Authority List........................ 324
Appendix F: Spectralink Certificates ................................... 335
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About This Guide
This Spectralink 84-Series Administration Guide provides advanced instructions for installing,
provisioning, and administering Spectralink handsets. It is a companion to the Spectralink
84-Series Deployment Guide which is your essential reference for understanding how to
provision and deploy Spectralink 84-Series handsets in any environment. This guide expands
upon the information provided in the Deployment Guide and provides additional data about how
the software works and provides descriptions of all applicable parameters. Specifically, this
Administration Guide will help you perform the following tasks:

Install and configure your handset on a network server or Web server

Configure your handset’s features and functions

Configure your handset’s user settings

Troubleshoot common handset issues
Who Should Read This Guide?
System administrators and network engineers should read this guide for advanced information
on configuring and understanding Spectralink 84-Series handsets. This guide describes
administration-level tasks and is not intended for end users.
Before reading this guide, you should be familiar with the following:

The information in the Spectralink 84-Series Deployment Guide is not duplicated in this
document. This document expands upon the basic configuration settings in the
Deployment Guide and this document assumes you are familiar with Deployment Guide
information.

Computer networking and driver administration for your operating system

An XML editor

The XML-based configuration file format that the Spectralink Software and its supported
handsets use.
What’s New in This Guide
The content in this guide has been significantly revised from the Polycom UCS version for use
with the Spectralink 84-Series handsets. It is designed for clarity and to provide more
information to system administrators who are already familiar with deploying Spectralink
84-Series handsets.
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Spectralink 84-Series Series Wireless Telephones Administration Guide
Recommended Software Tools
Spectralink recommends that you use an XML editor – such as XML Notepad – to create and
edit configuration files. In this way, all configuration files that you create will be valid XML files.
If the configuration files are not valid XML, they will not load on the handset and an error
message will be logged to the provisioning server.
See the Spectralink 84-Series Deployment Guide for a discussion on XML editor options,
usefulness and limitations.
Reading the Feature Parameter Tables
Each of the feature descriptions discussed in Part III: Configuring Features includes a table of
parameters that you configure to make the features work. Although there are three provisioning
methods you can use to configure a feature: a centralized provisioning server, the Web
Configuration Utility, or the local handset user interface, this document emphasizes the central
provisioning server method. It is the preferred method for deploying advanced configurations
such as those covered in this document as it is the only method that is available for every
feature. The Web Configuration Utility and the local handset user interface do not provide
access to all features.
The central provisioning server method requires you to configure parameters located in
template configuration files that Spectralink provides in XML format.
We recommend using the search feature of your XML editor to locate the parameters you need
to find.
Product Support
Spectralink wants you to have a successful installation. If you have questions please contact the
Customer Support Hotline at 1-800-775-5330.
The hotline is open Monday through Friday, 6 a.m. to 6 p.m. Mountain time.
For Technical Support: mailto:technicalsupport@spectralink.com
For Knowledge Base: http://support.spectralink.com
For Return Material Authorization: mailto:nalarma@spectralink.com
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Spectralink 84-Series Series Wireless Telephones Administration Guide
Spectralink References
All Spectralink documents are available at http://support.spectralink.com.
To go to a specific product page:
Select the Product Category and Product Type from the dropdown lists and then select the
product from the next page. All resources for that particular product are displayed by default
under the All tab. Documents, downloads and other resources are sorted by the date they were
created so the most recently created resource is at the top of the list. You can further sort the
list by the tabs across the top of the list to find exactly what you are looking for. Click the title to
open the link.
Specific Documents
The Spectralink 84-Series Wireless Telephone Administration Guide provides a comprehensive
list of every parameter available on Spectralink 84-Series Wireless Telephones.
Spectralink 84-Series Wireless Telephone Deployment Guide This document introduces
deployment concepts and the methods of provisioning the 84-Series handsets in any type of
facility. It is the fundamental text and a prerequisite to this Administration Guide, especially for
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Spectralink 84-Series Series Wireless Telephones Administration Guide
administrators who are new to the Spectralink 84-Series handsets or who may wish a refresher
course.
Quick Network Connect Administration Guide QNC provides step-by-step instructions for
configuring wireless settings required for the smartphones to associate with the wireless LAN.
QNC software and documents are available on the Spectralink support site at
http://support.spectralink.com/products/wi-fi/qnc.
The Spectralink 84-Series Barcode Administration Guide provides information about barcode
symbologies and how to configure and implement the barcode feature on the handset. The
Spectralink 84-Series User Guide also contains information about using the barcode feature.
Quick Barcode Connector Administration Guide Provides instruction for implementation of the
barcode application. The Spectralink 84-Series User Guide contains information about using the
barcode feature.
The Spectralink 84-Series User Guide offers comprehensive instructions on using each of the
features deployed on the handsets.
The Web Configuration Utility User Guide is used for troubleshooting in certain isolated cases
as explained in the text.
For information about combining Polycom desksets and Spectralink 84-Series handsets in the
same facility, see the Interoperability Guide: Spectralink 84-Series Coexistence with Polycom
Desksets.
For additional information about deploying Microsoft Skype for Business (formerly Lync) in your
phone environment, see Microsoft Skype for Business Interoperability Guide.
For information on IP PBX and softswitch vendors, see the Spectralink 84-Series Call Server
Interoperability Guide.
Technical Bulletins and Feature Descriptions explain workarounds to existing issues and
provide expanded descriptions and examples.
AP Configuration Guides explain how to correctly configure access points and WLAN controllers
(if applicable) and identify the optimal settings that support Spectralink 84-Series handsets. You
can find them on the VIEW Certified webpage.
Release Notes describe the new and changed features, and resolved issues in the latest
version of the software. Find them in the Downloads section of the support site.
Spectralink 84-Series Wireless Telephones Web Developer’s Guide assists with the
development of applications that run on the browser on the Spectralink 84-Series Wireless
Handsets.
For other references, look for the Web Info icon
throughout this Administration Guide.
White Papers
Spectralink White Papers are available at http://www.spectralink.com/resources/white-papers.
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Spectralink 84-Series Series Wireless Telephones Administration Guide
For the Spectralink 84-Series Wireless Telephones, please refer to Best Practices Guide for
Deploying Spectralink 84-Series Handsets for detailed information on wireless LAN layout,
network infrastructure, QoS, security and subnets.
For additional details on RF deployment please see The challenges of ensuring excellent voice
quality in a Wi-Fi workplace and Deploying Enterprise-Grade Wi-Fi Telephony.
These White Papers identify issues and solutions based on Spectralink’s extensive experience
in enterprise-class Wi-Fi telephony. It provides recommendations for ensuring that a network
environment is adequately optimized for use with Spectralink Wireless Telephones.
Conventions Used In This Document
Icons
Icons indicate extra information about nearby text.
Warning
The Warning icon highlights an action you must perform (or avoid) to avoid
exposing yourself or others to hazardous conditions.
Caution
The Caution icon highlights information you need to know to avoid a hazard that
could potentially impact device performance, application functionality, successful
feature configuration and/or affect handset or network performance.
Note
The Note icon highlights information of interest or important information that will
help you be successful in accomplishing a procedure or understanding a concept.
Tip
The Tip icon highlights information that may be valuable or helpful for users to
know, such as special techniques, shortcut methods, or information that will make
user tasks easier to perform.
Web
The Web Info icon highlights supplementary information available online such as
documents or downloads on support.spectralink.com or other locations.
Timesaver
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Spectralink 84-Series Series Wireless Telephones Administration Guide
A time-saving tip is typically used to mention or highlight a faster or alternative
method for users who may already be familiar with the operation or method being
discussed.
Admin Tip
This tip advises the administrator of a smarter, more productive or alternative
method of performing an administrator-level task or procedure.
Power User
A Power User Tip is typically reserved for information directed specifically at highlevel users who are familiar with the information or procedure being discussed and
are looking for better or more efficient ways of performing the task. For example,
this might highlight customization of a feature for a specific purpose.
Troubleshooting
This element can be used in any type of document and is typically used to highlight
information to help you solve a relevant problem you may encounter, or to point to
other relevant troubleshooting reference information.
Settings
The Settings icon highlights information to help you zero in on settings you need to
choose for a specific behavior, to enable a specific feature, or access
customization options.
Writing Conventions
Convention
Description
<MACaddress>
Indicates that you must enter information specific to your installation, phone, or
network. For example, when you see <MACaddress>, enter your phone’s 12-digit
MAC address. If you see <installed-directory>, enter the path to your installation
directory.
>
Indicates menu navigation. For example, Settings> Basic indicates that you need
to select Basic from the Settings menu.
parameter.*
Used for configuration parameters. If you see a parameter name in the form
parameter.* , the text is referring to all parameters beginning with parameter.
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Part I: Getting Started
Part I gives you an overview of the Spectralink 84-Series handsets and of the Spectralink
Software.
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Chapter 1: Welcome to the Spectralink
84-Series Handsets
This chapter introduces Spectralink 84-Series handsets used with Spectralink Software version
4.2.0 and above.
The Spectralink family of handsets provides a powerful, yet flexible wireless IP communications
solution for Ethernet TCP/IP networks. Not only do the handsets deliver excellent voice quality,
but also come with a high-resolution graphic display screen for call information, multiple
languages, directory access, and system status. The handsets can also support advanced
functionality, including multiple call and flexible line appearances, HTTPS secure provisioning,
presence, custom ringtones, and local conferencing.
Note: Indoor use only
This device is intended for indoor use only.
Caution: Product compatibility/safety
Spectralink 84-Series handsets are intended for operation only with Spectralink 84Series battery packs and Spectralink 84-Series chargers. These Spectralink
components are critical to product safety certification and may not be substituted.
Representative samples of the Spectralink 84-Series handsets, battery packs and
chargers have been tested as a complete system by an independent testing
organization and have been certified by that organization to meet applicable safety
standards. Use or operation of the Spectralink 84-Series handsets with batteries or
chargers other than those authorized by Spectralink has not been tested or safety
certified. Spectralink 84-Series handsets, battery packs or chargers used or
operated with products not authorized by Spectralink are not covered by the
Spectralink Limited Product Warranty.
Caution: Use authorized components only
Only Spectralink 84-Series battery packs and Spectralink 84-Series chargers are
authorized for use or operation with Spectralink 84-Series handsets.
Only Spectralink 84-Series battery packs are authorized for use or operation with a
Spectralink 84-Series charger and are not authorized to be used or operated in any
other charger.
Spectralink 84-Series Wireless Telephones were originally developed in conjunction with
Polycom Inc. In September 2012 the Spectralink 84-Series Wireless Telephone software code
and Polycom UCS deskset software code were split into two separate streams. The same code
no longer serves both Spectralink Wireless Telephones and Polycom wired desksets.
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Using Spectralink Handsets and Polycom Desksets in a combined
environment
Special configuration steps need to be taken in environments where both
Spectralink and Polycom phones are deployed. For more information on using
Spectralink 84-Series handsets and Polycom wired desksets in a facility see the
Interoperability Guide: Spectralink 84-Series Wireless Telephones and Polycom
Desksets.
From an administrator’s perspective, the handsets are endpoints in an overall network topology
designed to interoperate with other compatible equipment including application servers, media
servers, internet-working gateways, voice bridges, and other end points.
If you want to begin setting up your Spectralink handsets on the network, go to Setting Up Your
Device Network.
If you want to begin configuring the features available for your Spectralink handsets, go to Part
III: Configuring Features.
Support for Spectralink Handsets
You can find all documentation for all Spectralink handsets on Spectralink Support
Website. For more information, contact your Spectralink distributer.
Spectralink 8440/8441
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Spectralink 84-Series Series Wireless Telephones Administration Guide
Key Features of your Spectralink Handsets
Spectralink handsets running Spectralink Software include the following key features:



Award winning sound quality with a full-duplex speakerphone
○
Permits natural, high-quality, two-way conversations
○
Supports HDVoice
Easy-to-use
○
An easy transition from traditional PBX systems into the world of IP Communications
○
Four context-sensitive softkeys for further menu-driven activities
Platform independent
○

Faster Boot Time
○



○
Upgrade handsets as standards develop and protocols evolve
○
Extends the life of the handset to protect your investment
○
Application flexibility for call management and new telephony applications
Large LCD
○
Easy-to-use, easily readable, and intuitive interface
○
Support of rich application content, including multiple call appearances, presence
and instant messaging, and XML services
○
240 x 320 pixel graphical color LCD
Multiple language support
Set on-screen language to your preference. Select from Chinese (Simplified and
Traditional), Danish, Dutch, English (Canada, United Kingdom, and United States),
French, German, Italian, Japanese, Korean, Norwegian, Polish, Portuguese
(Brazilian), Russian, Slovenian, Spanish (International), and Swedish.
Web Browser
○

The time between handset reboot and obtaining a dial tone has been noticeably
reduced.
Field upgradeable
○

Supports multiple protocols and platforms enabling standardization of one handset
for multiple locations, systems, and vendors
Supports a subset of XHTML constructs that run like any other Web browser
XML status/control API
○
Ability to poll handsets for call status and device information
○
Ability to receive telephony notification events
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Chapter 2: System Overview
This chapter provides an overview of the Spectralink Software, providing an understanding of
how the handsets are deployed within the greater LAN and wireless LAN configuration. To
begin setting up your Spectralink handsets, refer to the Spectralink 84-Series Deployment
Guide and review the Infrastructure chapter.
The Spectralink handsets are deployed in an 802.1X wireless environment.
Deploying Spectralink Handsets in a Completely Wireless Environment
For more information on using these handsets in a completely wireless
environment, see the Spectralink 84-Series Wireless Telephone Deployment
Guide.
What is SIP?
The Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for
multimedia communications over IP. It is an ASCII-based, application-layer control protocol
(defined in RFC 3261) that can be used to establish, maintain, and terminate calls between two
or more endpoints. Like other voice over IP (VoIP) protocols, SIP is designed to address the
functions of signaling and session management within a packet telephony network. Signaling
allows call information to be carried across network boundaries. Session management provides
the ability to control the attributes of an end-to-end call.
Network Requirements
For Spectralink handsets to successfully operate as a SIP endpoint in your network, you will
require:

A working IP network

Routers configured for VoIP

VoIP gateways configured for SIP

The latest (or a compatible version) Spectralink Software image

An active, configured call server to receive and send SIP messages
For information on IP PBX and softswitch vendors, see the Spectralink 84-Series Call
Server Interoperability Guide.
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Spectralink 84-Series Series Wireless Telephones Administration Guide
Network Configuration
Many desktop phones connect physically through a Category 5 (Cat-5) cable to a standard
office twisted-pair (IEEE 802.3) 10/100/1000 megabits per second Ethernet LAN. Spectralink
wireless handsets, however, connect to a WLAN to send and receive all data using
802.11a/b/g/n technology to access telephony resources on the wired LAN.
There are many ways to set up a phone network using Spectralink wireless handsets and the
diagram shown next is just one example of a network setup.
Servers:
Time Server
LDAP Server
Radius Server
DHCP Server
Exchange Server
Application Server
OCS or Lync Server
Provisioning Server
Ethernet switch
Wi-Fi AP
WLAN Controller
Access point
area
SIP Call Server
QBC-enabled
computer
Wi-Fi AP
PSTN
8450/8452
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Spectralink 84-Series Series Wireless Telephones Administration Guide
Understanding Spectralink Phone Software
Architecture
The Spectralink handset software is made of four basic components:

Updater The software that loads & runs first when the handset is powered on

Spectralink Software
features

Configuration files The files that contain the handset’s parameter settings

Resource files Optional files that contain settings for advanced features
The software that implements the handset functions and
Figure 2-2: Spectralink phone software
What is the Updater?
The Updater is a small application that resides in the flash memory on the handset. The Updater
is installed at the factory and is already installed on your new Spectralink handsets.
When you start/boot/reboot the handset, the Updater performs the following tasks:
1
Enables you to open the setup menu so you can set various network and provisioning
options.
The Updater requests IP settings and accesses the provisioning server (also called the
boot server) to look for any changes to the Updater software.
If updates are found, they are downloaded and saved to flash memory, which overwrites
itself after verifying the integrity of the download.
2
If new updates are downloaded, the Updater formats the file system, removes any
application software and configuration files that were present.
3
Downloads the master configuration file.
The Updater and the application use this file to acquire a list of other files that the
handset needs.
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4
Examines the master configuration file for the name of the application file, and then
looks for this file on the provisioning server.
If the copy on the provisioning server is different than the one stored in device settings,
or there is no file stored in flash memory, the application file is downloaded.
5
Extracts the Spectralink Software from flash memory.
6
Installs the application into RAM, then uploads an event log file from the boot cycle.
The Updater will then terminate, and the Spectralink Software will take over.
What is the Spectralink Software?
The Spectralink Software manages the protocol stack, the digital signal processor (DSP), the
user interface, and the network interaction. The Spectralink Software implements the following
functions and features on the handsets:

VoIP signaling for a wide range of voice telephony functions using SIP signaling for call
setup and control

Industry standard security techniques for ensuring that all provisioning, signaling, and
media transactions are robustly authenticated and encrypted across the WLAN

Advanced audio signal processing for handset, headset, and speakerphone
communications using a wide range of audio codecs

Flexible provisioning methods to support single handset, small business, and large multisite enterprise deployments
The software is a binary file image and contains a digital signature that prevents tampering or
the loading of rogue software images.
Each release of software includes a new image file.
Both the Updater and Spectralink Software run on all 84-Series handset models that Spectralink
currently supports.
What are the configuration files?
The Spectralink Software that you download contains configuration file templates, valid XML
files that you can edit using an XML editor. These files contain all the parameters explained in
this document that provision the handsets with features and settings. The configuration files are
very flexible: you can rearrange the parameters within the file, move parameters to new files, or
create your own configuration files with only those parameters you want. This flexibility is useful
when you want to apply the same features and settings to a large number of handsets. Use of
the configuration files to provision the handsets with features and settings is called the
centralized provision method – the configuration files enable you to store a single set of
configuration files on a central provisioning server and configure all of your handsets to read the
same set of files. You can also configure a subset of handsets to use only specific files, thereby
deploying different handsets with different sets of features.
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Spectralink recommends that you configure handsets using the centralized provisioning method.
However, you can also configure individual handsets using the handset’s menu system,
accessible through the local user keypad interface, or you can configure select parameters by
using the Web Configuration Utility.
You will need to keep in mind that there is a hierarchy among the configuration methods and
settings. Using a higher-priority method will override settings you make using a lower-priority
method. The following lists all of the available ways to set features and settings for the
handsets. Spectralink strongly recommends becoming familiar with each of the configuration
methods.
Override files are maintained on the central provisioning server. See Understanding the Files
Written by the Handsets for additional information on how precedence is used by the handsets.
Configuration Methods
You can make changes to the handset’s configuration using any of the following configuration methods. Take note
that there is a precedence order among the configuration methods: changes made to settings using a higherpriority method override settings made using a lower-priority method. Configuration changes are uploaded to the
handset as override files that remain active until you remove them or reset to default.
The precedence order for configuration parameter changes is as follows (highest to lowest priority):
Local handset user interface
Web Configuration Utility
Central Provisioning Server
Default values (if Null then the value will be obtained from a higher priority method.)
Each of these configuration methods is detailed in Part III: Configuring Features.
What are the resource files?
In addition to the software and configuration files, the handsets may require resource files in
order to use some of the advanced features.
Examples of resource files include:

Language dictionaries

Ringtones

Contact directories

Custom backgrounds
If you need to remove resource files from a handset at a later date - for example, if you are
giving the handset to a new user - you will have to apply factory default settings to that handset.
For instructions on how to reset your handset to factory default settings, see the Spectralink
84-Series Deployment Guide.
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Part II: Setting Up Your
Environment
Part II provides you with essential information on how to set up your handset network and
provisioning server, and on the configuration methods you can use to set up handset features.
You will find basic and advanced instructions on how to set up a provisioning server, how to
deploy the Spectralink handsets from the provisioning server, and how to upgrade the software.
Part II consists of the following chapters:

Chapter 3: Setting Up Your Device Network

Chapter 4: Setting Up the Provisioning Server

Chapter 5: Understanding the Files Written by the Handsets
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Chapter 3: Setting Up Your Device
Network
The Spectralink 84-Series Wireless Handsets operate on a Wi-Fi LAN (WLAN). Local area
network design varies by organization and Spectralink handsets can be configured to
accommodate a number of network designs. This chapter shows you several automated and
manual ways to configure Spectralink handsets to operate in a LAN.
See the Spectralink 84-Series Deployment Guide for detailed information about how the
handset authenticates and associates with the WLAN.
Once the provisioning server discovery is complete the handset will initiate the provisioning
process described in Chapter 4: Setting Up the Provisioning Server.
Wireless Device Settings
You must configure wireless devices before they can establish a connection to a wireless
network. You can configure wireless devices manually, but it is more common to configure them
prior to deployment using the USB interface (USBNet) to the device (and the device.set
parameters in the configuration file). See the Spectralink 84-Series Wireless Telephone
Deployment Guide for full information about using the USB interface and basic wireless settings.
More advanced wireless settings that may need to be set up to connect your device to the
Wireless LAN (WLAN) are located in the Wi-Fi Menu section.
IP Communication Settings
When the handset has established network connectivity it needs to acquire several IP network
settings to proceed with provisioning. These settings are typically obtained automatically from a
DHCP server.
Tip: Novice administrator?
Read this section if you are new to this process or have never set up a provisioning
server before.
You have the option to set the IP communication settings manually from the handset UI, or to
pre-provision using a device.set capability.
When making the DHCP request the handset will include information in Option 60 that can
assist the DHCP server in delivering the appropriate settings to the device.
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Spectralink 84-Series Series Wireless Telephones Administration Guide
Timesaver: Reducing repetitive data entry
Spectralink recommends using DHCP where possible to eliminate repetitive
manual data entry.
The following table details the settings that are supported through the DHCP menu:
Option
SIP Parameter
Meaning
1
NA
Subnet mask
3
NA
Default gateway
6
DNSSRVR
DNS server
7
LOGSRVR
Syslog server logging
15
DOMAIN
Domain name
42
SNTPSRVR
NTP Server
43
sec.TLS.customCaCert.x
Auto discovery of the root CA certificate. If this setting is unavailable,
set the parameter per this guide.
66
TFTPSRVR
TFTP server
Web Info: RFC information on DHCP options
For more information on DHCP options, see RFC 2131 and RFC 2132.
Settings: Overriding the SNTP values set by DHCP
The configuration file value for SNTP server address and SNTP GMT offset can
be configured to override the DHCP value. See
tcpIpApp.sntp.address.overrideDHCP.
If you do not have control of your DHCP server or do not have the ability to set the DHCP
options, you will need to enable the handset to automatically discover the provisioning server
address. One way is to connect to a secondary DHCP server that responds to DHCP INFORM
queries with a requested provisioning server value. For more information, see RFC 3361 and
RFC 3925.
Provisioning Server Discovery
After the handset has established network connectivity it proceeds to the Configuration stage. In
this stage the following steps are carried out:

Software Update

Application of configuration settings relevant to a customer network
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Tip: Novice Administrator?
Read this section if you are new to this process or have never set up a provisioning
server before.
In many deployments a centralized provisioning server is used for the software update and
configuration functions. The handset supports several methods to ‘discover’ this provisioning
server:

Static You can manually configure the server address from the handset's user
interface or the Web Configuration Utility, or you can pre-provision the handset with an
initial provisioning server. The parameters are:
device.prov.serverName.set="1" and device.prov.serverName="" in a
configuration file.

DHCP DHCP option 66 is used to provide the address or URL of the provisioning
server.

DHCP INFORM The handset makes an explicit request for a DHCP option (which can
be answered by a server that is not the primary DHCP server). For more information,
see RFC 3361 and RFC 3925.
To change these parameters, go to Provisioning Server Menu.
Supported provisioning protocols
The Updater performs the provisioning functions of uploading log files, master configuration
files, software updates, and device setting menu changes.
By default, handsets are shipped with FTP enabled as the provisioning protocol. You can
change the provisioning protocol by updating the Server Type option. Or, you can specify a
transfer protocol in the Server Address, for example, http://usr:pwd@server (see Provisioning
Server Menu). The Server Address can be an IP address, domain string name, or URL. It can
be obtained through DHCP.
Supported protocols include: ftp, ftps, tftp, http and https.
In some cases a config file might need to be secured. You can use unique credentials to
connect to a server and include the transfer protocol in the configuration file name. For example,
https://usr:pwd@server/dir/file.cfg.
If a user name and password are specified as part of the server address or file name, they will
be used only if the server supports them. If a user name and password are required but not
specified, the device settings are sent to the server.
Settings: Choosing a valid URL
A URL should contain forward slashes (not back slashes) and should not contain
spaces. Escape characters are not supported. If a user name and password are
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not specified, the Server User and Server Password from device settings will be
used (see Provisioning Server Menu).
Note: Active and passive FTP methods
There are two types of FTP methods - active and passive. Spectralink Software is
not compatible with active FTP.
To guarantee software integrity, the Updater will download only cryptographically signed
Updater or Spectralink Software images. For HTTPS, widely recognized certificate authorities
are trusted by the handset and custom certificates can be added to the handset.
Web Info: Viewing trusted certificate authorities
For more information, see Appendix E: Trusted Certificate Authority List and
Technical Bulletin CS-13-06: Using custom certificates with Spectralink 8400
handsets.
Digest Authentication for Microsoft Internet Information Services (IIS)
If you want to use digest authentication against the Microsoft Internet Information Services server:




Use Microsoft Internet Information Server 6.0 or later.
Digest authentication needs the user name and password to be saved in reversible encryption.
The user account on the server must have administrative privileges.
The wildcard must be set as MIME type; otherwise, the handset will not download *.cfg, *.ld and other
required files. This is because the Microsoft Internet Information Server cannot recognize these extensions
and will return a “File not found” error. To configure wildcard for MIME type, see
http://support.microsoft.com/kb/326965.
For more information, see
http://www.microsoft.com/technet/prodtechnol/WindowsServer2003/Library/IIS/809552a3-3473-48a79683-c6df0cdfda21.mspx?mfr=true.
Network Configuration Menus
You can update the network configuration parameters in two ways:

During the Updater Phase. The setup menu is accessible during the auto-boot
countdown of the Updater phase of operation. While your handset boots up, press the
Cancel softkey, and press the Setup softkey to launch the setup menu. To access the
setup menu, you will have to enter the administrator’s password.

After your handset starts and is running Spectralink Software. The network
configuration menu is accessible from the handset’s main menu. Select Settings>
Advanced Settings> [enter password]> Administration Settings> Network
Configuration. To access the Advanced Settings menu, you will have to enter the
administrator’s password which is 456 by default.
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Admin: Changing the default administrator password
Spectralink recommends that you change the default administrative password. See
Passwords – User and Administrator.
You have the option of modifying the handset network configuration parameters in the following
menus and sub-menus:

Main Menu

Provisioning Server Menu

Network Interfaces Menu (Ethernet Menu)

TLS Menu

Syslog Menu
Use the softkeys, the arrow keys, and the Select and Delete keys to make changes.
Certain parameters are read-only due to the value of other parameters. For example, if the
DHCP client parameter is enabled, the Phone IP Address and Subnet Mask parameters are
grayed out or not visible since the DHCP server automatically supplies these parameters and
the statically assigned IP address and subnet mask will never be used in this configuration.
Settings: Resetting network configurations
The basic network configuration referred to in the subsequent sections can be reset
to factory default settings using the handset’s main menu: Select Settings>
Advanced Settings> [enter password]> Administration Settings> Reset to
Defaults> Reset Device Settings. Or use a multiple key combination, as
described in Multiple Key Combinations.
Network configuration menu
You can modify the following configuration parameters from the setup menu while the handset
boots, or from the Administrative Settings menu from a handset running Spectralink Software:
Name
Possible Values
Provisioning server menu
See Provisioning Server Menu.
Network interfaces menu or Ethernet menu
See Network Interfaces Menu (Ethernet Menu).
SNTP address
Dotted-decimal IP address OR Domain name string
The Simple Network Time Protocol (SNTP) server the handset obtains the current time from.
GMT offset
-13 through +12
The offset of the local time zone from Greenwich Mean Time (GMT) in half hour increments.
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Name
Possible Values
DNS server
Dotted-decimal IP address
The primary server the handset directs Domain Name System (DNS) queries to.
DNS INFORM server
Dotted-decimal IP address
The secondary server to which the handset directs DNS queries.
DNS domain
Domain name string
The handset’s DNS domain.
Hostname
hostname
The DHCP client hostname.
Syslog menu
See Syslog Menu.
Base profile
Generic, Lync*
* Select Lync if using Microsoft Server 2010 or 2013 or Skype for Business.
Provisioning server menu
The following configuration parameters can be modified on the Provisioning Server Menu.
Settings> Advanced Settings> [enter password]> Administration Settings> Network
Configuration> Provisioning Server:
Name
Possible Values
DHCP menu
See DHCP Menu. Note: This menu is disabled when the DHCP client is disabled.
Server type
0=FTP, 1=TFTP, 2=HTTP, 3=HTTPS, 4=FTPS
The protocol that the handset will use to obtain configuration and handset application files from the provisioning
server. See Supported Provisioning Protocols.
Note: Active FTP is not supported for BootROM version 3.0 or later. Passive FTP is supported. Only implicit FTPS
is supported.
Server address
Dotted-decimal IP address OR URL
Domain name string or a URL. All addresses can be followed by an optional directory. The address can also be
followed by the file name of a .cfg master configuration file, which the handset will use instead of the default
<MACaddress>.cfg file.
The provisioning server to use if the DHCP client is disabled, if the DHCP server does not send a boot server
option, or if the Boot Server parameter is set to Static.
The handset can contact multiple IP addresses per DNS name. These redundant provisioning servers must all use
the same protocol.
If a URL is used, it can include a user name and password. See Supported Provisioning Protocols. For information
on how to specify a directory and use the master configuration file, see Spectralink 84-Series Wireless Telephone
Deployment Guide.
Note: ":", "@", or "/" can be used in the user name or password if they are correctly escaped using the method
specified in RFC 1738.
Server user
String
The user name requested when the handset logs into the server (if required) for the selected Server type.
Note: If the Server address is a URL with a user name, this will be ignored.
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Name
Possible Values
Server password
String
The default password is 456.
File Transmit Tries
1 to 10 Default 3
The maximum number of attempts to transfer a file. (An attempt is defined as trying to download the file from all IP
addresses that map to a particular domain name.)
Retry Wait
0 to 300 seconds Default 1
The minimum amount of time that must elapse before retrying a file transfer. The time is measured from the start
of a transfer attempt, which is defined as the set of upload/download transactions made with the IP addresses that
map to a given provisioning server’s DNS. If the set of transactions in an attempt is equal to or greater than the
Retry Wait value, then there will be no further delay before the next attempt is started.
Tag SN to UA
Disabled, Enabled
If enabled, the handset’s serial number (MAC address) is included in the User-Agent header of HTTP/HTTPS
transfers and communications to the browser.
The default value is Disabled.
Upgrade Server
Non-editable string (auto-populated by the Web
Configuration Utility)
The Upgrade server is an alternate way of getting software updates into the handset through the Web
Configuration Utility. It is a completely different process than using a provisioning server method.
When a value is displayed in this field, it is the address/URL that has been accessed for software updates through
the Web Configuration Utility. This value is also stored in the handset’s override file on the provisioning server.
If this field in the handset menu is populated then you cannot get code onto the phone from any other method than
using the WebUI upgrade method because handset settings have highest precedence and this setting is basically
a mirror of the override file. If you want to download code into a particular handset using a provisioning server,
clear the value set by the WebUI, or edit the override file parameter to “” which will also delete the setting in the
handset.
Changing the Default Passwords
The Server User and Server Password parameters should be changed from the
default values.
DHCP Menu
The DHCP menu is accessible only when the DHCP client is enabled. You can update the
following DHCP configuration parameters from the DHCP menu:
Name
Possible Values
Boot Server
0=Option 66, 1=Custom, 2=Static, 3=Custom+Option 66
Option 66: The handset will look for option number 66 (string type) in the response received from the DHCP
server. The DHCP server should send address information in option 66 that matches one of the formats described
for Server Address in Provisioning Server Menu.
Custom: The handset will look for the option number specified by the Boot Server Option parameter (below), and
the type specified by the Boot Server Option Type parameter (below) in the response received from the DHCP
server.
Static: The handset will use the boot server configured through the Server Menu. For more information, see
Provisioning Server Menu.
Custom + Option 66: The handset will use the custom option first or use Option 66 if the custom option is not
present.
Note: If the DHCP server sends nothing, the following scenarios are possible:
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Name
Possible Values

If a boot server value is stored in flash memory and the value is not 0.0.0.0, then the value stored in
flash is used.

Otherwise the handset sends out a DHCP INFORM query.
○
If a single DHCP INFORM server responds, this is functionally equivalent to the scenario where the
primary DHCP server responds with a valid boot server value.
○
If no DHCP INFORM server responds, the INFORM query process will retry and eventually time out.
Boot Server Option
128 through 254 (Cannot be the same as VLAN ID Option)
When the Boot Server parameter is set to Custom, this parameter specifies the DHCP option number in which the
handset will look for its boot server.
Boot Server Option Type
0=IP Address, 1=String
When the Boot Server parameter is set to Custom, this parameter specifies the type of DHCP option in which the
handset will look for its provisioning server. The IP Address provided must specify the format of the provisioning
server. The String provided must match one of the formats described for Server Address in Provisioning Server
Menu.
Option 60 Format
0=RFC 3925 Binary, 1=ASCII String
RFC 3925 Binary: Vendor-identifying information in the format defined in RFC 3925.
ASCII String: Vendor-identifying information in ASCII.
For more information, see Technical Bulletin Using DHCP Vendor Identifying Options with Spectralink Handsets.
Note: DHCP option 125 containing the RFC 3295 formatted data will be sent whenever option 60 is sent. DHCP
option 43 data is ignored.
Multiple DHCP INFORM Servers
If multiple DHCP INFORM servers respond, the handset should gather the
responses from these DHCP INFORM servers. If configured for Custom+Option66,
the handset will select the first response that contains a valid custom option value.
If none of the responses contain a custom option value, the handset will select the
first response that contains a valid option66 value.
Network Interfaces Menu
You can select the following items in the Network Interfaces menu:

Wi-Fi Menu

USBNet Menu
Wi-Fi Menu
You can modify the following parameters from the Wi-Fi menu:
Table 3-10: Wi-Fi Menu
Name
Possible Values
Enabled
Yes, No
A flag to determine if the wireless interface is enabled or not.
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Name
Possible Values
DHCP
Enabled, Disabled
If enabled, DHCP will be used to obtain the parameters discussed in DHCP or Manual TCP/IP Setup.
DHCP Boot Server
Enabled, Disabled
A flag to determine if the DHCP server is accessible.
IP Address
Dotted-decimal IP address
The handset’s IP address.
Note: This option is not available when the DHCP parameter is Enabled.
Subnet Mask
Dotted-decimal subnet mask
The handset’s subnet mask.
Note: This option is not available when the DHCP parameter is Enabled.
IP Gateway
Dotted-decimal IP address
The handset’s default router.
AC Required
Yes, No
A flag to determine if handsets will connect only to APs (access points) that enforce access control (Wi-Fi
Multimedia Admission Control [WMM-AC]). (See Caution note below.)
SSID
string
The Service Set Identifier (SSID) of the wireless network.
Security
0=No security, 1=WEP, 2=WPA-PSK, 3=WPA2-PSK, 4=WPA2Enterprise
The wireless security mode.
WEP
See WEP Menu.
WPA(2)-PSK
See WPA (2) PSK Menu.
WPA2-Enterprise
See WPA2-Enterprise Menu.
Radio
See Radio Menu.
Caution: WMM-AC not supported by 87-Series handsets
When deploying both 84-Series and 87-Series handsets in the same facility using
the same Wireless LAN, Wi-Fi Multimedia Admission Control (aka access control,
AC or WMM-AC) must be disabled in any handset parameters and APs as it is not
supported by 87-Series handsets. Any parameter that requires or enforces AC
must be disabled.
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WEP Menu
You can modify the following Wired Equivalent Privacy (WEP) configuration parameters on the
WEP menu:
Table 3-11: WEP Menu
Name
Possible Values
Authentication
0=Open System (default), 1=Shared Key
The WEP authentication method.
Key Length
0=40 bits (default), 1=104 bits
The authentication key length.
Default Key
1 to 4
The default key. The default key is 1.
Encryption
Enabled, Disabled
A flag to determine if wireless data is encrypted.
Key1, Key2, Key3, Key4
Hexadecimal value
The authentication keys. There are four possible keys. The key length is determined by the Key Length
parameter.
WPA (2) PSK Menu
You can modify the following Wi-Fi Protected Access (WPA)/WPA2 Pre-Shared Key (PSK)
configuration parameters on the WPA(2)-PSK menu:
Table 3-22: WPA (2) PSK Menu
Name
Possible Values
PSK Type
0=Passphrase (default), 1=Hexadecimal key
The pre-shared key type.
Passphrase
8 to 63 character ASCII string
The authentication passphrase.
Note: This parameter is unavailable when PSK Type is 1.
Key
256 bit hexadecimal string
The authentication key.
Note: This parameter is unavailable when PSK Type is 0.
WPA2-Enterprise Menu
You can modify the following parameters from the WPA2-Enterprise menu:
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Table 3-3: WPA2-Enterprise Menu
Name
Possible Values
Fast Roaming Method
0=Opportunistic Key Caching (OKC) , 1= Cisco
Centralized Key Management (CCKM)
The fast roaming method. These fast roaming methods allow for the part of the key derived from the server to be
cached in the wireless network, thereby, shortening the time to renegotiate a secure handoff.
EAP Method
1=EAP-TLS, 2=EAP-PEAPv0/MSCHAPv2 (default),
6=EAP-FAST
The Extensible Authentication Protocol (EAP).
User ID
String
The authentication user name.
Password
String
The authentication password.
PAC File Info
See PAC File Information.
EAP-FAST Inband Provisioning
Enabled, Disabled
A flag to determine whether or not EAP-FAST Inband Provisioning is enabled. Note: This parameter is unavailable
when EAP Method is 2.
Table 3-8: 802.1X Menu
Name
Possible Values
EAP Method
0 = None, 1=EAP-TLS, 2=EAP-PEAPv0/MSCHAPv2 6=EAP-FAST
The selected EAP type to be used for authentication. For more information, see Supporting 802.1X Authentication.
User ID
UTF-8 encoded string
The identity (or user name) required for 802.1X authentication.
Password
UTF-8 encoded string
The password required for 802.1X authentication. The minimum length is 6 characters.
PAC File Info
See PAC File Information.
EAP-FAST Inband
Provisioning
Enabled, Disabled
A flag to determine whether EAP-FAST Inband Provisioning is enabled. This parameter is used only if EAP
Method is EAP-FAST.
PAC File Information
You can modify Protected Access Credential (PAC) File Information from the PAC File
Information menu:
Table 3-9: PAC File Information Menu
Name
Possible Values
Description
PAC File Password
UTF-8 encoded string
The password required to decrypt the PAC file.
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Name
Possible Values
Description
PAC File Name
UTF-8 encoded string
The path or URL of the PAC file for download.
Remove PAC File
UTF-8 encoded string
A flag to determine whether or not to delete the PAC file from
the handset.
Radio Menu
You can modify the following parameters from the Radio menu:
Table 3-44: Radio Menu
Name
Possible Values
Regulatory Domain
0, 1, 2 or 10
Available values specify the regulatory domain. The supported values are 1 (North America), 2 (Europe) and 10
(Australia). If 0, no regulatory domain is selected. You must set the regulatory domain before the handsets can be
used. There is no default setting for this option and the handsets will not associate with an access point (AP) until
you specify a value.
5 GHz
See 5 GHz Menu.
2.4 GHz
See 2.4 GHz Menu.
5 GHz Menu
You can modify the following parameters from the 5 GHz menu:
Table 3-55: 5 GHz Menu
Name
Possible Values
5 GHz Enable
Enabled, Disabled
A flag to determine if the 5 GHz band is enabled.
Sub-bandx Enable
Enabled, Disabled
A flag to determine if the 5 GHz sub-band is enabled. There are four sub-bands (x=1 to 4).
Sub-bandx Transmit Power
1 to 7
The maximum power that the handset uses to transmit in the 5 GHz sub-band. The “7” setting is also called Auto
in some menus and is the maximum allowable power for that channel and data rate. If no maximum is set, the
handset uses the P5 settings for each channel activated. For more information, see the
device.wifi.radio.band5GHz.subBandx.txPower set of parameters in Appendix D.
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2.4 GHz Menu
You can modify the following parameters from the 2.4 GHz menu:
Table 3-6: 2.4 GHz Menu
Name
Possible Values
2.4 GHz Enable
Enabled, Disabled
A flag to determine if the 2.4 GHz band is enabled.
2.4 GHz Transmit Power
1 to 7
The maximum power that the handset uses to transmit in the 2.4 GHz sub-band. The “7” setting is also called Auto
in some menus. If no maximum is set, the handset uses the P5 settings for each channel activated. Note that ESTI
regulations limit the maximum setting in Europe. If P4 or above is selected for domain 2, the handset will
broadcast at the maximum allowable power which is 12 mW.. For more information, see the
device.wifi.radio.band2.4GHz.subBandx.txPower set of parameters in Appendix D.
USBNet Menu
You can modify the following parameters from the USBNet menu:
Table 3-77: USBNet Menu
Name
Possible Values
Enabled
1=Yes, 2=No
A flag to determine if USB networking is supported. USBnet is used by QNC/SLIC and USB Setup for initial
provisioning.
QNC/SLIC disables it when the files are produced. Parameter: device.usbnet.enabled
IP Address
Dotted-decimal IP address
The handset’s dotted-decimal IP address on the USBNet interface. For Spectralink handsets, the default value is
169.254.1.2 .
Subnet Mask
Dotted-decimal subnet mask
The handset’s subnet mask. For Spectralink handsets, the default value is 255.255.0.0 .
IP Gateway
Dotted-decimal IP address
The handset’s default router. For Spectralink handsets, the default value is 169.254.1.1 .
DHCP
Enabled, Disabled
If enabled, DHCP will be used to obtain the parameters discussed in DHCP or Manual TCP/IP Setup.
Syslog Menu
Syslog is a standard for forwarding log messages in an IP network. The term ‘syslog’ is often
used for both the actual syslog protocol, as well as the application or library sending syslog
messages.
The syslog protocol is a simple protocol: the syslog sender sends a small textual message (less
than 1024 bytes) to the syslog receiver. The receiver is commonly called ‘syslogd’, ‘syslog
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daemon’ or ‘syslog server’. Syslog messages can be sent through UDP, TCP, or TLS. The data
is sent in cleartext.
Because syslog is supported by a wide variety of devices and receivers, syslog can be used to
integrate log data from many different types of systems into a central repository.
Web Info: Information on Syslog
For more information on the syslog protocol, see RFC 3164.
You can modify the following parameters from the Syslog Menu:
Table 3-83: Syslog Menu
Name
Possible Values
Server Address
Dotted-decimal IP address OR Domain name string
The syslog server IP address. The default value is Null.
Server Type
None=0, UDP=1, TCP=2, TLS=3
The protocol that the handset will use to write to the syslog server. If set to None (or 0), transmission is turned off,
but the server address is preserved.
Facility
0 to 23
A description of what generated the log message. For more information, see section 4.1.1 of RFC 3164.
The default value is 16, which maps to ‘local 0’.
Render Level
0 to 6
Specifies the lowest class of event that will be rendered to syslog. It is based on device.syslog.renderLevel
and can be a lower value. See Appendix D for device parameters.
Note: Use left and right arrow keys to change values when using the Admin menu on the handset.
Prepend MAC Address
Enabled, Disabled
If enabled, the handset’s MAC address is prepended to the log message sent to the syslog server. Spectralink
recommends enabling this parameter.
Login Credentials Menu
You can modify the following parameters from the Login Credentials menu:
Table 3-9: Login Credentials Menu
Name
Possible Values
Domain
UTF-8 encoded string
The domain name used by a server.
User
UTF-8 encoded string
The user name used to authenticate to a server.
Password
UTF-8 encoded string
The password used to authenticate to a server.
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TLS Security Menu
This section refers to the TLS Menu available in the Updater, not Spectralink Software. There is
another menu, called TLS Security, available in the Admin menu. Navigate to Advanced
Settings> [password] Administration Settings> TLS Security. You can modify the following
parameters from the TLS Menu:
Table 3-10: TLS Menu
Name
Possible Values
View or install a custom CA cert
URL
A CA certificate that is installed on the handset to be used for TLS authentication.
View or clear a custom device credentials
Yes, No
A flag to determine whether or not the device certificate can be removed from the handset.
Configure TLS Profiles
There are two TLS Platform Profiles and six TLS Application Profiles. See TLS Profile Menu.
Configure TLS Applications
See Applications Menu.
TLS Profile Menu
You can modify the following parameters from the TLS Profile Menu:
Table 3-11: TLS Profile
Name
Possible Values
SSL Cipher Suite
String
The global cipher suite.
Custom SSL Cipher Suite
String
A custom cipher suite.
CA Cert List
String
The CA certificate sources that are valid for this profile.
Device Cert List
String
The device certificate sources that are valid for this profile.
TLS Applications Menu
You can modify which platform profile is used for applications from the Applications Menu.
Although not listed here, SIP, browser and LDAP are similarly available:
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Table 3-12: Applications Menu
Name
Possible Values
802.1X
1 or 2
The TLS Profile to use for 802.1X authentication.
Provisioning
1 or 2
The TLS Profile to use for provisioning authentication.
Syslog
1 or 2
The TLS Profile to use for syslog authentication.
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Chapter 4: Setting Up the Provisioning
Server
This chapter provides instructions for setting up your Spectralink handsets with a central
provisioning server. If you are new to this process, it is important to read every section in this
chapter. Please also refer to the Spectralink 84-Series Wireless Telephone Deployment Guide
for basic information.
Because of the large number of optional installations and configurations that are available, this
chapter focuses on one particular way that the Spectralink Software and the required external
systems might initially be installed and configured in your network.
If you want to begin setting up handset features, go to Part III: Configuring Features.
Why Use a Provisioning Server?
Spectralink strongly recommends that you use a central provisioning server to install and
maintain your Spectralink handsets. You can set up a provisioning server on the local LAN or
anywhere on the Internet. A provisioning server maximizes the flexibility you have when
installing, configuring, upgrading, and maintaining the handsets, and enables you to store
configuration, log, directory, and override files on the server. If you allow the handset write
access to your provisioning server, the handset can use the server to upload all of the file types
and store administrator and user settings. The handset is designed such that if it cannot locate a
provisioning server when it boots up, it will operate with internally saved parameters. This is
useful when the provisioning server is not available.
The default number of provisioning servers is one and the maximum number is eight. For more
information on the protocol used, see Supported Provisioning Protocols.
Provisioning Server Redundancy
You can configure multiple (redundant) provisioning servers—one logical server with multiple
addresses—by mapping the provisioning server DNS name to multiple IP addresses. See
Server Redundancy for more information.
If you set up multiple provisioning servers, you must be able to reach all of the provisioning
servers with the same protocol and the contents on each provisioning server must be identical.
The parameters described in Provisioning Server Menu can be used to configure the number of
times each server will be tried for a file transfer and also how long to wait between each
attempt. You can configure the maximum number of servers to be tried. For more information,
contact your Certified Spectralink Reseller.
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Provisioning Server Security Notes
For organizational purposes, Spectralink recommends configuring a separate log file directory,
an override directory, and a contact directory. Each directory can have different access
permissions. Normally, LOG, CONTACTS, and OVERRIDES have full read and write access.
See Understanding the Files Written by the Handsets for complete information.
Ensure that the file permissions you create provide the minimum required access and that the
account has no other rights on the server.
Tip: Allowing File Uploads to Your Provisioning Server
Spectralink recommends that you allow file uploads to the provisioning server
where the security environment permits. File uploads allow event log files to be
uploaded to the provisioning server. File uploads provide backup copies (override
configuration files) of changes users make to the handset’s configuration settings
through the Web server and/or local user interface. These override and log files
help service providers and Spectralink provide customer support when diagnosing
issues that may occur with the handset operation.
The handset’s server account needs to be able to add files that it can write to in the log file
directory and the provisioning directory. It must also be able to access files in all directories
mentioned in the <MAC-address>.cfg file. All other files that the handset needs to read, such
as the application executable and the standard configuration files, should be made read-only
using file server file permissions.
Tip: Use RFC-Compliant Servers
Spectralink recommends that you use RFC-compliant servers.
Each handset may open multiple connections to the server.
The handset will attempt to upload log files, a configuration override file, and a directory file to
the server if changed. This requires that the handset’s account has delete, write, and read
permissions. The handset will still function without these permissions, but will not be able to
upload or save files.
If you know the handset is going to download a file from the server, you should mark the file as
read-only.
Setting up an FTP Server as Your Provisioning Server
A basic provisioning configuration uses File Transfer Protocol or FTP. FTP servers are free,
require installation, and use logins and passwords. A free and popular server, FileZilla Server, is
available for Windows. FileZilla Server (version 0.9.xx) has been tested with the Spectralink
Software.
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Tip: Choosing a Provisioning Protocol
By default, Spectralink sets FTP as the provisioning protocol on all Spectralink
handsets. This guide focuses on the FTP provisioning protocol. Other supported
protocols include FTPS, TFTP, HTTP, and HTTPS.
See the Spectralink 84-Series Wireless Telephone Deployment Guide for a full
explanation of setting up FTP directories for both initial and central provisioning.
To set up an FTP server using FileZilla Server:
1
Download and install the latest version of FileZilla Server.
2
After installation, a Connect to Server pop-up displays on your computer. Select OK to
open the administrative user interface.
3
To configure a user, select Edit> Users in the status bar.
4
Select Add.
5
Enter the user name for the handset and select OK.
For example, bill123.
6
Select the Password checkbox and enter a password.
For example, 1234. The handset will use this password to log in.
Settings: FTP username and password
When you set up the initial provisioning server as an FTP server, use
administrator for the username and admin123 for the password. Ensure all
checkboxes are checked.
7
Select Page> Shared folders to specify the server-side directory where the provisioning
files will be located (and the log files uploaded).
8
Select Add and pick the directory.
9
To allow the handset to upload logs onto the provisioning server, select the Shared
Folders> Files> select Write and Delete checkboxes, and then select OK.
10
Determine the IP address of the FTP server by entering cmd in the Run dialog on your
Start menu, and ipconfig in the command prompt.
IP addresses of your network cards are displayed. One of them (if there are more than
one) will be your FTP server.
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Downloading Spectralink Software Files to the
Provisioning Server
This section explains how to download the Spectralink Software to the provisioning server.
Admin Tip: Upgrading Software
If you need to upgrade software on your handsets, please see Appendix B in the
Spectralink 84-Series Wireless Telephone Deployment Guide.
Microsoft® Skype for Business compatibility
Note: What is Skype for Business?
Microsoft has re-branded its software products formerly sold under the Lync name
to Skype for Business. In this document, when Skype for Business is referenced,
former Lync products are included.
Spectralink software is available in two variants – Skype for Business and non-Skype for
Business (or open SIP/Generic). Starting with Spectralink software 4.3/4.4, even numbered
releases support both Skype for Business and open SIP and odd numbered releases support
open SIP only. Release 4.7 and above differentiate Skype for Business and open SIP releases
by the build number. The last four digits of the build number denotes the build ID. For
Lync/Skype for Business releases the first digit is set to 1. For non-Lync/SIP/Generic releases,
the first digit is set to 2. For example:

Skype for Business – 5.0.0.1077

SIP/Generic – 5.0.0.2077
Handsets purchased without Skype for Business capability will not run Skype for Business
software releases, e.g. 5.0.0.1xxx. Handsets with Lync/Skype for Business compatibility will run
BOTH Skype for Business and SIP/Generic software releases.
All legacy phones manufactured prior to June 2013 support Skype for Business.
Manufacturing date
Support Skype for Business?
Prior to June 2013
Yes
June 2013 and later
Two handset variations:
1. Skype for Business-enabled: supports Skype for Business and SIP/Generic
2. SIP/Generic: does not support Skype for Business
All handsets have product ID’s that identify them as Skype for Business (WITH LYNC) or nonSkype for Business (WITHOUT LYNC) compatible.
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84-Series Product IDs with Microsoft Skype for Business Support
Model
Skype for Business SKUs
8440:
2200-37149-001, 2200-37150-001
2200-37174-101, 2200-37175-101
8441:
2200-37290-001, 2200-37290-101
8452:
2200-37172-001, 2200-37173-001
2200-37198-101, 2200-37199-101
8453:
2200-37294-001, 2200-37294-101
84-Series Product IDs without Microsoft Skype for Business Support
Model
Open SIP SKUs
8440:
2200-37147-001, 2200-37148-001
2200-37165-101, 2200-37164-101
8441:
2200-37288-001, 2200-37288-101
8450:
Legacy 8450 models support Microsoft Lync
Server 2013 or 2010
8452:
2200-37163-001, 2200-37162-001
2200-37161-101, 2200-37160-101
8453:
2200-37292-001, 2200-37292-101
Label example
Label example
Admin Tip
Handsets manufactured prior to June 2013 are not differentiated on the label as to
Lync or Lync. All handsets manufactured before June 2013 are Microsoft Lync
Server 2013/2010 compatible. Please consult the label date to determine Lync
compatibility.
To help understand if your 84-Series handset supports Skype for Business or Microsoft Lync
Server 2013 or 2010, look at the manufacturing date on the label in the battery compartment.
ALL 84-Series handsets produced before June 2013 support Microsoft Lync Server 2013 or
2010. For handsets built during or after June 2013, check the label text. The product ID and the
“with Lync” or “without Lync” text on the label will confirm whether or not the handset is enabled
for Skype for Business or Microsoft Lync Server 2013 or 2010.
Spectralink 84-Series Hardware IDs
Each Spectralink 84-Series model has a unique hardware ID. You can find this number printed
on the handset’s label inside the battery compartment. This number enables the handset model
to identify itself and provides flexibility to the administrator, permitting different models to load
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different code versions. See Product Model Number and Hardware ID Mapping for more
information about how hardware IDs are used.
Model Name
Hardware ID
SL8440
SL8450
SL8452
SL8441
SL8453
3111-36150-001
3111-36152-001
3111-36154-001
3111-67360-001
3111-67361-001
Go to the Spectralink Support Website to download current and past releases and access
supporting documentation.
Spectralink provides the Spectralink Software download in ZIP file format.
To download the Spectralink Software :
1
Access Spectralink Software from the Spectralink Support Website.
2
Acknowledge that you read the notices, accept the agreement, and choose Submit.
3
Save the Spectralink Software ZIP file download.
4
Extract (uncompress) the ZIP file.
Copy all files from the distribution ZIP file to working directory on the provisioning server,
maintaining the same folder hierarchy. To simplify provisioning, Spectralink
recommends, as a best practice, to start creating new configuration files from unedited
template files containing the default values. Rename the template file to your specific file
name as you configure and add specific parameter values for your site.

You will see one sip.ld file that includes all handset models. Accompanying folders
contain the configuration file templates and localization files.
See the Release Notes for a Description of all Parameters for a Spectralink
Software Release
For a description of each file in a Spectralink Software distribution, see the
Spectralink Software Release Notes for a particular Spectralink Software release
on the Spectralink Software Support Center.
Deploying and Updating Spectralink Handsets with a
Provisioning Server
This section explains how to deploy and update Spectralink handsets from a provisioning
server. If you are provisioning the handsets using a provisioning server for the first time, follow
the provisioning process described in the Spectralink 84-Series Wireless Telephone
Deployment Guide.
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You can create as many configuration files as you want and your configuration files can contain
any combination of parameters you put in them. You can put all parameters into one file or, for
example, you can put SIP server parameters in one file and handset features parameters in
another file. Configuration file variances are explained in the Spectralink 84-Series Wireless
Telephone Deployment Guide.
For large-scale deployments, the centralized provisioning method using configuration files is
strongly recommended. For smaller scale deployments, the Web Configuration Utility or local
interface may be used, but administrators need to be aware that settings made using these
methods will override settings made using configuration files.
For instructions on how to encrypt your configuration files, see Encrypting Configuration Files.
Shortcut Method to Deploy Spectralink Handsets with a Provisioning Server
The following steps are a shortcut method for provisioning procedure. Please use the
Spectralink 84-Series Wireless Telephone Deployment Guide for detailed instructions for each
of these steps.
To deploy handsets with a provisioning server using a shortcut method:
1
Obtain a list of MAC addresses for the handsets you want to deploy.
The MAC address is a 12-digit hexadecimal number on a label on the back of the
handset and on the outside of the shipping box. It is also available on the Status menu.
2
Create a per-handset <MACaddress>-ext.cfg file.
Do NOT use these names for a per-handset configuration file
Do NOT use the following file names as your per-handset file name:
<MACaddress>-phone.cfg,
<MACaddress>-Web.cfg,
<MACaddress>-app.log,
<MACaddress>-boot.log, or
<MACaddress>-license.cfg.
These file names are used by the handset itself to store user preferences
(overrides) and logging information.
Add handset registration parameters to the file, for example reg.1.address,
reg.1.label, and reg.1.type.
3
Create a per-site site<location>.cfg file.
For example, add the SIP server or feature parameters like
voIpProt.server.1.address and feature.corporateDirectory.enabled.
Settings: Configuring Your Phone for Local Conditions
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Some of the default settings are typically adequate; however, if SNTP settings are
not available through DHCP, you will need to edit the SNTP GMT offset, and
(possibly) the SNTP server address for the correct local conditions. Changing the
default daylight savings parameters will likely be necessary outside of North
America. Disable the local Web (HTTP) server or change its signaling port if the
local security policy dictates (see <httpd/>). Change the default location settings for
user interface language and time and date format (see Time and Date Display).
Modify any settings to match your WLAN and network as needed.
4
Create a master configuration file by performing the following steps:
a
Enter the name of each per-handset and per-site configuration files created in steps
2 and 3 in the CONFIG_FILES attribute of the master configuration file
(000000000000.cfg).
b
Optional) Edit the LOG_FILE_DIRECTORY attribute of master configuration file so
that it points to the log file directory.
c
(Optional) Edit the CONTACT_DIRECTORY attribute of master configuration file so
that it points to the organization’s contact directory.
d
(Optional) Edit the USER_PROFILES_DIRECTORY attribute of master configuration
file, if you intend to enable the User Login feature, so that it points to the directory
where user profile files are stored.
e
(Optional) Edit the CALL_LISTS_DIRECTORY attribute of master configuration file
so that it points to the user call lists.
5
Perform the following steps to configure the handset to point to the IP address of the
provisioning server and set up the user:
6
On the handset’s Home screen or idle display, select Settings> Advanced Settings>
[enter password]> Administration Settings> Network Configuration> Provisioning
Server> DHCP Menu.
When prompted for the administrative password, enter 456.
a
Open the DHCP Server menu by pressing the Select softkey. Set Boot Server to
Static and press
to return to the Provisioning Server menu.
b
Scroll down to Server Type and ensure that it is set to FTP.
c
Scroll down to Server Address and enter the IP address of your provisioning server.
d
Press the Edit softkey to edit the value and the OK softkey to save your changes.
e
Scroll down to Server User and Server Password and enter the user name and
password of the profile you created on your provisioning server.
In Setting up an FTP Server as Your Provisioning Server the example user given
was bill1234 and the example password was 1234.
f
Press the Back softkey twice.
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g
Scroll down to Save & Reboot, and then press the Select softkey.
The handset reboots.
After this step, if the file does not exist, the Spectralink Software will try the
unmodified APPLICATION APP_FILE_PATH attribute (sip.ld).
For more information, see Parsing Vendor ID Information.
7
Ensure that the configuration process completed correctly.
On the handset, press the arrow key, and then select Settings> Status> Platform>
Application> Main> OK to see the Spectralink Software version and Settings> Status>
Platform> Configuration> OK to see the configuration files downloaded to the handset.
Monitor the provisioning server event log and the uploaded event log files (if permitted).
All configuration files used by the provisioning server are logged.
The handset will upload two logs files to the LOG_DIRECTORY directory:
<MACaddress>-app.log and <MACaddress>-boot.log.
You can now test your deployment by making calls and testing features.
Upgrading Spectralink Software
You can upgrade the software that is running on the Spectralink handsets in your organization.
The upgrade process varies with the version of Spectralink Software that is currently running on
your handsets and with the version that you want to upgrade to. The Updater, Spectralink
Software executable, and configuration files can all be updated using centralized provisioning.
Updating Spectralink Software on a Single Phone
Starting with Spectralink Software 4.3.x, you can use the Software Upgrade tool in
the Web Configuration Utility to update the Spectralink Software version running on
a single handset. Note that configuration changes made to individual handsets
using the Web Configuration Utility will override configuration settings made using
central provisioning. For instructions on how to update Spectralink Software, see
Technical Bulletin Using the Software Upgrade Tool in the Web Configuration
Utility.
To continue setting up a provisioning server, use the instructions in the Spectralink 84-Series
Wireless Telephone Deployment Guide.
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Chapter 5: Understanding the Files
Written by the Handsets
The Spectralink 84-Series handsets can write several types of information into directories set up
on the central provisioning server. These directories serve as repositories for the different types
of files written by the handsets and allow an administrator to easily find needed information. If
separate directories are not provisioned, and if the handset has write privileges, all files will be
written to the root directory.
The Spectralink 84-Series Deployment Guide explains how to set up directories on the central
provisioning server (or elsewhere) so that the files written by the handset can be sorted and
viewed efficiently. These directories must have full read and write access. The “Directory
element” in the table below is referring to the directories that are provisioned in the master
configuration file. The Directory names tell the handset where to write the files it produces. For
example:
Filenames for the files produced by the handset start with the MACaddress and use an
extension to signify the type of information in it. E.g. 00907a0cd989-phone.cfg is the filename of
an override file created when the user changed a parameter in the handset keypad menus.
If User Profiles are deployed, files start with the login name instead of the MACaddress. E.g.
lgates-phone.cfg.
Log files are written and stored as text files. All other files are written and stored as .xml files.
Type of file
Directory element
Contents
Extension
Log
LOG_FILE_DIRECTORY
This is the directory where the
handset will write its log files.
-boot.log
-app.log
Override
OVERRIDES_DIRECTORY
Parameters set in the .cfg files can
be overridden when changed in the
Web Configuration Utility or in the
handset menus using its keypad.
This directory stores these
overrides by MACaddress so that
they are available when the handset
restarts.
-phone.cfg
-web.cfg
Contacts
CONTACTS_DIRECTORY
Contacts are stored by
MACaddress in this directory as a
backup so that they can be
reloaded when the handset reboots.
-directory.xml
Calls
CALL_LISTS_DIRECTORY
Call lists are stored by handset or
by User Profile.
-calls.xml
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Log Files
Log files are discussed in detail in the Troubleshooting section.
Overrides
Certain configuration methods take precedence over other methods and produce override files
on the central provisioning server. The handset reads these files when it checks in with the
central provisioning server and uses the information stored in them for operational information
and parameter configuration.
The precedence order for configuration methods follows this sequence (highest to lowest
priority):
1
Local handset user interface
2
Web Configuration Utility
3
Central Provisioning Server
4
Default values
If you set a parameter in the central provisioning server and the user later changes it in the
handset menus, the setting made by the user will take precedence and override the setting in
the central provisioning server. This action updates an overrides file that is stored in the
Overrides directory, if specified.
Note that although certain parameters take precedence and overrule other parameters, this type
of behavior does NOT produce an override file. This behavior is fully explained in the given
parameter description. We use the term “overrule” to describe parameter precedence. We use
the term “override” to describe configuration method precedence.
Example:
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This example shows an overrides file for one handset that shows what the user has done to
change from the default or configured settings:

set the normal notification profile (np) to a silent ring, etc.

set the selected notification profile to silent,

has forwarded a call to extension 0567,

changed various PTT channel settings.
Deleting the override file
The administrator might want to delete all overrides. This can be done both remotely and if you
have access to the phone.
With phone access
1
Power off the phone.
2
Modify the override files to have empty <PHONE_CONFIG> tags. Open the override file
and remove all tags. The file should only have these active parameters:
<PHONE_CONFIG>
</PHONE_CONFIG>
3
Power the phone back on.
Remotely
The remote method could work in conjunction with a shift change where the likelihood of the
phones power cycling is high. It is also possible to automate this process with scripts.
1
Modify the override files to have empty <PHONE_CONFIG> tags. Open the override file
and remove all tags. The file should only have these active parameters:
<PHONE_CONFIG>
</PHONE_CONFIG>
2
Make the override file read-only. This can be done at the individual file level, or (if a
specific Override directory is specified in the configuration files) by making the entire
directory read-only.
3
Wait for the phone to power cycle at least once.
4
Allow the override file to be writable, if desired.
Contacts
Each handset can store up to 9999 contacts in its local Contact Directory. The Contact Directory
can be viewed on the handset by navigating to Home> Contacts / Call Lists> Contact
Directory. This contact list is also written to the central provisioning server and stored in a
Contacts Directory whenever any information is changed. Contact information is also stored
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locally in the handset so the file on the central provisioning server serves as a backup in case
the handset loses its memory.
Call List
The 84-Series handset records missed, received and placed calls in a call list that can be
viewed on the handset by navigating to Home> Contacts / Call Lists> Call Lists. Call history
is stored locally on the handset and will survive a restart or reboot unless the list has been
cleared. The user can use the call list to redial previous outgoing calls, return incoming calls,
and save contact information from call list entries to the contact directory.
Such calls are logged in the Call List directory that resides on the central provisioning server.
These call logs contain call information such as remote party identification, time and date of the
call, and call duration. All call logs are enabled by default but can be disabled by turning off the
unwanted parameter.
The Call List directory on the central provisioning server is a real-time file. When a new call is
made, the call is logged. When the lists are cleared on the handset, the calls in the cleared list
are also cleared out of the Call List directory on the central provisioning server.
Table 6-1: Configuring the Call Lists
Parameter
feature.callList.enabled
All locally controlled call lists.
feature.callListMissed.enabled1
The missed calls list.
feature.callListPlaced.enabled1
The placed calls list.
feature.callListReceived.enabled1
The received calls list.
Permitted Values
0 or 1
Default
1
0 or 1
1
0 or 1
1
0 or 1
1
If 0, the call list is disabled. If 1, the call list is enabled. To enable the Missed, Placed, or Received call lists,
feature.callList.enabled must be enabled.
Call log example
A logged call from the call list looks like this:
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The next table describes each element and attribute that displays in the call log
Table 6-2: Call Log Elements and Attributes
Element
Permitted Values
direction
In, Out
Call direction with respect to the user.
disposition
Busy, Forwarded, Normal, Partial, Preempted,
Rejected, RemotelyHandled, Transferred
What happened to the call. When a call entry is first created, the disposition is set to Partial.
line
Positive integer
The line (or registration) index.
protocol
SIP
The line protocol.
startTime
String
The start time of the call. For example: 2010-01-05T12:38:05 in local time.
duration
String
The duration of the call, beginning when it is connected and ending when the call is terminated.
For example: PT1H10M59S.
count
Positive Integer
The number of consecutive missed and abandoned calls from a call destination.
destination
Address
The original destination of the call.
For outgoing calls, this parameter designates the outgoing call destination; the name is initially supplied by the
local handset (from the name field of a local contact entry) but may later be updated via call signaling. This field
should be used for basic redial scenarios.
For incoming calls, the called destination identifies the requested party, which may be different than any of the
parties that are eventually connected (the destination may indicate a SIP URI which is different from any SIP URI
assigned to any lines on the handset).
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Element
Permitted Values
source
Address
The source of the call (caller ID from the call recipient’s perspective).
Connection
Address
An array of connected parties in chronological order.
As a call progresses, the connected party at the far end may change, for example, if the far end transfers the call
to someone else. The connected element allows the progression of connected parties, when known, to be saved
for later use. All calls that contain a connected state must have at least one connection element created.
finalDestination
Address
The final connected party of a call that has been forwarded or transferred to a third party.
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Part III: Configuring Features
Part III provides you with an in-depth look at advanced features and parameters. The
Spectralink 84-Series Wireless Telephone Deployment Guide covers parameters that are used
by 80% of all installations. However, since requirements vary, Part III covers the entire range of
configurable parameters for every type of feature and functional requirement.
Part III is arranged by types of features and settings that you might need to deploy to further
customize your installation. This Part is divided into four Chapters that cover:

An explanation of inherent features that are not configurable,

User settings that can be set by the administrator but the user can freely alter through
the user menus. Some of these options can be locked or made unavailable by the
administrator.

Feature settings that are configured by the administrator. These settings are optional
and only available if configured by an administrator. In some cases once the feature is
configured, user menus contain options that the user can configure to customize the
feature for individual use and preferences.

System settings that the administrator configures to adjust the way the handset interacts
with the greater infrastructure.
Configuration parameters are named for the function they provide but frequently several types
of parameters must be configured to provision a feature or function. Therefore this section is
arranged by feature and function, not by parameter name.
The parameters listed in this section are available in the Config folder that is downloaded with
the software. For any parameters not available in the scenario templates, look in the
Troubleshooting folder for the “everything.cfg” file. You will find an alphabetical folder hierarchy
of all the parameters detailed in this document. To find the exact parameters you want to use,
open the file and use your search tools.
The easiest way to provision any parameter is to simply drag and drop or copy/paste it from the
source template into the .cfg file you will use to deploy the handsets. Edit it in your final .cfg file
per the instructions in the parameter list.
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Chapter 6: Features that Cannot be
Configured
Audio Processing Features
The Spectralink 84-Series handsets have these built-in audio processing features: automatic
gain control, background noise suppression, comfort noise fill, dynamic noise reduction, jitter
buffer and packet error concealment, and low delay audio packet transmission. These features
work automatically, without configuration changes.
Automatic Gain Control
Automatic Gain Control (AGC) is applicable to handsfree operation and is used to boost the
transmit gain of the local talker in certain circumstances. This increases the effective userhandset radius and helps with the intelligibility of soft-talkers.
Background Noise Suppression
Background noise suppression (BNS) is designed primarily for handsfree operation and reduces
background noise to enhance communication in noisy environments.
Comfort Noise Fill
Comfort noise fill is designed to help provide a consistent noise level to the remote user of a
handsfree call. Fluctuations in perceived background noise levels are an undesirable side effect
of the non-linear component of most acoustic echo cancellation (AEC) systems. This feature
uses noise synthesis techniques to smooth out the noise level in the direction toward the remote
user, providing a more natural call experience. This feature is different from Voice Activity
Detection.
Dynamic Noise Reduction
Dynamic noise reduction (DNR) provides maximum microphone sensitivity, while automatically
reducing background noise— from fans, projectors, heating and air conditioning—for clearer
sound and more efficient conferencing.
Jitter Buffer and Packet Error Concealment
The handset employs a high-performance jitter buffer and packet error concealment system
designed to mitigate packet inter-arrival jitter and out-of-order, or lost or delayed (by the
network) packets. The jitter buffer is adaptive and configurable for different network
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environments. When packets are lost, a concealment algorithm minimizes the resulting negative
audio consequences.
Low-Delay Audio Packet Transmission
The handset is designed to minimize latency for audio packet transmission.
Call Timer
A call timer displays on the handset’s screen. A separate call duration timer displays the hours,
minutes, and seconds of each call in progress.
There are no related configuration changes.
Called Party Identification
By default, the handset displays and logs the identity of parties called from the handset. The
handset obtains called party identity from the network signaling. Because Called Party
Identification is a default state, the handset will display caller IDs matched to the call server and
does not match IDs to entries in the Local Contact Directory or Corporate Directory.
There are no related configuration changes.
Connected Party Identification
By default, the handset displays and logs the identity of remote parties you connect to if the call
server can derive the name and ID from the network signaling. Note that in cases where remote
parties have set up certain call features, the remote party you connect to—and the caller ID that
displays on the handset—may be different than the intended party. For example, Bob places a
call to Alice, but Alice has call diversion configured to divert Bob’s incoming calls to Fred. In this
case, the handset will log and display the connection between Bob and Fred. Note that the
handset does not match party IDs to entries in the contact directory or the corporate directory.
Microphone Mute
The handsets have a microphone mute softkey. When you activate microphone mute, a mute
icon will display on the status bar.
No configuration changes can be made to the microphone mute feature.
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Synthesized Call Progress Tones
Spectralink handsets play call signals and alerts, called call progress tones, such as busy
signals, ringback sounds, and call waiting tones. The built-in call progress tones on your
handset match standard North American tones.
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Chapter 7: Configurable Features on the
User Menus
Features that are available on the User menus, such as ring tones, can be configured by the
administrator in the central provisioning server but may be changed by the user, creating
override files stored on the central provisioning server. Some user-level parameters that are
made available to users by default can be disabled by the administrator if certain usability
factors need to be enforced. This chapter covers features that are available on the User menus
that have user-level parameters that can be configured on the central provisioning server.
Some features, such as Push-to-talk, are set up by the administrator and are designed to be
customized by the user, like subscribing to certain channels that have been enabled by the
administrator. These features are covered in the next chapter.
Features that are not on the User menus but can be set through the Web Configuration Utility
are covered in the next chapter.
Call Forwarding
The handset provides a flexible call forwarding feature that enables you to forward incoming
calls to another destination. To enable and set call forwarding from the handset, navigate to
Home> Settings> Feature Settings> Forward. From this menu you can apply call forwarding
in the following ways:
 To all calls
 When your handset is busy
(Since the 84-Series handsets support 24 incoming calls by default [call.callsPerLineKey]
they won’t offer a busy signal to the PBX unless it’s handling 24 calls to any given
handset, an unlikely event. In order to forward on busy, the callsPerLineKey must be set
to 1 or 2 instead of 24.)
 When the handset has not been answered within a specified number of rings
Call forwarding is also available as a divert function in the Contacts Directory:
 To incoming calls from a specific caller or extension
 You can have incoming calls forwarded automatically to a predefined destination you
choose or you can manually forward calls to a destination.
To enable server-based call forwarding, see your PBX manual and the Interop Guide relating to
that server.
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Troubleshooting: If Call Forwarding Does Not Work
The server-based and local call forwarding features do not work with the Shared
Call Appearance (SCA) and Bridged Line Appearance (BLA) features. If you have
SCA or BLA enabled on your handset, you will need to disable the feature before
you can use call forwarding.
The call server uses the Diversion field with a SIP header to inform the handset of a call’s
history. For example, when you enable call forwarding, the Diversion header allows the
receiving handset to indicate who the call was from, and the handset number it was forwarded
from.
Summary
Parameter
Used to:
voIpProt.SIP.serverFeatureControl.cf
Enable or disable server-based call forwarding
voIpProt.SIP.serverFeatureControl.localProcessing.cf
Enable or disable local call forwarding behavior when
server-based call forwarding is enabled
voIpProt.SIP.header.diversion.*
Enable or disable the display of the Diversion header and
the order in which to display the caller ID and number
divert.*
Set all call diversion settings including a global forward-to
contact and individual settings for call forward all, call
forward busy, call forward no-answer.
reg.x.fwd.*
Enable or disable server-based call forwarding as a perregistration feature
Table 7-1: Configuring Call Forwarding
Parameter
Permitted Values
Default
voIpProt.SIP.serverFeatureControl.cf1
0 or 1
0
If set to 1, server-based call forwarding is enabled. The call server has control of call forwarding.
If set to 0, server-based call forwarding is not enabled.
voIpProt.SIP.serverFeatureControl.localProcessing.cf
0 or 1
1
If set to 0 and voIpProt.SIP.serverFeatureControl.cf is set to 1, the handset will not perform local Call
Forward behavior.
If set to 1, the handset will perform local Call Forward behavior on all calls received.
voIpProt.SIP.header.diversion.enable1
0 or 1
0
If set to 1, the diversion header is displayed if received. If set to 0, the diversion header is not displayed.
voIpProt.SIP.header.diversion.list.useFirst1
0 or 1
1
If set to 1, the first diversion header is displayed. If set to 0, the last diversion header is displayed.
reg.x.fwdcontact
string
Null
The forward-to contact for calls when always is specified. If Null, calls are not forwarded.
reg.x.fwd.noanswer.status
0 or 1
0
If 0, calls are not forwarded if there is no answer. If 1, calls are forwarded to the contact specified by
reg.x.noanswer.contact after ringing for the length of time specified by reg.x.fwd.noanswer.ringCount.
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Parameter
Permitted Values
Default
reg.x.fwd.noanswer.contact
string
Null
The forward-to contact used for calls forwarded due to no answer. If Null, the contact specified by
divert.x.contact will be used.
reg.x.fwd.noanswer.ringCount
0 to 65535
0
The number of seconds the handset should ring for before the call is forwarded because of no answer. Note: The
maximum value accepted by some call servers is 20.
reg.x.fwd.busy.status
0 or 1
0
If 0, incoming calls that receive a busy signal will not be forwarded. If 1, busy calls are forwarded to the contact
specified by reg.x.fwd.busy.contact.
reg.x.fwd.busy.contact
string
Null
The forward-to contact for calls forwarded due to busy status. If Null, the contact specified by divert.x.contact
will be used.
1
Change causes handset to restart or reboot.
Example Call Forwarding Configuration
In the example configuration shown next, the call forwarding parameters for registration 1 have
been changed from the default values.

The forward-always contact for registration 1 is 5557 and this number will be used if the
parameters divert.busy, divert.dnd, or divert.noanswer are not set.

Parameters you set in those fields will overrule divert.1.contact.

To enable these three divert options for each registration, you will need to enable the
divert.fwd.x.enabled parameter and the .enabled parameter for each of the
three forwarding options you want to enable.

In this example, divert.fwd.1.enabled has been disabled; all calls to registration 1
will be diverted to 5557 and you do not have the option of enabling any of the three
forwarding options on the handset.

The three divert options are enabled for registration 2 in the divert.fwd.2.enabled
parameter, giving you the option to enable or disable any one of the three forwarding
options on the handset.

When do not disturb (DND) is turned on, you can set calls to registration 2 to be diverted
to 6135559874 instead of 5557.

The parameter divert.noanswer.2.enabled is enabled so that, on the handset,
you can set calls to registration 2 that ring for more than 15 seconds, specified in
divert.noanswer.2.timeout, to be diverted to 2987, as set in
divert.noanswer.2.contact.
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The handset has a flexible call forward/diversion feature for each registration. In all cases, a call
will only be diverted if a non-Null contact has been configured.
In the following table, x is the registration number. SL8400: x=1-6.
Table 7-2: Call Diversion (Call Forwarding) Parameters
Parameter
Permitted Values
Default
divert.x.contact1
contact address: ASCII encoded string containing digits
(the user part of a SIP URL) or a string that constitutes a
valid SIP URL (6416 or 6416@Spectralink.com)
Null
The forward-to contact used for all automatic call diversion features. All automatically forwarded calls will be
directed to this contact. The contact can be overridden by a busy contact, DND contact, or no-answer contact as
specified by the busy, dnd, and noAnswer parameters that follow.
divert.x.sharedDisabled1
0 or 1
1
If 0, call diversion features can be used on shared lines. If 1, call diversion features are disabled on shared lines.
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Parameter
Permitted Values
Default
divert.x.autoOnSpecificCaller2
0 or 1
1
If 0, the Auto Divert feature of the contact directory is disabled for registration x. If 1, calls on registration x may be
diverted using Auto Divert, you may specify to divert individual calls or divert all calls.
divert.busy.x.enabled2
divert.busy.x.contact1
0 or 1
contact address
1
Null
Divert incoming calls that reach a busy signal. If enabled is set to 1, calls will be diverted when registration x is
busy. Calls will be sent to the busy contact’s address if it is specified; otherwise calls will be sent to the default
contact specified by divert.x.contact. If enabled is set to 0, calls will not be diverted if the line is busy.
divert.dnd.x.enabled2
divert.dnd.x.contact1
0 or 1
contact address
0
Null
Divert calls when Do Not Disturb is enabled. If enabled is set to 1, calls will be diverted when DND is enabled on
registration x. Calls will be sent to the DND contact’s address if it is specified; otherwise calls will be sent to the
default contact specified by divert.x.contact.
divert.fwd.x.enabled2
0 or 1
1
If 0, the user cannot enable universal call forwarding (automatic forwarding for all calls on registration x). If 1, a
Forward softkey will display on the flyout menu when you press the Features softkey.
divert.noanswer.x.enabled2
divert.noanswer.x.contact1
divert.noanswer.x.timeout1
0 or 1
contact address
positive integer
1
Null
55
If no-answer call diversion is enabled, calls that are not answered after the number of seconds specified by
timeout will be sent to the no-answer contact. If the no-answer contact is set to Null, the call will be sent to the
default contact specified by divert.x.contact. If enabled is set to 0, calls will not be diverted if they are not
answered.
1
Change causes handset to restart or reboot.
2
Change causes handset to restart or reboot. If server-based call forwarding is enabled, this parameter is disabled.
Keypad Lock
Keypad Lock is enabled by default. This feature locks the keypad to prevent inadvertent dialing
while the phone is idle and/or while in a call. The feature adds a Keypad Lock and/or an In-call
Keypad Lock option on the Features softkey menu. Once the handset is locked for idle, an
Unlock softkey appears. When pressed it will unlock the keypad. No password is required. The
in-call keypad lock will expire when the phone call is ended.
If your deployment requires additional handset security, please see the Phone Lock section.
Table 7-3: Keypad Lock
Parameter
Permitted Values
Default
keypadLock.enabled1
0 or 1
0
If 0, the keypad lock feature is disabled. If 1, the feature is enabled.
keypadLock.idleTimeout1
0 to 65535
0
The maximum time (in seconds) the handset can be idle before the keypad will lock.
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Parameter
Permitted Values
Default
inCall.keypadLock.enabled
0 or 1
0
If enabled the In-call Keypad Lock item will appear on the Profile soft key when a call is connected. Note that if
more than one call connected at the same time, this item will not show up as the space on the flyout is used by
other items.
inCall.keypadLock.autoLockTime
0 to 30
0
If 0, auto-lock is not enabled.
If not zero then the In-call Keypad Lock will be engaged nn seconds after a call is fully connected.
Only 1 call can engage the lock. If a second call is started during the timeout period for the first call, the auto-lock
for the second call will be ignored. The In-call Keypad Lock item on the Profile soft key flyout can still be used even
if auto-lock is configured.
1
Change causes handset to restart or reboot.
Multi Key Answer
Multikey Answer enables you to answer incoming calls by pressing any key on the handset’s
keypad. Multikey Answer is disabled by default but may be enabled by the administrator. You
cannot use the Multi Key Answer feature for Open Application Interface (OAI) calls, Group
Paging, or Push-to-talk (PTT) calls. Navigate to the option by going to Settings> Basic
Settings> Preferences> Multi Key Answer.
Table 7-4: Enabling Multi Key Answer
Parameter
Permitted Values
Default
up.multiKeyAnswerEnabled
0 or 1
0
If 1, incoming calls can be answered by pressing any key. If 0, incoming calls can only be answered using the Talk
button or the Start key.
1
Change causes handset to restart or reboot.
Notification Profiles
The Spectralink handsets support four profiles for notification alerts: Normal, Silent, Meeting,
and Custom1. You can customize each profile with a unique name, unique ringtones, alerts,
and vibrations for specific situations. For example, you can customize barcode scan alerts or
when you receive an instant message.
Notification Profiles are selectable on the handset by selecting a profile from the standby mode
Profile softkey or by navigating to Settings> Basic Settings> Notification Profiles where
profiles can be modified by the user.
By default, the ringing and alert volumes are at the same level. You can configure the ringer
volume for ringing only and set a distinct alert volume for each alert type. By default, the
handset will maintain changes you make to the ringer volume when the handset reboots or
restarts.
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This section shows you how to choose a default notification profile from four available types Normal, Silent, Meeting, Custom1 - and shows you the parameters you can set for each type.
Each profile is defined by an alert type and a ring type; there are 15 alert types and three ringing
types.
For each alert type:

You can select a tone pattern from the patterns defined in se.pat.misc. These patterns
include: custom1 to custom10, instantMessaging, localHoldNotification,
messageWaiting, misc1 to misc9, negativeConfirm, positiveConfirm,
remoteHoldNotification, silent, and welcome. For information on customizing these
parameters, refer to se.pat.misc.

You can determine if the handset should vibrate for the alert. Set the vibrate parameter
to 0 to disable vibration or 1 to enable vibration.
For each ringer type:

You can choose a tone pattern from the patterns defined in se.pat.ringer. These
patterns include: default, ringer1 to ringer 24, and 1 to 22.

You can also set the vibration type for the ringer. You can select off, continuous,
shortPulse, or longPulse.
Configure the default notification profile by setting the parameter shown in the following table:
Table 7-5: Notification Profile Selection Parameter
Parameter
Permitted Values
Default
np.selected
Normal, Silent, Meeting, Custom1
Normal
The initial profile that is selected when the handset powers on and active during operation. The user can override
this default profile to set a new default profile that will be selected when the handset powers on the next time.
Table 7-6: Normal Profile Alert Parameters
Parameter
Permitted Values
Default
np.normal.label
String
Normal
Any tone (see se.pat.misc)
0 or 1
misc2
0
The name of the profile type.
np.normal.alert.barcodeBeep.tonePattern
np.normal.alert.barcodeBeep.vibration
The tone pattern and vibration (1 to enable) for the alert played when a barcode is scanned.
np.normal.alert.docked.tonePattern
np.normal.alert.docked.vibration
Any tone (see se.pat.misc)
0 or 1
positiveConfirm
0
The tone pattern and vibration (1 to enable) for the alert played when the handset is docked.
np.normal.alert.undocked.tonePattern
np.normal.alert.undocked.vibration
Any tone (see se.pat.misc)
0 or 1
negativeConfirm
0
The tone pattern and vibration (1 to enable) for the alert played when the handset is undocked.
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Parameter
Permitted Values
Default
np.normal.alert.instantMessaging.tonePattern
np.normal.alert.instantMessaging.vibration
Any tone (see se.pat.misc)
0 or 1
instantMessage
0
The tone pattern and vibration (1 to enable) for the instant message alert.
np.normal.alert.localHoldNotification.tonePattern
np.normal.alert.localHoldNotification.vibration
Any tone (see se.pat.misc)
0 or 1
localHoldNotification
0
The tone pattern and vibration (1 to enable) for the local hold notification alert.
np.normal.alert.lossOfNetwork.tonePattern
np.normal.alert.lossOfNetwork.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the network is lost.
np.normal.alert.lowBattery.tonePattern
np.normal.alert.lowBattery.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the battery is low.
np.normal.alert.veryLowBattery.tonePattern
np.normal.alert.veryLowBattery.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the battery is very low.
np.normal.alert.messageWaiting.tonePattern
np.normal.alert.messageWaiting.vibration
Any tone (see se.pat.misc)
0 or 1
messageWaiting
0
The tone pattern and vibration (1 to enable) for the alert played if there is a message waiting.
np.normal.alert.negativeConfirm.tonePattern
np.normal.alert.negativeConfirm.vibration
Any tone (see se.pat.misc)
0 or 1
negativeConfirm
0
The tone pattern and vibration (1 to enable) for the negative confirmation alert.
np.normal.alert.positiveConfirm.tonePattern
np.normal.alert.positiveConfirm.vibration
Any tone (see se.pat.misc)
0 or 1
positiveConfirm
0
The tone pattern and vibration (1 to enable) for the positive confirmation alert.
np.normal.alert.pttTransmit.tonePatterna
np.normal.alert.pttTransmit.vibration
Any tone (see se.pat.misc)
0 or 1
misc3
0
The tone pattern and vibration (1 to enable) for the alert played if sending a push-to-talk page.
np.normal.alert.pttWait.tonePattern
np.normal.alert.pttWait.vibration
Any tone (see se.pat.misc)
0 or 1
misc4
0
The tone pattern and vibration (1 to enable) for the push-to-talk wait alert.
np.normal.alert.welcome.tonePattern
np.normal.alert.welcome.vibration
Any tone (see se.pat.misc)
0 or 1
Welcome
0
The tone pattern and vibration (1 to enable) for the alert played when the handset turns on.
np.normal.ringing.calls.tonePattern
A ringer (see se.pat.ringer)
default
The ringtone (see se.pat.ringer) and vibration (1 to enable) for normal calls.
np.normal.ringing.calls.vibration
off, continuous, shortPulse,
longPulse
off
The ringtone (see se.pat.ringer) and vibration (1 to enable) for normal calls.
np.normal.ringing.oai1.tonePattern
A ringer (see se.pat.ringer)
ringer2
The ringtone (see se.pat.ringer) for Open Application Interface (OAI) communications.
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Parameter
Permitted Values
Default
np.normal.ringing.oai1.vibration
off, continuous, shortPulse,
longPulse
off
The vibration pattern for Open Application Interface (OAI) communications.
np.normal.ringing.oai2.tonePattern
A ringer (see se.pat.ringer)
ringer2
The ringtone (see se.pat.ringer) and vibration (1 to enable) for Open Application Interface (OAI) version 2.2
communications.
np.normal.ringing.oai2.vibration
off, continuous, shortPulse,
longPulse
off
The vibration pattern for Open Application Interface (OAI) version 2.2 communications.
np.normal.ringing.privateLine.tonePattern
default, ringer1 to ringer24
default
The ringtone (see se.pat.ringer) for a private line registered to Microsoft Skype for Business or Microsoft Lync
Server 2013 or 2010.
np.normal.ringing.privateLine.vibration
off, continuous, shortPulse,
longPulse
shortPulse
The vibration pattern for a private line registered to Microsoft Skype for Business or Microsoft Lync Server 2013 or
2010.
np.normal.ringing.toneVolume.handset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Handset and Normal profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.normal.ringing.toneVolume.headset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Headset and Normal profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.normal.ringing.toneVolume.chassis
-1000 to 1000
0
The attribute is set (on adjusting ring volume) when ringing termination is Chassis and Normal profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.normal.ringing.toneVolume.dock
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when handset is at the speakerphone dock and Normal profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.normal.ringing.toneVolume.bluetoothHeadset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Bluetooth Headset and Normal profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.normal.ringing.toneVolume.reserved
-1000 to 1000
-21
-1000 to 1000
-21
Not currently used. Reserved for future use.
np.normal.ringing.toneVolume.usbHeadset
The attribute is set (on adjusting ring volume) when ringing termination is a USB headset and Normal profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
Table 7-7: Silent Profile Alert Parameters
Parameter
Permitted Values
Default
np.silent.label
String
silent
The name of the profile type.
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Parameter
Permitted Values
Default
np.silent.alert.barcodeBeep.tonePattern
np.silent.alert.barcodeBeep.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played when a barcode is scanned.
np.silent.alert.docked.tonePattern
np.silent.alert.docked.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played if the handset is docked.
np.silent.alert.undocked.tonePattern
np.silent.alert.undocked.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played if the handset is undocked.
np.silent.alert.instantMessaging.tonePattern
np.silent.alert.instantMessaging.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the instant message alert.
np.silent.alert.localHoldNotification.tonePattern
np.silent.alert.localHoldNotification.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the local hold notification alert.
np.silent.alert.lossOfNetwork.tonePattern
np.silent.alert.lossOfNetwork.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played if the network is lost.
np.silent.alert.lowBattery.tonePattern
np.silent.alert.lowBattery.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played if the battery is low.
np.silent.alert.veryLowBattery.tonePattern
np.silent.alert.veryLowBattery.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played if the battery is very low.
np.silent.alert.messageWaiting.tonePattern
np.silent.alert.messageWaiting.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played if there is a message waiting.
np.silent.alert.negativeConfirm.tonePattern
np.silent.alert.negativeConfirm.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the negative confirmation alert.
np.silent.alert.positiveConfirm.tonePattern
np.silent.alert.positiveConfirm.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the positive confirmation alert.
np.silent.alert.pttTransmit.tonePatterna
np.silent.alert.pttTransmit.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played if sending a push-to-talk page.
np.silent.alert.pttWait.tonePattern
np.silent.alert.pttWait.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the push-to-talk wait alert.
np.silent.alert.welcome.tonePattern
np.silent.alert.welcome.vibration
Any tone (see se.pat.misc)
0 or 1
silent
0
The tone pattern and vibration (1 to enable) for the alert played when the handset turns on.
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Parameter
Permitted Values
Default
np.silent.ringing.calls.tonePattern
A ringer (see se.pat.ringer)
ringer1
The ringtone (see se.pat.ringer) and vibration (1 to enable) for normal calls.
np.silent.ringing.calls.vibration
off, continuous, shortPulse,
longPulse
off
The ringtone (see se.pat.ringer) and vibration (1 to enable) for normal calls.
np.silent.ringing.oai1.tonePattern
A ringer (see se.pat.ringer)
ringer1
The ringtone (see se.pat.ringer) for Open Application Interface (OAI) communications.
np.silent.ringing.oai1.vibration
off, continuous, shortPulse,
longPulse
off
The vibration pattern for Open Application Interface (OAI) communications.
np.silent.ringing.oai2.tonePattern
A ringer (see se.pat.ringer)
ringer1
The ringtone (see se.pat.ringer) and vibration (1 to enable) for Open Application Interface (OAI) version 2.2
communications.
np.silent.ringing.oai2.vibration
off, continuous, shortPulse,
longPulse
off
The vibration pattern for Open Application Interface (OAI) version 2.2 communications.
np.silent.ringing.privateLine.tonePattern
default, ringer1 to ringer24
ringer1
The ringtone (see se.pat.ringer) for a private line registered to Microsoft Skype for Business or Microsoft Lync
Server 2013 or 2010.
np.silent.ringing.privateLine.vibration
off, continuous, shortPulse,
longPulse
shortPulse
The vibration pattern for a private line registered to Microsoft Skype for Business or Microsoft Lync Server 2013 or
2010.
np.silent.ringing.toneVolume.handset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Handset and Silent profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.silent.ringing.toneVolume.headset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Headset and Silent profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.silent.ringing.toneVolume.chassis
-1000 to 1000
0
The attribute is set (on adjusting ring volume) when ringing termination is Chassis and Silent profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.silent.ringing.toneVolume.dock
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when handset is at the speakerphone dock and Silent profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.silent.ringing.toneVolume.bluetoothHeadset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Bluetooth Headset and Silent profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.silent.ringing.toneVolume.reserved
-1000 to 1000
-21
Not currently used. Reserved for future use.
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Parameter
Permitted Values
Default
np.silent.ringing.toneVolume.usbHeadset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is a USB headset and Silent profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
Table 7-8: Meeting Profile Alert Parameters
Parameter
Permitted Values
Default
np.meeting.label
String
Meeting
Any tone (see se.pat.misc)
0 or 1
misc2
0
The name of the profile type.
np.meeting.alert.barcodeBeep.tonePattern
np.meeting.alert.barcodeBeep.vibration
The tone pattern and vibration (1 to enable) for the alert played when a barcode is scanned.
np.meeting.alert.docked.tonePattern
np.meeting.alert.docked.vibration
Any tone (see se.pat.misc)
0 or 1
postiveConfirm
0
The tone pattern and vibration (1 to enable) for the alert played if the handset is docked.
np.meeting.alert.undocked.tonePattern
np.meeting.alert.undocked.vibration
Any tone (see se.pat.misc)
0 or 1
negativeConfirm
0
The tone pattern and vibration (1 to enable) for the alert played if the handset is undocked.
np.meeting.alert.instantMessaging.tonePattern
np.meeting.alert.instantMessaging.vibration
Any tone (see se.pat.misc)
0 or 1
instantMessage
0
The tone pattern and vibration (1 to enable) for the instant message alert.
np.meeting.alert.localHoldNotification.tonePattern
np.meeting.alert.localHoldNotification.vibration
Any tone (see se.pat.misc)
0 or 1
localHoldNotification
0
The tone pattern and vibration (1 to enable) for the local hold notification alert.
np.meeting.alert.lossOfNetwork.tonePattern
np.meeting.alert.lossOfNetwork.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the network is lost.
np.meeting.alert.lowBattery.tonePattern
np.meeting.alert.lowBattery.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the battery is low.
np.meeting.alert.veryLowBattery.tonePattern
np.meeting.alert.veryLowBattery.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the battery is very low.
np.meeting.alert.messageWaiting.tonePattern
np.meeting.alert.messageWaiting.vibration
Any tone (see se.pat.misc)
0 or 1
messageWaiting
0
The tone pattern and vibration (1 to enable) for the alert played if there is a message waiting.
np.meeting.alert.negativeConfirm.tonePattern
np.meeting.alert.negativeConfirm.vibration
Any tone (see se.pat.misc)
0 or 1
negativeConfirm
0
The tone pattern and vibration (1 to enable) for the negative confirmation alert.
np.meeting.alert.positiveConfirm.tonePattern
np.meeting.alert.positiveConfirm.vibration
Any tone (see se.pat.misc)
0 or 1
positiveConfirm
0
The tone pattern and vibration (1 to enable) for the positive confirmation alert.
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Parameter
Permitted Values
Default
np.meeting.alert.pttTransmit.tonePatterna
np.meeting.alert.pttTransmit.vibration
Any tone (see se.pat.misc)
0 or 1
misc3
0
The tone pattern and vibration (1 to enable) for the alert played if sending a push-to-talk page.
np.meeting.alert.pttWait.tonePattern
np.meeting.alert.pttWait.vibration
Any tone (see se.pat.misc)
0 or 1
misc4
0
The tone pattern and vibration (1 to enable) for the push-to-talk wait alert.
np.meeting.alert.welcome.tonePattern
np.meeting.alert.welcome.vibration
Any tone (see se.pat.misc)
0 or 1
Welcome
0
The tone pattern and vibration (1 to enable) for the alert played when the handset turns on.
np.meeting.ringing.calls.tonePattern
A ringer (see se.pat.ringer)
ringer1
The ringtone (see se.pat.ringer) and vibration (1 to enable) for normal calls.
np.meeting.ringing.calls.vibration
off, continuous, shortPulse,
longPulse
continuous
The ringtone (see se.pat.ringer) and vibration (1 to enable) for normal calls.
np.meeting.ringing.oai1.tonePattern
A ringer (see se.pat.ringer)
ringer1
The ringtone (see se.pat.ringer) for Open Application Interface (OAI) communications.
np.meeting.ringing.oai1.vibration
off, continuous, shortPulse,
longPulse
continuous
The vibration pattern for Open Application Interface (OAI) communications.
np.meeting.ringing.oai2.tonePattern
A ringer (see se.pat.ringer)
ringer1
The ringtone (see se.pat.ringer) and vibration (1 to enable) for Open Application Interface (OAI) version 2.2
communications.
np.meeting.ringing.oai2.vibration
off, continuous, shortPulse,
longPulse
continuous
The vibration pattern for Open Application Interface (OAI) version 2.2 communications.
np.meeting.ringing.privateLine.tonePattern
default, ringer1 to ringer24
ringer9
The ringtone (see se.pat.ringer) for a private line registered to Microsoft Skype for Business or Microsoft Lync
Server 2013 or 2010.
np.meeting.ringing.privateLine.vibration
off, continuous, shortPulse,
longPulse
shortPulse
The vibration pattern for a private line registered to Microsoft Skype for Business or Microsoft Lync Server 2013 or
2010.
np.meeting.ringing.toneVolume.handset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Headset and Meeting profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.meeting.ringing.toneVolume.headset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Headset and Meeting profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.meeting.ringing.toneVolume.chassis
-1000 to 1000
0
The attribute is set (on adjusting ring volume) when ringing termination is Chassis and Meeting profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
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Parameter
Permitted Values
Default
np.meeting.ringing.toneVolume.dock
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when handset is at the speakerphone dock and Meeting profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.meeting.ringing.toneVolume.bluetoothHeadset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Bluetooth Headset and Meeting profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.meeting.ringing.toneVolume.reserved
-1000 to 1000
-21
-1000 to 1000
-21
Not currently used. Reserved for future use.
np.meeting.ringing.toneVolume.usbHeadset
The attribute is set (on adjusting ring volume) when ringing termination is a USB headset and Meeting profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
Table 7-9: Custom1 Profile Alert Parameters
Parameter
Permitted Values
Default
np.custom1.label
String
Custom1
Any tone (see se.pat.misc)
0 or 1
misc2
0
The name of the profile type.
np.custom1.alert.barcodeBeep.tonePattern
np.custom1.alert.barcodeBeep.vibration
The tone pattern and vibration (1 to enable) for the alert played when a barcode is scanned.
np.custom1.alert.docked.tonePattern
np.custom1.alert.docked.vibration
Any tone (see se.pat.misc)
0 or 1
postiveConfirm
0
The tone pattern and vibration (1 to enable) for the alert played if the handset is docked.
np.custom1.alert.undocked.tonePattern
np.custom1.alert.undocked.vibration
Any tone (see se.pat.misc)
0 or 1
negativeConfirm
0
The tone pattern and vibration (1 to enable) for the alert played if the handset is undocked.
np.custom1.alert.instantMessaging.tonePattern
np.custom1.alert.instantMessaging.vibration
Any tone (see se.pat.misc)
0 or 1
instantMessage
0
The tone pattern and vibration (1 to enable) for the instant message alert.
np.custom1.alert.localHoldNotification.tonePattern
np.custom1.alert.localHoldNotification.vibration
Any tone (see se.pat.misc)
0 or 1
localHoldNotification
0
The tone pattern and vibration (1 to enable) for the local hold notification alert.
np.custom1.alert.lossOfNetwork.tonePattern
np.custom1.alert.lossOfNetwork.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the network is lost.
np.custom1.alert.lowBattery.tonePattern
np.custom1.alert.lowBattery.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the battery is low.
np.custom1.alert.veryLowBattery.tonePattern
np.custom1.alert.veryLowBattery.vibration
Any tone (see se.pat.misc)
0 or 1
misc1
0
The tone pattern and vibration (1 to enable) for the alert played if the battery is very low.
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Parameter
Permitted Values
Default
np.custom1.alert.messageWaiting.tonePattern
np.custom1.alert.messageWaiting.vibration
Any tone (see se.pat.misc)
0 or 1
messageWaiting
0
The tone pattern and vibration (1 to enable) for the alert played if there is a message waiting.
np.custom1.alert.negativeConfirm.tonePattern
np.custom1.alert.negativeConfirm.vibration
Any tone (see se.pat.misc)
0 or 1
negativeConfirm
0
The tone pattern and vibration (1 to enable) for the negative confirmation alert.
np.custom1.alert.positiveConfirm.tonePattern
np.custom1.alert.positiveConfirm.vibration
Any tone (see se.pat.misc)
0 or 1
positiveConfirm
0
The tone pattern and vibration (1 to enable) for the positive confirmation alert.
np.custom1.alert.pttTransmit.tonePatterna
np.custom1.alert.pttTransmit.vibration
Any tone (see se.pat.misc)
0 or 1
misc3
0
The tone pattern and vibration (1 to enable) for the alert played if sending a push-to-talk page.
np.custom1.alert.pttWait.tonePattern
np.custom1.alert.pttWait.vibration
Any tone (see se.pat.misc)
0 or 1
misc4
0
The tone pattern and vibration (1 to enable) for the push-to-talk wait alert.
np.custom1.alert.welcome.tonePattern
np.custom1.alert.welcome.vibration
Any tone (see se.pat.misc)
0 or 1
Welcome
0
The tone pattern and vibration (1 to enable) for the alert played when the handset turns on.
np.custom1.ringing.calls.tonePattern
A ringer (see se.pat.ringer)
ringer2
The ringtone (see se.pat.ringer) and vibration (1 to enable) for normal calls.
np.custom1.ringing.calls.vibration
off, continuous, shortPulse,
longPulse
continuous
The ringtone (see se.pat.ringer) and vibration (1 to enable) for normal calls.
np.custom1.ringing.oai1.tonePattern
A ringer (see se.pat.ringer)
ringer2
The ringtone (see se.pat.ringer) for Open Application Interface (OAI) communications.
np.custom1.ringing.oai1.vibration
off, continuous, shortPulse,
longPulse
continuous
The vibration pattern for Open Application Interface (OAI) communications.
np.custom1.ringing.oai2.tonePattern
A ringer (see se.pat.ringer)
ringer2
The ringtone (see se.pat.ringer) and vibration (1 to enable) for Open Application Interface (OAI) version 2.2
communications.
np.custom1.ringing.oai2.vibration
off, continuous, shortPulse,
longPulse
continuous
The vibration pattern for Open Application Interface (OAI) version 2.2 communications.
np.custom1.ringing.privateLine.tonePattern
default, ringer1 to ringer24
ringer9
The ringtone (see se.pat.ringer) for a private line registered to Microsoft Skype for Business or Microsoft Lync
Server 2013 or 2010.
np.custom1.ringing.privateLine.vibration
off, continuous, shortPulse,
longPulse
shortPulse
The vibration pattern for a private line registered to Microsoft Skype for Business or Microsoft Lync Server 2013 or
2010.
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Parameter
Permitted Values
Default
np.custom1.ringing.toneVolume.handset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Headset and Custom1 profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.custom1.ringing.toneVolume.headset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Headset and Custom1 profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.custom1.ringing.toneVolume.chassis
-1000 to 1000
0
The attribute is set (on adjusting ring volume) when ringing termination is Chassis and Custom1 profile is active.
Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.custom1.ringing.toneVolume.dock
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when handset is at the speakerphone dock and Custom1 profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.custom1.ringing.toneVolume.bluetoothHeadset
-1000 to 1000
-21
The attribute is set (on adjusting ring volume) when ringing termination is Bluetooth Headset and Custom1 profile
is active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
np.custom1.ringing.toneVolume.reserved
-1000 to 1000
-21
-1000 to 1000
-21
Not currently used. Reserved for future use.
np.custom1.ringing.toneVolume.usbHeadset
The attribute is set (on adjusting ring volume) when ringing termination is a USB headset and Custom1 profile is
active. Although the permitted values are -1000 to 1000, the practical limits used by the handset are -50 to 10.
Time and Date Display
A clock and calendar are enabled by default. View/edit by navigating to Home> Settings>
Basic Settings> Preferences> Time & Date. You can display the time and date for your time
zone in several formats, or you can turn it off altogether. You can also set the time and date
format to display differently when the handset is in certain modes. For example, the display
format can change when the handset goes from idle mode to an active call.
There are multiple formats to this parameter and as a result there is a hierarchy of precedence.
Example: the device.sntp parameter would be considered a base level parameter whereas the
tcpipApp.sntp.address parameter takes precedence over the device.sntp parameter. Further still
DHCP can override both and take precedence. However, tcpipApp.sntp.address.overrideDHCP
can be configured to override DHCP.
The templates contain the parameter device.sntp in the wireless.cfg file. No other template file
contains any of these sntp base parameters. They are available in the everything.cfg.
In order to display the time and date accurately, you will have to synchronize the handset to the
Simple Network Time Protocol (SNTP) time server. Until a successful SNTP response is
received, the handset will continuously flash a time and date to indicate that they are not
accurate.
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Caution: Some language parameters may overrule lcl.dateline parameters.
Certain languages use date and time settings that overrule lcl.dateline parameters.
See the Languages section for details.
Table 7-10: Setting the Time and Date Display
Parameter
Permitted Values
Default
up.localClockEnabled
0 or 1
1
If 0, the date and time are not shown on the idle display. If 1, the date and time and shown on the idle display.
string which includes ‘D’, ‘d’ and
‘M’ and two optional commas
lcl.datetime.date.format
D,Md
Controls format of date string. D = day of week, d = day, M = month.
Up to two commas may be included.
For example: D,dM = Thursday, 3 July or Md,D = July 3, Thursday
The field may contain 0, 1 or 2 commas which can occur only between characters and only one at a time. For
example: “D,,dM” is illegal.
lcl.datetime.date.longFormat
0 or 1
1
If set to 1, display the day and month in long format (Friday/November), otherwise, use abbreviations (Fri/Nov).
lcl.datetime.time.24HourClock
0 or 1
0
If set to 1, display time in 24-hour clock mode rather than a.m./p.m.
Table 7-11: Date Formats
lcl.datetime.date.format
lcl.datetime.date.longformat
Date Displayed on Phone
dM,D
0
19 Aug, Fri
dM,D
1
19 August, Friday
Md,D
0
Aug 19, Fri
Md,D
1
August 19, Friday
D,dM
0
Fri, 19 Aug
D,dM
1
Friday, 19 August
D,Md
0
Fri, Aug 19
D,Md
1
Friday, August 19
DD/MM/YY
n/a
19/08/11
DD/MM/YYYY
n/a
19/08/2011
MM/DD/YY
n/a
08/19/11
MM/DD/YYYY
n/a
08/19/2011
YY/MM/DD
n/a
11/08/19
YYYY/MM/DD
n/a
2011/08/19
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Synchronizing with SNTP
The following table describes the Simple Network Time Protocol (SNTP) parameters used to set
up time synchronization and daylight savings time. The default values will enable and configure
daylights savings time (DST) for North America.
Daylight savings time defaults:
 Do not use fixed day, use first or last day of week in the month.
 Start DST on the second Sunday in March at 2am.
 Stop DST on the first Sunday in November at 2am.
Table 7-12: Simple Network Time Protocol (SNTP) Parameters
Parameter
Permitted Values
Default
tcpIpApp.sntp.address
Valid hostname or IP
address
Null
0 or 1
0
The address of the SNTP server.
tcpIpApp.sntp.address.overrideDHCP
If 0, the DHCP values for the SNTP server address will be used. If 1, the SNTP parameters will overrule the DHCP
values.
tcpIpApp.sntp.daylightSavings.enable
0 or 1
1
If 0, daylight savings time rules are not applied to the displayed time. If 1, the daylight savings rules apply.
tcpIpApp.sntp.daylightSavings.fixedDayEnable
0 or 1
0
If 0, month, date, and dayOfWeek are used in the DST calculation. If 1, only month and date are used.
tcpIpApp.sntp.daylightSavings.start.date
1 to 31
8
The start date for daylight savings time. If fixedDayEnable is set to 1, the value of this parameter is the day of
the month to start DST. If fixedDayEnable is set to 0, this value specifies the occurrence of dayOfWeek when
DST should start. Set 1 for the first occurrence in the month, set 8 for the second occurrence, 15 for the third
occurrence, or 22 for the fourth occurrence. For example, if set to 15, DST starts on the third dayOfWeek of the
month.
tcpIpApp.sntp.daylightSavings.start.dayOfWeek
1 to 7
1
The day of the week to start DST. 1=Sunday, 2=Monday, … 7=Saturday. Note: this parameter is not used if
fixedDayEnable is set to 1.
tcpIpApp.sntp.daylightSavings.start.dayOfWeek.lastInMonth
0 or 1
0
If 1, DST starts on the last dayOfWeek of the month and the start.date is ignored. Note: this parameter is not
used if fixedDayEnable is set to 1.
tcpIpApp.sntp.daylightSavings.start.month
1 to 12
3 (March)
0 to 23
2
The month to start DST. 1=January, 2=February… 12=December.
tcpIpApp.sntp.daylightSavings.start.time
The time of day to start DST – in 24 hour clock format. 0= 12am, 1= 1am,… 12= 12pm, 13= 1pm, … 23= 11pm.
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Parameter
Permitted Values
Default
tcpIpApp.sntp.daylightSavings.stop.date
1 to 31
1
The stop date for daylight savings time. If fixedDayEnable is set to 1, the value of this parameter is the day of
the month to stop DST. If fixedDayEnable is set to 0, this value specifies the occurrence of dayOfWeek when
DST should stop. Set 1 for the first occurrence in the month, set 8 for the second occurrence, 15 for the third
occurrence, or 22 for the fourth occurrence. For example, if set to 22, DST stops on the fourth dayOfWeek of the
month.
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek
1 to 7
1
The day of the week to stop DST. 1=Sunday, 2=Monday, … 7=Saturday. Note: this parameter is not used if
fixedDayEnable is set to 1.
tcpIpApp.sntp.daylightSavings.stop.dayOfWeek.lastInMonth
0 or 1
0
If 1, DST stops on the last dayOfWeek of the month and the stop.date is ignored. Note: this parameter is not
used if fixedDayEnable is set to 1.
tcpIpApp.sntp.daylightSavings.stop.month
1 to 12
11
0 to 23
2
The month to stop DST. 1=January, 2=February… 12=December.
tcpIpApp.sntp.daylightSavings.stop.time
The time of day to stop DST – in 24 hour clock format. 0= 12am, 1= 1am,… 12= 12pm, 13= 1pm, … 23= 11pm.
tcpIpApp.sntp.gmtOffset
positive or negative
integer
0
The offset in seconds of the local time zone from GMT.3600 seconds = 1 hour, -3600 seconds = -1 hour.
tcpIpApp.sntp.gmtOffset.overrideDHCP
0 or 1
0
If 0, the DHCP values for the GMT offset will be used. If 1, the SNTP values for the GMT offset will be used.
tcpIpApp.sntp.resyncPeriod
positive integer
86400
The period of time (in seconds) that passes before the handset resynchronizes with the SNTP server. Note: 86400
seconds is 24 hours.
User Preferences Parameters
Options on various user menus can be pre-configured by the administrator.
Table 7-13: User Preferences Parameters
Parameter
Permitted Values
Default
up.backlight.idleIntensity
0, 1, 2, or 3
0
The brightness of the LCD backlight when the handset is idle. 0 – off, 1 – low, 2 – medium, 3 – high. Note: If this is
higher than the active backlight brightness (onIntensity), the active backlight brightness is used. When the
phone is in the charger, this parameter does not take effect, the backlight is always on.
up.backlight.onIntensity
0, 1, 2, or 3
3
The brightness of the LCD backlight when the handset is active (in use). 0: off, 1 – low, 2 – medium, 3 – high
up.backlight.timeout
5 to 60
10
The number of seconds to wait before the backlight dims from the active intensity to the idle intensity.
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Parameter
Permitted Values
Default
up.hearingAidCompatibility.enabled
0 or 1
0
If set to 1, the handset audio Rx (receive) equalization is disabled for hearing aid compatibility. If 0, audio Rx
equalization is enabled.
up.idleTimeout1
0 to 65535, seconds
40
The number of seconds that the handset can be idle before automatically leaving a menu and showing the idle
display. If 0, there is no timeout and the handset does not automatically exit to the idle display.
up.numberFirstCID1
0 or 1
0
If 0, the caller ID display will show the caller’s name first. If 1, the caller’s handset number will be shown first.
up.onHookDialingEnabled
0 or 1
1
If 0, on hook dialing is disabled. If 1, on-hook dialing is enabled.
up.screenCapture.enabled1
0 or 1
0
If 0, screen captures are disabled. If 1, the user can enable screen captures from the Screen Capture menu on the
handset. Note: when the handset reboots, screen captures are disabled from the Screen Capture menu on the
handset.
up.simplifiedSipCallInfo
0 or 1
0
If 1, the displayed host name is trimmed for both incoming and outgoing calls and the protocol tag/information is
not displayed for incoming and outgoing calls.
up.warningLevel1
0 to 2
0
If 0, the handset’s warning icon and a pop-up message display on the handset for all warnings. If 1, the warning
icon and pop-up messages are only shown for critical warnings. If 2, no warnings are displayed. Note: All warnings
are listed in the Warnings menu (navigate to Menu> Status> Diagnostics> Warnings on the handset).
up.welcomeSoundEnabled1
0 or 1
1
If 0, the welcome sound is disabled. If 1, the welcome sound is enabled and played each time the handset reboots.
up.welcomeSoundOnWarmBootEnabled1
0 or 1
0
If 0, the welcome sound is played when the handset powers up (cold boot), but not after it restarts or reboots
(warm boot). If 1, the welcome sound plays each time the handset powers up, reboots, or restarts.
1
Change causes handset to restart or reboot.
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Chapter 8: Features Configured by the
Administrator
Certain features are entirely set by the administrator and for the most part the user cannot
change them. However, some have user-configurable options available on the user menus.
Features of this type are covered in this chapter.
Finding the Parameters in the Config files
The parameters detailed in this chapter are mostly found in the “everything.cfg” file
located in the troubleshooting folder in the Config folder that you download with the
software starting with Spectralink software version 4.2.x.
AutoComplete List
The autocomplete list displays when the user goes off hook and when transferring a call. It is
composed of your Call List and Contact Directory entries, sorted alphabetically.
When the autocomplete list is presented, the user can type in numbers or letters (using the Dial
mode softkey to switch between modes) to get a list of matches.
The default method of finding matches is to match the entered characters to the starting
characters in the fields checked following this logic:

For the Call List entries, the handset checks for matches within the name and the
contact fields.

For the Contact Directory items, the handset checks for matches within the first name,
last name, and contact fields.
The search is always case insensitive.
The administrator can configure the handset to have the search find matches that “contain” the
entered characters.
Parameter
Permitted Values
Default
autoComplete.useContainsSearch
0 or 1
0
When set to 0, the search will compare the entered characters against the starting characters of each field
searched. This is the default behavior. When set to 1, the search will look for the entered characters anywhere
inside each field searched.
Note that the handset limits the search to a certain period of time (because otherwise it can
interfere with entering letters in the search field) so not all possible matches may be shown in
the list after the user enters a single letter or number. This is especially true when using a
“contains” search. The user might need to enter 2 or 3 characters to narrow down the search
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enough for the handset to show all the possible matches. This should only be necessary when
very large lists are present.
Audio Settings
Context Sensitive Volume Control
The parameters shown below enable you to adjust the volume of handset sound effects — such
as the ringer and the volume of receiving call audio — separately for the speakerphone,
handset, and headset. While transmit levels are fixed according to the TIA/EIA-810-A standard,
you can adjust the receive volume.
In some countries, regulations state that a handset’s receiver volume must be reset to a nominal
level for each new call. This is the handset’s default behavior. Using this parameter, you can set
the receiver volume to persist across calls each time a user makes changes to the default
volume level.
<voice.volume/>
Table 8-1: Context Sensitive Volume Control
Parameter
Permitted Values
Default
voice.volume.persist.bluetooth.headset1
0 or 1
0
If 0, the Bluetooth headset receive volume will automatically reset to a nominal level after each call. If
1, the volume for each call will be the same as the previous call.
voice.volume.persist.handset1
0 or 1
0
If 0, the handset receive volume will automatically reset to a nominal level after each call. If 1, the
volume for each call will be the same as the previous call.
voice.volume.persist.headset1
0 or 1
0
If 0, the headset receive volume will automatically reset to a nominal level after each call. If 1, the
volume for each call will be the same as the previous call.
voice.volume.persist.handsfree1
0 or 1
1
If 0, the speakerphone receive volume will automatically reset to a nominal level after each call. If 1,
the volume for each call will be the same as the previous call.
voice.volume.persist.usb.handsfree1
0 or 1
1
If 0, the USB headset receive volume will automatically reset to a nominal level after each call. If 1, the
volume for each call will be the same as the previous call.
voice.volume.persist.usbHeadset1
0 or 1
0
If 0, the USB headset receive volume will automatically reset to a nominal level after each call. If 1, the
volume for each call will be the same as the previous call.
1
Change causes handset to restart or reboot.
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<voice/>
The <voice/> parameter controls the settings related to the audio on the handset.
Table 8-2: Voice Parameters
Parameter
Permitted Values
Default
voice.txPacketDelay1
low, normal, Null
Null
If set to normal or Null, no audio parameters are changed.
If set to low and there are no precedence conflicts, the following changes are made:
• voice.codecPref.G722="1"
• voice.codecPref.G711_Mu="2"
• voice.codecPref.G711_A="3"
• voice.codecPref.<OtherCodecs>=""
• voice.audioProfile.G722.payloadSize="10"
• voice.audioProfile.G711Mu.payloadSize= "10"
• voice.audioProfile. G711A.payloadSize= "10"
• voice.aec.hs.enable="0"
• voice.ns.hs.enable="0"
voice.txPacketFilter1
0 or 1
0
If 0, no Tx filtering is performed. If 1, narrowband Tx high pass filter is enabled.
1
Change causes handset to restart or reboot.
<rxQoS/>
The following table lists the jitter buffer parameters for wired network interface voice traffic,
wireless network interface voice traffic, and push-to-talk interface voice traffic.
Caution: Do not change these settings.
Changing any rxQoS settings can impair system operation. Do not change any
rxQoS parameters without prior consultation with Spectralink Technical Support.
Table 8-3: Voice Jitter Buffer Parameters
Parameter
Permitted Values
Default
voice.rxQoS.avgJitter1
0 to 80
20
voice.rxQoS.maxJitter1
0 to 200
160
The average and maximum jitter in milliseconds for wired network interface voice traffic.
avgJitter – The wired interface minimum depth will be automatically configured to adaptively handle this level of
continuous jitter without packet loss.
maxJitter – The wired interface jitter buffer maximum depth will be automatically configured to handle this level
of intermittent jitter without packet loss.
Actual jitter above the average but below the maximum may result in delayed audio play out while the jitter buffer
adapts, but no packets will be lost. Actual jitter above the maximum value will always result in packet loss. Note
that if legacy voice.audioProfile.x.jitterBuffer.* parameters are explicitly specified, they will be used
to configure the jitter buffer and these voice.rxQoS parameters will be ignored.
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Parameter
Permitted Values
Default
voice.rxQoS.wireless.avgJitter1
0 to 200
70
voice.rxQoS.wireless.maxJitter1
20 to 500
300
The average and maximum jitter in milliseconds for wireless network interface voice traffic.
avgJitter – The wireless interface minimum depth will be automatically configured to adaptively handle this
level of continuous jitter without packet loss.
maxJitter – The wireless interface jitter buffer maximum depth will be automatically configured to handle this
level of intermittent jitter without packet loss.
Actual jitter above the average but below the maximum may result in delayed audio play out while the jitter buffer
adapts, but no packets will be lost. Actual jitter above the maximum value will always result in packet loss.
Note: if legacy voice.audioProfile.x.jitterBuffer.* parameters are explicitly specified, they will be
used to configure the jitter buffer and these voice.rxQoS parameters will be ignored for wireless interfaces.
voice.rxQoS.ptt.avgJitter1
0 to 200
150
voice.rxQoS.ptt.maxJitter1
20 to 500
480
The average and maximum jitter in milliseconds for IP multicast voice traffic (wired or wireless).
avgJitter – The PTT/Paging interface minimum depth will be automatically configured to adaptively handle this
level of continuous jitter without packet loss.
maxJitter – The PTT/Paging interface jitter buffer maximum depth will be automatically configured to handle this
level of intermittent jitter without packet loss.
Actual jitter above the average but below the maximum may result in delayed audio play out while the jitter buffer
adapts, but no packets will be lost. Actual jitter above the maximum value will always result in packet loss.
Note: if legacy voice.audioProfile.x.jitterBuffer.* parameters are explicitly specified, they will be
used to configure the jitter buffer and these voice.rxQoS parameters will be ignored for PTT/Paging interface
interfaces.
1
Change causes handset to restart or reboot.
Automatic Off-Hook Call Placement
You can configure the handset to automatically place a call to a specified number when you go
off-hook. This feature is sometimes referred to as Hot Dialing. The handset goes off-hook when
you lift the handset, press the New Call softkey, or press the headset or speakerphone buttons
on the handset. You can specify an off-hook call contact and enable or disable the feature for
specific line registrations.
Table 8-4: Enabling Automatic Off-Hook Call Placement
Parameter
Permitted Values
Default
call.autoOffHook.x.enabled1
0 or 1
0
Enable or disable the feature
If enabled is set to 0, no call is placed automatically when the handset goes off hook, and the other parameters are
ignored. If enabled is set to 1, a call is automatically placed to the contact.
call.autoOffHook.x.contact1
a SIP URL
Null
The contact address to where the call is placed
The contact must be an ASCII-encoded string containing digits, either the user part of a SIP URL (for example,
6416), or a full SIP URL (for example, 6416@Spectralink.com).
1
Change causes handset to restart or reboot.
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Background Images
The Spectralink 84-Series wireless handsets include a feature that enables you to add a custom
background using a digital image. Using a digital image enables you to display a company logo
or product brand as the background on your handset. Supported graphics files are PNG, JPEG,
or BMP images. The maximum supported is size is 240x270 pixels. Progressive or multiscan
JPEG images are not supported.
After one or more backgrounds have been provisioned, the user can select one of them by
navigating to Menu> Settings> Basic> Preferences> Background.
Choosing a Graphic Display Background
Depending on the image you use, the graphic display background may affect the
visibility of text and numbers on the handset screen. As a general rule,
backgrounds should be light in shading for better handset and feature usability.
Configuring Background Images
You may want to define a set of images that the user can select from and set one of them as the
default. This feature is commonly used to set the company logo as a background or to
distinguish handsets that are provisioned for specific purposes or in specific groups. Limit the
configurable options by limiting the number of images available.
Table 8-5: Background Parameters
Parameter
Permitted Values
Default
bg.color.selection
w,x
1,1
Set the background. Specify which type of background (w) and index (x) for that type is selected on reboot. The
default selection is 1,1 the first solid background.
Use w=1 and x=1 (1,1) to select the built-in image.
Use w=3 and x= 1 to 6 to select one of the six background bm images
bg.color.bm.x.name
Phone screen background image file
URL or file path of a BMP or JPEG
image
null
The name of the image file (including extension).
Example configuration
In this example, four of the graphic images are located in the root directory of the central
provisioning server. The fifth image, Jellyfish.jpg, is located in a subdirectory. Each of these
images will appear on the Background menu and is user-selectable.
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The figure shown next is an example of a digital image background on a Spectralink 84-Series
handset.
Backlight Off while Phone is Charging
The administrator can configure the phone so that the backlight turns off when the phone is in
the desktop charger. This configuration parameter is available in the everything.cfg file.
Parameter
up.backlight.onAcCharger.enabled
Modification
Default
Added
1
If 1 after an idle timeout display backlight is always ON to low intensity.
If 0 after an idle timeout display backlight will be OFF.
Battery End-of-life Alert
When a battery degrades through normal use, it will eventually not last as long between
charges. When a worn out battery is detected a warning message will appear on the display
which the user can clear. A warning alert icon will remain in the status bar until the battery is
replaced.
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These configuration parameters are available in the everything.cfg file.
Parameter
battery.check.fullCapacity1100Min
battery.check.fullCapacity1750Min
Modification
Added
Added
Min
Max
Default
0
0
9999
9999
0
0
If the customer decides to screen batteries, these are the recommended settings:
For 1100 (standard batteries) the recommended value is 1010.
For 1750 (extended batteries) the recommended value is 1600.
If set above the recommended value, perfectly functional batteries might be detected as worn out. If set below the
recommended setting, worn out batteries might not be detected at all.
The default of 0 ensures that batteries will not be screened at all.
When a worn out battery is detected a warning message will appear on the display and a warning alert icon will
remain in the status bar until the battery is replaced.
Feature and Basic Settings Menu Password
Certain installations need to restrict access to options on the Settings menu. The Advanced
Settings option is already behind an admin password. These parameters allow you to require a
password to access Basic Settings and Feature Settings.
Parameter
Permitted values
Defaul
settingsLock.basicSettingsPassword
String (1-32 characters)
null
If set, the indicated password is required to enter the Basic Settings menu. Also causes Edit item to be removed
from the Profiles softkey flyout menu. This allows user to change which profile is current, but not modify the
settings for each profile. Defaults to Null which means no password required.
settingsLock.featureSettingsPassword
String (1-32 characters)
Null
If set, the indicated password is required to enter the Feature Settings menu. Also causes the Forward item to be
removed from the Features softkey flyout menu. Defaults to Null which means no password required. Note that the
Feature Settings menu has DND, Forward and also the Microsoft Skype for Business Signin/Signout menus.
However, the Skype for Business Signin/Signout items can be configured to be on the Features softkey flyout
menu so one can always signin and signout.
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Parameter
Permitted values
Defaul
settingsLock.disallowProfileSoftkey
0 or 1
0
If 1, the Profile softkey will not be shown at all. Default is 0 which is current behavior.
Call Hold
The 84-Series handset has local hold functionality. Rarely, a server may not recognize this
functionality and the call can be lost. If this situation occurs, a server-based call hold feature
may be required and the parameters described in this section can be used.
The purpose of call hold is to pause activity on one call so that you can use the handset for
another task, for example, to place or receive another call or to search your handset’s menu for
information. When you place an active call on hold, a message will inform the held party that
they are on hold. You can also configure a call hold alert to remind you after a period of time
that a call is still on hold.
Table 8-6: Enabling Call Hold
Parameter
Permitted Values
Default
voIpProt.SIP.useRFC2543hold
0 or 1
0
If set to 0, use SDP media direction parameters (such as a=sendonly) per RFC 3264 when initiating a call.
Otherwise use the obsolete c=0.0.0.0 RFC2543 technique. In either case, the handset processes incoming hold
signaling in either format.
Note: voIpProt.SIP.useRFC2543hold is effective only when the call is initiated.
voIpProt.SIP.useSendonlyHold
0 or 1
1
If set to 1, the handset will send a reinvite with a stream mode parameter of “sendonly” when a call is put on hold.
This is the same as the previous behavior.
If set to 0, the handset will send a reinvite with a stream mode parameter of “inactive” when a call is put on hold.
NOTE: The handset will ignore the value of this parameter if set to 1 when the parameter
voIpProt.SIP.useRFC2543hold is also set to 1 (default is 0).
call.hold.localReminder.enabled1
0 or 1
0
If 1, users are reminded of calls that have been on hold for an extended period of time. If 0, there is no hold
reminder.
call.hold.localReminder.period1
non-negative integer
60
non-negative integer
90
a SIP URI
Null
Specify the time in seconds between subsequent hold reminders.
call.hold.localReminder.startDelay1
Specify a time in seconds to wait before the initial hold reminder.
voIpProt.SIP.musicOnHold.uri
A URI that provides the media stream to play for the remote party on hold. This parameter is used if
reg.x.musicOnHold.uri is Null.
Note: The SIP URI parameter transport is supported when configured with the values of UDP, TCP, or TLS.
1
Change causes handset to restart or reboot.
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Example Call Hold Configuration
The following two illustrations show a sample configuration for the call hold feature. In the first
illustration, the three localReminder.* parameters have been configured to play a tone to
remind you of a party on hold, that the tone will begin to play 45 seconds after you put a party
on hold, and that the tone will repeat every 30 seconds.
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In the second illustration, the musicOnHold.uri parameter has been configured so the party
on hold will hear music played from SIP URI moh@example.com.
Call Handling Features
The call handling features described in this section require support from a SIP server and setup
of these features depend on the SIP server. For example, while some SIP servers implement
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group call pick-up using a particular star-code sequence, others implement the feature using
network signaling.
The Spectralink 84-Series handsets can bypass some SIP server requirements and implement
various call handling features using Enhanced Feature Keys (EFK) instead. Please refer to the
EFK section for more information.
Call Park and Retrieve
You can park an active call and retrieve parked calls from any handset. Whereas call hold
keeps the held call on the same line, call park moves the call to a separate address where the
call can be retrieved by any handset.
Table 8-7: Configuring Call Park and Retrieve
Parameter
Permitted Values
Default
feature.callPark.enabled1
0 or 1
0
If 0, the call park and call retrieve features are disabled. If 1, the features are enabled.
1
Change causes handset to restart or reboot.
Call Waiting Alerts
By default, the handset will alert you to incoming calls while you are in an active call.
Parameter
Permitted Values
Default
call.callWaiting.enable
0 or 1
1
If 1, the handset alerts you to an incoming call while you are in an active call. If 0, you are not alerted to incoming
calls while in an active call and the incoming call is treated as if you did not answer it. If 1, and you end the active
call during a second incoming call, you are alerted to the second incoming call.
call.callWaiting.ring1
beep, ring, silent
beep
Specifies the ringtone of incoming calls when another call is active. If set to Null, the default value is beep.
1
Change causes handset to restart or reboot.
Calling Party Identification
By default, the handset displays the identity of incoming callers if available to the handset
through the network signal. If the caller is in the contact directory, you can choose to display that
name instead. Note that the handset cannot match the identity of calling parties to entries in the
Corporate Directory.
Table 8-8: Configuring Calling Party Identification
Parameter
Permitted Values
Default
up.useDirectoryNames1
0 or 1
1
If 0, names provided through network signaling are used for caller ID. If 1, the name field in the local contact
directory will be used as the caller ID for incoming calls from contacts in the local directory. Note: Outgoing calls and
corporate directory entries are not matched.
1
Change causes handset to restart or reboot.
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Missed Call Notification
By default, the missed call notification displays on the 84-Series handset’s status bar. A counter
shows the number of missed calls. The counter is reset by viewing the Missed Calls list on the
handset.
The Missed Call Notification can be disabled for each registered line on a handset.
Table 8-9: Disabling Missed Call Notification
Parameter
Permitted Values
Default
call.missedCallTracking.x.enabled1
0 or 1
1
If set to 1, missed call tracking is enabled.
If call.missedCallTracking.x.enabled is set to 0, then missedCall counter is not updated regardless of
what call.serverMissedCalls.x.enabled is set to (and regardless of how the server is configured). There is
no Missed Call List provided under Menu> Features of the handset.
If call.missedCallTracking.x.enabled is set to 1 and call.serverMissedCalls.x.enabled is set to 0, then the
number of missedCall counter is incremented regardless of how the server is configured.
If call.missedCallTracking.x.enabled is set to 1 and call.serverMissedCalls.x.enabled is set to
1, then the handling of missedCalls depends on how the server is configured.
1
Change causes handset to restart or reboot.
Example Missed Call Notification Configuration
In the following example, the missed call counter is enabled by default for registered lines 1 and
2, and only server-generated missed calls will be displayed on line 1.
Call Transfer
Two types of call transfer behavior is available on the 84-Series handsets. When Party A is
talking to Party B and wishes to transfer the call to Party C:
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
Blind Transfer Party A calls Party C and presses the Transfer softkey while the call is
ringing. The call between Party A and Party B is ended when the Transfer softkey is
pressed.

Consultative Transfer Party A calls Party C while Party B is on hold. The call between
Party A and Party B is ended when Party A presses the Transfer softkey.
By default, the Transfer softkey uses the consultative transfer functionality and the blind transfer
is available on the Features softkey menu. This parameter allows you to change the Transfer
softkey to use the blind transfer functionality.
Table 8-10: Using Call Transfer
Parameter
Permitted Values
Default
call.transfer.blindPreferred
0 to 1
0
If set to 1, the default softkey will be for blind transfer.
If set to 0, the default softkey will be for consultative transfer.
The other method is always available on the Features softkey menu when the handset is in call.
Example Call Transfer Configuration
In the following example configuration, the parameter allowTransferOnProceeding has
been disabled so that the Transfer softkey will not display while the third-party handset is
ringing, the proceeding state. Once you have connected to the third-party, the Transfer softkey
will display. If the third-party does not answer, you can press the Cancel softkey to return to the
active call.
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Call Lists
Table 8-11: Call List (Call Log) Parameters
Parameter
Permitted Values
Default
feature.callList.enabled1
All locally controlled call lists.
feature.callListMissed.enabled1
The missed calls list.
feature.callListPlaced.enabled1
The placed calls list.
feature.callListReceived.enabled1
The received calls list.
0 or 1
1
0 or 1
1
0 or 1
1
0 or 1
1
If 0, the call list is disabled. If 1, the call list is enabled. To enable the Missed, Placed, or Received call lists,
feature.callList.enabled must be enabled.
callLists.grouping
Unified, InOut,
InOutMissed
Unified
Used by callLists.collapseDuplicates and callLists.size.
Unified - apply the limit in callLists.size to the total number of call list entries, regardless of their type.
InOut - apply the limit to incoming (Received + Missed) calls, and apply it separately to outgoing (Placed) calls.
Thus, the total number of call list entries is actually twice that specified in callLists.size.
InOutMissed - apply the limit to Missed calls, and then separately to Placed Calls, and then separately again to
Received calls. Thus, the total number of allowed call list entries is three times that specified in callLists.size.
callLists.collapseDuplicates
0 or 1
1
If 0, all calls are archived and presented in the call lists. If 1, consecutive incomplete calls between the same party
in the same direction (outgoing/incoming) are collapsed into one record with the most recent call displaying.
callLists.logConsultationCalls
0 or 1
0
If 1, all consultation calls are logged. (Calls made to a third party—while the original party is on hold—when
settings up a conference call are called consultation calls.)
If 0, consultation calls are not logged.
callLists.size
10 to 99
99
The maximum number of retained records of each type (incoming, outgoing, and missed). When the maximum
number is reached, new records will overwrite existing records. You can clear the list using the handset’s menu
system. If you want to prevent the records from uploading to the provisioning server, enter a false URL in the
CALL_LISTS_DIRECTORY field in the master configuration file.
callLists.writeDelay.journal
1 to 600
5
The delay (in seconds) before changes due to an in-progress call are flushed to the file system as a journal.
callLists.writeDelay.terminated
10 to 600
60
The minimum period between writing out the complete XML file to the local file system and, optionally, to the
provisioning server.
1
Change causes handset to restart or reboot.
Miscellaneous Call Handling Parameters
The handset supports an optional per-registration feature that enables automatic call placement
when the handset is off-hook.
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The handset supports a per-registration configuration that determines which events will cause
the missed-calls counter to increment.
You can enable/disable missed call tracking on a per-line basis.
Note: Reading the Call Parameter Table
In the following table, x is the registration number. For the Spectralink 84-Series
handsets, x=6.
This per-site and per-handset configuration parameters are defined as follows:
Table 8-12: Call Parameters
Parameter
Permitted Values
Default
feature.ringDownload.enabled1
0 or 1
1
If 0, the handset will not download ringtones when it starts up. If 1, the handset will download ringtones when it
starts up.
feature.nonVolatileRingerVolume.enabled
0 or 1
1
If 0, user changes to the ringer volume are reset to default when the handset reboots. If 1, user changes to the
ringer volume are saved and maintained when the handset reboots.
call.autoAnswer.micMute
0 or 1
1
If 0, the microphone is active immediately after a call is auto-answered. If 1, the microphone is initially muted after
a call is auto-answered.
call.autoAnswer.ringClass
see the list of ring classes.
ringAutoAnswer
The ring class to use when a call is to be automatically answered using the auto-answer feature. If set to a ring
class with a type other than answer or ring-answer, the setting will be overridden such that a ringtone of
visual (no ringer) applies.
call.dialtoneTimeOut1
positive integer
60
The time is seconds that a dial tone will play before a call is dropped. If set to 0, the call is not dropped.
call.enableOnNotRegistered1
0 or 1
1
If 1, users can make calls when the handset is not registered. If 0, calls are not permitted without registration.
call.offeringTimeOut1
positive integer
60
Specify a time in seconds that an incoming call will ring before the call is dropped, 0=infinite.
Note: The call diversion, no answer feature will overrule this feature if enabled.
call.ringBackTimeOut1
positive integer
60
Specify a time in seconds to allow an outgoing call to remain in the ringback state before dropping the call,
0=infinite.
call.stickyAutoLineSeize1
0 or 1
0
If set to 1, the handset uses sticky line seize behavior. This will help with features that need a second call object to
work with. The handset will attempt to initiate a new outgoing call on the same SIP line that is currently in focus on
the LCD (this was the behavior in SIP 1.6.5). Dialing through the call list when there is no active call will use the
line index for the previous call. Dialing through the call list when there is an active call will use the current active
call line index. Dialing through the contact directory will use the current active call line index.
If set to 0, the feature is disabled (this was the behavior in SIP 1.6.6). Dialing through the call list will use the line
index for the previous call. Dialing through the contact directory will use a random line index.
Note: This may fail due to glare issues in which case the handset may select a different available line for the call.
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Parameter
Permitted Values
Default
call.stickyAutoLineSeize.onHookDialing1
0 or 1
0
If call.stickyAutoLineSeize is set to 1, this parameter has no effect. The regular stickyAutoLineSeize
behavior is followed.
If call.stickyAutoLineSeize is set to 0 and this parameter is set to 1, this overrules the stickyAutoLineSeize
behavior for hot dial only. (Any new call scenario seizes the next available line.)
If call.stickyAutoLineSeize is set to 0 and this parameter is set to 0, there is no difference between hot
dial and new call scenarios.
Note: A hot dial occurs on the line which is currently in the call appearance. Any new call scenario seizes the next
available line.
call.suppressFullUrlDisplay.enabled1
0 or 1
0
This parameter affects the display on the connected in-call screen.
If 0, the handset displays the full SIP URL of an incoming call when the call is from a server the phone is not
registered to.
If 1, the handset displays only the number of an incoming call when the call is from a server the phone is not
registered to.
call.suppressIgnoreSoftkey1
0 or 1
0
If this is 1, the "Ignore" softkey will not show up on the incoming call screen. Instead, the softkey will be blank.
1
Change causes handset to restart or reboot.
CMS 2.0
Configuration Management Software is a UI administrative tool to accelerate configuration of
individual handsets and facilitate management of all handsets in a facility. It is a licensed
product accessible through an account key which must be present on every handset.
Two parameters ensure that the handset’s heartbeat can reach CMS and that CMS recognizes
the handset. Both of these parameters must set to a non-Null value for the CMS heartbeat to be
enabled.
Parameter
Permitted Values
Default
cms.heartbeat.URL
A string of 0-256 characters
Null
This is the URL to which the phone sends the heartbeats. It must be HTTPS.
cms.heartbeat.accountKey
A string of 0-100 characters
Null
The unique id that identifies the customer. (Provided by Spectralink)
Conference Calls
Local conferences require a host handset, which processes the audio of all parties. All handsets
support three-party local conferencing.
Parameter
Permitted Values
Default
call.localConferenceEnabled1
0 or 1
1
If set to 0, the Conference and Join softkeys do not display during an active call and you cannot establish
conferences on the handset.
If set to 1, the Conference and Join softkeys display during an active call and you can establish conferences on
the handset.
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Parameter
Permitted Values
call.transferOnConferenceEnd1
Default
0 or 1
1
The behavior when the conference host exits a conference. If 0, all parties are disconnected when the conference
host exits the conference. If 1, the other parties are left connected when the host exits the conference.
call. localConferenceCallHold1
0 or 1
0
If set to 0, a hold will happen for all legs when conference is put on hold.
If set to 1, only the host is out of the conference, all other parties in conference continue to talk.
1
Change causes handset to restart or reboot.
Corporate Directory
You can connect your handset to a corporate directory server that supports the Lightweight
Directory Access Protocol (LDAP) version 3. The corporate directory is a flexible feature and
provides you with the parameters you can configure. Once set up on the handsets, the
corporate directory can be browsed or searched. The user can call numbers and save entries
retrieved from the LDAP server to the local contact directory on the handset.
Spectralink 84-Series handsets currently support the following LDAP servers:
 Microsoft® Active Directory 2003 SP2
 Sun ONE Directory Server 5.2 p6
 Open LDAP Directory Server 2.4.12
 Microsoft Active Directory Application Mode (ADAM) 1.0 SP1
Spectralink handsets support corporate directories that support server-side sorting and those
that do not. For handsets that do not support server-side sorting, sorting is performed on the
handset.
Tip: Better Performance With Server-Side Sorting
Spectralink recommends using corporate directories that have server-side sorting
for better performance. Consult your LDAP Administrator when making any
configuration changes for the corporate directory. For more information on LDAP
attributes, see RFC 4510 - Lightweight Directory Access Protocol (LDAP):
Technical Specification Road Map.
Web Info: Supported LDAP Directories
Configuration of a corporate directory depends on the LDAP server you use. For
detailed explanations and examples of all currently supported LDAP directories,
see Technical Bulletin CS-14-19 Corporate Directory Best Practices.
A portion of the corporate directory is stored in flash memory on the handset. Spectralink
84-Series handsets have 256Mb of flash memory.
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Table 8-13: Using the Corporate Directory
Parameter
Permitted Values
feature.corporateDirectory.enabled
0 or 1
Default
0
If 0, the corporate directory feature is disabled. If 1, the feature is enabled.
dir.corp.address1
dotted-decimal IP address or
hostname or FQDN
Null
The IP address or hostname of the LDAP server interface to the corporate directory. For example,
host.domain.com.
dir.corp.attribute.x.filter1
UTF-8 encoded string
Null
The filter string for this parameter, which is edited when searching.
dir.corp.attribute.x.label1
UTF-8 encoded string
Null
UTF-8 encoded string
Null
The label when data is displayed.
dir.corp.attribute.x.name1
The name of the parameter to match on the server. Each name must be unique; however, an LDAP entry can
have multiple parameters with the same name. Up to eight parameters can be configured (x = 1 to 8).
dir.corp.attribute.x.searchable1
0 or 1
0
If 0, quick search on parameter x (if x is 2 or more) is disabled. If 1, quick search on x (if x is 2 or more) is enabled.
dir.corp.attribute.x.sticky1
0 or 1
0
If 0, the filter criteria for attribute x is reset after a reboot. If 1, the filter criteria are retained through a reboot. If you
set an attribute to be sticky (set this parameter to 1), a ‘*’ will display before the label of the attribute on the
handset.
dir.corp.attribute.x.type1
first_name, last_name,
phone_number, SIP_address,
H323_address URL, other
last_name
Defines how parameter x is interpreted by the handset. Entries can have multiple parameters of the same type.
The value other is used for display purposes only.
If the user saves the entry to the local contact directory on the handset, first_name, last_name, and
phone_number are copied. The user can place a call to the phone_number and SIP_address from the
corporate directory.
dir.corp.autoQuerySubmitTimeout1
0 to 60 seconds
0
The timeout (in seconds) between when the user stops entering characters in the quick search and when the
search query is automatically submitted. If 0, there is no timeout (automatic submit is disabled).
dir.corp.backGroundSync1
0 or 1
0
If 0, background downloading from the LDAP server is disabled. If 1, background downloading is enabled.
dir.corp.backGroundSync.period1
3600 to 604800
86400
The corporate directory cache is refreshed after the corporate directory feature has not been used for this period of
time in seconds. The default period is 24 hours (86400 seconds). The minimum is 1 hour and the maximum is 7
days.
dir.corp.baseDN1
UTF-8 encoded string
Null
The base domain name. This is the starting point for making queries on the LDAP server.
dir.corp.bindOnInit1
0 or 1
1
If 0, do not use bind authentication on initialization. If 1, use bind authentication on initialization.
dir.corp.cacheSize1
8 to 256
128
The maximum number of entries that can be cached locally on the handset.
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Parameter
Permitted Values
Default
dir.corp.filterPrefix1
UTF-8 encoded string
(objectclass=person)
8 to 64
32
The predefined filter string for search queries.
dir.corp.pageSize1
The maximum number of entries requested from the corporate directory server with each query.
dir.corp.password1
UTF-8 encoded string
Null
The password used to authenticate to the LDAP server.
dir.corp.port1
0, 1 to 65535
389 (TCP) 636 (TLS)
The port that connects to the server if a full URL is not provided. When the value is set to 0, the default value will
be used.
dir.corp.scope1
one, sub, base
sub
The type of search that is performed. If one, a search of one level below the base domain name (DN). If sub, a
recursive search of all levels below the base DN. If base, a search at the base DN level.
dir.corp.sortControl1
0 or 1
0
Controls how a client can make queries and sorts entries locally. If 0, leave sorting as negotiated between the
client and server. If 1, force sorting of queries (this causes excessive LDAP queries and should only be used to
diagnose LDAP servers with sorting problems).
dir.corp.transport1
TCP, TLS, Null
TCP
Specifies whether a TCP or TLS connection is made with the server, if a full URL is not provided.
dir.corp.useContainsSearch1
0 or 1
0
When set to 0, the search will compare the entered characters against the starting characters of each field
searched. This is the default behavior. When set to 1, the search will look for the entered characters anywhere
inside each corporate directory field.
dir.corp.user1
UTF-8 encoded string
Null
The user name used to authenticate to the LDAP server.
dir.corp.viewPersistence1
0 or 1
0
If 0, the corporate directory search filters and browsing position are reset each time the user accesses the
corporate directory. If 1, the search filters and browsing position from the previous session are displayed each time
the user accesses the corporate directory.
dir.corp.vlv.allow1
0 or 1
0
If 0, virtual view list (VLV) queries are disabled. If 1, VLV queries are enabled and can be made if the LDAP server
supports VLV.
dir.corp.vlv.sortOrder1
list of parameters
Null
The list of parameters —in exact order — for the LDAP server to use when indexing. For example: sn,
givenName, telephoneNumber.
1
Change causes handset to restart or reboot.
The following illustration points you to the minimum parameters you need to set. You will need
to enter a corporate directory address in dir.corp.address. You will need to specify where
on the corporate directory server you want to make queries in dir.corp.baseDN. In addition,
you will require a user name and password. The dir.corp.attribute.x.name must match
the attributes in the server.
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Default Ring Tones and Alert Tones
In different countries, different ring and tone patterns are used for a busy signal or reorder tone,
for example. This section explains how to change the default ring tones and alert tones to the
customary tones used in your area. See the sound effects pattern types table for more
information.
Additionally, the handset can use built-in wave files for some sound effects. The built-in wave
files can be replaced with files downloaded from the provisioning server or from the Internet.
These are stored in volatile memory so the files will need to remain accessible should the
handset need to be rebooted. Files will be truncated to a maximum size of 300 kilobytes.
The following sampled audio WAVE (.wav) file formats are supported:

mono 8 kHz G.711 u-Law

G.711 A-Law
Your custom sampled audio files must be available at the path or URL specified by saf.x so
the handset can download them at bootup. Include the name of the file and the .wav extension
in the path. In the following example, you can see that the tones are at the top level of the zip
file structure when the code is downloaded.
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Summary
Parameter
Used to:
saf.x
Specify a path or URL for the handset to download a custom audio file
se.pat.*
Specify the name, type, and value for a custom sound effect
Table 8-14: Sampled Audio File Parameter
Parameter
Permitted Values
saf.x
Null or valid path name or an RFC 1738-compliant URL to a HTTP,
FTP, or TFTP wave file resource.
LoudRing.wav
Default
Where “x” is the saf # that corresponds to the ringtone slot number you want the.wav file to appear in on the
phone’s menu.
The welcome tone is programmed to saf.1 which is menu slot 15. You can program this slot to a different welcome
tone by programming saf.1. Slots 15 through 22 are available for custom tones. If Null, the handset will use a builtin file.
If set to a path name, the handset will attempt to download this file at boot time from the provisioning server.
If set to a URL, the handset will attempt to download this file at boot time from the Internet.
Note: A TFTP URL is expected to be in the format: tftp://<host>/[pathname]<filename>, for example:
tftp://somehost.example.com/sounds/example.wav .
Note: See the above wave file format restrictions.
Specifying the saf.x parameter simply tells the phone to make the ringtone available in the
ringtone menu list at the specified slot. Per the normal use of the saf parameter, the ‘x’ is an
offset from the end of the default menu (slots 1-14). So saf.1 points to slot 15, etc…The table
below provides the correspondences amoung saf#, menu slot# and pattern reference.
Notes on the Loudring.wav option
The LoudRing.wav file is a special case that makes this custom tone available on the phones
rather than having to download a wav file to the phone from a provisioning server. Therefore, it
is not necessary to have the file on the provisioning server since it is already on the phone.
Once this configuration option is loaded on the phone the LoudRing.wav will be available even if
the phone cannot connect to the provisioning server.
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Table 8-15: Default Sample Audio File Usage
SAF #
Menu
slot #
Default Use (Pattern Reference)
1
15
Ringer 12 (se.pat.misc.welcome)
2
16
Ringer 13 (se.pat.ringer.ringer15)
3
17
Ringer 14 (se.pat.ringer.ringer16)
4
18
Ringer 15 (se.pat.ringer.ringer17)
5
19
Ringer 16 (se.pat.ringer.ringer18)
6
20
Ringer 17 (se.pat.ringer.ringer19)
7
21
Ringer 18 (se.pat.ringer.ringer20)
8
22
Ringer 19 (se.pat.ringer.ringer21)
9
23
Ringer 20 (se.pat.ringer.ringer22)
10
24
Ringer 21 (se.pat.ringer.ringer23)
Sampled
Audio File
Number
11
Ringer 22 (se.pat.ringer.ringer24)
12 to 24
Not Used
The handset uses both synthesized (based on the chord-sets, see <chord/>) and sampled audio
sound effects. Sound effects are defined by patterns: rudimentary sequences of chord-sets,
silence periods, and wave files.
Table 8-16: Sound Effect Parameter
Parameter
Permitted Values
Default
se.appLocalEnabled1
0 or 1
1
If set to 1, local user interface sound effects such as confirmation/error tones, will be enabled.
se.stutterOnVoiceMail
0 or 1
1
If set to 1, a stuttered dial tone is used in place of a normal dial tone to indicate that one or more voicemail
messages are waiting at the message center.
1
Change causes handset to restart or reboot.
Patterns use a simple script language that allows different chord sets or wave files to be strung
together with periods of silence. The script language uses the following instructions:
Table 8-17: Sound Effects Pattern Types
Instruction
Meaning
sampled (n)
Play sampled audio file n
Example:
se.pat.misc.SAMPLED_1.inst.1.type =”sampled” (sampled audio file instruction type)
se.pat.misc.SAMPLED_1.inst.1.value =”2” (specifies sampled audio file 2)
chord (n, d)
Play chord set n (d is optional and allows the chord set ON duration to
be overridden to d milliseconds)
Example:
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Instruction
Meaning
se.pat.callProg.busyTone.inst.2.type = “chord” (chord set instruction type)
se.pat.callProg.busyTone.inst.2.value = “busyTone” (specifies sampled audio file busyTone)
se.pat.callProg.busyTone.inst.2.param = “2000” (overrule ON duration of chord set to 2000
milliseconds)
silence (d)
Play silence for d milliseconds (Rx audio is not muted)
Example:
se.pat.callProg.bargeIn.inst.3.type = “silence” (silence instruction type)
se.pat.callProg.bargeIn.inst.3.value = “300” (specifies silence is to last 300 milliseconds)
branch (n)
Advance n instructions and execute that instruction (n must be negative
and must not branch beyond the first instruction)
Example:
se.pat.callProg.alerting.inst.4.type = “branch” (branch instruction type)
se.pat.callProg.alerting.inst.4.value = “-2” (step back 2 instructions and execute that instruction)
In the following table, x is the pattern name, y is the instruction number. Both x and y need to be
sequential. There are three categories cat of sound effect patterns: callProg (Call Progress
Patterns), ringer (Ringer Patterns) and misc (Miscellaneous Patterns).
Table 8-18: Sound Effects Pattern Parameters
Parameter
Permitted Values
se.pat.cat.x.name
UTF-8 encoded string
Sound effects name, where cat is callProg, ringer, or misc.
se.pat.cat.x.inst.y.type
sampled, chord, silence, branch
Type of sound effect, where cat is callProg, ringer, or misc.
se.pat.cat.x.inst.y.value
String
The instruction: sampled – sampled audio file number, chord – type of sound effect,
silence – silence duration in ms, branch – number of instructions to advance. cat is callProg, ringer, or
misc.
Call Progress Patterns
The following table shows the call progress pattern names and their descriptions:
Table 8-19: Call Progress Tone Pattern Names
Call Progress Pattern Name
Description
alerting
Alerting
bargeIn
Barge-in tone
busyTone
Busy tone
callWaiting
Call waiting tone
callWaitingLong
Call waiting tone long (distinctive)
confirmation
Confirmation tone
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Call Progress Pattern Name
Description
dialTone
Dial tone
howler
Howler tone (off-hook warning)
intercom
Intercom announcement tone
msgWaiting
Message waiting tone
precedenceCallWaiting
Precedence call waiting tone
precedenceRingback
Precedence ringback tone
preemption
Preemption tone
precedence
Precedence tone
recWarning
Record warning
reorder
Reorder tone
ringback
Ringback tone
secondaryDialTone
Secondary dial tone
stutter
Stuttered dial tone
Example Configuration
The following example configuration illustrates how to add a custom sound effect from a
sampled audio file. In the example, the custom audio files Classic phone.wav and Chirp.wav
have been added as sound effects 12 and 13. The welcome sound has been customized to use
the sampled audio file 13 (Chirp.wav) with the label Birds. Ringtone 19 is named Classic phone
and is configured to use sampled audio file 12 (Classic phone.wav).
The following illustration shows the custom ring tone Classic phone as it displays on the
handset menu:
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Do Not Disturb
You can use the Do Not Disturb (DND) feature to temporarily stop incoming calls. Incoming calls
received while DND is turned on are logged as missed. DND is enabled locally through the
handset by navigating to Settings> Feature Settings> Do Not Disturb.
Table 8-20: Configuring Do Not Disturb
Parameter
Permitted Values
Default
call.rejectBusyOnDnd1
0 or 1
1
If 1, and DND is turned on, the handset rejects incoming calls with a busy signal. If set to 0, and DND is turned on,
the handset gives a visual alert of incoming calls and no audio ringtone alert.
Note: This parameter does not apply to shared lines since not all users may want DND enabled.
call.donotdisturb.perReg1
0 or 1
0
This parameter determines if the Do-Not-Disturb feature will apply to all registrations on the handset (globally), or
apply on a per-registration basis. If 0, DND will apply to all registrations on the handset when it is active. If 1, the
user can activate DND on a per-registration basis.
1
Change causes handset to restart or reboot.
Dual Tone Multi-Frequency (DTMF) Tones
The handset generates dual tone multi-frequency (DTMF) tones in response to user dialing on
the dial pad. The parameters in the following table will help you set up this feature. These tones,
commonly referred to as touch tones, are transmitted in the real-time transport protocol (RTP)
streams of connected calls. The handset can encode the DTMF tones using the active voice
codec or using RFC 2833-compatible encoding. The coding format decision is based on the
capabilities of the remote end point.
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Summary
Parameter
Used to:
tone.dtmf.chassis.masking
Specify if DTMF tones should be played through the speakerphone
tone.dtmf.level
Specify the frequency level of DTMF digits
tone.dtmf.offTime
Specify how long the handset should wait between DTMF digits
tone.dtmf.onTime
Specify how long the handset should play each DTMF tone for
tone.dtmf.viaRtp
Enable or disable DTMF encoding in an RTP stream
Table 8-21: DTMF Tone Parameters
Parameter
Permitted Values
Default
tone.dtmf.chassis.masking1
0 or 1
0
If 0, DTMF tones will be played through the speakerphone in handsfree mode. If 1 (set only if
tone.dtmf.viaRtp is set to 0), DTMF tones will be substituted with non-DTMF pacifier tones when dialing in
handsfree mode – this is to prevent the tones from broadcasting to surrounding telephony devices or being
inadvertently transmitted in-band due to local acoustic echo.
tone.dtmf.level1
-33 to 3
-15
The level of the high frequency component of the DTMF digit measured in dBm0; the low frequency tone will be
two dB lower.
tone.dtmf.offTime1
positive integer
50
When a sequence of DTMF tones is played out automatically, this is the length of time in milliseconds the handset
will pause between digits. This is also the minimum inter-digit time when dialing manually.
tone.dtmf.onTime1
positive integer
50
When a sequence of DTMF tones is played out automatically, this is the length of time in milliseconds the tones
will be played for. This is also the minimum time the tone will be played when dialing manually (even if key press is
shorter).
tone.dtmf.viaRtp1
0 or 1
1
If set to 1, encode DTMF in the active RTP stream. Otherwise, DTMF may be encoded within the signaling
protocol only when the protocol offers the option. Note: If this parameter is set to 0,
tone.dtmf.chassis.masking should be set to 1.
1
Change causes handset to restart or reboot.
DTMF Event RTP Payload
The handset is compatible with RFC 2833—RTP Payload for DTMF Digits, Telephony Tones,
and Telephony Signals. RFC 2833 describes a standard RTP-compatible technique for
conveying DTMF dialing and other telephony events over an RTP media stream. The handset
generates RFC 2833 (DTMF only) events but does not regenerate – or otherwise use – DTMF
events received from the remote end of the call. Use the next table to set up this feature.
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Table 8-22: Dual Tone Multi-Frequency (DTMF) Event RTP Payload
Parameter
Permitted Values
Default
tone.dtmf.rfc2833Control1
0 or 1
1
If set to 1, the handset will indicate a preference for encoding DTMF through RFC 2833 format in its Session
Description Protocol (SDP) offers by showing support for the handset-event payload type. This does not affect SDP
answers; these will always honor the DTMF format present in the offer since the handset has native support for
RFC 2833.
tone.dtmf.rfc2833Payload1
96 to 127
127
The handset-event payload encoding in the dynamic range to be used in SDP offers.
1
Change causes handset to restart or reboot.
Emergency Calls
Emergency numbers can be programmed to appear on the Speed Dial list and Favorites menu.
These numbers can be dialed when the handset is registered and available for calls. See Call
List for more information.
When the phone’s keypad is locked or a user is logged out, an emergency call can still be made
if emergency dial numbers are programmed to appear on an Authorized Call menu.
Additionally, the PTT button on the 8453 handset can be programmed to be used as a duress
button if PTT is not programmed. These two contingencies are explained below.
Emergency Dial via Authorized Call menu
When the phone is locked and requires a PIN before it will respond to keypad presses, a New
Call softkey is available for emergency calls. When pressed, it opens the Authorized Call menu
which lists numbers that call be reached while the phone is locked. Please see the Phone Lock
section for information about phoneLock parameters that program the numbers to appear on the
Authorized Call menu. For these numbers to go through, the phone must be registered and the
numbers must be recognized by the call server. There are no phoneLock parameters that
specify which server to use.
When User Profiles are deployed, when a user is logged out the phone is not available to make
any calls except through the Authorized Call menu, accessed by pressing the Start button at the
login screen. When users are logged out, the phone is not registered and calls usually cannot
be made. However an emergency call server can be configured to allow anonymous calls and
the numbers programmed to that server will go through. How to program the
dialplan.routing.emergency parameters for server access and authorized numbers is covered in
the User Profiles section Placing Authorized (Emergency) Calls without Logging In.
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Caution: What numbers are on the Authorized Call menu
All numbers programmed through phoneLock.authorized.x and
dialplan.routing.emergency.x parameters will appear on the Authorized Call menu.
If the phone is registered, any number listed on the Authorized Call menu should go
through. The phoneLock numbers will be routed through the SIP call server and the
dialPlan numbers will be routed through the emergency call server.
If the phone is not registered, only those numbers programmed to the emergency
call server that allows anonymous SIP calls will go through.
Emergency Dial via Duress Button
Emergency dial, also known as the “duress button” is enabled by programming the PTT key on
the left side of the handset to function as a speed dial button programmed to call an emergency
number. The Emergency Dial feature will allow the user to place the emergency call without
having to unlock the keypad or unlock the handset. When the button is pressed twice within two
seconds, a call is placed to the programmed number. A pop-up window on the display will
inform the user that this mode is being activated.
If you enable Emergency Dial, PTT cannot be deployed. If both are enabled, Emergency Dial
will overrule and PTT will not work.
The Emergency Dial feature places the call on Line 1. Therefore the PBX that line 1 is assigned
to must be able to place a local emergency call. Any active call on line 1 will be preempted by
the emergency call.
Two dial modes are configurable. Both modes are triggered by remapping the Talk key to a
speed dial key. The software detects this remapping, which triggers the special handling of the
key presses for this mode.

The dial mode, which uses only a dial plan number for the Emergency Call, and

the macro mode, which allows a more complex dial pattern for the Emergency Dial
feature.
Dial Mode
This mode is configured in the site.cfg and ptt.cfg templates. A sample file is shown below:
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Example PTT Emergency Dial -->
<handsetConfig>
<key
key.20.function.prim="SpeedDial"
/>
<ptt
ptt.pttMode.enable="0"
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ptt.emergencyDial.description="CODE BLUE Emergency"
ptt.emergencyDial.number="6183"
/>
</handsetConfig>
The key section remaps the Talk key to a speed dial key. The ptt section turns off the .pttMode,
provides a .description of the call which will appear in the To: line when the call is placed, and
provides the extension .number to be dialed. You may customize the yellow highlighted fields
for the display and the number to be dialed.
Dial Mode example in the ptt.cfg template (also available in the site.cfg template)
Macro Mode
The macro mode is configured by programming a macro as the dial string for the emergency
number. EFK must be enabled. A sample file is shown below:
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Example PTT Emergency Dial Macro Mode-->
<handsetConfig>
<key
key.20.function.prim="SpeedDial"
/>
<ptt
ptt.pttMode.enable="0"
ptt.emergencyDial.description="CODE BLUE Emergency"
ptt.emergencyDial.number="^6183$Tinvite$$Cwc$456$Cp2$$Tdtmf$"
/>
<feature
feature.enhancedFeatureKeys.enabled=”1”
/>
</handsetConfig>
The difference between the two files is in the .number parameter. This file uses a macro to
describe the dialing sequence to be used. The example shown above will place a SIP call to
6183, wait for the call to be connected, wait for two seconds, and then send DTMF signaling for
456.
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The first entry in the macro, 6183, is the extension the call will be placed to. The entry,
$Tinvite$, specifies that the call will be placed by using the SIP Invite method. $Cwc$ specifies
that the macro should pause until a connection is established with the far end. This could also
be specified as $Cwaitconnect$. 456 specify the next digits to be dialed. $Cp2$ or alternatively
$Cpause2$ causes a two second delay to occur prior to dialing after the connection is
established. This can be any integer value from 1 to 10. $Tdtmf$ specifies DTMF dialing for the
second series of digits.
The Enhanced Feature Key (EFK) macro language is described elsewhere in this document.
NOTE: Currently, the .description parameter is ignored for macro based dialing. The To: line will
appear as “To:EFK Dial”.
NOTE: For the macro dialing mode to function, the Enhanced Feature Keys mode must be
enabled.
Default Values
If the PTT Emergency Dial feature is activated via the remapping of the Talk key to a Speed Dial
key, the default values for the .description and .number parameters are “Emergency Call” and
“911” respectively.
Table: Emergency Dial Parameters
Parameter
Permitted values
Default
key.20.function.prim
SpeedDial
Null
Changes key 20, the PTT button, to a speed dial function.
ptt.pttMode.enable
0 or 1
0
If Emergency Dial is enabled, disable PTT by setting this value to 0.
ptt.emergencyDial.description
string
Emergency Call
Enter a label description which will appear on the display when the number is called.
ptt.emergencyDial.number
string
911
Enter the number to be dialed.
Enhanced Feature Keys
Enhanced Feature Keys (EFK) enables you to customize the functions of line keys and
softkeys. You can use EFK to assign frequently used functions to line keys and softkeys or to
create menu shortcuts to frequently used handset settings.
Enhanced feature key functionality is implemented using star code sequences (like *69) and SIP
messaging. Star code sequences that define EFK functions are written as macros that you
apply to line and softkeys. The EFK macro language was designed to follow current
configuration file standards and to be extensible. The macros are case sensitive.
Different rules apply to configuring EFK for line keys and softkeys. Before using EFK, please
become familiar with the macro language shown in this section.
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Summary
Parameter
Used to:
reg.x.callsPerLineKey
Specify at least two calls per line key
feature.enhancedFeatureKeys.enabled
Enable or disable Enhanced Feature Keys
efk.efklist.x.*
Specify the EFK List parameters
efk.efkprompt.x.*
Specify the EFK Prompts
Because line keys and their functions are linked to fields in the contact directory file 000000000000-directory.xml (global) or <MACaddress>-directory.xml (per handset),- you will
need to match the contact field (ct) in the directory file to the macro name field (mname) in the
configuration file that contains the EFK parameters. When you enter macro names to the
contact field (ct) in the directory file, add the ‘!’ prefix to the macro name. For more detailed
information on using the contact directory, see Local Contact Directory.
Guidelines for Configuring Enhanced Feature Keys
The following guidelines will help you to configure EFK efficiently:

Activation of EFK functions requires valid macro construction.

All failures are logged at level 4 (minor), in the EFK logging module.

If two macros have the same name, the first one will be used and the subsequent ones
will be ignored.

A sequence of characters prefixed with “!” are parsed as a macro name. The exception
is the speed dial reference, which starts with “!” and contains digits only.

A sequence of characters prefixed with “^” is the action string.

“'!” and “^” macro prefixes cannot be mixed in the same macro line.

The sequence of characters must be prefixed by either “!” or “^” so it will be processed
as an enhanced feature key. All macro references and action strings added to the local
directory contact field must be prefixed by either “!” or “^”.

Action strings used in softkey definitions do not need to be prefixed by “^”. However, the
“!” prefix must be used if macros or speed dials are referenced.

A sequence of macro names in the same macro is supported (for example, “!m1!m2” ).

A sequence of speed dial references is supported (for example, “!1!2” ).

A sequence of macro names and speed dial references is supported (for example,
“!m1!2!m2” ).

Macro names that appear in the local contact directory must follow the format “!<macro
name>” , where <macro name> must match an <elklist> mname entry. The maximum
macro length is 100 characters.

A sequence of macros is supported, but cannot be mixed with other action types.

Action strings that appear in the local contact directory must follow the format “^<action
string>”. Action strings can reference other macros or speed dial indexes. Protection
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against recursive macro calls exists (the enhanced feature keys fails once you reach 50
macro substitutions).
Table 8-23: Enhanced Feature Key (EFK) List Parameters
Parameter Name
Permitted Values
Default
efk.efklist.x.action.string
string
Null
The action string contains a macro definition of the action that the feature key will perform. If EFK is enabled, this
parameter must have a value (it cannot be Null). For a list of macro definitions and example macro strings, see
Understanding Macro Definitions.
efk.efklist.x.label
string
Null
The text string that will be used as a label on any user text entry screens during EFK operation. If Null, the Null
string is used. Note: If the label does not fit on the screen, the text will be shortened and ‘…’ will be appended.
efk.efklist.x.mname
string
expanded_macro
The unique identifier used by the speed dial configuration to reference the enhanced feature key entry. Cannot
start with a digit. Note that this parameter must have a value, it cannot be Null.
efk.efklist.x.status
0 or 1
0
Parameter Name
Permitted Values
Default
feature.enhancedFeatureKeys.enabled
0 or 1
0
If 0, key x is disabled. If 1, the key is enabled.
Table 8-24: Enhanced Feature Key (EFK) Parameters
If 0, the enhanced feature keys feature is disabled. If 1, the feature is enabled.
efk.version
2 (1 for SIP 3.0 and
earlier)
2
The version of the EFK elements. For SIP 3.0.x or earlier, 1 is the only supported version. For SIP 3.1 and later, 2
is the only supported version. If this parameter is Null, the EFK feature s disabled. This parameter is not required if
there are no efk.efklist entries.
Enhanced Feature Key Examples
The following illustration shows the default value 24 calls per line key. Ensure that you specify at
least two calls per line key. Failure to do so will prevent the macro from activating as additional
calls per line key are used when activating a macro.
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Find the enhanced feature keys feature in the everything.cfg template file. Use this file as the
source for the EFK parameters. Copy/paste or drag them to the .cfg file you are developing for
your facility.
In the following illustration, the EFK parameters are located in the everything.cfg file. In the
efk.efklist.x.* parameters, line key ‘1’ has been assigned a Call Park address
(1955) and line key ‘2’ a Call Retrieve function. The parameter acton.string shows you
the macro definition for these two functions. In addition, status is enabled (1) and a label has
been specified to display next to the line key. The entry in the mname parameter corresponds to
the contact (ct) field in the contact directory.
In the efk.prompt.* parameters, status has been enabled (1). The label on the user
prompt has been defined as Enter Number: and this prompt will display on the handset
screen. The type parameter has been set to numeric to allow only numbers and because
userfeedback has been specified as visible, you will be able to see the numbers you enter
into the prompt.
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Understanding Macro Definitions
The efk.efklist.x.action.string can be defined by one of the following:

Macro Action

Prompt Macro Substitution

Expanded Macros
Macro Action
The action string is executed in the order it displays, i.e. as it is read from left to right. If
necessary, user input is collected before any action is taken. The action string can contain the
following fields.
To aid in the understanding of each of the following fields we will reference an example macro
action string:
$LCallPark$20371$P1N4$$Trefer$$Cwaitconnect$
Note that this example does not contain all actions.
Macro Actions and Descriptions
$L<label>$
This is the label that describes the macro action to be performed and will be displayed on the line key or softkey
on the handset. The value can be any string including the null string (in this case, no label displays). This label
will be used if no other label has been defined (up to the point where this field is parsed in the macro). Make
this the first entry in the action string to be sure this label is used; otherwise another label may be used and this
one ignored.
In the above example, the label portion is the $LCallPark$. This field is not required but can be helpful to
include when later referencing or troubleshooting macro setup.
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digits
The digits to be sent. The appearance of this parameter depends on the action string. If no digits are to be sent
it may not be necessary to include them. For example, if the macro includes a prompt to gather digits from the
user then it may not be necessary to supply digits. Valid digits are 0-9 and ‘*’ and ‘#’.
In the above example, the digits being sent by the macro action string are 20371. Note that the digits do not get
entered into the string with the ‘$’ on either end of the digits. This is because the digits are not a function that
must be interpreted by the handset.
$C<command>$
This is a command that will be performed while processing the macro action string. It can appear anywhere in
the action string and may appear multiple times. Supported commands (or shortcuts) include:
• hangup (hu)
• hold (h)
• waitconnect (wc)
• pause <number of seconds> (p <num sec>) where the maximum value is 10
Note that the shortcut version of each command can be entered into the macro string instead of the full
command name.
In the above example, the command portion of the macro action string can be identified by the capital ‘C’. In this
case we are issuing a wait connect command, $Cwaitconnect$. The wait connect command causes the handset
to stop processing the macro action string until the call is connected to the far end. Using it at the end of the line
like this means that the handset will process everything before and then stop to wait for the call to connect. Since
we were collecting digits from the user the handset will not send these digits until the call is connected.
$T<type>$
The type of action that will be performed if the macro action string is intended to initiate a call. Multiple actions
can be defined. Supported action types include:
• invite
• dtmf
• refer
Note: Spectralink recommends that you always define this field. If it is not defined, the supplied digits will be
dialed using INVITE (if no active call) or DTMF (if an active call). The use of refer method is call server
dependent and may require the addition of star codes. The refer method is the equivalent of a call transfer.
In the above example, the type of action being performed is, $Trefer$. Because this example is for a call park
scenario the refer action type is used because we are going to transfer the active call on the handset to a
network park location.
$M<macro>$
If you need to define multiple macros you can embed macros into a single macro string by using this option. The
<macro> string must begin with a letter rather than a digit. If the macro name is not defined, the execution of the
entire action string fails. Note that the <macro> is the actual name of another macro, not the macro action string.
You would define this embedded macro in the same you are defining this current macro. The macro name is the
efk.efklist.*.mname parameter.
$P<prompt num>N<num digits>$
You can use this option to cause the macro to prompt the user for additional input. That might be a name, a
phone number or whatever you need them to enter. See Prompt Macro Substitution for more details on how to
use this option.
In the above example, we defined a user prompt using, $P1N4$. This method uses the Prompt Macro
Substitution configuration to create the text that is displayed on the handset to prompt the user for more input. It
then collects the input, in this case, a maximum of 4 characters, and feeds that back into the macro for additional
processing.
$S<speed dial index>$
Using this option you can define a directory entry as the location to collect the input needed to process the
macro. Only digits are valid for the <speed dial index> as only digits are used in the directory to define an index
location. The input collected is found in the contact field of the local directory entry pointed to by the index
number.
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$F<internal function>$
This particular option can be very powerful and equally useful when defining a macro. There are a number of
internal handset functions that are pre-defined that you can call using this option. For example, you can use the
internal function that allows you to navigate on the handset or adjust the handset volume.
URL
Entering a URL into a macro string can be used to cause the handset to browse to a specific web page or to
access a server location using the handsets browser. Only one URL per action string is supported.
Prompt Macro Substitution
The efk.efklist.x.action.string can be configured to include a prompt using, $PnNn$
where:

P is the prompt n as defined by efk.efkprompt.x where n=x.

N, where n is the number of digits or letters that the user can enter. The value must be
between 1 and 32 characters; otherwise the macro execution will fail. The user needs to
press the Enter softkey to complete data entry.
The macros provide a generic and easy to manage way to define the prompt to be displayed to
the user and the maximum number of characters that the user can input. The macros are case
sensitive so you will need to ensure that the P and N are both capitalized.
If a macro attempts to use a prompt that is disabled, the macro execution fails. A prompt is not
required for every macro.
Enhanced Feature Key (EFK) Prompt Parameters
Parameter Name
efk.efkprompt.x.label
1
Permitted Values
Default
string
Null
The prompt text that is presented to the user on the user prompt screen. If Null, no prompt displays. Note: If the
label does not fit on the screen, the label will be shortened and ‘…’ will be appended.
efk.efkprompt.x.status
1
0 or 1
0
If 0, key x is disabled. If 1, the key is enabled. This parameter must have a value, it cannot be Null. Note: If a
macro attempts to use a prompt that is disabled or invalid, the macro execution will fail.
efk.efkprompt.x.type
1
numeric or text
text
The type of characters entered by the user. If set to numeric, the characters are interpreted as numbers. If set to
text, the characters are interpreted as letters. If Null, numeric is used. If this parameter has an invalid value, this
prompt, and all parameters depending on this prompt, are invalid. Note: A mix of numeric and text is not
supported.
efk.efkprompt.x.userfeedback
1
visible or masked
visible
The user input feedback method. If set to visible, the text is visible. If set to masked, the text displays as asterisk
characters (*), this can be used to mask password fields. If Null, visible is used. If this parameter has an invalid
value, this prompt, and all parameters depending on this prompt, are invalid.
1
Change causes handset to restart or reboot.
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Expanded Macros
Expanded macros are prefixed with the ^ character and are inserted directly into the local
directory contact field. For more information, see Local Contact Directory.
Special Characters
The following special characters are used to implement the enhanced feature key functionality.
Macro names and macro labels cannot contain these characters. If they do, you may
experience unpredictable behavior.

! The characters following it are a macro name.

' or ASCII (0x27) This character delimits the commands within the macro.

$ This character delimits the parts of the macro string. This character must exist in pairs.

^ This character indicates that the following characters represent the expanded macro
(as in the action string).
Example Macro
The action string:
$Changup$*444*$P1N4$$Tinvite$$Cwaitconnect$$P2N3$$Cpause2$$Tdtmf$$Ch
angup$
is executed in order as follows:
1
The handset terminates any active call.
2
The handset receives digits *444*.
3
The user is prompted for 4 digits. For example, 1234.
4
The handset issues a new call to *444*1234 using the INVITE method.
5
The handset waits until the call connects before processing the macro further.
6
The user is prompted for 3 digits. For example, 567.
7
The handset pauses for 2 seconds
8
The handset sends the collected digits, 567, using DTMF tones.
9
The active call is disconnected.
Because line keys and their functions are linked to fields in the directory file, a macro name you
enter in efk.list.x.mname must match the name you enter to the contact (cn) field in
the directory file. The macro name you enter in the (ct) field of the directory file must begin with
the ‘!’ prefix. The following example directory file shows a line key configured with Call Park, Call
Retrieve, and a speed dial contact Lisa Woo.
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For an explanation of all fields in the directory file, see Local Contact Directory.
All line keys will be represented under the Features softkey fly-out menu on the handset.
Speed Dial Example
If your organization’s voicemail system is accessible through 7700 and your voicemail password
is 2154, you can use a speed dial key to access your voicemail by entering
7700$Cpause3$2154 as the contact number in the contact (ct) element.
Tip: Ensuring Users Do Not Delete Definitions in the Contact Directory
To avoid users accidentally deleting the definitions in the contact directory, make
the contact directory read only.
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Features Softkey Menu Options Customization
On the Spectralink handsets, you can customize the flyout menu of the Features softkey to add
options. This feature is typically used to access frequently used functions. As with EFK line
keys, you assign functions to Features options using macros.
You can configure the custom option to display depending on the handset’s menu level or call
state. For example, you can make a Call Park option available when the handset is in an active
call state.
Custom options can be added in the following call states:
 Idle There are no active calls.
 Active This state starts when a call is connected. It stops when the call stops or
changes to another state (like hold or dial tone).
 Alerting (or ringing or incoming proceeding)
 Dial tone
The handset is ringing.
You can hear a dial tone.
 Proceeding (or outgoing proceeding) This state starts when the handset sends a
request to the network. It stops when the call is connected.
 Setup This state starts when the user starts keying in a handset number. This state
ends when the Proceeding state starts.
 Hold The call is put on hold locally.
New Features options can be created as:
 An Enhanced Feature Key sequence
 A speed dial contact directory entry
 An Enhanced Feature Key macro
 A URL
 A chained list of actions
Up to 10 custom options can be configured. If more softkeys are configured than fit on the
handset’s screen, a More softkey displays. Press the More softkey to view the remaining
softkeys.
This feature is part of Enhanced Feature Keys (EFK) and you must enable the enhanced
feature keys parameter to configure softkeys.
Summary
Parameter
Used to:
feature.enhancedFeatureKeys.enabled
To turn Enhanced Feature Keys on (required)
softkey.x.action
Specify the macro for a line key or softkey function
softkey.x.enable
To enable a custom softkey
softkey.x.insert
Specify the position of the softkey on the handset screen
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Parameter
Used to:
softkey.x.label
Specify the text to display on the softkey label
softkey.x.precede
To position the custom softkey before the default softkeys
softkey.x.use.*
Specify which call states the softkey will display in
softkey.feature.*
To display softkeys for various handset features, including default
softkeys
Note that feature.enhancedFeatureKeys.enabled must be enabled (set to 1) to use the
Configurable Softkey feature.
Table 8-25: Softkey Customization Parameters
Parameter
Permitted Values
Default
softkey.feature.basicCallManagement.redundant
0 or 1
1
Control the display of the Hold, Transfer, and Conference softkeys. If set to 0 and the handset has hard keys
mapped for Hold, Transfer, and Conference functions (all must be mapped), none of the softkeys are displayed.
If set to 1, all of these softkeys are displayed.
softkey.feature.callers
0 or 1
0
If 1, the Callers softkey displays on all platforms. If 0, the Callers softkey is disabled for all platforms.
The default is 0.
softkey.feature.directories
0 or 1
0
If 1, the Dir softkey displays on all platforms. If 0, the Dir softkey is disabled for all platforms.
The default value is 0.
softkey.feature.endcall
0 or 1
1
If 0, the End Call softkey is not displayed. If 1, the softkey is displayed.
softkey.feature.mystatus
0 or 1
1
If 0, the MyStatus softkey is not displayed. If 1, the softkey is displayed (if pres.idleSoftKeys is set to 1). Only
used with Skype for Business.
softkey.feature.newcall
0 or 1
1
If 0, the New Call softkey is not displayed when there is an alternative way to place a call. If 1, the New Call
softkey is displayed.
softkey.feature.simplifiedSignIn
0 or 1
0
If 0, the SignIn softkey is not displayed. If 1 and voIpProt.server.x.specialInterop is lync2010,
lync2013 or Skype for Business, the SignIn softkey is displayed. The Lync Base Profile sets it to 1 by
default and signin options are available in the Features softkey menu. See the Lync Interoperability Guides for
more information.
softkey.x.action
macro action string, 256
characters
Null
The action or function for custom softkey x. This value uses the same macro action string syntax as an Enhanced
Feature Key. For a list of actions, see Macro Action.
softkey.x.enable
0 or 1
0
If 0, the softkey x is disabled. If 1, the softkey is enabled.
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Parameter
Permitted Values
Default
softkey.x.insert
0 to 10
0
The position on the handset screen for softkey x. For example, if the value is 3, the softkey will be displayed on the
screen in the third position from the left. Note: If softkey.x.precede is configured, this value is ignored. If the
insert location is greater than the number of softkeys, the key will be positioned last, after the other softkeys.
softkey.x.label
string
Null
The text displayed on the softkey label. If Null, the label is determined as follows:

If the softkey performs an Enhanced Feature Key macro action, the label of the macro will be used.

If the softkey calls a speed dial, the label of the speed dial contact will be used.

If the softkey performs chained actions, the label of the first action is used.

If the softkey label is Null and none of the preceding criteria are matched, the label will be blank.
softkey.x.precede
0 or 1
0
If 0, softkey x is positioned in the first empty space from the left. If 1, the softkey is displayed before (to the left of)
the first default softkey.
softkey.x.use.active
Display in the active call state
softkey.x.use.alerting
Display in the alerting state
softkey.x.use.dialtone
Display in the dial tone state
softkey.x.use.hold
Display in the hold state
softkey.x.use.idle
Display in the idle state
softkey.x.use.proceeding
Display in the proceeding state
softkey.x.use.setup
0 or 1
0
0 or 1
0
0 or 1
0
0 or 1
0
0 or 1
0
0 or 1
0
0 or 1
0
If 0, the softkey is not displayed when the handset is in the x state. If 1, the softkey is displayed when the handset
is in the x state.
1
Change causes handset to restart or reboot.
Example Softkey Configurations
This section provides a few examples of available softkey configurations.
Web Info: Using Configurable Softkeys
For more examples, see Technical Bulletin 42250: Using Enhanced Feature Keys
and Configurable Softkeys on Spectralink handsets.
To map a chained list of actions to a softkey:
1
Configure speed dial index 2 in the contact directory file with a handset address. For
example, enter ‘2900’ in the contact (ct) field.
2
In the contact directory, enter ‘!2’ in the contact (ct) field of speed dial index 1.
3
Update the configuration file as follows:
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softkey.1.label = ChainAct
softkey.1.action = $S1$$Tinvite$
softkey.1.use.idle = 1
4
Reboot the handset.
A softkey ChainAct displays. Press ChainAct to dial the handset number 2900.
To map the Do Not Disturb Enhanced Feature Key sequence to a softkey:
1
2
Update the configuration file as follows:
softkey.1.label = DND
softkey.1.action = $FDoNotDisturb$
softkey.1.use.idle = 1
Reboot the handset.
A DND softkey is displayed on the handset when it is in the idle state. When the DND
softkey is pressed, the Do Not Disturb icon is displayed.
To map a Send-to-Voicemail Enhanced Feature Key sequence to a softkey:
1
2
Update the configuration file as follows:
softkey.2.label = ToVMail
softkey.2.action = ^*55$P1N10$$Tinvite$
softkey.2.use.alerting = 1
Reboot the handset.
When another party calls, the ToVMail softkey is displayed. When the user presses the
ToVMail softkey, the other party is transferred to voicemail.
Tip: Active Call Transfer Star Codes Depend On Your Call Server
The exact star code to transfer the active call to Voicemail depends on your call
server.
The following example enables a softkey in the handset’s idle state that navigates to a
handset’s administrator settings. The soft is inserted in softkey position 3, after the default
softkeys. Note the macro action string:
$FMenu$$FDialpad3$$FDialpad2$$FDialpad4$$FDialpad5$$FDialpad6$$FSoft
Key1$
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<softkey/>
The configuration parameter is defined as follows (where x=1 to a maximum number of defined
softkeys).
<efk/>
Use the following three tables to configure the Enhanced Feature Key feature on your handset.
Table 8-26: Enhanced Feature Key (EFK) Parameters
Parameter Name
Permitted Values
Default
feature.enhancedFeatureKeys.enabled
0 or 1
0
If 0, the enhanced feature keys feature is disabled. If 1, the feature is enabled.
efk.version
2 (1 for SIP 3.0 and
earlier)
2
The version of the EFK elements. For SIP 3.0.x or earlier, 1 is the only supported version. For SIP 3.1 and later, 2
is the only supported version. If this parameter is Null, the EFK feature s disabled. This parameter is not required if
there are no efk.efklist entries.
Table 8-27: Enhanced Feature Key (EFK) List Parameters
Parameter Name
Permitted Values
Default
efk.efklist.x.action.string
The action string contains a macro definition of the action that the feature key will perform. If EFK is enabled, this
parameter must have a value (it cannot be Null). For a list of macro definitions and example macro strings, see
Macro Action.
efk.efklist.x.label
string
Null
The text string that will be used as a label on any user text entry screens during EFK operation. If Null, the Null
string is used. Note: If the label does not fit on the screen, the text will be shortened and ‘…’ will be appended.
efk.efklist.x.mname
expanded_macro
The unique identifier used by the speed dial configuration to reference the enhanced feature key entry. Cannot
start with a digit. Note that this parameter must have a value, it cannot be Null.
efk.efklist.x.status
0 or 1
0
If 0 or Null, key x is disabled. If 1, the key is enabled.
efk.efklist.x.type
invite
The SIP method to be performed. If set to invite, the action required is performed using the SIP INVITE method.
Note: This parameter is included for backwards compatibility. Do not use if possible. If efk.x.action.string
contains types, this parameter is ignored. If Null, the default of INVITE is used.
Table 8-28: Enhanced Feature Key (EFK) Prompt Parameters
Parameter Name
Permitted Values
Default
efk.efkprompt.x.label1
string
Null
The prompt text that is presented to the user on the user prompt screen. If Null, no prompt displays. Note: If the
label does not fit on the screen, the label will be shortened and ‘…’ will be appended.
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Parameter Name
Permitted Values
Default
efk.efkprompt.x.status1
0 or 1
0
If 0, key x is disabled. If 1, the key is enabled. This parameter must have a value, it cannot be Null. Note: If a
macro attempts to use a prompt that is disabled or invalid, the macro execution will fail.
efk.efkprompt.x.type1
numeric or text
text
The type of characters entered by the user. If set to numeric, the characters are interpreted as numbers. If set to
text, the characters are interpreted as letters. If Null, numeric is used. If this parameter has an invalid value, this
prompt, and all parameters depending on this prompt, are invalid. Note: A mix of numeric and text is not
supported.
efk.efkprompt.x.userfeedback1
visible or masked
visible
The user input feedback method. If set to visible, the text is visible. If set to masked, the text displays as
asterisk characters (*), this can be used to mask password fields. If Null, visible is used. If this parameter has an
invalid value, this prompt, and all parameters depending on this prompt, are invalid.
1
Change causes handset to restart or reboot.
Handsfree Settings
Spectralink handsets have built-in speakerphones and support Bluetooth v2.1 headsets with
Enhanced Data Rate (EDR) and Headset Profile (HSP v1.2). You can enable and disable each
of these options.
Table 8-29: Audio Options for the Handset, Headset, and Speakerphone
Parameter
Permitted Values
Default
up.headsetOnlyAlerting
0 or 1
0
If 1, only an auxiliary or Wi-Fi headset is used for alerting (such as incoming call alerting), when the headset is
present.
up.headset.phoneVolumeControl1
disable, enable, auto
auto
When a headset is connected to the handset, the handset’s behavior with respect to volume control events from
certain headsets is different.
enable – The handset responds to volume up/down events from the headset by displaying the volume widget in the
handset’s user interface and adjusting the handset's internal volume.
disable – The handset shall ignore volume up/down events from the headset; pressing the headset's volume
controls has no effect on the handset.
auto – The handset shall automatically select which of the above two behaviors to apply, based upon the type and
model version of headset that is attached.
up.analogHeadsetOption
0, 1, or 3
0
The Electronic Hookswitch mode for the handset’s analog headset jack. 0 – no EHS-compatible headset is
attached. 1 – a Jabra EHS-compatible headset is attached. 2 – a Plantronics EHS-compatible headset is attached.
3 – a Sennheiser EHS-compatible headset is attached.
bluetooth.radioOn
0 or 1
0
If 0, the Bluetooth radio (transmitter/receiver) is off. If 1, the Bluetooth radio is on. The Bluetooth radio must be
turned on before the handset can use a Bluetooth headset.
feature.bluetooth.enabled
0 or 1
1
If 0, the Bluetooth headset feature is disabled. If 1, the feature is enabled.
1
Change causes handset to restart or reboot.
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Bluetooth Headset Support
You can use Bluetooth v2.1 headsets with your Spectralink handsets. To use a Bluetooth
headset, you need to enable the Bluetooth headset feature and turn on the Bluetooth radio, as
shown in the next table.
Troubleshooting: Using a Bluetooth Headset Affects my Phone’s Voice
Quality
You may not experience the highest voice quality if you use a Bluetooth headset
while the 2.4 GHz band is enabled or while you are in an environment with many
other Bluetooth devices or other 2.4 GHz wireless devices. This possible loss in
voice quality is due to inherent limitations with Bluetooth technology.
Table 8-30: Bluetooth Headset Support
Parameter
Permitted Values
Default
bluetooth.radioOn
0 or 1
0
If 0, the Bluetooth radio (transmitter/receiver) is off. If 1, the Bluetooth radio is on. The Bluetooth radio must be
turned on before the handset can use a Bluetooth headset.
feature.bluetooth.enabled
0 or 1
1
If 0, the Bluetooth headset feature is disabled. If 1, the feature is enabled.
Language Support
The handset language is selectable by the user by navigating to Settings> Basic Settings>
Preferences> Language. The default language is English but a different language may be
configured.
Supported languages are: Simplified Chinese, Traditional Chinese, Danish, Dutch, English,
French, German, Italian, Japanese, Korean, Norwegian, Polish, Brazilian Portuguese, Russian,
Slovenian, International Spanish, and Swedish.
Table 8-31: Setting the Phone Language
Parameter
Permitted Values
lcl.ml.lang
Null or an exact match for one of the
label names stored in
lcl.ml.lang.menu.x.label
Default
If Null, the default internal language (US English) will be used, otherwise, the language to be used may be
specified in the format of lcl.ml.lang.menu.x.label. Spectralink edits the following languages to match
code updates:
English_Canada
English_United_Kingdom
English_United_States
French_France
German_Germany
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Parameter
Permitted Values
Default
string
Null
0 or 1 or Null
Null
Italian_Italy
Spanish_Spain
lcl.ml.lang.charset1
The language character set.
lcl.ml.lang.clock.x.24HourClock
If parameter present, overrules lcl.datetime.time.24HourClock
If 1, display time in 24-hour clock mode rather than am/pm.
If Null, use value in lcl.datetime.time.24HourClock.
lcl.ml.lang.clock.x.format
string which includes ‘D’, ‘d’ and ‘M’ and
two optional commas
D,dM
If parameter present, overrules lcl.datetime.date.format;
D = day of week d = day M = month. Up to two commas may be included.
For example: D,dM = Thursday, 3 July or Md,D = July 3, Thursday
The field may contain 0, 1 or 2 commas which can occur only between characters
and only one at a time. For example: “D,,dM” is illegal.
lcl.ml.lang.clock.x.longFormat
0 or 1
[none]
If parameter present, overrules lcl.datetime.date.longFormat.
If 1, display the day and month in long format (Friday/November), otherwise use
abbreviations (Fri/Nov).
lcl.ml.lang.list1
a comma-separated list
A list of the languages supported on the handsets.
1
Change causes handset to restart or reboot.
Example Phone Language Configuration
The following illustration shows you how to change the handset language.
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.
From the list, select the language you want to use and enter it in lcl.ml.lang . In the following
example, the handset is set to use the Korean language.
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Once configured, the handset will use Korean characters. The language can be changed on the
Settings menu.
Local Contact Directory
The Spectralink 84-Series handsets feature a contact directory you can use to store frequently
used contacts. Open the Contact Directory by navigating to Home> Contacts/Call Lists>
Contact Directory.
When it first starts up, the handset locates the contact directory by following a specific order:
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1
First the handset looks in its local memory. If the user has populated the contact
directory or makes any change to it, it will write to the MACaddress-directory.xml file on
the central provisioning server and will use that file. If the user has not made any entries
it will go to the second location.
If the contact directory has been populated, it will be written to the central provisioning
server as an xml file named MACaddress-directory.xml every time any changes are
made to it. See Understanding the Files Written by the Handsets for more information
about how handset-written files work.
2
If there is no MACaddress-directory.xml file, the handset will look for a “seed directory”
named 000000000000-directory.xml. This file can be provisioned at initial deployment to
provide contact information to all handsets in the deployment. Once the handset finds
this file, it populates the contact directory. If the user makes any changes to the contact
information, the contact directory is written to the central provisioning server usings its
MACaddress as the filename as explained in step 1.
3
Failing the above 3 locations, the contact directory will remain empty.
The only time the handset “makes it” to #3 is when has been restored to factory defaults and
there is no seed directory provisioned.
Provisioning the Seed Directory
The seed directory gives the administrator a way to populate the contact directory with global
contact information at initial deployment. Unless a handset is returned to factory defaults, it will
never again look for the seed directory because its local memory will be populated as in step 1
and any changes will be stored on the central provisioning server as in step 1.
Two features make the seed directory extremely useful in certain situations: you can provision
an emergency number accessible on the speed dial list and you can apply distinctive incoming
call treatment to specific contacts. You could also use it instead of a corporate directory and
populate users’ contact directories with a full database of current personnel. (see Speed Dial)
(see Distinctive Incoming Call Treatment)
Contact directory is editable by the user
All entries in the users’ contact directories can be edited by the user unless you set
the parameter to read-only.
Setting up the seed directory:
1
Locate the seed directory template in the Config folder you downloaded with the
software. It is named 00000000000-directory~.xml.
2
Remove the tilde from the filename.
3
Populate the entries with your own contact information.
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Table 8-32: Local Contact Directory elements
Element
Definition
Permitted Values
fn
First Name
UTF-8 encoded string of up to 40 bytes1
The contact’s first name.
ln
Last Name
UTF-8 encoded string of up to 40 bytes1
The contact’s last name.
ct
ct2
ct3
ct4
ct5
Contact
UTF-8 encoded string containing digits (the user part of a SIP
URL) or a string that constitutes a valid SIP URL
Used by the handset to address a remote party in the same way that a string of digits or a SIP URL are dialed
manually by the user. This element is also used to associate incoming callers with a particular directory entry. The
maximum field length is 128 characters.
Note: This field cannot be null or duplicated.
Up to 5 contact parameters may be configured, but only 1 is required (ct). Use dir.contact.attribute.x.label to
configure the field name for each of the contact fields. E.g. cell phone, home phone, etc. Ensure the values
entered for the contact align with the configured field names.
sd
Speed Dial Index
Null, 1 to 9999
Associates a particular entry with a speed dial key for one-touch dialing or dialing from the speed dial menu.
lb
Label
UTF-8 encoded string of up to 40 bytes1
An element in the displayed name of the contact. The label field is usually used for a title. If the label field is not
populated, the first and last names will display. . If the label field is populated, then the label will precede the first
and last names: “label” “first name” “last name”. E.g. Dr. John Smith will display when John Smith’s label field is
populated with Dr.
pt
Protocol
SIP, H323, or Unspecified
The protocol to use when placing a call to this contact.
rt
Ring Tone
Null, 1 to 21
When incoming calls match a directory entry, this field specifies the ringtone that will be used.
dc
Divert Contact
UTF-8 encoded string containing digits (the user part of a SIP
URL) or a string that constitutes a valid SIP URL
The address to forward calls to if the Auto Divert feature is enabled.
ad
Auto Divert
0 or 1
If set to 1, callers that match the directory entry are diverted to the address specified for the divert contact element.
Note: If auto-divert is enabled, it overrules auto-reject.
ar
Auto Reject
0 or 1
If set to 1, callers that match the directory entry specified for the auto-reject element are rejected.
Note: If auto divert is also enabled, it overrules auto reject.
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Example seed directory with Security set to speed dial 1 and ring tone 5:
Configuring the Contact Directory
The defaults can be changed to accommodate specialized applications of the contact directory.
You can enable or disable the local contact directory. If disabled it will not appear on the menu.
You can specify read-only for the contact directory, in which case no contact information can be
changed by the user. You can specify the maximum number of entries to conserve handset
memory and you can specify whether the search of the directory is by last or first name.
The field names in the Contact Directory are fixed except for the contact field(s). The contact
field value is the phone number for that entry. Up to five contact fields can be configured, for
example a home, mobile and work number. These fields are intended to mimic corresponding
fields that might be configured in the LDAP server. When Corporate Directory entries are saved,
the values are saved to the corresponding fields in the Contact Directory. See the Tech Bulletin
Configuring Contact Fields.
The field names for contact fields are drawn first from the dir.contact.attribute.x.label parameter,
then from the dir.corp.attribute parameter and if these are not configured, the defaults are used,
as shown below.
Table 8-33: Contact Directory elements
Parameter
Permitted Values
Default
feature.directory.enabled
0 or 1
1
If 0, the local contact directory is disabled. If 1, the directory is enabled.
feature.urlDialing.enabled
0 or 1
1
You can enter a url as the contact number if the Mode softkey is enabled when editing the "Contact" field of a
Contact Directory item.
If 0, URL/name dialing is not available. If 1, URL/name dialing is available from private lines. Note: If enabled,
unknown callers will be identified on the display by their phone's IP address.
dir.local.readonly1
0 or 1
0
If 0, the local contact directory can be edited. If 1, the local contact directory is read-only.
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Parameter
Permitted Values
Default
dir.local.contacts.maxNum1
1 to 9999
9999
Maximum number of contacts allowed in the local contact directory.
dir.search.field
0 or 1
0
If 0, search the contact directory by contact’s last name. If 1, search by first name.
dir.contact.attribute.x.label
String x characters
Primary,
Contact 2,
Contact 3,
Contact 4,
Contact 5
Sets the label for a contact field in the Contacts Directory. If x is 1, the default is Primary. The x value corresponds
to the Contact number. E.g. if the x value or the label parameter is 2, the default value for the contact field is
Contact 2. Note that the Primary contact number is used when setting a Speed Dial number to a contact or for
default dialing. This parameter aligns with the ct values configured in the contact directory elements. I.e ct2 is the
value for the Contact 2 field.
1
Change causes handset to restart or reboot.
Editing the Users’ MACaddress-directory.xml File
In certain rare cases, an administrator might want to edit the user’s contact directory file that has
been written to the central provisioning server. Simply open the file with an xml editor and make
the changes using the table above to identify the fields that need to be changed.
Specialized Caller Treatments
When setting up a seed directory or editing a user’s contact directory, several specialized caller
treatments can be applied. The user can also apply these treatments by opening and editing the
contact in the contact directory. See Table 8-32: Local Contact Directory elements.
Distinctive Incoming Call Treatment
You can apply distinctive treatment to specific calls and contacts in the contact directory. You
can set up distinctive treatment for each of the contacts by specifying a Divert Contact,
enabling Auto-Reject, or by enabling Auto-Divert for a specific contact.
Example Call Treatment Configuration
In the following example, the Auto Divert feature has been enabled in ad so that incoming calls
from John Doe will be diverted to SIP address 3339951954 as specified in dc. Incoming calls
from Bill Smith have been set to Auto Reject in ar and will be sent to voicemail.
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Note that if you enable both the Auto Divert and Auto Reject features, Auto Divert overrules
Auto Reject.
Speed Dial
You can link entries in the local contact directory to speed dial contacts on the handset. The
range of speed dial numbers is from 1 to 9999. Usually this link is made by the user when the
contact is set up or edited. However, it can be added in the seed directory or in the
MACaddress-directory.xml files through the sd element as described in the Contact Directory
Elements table.
Distinctive Call Waiting
You can use the alert-info values and class fields in the SIP header to map calls to distinct callwaiting types. You can apply three call waiting types: beep, ring, and silent. This feature
requires call server support.
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Table 8-34: Applying Distinctive Call Waiting
Parameter
Permitted Values
Default
voIpProt.SIP.alertInfo.x.class
See the list of ring
classes in Ring
Tones <rt/>
default
Alert-Info fields from INVITE requests will be compared against as many of these parameters as are specified
(x=1, 2, ..., N) and if a match is found, the behavior described in the corresponding ring class is applied.
Example Distinctive Call Waiting Configuration
In the following illustration, voIpProt.SIP.alertInfo.1.value is set to http://<SIP
headerinfo>. An incoming call with this value in the SIP alert-info header will cause the handset
to ring in a manner specified by voIpProt.SIP.alertInfo.x.class. In this example, the
handset will display a visual LED notification, as specified by the value visual.
Location Services (Ekahau)
You can use location services to send reports for Ekahau® Real-Time Location Systems (RTLS)
on the Spectralink handsets. You can select a transmit interval and enter a static IP address for
the Ekahau Positioning Engine™ (EPE). Location services are provided by the EPE 4.0 using
Ekahau Location Protocol (ELP). For more information, see the Ekahau website description:
Ekahau Real Time Location System.
Table 8-35: Enabling Location Services
Parameter
Permitted Values
Default
wifi.rtls.ekahau.address
IP-address
169.254.10.10
0 or 1
0
The IP address of the Ekahau Positioning Engine.
wifi.rtls.ekahau.enable
If 0, the Ekahua Real-Time Location System (RTLS) is disabled. If 1, the Ekahua RTLS is enabled.
wifi.rtls.ekahau.port
0 to 65535
8552
0 to 2
0
The port number of the Ekahau Positioning Engine.
wifi.rtls.ekahau.txInterval
The maximum time between transmit intervals. If set to 0, the transmit interval is 1-minute. If set to 1, the transmit
interval is 5-minutes. If set to 2, the transmit interval is 10-minutes.
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Parameter
Permitted Values
Default
wifi.rtls.ekahau.txIntervalSeconds
10-600
Null
Used in conjunction with Personal Alarms to control the interval with more precision. See the PersonalAlarms.cfg
template and Personal Alarms configurable parameters. This setting will override any existing
wifi.rtls.ekahau.txInterval parameter. If this setting is not defined or specified incorrectly, the existing
wifi.rtls.ekahau.txInterval value will be used.
During the alarm state, if RTLS is enabled, the handset will automatically use the shortest txInterval of 10 seconds
regardless of the setting of any parameter. Once the alarm is cleared, the txInterval setting will revert to its former
value.
Note that this setting will overrule any interval set in the handset Administration Settings menu.
Example Location Service Integration Configuration
To use RTLS, enable the wifi.rtls.ekahau.enable parameter, as shown next. All other
Ekahau parameter vales shown in the following example are the default values.
Microsoft Exchange Calendar Integration
Spectralink handsets can display the Microsoft Exchange 2007 and 2010 calendar. The
calendar gives you quick access to meeting information and you can dial in to conference calls.
To integrate the Microsoft Exchange Calendar features with your handset, configure the
parameters in the next table.
You can access the feature from the Applications menu on the Spectralink handsets.
You will need valid Microsoft Windows credentials to access the Microsoft Exchange Calendar
information on the handset. You can manage these credentials through the Login Credentials,
which are available through Home> Settings> Basic Settings> Login Credentials.
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You can view the calendar information in day or month format. The meeting details overlap the
calendar view.
All possible handset numbers that you can dial to place a call to the meeting will display in the
meeting details. You can automatically place a call by pressing a softkey.
A reminder pop-up is displayed 15 minutes before a scheduled meeting. You can dismiss the
reminder, select snooze to have the reminder pop up again and open the meeting details view.
A tone will be played along with the reminder pop-up.
Web Info: Using Microsoft Exchange Calendar Integration
For user instructions on how to use calendar integration, refer to the Spectralink
84-Series Wireless Handset User Guide.
Table 8-36: Enabling Microsoft Exchange Calendar Integration
Parameter
Permitted Values
Default
feature.exchangeCalendar.enabled1
0 or 1
0
If 0, the calendaring feature is disabled. If 1, the feature is enabled.
exchange.server.url1
String
Null
String
Null
The Microsoft Exchange server address.
exchange.meeting.phonePattern
The pattern used to identify phone numbers in meeting descriptions, where "x" denotes any digit and "|" separates
alternative patterns (for example, xxx-xxx-xxxx|604.xxx.xxxx).
exchange.meeting.reminderEnabled
0 or 1
1
If 0, meeting reminders are disabled. If 1, they are enabled.
1
Change causes handset to restart or reboot.
After you enable the feature, specify the Microsoft Exchange Server address in
applications.cfg template as shown next. In this example, a pattern has been specified for
meeting numbers. When you specify a pattern, any number in your meeting invitation that
matches the pattern will display on a meeting participants’ handsets as a softkey. Then,
participants can press the softkey to dial in to the meeting. You can specify multiple patterns,
separated by a bar. In the following example, two patterns are specified.
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Open Application Interface
Spectralink’s Open Application Interface (OAI) enables you to use the Spectralink handsets to
retrieve and respond to information on third-party computer applications.
Each handset that uses OAI features must be configured with its line number and MAC address
in the OAI Gateway so that it can register to the OAI Gateway and receive messages from it.
You can configure the OAI Gateway to recognize either the handset’s MAC address or a “virtual
MAC Address” substitute. The substitute is useful when deploying User Profiles where the
handset user could use different handsets on different shifts. In this scenario the MAC address
will not identify the user. Configuration of the oai.userid parameter allows you to configure an ID
for a specific user. That way that user has the some ID whenever they are logged in.
Web Info: Using the Spectralink 8000 OAI Gateway
OAI v2.2 is supported by the Spectralink 84-Series handsets. For more information,
see the Spectralink 8000 Open Applications Interface (OAI) Gateway
Administration Guide.
Table 8-37: Configuring the Open Application Interface (OAI)
Parameter
Permitted values
Default
oai.gateway.address
IP address
Null
String of eight hexadecimal characters
Null
The address of the OAI server.
oai.userId
The lower four bytes of the six-byte OAI handset identifier in the OAI gateway.
If the value is null or invalid, the handset identifies itself to the OAI gateway using the MAC address of the handset;
otherwise, the upper two bytes are zero and the lower four bytes are as specified.
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Example OAI Configuration
The following example shows the connection parameters you need to set for OAI
communications with Spectralink handsets. You will need to specify the OAI user ID and
gateway address.
If deploying User Profiles, specify the User ID in the login.cfg file:
Passwords – User and Administrator
The parameters in this section regulate access to the admin menus on the handsets and user
settings on the Web Configuration Utility. The handset will prompt you for a user or
administrator password before you can access certain menu options. If the handset requires the
administrator password, you may be able to use the user password, but you will be presented
with limited menu options. If the handset prompts you for the user password, you may use the
administrator password (you will see the same menus as the user). The Web Configuration
Utility is protected by the user and administrator password and displays different features and
options depending on which password you use.
The default user password is 123 and the default administrator password is 456. You should
change the administrator password from the default value. The admin password can be set in
the wireless.cfg template when the handsets are first deployed. You may want to change the
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user password for security reasons. Note that the user login password when deploying User
Profiles is a separate password type. Please refer to the User Profiles section for more
information about passwords and User Profiles.
Passwords are also used with Microsoft Skype for Business servers, with certain security
configurations and for access to configuration files using secure protocols. Please see the
corresponding section for more information about provisioning handsets in secure
environments.
Summary
Parameter
Used to:
sec.pwd.length.admin
Set the minimum length for the administrator password
sec.pwd.length.user
Set the minimum length for the user password
device.auth.localAdminPassword
Set the handset’s local administrator password
device.auth.localUserPassword
Set the handset’s local user password
Table 8-38: Local User and Administrator Password Settings
Parameter
Permitted Values
Default
sec.pwd.length.admin1
0-32
1
The minimum length for administrator passwords changed using the handset. Use 0 to allow null passwords.
sec.pwd.length.user1
0-32
2
The minimum length for user passwords changed using the handset. Use 0 to allow null passwords.
device.auth.localAdminPassword
string (32 character max)
456
The handset’s local administrative password. The minimum length is defined by sec.pwd.length.admin. If the
value is Null, 456 will be used.
device.auth.localUserPassword
string (32 character max)
Null
The handset user’s local password. The minimum length is defined by sec.pwd.length.user. If the value is Null,
123 will be used.
1
Change causes handset to restart or reboot.
Table 8-39: Password obfuscation
Parameter
Permitted Values
Default
webutility.outputKeysInClearText
0 or 1
0
Note: Parameter only available in Release 4.5 and later.
This parameter will cause passwords to be obfuscated when they are exported from the WebUI.
0 = Any key or passphrase output in an exported configuration file or backup file created using the WebUI, will be
output as “********” if it has been set and as “” if not.
1 = Any key or passphrase output in a exported configuration file or backup file created using the WebUI will be
output in clear text if it has been set and as “” if it has not.
NOTE: Passwords that are entered into menus on the handset are not shown at all. Only values that are entered
into Config files are affected.
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Affected parameters:
device.wifi.wep.key1
device.wifi.wep.key2
device.wifi.wep.key3
device.wifi.wep.key4
device.wifi.psk.key
device.wifi.wpa2Ent.password
device.sec.configEncryption.key
device.logincred.password
device.net.dot1x.password
device.pacfile.password
device.prov.password
device.auth.localAdminPassword
device.auth.localUserPassword
qbc.connect.passphrase
sec.TLS.customDeviceKey
tcpIpApp.ice.password
prov.login.defaultPassword
prov.login.localPassword
apps.statePolling.password
apps.push.password
dir.corp.password
diags.sshc.gateway.password
Use the device.set parameter for each device.x parameter.
Each <device/> parameter has a corresponding .set parameter that enables or
disables the value for that device parameter. You will need to enable the
corresponding .set parameter for each parameter you want to apply.
Personal Alarms
Workers can be at risk during security breaches or if personal incidents require immediate
attention. Spectralink 8441 and 8453 handsets offer personal monitoring and duress call
functionality, including “man down” alarms, “running” alarms and duress calls to an emergency
number. Coupled with a security alarm application program, real time location information from
the alarming handset can be displayed on security monitors and sent to other Spectralink
84-Series handsets for mobile response. The existing functionality of Location Services allows
an alarming handset’s location to be pinpointed so that aid can be directed to the exact scene.
When deployed in conjunction with a security alarm application, Spectralink Personal Alarms
provide unparalleled support for isolated workers or other at-risk personnel in potentially
threatening situations.
Duress call alarms can also be deployed within the functionality of the 8440 and 8452 models.
The emergency dial feature can be programmed to sound a local alarm through the built-in
speakerphone when an emergency call is dialed. Coupled with a security alarm application, this
duress alarm can be used to identify the handset, the user and the location of the alarming
handset.
Warning
The reliability of the Spectralink Personal Alarms application depends on the
functionality and reliability of the greater infrastructure – the wireless LAN, the LAN,
the call server, the central provisioning server, the server hosting location services,
the central security system and its servers, the correct configuration of the
Spectralink 84-Series handsets, correct installation and central provisioning
server(s), and thorough training of personnel.
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Four conditions of alerting can be activated and each is
configurable to the requirements of the facility. If any of
the running/tilt/still conditions occur, the handset will first
warn the user about an impending alarm and if the user
does not cancel the warning within a configurable
number of seconds the handset will start to alarm.
Duress
8440
8441
8452
8453
X
X
X
X
Running
X
X
Still
X
X
Tilt
X
X

Duress – should the user press the emergency button on the left side of the handset a
call is placed to an emergency number. A call can also be placed automatically if a
running/tilt/still alarm is initiated,

“Running” – the handset detects shaking in case a user runs for a configurable number
of seconds,

Tilt – the handset is tilted (not vertical) for a configurable number of seconds (aka “man
down”),

Still – the handset remains unmoved for a configurable number of seconds, potentially
indicating the user is no longer moving.
Administrator Configurable Options
Alarm detection and sensitivity parameters are configurable only by an administrator through the
configuration files on a central provisioning server. The personalAlarm.cfg file contains the
parameters that are listed in this document. Spectralink Personal Alarms configuration options
are not offered on the handset menus or through the Web Configuration Utility. They are only
configurable through the central provisioning server configuration files. See the Spectralink
84-Series Deployment Guide for a full explanation of how to use a central provisioning server for
Spectralink 84-Series handset provisioning and deployment.
Configurable options are

The Motion events -- Running, Tilt, and Still -- may be configured for degree of sensitivity
and duration before the warning becomes an alarm condition.

The Duress alert is a call to an emergency number when the user either a) presses the
Emergency Dial button on the left side of the handset twice within two seconds or b)
presses and holds the button for x seconds. This button must be configured to dial an
emergency number instead of its usual Push-to-talk function. The two features are
mutually exclusive.

The Motion events -- Running, Tilt, and Still -- can prompt an automatic emergency call.

The emergency call can be configured to force use of the speakerphone instead of
handset, headset or Bluetooth methods.

A Suspend function can be configured to allow the user to temporarily disable the
running/tilt/still sensing mechanism.
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
A tone/vibrate/LED flash alarm notification can be configured for the handset when the
warning/alarm event occurs.

An XML API alarm event may be enabled/disabled for alarm events. See Web
Application Parameters.

Already configured Location services are forced to the shortest txInterval of 10 seconds
during an alarm condition.

Tone/vibrate patterns for the Silent notification profile must be changed from the default
of “silent” to a warning and alarm tone and/or vibration. Otherwise the alert and warning
tones will not be played when the handset is in the Silent profile.

The feature is disabled by ensuring the sensitivity for all motion events is set to zero.
Note: Emergency call functionality
The ability to make an emergency call can be configured on every Spectralink
84-Series handset model and is not unique to Spectralink Personal Alarms which
are offered only on the 8441 and 8453 models. How to configure the Emergency
Dial feature is described in detail in the Spectralink 84-Series Deployment Guide.
Spectralink Personal Alarms can be configured to automatically generate an
emergency call when a Running/Tilt/Still alarm is triggered if the Emergency Dial
feature is also configured.
Diagram
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User Experience
Optimal user experience requires an understanding of which Spectralink Personal Alarm
features have been activated by the administrator. Users with 8441 and 8453 handsets should
be advised if Running, Tilt, and/or Still alarms are enabled and all users should be advised if the
Duress/Emergency Call functionality is enabled. All users need to know what the result will be if
an alarm is activated, either intentionally or unintentionally. The user cannot permanently
disable Spectralink Personal Alarms or turn it on and off.
The Spectralink Personal Alarms icon on the status bar indicates that the Spectralink Personal
Alarms functionality is active. Colors and shapes signify its status:
= Spectralink Personal Alarms monitoring active
= Spectralink Personal Alarms alarm triggered
= Spectralink Personal Alarms temporarily suspended (if configured)
No icon = handset in charger (or the feature has been disabled by the administrator)
Note: Personal alarm icon for the 8440/8452 handsets
The Spectralink 8440/8452 models display the red icon during an Emergency Call
until the alarm is cleared. No other icon is displayed.

A security application may override a user or the notification profile setting.

The administrator can program the warning and alarm alerts to sound even during the
silent notification profile by overriding the default configuration for the silent profile. See
the np.xxx.alert setting later in this document.

The user can suspend Spectralink Personal Alarms running/tilt/still sensing mechanism
for a period of time by selecting the Suspend monitor option on the Features softkey
menu. The user may restart Spectralink Personal Alarms before the time elapses by
selecting the Resume monitor option. Spectralink Personal Alarms suspend is a
configurable option controlled by the administrator. The Duress button functionality is not
suspended.
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
If a running/tilt/still event has been detected, the handset goes into a “warning” state in
which it presents a warning window and sounds the alert and/or vibrates. The user can
cancel an impending alarm during the warning state by pressing the Cancel softkey and
the alert audio will turn off and an alarm will not be sent to any configured security
application and the handset resumes monitoring.

If the user does not cancel during the running/tilt/still warning state, the handset goes
into the alarm state. The alarm sounds and the alarm window appears. If configured, an
XML alarm event is generated and sent to a security application, and an emergency call
is placed.
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
When you are in an alarm state, the alerting ring tone or vibration can be turned off by
selecting the Silence alarm option on the Features softkey menu. Only the
audio/vibration is stopped. The alarm state will continue until cleared. The flashing LED
continues until the alarm state is cleared.

Once an alarm state is reached, the user cannot retract the alarm but can cancel the
alarming state and return the handset to the monitoring state by selecting the Clear
alarms option on the Features softkey menu. Clear alarm returns the handset to the
monitoring state and sends an xml notification of the end of the alarm. If an emergency
call has been placed as well, the call must be terminated as would any call.

If the Duress alarm is activated, a similar process occurs as for running/tilt/still alarms
except there is no warning state and an emergency call is placed when the button is
pressed twice within two seconds or with one long press. An emergency call preempts
any existing active call. The emergency call is ended by pressing the END key or by the
far end hanging up. The handset remains in the alarm state until manually cleared by the
user.
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
New alarm detection is inactive when the handset is connected to a charger – USB, Dual
Charger or Speakerphone Dock. No icon displays during the inactive state. Once the
handset is removed from the charger, monitoring begins and the icon turns green.

Alarms must be manually cleared. They cannot be cleared by placing the handset into a
charger.
Integration with Third Party Applications
Customers will receive the most benefit with Spectralink Personal Alarms by integrating the
handsets with monitoring applications and real-time location-tracking (RTLS) systems.
Monitoring applications, such as lone worker systems, can monitor workers in real time and
implement sophisticated response management procedures such as notifying response teams
in the case of an alarm. Such applications can be developed by third parties and customized
according to a facility’s needs.
Location tracking systems can provide pin-point accuracy in real-time to reduce the time spent
finding the incident location. The following section describes several integration options when
employing mandown handsets.
See the Spectralink 84-Series Web Developer’s Guide for full information about API
applications.
Note: Web applications parameters
See Web Application Parameters for information about applications parameters.
Implementation without application integration or Location Systems
Running/tilt/still alarms can be configured along with the automatic emergency call (aka
Emergency Dial). In the event of any one of the three motion alarms being activated this will
prompt an automatic emergency call to the configured emergency dial number. No application
integration or RTLS is required for this first level of implementation. The emergency call
provides notification of an alarm event. If the handset is configured for local notification, when
an alarm is triggered the handset will emit noises to help rescuers locate the handsets.
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Implementation with Location Systems
All Spectralink 84-Series handsets support integration with the Ekahau RTLS system. Handsets
send periodic location information to an Ekahau server allowing the server to pinpoint the
handsets’ location. To maximize battery life, an administrator may set this update interval
conservatively, e.g. 1 minute. In the event the handset enters an alarm state, the handset can
be configured send updates more frequently, e.g. 10 seconds. This allows the Ekahau RTLS
system to provide updated positions of the alarming handset more frequently. Additionally, if the
Ekahau system is integrated into a management application these positions can be sent to
responders.
Additional 3rd-party RTLS systems, e.g. Aeroscout, may also be able to provide the location of
Spectralink 84-Series handsets.
Examples of implementation with third-party applications
Third-Party applications, such as Lone Worker Systems, will use the Spectralink 84-Series XML
API to receive handset events and provide notifications to the user. Specific application
capabilities are dependent on the developer and customer requirements. A typical application
may monitor 84-Series handsets for Spectralink Personal Alarms events, provide alarm
escalation logic to control who should receive notification, and then notify personnel providing
relevant alarm information. Obviously the application sophistication and degree of integration
with other systems may vary.
These examples assume that one or more of the Running, Tilt, and Still conditions are
configured and that the Duress call is also configured (aka Emergency Dial). The automatic
emergency call may or may not be configured.
1
When a handset enters the alarm state, if XML notification for alarms is configured, the
handset will send an alarm event (AlarmNotificationEvent) to the application. A
receiving application can use this to detect a handset has raised an alarm and perform
the appropriate alarm response. The parameters set as part of the alarm event are
covered in the XML API Detail section and the Spectralink 84-Series Wireless
Telephone Administration Guide at the Spectralink Support Website.
2
If configured, the emergency call triggered by the handset may be valuable identifying
the severity of the situation and if the user of the alarming handset is responsive. And/or
the application can use the XML API to send a message to the alarming handset to
determine if the user intended to send an alarm and if the user is responsive or not.
3
If the application is integrated with an RTLS system, the application may attempt to
locate the alarming handset before notifying the appropriate personnel to respond.
4
If the user of the alarming handset clears the alarm, if XML alarm notification is
configured the handset will send an alarm notification to the application indicating the
alarm has been cleared.
5
Assuming the application or application user needs to notify other personnel to respond
to the alarm, the application can send appropriate messages to these response
personnel. The XML API allows an application to deliver either simple messaging or
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complex information such as a web-page with a site plan map showing the approximate
location of the user.
6
To ensure a user is aware of these alarm response notifications the application can even
override a user’s volume settings.
Personal Alarms configurable parameters
Alarm detection and sensitivity parameters are configurable only by an administrator through the
configuration files on a central provisioning server. The Spectralink Personal Alarms .cfg file
contains the parameters that are listed in this document.
Caution
When a motion timeout setting is changed, the new value does not take effect until
the handset detects a new occurrence of that motion condition and starts timing its
duration to determine if warning should be raised. If the handset has already
detected that motion condition (e.g. handset is currently tilted) and is currently
timing the duration, the new timeout value will not be used until the handset exits
that motion condition state (e.g. no longer tilted) and then enters it again.
Table 8-40: Personal alarms
Parameter
Permitted values
Default
up.PersonalAlarm.still.sensitivity
0-7
0
0 disable, 1 least sensitive, 7 most sensitive
As the sensitivity increases the handset must be more stationary to trigger a no movement alarm. The specific
sensitivity setting appropriate for a given application is site/user specific.
up.PersonalAlarm.still.timeout
5-300 (secs)
7
The still condition must persist for this amount of time before a warning occurs.
up.PersonalAlarm.tilt.sensitivity
0-7
0
0 disable, 1 least sensitive, 7 most sensitive
Indicates the degree of tilt from horizontal that triggers an alarm. This setting
is modified by the timeout setting that determines the amount of time the
position (or lower) is maintained before an alarm is triggered.
In the most sensitive position setting, a slightly leaning position will trigger the
alarm.
In the least sensitive position setting, the body must be nearly horizontal
before an alarm triggers.
Experiment with these settings until you find the right sensitivity for your
facility.
Horizontal movement can interfere with the handset's ability to sense the tilt
condition. If the tilt alarm is the only motion alarm configured, we recommend
using a relatively high sensitivity setting.
up.PersonalAlarm.tilt.timeout
5-300 (secs)
7
The tilt condition must persist for this amount of time before a warning occurs.
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Parameter
Permitted values
Default
up.PersonalAlarm.running.sensitivity
0-7
0
0 disable, 1 least sensitive, 7 most sensitive
As the sensitivity increases the handset requires less movement to trigger a running alarm. The specific sensitivity
setting appropriate for a given application is site/user specific.
up.PersonalAlarm.running.timeout
5-60 (secs)
7
The running condition must persist for this amount of time before a warning occurs.
up.PersonalAlarm.suspendMonitoring
0- 300 (secs)
0
0, no suspension of monitoring is allowed, otherwise duration in seconds of the amount of time during which
motion event monitoring is disabled.
up.PersonalAlarm.warningTimeout
5-60 (secs)
10
This is the number of seconds a user has to disable the warning on the handset before the warning automatically
triggers an alarm.
up.PersonalAlarm.notificationEnable
0-1
1
Disable/enable local alarm notification (tone, vibrate, LED flash) on motion event detection in both warning and
alarm states.
up.PersonalAlarm.emergencyDialEnable
0-1
0
This parameter requires correct configuration of the Emergency Dial feature as detailed in the Spectralink
84-Series Administration Guide. This parameter is disabled by default.
If enabled, an emergency call is automatically placed when a running/tilt/still alarm occurs.
If disabled, and a running/tilt/still alarm occurs the emergency call will NOT be placed but the alarm state will be
activated.
Either way, the handset will transition into the alarm state and will generate an XML alarm notification if
apps.telNotification.alarmEvent is enabled.
up.PersonalAlarm.emerDialForceSpeakerPhone
0-1
1
This parameter requires correct configuration of the Emergency Dial feature as detailed in the Spectralink
84-Series Administration Guide. This parameter is enabled by default.
Allows you to force the speakerphone on an emergency call. If 1 force speakerphone on emergency call, if 0 use
normal audio termination routing rules (headset if connected, …).
ptt.emergencyDial.emergencyDialEnable
0-1
1
This parameter requires correct configuration of the Emergency Dial feature as detailed in Emergency . This
parameter is enabled by default.
If enabled, an emergency call will be placed if the duress button is pressed twice within two seconds.
If disabled, and the duress button is pressed twice within two seconds, the emergency call will NOT be placed but
the alarm state will be activated.
ptt.emergencyDial.longKeypressEnable
0-1
0
This parameter enables the long press method of activating the Emergency Dial function. It is disabled by default.
When set (1), it overrides the default double press method (0).
ptt.emergencyDial.longKeypressDuration
1,2,3,4
2
If the long press method of activating the Emergency Dial function is enabled, this parameter sets the length of
time in seconds that the key must be pressed in order for a call to be initiated.
ptt.emergencyDial.notificationEnable
0-1
0
This parameter requires correct configuration of the Emergency Dial feature as detailed in Emergency . This
parameter is disabled by default.
Disable/enable local alarm notification (tone,vibrate,LED flash) on manual button press (a duress or emergency
dial call). This parameter works in conjunction with Emergency Dial activation and applies to all handset models.
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Parameter
Permitted values
Default
wifi.rtls.ekahau.txIntervalSeconds
10-600
Null
Used in conjunction with Location Services server to control the interval with more precision. See the RTLS.cfg
template and Location Services (Ekahau). This setting will override any existing wifi.rtls.ekahau.txInterval
parameter. If this setting is not defined or specified incorrectly, the existing wifi.rtls.ekahau.txInterval value will be
used.
During the alarm state, if RTLS is enabled, the handset will automatically use the shortest txInterval of 10 seconds
regardless of the setting of any parameter. Once the alarm is cleared, the txInterval setting will revert to its former
value.
Note that this setting will overrule any interval set in the handset Administration Settings menu.
np.xxx.alert.PersonalAlarm.tonepattern
Any tone (see se.pat.misc)
np.xxx.alert.PersonalAlarm.vibration
0 or 1
np.xxx.alert.PersonalWarning.tonepattern
Any tone (see se.pat.misc)
np.xxx.alert.PersonalWarning.vibration
0 or 1
Normal = misc2
silent = silent
meeting = silent
custom1 = custom2
0
Normal = misc1
silent = silent
meeting = silent
custom1 = custom1
0
If a consistent audible alarm or vibration is desired during the warning or alarm period, no matter the profile, the
behavior must be set in the configuration file. These tones cannot be overridden by the user as these alerts do not
appear on the editable alert menu for the notification profiles. This is especially important for the silent notification
profile as the silent notification profile default behavior is silent for all alerts.
Alert settings can be programmed in the personal alarm application to play a tone pattern or vibration according to
the notification profile. See Spectralink 84-Series Wireless Telephones Web Developer’s Guide for more
information.
XML API Detail
An XML API allows you to use the telephone event notifications to develop a response
application when the Spectralink Personal Alarms alarm is triggered.
The Spectralink 84-Series Web Application Developers’ Guide explains how to install and use
the Spectralink 84-Software Development Kit (SDK) to plan, create, and develop Web
applications that will run on Spectralink 84-Series Wireless Telephones using Spectralink
software 4.2.x or later. Consult that reference for more information.
Viewing an Alarm Event
The configuration parameter for this new event type is in the applications.cfg file.
Table 8-41: Alarm events
Parameter
Permitted Values
Default
apps.telNotification.alarmEvent
0 or 1
0
If 0, alarm event notification is disabled. If 1, notification is enabled. When this parameter is enabled, an XML
notification is sent when an alarm event occurs. Alarm events occur when Running, Tilt, and Still alarms go off
and when Duress/Emergency Calls are made.
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The Alarm event notifies an application that a (personal security) alarm condition has been
detected or cleared on the handset.
Use the following format when viewing the alarm event:
<PolycomIPPhone>
<AlarmNotificationEvent>
<PhoneIP>172.29.71.157</PhoneIP>
<MACAddress>00907a0e4459</MACAddress>
<BSSID>0023ebe4ebaf</BSSID>
<StillAlarm>0</StillAlarm>
<TiltAlarm>1</TiltAlarm>
<RunningAlarm>0</RunningAlarm>
<DuressAlarm>0</DuressAlarm>
<TimeStamp>2012-12-10T08:11:25-07:00</TimeStamp>
</AlarmNotificationEvent>
</PolycomIPPhone>
Where
Phone IP is the IP address of the handset
MAC Address is the MAC address of the handset
BSSID is the MAC address of the AP the handset is currently using
StillAlarm is the current state of this alarm detector 0 = no alarm, 1 = alarm
TiltAlarm is the current state of this alarm detector 0 = no alarm, 1 = alarm
RunningAlarm is the current state of this alarm detector 0 = no alarm, 1 = alarm
DuressAlarm is the current state of this alarm detector 0 = no alarm, 1 = alarm
TimeStamp is the recorded time of the event
Configuration Template
The PersonalAlarm.cfg template is provided with the software download in the Config>Features
folder. Use it to as a starting point to customize your deployment of this feature.
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Phone Lock
The Phone Lock feature allows you to add an option to the Features softkey menu to lock the
handset and require a password to unlock it. This can be important in certain settings where
security is an important factor in handset deployment. Keypad lock is available for inadvertent
key presses. This is a more advanced security feature.
Please review the Enhanced Feature Keys section for complete information about configuring
additional menu keys. Enhanced Feature Keys must be activated to configure this feature.
When a handset is in a call, IM or OAI session and another call comes in, the handset behaves
normally and only locks when it returns to the idle state for the configured amount of time.
A handset may be set to alert when it is locked or it may be set to DND. The following table
details the different handset behaviors when Alert or DND is turned on.
Table 8-42: Phone Lock Behavior
Alert
DND
Incoming Phone
call
Call Alert screen with caller ID info. User
presses green key and is prompted for user
PIN, on successful entry call is answered.
Phone automatically locks on end of call.
Voice call goes right to voice mail. Phone
remains locked.
Incoming OAI
OAI alert screen (app can put more info on
screen while ringing once ring ack’d). User
presses green key and is prompted for user
No OAI alert screen displayed. Phone sends
“user did not respond to ring” response
immediately so next level escalation if
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Incoming IM
Alert
DND
PIN, on successful entry call is answered.
Phone automatically locks on end of call.
appropriate can begin right away. Phone
remains locked.
IM alert screen with first IM message
contents. User presses green key and is
prompted for user PIN, on successful entry
IM session is on hold. User presses resume
and IM session proceeds. Phone
automatically locks on end of call.
No IM alert screen displayed. Phone sends a
“486” response which means “The user was
contacted but is currently not willing or able to
take the call.”
The first IM message is added to the
conversation list and the missed message count
is incremented so the user can see they missed
a message.
Phone remains locked.
Table 8-43: Phone Lock Parameters
Parameter
Permitted Values
Default
phoneLock.authorized.x.description
The name or description of an authorized number
phoneLock.authorized.x.value
The number or address for an authorized call
recipient.
String
Null
string
Null
The numbers configured by these parameters appear on the Authorized Call menu when the New Call softkey is
pressed while the phone is locked.
Up to five (x=1 to 5) authorized contacts can be configured. Each contact requires a description that displays on
the screen, and a phone number or address value for the handset to dial.
These calls will only go through if the phone is registered and the numbers are recognized by the
call server.
Additional numbers configured in the dialplan may also appear on the Authorized Call menu. See Emergency Dial
via Authorized Call menu for additional information.
phoneLock.browserEnabled
0 or 1
0
If 0, the browser or browser is not displayed while the handset is locked and the handset cannot receive any
updates from a browser application. If 1, the browser is displayed while the handset is locked.
phoneLock.dndWhenLocked
0 or 1
0
If 0, the handset can receive calls while it is locked. If 1, the handset enters Do-Not-Disturb mode while it is locked.
The user can enable DND even if this parameter is set to 0 which will cause the handset to demonstrate the
behavior described in the above table until the user disables DND.
phoneLock.enabled1
0 or 1
0
If 0, the handset lock feature is disabled. If 1, the handset lock feature is enabled. Note: To ‘unlock’ the handset
remotely (in conjunction with deleting/modifying the overrides files), disable and re-enable this parameter.
phoneLock.idleTimeout
0 to 65535
0
The amount of time (in seconds) the handset can be idle before it automatically locks. If 0, automatic locking is
disabled.
phoneLock.lockState
0 or 1
0
The value for this parameter indicates whether the handset is locked or unlocked and changes each time you lock
or unlock the handset. If 0, the handset is unlocked. If 1, the handset is locked. Note that the handset stores and
uploads the value each time it changes via the MAC-phone.cfg. You can set this parameter remotely using the
Web Configuration Utility.
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Parameter
Permitted Values
Default
phoneLock.powerUpUnlocked
0 or 1
0
Use this parameter to override phoneLock.lockState. If 0, the handset retains the value in
phoneLock.lockState. If 1, you can restart, reboot, or power cycle the handset to override the value for
phoneLock.lockState in the MAC-phone.cfg and start the handset in an unlocked state. You can then lock
or unlock the handset locally. Spectralink recommends that you do not leave this parameter enabled as its use is
primarily administrative.
1
Change causes handset to restart or reboot.
Provisional Polling of Spectralink Handsets
You can control how your handset handles automatic provisioning by configuring the
parameters in Table 8-44: Provisional Polling of Spectralink handsets.
You can set the handset's automatic provisioning behavior to be:
 Absolute
The handset polls at the same time every day.
 Relative The handset polls every x seconds, where x is a number greater than 3600.
 Random The handset polls randomly based on a time interval you set.
○ If the time period is less than a day or equal to one day, the first poll is at a random time
between the handset starting up and the polling period. Afterwards, the handset will poll
every x seconds.
○ If you set the polling period to be greater than one day, the handset polls on a random
day based on the handset’s MAC address.
For example:
 If prov.polling.mode is set to rel and prov.polling.period is set to 7200, the
handset polls every two hours.
 If prov.polling.mode is set to abs and prov.polling.time is set to 04:00, the
handset polls at 4am every day.
 If prov.polling.mode is set to random, prov.polling.period is set to 86400,
prov.polling.time is set to 01:00, prov.polling.timeRandomEnd is set to 05:00, the
handset polls randomly between 1am and 5am every day. This parameter is only used
when prov.polling.mode is set to random.

If prov.polling.mode is set to abs and prov.polling.period is set to 2328000,
the handset polls every 20 days.
Table 8-44: Provisional Polling of Spectralink handsets
Parameter
Permitted Values
Default
prov.polling.enabled
0 or 1
1
If 0, the provisioning server is not automatically polled for upgrades. If 1, the provisioning server is polled.
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Parameter
Permitted Values
Default
prov.polling.mode
abs, rel, random
abs
The polling mode.
abs The handset polls every day at the time specified by prov.polling.time.
rel The handset polls after the number of seconds specified by prov.polling.period.
random The handset polls at random between a starting time set in prov.polling.time and an end time set
in prov.polling.timeRandomEnd. Note that if you set the polling period in prov.polling.period to a time
greater than 86400 (one day) polling occurs on a random day between the start and end times based on the
handset’s MAC address.
prov.polling.period
integer> 3600
86400
The polling period in seconds. The polling period is rounded up to the nearest number of days in absolute mode. In
relative mode, the polling period starts once the handset boots. In random mode, if this is set to a time greater than
86400 (one day) polling occurs on a random day based on the handset’s MAC address.
prov.polling.time
hh:mm
03:00
The polling start time. Used in absolute and random modes.
prov.polling.timeRandomEnd
hh:mm
Null
The polling stop time. Only used in random mode.
Example Provisional Polling Configuration
The following illustration shows the default sample random mode configuration for the
provisional polling feature in the everything.cfg file.
Push-to-talk and Group Paging
The Push-to-talk (PTT) and Group Paging features are supported on all Spectralink 84-Series
handset models.
The Group Paging feature enables pages —one-way audio announcements — to users
subscribed to a page group. Paging mode was originally intended primarily for desktop phones
but has some use for Wi-Fi handsets that may or may not also be using PTT. In Page mode,
announcements play only through the handset’s speakerphone.
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The Push-to-talk (PTT) feature is a collaborative tool that enables users to exchange radio
broadcasts to other users subscribed to a PTT channel. In PTT mode, the handset behaves
like a walkie-talkie; users can broadcast audio to a PTT channel and recipients subscribed to
that channel can respond to your message. PTT broadcasts can be transmitted using the
handset, headset, or speakerphone. They can be rejected, placed on hold and ended at any
time. PTT broadcasts can be received on the speakerphone, handset, and headset.
Administrators must enable Paging and PTT before users can subscribe to a page group or PTT
channel. You can enable one of these features or you can operate both simultaneously. Paging
and PTT each have 25 groups/channels you can enable.
Note that you can enter a display name for sent PTT broadcasts in ptt.displayName and for
sent page announcements in ptt.pageMode.displayName .
Web Info: Using a Different IP multicast address
The Push-to-talk and Group Paging features use a IP multicast address. If you
want to change the default IP multicast address, ensure that the new address does
not already have an official purpose as specified in the IPv4 Multicast Address
Space Registry.
Push-to-talk
You specify the same IP multicast address in the parameter ptt.address for both PTT and
Paging mode. PTT administrator settings are located in the site.cfg template file. PTT channels
settings are located in the features.cfg template file.
Tip: Compatibility With Earlier/Later Spectralink Handsets
You can configure the PTT feature to be compatible with the earlier Spectralink
8020 and 8030 Series Wireless Handsets by setting the
ptt.compatibilityMode parameter to ‘1’. If you are deploying Spectralink 87Series handsets and need to maintain PTT compatibility, this parameter must be
set to 0 as the 87-Series handset does not support the G.726 codec.
Table 8-45: Enable Push-to-talk
Parameter
Permitted Values
Default
ptt.pttMode.enable
0 or 1
0
If 0, push-to-talk is disabled. If 1, push-to-talk is enabled.
ptt.address
multicast IP address
224.0.1.116
The multicast IP address to send page audio to and receive page audio from. Use default.
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Parameter
Permitted Values
Default
ptt.compatibilityMode
0 or 1
1
If 0, the PTT protocol behavior is not compatible with Spectralink handset models 8020/8030 or older. If 1, all PTT
protocol behavior is compatible with the older Spectralink handsets, even if some configuration parameters are
incompatible. For example, if this parameter is enabled and ptt.codec is set to G.722, the G.726QI codec will be
used for outgoing PTT audio to maintain compatibility.
If you are deploying Spectralink 87-Series handsets and need to maintain PTT compatibility, this parameter must
be set to 0 as the 87-Series handset does not support the G.726 codec.
ptt.defaultChannel
1 to 25
1
The PTT channel used to transmit an outgoing page if the user does not explicitly specify a channel.
ptt.payloadSize
10 to 80
20
The audio payload size in milliseconds. Use default.
ptt.priorityChannel
1 to 25
24
1 to 25
25
0 or 1
1
0 or 1
1
0 or 1
1
string
ch1: All, ch24: Priority,
ch25: Emergency, others: Null
0 or 1
ch1, 24, 25: 1, others: 0
The channel assigned for priority broadcasts.
ptt.emergencyChannel
The channel assigned for emergency broadcasts.
ptt.channel.x.available
Make the channel available to the user
ptt.channel.x.allowTransmit
Allow outgoing broadcasts on the channel
ptt.channel.x.allowReceive
Allow incoming broadcasts on the channel
ptt.channel.x.label
The label to identify the channel
ptt.channel.x.subscribed
Subscribe the handset to the channel
A push-to-talk channel x, where x= 1 to 25. The label is the name used to identify the channel during broadcasts.
If available is disabled (0), the user cannot access the channel or subscribe and the other channel parameters
will be ignored. If enabled, the user can access the channel and choose to subscribe.
If allowTransmit is disabled (0), the user cannot sent PTT broadcasts on the channel. If enabled, the user may
choose to send PTT broadcasts on the channel.
If allowReceive is disabled (0), the user cannot receive PTT broadcasts on the channel. If enabled, the user
may choose to receive PTT broadcasts on the channel.
If subscribed is disabled, the handset will not be subscribed to the channel. If enabled, the handset will
subscribe to the channel.
Table 8-46: Push-to-talk and Group Paging Parameters
Parameter
Permitted Values
Default
ptt.address
multicast IP address
224.0.1.116
The multicast IP address to send page audio to and receive page audio from.
ptt.callWaiting.enable
0 or 1
0
If 0, incoming PTT sessions do not produce standard call waiting. If 1, incoming PTT sessions produce standard
call waiting behavior on the active audio channel.
ptt.compatibilityMode
0 or 1
1
If 0, the PTT protocol behavior is not compatible with Spectralink handset models 8020/8030 or older. If 1, all PTT
protocol behavior is compatible with the older Spectralink handsets, even if some configuration parameters are
incompatible. For example, if this parameter is enabled and ptt.codec is set to G.722, the G.726QI codec will be
used for outgoing PTT audio to maintain compatibility.
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Parameter
Permitted Values
Default
ptt.emergencyChannel.volume
-57 to 0
-10
The volume of emergency pages relative to the maximum speakerphone volume of the handset. Positive values
are louder than the maximum and negative values are quieter. The gain to use for emergency page/PTT is the
maximum termination gain plus this parameter. Note: To enter a negative number, press the * key first.
ptt.port
0 to 65535
5001
Parameter
Permitted Values
Default
ptt.allowOffHookPages
0 or 1
0
The port to send audio to and receive audio from.
Table 8-47: Additional Push-to-talk parameters
If 0, PTT broadcasts will not play out on the handset during an active call — except for Priority and Emergency
pages. If 1, PTT broadcasts will play out on the handset during an active call.
ptt.codec
G.711mu, G.726QI, G.722
G.722
The audio codec to use for outgoing PTT broadcasts. Incoming PTT audio will be decoded according to the codec
specified in the incoming message.
ptt.defaultChannel
1 to 25
1
The PTT channel used to transmit an outgoing page if the user does not explicitly specify a channel.
ptt.displayName
string
Null
This display name is shown in the caller ID field of outgoing pages. If Null, the value from reg.1.displayName
will be used. If the reg.1 display name is also Null, the handset’s MAC address will be used.
ptt.emergencyChannel
1 to 25
25
10 to 80
20
1 to 25
24
0 or 1
0
The channel assigned for emergency pages.
ptt.payloadSize
The audio payload size in milliseconds.
ptt.priorityChannel
The channel assigned for priority pages.
ptt.pttMode.enable
If 0, push-to-talk is disabled. If 1, push-to-talk is enabled.
Group Paging
You specify the same IP multicast address in the parameter ptt.address for both PTT and
Paging mode. Paging administrator settings are located in the site.cfg template file. Page group
settings are located in the features.cfg template file.
Table 8-48: Group Paging Parameters
Parameter
Permitted Values
Default
ptt.pageMode.allowOffHookPages
0 or 1
0
If 0, group pages will not play out on the handset during an active call — except for Priority and Emergency pages.
If 1, group pages will play out on the handset during an active call.
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Parameter
Permitted Values
Default
ptt.pageMode.codec
G.711Mu, G.726QI, or
G.722
G.722
The audio codec to use for outgoing group pages. Incoming pages will be decoded according to the codec
specified in the incoming message.
ptt.pageMode.defaultGroup
1 to 25
1
The paging group used to transmit an outgoing page if the user does not explicitly specify a group.
ptt.pageMode.displayName
up to 64 octet UTF-8
string
Null
This display name is shown in the caller ID field of outgoing group pages. If Null, the value from
reg.1.displayName will be used. If the reg.1 display name is also Null, the handset’s MAC address will be
used.
ptt.pageMode.emergencyGroup
1 to 25
25
0 or 1
0
The paging group to use for emergency pages.
ptt.pageMode.enable
If 0, group paging is disabled. If 1, group paging is enabled.
ptt.pageMode.group.x.available
Make the group available to the user
ptt.pageMode.group.x.allowTransmit
Allow outgoing announcements to the group
ptt.pageMode.group.x.allowReceive
Allow receipt of announcements to the group
ptt.pageMode.group.x.label
The label to identify the group
0 or 1
1
0 or 1
1
0 or 1
1
string
ch24: Priority,
ch25: Emergency, others: Null
ptt.pageMode.group.x.subscribed
Subscribe the handset to the group
0 or 1
ch1, 24, 25: 1, others: 0
A page mode group x, where x= 1 to 25. The label is the name used to identify the group during pages.
If available is disabled (0), the user cannot access the group or subscribe and the other page mode group
parameters will be ignored. If enabled, the user can access the group and choose to subscribe.
If allowTransmit is disabled (0), the user cannot send outgoing pages to the group. If enabled, the user may
send outgoing pages.
If allowReceive is disabled (0), the user cannot receive incoming pages to the group. If enabled, the user may
receive incoming pages.
If subscribed is disabled, the handset will not be subscribed to the group. If enabled, the handset will subscribe
to the group.
ptt.pageMode.payloadSize
10, 20, ..., 80
milliseconds
20
1 to 25
24
0 to 65535
60
The page mode audio payload size.
ptt.pageMode.priorityGroup
The paging group to use for priority pages.
ptt.pageMode.transmit.timeout.continuation
The time (in seconds) to add to the initial timeout (ptt.pageMode.transmit.timeout.initial) for
terminating page announcements. If this value is non-zero, an Extend softkey will display on the handset.
Pressing the Extend softkey continues the initial timeout for the time specified by this parameter. If 0,
announcements cannot be extended.
ptt.pageMode.transmit.timeout.initial
0 to 65535
0
The number of seconds to wait before automatically terminating an outgoing page announcement. If 0, page
announcements will not automatically terminate.
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Quick Barcode Connector Application
If you are using Spectralink 8450/8452 handsets, the Spectralink Quick Barcode Connector™
(QBC) application enables you to capture and decode barcode patterns with the handset and
transfer the data to applications running on one or more host computers. Data can be
transferred in single endpoint mode (one host computer) or multiple endpoint mode (many host
computers). Please refer to the Quick Barcode Connector Administration Guide for complete
information.
Registrations
Each registration can optionally be associated with a private array of servers for completely
segregated signaling. The Spectralink handsets support six registrations.
In the following tables, x is the registration number which can be from 1-6.
Table 8-49: Registration Parameters
Parameter
Permitted Values
Default
reg.x.acd-login-logout
reg.x.acd-agent-available
0 or 1
0 or 1
0
0
If both ACD login/logout and agent available are set to 1 for registration x, the ACD feature will be enabled for that
registration.
reg.x.address
string address
Null
The user part (for example, 1002) or the user and the host part (for example, 1002@Spectralink.com) of the
registration SIP URI.
reg.x.applyServerDigitMapLocally
0 or 1
0
If 1 and reg.x.server.y.specialInterop is set to lync2010, lync2013 or Skype for Business, the
handset uses the dialplan from the Microsoft Lync Server. Any dialed number will apply the dial plan locally.
If 0, the dialplan from the Microsoft Lync Server is not used.
reg.x.auth.domain
string
Null
The domain of the authorization server that is used to check the user names and passwords.
reg.x.auth.password
string
Null
The password to be used for authentication challenges for this registration. If the password is non-Null, it will
overrule the password entered into the Authentication submenu on the Settings menu of the handset.
reg.x.auth.userId
string
Null
User ID to be used for authentication challenges for this registration. If the User ID is non-Null, it will overrule the
user parameter entered into the Authentication submenu on the Settings menu of the handset.
reg.x.csta
0 or 1
0
If 0, the uaCSTA (User Agent Computer Supported Telecommunications Applications) feature is disabled. If 1,
uaCSTA is enabled (overrules the global parameter voIpProt.SIP.csta).
reg.x.dialPlanName
String
Null
If reg.x.server.y.specialInterop is set to lync2010, lync2013 or Skype for Business, the
dialplan name from the Microsoft Lync Server is stored here. Each registration has its own name for this dialplan.
Note: Do not change this parameter if set by Microsoft Lync.
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Parameter
Permitted Values
Default
reg.x.displayName
UTF-8 encoded
string
Null
The display name used in SIP signaling and/or the H.323 alias used as the default caller ID.
reg.x.ice.turn.callAdmissionControl.enabled
0 or 1
0
If 0, call admission control is disabled. If 1, call admission control is enabled for calls using the Microsoft Skype for
Business or Microsoft Lync Server 2013 or 2010..
When deploying both 84-Series and 87-Series handsets in the same facility using the same Wireless LAN, Wi-Fi
Multimedia Admission Control (aka access control, AC or WMM-AC) must be disabled in any handset parameters
and APs as it is not supported by 87 Series handsets. Any parameter that requires or enforces AC must be
disabled.
reg.x.label
UTF-8 encoded
string
Null
The text label that displays next to the line key for registration x. If Null, the user part of reg.x.address is used.
reg.x.lcs
0 or 1
0
If 0, the Microsoft Live Communications Server (LSC) is not supported for registration x. If 1, LSC is supported.
reg.x.lineKeys
1 to 6
1
Specify the number of line keys to use for a single registration. The maximum number of line keys is 6.
reg.x.lisdisclaimer
string, 0 to 256
characters
Null
This parameter sets the value of the location policy disclaimer. For example, the disclaimer may be ”Warning: If you
do not provide a location, emergency services may be delayed in reaching your location should you need to call for
help.” This parameter is set by in-band provisioning when the handset is registered to Microsoft Skype for Business
or Microsoft Lync Server 2013 or 2010.
reg.x.lync.autoProvisionCertLocation
0 to 6
6
If 0, the certificate download is disabled. If non-0, the certificate corresponding to the index of the appropriate
sec.TLS.customCaCert.X is downloaded.
reg.x.musicOnHold.uri
a SIP URI
Null
A URI that provides the media stream to play for the remote party on hold. If present and not Null, this parameter
overrules voIpProt.SIP.musicOnHold.uri.
reg.x.outboundProxy.address
dotted-decimal IP
address or
hostname
Null
The IP address or hostname of the SIP server to which the handset sends all requests.
reg.x.outboundProxy.port
0, 1 to 65535
0
The port of the SIP server to which the handset sends all requests.
reg.x.outboundProxy.transport
DNSnaptr,
TCPpreferred,
UDPOnly, TLS,
TCPOnly
DNSnaptr
The transport method the handset uses to communicate with the SIP server.
Null or DNSnaptr – if reg.x.outboundProxy.address is a hostname and reg.x.outboundProxy.port is 0
or Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If
reg.x.outboundProxy.address is an IP address, or a port is given, then UDP is used.
TCPpreferred – TCP is the preferred transport, UDP is used if TCP fails.
UDPOnly – only UDP will be used.
TLS – if TLS fails, transport fails. Leave port field empty (will default to 5061) or set to 5061.
TCPOnly – only TCP will be used.
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Parameter
Permitted Values
Default
reg.x.proxyRequire
string
Null
The string that needs to be entered in the Proxy-Require header. If Null, no Proxy-Require will be sent.
reg.x.ringType.privateLine
default, ringer1 to
ringer24
default
The ringer to be used for calls received by a private line connected to Microsoft Skype for Business or Microsoft
Lync Server 2013 or 2010.
reg.x.serverAutoDiscovery
0 or 1
1
Determines whether or not to discover the server address automatically. This parameter is used with Microsoft
Skype for Business or Microsoft Lync Server 2013 or 2010.
reg.x.serverFeatureControl.cf1
0 or 1
0
If 0, server-based call forwarding is not enabled. If 1, server based call forwarding is enabled. This parameter
overrules voIpProt.SIP.serverFeatureControl.cf.
reg.x.serverFeatureControl.dnd1
0 or 1
0
If 0, server-based do-not-disturb (DND) is not enabled. If 1, server-based DND is enabled and the call server has
control of DND. This parameter overrules voIpProt.SIP.serverFeatureControl.dnd.
reg.x.serverFeatureControl.localProcessing.cf
0 or 1
1
If 0 and reg.x.serverFeatureControl.cf is set to 1, the handset will not perform local Call Forward behavior.
If set to 1, the handset will perform local Call Forward behavior on all calls received. This parameter overrules
voIpProt.SIP.serverFeatureControl.localProcessing.cf.
reg.x.serverFeatureControl.localProcessing.dnd
0 or 1
1
If 0 and reg.x.serverFeatureControl.dnd is set to 1, the handset will not perform local DND call behavior. If
set to 1, the handset will perform local DND call behavior on all calls received. This parameter overrules
voIpProt.SIP.serverFeatureControl.localProcessing.dnd.
reg.x.serverFeatureControl.signalingMethod
subscribeAsFeatureEvent,
inviteFACSubscribePresence,
serviceMsForwardContact
serviceMsForwardContact
Controls the method used to perform call forwarding requests to the server.
reg.x.server.y.registerRetry.maxTimeout
180 seconds
Set the maximum period of time in seconds that you want the handset to try registering with the server.
reg.x.srtp.enable1
0 or 1
1
If 0, the registration always declines SRTP offers. If 1, the registration accepts SRTP offers.
reg.x.srtp.offer1
0 or 1
0
If 1, the registration includes a secure media stream description along with the usual non-secure media description
in the SDP of a SIP INVITE. This parameter applies to the registration initiating (offering) a phone call. If 0, no
secure media stream is included in SDP of a SIP invite.
reg.x.srtp.require1
0 or 1
0
If 0, secure media streams are not required. If 1, the registration is only allowed to use secure media streams. Any
offered SIP INVITEs must include a secure media description in the SDP or the call will be rejected. For outgoing
calls, only a secure media stream description is included in the SDP of the SIP INVITE, meaning that the nonsecure media description is not included. If this parameter set to 1, reg.x.srtp.offer will also be set to 1,
regardless of the value in the configuration file.
reg.x.srtp.simplifiedBestEffort
0 or 1
0
If 0, no SRTP is supported. If 1, negotiation of SRTP compliant with Microsoft Session Description Protocol Version
2.0 Extensions is supported. This parameter overrules sec.srtp.simplifiedBestEffort.
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Parameter
Permitted Values
Default
reg.x.strictLineSeize
0 or 1
0
If 1, the handset is forced to wait for 200 OK on registration x when receiving a TRYING notify. This parameter
overrules voIpProt.SIP.strictLineSeize for registration x.
reg.x.telephony
0 or 1
1
If 0, telephony calls are not enabled on this registration (use this value if the registration is used for IM with Microsoft
Office Communications Server 2007 R2 or Microsoft Skype for Business or Microsoft Lync Server 2013 or 2010.). If
1, telephony calls are enabled on this registration.
reg.x.thirdPartyName
string address
Null
This field must match the reg.x.address value of the registration which makes up the part of a bridged line
appearance (BLA). It must be Null in all other cases.
reg.x.type
private or shared
private
If set to private, use standard call signaling. If set to shared, augment call signaling with call state subscriptions and
notifications and use access control for outgoing calls.
1
Change causes handset to restart or reboot.
Multiple Registrations
You can list multiple registration servers for fault tolerance. In the following table, you can list 4
servers by using y=1 to 4. If the reg.x.server.y.address is not null, all of the parameters
in the following table will overrule the parameters specified in voIpProt.server.*. The
server registration parameters are listed in the following table:
Table 8-50: Registration Server Parameters
Parameter
Permitted Values
Default
reg.x.server.y.address
dotted-decimal IP
address or
hostname
Null
The IP address or host name of a SIP server that accepts registrations. If not Null, all of the parameters in this
table will overrule the parameters specified in voIpProt.server.*. Notes: If this parameter is set, it will overrule
even if the DHCP server is available. If this registration is used for Microsoft Office Communications Server 2007
R2 on Spectralink handsets, this parameter must be in the form OCShostname.OSCdomain_name.
reg.x.server.y.expires
positive integer,
minimum 10
3600
The handset’s requested registration period in seconds. Note: The period negotiated with the server may be
different. The handset will attempt to re-register at the beginning of the overlap period. For example, if
expires=”300” and overlap=”5”, the handset will re-register after 295 seconds (300–5).
reg.x.server.y.expires.lineSeize
0 to 65535
30
5 to 65535
60
Requested line-seize subscription period.
reg.x.server.y.expires.overlap
The number of seconds before the expiration time returned by server x at which the handset should try to reregister. The handset will try to re-register at half the expiration time returned by the server if the server value is
less than the configured overlap value.
reg.x.server.y.lcs
0 or 1
0
If 0, the Microsoft Live Communications Server (LSC) is not supported. If 1, LCS is supported for registration x.
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Parameter
Permitted Values
Default
reg.x.server.y.useOutboundProxy
0 or 1
1
Specify whether or not to use the outbound proxy specified in reg.x.outboundProxy.address for server x.
This parameter overrules voIpProt.server.x.useOutboundProxy for registration x.
reg.x.server.y.port
0, 1 to 65535
0
The port of the sip server that specifies registrations. If 0, the port used depends on
reg.x.server.y.transport.
reg.x.server.y.register
0 or 1
1
10 - 120
60
If 0, calls can be routed to an outbound proxy without registration.
reg.x.server.y.registerRetry.baseTimeOut
The base time period to wait before a registration retry. Used in conjunction with
reg.x.server.y.registerRetry.maxTimeOut to determine how long to wait. The algorithm is defined in
RFC 5626.
reg.x.server.y.registerRetry.maxTimeOut
60 - 1800
60
The maximum time period to wait before a registration retry. Used in conjunction with
reg.x.server.y.registerRetry.baseTimeOut to determine how long to wait. The algorithm is defined in
RFC 5626.
reg.x.server.y.retryMaxCount
0 to 20
3
If set to 0, 3 is used. The number of retries that will be attempted before moving to the next available server.
reg.x.server.y.retryTimeOut
0 to 65535
0
The amount of time (in milliseconds) to wait between retries. If 0, use standard RFC 3261 signaling retry behavior.
reg.x.server.y.specialInterop
standard,
ocs2007r2,
lcs2005,
lync2010,
lync2013
Skype for
Business
standard
Specify if this registration should support Microsoft Office Communications Server 2007 R2 (ocs2007r2), Microsoft
Live Communications Server 2005 (lcs2005), or Microsoft Skype for Business or Microsoft Lync Server 2013 or
2010. (. Note: To use instant messaging on Spectralink handsets, set this parameter to ocs2007r2.
reg.x.server.y.transport
DNSnaptr,
TCPpreferred,
UDPOnly, TLS,
TCPOnly
DNSnaptr
The transport method the handset uses to communicate with the SIP server.
Null or DNSnaptr – if reg.x.server.y.address is a hostname and reg.x.server.y.port is 0 or Null,
do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If
reg.x.server.y.address is an IP address, or a port is given, then UDP is used.
TCPpreferred – TCP is the preferred transport; UDP is used if TCP fails.
UDPOnly – only UDP will be used.
TLS – if TLS fails, transport fails. Leave port field empty (will default to 5061) or set to 5061.
TCPOnly – only TCP will be used.
Multiple Concurrent Calls
You can enable each registered phone line to support multiple concurrent calls and have each
concurrent call display on the handset’s user interface. For example, you can place one call on
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hold, switch to another call on the same registered line, and have both calls display. As shown
in the next tables, you can set the maximum number of concurrent calls per registered line and
the default number of calls per line key.
Summary
Parameter
Used to:
call.callsPerLineKey
Set the default number of concurrent calls for all line keys
reg.x.callsPerLineKey
Overrule the default number of calls per line key for a specific line
Table 8-51: Enabling Multiple Call Appearances
Parameter
Permitted Values
Default
call.callsPerLineKey
1-24
24
Set the maximum number of concurrent calls per line key. This parameter applies to all registered lines.
Note that this parameter may be overruled by the per-registration parameter of reg.x.callsPerLineKey.
reg.x.callsPerLineKey1
1-24
24
Set the maximum number of concurrent calls for a single registration x. This parameter applies to all line keys using
registration x. If registration x is a shared line, an active call counts as a call appearance on all handsets sharing
that registration. This parameter overrules call.callsPerLineKey.
1
Change causes handset to restart or reboot.
Example Multiple Call Appearances Configuration
The following illustration shows parameters in the everything.cfg file. It shows how you can
enable line 1 on your handset with three call appearances.
Once you have set the reg.1.callsPerLineKey parameter to three, you can have three call
appearances on line 1. By default, additional incoming calls will be automatically forwarded to
your voicemail. If you have more than two call appearances, a call appearance counter will
display at the top right corner of your handset’s screen as shown next.
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Flexible Call Appearances
A number of features are associated with Flexible Call Appearances. Use the following
information to understand how you can organize registrations, line keys per registration, and
concurrent calls per line key.
In the following table:
Registrations The maximum number of user registrations
Line Keys The maximum number of line keys
Calls Per Line Key The maximum number of concurrent calls per line key
Concurrent Calls (includes Conference Legs) The runtime maximum number of concurrent
calls. (The number of conference participants minus the moderator.)
Phone Model
Registrations
Line Keys
Calls Per Line Key
Concurrent Calls*
Spectralink 84xx
6
6
24
24 (2)
* Note that each conference leg counts as one call. The total number of concurrent calls in a
conference indicated in this table includes all conference participants minus the moderator.
User Profiles
User Profiles are designed to be used when handsets are shared by several users such as in
shift situations. Users log on to any handset with a username and password and the handset
uses their personal settings. The default password is 123.
If a user changes any settings while logged in to a handset, the settings will be saved in an
override file and loaded onto the phone the next time the same user logs in to a handset.When
a user logs out, the user’s personal handset settings are no longer loaded on the phone.
If the User Profile feature is set up on your company’s handsets, users can:
 Log in to a handset to access their personal handset settings.
 Log out of a handset after they finish using it. (with or without a password)
 Place a call to an authorized number from a handset that is in the logged out state.
 Change their user password.
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When you set up the User Profile feature, you will have to decide whether you want to require
users to always log in to a handset. If the User Profile feature is enabled, but not required, users
can choose to use the handset as is (that is, without access to their personal settings), or they
can log in to have access to their personal settings. You can specify if a user is logged out of
the handset when the handset restarts or reboots, or if they remain logged in.
You can also choose to define default credentials for the handset. If you specify a default user
ID and password, the handset automatically logs itself in each time an actual user logs out or
the handset restarts or reboots. When the handset logs itself in using the default login
credentials, a default handset profile is displayed (as defined in the handset’s master
configuration file on the provisioning server). In this scenario, users will still have the option to
log in and view their personal settings.
Resetting a User’s Password
Spectralink recommends that you create a single default user password for all
users. You can restore this default password by removing the password parameter
from the override file. This will cause the handset to use the default password in the
<user>.cfg file.
Summary
Parameter
Used to:
prov.login.enabled
Enable or disable the user profile feature
prov.login.automaticLogout
Specify the amount of time before a non-default user is logged out
prov.login.defaultPassword
Specify the default password for the default user
prov.login.defaultOnly
Specify if the handset can have users other than the default user
prov.login.defaultUser
Specify the name of the default user
prov.login.localPassword
Specify the password used to validate the user login
prov.login.persistent
Specify if a user should remain logged in after the handset reboots
prov.login.required
Specify if a user must log in while the feature is enabled
settingsLock.userProfileLogoutPassword
Specify a logout password
Table 8-52: User Profile Settings
Parameter
Permitted Values
Default
prov.login.enabled
0 or 1
0
If 0, the user profile feature is disabled. If 1, the user profile feature is enabled.
prov.login.defaultPassword
String
Null
0 or 1
0
The login password for the default user.
prov.login.defaultOnly
If 1, the default user is the only user who can log in. If 0, other users can log in.
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Parameter
Permitted Values
Default
prov.login.defaultUser
String
Null
The username for the default user. If present, the user is automatically logged in when the handset boots up and
logged in after another user logs out.
prov.login.localPassword
String
123
The password used to validate the user login. It is stored either as plain text or encrypted (an SHA1 hash).
prov.login.persistent
0 or 1
0
If 0, users are logged out if the handset reboots. If 1, users remain logged in when the handset reboots.
Set this parameter to “1” if PTT parameters are specified in user-profile-specific cfg files. This way, if the phone
reboots, the PTT settings persist. PTT settings (and other settings) will persist until the user logs out.
prov.login.required
0 or 1
0
If 1, a user must log in when the login feature is enabled. If 0, the user does not have to log in.
prov.loginCredPwdFlushed.enabled
0 or 1
1
If 1, when a user logs in or logs out, the login credential password is cleared. If 0, the login credential password is
not cleared.
prov.login.automaticLogout
0 to 46000
0
The time (in minutes) before a non-default user is automatically logged out of the handset. If 0, the user is not
automatically logged out.
settingsLock.userProfileLogoutPassword
String [1-32 characters]
0
Logout password. When set, and the phone is configured for user profiles, the phone will:
1. require a password to access the following menu: Settings> Feature Settings> User Login> Logout
This means when the user selects "Logout" they will be prompted for the password set in the above config item.
2. the Logout item on the Feature softkey will be greyed out - this will be unusable, it will not prompt for a password.
Placing Authorized (Emergency) Calls without Logging In
You may wish to allow handsets that normally require a user to provide log in credentials to be
used for emergency dialing purposes without requiring the caller to log in. You can provision a
list of authorized calls that can be made without logging in. The required parameters are
detailed in the Local Digit Map section in the chapter on Special Use Cases and illustrated
below. Also see Emergency Dial via Authorized Call menu for additional information about the
Authorized Call menu.
1
Enable the prov.login parameter prov.login.required. When enabled, the handset will
continuously display the User Login screen as shown below.
2
Configure the requisite parameters for emergency dial routing.
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Parameters for enabling the emergency dialing feature should be implemented in a
global/site configuration file so that all handsets have the emergency dialing feature
enabled.
Caution: Emergency server must allow anonymous inbound SIP calls
A logged-out handset is not registered to the SIP call server. Therefore, the
emergency SIP call server must be configured to allow anonymous inbound SIP
calls. Otherwise the emergency calls will not be recognized.
User experience
The handset will display the login screen. When the off-hook START key is pressed, the user
will be presented with the Place an Authorized Call screen enabling the user to place
emergency calls.
Voicemail Integration
The Spectralink 84-Series handset is compatible with voicemail servers. You can configure
each handset or line registration per handset to subscribe with a SIP URL to a voicemail server
contact. You can also configure the handset to access voicemail with a single key, for example,
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the Messages icon on the handset’s Home screen. When you access the voicemail server, the
handset gives a visual and audio alert; you can also configure a message waiting alert to
indicate that you have unread voicemail messages.
Table 8-53: Voicemail Integration
Parameter
Permitted Values
Default
up.oneTouchVoiceMail1
0 or 1
0
If set to 1, the voicemail summary display is bypassed and voicemail is dialed directly (if configured).
msg.mwi.x.subscribe
ASCII encoded string containing
digits (the user part of a SIP URL)
or a string that constitutes a valid
SIP URL (6416 or
6416@spectralink.com)
Null
If non-Null, the handset will send a SUBSCRIBE request to this contact after boot-up.
msg.mwi.x.callBackMode
contact, registration, disabled
registration
The message retrieval mode and notification for registration x.
contact – a call is placed to the contact specified by msg.mwi.x.callback.
registration – the registration places a call to itself (the handset calls itself).
disabled – message retrieval and message notification are disabled.
msg.mwi.x.callBack
ASCII encoded string containing
digits (the user part of a SIP URL)
or a string that constitutes a valid
SIP URL (6416 or
6416@spectralink.com)
Null
The contact to call when retrieving messages for this registration if msg.mwi.x.callBackMode is set to
contact. For R4.14 and higher the maximum length is 126 characters. Earlier versions are limited to 64
characters.
up.mwiVisible1
0 or 1
0
If set is 0, the incoming MWI notifications for lines where the MWI callback mode is disabled
(msg.mwi.x.callBackMode is set to 0) are ignored, and do not appear in the message retrieval menus.
If set to 1, the MWI for lines whose MWI is disabled will display (pre-SIP 2.1 behavior), even though MWI
notifications have been received for those lines.
1
Change causes handset to restart or reboot.
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Example Voicemail Configuration
The following illustration shows you how to enable one-touch access to the voicemail server. In
the next illustration, line 2 is configured to subscribe to the voicemail server at
voicemail.Spectralink.com.
<voIpProt/>
VoIP Protocol settings are used in wireless environments but may be overruled by <reg>
parameters. You can use VoIP Protocol parameters to set up the call server and DTMF
signaling.
Settings: many voIpProt settings are overruled by counterpart reg.x settings
Many voIpProt parameters have been replaced by reg.x parameters which will
overrule their counterpart voIpProt parameters if both are set. We recommend
using reg.x parameters whenever possible. Some but not all voIpProt parameters
have replacement reg.x parameters noted in their parameter descriptions.
Table 8-54: VoIP Server Parameters
Parameter
Permitted Values
Default
voIpProt.server.dhcp.available1
0 or 1
0
If 0, do not check with the DHCP server for the SIP server IP address. If 1, check with the server for the IP
address.
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Parameter
Permitted Values
Default
voIpProt.server.dhcp.option1
128 to 254
128
The option to request from the DHCP server if voIpProt.server.dhcp.available= 1.
Note: If reg.x.server.y.address is non-Null, it overrules even if the DHCP server is available.
voIpProt.server.dhcp.type1
0 or 1
0
Type to request from the DHCP server if voIpProt.server.dhcp.available is set to 1.If this parameter is
set to 0, IP request address. If set to 1, request string.
voIpProt.server.x.address
dotted- decimal IP
address or
hostname
Null
The IP address or hostname of a SIP server that accepts registrations. Multiple servers can be listed starting with
x=1 to 4 for fault tolerance. Overruled by reg.x.server.y.address.
voIpProt.server.x.port
0, 1 to 65535
0
The port of the server that accepts registrations. If 0, the port used depends on
voIpProt.server.x.transport. Overruled by reg.x.server.y.port.
voIpProt.server.x.registerRetry.baseTimeOut
10 - 120
60
The base time period to wait before a registration retry. Used in conjunction with
voIpProt.server.x.registerRetry.maxTimeOut to determine how long to wait. The algorithm is defined
in RFC 5626.
If both parameters voIpProt.server.x.registerRetry.baseTimeOut and
reg.x.server.y.registerRetry.baseTimeOut are set, the value of
reg.x.server.y.registerRetry.baseTimeOut overrules the similar voIpProt parameter.
voIpProt.server.x.registerRetry.maxTimeOut
60 - 1800
60
The maximum time period to wait before a registration retry. Used in conjunction with
voIpProt.server.x.registerRetry.maxTimeOut to determine how long to wait. The algorithm is defined
in RFC 5626.
If both parameters voIpProt.server.x.registerRetry.maxTimeOut and
reg.x.server.y.registerRetry.maxTimeOut are set, the value of
reg.x.server.y.registerRetry.maxTimeOut overrules the similar voIpProt parameter.
voIpProt.server.x.transport
DNSnaptr,
TCPpreferred,
UDPOnly, TLS,
TCPOnly
DNSnaptr
The transport method the handset uses to communicate with the SIP server. Overruled by
reg.x.server.y.transport values.
Null or DNSnaptr – if voIpProt.server.x.address is a hostname and voIpProt.server.x.port is 0 or
Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If
voIpProt.server.x.address is an IP address, or a port is given, then UDP is used.
TCPpreferred – TCP is the preferred transport; UDP is used if TCP fails.
UDPOnly: only UDP will be used.
TLS – if TLS fails, transport fails. Leave port field empty (will default to 5061) or set to 5061.
TCPOnly – only TCP will be used.
voIpProt.server.x.protocol.SIP
0 or 1
1
If 1, server is a SIP proxy/registrar. Note: if set to 0, and the server is confirmed to be a SIP server, then the value
is assumed to be 1.
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Parameter
Permitted Values
Default
voIpProt.server.x.expires
positive integer,
minimum 10
3600
The handset’s requested registration period in seconds. Note: The period negotiated with the server may be
different. The handset will attempt to re-register at the beginning of the overlap period. For example, if
expires=”300” and overlap=”5”, the handset will re-register after 295 seconds (300–5). Overruled by
reg.x.server.y.expires values.
voIpProt.server.x.expires.overlap
5 to 65535
60
The number of seconds before the expiration time returned by server x at which the handset should try to reregister. The handset will try to re-register at half the expiration time returned by the server if the server value is
less than the configured overlap value.
voIpProt.server.x.expires.lineSeize
0 to 65535
30
0 or 1
0
Requested line-seize subscription period.
voIpProt.server.x.lcs
If 0, the Microsoft Live Communications Server (LSC) is not supported. If 1, LCS is supported for registration x.
This parameter overrules voIpProt.SIP.lcs .
voIpProt.server.x.register
0 or 1
1
If 0, calls can be routed to an outbound proxy without registration. See reg.x.server.y.register.
voIpProt.server.x.retryTimeOut
0 to 65535
0
The amount of time (in milliseconds) to wait between retries. If 0, use standard RFC 3261 signaling retry behavior.
voIpProt.server.x.retryMaxCount
0 to 20
3
If set to 0, 3 is used. The number of retries that will be attempted before moving to the next available server.
voIpProt.server.x.specialInterop
standard,
ocs2007r2, lcs2005,
lync2010, lync2013
Skype for Business
standard
Specify if this registration should support Microsoft Office Communications Server 2007 R2 (ocs2007r2), Microsoft
Live Communications Server 2005 (lcs2005), or Microsoft Skype for Business or Microsoft Lync Server 2013 or
2010.).
voIpProt.server.x.useOutboundProxy
0 or 1
1
Specify whether or not to use the outbound proxy specified in voIpProt.SIP.outboundProxy.address for
server x.
1
Change causes handset to restart or reboot.
Table 8-55: Session Description Protocol (SDP) Parameters
Parameter
Permitted Values
Default
voIpProt.SDP.answer.useLocalPreferences
0 or 1
0
If set to 1, the handset uses its own preference list when deciding which codec to use rather than the preference
list in an offer. If set to 0, it is disabled.
voIpProt.SDP.early.answerOrOffer
0 or 1
0
If set to 1, an SDP offer or answer is generated in a provisional reliable response and PRACK request and
response. If set to 0, an SDP offer or answer is not generated.
Note: An SDP offer or answer is not generated if reg.x.musicOnHold.uri is set.
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Parameter
Permitted Values
Default
voIpProt.SDP.useLegacyPayloadTypeNegotiation
0 or 1
0
If set to 1, the handset transmits and receives RTP using the payload type identified by the first codec listed in the
SDP of the codec negotiation answer.
If set to 0, RFC 3264 is followed for transmit and receive RTP payload type values.
Table 8-56: Session Initiation Protocol (SIP) Parameters
Parameter
Permitted Values
Default
voIpProt.SIP.acd.signalingMethod1
0 or 1
0
If set to 0, the ‘SIP-B’ signaling is supported. (This is the older ACD functionality.)
If set to 1, the feature synchronization signaling is supported. (This is the new ACD functionality.)
voIpProt.SIP.CID.sourcePreference
ASCII string up to
120 characters
long
Null
Specify the priority order for the sources of caller ID information. The headers can be in any order.
If Null, caller ID information comes from P-Asserted-Identity, Remote-Party-ID, and From in that order.
The values From,P-Asserted-Identity, Remote-Party-ID and P-Asserted-Identity, RemoteParty-ID are also valid.
voIpProt.SIP.compliance.RFC3261.validate.contentLanguage
0 or 1
1
If set to 1, validation of the SIP header content language is enabled. If set to 0, validation is disabled.
voIpProt.SIP.compliance.RFC3261.validate.contentLength
0 or 1
1
If set to 1, validation of the SIP header content length is enabled. If set to 0, validation is disabled.
voIpProt.SIP.compliance.RFC3261.validate.uriScheme
0 or 1
1
If set to 1, validation of the SIP header URI scheme is enabled. If set to 0, validation is disabled.
voIpProt.SIP.conference.address
ASCII string up to
128 characters
long
Null
If Null, conferences are set up on the handset locally.
If set to some value, conferences are set up by the server using the conferencing agent specified by this address.
Acceptable values depend on the conferencing server implementation policy.
voIpProt.SIP.conference.parallelRefer
0 or 1
0
If 1, a parallel REFER is sent to the call server. Note: This parameter must be set for Siemens Openscape
Centralized Conferencing.
voIpProt.SIP.connectionReuse.useAlias
0 or 1
0
If set to 0, this is the old behavior.
If set to 1, handset uses the connection reuse draft which introduces "alias".
voIpProt.SIP.csta
0 or 1
0
If 0, the uaCSTA (User Agent Computer Supported Telecommunications Applications) feature is disabled. If 1,
uaCSTA is enabled (If reg.x.csta is set, it will overrule this parameter).
voIpProt.SIP.dialog.strictXLineID
0 or 1
0
If 0, the handset will not look for x-line-id (call appearance indec) in a SIP INVITE message, if one is not present.
Instead, when it receives INVITE, the handset will generate the call appearance locally and pass that information to
other parties involved in the call.
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Parameter
Permitted Values
Default
voIpProt.SIP.dialog.usePvalue
0 or 1
0
If set to 0, handset uses a pval field name in the Dialog. This obeys the draft-ietf-sipping-dialog-package-06.txt
draft.
If set to 1, the handset uses a field name of pvalue.
voIpProt.SIP.dialog.useSDP
0 or 1
0
If set to 0, a new dialog event package draft is used (no SDP in dialog body).
If set to 1, for backwards compatibility, use this setting to send SDP in the dialog body.
voIpProt.SIP.dtmfViaSignaling.rfc29761
0 or 1
0
If set to 1, DTMF digit information is sent in RFC2976 SIP INFO packets during a call.
If set to 0, no DTMF digit information is sent.
voIpProt.SIP.enable1
0 or 1
1
A flag to determine if the SIP protocol is used for call routing, dial plan, DTMF, and URL dialing.
If set to 1, the SIP protocol is used.
voIpProt.SIP.IM.autoAnswerDelay
0 to 40
10
The time interval in seconds from receipt of the instant message invitation to automatically accepting the invitation.
voIpProt.SIP.keepalive.sessionTimers
0 or 1
0
If set to 1, the session timer will be enabled. If set to 0, the session timer will be disabled, and the handset will not
declare “timer” in “Support” header in an INVITE. The handset will still respond to a re-INVITE or UPDATE. The
handset will not try to re-INVITE or UPDATE even if the remote end point asks for it.
voIpProt.SIP.lineSeize.retries
3 to 10
10
Controls the number of times the handset will retry a notify when attempting to seize a line (BLA).
voIpProt.SIP.local.port1
0 to 65535
5060
The local port for sending and receiving SIP signaling packets.
If set to 0, 5060 is used for the local port but is not advertised in the SIP signaling.
If set to some other value, that value is used for the local port and it is advertised in the SIP signaling.
voIpProt.SIP.mtls.enable
0 or 1
1
If 0, TLS with mutual authentication is disabled. If 1, TLS with mutual authentication is enabled. Used in conjunction
with Microsoft Skype for Business or Microsoft Lync Server 2013 or 2010..
voIpProt.SIP.outboundProxy.address
dotted-decimal IP
address or
hostname
Null
The IP address or hostname of the SIP server to which the handset sends all requests.
voIpProt.SIP.outboundProxy.port
0 to 65535
0
DNSnaptr,
TCPpreferred,
UDPOnly, TLS,
TCPOnly
DNSnaptr
The port of the SIP server to which the handset sends all requests.
voIpProt.SIP.outboundProxy.transport
The transport method the handset uses to communicate with the SIP server.
Null or DNSnaptr – if reg.x.outboundProxy.address is a hostname and reg.x.outboundProxy.port is
0 or Null, do NAPTR then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If
reg.x.outboundProxy.address is an IP address, or a port is given, then UDP is used.
TCPpreferred – TCP is the preferred transport, UDP is used if TCP fails.
UDPOnly – only UDP will be used.
TLS – if TLS fails, transport fails. Leave port field empty (will default to 5061) or set to 5061.
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Parameter
Permitted Values
Default
0 to 3600
0
TCPOnly – only TCP will be used.
voIpProt.SIP.pingInterval
The number in seconds to send "PING" message. This feature is disabled by default.
voIpProt.SIP.pingMethod
PING, OPTIONS
PING
0 or 1
0
The ping method to be used.
voIpProt.SIP.presence.nortelShortMode1
Different headers sent in SUBSCRIBE when used for presence on an Avaya (Nortel) server. Support is indicated by
adding a header Accept-Encoding: x-nortel-short. A PUBLISH is sent to indicate the status of the
handset.
voIpProt.SIP.requestValidation.digest.realm1
wirelessIpPhone
wirelessIpPhone
0 or 1
0
0 or 1
0
Determines the string used for Realm.
voIpProt.SIP.requestURI.E164.addGlobalPrefix
If set to 1, ‘+’ global prefix is added to the E.164 user parts in sip: URIs.
voIpProt.SIP.sendCompactHdrs
If set to 0, SIP header names generated by the handset use the long form, for example From.
If set to 1, SIP header names generated by the handset use the short form, for example f.
voIpProt.SIP.serverFeatureControl.dnd
0 or 1
0
If set to 1, server-based DND is enabled. The call server has control of DND.
If set to 0, server-based DND is not enabled.
voIpProt.SIP.serverFeatureControl.missedCalls1
0 or 1
0
If set to 1, server-based missed calls is enabled. The call server has control of missed calls.
If set to 0, server-based missed calls is not enabled. This is the old behavior.
voIpProt.SIP.serverFeatureControl.localProcessing.dnd
0 or 1
1
If set to 0 and voIpProt.SIP.serverFeatureControl.dnd is set to 1, the handset will not perform local DND
call behavior.
If set to 1, the handset will perform local DND call behavior on all calls received.
voIpProt.SIP.specialEvent.checkSync.alwaysReboot1
0 or 1
0
If set to 1, always reboot when a NOTIFY message is received from the server with event equal to check-sync.
If set to 0, only reboot if any of the files listed in <MAC-address>.cfg have changed on the FTP server when a
NOTIFY message is received from the server with event equal to check-sync.
voIpProt.SIP.specialEvent.lineSeize.nonStandard1
0 or 1
1
If set to 1, process a 200 OK response for a line-seize event SUBSCRIBE as though a line-seize NOTIFY with
Subscription State: active header had been received,. This speeds up processing.
voIpProt.SIP.strictLineSeize
0 or 1
0
If set to 1, The handset is forced to wait for a 200 OK response when receiving a TRYING notify.
If set to 0, this is old behavior.
voIpProt.SIP.strictUserValidation
0 or 1
0
If set to 1, the handset is forced to match the user portion of signaling exactly.
If set to 0, the handset will use the first registration if the user part does not match any registration.
voIpProt.SIP.tlsDsk.enable
0 or 1
0
If 0, TLS DSK is disabled. If 1, TLS DSK is enabled.
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Parameter
Permitted Values
Default
voIpProt.SIP.turnOffNonSecureTransport1
0 or 1
0
0 or 1
0
If set to 1, stop listening to port 5060 when using AS-SIP enabled.
voIpProt.SIP.use486forReject
If set to 1 and the handset is indicating a ringing inbound call appearance, the handset will transmit a 486 response
to the received INVITE when the Reject softkey is pressed.
If set to 0, no 486 response is transmitted.
voipPort.SIP.useCompleteUriForRetrieve
0 or 1
1
If set to 1, the target URI in BLF signaling will use the complete address as provided in the xml dialog document.
If set to 0, only the user portion of the XML dialog document is used and the current registrar's domain is appended
to create the full target URI.
voIpProt.SIP.useContactInReferTo
0 or 1
0
If set to 0, the “To URI” is used in the REFER.
If set to 1, the “Contact URI” is used in the REFER.
1
Change causes handset to restart or reboot.
Web Browser
There are two aspects to the web browser. One is the list of apps that are accessible from the
Applications icon. The other is the development of these apps and the parameters that are used
to implement them.
The Spectralink handsets support a full Web browser that the user can launch by navigating to
Home> Applications. The label and url for the web application is set in the parameters listed
below. Other <apps/> parameters are used to control telephone notification events, state polling
events, and push server controls.
For more information on how to use the control parameters, see the Spectralink Web
Application Developer’s Guide.
Table 8-57: Apps listed on the handset
Parameter
Permitted Values
Default
apps.x.label2
The descriptive text that displays in the Applications menu
apps.x.url2
The URL of an application
String
null
The label and URL of up to 12 applications (for x = 1 to 12).
2
For the Spectralink 84-Series handsets, the toolbar autohide is disabled by default.
Example Web Browser Configuration
The following example shows you how to set the interactive Web browser’s home page on the
Spectralink handsets.
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The following illustration shows the Web browser’s interactive home page on the Spectralink
handset.
Table 8-58: SDK Application Parameters
Parameter
Permitted Values
Default
apps.telNotification.heartbeatTimeoutSeconds
0 – 65,535
0
Implements a "heartbeat" timer for telephony notifications that repeats the line registration notification on a periodic
basis if set. The default value for this parameter is 0, where the timer does not start at all, so there is no change to
the behavior unless the apps.telNotification.heartbeatTimeoutSeconds parameter is set to a non-zero value.
With a heartbeat set, when the phone goes out of range or is power cycled, notifications are not disrupted.
apps.push.messageType
0 to 5
0
Choose a priority level for push messages from the application server to the handset.
0: (None) Discard push messages
1: (Normal) Allows only normal push messages
2: (Important) Allows only important push messages
3: (High) Allows only priority push messages
4: (Critical) Allows only critical push
5: (All) Allows all push messages
apps.push.password
string
null
The password to access the push server URL.
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Parameter
Permitted Values
Default
apps.push.secureTunnelEnabled
0 or 1
1
If 0, the Web server is not connected through a secure tunnel. If 1, the Web server is connected through a secure
tunnel. To disable the non-secure push (HTTP) requires that both apps.push.secureTunnelEnabled and
apps.push.secureTunnelRequired be set to 0.
apps.push.secureTunnelPort
1 to 65535
443
The port that the handset should use to communicate to the Web server when the secure tunnel is used.
apps.push.secureTunnelRequired
0 or 1
0
If 0, communications to the Web server do not require a secure tunnel. If 1, communications require a secure
tunnel. To disable the non-secure push (HTTP) requires that both apps.push.secureTunnelEnabled and
apps.push.secureTunnelRequired be set to 0.
apps.push.serverRootURL
URL
null
The URL of the application server you enter here is combined with the handset address and sent to the handset’s
browser. For example, if the application server root URL is http://172.24.128.85:8080/sampleapps and the relative
URL is /examples/sample.html, the URL that is sent to the browser is
http://172.24.128.85:8080/sampleapps/examples/sample.html. Can be either HTTP or HTTPS.
apps.push.username
string
null
The user name to access the push server URL.
Note: To enable the push functionality, the parameters apps.push.username and apps.push.password must be set
(not null).
apps.statePolling.password
string
null
Enter the password that the handset requires to authenticate handset state polling.
apps.statePolling.URL
URL
null
The URL to which the handset sends call processing state/device/network information. The protocol used can be
either HTTP or HTTPS. Note: To enable state polling, the parameters apps.statePolling.URL,
apps.statePolling.username, and apps.statePolling.password must be set to non-null values.
apps.statePoling.responseMode
0 or 1
1
The mode of sending requested polled data. If 1, requested polled data is sent to a configured URL. If 0, the data
is sent in the HTTP response.
apps.statePolling.username
string
null
Enter the user name that the handset requires to authenticate handset state polling.
apps.telNotification.callStateChangeEvent
0 or 1
0
If 0, call state change notification is disabled. If 1, notification is enabled.
apps.telNotification.incomingEvent
0 or 1
0
If 0, incoming call notification is disabled. If 1, notification is enabled.
apps.telNotification.lineRegistrationEvent
0 or 1
0
If 0, line registration notification is disabled. If 1, notification is enabled.
apps.telNotification.networkUpEvent
0 or 1
0
If 0, network up notification is disabled. If 1, notification is enabled.
apps.telNotification.offhookEvent
0 or 1
0
If 0, off-hook notification is disabled. If 1, notification is enabled.
apps.telNotification.onhookEvent
0 or 1
0
If 0, on-hook notification is disabled. If 1, notification is enabled.
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Parameter
Permitted Values
Default
apps.telNotification.outgoingEvent
0 or 1
0
If 0, outgoing call notification is disabled. If 1, notification is enabled.
apps.telNotification.scanBarcodeEvent
0 or 1
0
If 0, event notification is disabled. If 1, notification is enabled.
apps.telNotification.uiInitializationEvent
0 or 1
0
If 0, user interface initialization notification is disabled. If 1, notification is enabled.
apps.telNotification.URL
URL
null
The URL to which the handset sends notifications of specified events. Can be either HTTP or HTTPS.
apps.telNotification.x.URL
URL
null
The URL to which the handset sends notifications of specified events, where x 1 to 9. Can be either HTTP or
HTTPS.
apps.telNotification.userLogInOutEvent
0 or 1
0
If 0, user login/logout notification is disabled. If 1, notification is enabled.
The UserLoginOut Event can be used to detect when a user signs into or out of a phone using a profile. A
notification will also be sent whenever the phone is power cycled, whether or not the same login is used (i.e.
persistent). This way you can verify who is using the phone through a power cycle as well as any login/logout
cycle.
<mb/>
This parameter’s settings control the home page, proxy and size limits to be used by the
browser when it is selected to provide services.
Table 8-59: Web Browser Parameters
Parameter
Permitted Values
Default
mb.main.autoBackKey1
0 or 1
1
If 0, the handset does not provide a Back softkey; all softkeys are created and controlled by the application. If 1,
the handset automatically supplies a Back softkey in all main browser screens. The Back softkey will take the
user back to the previous page in the browser history.
mb.main.home
Any fully formed valid HTTP URL.
Length up to 255 characters.
Null
The URL of the browser's Home page. For example: http://www.example.com/xhtml/frontpage/home. If blank, the
browser will notify the user that a blank home-page was used.
mb.main.idleTimeout
0 to 600
40
The timeout, in seconds, for the interactive browser. If the interactive browser remains idle for the defined period of
time, the handset returns to idle. If 0, there is no timeout.
mb.proxy
Null or domain name or IP address
in the format <address>:<port>
Null. Default port = 8080
The address of the HTTP proxy to be used by the browser. If blank, normal unproxied HTTP is used by the
browser.
mb.main.toolbar.autoHide.enabled
0 or 1
1
If 0, the toolbar displays continually. If 1, the toolbar disappears if not selected.
1
Change causes handset to restart or reboot.
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<oai/>
The Spectralink handsets support communications using the Open Application Interface (OAI).
Also see Open Application Interface.You can set the connection parameters using the table
shown next:
Table 8-60: Open Application Interface (OAI) Parameters
Parameter
Permitted values
Default
oai.gateway.address
IP address
Null
String of eight hexadecimal characters
Null
The address of the OAI server.
oai.userId
The lower four bytes of the six-byte OAI handset identifier in the OAI gateway.
If the value is null or invalid, the handset identifies itself to the OAI gateway using the MAC address of the handset;
otherwise, the upper two bytes are zero and the lower four bytes are as specified.
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Chapter 9: Web Application Parameters
Web application parameters allow the handset to work with an application, providing
information, alerts and notifications through configuration of parameters that are designed for
interfacing with third party applications. Consult with your application developer for parameter
requirements.
The parameters described in this chapter include those for:

Web applications

Push requests

Telephone event notifications

Phone state polling
A sample configuration is shown in Sample Configurations.
Application menu configuration <apps.>
Table 9-1: Apps listed on the phone
Parameter
Permitted Values
Default
apps.x.label2
The descriptive text that displays in the Applications menu
apps.x.url2
The URL of an application
String
null
The label and URL of up to 12 applications (for x = 1 to 12).
2
For the Spectralink 84-Series handsets, the toolbar autohide is disabled by default.
Example Web browser Configuration
The following example illustrates the use of these two parameters and how they look on the
Application menu.
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Web browser parameters <mb.>
The <mb.> parameters control the home page, proxy and size limits to be used by the browser
when it is selected to provide services.
Table 9-2: Web browser Parameters
Parameter
Permitted Values
Default
mb.main.autoBackKey1
0 or 1
1
If 0, the phone does not provide a Back soft key; all soft keys are created and controlled by the application. If 1,
the phone automatically supplies a Back soft key in all main browser screens. The Back soft key will take the user
back to the previous page in the browser history.
mb.main.home
Any fully formed valid HTTP URL.
Length up to 255 characters.
Null
The URL of the browser's Home page. For example: http://www.example.com/xhtml/frontpage/home. If no
apps.x.url applications are configured, and mb.main.home is configured, then when the user selects the
Applications icon from the carousel, the mb.main.hone will be immediately loaded.
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Parameter
Permitted Values
Default
mb.main.idleTimeout
0 - 600, seconds
40
The timeout, in seconds, for the interactive browser. If the interactive browser remains idle for the defined period of
time, the phone returns to the idle browser. If 0, there is no timeout.
mb.proxy
Null or domain name or IP address
in the format <address>:<port>
Null. Default port = 8080
The address of the HTTP proxy to be used by the browser. If blank, normal unproxied HTTP is used by the
browser.
1
Change causes phone to restart or reboot.
State Polling Parameters <apps.statePolling.>
The <apps.statePolling/> parameter is used to control state polling events.
Table 9-3: Telephone Event Notification Parameters
Parameter
Permitted Values
Default
apps.statePolling.password
string
null
Enter the password that the phone requires to authenticate phone state polling.
apps.statePolling.URL
URL
null
The URL to which the phone sends call processing state/device/network information. The protocol used can be
either HTTP or HTTPS. Note: To enable state polling, the parameters apps.statePolling.URL,
apps.statePolling.username, and apps.statePolling.password must be set to non-null values.
apps.statePolling.responseMode
0 or 1
1
The mode of sending requested polled data. If 1, requested polled data is sent to a configured URL. If 0, the data
is sent in the HTTP response.
apps.statePolling.username
string
null
Enter the user name that the phone requires to authenticate phone state polling.
Push Request Parameters <apps.push.>
The <apps.push/> parameters are used to control push server controls.
Settings: Enabling Data URL Push
Both apps.push.username and apps.push.password must be set for Data URL
Push to be enabled.
Note: Parameter available in Release 4.5, 4.7 and later
The expanded np.normal.alert.x parameters have been implemented in Release
4.5.x, 4.7.x and later.
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Table 9-4: Push Parameters
Parameter
Permitted Values
Default
np.normal.alert.apiPush.tonePattern
silent, messageWaiting,
instantMessage,
remoteHoldNotification,
localHoldNotification,
positiveConfirm,
negativeConfirm,
welcome, misc1, misc2,
misc3, misc4, misc5,
misc6, misc7, custom1,
custom2
silent
0 or 1
0
If silent, there is no sound when an alert is pushed.
np.normal.alert.apiPush.vibration
If 0, there is no sound when an alert is pushed. If 1, the handset’s selected ring profile’s configured API Push alert
tone is played.
np.silent.alert.apiPush.tonePattern
0 or 1
0
If 0, there is no sound when an alert is pushed. If 1, the handset’s selected ring profile’s configured API Push alert
tone is played.
np.silent.alert.apiPush.vibration
0 or 1
0
If 0, there is no sound when an alert is pushed. If 1, the handset’s selected ring profile’s configured API Push alert
tone is played.
np.meeting.alert.apiPush.tonePattern
0 or 1
0
If 0, there is no sound when an alert is pushed. If 1, the handset’s selected ring profile’s configured API Push alert
tone is played.
np.meeting.alert.apiPush.vibration
0 or 1
0
If 0, there is no sound when an alert is pushed. If 1, the handset’s selected ring profile’s configured API Push alert
tone is played.
np.custom1.alert.apiPush.tonePattern
0 or 1
0
If 0, there is no sound when an alert is pushed. If 1, the handset’s selected ring profile’s configured API Push alert
tone is played.
np.custom1.alert.apiPush.vibration
0 or 1
0
If 0, there is no sound when an alert is pushed. If 1, the handset’s selected ring profile’s configured API Push alert
tone is played.
apps.push.messageType
0 to 5
0
Choose a priority level for push messages from the application server to the phone.
0: (None) Discard push messages
1: (Normal) Allows only normal push messages
2: (Important) Allows only important push messages
3: (High) Allows only priority push messages
4: (Critical) Allows only critical push messages
5: (All) Allows all push messages
See Caution: Critical message conflict note after this table.
apps.push.password
string
null
The password to access the push server URL. Used with the username to respond to the HTTP Digest Challenge
from the handset.
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Parameter
Permitted Values
Default
apps.push.play.incall.volume.scaling
-36dB to 0dB
0dB
This parameter allows the administrator to attenuate the volume level of alert tones played in-ear
when a user is in the call. Some alert tones may be perceived by users as being loud and this
parameter allows an administrator to apply volume reduction to all tones played by the API when
in call.
If 0, the volume of the push alert is the same as the call volume. Settings of -10, -20 and -36 get progressively
quieter.
apps.push.secureTunnelEnabled
0 or 1
1
If 0, the Web server is not connected through a secure tunnel. If 1, the Web server is connected through a secure
tunnel.
apps.push.secureTunnelPort
1 to 65535
443
The port that the phone should use to communicate to the Web server when the secure tunnel is used.
apps.push.secureTunnelRequired
0 or 1
0
If 0, communications to the Web server do not require a secure tunnel. If 1, communications require a secure
tunnel.
apps.push.serverRootURL
URL
null
The URL of the application server you enter here is combined with the phone address and sent to the phone’s
browser. For example, if the application server root URL is http://172.24.128.85:8080/sampleapps and the relative
URL is /examples/sample.html, the URL that is sent to the microbrowser is
http://172.24.128.85:8080/sampleapps/examples/sample.html. Can be either HTTP or HTTPS.
apps.push.username
string
null
The user name to access the push server URL. Used with the password to respond to the HTTP Digest Challenge
from the handset.
Note: To enable the push functionality, the parameters apps.push.username and
apps.push.password must be set (not null).
Caution: Critical message conflict
Any new critical message overlays the previous one, but when 2 critical messages
come to the phone too close together and both require the phone to retrieve files,
the retrieval of the first file can be cut short by the retrieval of the second. This will
cause the first file to not be processed properly or not at all.
For instance, if the first critical message causes the phone to retrieve an html file
with javaScript in it to write text on the phone's display and the second critical
message causes the phone to retrieve a wav file to play a tune, the user may not
see the display on the phone but will just hear the tune.
Telephony Notification Parameters
(apps.telNotification.>
The <apps.telNotification/> parameter is used to control telephone notification events.
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Table 9-5: Telephone Heartbeat Parameter
Parameter
Permitted Values
Default
apps.telNotification.heartbeatTimeoutSeconds
0 – 65,535
0
Implements a "heartbeat" timer for telephony notifications that repeats the line registration notification on a periodic
basis if set. The default value for this parameter is 0, where the timer does not start at all, so there is no change to
the behavior unless the apps.telNotification.heartbeatTimeoutSeconds parameter is set to a non-zero value.
With a heartbeat set, when the phone goes out of range or is power cycled, notifications are not disrupted.
Table 9-6: Telephone Event Notification Parameters
Parameter
Permitted Values
Default
apps.telNotification.alarmEvent
0 or 1
0
Used with Personal Alarms. If 0, alarm event notification is disabled. If 1, notification is enabled. When this
parameter is enabled, an XML notification is sent when an alarm event occurs. Alarm events occur when Running,
Tilt, and Still alarms go off and when Duress/Emergency Calls are made.
apps.telNotification.callStateChangeEvent
0 or 1
0
If 0, call state change notification is disabled. If 1, notification is enabled.
apps.telNotification.incomingEvent
0 or 1
0
If 0, incoming call notification is disabled. If 1, notification is enabled.
apps.telNotification.InChargerEvent
0 or 1
0
0 or 1
0
0 or 1
0
If 0, notification is disabled. If 1, notification is enabled.
apps.telNotification.OutChargerEvent
If 0, is disabled. If 1, notification is enabled.
apps.telNotification.lineRegistrationEvent
If 0, line registration notification is disabled. If 1, notification is enabled.
apps.telNotification.networkUpEvent
0 or 1
0
If 0, network up notification is disabled. If 1, notification is enabled.
apps.telNotification.offhookEvent
0 or 1
0
If 0, off-hook notification is disabled. If 1, notification is enabled.
apps.telNotification.onhookEvent
0 or 1
0
If 0, on-hook notification is disabled. If 1, notification is enabled.
apps.telNotification.outgoingEvent
0 or 1
0
If 0, outgoing call notification is disabled. If 1, notification is enabled.
apps.telNotification.unInitializationEvent
0 or 1
0
If 0, line unregisteration notification is disabled. If 1, notification is enabled.
apps.telNotification.URL
URL
null
The URL to which the phone sends notifications of specified events. Can be either HTTP or HTTPS.
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Parameter
Permitted Values
Default
apps.telNotification.x.URL
URL
null
The URL to which the phone sends notifications of specified events, where x 1 to 9. Can be either HTTP or
HTTPS.
apps.telNotification.userLogInOutEvent
0 or 1
0
If 0, user login/logout notification is disabled. If 1, notification is enabled.
Admin Tip: Limitation for server URLs
The configured events will be sent to all apps.telNotification. URL and all
appstelNotification.x.URLs that are configured. There is no way you can configure
a few events for a specific server and remaining events for another server.
Open Application Interface parameters <oai.>
The Spectralink handsets support communications using the Open Application Interface (OAI).
You can set the connection parameters using the table shown next:
Table 9-6: Open Application Interface (OAI) Parameters
Parameter
Permitted values
Default
oai.gateway.address
IP address
Null
String of eight hexadecimal characters
Null
The address of the OAI server.
oai.userId
The lower four bytes of the six-byte OAI handset identifier in the OAI gateway.
If the value is null or invalid, the handset identifies itself to the OAI gateway using the MAC address of the handset;
otherwise, the handset identifies itself to the OAI gateway as 00:00:ww:xx:yy:zz. The upper two bytes are zero and
the lower four bytes are the oai.userid.
oai.keyRepeatAcceleration.enabled
0, 1
1
The default is "1", enabled, which is the usual behavior where the key repeat accelerates to one keypress every
50ms. To prevent excessive messaging when the key is held down, set the parameter to “0” which reduces the
key repeat rate to 500ms. This parameter only affects OAI calls.
Sample Configurations
The following sample configurations show how configuration parameters are used in
applications.
Push

apps.push.messageType is set to the appropriate display priority. For example, 3 –
Important Priority messages only.
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
apps.serverRootURL is set to the application server root URL. For example,
http://172.24.128.85:8080/sampleapps .

apps.push.username is set to the appropriate user name. For example, bob.

apps.push.password is set to the appropriate password. For example, 1234.
Telephony Notifications

apps.telNotification.URL is set to the URL where notifications should be sent.
For example, http://172.24.128.85:8080.

apps.telNotification.offhookEvent is set to 1 to enable notifications for offhook events.

apps.telNotification.onhookEvent is set to 1 to enable notifications for on-hook
events.

apps.telNotification.userLogInOut is set to 1 to enable notifications for user
login and logout events.

apps.telNotification.callStateChange is set to 1 to enable notifications for
call state change events.
State Polling

apps.statePolling.URL is set to the location where requested state polling
information should be sent. For example, http://172.24.128.85:8080.

apps.statePolling.responseMode is set to send the requested state polling
information to the configured URL, 1, instead of back to the requestor.

apps.statePolling.username is set to the appropriate username. For example,
bob.

apps.statePolling.password is set to the appropriate password. For example,
1234.
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Personal Alarms
The PersonalAlarm.cfg template is provided with the Spectralink software download in the
Config >Features folder. Use it to as a starting point to customize your deployment of this
feature.
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Chapter 10: System-Level Parameters
Certain parameters address system-level settings, such as DHCP.
Configuration File Encryption
You can encrypt configuration files (excluding the master configuration file), contact directories,
and configuration override files. See Encrypting Configuration Files for how to encrypt
configuration files. This section allow you to regulate the behavior of encrypted files.
Summary
Parameter
Used to:
sec.encryption.upload.callLists
Specify if the call list overrides file should be encrypted when it is uploaded
from the handset to the server
sec.encryption.upload.config
Specify if configuration files uploaded from the handset to the provisioning
server should be encrypted
sec.encryption.upload.dir
Specify if the contact directory is encrypted when it is uploaded from the
handset to the provisioning server
sec.encryption.upload.overrides
Specify if the configuration overrides file should be encrypted when it is
uploaded from the handset to the server
device.sec.configEncryption.key
Specify an encryption key so the handset can download encrypted files from
the provisioning server.
Table 10-1: Configuration File Encryption
Parameter
Permitted Values
Default
sec.encryption.upload.callLists1
0 or 1
0
The encryption on the handset-specific call lists that is uploaded to the provisioning server.
If 0, the file is uploaded unencrypted, and overwrites whatever handset-specific configuration file is on the server,
even if the file on the server is encrypted.
If 1, the call list is uploaded encrypted regardless of how it was downloaded. The file replaces any existing
handset-specific call lists file on the server.
sec.encryption.upload.config
0 or 1
0
The encryption on the handset-specific configuration file created and uploaded to the provisioning server when the
user selects Upload Configuration from the handset menu.
If 0, the file is uploaded unencrypted, and overwrites whatever handset-specific configuration file is on the server,
even if the file on the server is encrypted.
If 1, the file is uploaded encrypted and replaces any existing handset-specific configuration file on the server.
sec.encryption.upload.dir1
0 or 1
0
The encryption on the handset-specific contact directory that is uploaded to the provisioning server.
If 0, the directory is uploaded unencrypted regardless of how it was downloaded, the directory replaces whatever
handset-specific contact directory is on the server, even if the file on the server is encrypted.
If 1, the directory is uploaded encrypted regardless of how it was downloaded. The file replaces any existing
handset-specific contact directory file on the server.
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Parameter
Permitted Values
Default
sec.encryption.upload.overrides
0 or 1
0
The encryption on the handset-specific <MACaddress>-phone.cfg override file that is uploaded to the server.
If 0, the file is uploaded unencrypted regardless of how it was downloaded, the file replaces whatever file was on
the server, even if the file on the server is encrypted.
If 1, the file is uploaded encrypted regardless of how it was downloaded. The file replaces any existing handsetspecific override file on the server.
1
Change causes handset to restart or reboot.
Understanding Digital Certificates
Spectralink handsets are installed with a Spectralink-authenticated RSA device certificate. You
can use this certificate to create a secure connection between handset and server when
initiating Transport Layer Security (TLS) communications over protocols such as FTPS, HTTPS
and SIP. A device certificate isused with WPA2 Enterprise EAP-TLS security method.
You can download the Spectralink Root CA from
http://pki.Spectralink.com/aia/Spectralink%20Issuing%20CA.crt. Entering this URL exactly as
shown will download the certificate to your computer (you will be prompted on whether to save
or open the certificate. Note that the certificate is set to expire on March 9, 2044.
Note: How do I get the Spectralink certificates?
Spectralink CA certificates can be obtained from:
http://pki.spectralink.com/aia/Spectralink%20Issuing%20CA.crt
http://pki.spectralink.com/aia/Spectralink%20Root%20CA.crt
http://pki.spectralink.com/aia/Spectralink%20Issuing%20CA%20BLCAI01.crt
Spectralink uses the X.509 standard, which defines what information can go into a certificate.
An X.509 digital certificate is a digitally signed statement. All X.509 certificates have the
following fields, in addition to the signature:
 Version—This identifies which version of the X.509 standard applies to this certificate,
which in turn affects what information can be specified in the certificate.
 Serial Number—The entity that created the certificate is responsible for assigning it a
serial number to distinguish it from other certificates it issues.
 Signature Algorithm Identifier—This identifies the algorithm used by the Certificate
Authority (CA) to sign the certificate.
 Issuer Name—The X.500 name of the entity that signed the certificate. This is normally a
CA. Using this certificate means trusting the entity that signed this certificate.
 Validity Period—Each certificate is valid for a limited amount of time. This period is
described by a start date and time and an end date and time, and can be as short as a
few seconds or almost as long as a century.
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 Subject Name—The name of the entity whose public key the certificate identifies. This
name uses the X.500 standard, so it is intended to be unique across the Internet.
 Subject Public Key Information—This is the public key of the entity being named,
together with an algorithm identifier that specifies to which public key cryptographic
system this key belongs and any associated key parameters.
The following is an example of a Spectralink device certificate when opened in Microsoft
Windows.
The device certificate and associated private key are stored on the handset in its non-volatile
memory as part of the manufacturing process. For more information on digital certificates, see
Public Key Infrastructure (X.509) and RFC 2459: Internet X.509 Public Key Infrastructure .
Web Info: Using custom certificates with Spectralink handsets
As of 84-Series software 4.0.0, you can install custom device certificates on your
Spectralink handsets. These certificates are installed in the same way custom CA
certificates are installed. See Technical Bulletin CS-13-06: Using custom
certificates with Spectralink 8400 handsets.
To determine if there is a custom device certificate on a Spectralink handset:
Navigate to Settings> Advanced Settings> [enter password]> Administration
Settings> TLS Security> Custom Device Credentials.
To view the status of the Spectralink device certificate on the handset:
Navigate to Settings > Status> Platform> Phone> Device Certificate
○
Factory Installed is displayed if the certificate is available in flash memory, all the
certificate fields are valid (listed above), and the certificate has not expired.
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Signed by: displays the common name of the signing Certificate Authority (CA), e.g.
“Spectralink”.
○
Self-signed is displayed if other certificates have been installed and the Signed by
field could be a MAC address.
○
Device Certificate: Not Installed is displayed if the certificate is not available in
flash memory (or the flash memory location where the device certificate is to be
stored is blank).
○
Device Certificate: Invalid is displayed if the certificate is not valid.
Note: Device Certificate Shown as Self-Signed
Some Spectralink handsets manufactured after December, 2011 report the device
certificate as ‘self-signed’ and not as ‘Factory Installed’. The difference indicates
that different issuing CAs were used to generate the certificates. As long as the
authenticating server trusts the Spectralink Root CA that issued these certificates,
the handsets will operate correctly.
About Digital Certificates
Certificates enable handsets and servers to authenticate each other before permitting any
exchange of data. Certificates are utilized by the Transport Layer Security (TLS) protocol which
ensures that no third party may eavesdrop or tamper with any message. Certificates also allow
for securely encrypted data to be passed between a SIP client and the PBX, (including Skype
for Business), for secure provisioning under the FTPS, HTTPS protocols, for secure browser
communications and for secure syslogging.
A number of commonly-used certificates are loaded in the phone at the factory. These are
called “built-in” certificates. See Appendix E: Trusted Certificate Authority List for the list of
authorities.
Types of certificates
CA certificate used for 802.1x Authentication
A CA certificate can be used for 802.1x Authentication. EAP-TLS and EAP-FAST.
The client (the handset) uses the CA certificate to verify the CA signature of the Authentication
server (the RADIUS server) certificate before establishing a secure connection. This way the
handset knows it’s talking to the correct far side. It is provided by a Certificate Authority (hence
CA) or an IT administrator and must be loaded on the handset during configuration. This type of
certificate is also called a server certificate as it certifies the server. The certificate is usually the
same for every device and is therefore also known as a “public” certificate.
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CA certificate used for Secure Syslog
A CA certificate can be used for Secure Syslog. Syslog can be configured to use a secure TLS
Tunnel using the CA certificate.
CA certificate used for Secure Provisioning
A CA Certificate can be used for Secure Provisioning: FTPS and HTTPS. This prevents the
configuration parameters of a device from being exposed during wireless transfer.
CA certificate used for SIP Communication
Some PBXes allow for communication between the handset and the device to occur using TLS.
A CA certificate can be used to set up MTLS (mutual TLS) between the handset and the PBX in
this case.
Other
CA Certificate used for Browser Communication
CA certificate used for LDAP Communication
Device certificate used for 802.1x Authentication
A device certificate validates the handset to the RADIUS server during EAP-TLS Authentication.
Spectralink 84-Series handsets are shipped with a Spectralink device certificate and its
associated private key known only to the phone which can be used by EAP-TLS for Wi-Fi
security. The Spectralink device certificate uses the handset’s MAC address as its common
name which is also its Identity.
PAC file
The Protected Access Credential (PAC) is a proprietary Cisco method for provisioning
certificates. The PAC can be either a specific to a device or common to a group of devices. It is
generated by the RADIUS server and must be loaded either manually or automatically. EAPFAST is used with Cisco® products and by a number of other WLAN vendors.
Configuring certificates
To configure a certificate, you will follow these general steps:
1
Loading the certificate
2
Assigning the certificate to a Platform or Application profile
3
Connecting the platform to how it will be used
4
Configuring additional parameters (if necessary)
For purposes of configuration, certificates are divided into two different categories—Platform
and Application. If set, the two platform certificates are stored in the device’s flash memory and
are used by both the Updater and the application parts of the software. If any are set, the six
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application certificates are stored in the device’s RAM and are used by the application part of
the software.
Platform certificates
Platform types are designed to be used for parameters that are required for initial wireless
configuration and provisioning. Their configuration parameters begin with device. Two
Platform certificate “slots” are available. Platform profiles are used by:

802.1x (parameters are called “dotx” and refer to Wi-Fi security methods)

Provisioning server transport protocol

Syslog server
#1 Loading Platform certificates
You will load the Platform certificates using the below parameters in the wireless.cfg file.
Spectralink recommends that you load Wi-Fi certificates in slot1 and provisioning/syslog server
certificates in slot2. This parallels where QNC/SLIC loads certificates for the Wi-Fi security
method and for the provisioning server.

device.sec.TLS.customCaCert1 “slot1”

device.sec.TLS.customCaCert2 “slot2”
#2 Assigning the certificates to a Profile List
You can broadly or narrowly define which certificates in the handset’s certificate store can be
used for authentication purposes. There are two profile lists, CaCertList1 and CaCertList2. The
default is All which means the handset will look through all certs loaded on the phone to locate
the one it needs for a particular use. The options are single Platform1 or Platform2 or various
combinations.
All references any loaded certificate. All is the default.
Platform1 references the certificate loaded into device.sec.TLS.customCaCert1
Platform2 references the certificate loaded into device.sec.TLS.customCaCert2
Platform1AndPlatform2 references either one of the two platform certs.
Builtin references the large number of well-known certificate authorities pre-loaded into the
handset.
Instead of having the application search through all the certificates, you can use the CaCertList
parameter to point the application to the exact certificate(s) you want it to use which speeds up
the process and makes bringing up the handsets more efficient. Spectralink recommends
assigning Platform1 to CaCert1 and Platform2 to CaCert2.Use the following parameters to
configure the Profile Lists:

Profile List 1 is configured with device.sec.TLS.profile.CaCertList1

Profile List 2 is configured with device.sec.TLS.profile.CaCertList2
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#3 Connect the Platform to a use
In this step you assign the program that is using the certificate(s) to a Profile List that you
created above. More than one program can use a certificate. Spectralink recommends using
Platform 1 or Platform 2 for Platform uses; provisioning server, syslog server and Wi-Fi security:

Provisioning server: device.sec.TLS.profileSelection.provisioning

Wi-Fi security: device.sec.TLS.profileSelection.dot1x

Syslog server: device.sec.TLS.profileSelection.syslog
Example: 802.11X with EAP-TLS
How the parameters are used to set Wi-Fi security method EAP-TLS.
How the parameters work:
device.wifi.securityMode=”WPA2-Enterprise”
Establishes WPA2-Enterprise as the type of Wi-Fi security
device.wifi.wpa2Ent.method=”EAP-TLS”
Establishes the WPA2-Enterprise method
device.wifi.wpa2Ent.user=”[MACaddress]”
Sets the identity for the method. The MACaddress is used for the device certificate. A generic identity that is
accepted by the RADIUS server is configurable by QNC/SLIC.
device.sec.TLS.customCaCert1=”[certificate]”
Loads the actual certificate onto the handset. Referred to as Platform1.
device.sec.TLS.profile.caCertList1=”Platform1”
Directs the handset to use the Platform2 certificate which is the customCaCert1 loaded above.
device.sec.TLS.profileSelection.dot1x
Tells the handset that for EAP-TLS authentication it will find the certificate it needs in Platform Profile 1.
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Servers that use Platform certificates
Both provisioning servers and syslog servers can use certificates. If you use HTTPS or FTPS
you will need to configure a certificate for the server. These are loaded and assigned exactly as
described above for Wi-Fi security methods EAP-TLS or PEAP.
Spectralink recommends that you use different CA certificates for servers than you do for Wi-Fi
security methods and assign the Platform Profiles accordingly. Provisioning server certificates
can be configured with QNC/SLIC.
Parameter
Permitted Values
Default
device.sec.TLS.profileSelection.provisioning1
PlatformProfile1,
PlatformProfile2
Null
The TLS Platform Profile to use for provisioning, either Platform Profile 1 or TLS Platform Profile 2.
device.sec.TLS.profileSelection.syslog1
PlatformProfile1,
PlatformProfile2
Null
The TLS Platform Profile to use for syslog, either Platform Profile 1 or TLS Platform Profile 2.
Example: FTPS
Your configuration for implementing FTPS on a provisioning server could look like this:
How the parameters work:
device.prov.serverType=”FTPS”
Sets the FTPS protocol on the provisioning server.
device.sec.TLS.customCaCert2=[certificate]
Loads the actual certificate onto the handset. Also referred to as Platform2.
device.sec.TLS.profile.caCertList2="Platform2"
Directs the handset to use the Platform2 certificate which is the customCaCert2 loaded above.
device.sec.TLS.profileSelection.provisioning="PlatformProfile2"
Tells the handset that for provisioning server authentication it will find the certificate it needs in Platform Profile
2.
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#4 Common name validation
When a provisioning or syslog server sends its certificate to the client to be validated one of the
default steps the phone will do is to verify that the Common Name tied to the certificate matches
the name of the server it is talking to. In most cases this won’t be much of a problem, but it
could be an issue if you are using self-signed certificates. If the names don’t match, the default
behavior is to deny the connection and disconnect.
You can disable common name validation and allow the phone to ignore the name of the server
and the name in the certificate. Note that this is less secure but for many environments it will be
perfectly acceptable. If you need to disable common name validation use the following
parameters.
Parameter
Permitted Values
Default
device.sec.TLS.dot1x.strictCertCommonNameValidation
0 or 1
1
If set to 1, 802.1X always verifies the server certificate for commonName/SubjectAltName match with the server
hostname that the phone is trying to connect.
device.sec.TLS.prov.strictCertCommonNameValidation
0 or 1
1
If set to 1, provisioning always verifies the server certificate for commonName/SubjectAltName match with the
server hostname that the handset is trying to connect. Must be disabled for FTPS.
device.sec.TLS.syslog.strictCertCommonNameValidation
0 or 1
1
If set to 1, syslog always verifies the server certificate for commonName/SubjectAltName match with the server
hostname that the handset is trying to connect.
sec.TLS.SIP.strictCertCommonNameValidation
0 or 1
1
If 1, enable common name validation for SIP.
If you are not sure whether or not you need to disable common name validation, check the logs.
What you will likely see in the log of your FTPS or HTTPS server is that the device will establish
a connection but will not send its authentication credentials.
Application certificates
Application certificates can be used by any application that requires a certificate that is not a
Platform type. Six Application “slots” are available. Loading and assigning application
certificates follows the same sequence as detailed above for Platform certificates. Application
certificates are not configured by QNC/SLIC.
This section also offers two Platform certificate slots that are separate and different from the two
device parameters that can be configured by QNC/SLIC.
An Application may have both its own application certificate and a corresponding custom device
certificate.
Either Platform or Application slots can be used by:

SIP

Browser

LDAP
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#1 Loading Application certificates
Application certificates are assigned a number (1-6) and that number is used to identify the
application when installing and assigning the certificate it requires. Application certificates are
installed using the following parameters for application slots:
Parameter
Permitted Values
Default
sec.TLS.customCaCert.x
String
Null
The custom certificate for Application Profile x (x= 1 to 6). E.g. The certificate loaded in
sec.TLS.customCaCert.1 is referred to as ApplicationProfile1.
Example of an ApplicationProfile1 certificate:
#2 Connecting the certificate to the application
In this step, you will link the certificate loaded to the defined profile. You have these options

Platform1 or Platform2

Applicationx (1-6) the certificates loaded above for an application.
Parameter
Permitted Values
Default
sec.TLS.profileSelection.SIP
sec.TLS.profileSelection.syslog
sec.TLS.profileSelection.LDAP
sec.TLS.profileSelection.browser
PlatformProfile1,
PlatformProfile2,
ApplicationProfile1,
ApplicationProfile2,
ApplicationProfile3,
ApplicationProfile4,
ApplicationProfile5,
ApplicationProfile6
PlatformProfile1
The Platform or Application Profile to use for the application identified in the parameter. Note that you can assign a
Platform certificate to an application using this parameter.
The TLS platform profile orTLS application profile to use for applications.
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Parameter
Permitted Values
Default
The TLS platform profile or TLS application profile (see preceding list) to use for the
Corporate Directory.
The TLS platform profile or TLS application profile (see preceding list) to use for SIP
operations.
The TLS platform profile to use for syslog operations.
Example of SIP using an Application profile.
Device certificates
Device Certificates are used in the following situations:

Mutual TLS Authentication: Allows a server to verify that a device is truly a Spectralink
device (and not a malicious endpoint or software masquerading as a Spectralink device).
This could be used for tasks like provisioning, or SIP signaling using TLS signaling.

Secure HTTP (https) access to the web server on the phone at https://<IP ADDRESS
OF PHONE>. The web server is used for certain configuration and troubleshooting
activities.

Secure communications utilizing the Spectralink Applications API.
The device can be configured to use different device certificates for each operation (or the
same device certificate can be used for multiple operations). The operations available are:

802.1X

Syslog

Provisioning
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
SIP

Browser

Presence

LDAP
This configuration can be done:
 Using configuration files.
 From the phone menu.
There are several options for utilizing device certificates on the phone.

A factory installed device certificate. This certificate is installed at the time of
manufacture and is unique to a device (based on the MAC address) and signed by the
Spectralink Certificate Authority (CA). Since it is installed at the time of manufacture, it is
the easiest option for out-of-box activities; in particular, device provisioning.

Two platform device certificates. These certificates are loaded onto the device by the
system administrator and can be configured to be used for any of the following
purposes: 802.1X Authentication, provisioning, syslog, SIP signaling, browser
communications, presence, and LDAP.

Six Application device certificates. These certificates are loaded onto the device by the
system administrator and can be used for all of the operations listed above for platform
certificates with the exception of 802.1X, syslog, and provisioning.
Configuration options are used to select which type of device certificate is used for each of the
secure communication options. By default, all operations will utilize the factory installed device
certificate.
To configure your web servers and/or clients to trust Spectralink factory installed device
certificates, you will need to download the Spectralink Issuing CA certificate and the Root CA
will need to be installed on the server. See Appendix F: Spectralink Certificates for link. You
may also need to download the Intermediate CA certificates; this is determined by the
authenticating server.
Platform Device Certificates
Platform device certificates can be installed using one of the following methods:

Using a configuration file. You must enter the certificate in PEM or PKCS# certificate
format.
device.sec.TLS.customDeviceCertX.publicCert
device.sec.TLS.customDeviceCertX.privateKey
device.sec.TLS.customDeviceCertX.set
where X = 1 or 2.
Assign the certificate to Platform Profile 1 or 2 using:
device.sec.TLS.profile.deviceCertx
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where X = 1 or 2.
Parameter
Permitted Values
device.sec.TLS.profile.deviceCert1 (TLS Platform Profile 1)
device.sec.TLS.profile.deviceCert2 (TLS Platform Profile 2)
Builtin, Platform1,
Platform2
Default
Null
Choose the device certificate(s) for TLS Platform Profile 1 and TLS Platform Profile 2 to use for authentication.
device.sec.TLS.customDeviceCert1.publicCert
device.sec.TLS.customDeviceCert2.publicCert
Enter the signed custom
device certificate in PEM
format (X.509)
Null
device.sec.TLS.customDeviceCert1.privateKey
device.sec.TLS.customDeviceCert2.privateKey
Enter the corresponding
signed private key in PEM
format (X.509)
Null
device.sec.TLS.customDeviceCert1.set
device.sec.TLS.customDeviceCert2.set
0 or 1
0
Note that you use a single .set parameter to enable or disable only these two related <device/> parameters device.sec.TLS.customDeviceCertX.publicCert and
device.sec.TLS.customDeviceCertX.publicKey. All other <device/> parameters have their own
corresponding .set parameter that will enable or disable that parameter.

From the phone. Navigate to Settings > Advanced Setti ngs> Admin Settings > TLS
Security > Configure TLS Profiles > Custom Device Credentials. You must enter a
URI linking to a PEM formatted certificate in PKCS #7 certificate format.

By generating a Certificate Signing Request (CSR). See Generating a Certificate Signing
Request.

The total size of the platform certificate plus private key is restricted as follows:
○
Platform Certificate –8192 bytes.
○
Platform Private Key–4096 bytes.
If the administrator attempts to download a certificate that is too big, ‘Failed to save
certificate’ displays on the phone’s screen and a message appears in the log file (shown
next).
0529103935|tls |4|03|Device credential invalid: Cert is not proper in
the certificate
Application Device Certificates
Application certificates can be installed using one of the following methods:

Using a configuration file. You must enter the certificate in PKCS #7 certificate format.
The configuration parameters are:
sec.TLS.customDeviceCert.x
sec.TLS.customDeviceKey.x
Assign the certificate to a Platform or Application Profile using:
sec.TLS.profile.x.deviceCert
where x = 1-6.
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Parameter
Permitted Values
Default
sec.TLS.customDeviceCert.x
String
Null
The custom device certificate for TLS Application Profile x (x= 1 to 6).
sec.TLS.customDeviceKey.x
String
Null
The custom device certificate private key for TLS Application Profile x (x= 1 to 6).
sec.TLS.profile.x.deviceCert
Factory, Platform1, Platform2,
Application1, Application2,
Application3, Application4,
Application5, Application6
Factory
The device certificate to use for TLS Application Profile x (x = 1 to 6).
Caution: Exposing the device key is insecure
This method is available but exposes the private key. This is not secure and is
generally not recommended. Allow the phone to generate a certificate signing
request instead.
Example custom device Cert and Key

From the phone. Navigate to Settings > Advanced > Admin Settings > TLS Security
> Configure TLS Profiles > Custom Device Credentials. You must enter a URI linking
to a PEM formatted certificate in PKCS #7 certificate format.

By generating a Certificate Signing Request (CSR). See below. There is no size
constraint on the application certificate and private key.
Cipher Suite Parameters
The handset administrator can control which cipher suites will be offered/accepted during TLS
session negotiation. The handset supports the cipher suites shown in the next table. The ‘Null
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Cipher’ is a special case option which will not encrypt the signaling traffic, and is useful for
troubleshooting purposes.
You will only need to configure cipher suite parameters if you are not using the default.
Cipher
Cipher Suite
ADH
ADH-RC4-MD5, ADH-DES-CBC-SHA, ADH-DES-CBC3-SHA, ADH-AES128-SHA, ADHAES256-SHA
AES128
AES128-SHA
AES256
AES256-SHA
DES
DES-CBC-SHA, DES-CBC3-SHA
DHE
DHE-DSS-AES128-SHA, DHE-DSS-AES256-SHA, DHE-RSA-AES128-SHA, DHE-RSAAES256-SHA
EXP
EXP-RC4-MD5, EXP-DES-CBC-SH, EXP-EDH-DSS-DES-CBC-SHA, EXP-DES-CBC-SHA,
EXP-ADH-RC4-MD5, EXP-ADH-DES-CBC-SHA, EXP-EDH-RSA-DES-CBC-SHA
EDH
EDH-RSA-DES-CBC-SHA, EDH-DSS-DES-CBC3-SHA, EDH-DSS-CBC-SHA
NULL
NULL-MD5, NULL-SHA
RC4
RC4-MD5, RC4-SHA
Parameter
Permitted Values
device.sec.TLS.profile.cipherSuite1 (Platform Profile 1)
device.sec.TLS.profile.cipherSuite2 (Platform Profile 2)
Default
String
Null
The cipher suites to use for Platform Profile 1 and Platform Profile 2) Use the Cipher Suites table above.
device.sec.TLS.profile.cipherSuiteDefault1 (Platform Profile 1)
device.sec.TLS.profile.cipherSuiteDefault2 (Platform Profile 2)
0 or 1
Null
The cipher suite to use for Platform Profile 1 and Platform profile 2. If set to 0, the custom cipher suite will be used.
If set to 1, the default cipher suite will be used.
sec.TLS.cipherList
String
“RSA:!EXP:!LOW:!NUL
L:!MD5:@STRENGTH”
String
NoCipher
String
NoCipher
String
NoCipher
String
NoCipher
String
NoCipher
The global cipher list parameter.
sec.TLS.browser.cipherList
The cipher list for the browser.
sec.TLS.LDAP.cipherList
The cipher list for the corporate directory.
sec.TLS.prov.cipherList
The cipher list for provisioning.
sec.TLS.SIP.cipherList
The cipher list for SIP.
sec.TLS.syslog.cipherList
The cipher list for syslog.
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Generating a Certificate Signing Request
You may need a certificate to perform a number of tasks, for example, TLS with mutual
authentication. For version 4.12 and above, the CSR is 2048 bits, while 4.11 and below generated
512 bit keys.
To obtain a certificate you need to:

Request a certificate from a Certificate Authority (CA) by creating a certificate signing
request (CSR).

Forward the CSR to a CA to create a certificate. If your organization doesn’t have its
own CA, you will need to forward the CSR to a company like Symantec. If successful,
the CA will send back a certificate that has been digitally signed with their private key.
After you receive the certificate, you can download it to the handset:

Using a configuration file

Through the handset’s user interface
To generate a certificate signing request on a Spectralink handset:
1
Navigate to Settings> Advanced Settings> [enter password]> Administration
Settings> Generate CSR.
2
From the Generate CSR Screen, enter information as shown next. You must fill in the
Common Name and Country fields. The Organization, Email Address, Country, and
State fields are optional.
3
Press Generate.
A message CSR generation completed displays on the handset’s screen.
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Downloading Certificates to a Spectralink Phone
You can download certificates to a Spectralink handset by specifying a URL where the
certificate is currently stored. You can install up to eight CA certificates and eight device
certificates on the handset. You can refresh certificates when they expire or are revoked. You
can delete any CA certificate or device certificate that you install.
Maximum Size for Certificates
The maximum certificate size on both Platform CA1 and Platform CA2 is 4KB.
To download a certificate to a Spectralink handset:
1
Navigate to Settings> Advanced Settings> [enter password]> Administration
Settings> TLS Security and select Custom CA Certificates or Custom Device
Certificates.
2
Select one of the certificate slots from the displayed list, then press the Install softkey.
When prompted, enter the administrative password and press the Enter softkey. The
default administrative password is 456.
3
Enter the URL where the certificate is stored.
For example, http://bootserver1.vancouver.Spectralink.com/ca.crt
4
Select the Enter softkey.
The certificate is downloaded. The certificate’s MD5 fingerprint displays to verify that the
correct certificate is to be installed.
5
Select the Accept softkey.
The certificate is installed successfully.
The appropriate certificate menu displays the certificate’s common name.
DNS SIP Server Name Resolution
If a DNS name is given for a proxy/registrar address, the IP address(es) associated with that
name will be discovered as specified in RFC 3263. DNS record lookup behavior may
additionally be influenced if the transport mechanism or port number are specified in the
phone's configuration. If the phone is allowed to operate using the default configuration
parameter combination of transport method and port number, the handset will attempt record
lookups in the following order until it receives a successful lookup response; NAPTR, then SRV,
then A record.
If a transport method is specified to be something other than DNS NAPTR, the phone will skip
the attempt to query for a DNS NAPTR record, and will instead attempt a SRV record lookup for
the proxy/registrar address using the transport mechanism specified, followed by an A name
record lookup if the SRV record lookup fails. If a port number is specified via the configuration
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parameters, the phone will not attempt NAPTR or SRV record lookups. The only lookup will be
an A record. If no port is specified via configuration parameters, and none is found through DNS
mechanisms, port 5060 will be used by default for TCP or UDP transport types, and 5061 will be
used as the port number for TLS.See RFC 3263 for an example.
Caution: No DNS Resolution Will Cause Failover
Failure to resolve a DNS name is treated as signaling failure that will cause a
failover.
Behavior When the Primary Server Connection Fails
For Outgoing Calls (INVITE Fallback)
When the user initiates a call, the handset will go through the following steps to connect the call:
1
The handset will try to call the working server.
2
If the working server does not respond correctly to the INVITE, the handset will try and
make a call using the next server in the list (even if there is no current registration with
these servers). This could be the case if the Internet connection has gone down, but the
registration to the working server has not yet expired.
3
If the second server is also unavailable, the handset will try all possible servers (even
those not currently registered) until it either succeeds in making a call or exhausts the list
at which point the call will fail.
At the start of a call, server availability is determined by SIP signaling failure. SIP signaling
failure depends on the SIP protocol being used:
 If TCP is used, then the signaling fails if the connection fails or the Send fails.
 If UDP is used, then the signaling fails if ICMP is detected or if the signal times out. If the
signaling has been attempted through all servers in the list and this is the last server, then
the signaling fails after the complete UDP timeout defined in RFC 3261. If it is not the last
server in the list, the maximum number of retries using the configurable retry timeout is
used.
Caution: Use Long TTLs to Avoid DNS Timeout Delays
If DNS is used to resolve the address for Servers, the DNS server is unavailable,
and the TTL for the DNS records has expired, the handset will attempt to contact
the DNS server to resolve the address of all servers in its list before initiating a call.
These attempts will timeout, but the timeout mechanism can cause long delays (for
example, two minutes) before the handset call proceeds using the working server.
To prevent this issue, long TTLs should be used. Spectralink recommends
deploying an on-site DNS server as part of the redundancy solution.
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Phone Configuration
The handsets at the customer site are configured as follows:
 Server 1 (the primary server) will be configured with the DNS name of the service provider
call server. The IP address of the server(s) will be provided by the DNS server, for
example: reg.1.server.1.address=voipserver.serviceprovider.com .
 Server 2 (the fallback server) will be configured to the address of the router/gateway that
provides the fallback telephony support and is on-site, for example:
reg.1.server.2.address=172.23.0.1 .
Note: Caution When Using Multiple Servers Per Registration
It is possible to configure the handset for more than two servers per registration,
but you need to exercise caution when doing this to ensure that the handset and
network load generated by registration refresh of multiple registrations does not
become excessive. This would be of particular concern if a handset had multiple
registrations with multiple servers per registration and it is expected that some of
these servers will be unavailable.
Phone Operation for Registration
After the handset has booted up, it will register to all the servers that are configured.
Server 1 is the primary server and supports greater SIP functionality than other servers. For
example, SUBSCRIBE/NOTIFY services used for features such as shared lines, presence, and
BLF will be established only with Server 1.
Upon the registration timer expiry of each server registration, the handset will attempt to reregister. If this is unsuccessful, normal SIP re-registration behavior (typically at intervals of 30 to
60 seconds) will proceed and continue until the registration is successful (for example, when the
Internet link is once again operational). While the primary server registration is unavailable, the
next highest priority server in the list will serve as the working server. As soon as the primary
server registration succeeds, it will return to being the working server.
Failover to Servers that are Not Registered
If reg.x.server.y.register is set to 0, the handset will not register to that
server. However, an INVITE will fail over to that server if all higher priority servers
are down.
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Incoming Signaling Validation
You can choose from three optional levels of security for validating incoming network signaling:
 Source IP address validation
 Digest authentication
 Source IP address validation and digest authentication
Summary
Parameter
Used to:
voIpProt.SIP.requestValidation.x.method
Specify what type of validation to perform
voIpProt.SIP.requestValidation.x.request
Set the name of the method for which validation will be applied
voIpProt.SIP.requestValidation.x.request.y.event
Determine which events within the Event header should be
validated
Table 10-2: Incoming Signal Validation
Parameter
Permitted Values
Default
voIpProt.SIP.requestValidation.x.method1
Null, source, digest, both,
all
Null
If Null, no validation is made. Otherwise this sets the type of validation performed for the request:
source: ensure request is received from an IP address of a server belonging to the set of target registration servers;
digest: challenge requests with digest authentication using the local credentials for the associated registration (line);
both or all: apply both of the above methods
voIpProt.SIP.requestValidation.x.request1
Null, INVITE, ACK , BYE,
REGISTER, CANCEL,
OPTIONS, INFO,
MESSAGE, SUBSCRIBE,
NOTIFY, REFER, PRACK,
UPDATE
Null
Sets the name of the method for which validation will be applied.
Note: Intensive request validation may have a negative performance impact due to the additional signaling required
in some cases.
voIpProt.SIP.requestValidation.x.request.y.event
A valid string
Null
1
Determines which events specified with the Event header should be validated; only applicable when
voIpProt.SIP.requestValidation.x.request is set to SUBSCRIBE or NOTIFY.
If set to Null, all events will be validated.
1
Change causes handset to restart or reboot.
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Instant Messaging
You can use Microsoft Office Communications Server (OCS) 2007 R2 and Microsoft Skype for
Business servers for instant messaging.
In our IM example, line 2 becomes the IM line. Skype for Business or Microsoft Lync Server
2013 or 2010 (depending on your software version) is deployed. For User Profiles Deployment,
the reg.2 parameters are located in the login.cfg template.
Table 10-3: IM
Parameter
Permitted Values
Default
feature.messaging.enabled
0 or 1
0
If 0, the instant messaging feature is disabled. If 1, the feature is enabled.
feature.presence.enabled
0 or 1
0
If 0, the presence feature — including user status — is disabled. If 1, the presence feature is enabled with the
status options.
reg.x.telephony
0 or 1
1
If 0, telephony calls are not enabled on this registration (use this value if the registration is used with Microsoft
Office Communications Server 2007 R2 or Microsoft Skype for Business or Microsoft Lync Server 2013 or 2010.. If
1, telephony calls are enabled on this registration.
reg.x.auth.useLoginCredentials
0 or 1
0
If 0, login credentials are not used for authentication to the server on registration x. If 1, login credentials are used
for authentication to the server. Note: This must be set to 1 for instant messaging on the Spectralink handsets.
reg.x.server.y.address
dotted-decimal IP address
or hostname
Null
The IP address or host name of a SIP server that accepts registrations. If not Null, all of the parameters in this
table will overrule the parameters specified in voIpProt.server.*.
Note: If this parameter is set, it will overrule even if the DHCP server is available. If this registration is used for
Microsoft Office Communications Server 2007 R2, this parameter must be in the form
OCShostname.OSCdomain_name.
For IM and Presence, use reg.2.server.1.address to contact and register with the Skype for Business server.
reg.x.server.y.port
0, 1 to 65535
0
The port of the sip server that specifies registrations. If 0, the port used depends on
reg.x.server.y.transport.
reg.x.server.y.register
0 or 1
1
If the outbound proxy can route calls without the handset being registered to it, set this value to 0.
reg.x.server.y.specialInterop
standard, ocs2007r2,
lcs2005, lync2010,
lync2013, Skype for
Business
standard
Specify if this registration should support Microsoft Office Communications Server 2007 R2 (ocs2007r2), Microsoft
Live Communications Server 2005 (lcs2005), or Microsoft Skype for Business or Microsoft Lync Server 2013 or
2010..
Note: To use instant messaging, set this parameter to ocs2007r2.
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Parameter
Permitted Values
Default
reg.x.server.y.transport
Null, DNSnaptr,
TCPpreferred, UDPOnly,
TLS, TCPOnly
DNSnaptr
The transport method the handset uses to communicate with the SIP server.
Null or DNSnaptr – if reg.x.server.y.address is a hostname and reg.x.server.y.port is 0, do NAPTR
then SRV look-ups to try to discover the transport, ports and servers, as per RFC 3263. If
reg.x.server.y.address is an IP address, or a port is given, then UDP is used.
TCPpreferred – TCP is the preferred transport; UDP is used if TCP fails.
UDPOnly – only UDP will be used.
TLS – if TLS fails, transport fails. Leave port field empty (will default to 5061) or set to 5061.
TCPOnly – only TCP will be used.
roaming_buddies.reg
Null, 1 to 34
Null
The index of the registration which has roaming buddies support enabled. If Null, the roaming buddies feature is
disabled.
Note: This parameter must be set if the call server is Microsoft Live Communications Server 2005, Microsoft Office
Communications Server 2007 R2, or Microsoft Skype for Business.
sec.TLS.customCaCert.x
String
Null
The custom certificate for TLS Application Profile x (x= 1 to 6). This parameter is not in the template but may need
to be added. It is not required if you use DHCP option 43 to tell the handset where to get its certificate
automatically but if that is not available , then add the certificate using this parameter.
sec.TLS.profileSelection.SIP
a TLS profile
PlatformProfile1
The TLS platform profile or TLS application profile to use for SIP operations. Permitted values are:

PlatformProfile1

PlatformProfile2

ApplicationProfile1

ApplicationProfile2

ApplicationProfile3

ApplicationProfile4

ApplicationProfile5

ApplicationProfile6
voIpProt.SIP.mtls.enable
0 or 1
1
If 0, TLS with mutual authentication is disabled. If 1, TLS with mutual authentication is enabled. Used in
conjunction with Microsoft Skype for Business or Microsoft Lync Server 2013 or 2010..
voIpProt.SIP.IM.autoAnswerDelay
0 to 40, seconds
10
The time interval from receipt of the instant message invitation to automatically accepting the invitation.
If users have a PC that is logged to their IM account, should the PC auto-answer incoming IMs if no action is taken
on the handset? If yes, set to 30, if no, set to 10.
Yes: voIpProt.SIP.IM.autoAnswerDelay=”30”
No: voIpProt.SIP.IM.autoAnswerDelay=”10”
Table 10-4: Setting Up Microsoft Office Communications Server 2007 R2 Integration
Parameter
Used to:
feature.presence.enabled
Turn the presence feature on or off
feature.messaging.enabled
Turn the messaging feature on or off
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Parameter
Used to:
reg.x.lineKeys
Specify the number of line keys to use for a single registration
Parameter
Permitted Values
feature.presence.enabled1
0 or 1
Default
0
If 0, the presence feature — including user status — is disabled. If 1, the presence feature is enabled with status
options.
feature.messaging.enabled
0 or 1
0
If 0, the instant messaging feature is disabled. If 1, the feature is enabled.
reg.x.lineKeys
1 to 24
1
Specify the number of line keys to use for a single registration.
1
Change causes handset to restart or reboot.
Example Instant Messaging Configuration
The following illustration shows you how to enable instant messaging in the site.cfg template.
Table 10-5: Other Instant Messaging Parameters
Parameter
Permitted Values
Default
messaging.maxImMessages
10 to 1000
1000
The maximum number of instant messages allowed.
messaging.quickNotes.x
String of up to 128 characters
Null
Up to 10 (x =1 to 10) quick notes for use in instant messages
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IP Type-of-Service
The type-of-service field in an IP packet header consists of four type-of-service (TOS) bits and a
3-bit precedence field. See the next table for available parameters. Each TOS bit can be set to
either 0 or 1. The precedence field can be set to a value from 0 through 7. The type of service
can be configured specifically for RTP packets and call control packets, such as SIP signaling
packets.
Table 10-6: IP Type-of-Service (ToS)
Parameter
Permitted Values
Default
qos.ip.callControl.max_reliability1
qos.ip.callControl.max_throughput1
qos.ip.callControl.min_cost1
qos.ip.callControl.min_delay1
qos.ip.callControl.precedence1
0 or 1
0 or 1
0 or 1
0 or 1
0 -7
0
0
0
1
5
Set the bits in the IP ToS field of the IP header used for call control. Specify whether or not to set the max reliability
bit, the max throughput bit, the min cost bit, the min delay bit, and the precedence bits.
If 0, the bit in the IP ToS field of the IP header is not set. If 1, the bit is set.
qos.ip.rtp.dscp1
Null, 0 to 63
Null
Specify the DSCP of packets. If the value is not null, this parameter will overrule the other qos.ip.rtp.*
parameters. The default value is Null, so the other qos.ip.rtp.* parameters will be used.
qos.ip.rtp.max_reliability1
qos.ip.rtp.max_throughput1
qos.ip.rtp.min_cost1
qos.ip.rtp.min_delay1
qos.ip.rtp.precedence1
0 or 1
0 or 1
0 or 1
0 or 1
0 -7
0
1
0
1
5
Set the bits in the IP ToS field of the IP header used for RTP. Specify whether or not to set the max reliability bit, the
max throughput bit, the min cost bit, the min delay bit, and the precedence bit.
If 0, the bit in the IP ToS field of the IP header is not set. If 1, the bit is set.
1
Change causes handset to restart or reboot.
<qos/>
These parameters control the Quality of Service (QoS) options:
 The 802.1p/Q user_priority field RTP, call control, and other packets
 The “type of service” field RTP and call control packets
Troubleshooting: Trouble with WMM-AC?
Phones deployed prior to software release 4.3 use different defaults for the three
Ethernet parameters listed below. If you have trouble with connectivity when WMMAC is deployed, configure these three parameters with the defaults as shown. You
can find them in the everything.cfg file.
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Table 10-7: Quality of Service (Type-of-Service) Parameters
Parameter
Permitted Values
Default
qos.ip.callControl.dscp1
Null, 0 to 63 or EF or any of AF11,AF12,
AF13,AF21, AF22,AF23, AF31,AF32,
AF33,AF41, AF42,AF43
Null
Specify the DSCP of packets. If the value is not null, this parameter will overrule the other
qos.ip.callControl.* parameters. The default value is Null, so the other qos.ip.callControl.*
parameters will be used if no value is entered.
qos.ethernet.rtp.user_priority
0-7
6
0-7
4
0-7
0
Real-Time Protocol (RTP) packets.
qos.ethernet.callControl.user_priority
User-priority for call control packets.
qos.ethernet.other.user_priority
User-priority for packets that do not have a per-protocol setting.
1
Change causes handset to restart or reboot.
Logging Parameters
Logs formed by these logging parameters are written to the provisioning server. A small subset
of these parameters can be set for a single phone in the Admin menu at Advanced Settings>
Administrative Settings> Logging. (Skype for Business only.)
Caution: Changing the Logging Parameters
Logging parameter changes can impair system operation. Do not change any
logging parameters without prior consultation with Spectralink Technical Support.
The event logging system supports the following classes of events:
Table 10-8: Logging Levels
Logging Level
Interpretation
0
Debug only
1
High detail event class
2
Moderate detail event class
3
Low detail event class
4
Minor error – graceful recovery
5
Major error – will eventually incapacitate the system
6
Fatal error
Each event in the log contains the following fields separated by the | character:

time or time/date stamp

1-5 character component identifier (such as “so”)
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
event class

cumulative log events missed due to excessive CPU load

free form text - the event description
Three formats are available for the event timestamp:
Table 10-9: Event Timestamp Formats
Type
Example
0 - seconds.milliseconds
011511.006 -- 1 hour, 15 minutes, 11.006 seconds since
booting.
1 - absolute time with minute resolution
0210281716 -- 2002 October 28, 17:16
2 - absolute time with seconds resolution
1028171642 -- October 28, 17:16:42
Two types of logging are supported:

<level/> <change/>and<render/>

<sched/>
<level/> <change/>and<render/>
Table 10-10: Logging Level, Change, and Render Parameters
Parameter
Permitted Values
Default
log.level.change.xxx
0 to 6
4
Controls the logging detail level for individual modules. These are the input filters into the internal memory-based
log system which produces the [MACaddress]-app.log file.
The most commonly-used values for xxx are listed below. Additional log modules can be found in the log.cfg file.
app1
push
brow
sip
cfg
so
dot1x
tickt
httpd
tls
ice
utilm
key
wlan
pps
wmgr
This parameter and the associated render level (see below) can be set for a single phone in the Admin menu at
Advanced Settings> Administrative Settings> Logging. (Skype for Business only.)
log.render.file
0 or 1
1
Set to 1. Spectralink recommends that you do not change this value.
log.render.file.size
positive integer, 1 to 180
32
Maximum size of flash memory for logs in Kbytes. When this size is about to be exceeded, the handset will upload
all logs that have not yet been uploaded, and erase half of the logs on the handset. The administrator may use
Web browser to read all logs on the handset.
log.render.file.upload.append
0 or 1
1
If set to 1, use append mode when uploading log files to server.
Note: HTTP and TFTP don’t support append mode unless the server is set up for this.
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Parameter
Permitted Values
Default
log.render.file.upload.append.limitMode
delete, stop
delete
Behavior when server log file has reached its limit.
delete=delete file and start over stop=stop appending to file
log.render.file.upload.append.sizeLimit
positive integer
512
Maximum log file size that can be stored on provisioning server in Kbytes.
log.render.file.upload.period
positive integer
86400
Time in seconds between log file uploads to the provisioning server. The default is 86400 which is 24 hours.
Note: The log file will not be uploaded if no new events have been logged since the last upload.
log.render.level
0 to 6
1
Specifies the lowest class of event that will be rendered to the [MACaddress]-app.log files.
The log.render.level maps to severity as follows:
0 -> Debugging
1 -> High Detail
2 -> Moderate Detail
3 -> Low Detail
4 -> Minor Error
5 -> Major Error
6 -> Fatal Error
log.render.realtime
0 or 1
1
Set to 1. Spectralink recommends that you do not change this value.
Debuggiung) d32
0 or 1
0
Set to 1. Spectralink recommends that you do not change this value. Note that on Spectralink handsets, the
default value is 0.
log.render.type
0 to 2
2
Refer to Table 10-9: Event Timestamp Formats for timestamp type.
<sched/>
The handset can be configured to schedule certain advanced logging tasks on a periodic basis.
These parameters should be set in consultation with Spectralink Technical Support. Each
scheduled log task is controlled by a unique parameter set starting with log.sched.x where x
identifies the task. A maximum of 10 schedule logs is allowed.
Table 10-11: Logging Schedule Parameters
Parameter
Permitted Values
log.sched.x.level
0 to 5, default 3
Event class to assign to the log events generated by this command. This needs to be the same or higher than
log.level.change.slog for these events to display in the log.
log.sched.x.name
alphanumeric string
Name of an internal system command to be periodically executed. To be supplied by Spectralink.
log.sched.x.period
positive integer, default 15
Seconds between each command execution. 0=run once
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Parameter
Permitted Values
log.sched.x.startDay
0 to 7
When startMode is abs, specifies the day of the week to start command execution. 1=Sun, 2=Mon, ..., 7=Sat
log.sched.x.startMode
abs, rel
Start at an absolute time or relative to boot.
log.sched.x.startTime
positive integer OR hh:mm
Seconds since boot when startMode is rel or the start time in 24-hour clock format when startMode is abs.
Microsoft Skype for Business Integration
See the Microsoft Interoperability Guides for Microsoft Skype for Business or Microsoft Lync
Server 2013 or 2010. for complete information.
Network Address Translation (NAT)
The handset can work with certain types of network address translation (NAT). NAT enables a
local area network (LAN) to use one set of IP addresses for internal traffic and another set for
external traffic. The handset’s signaling and Real-Time Transport Protocol (RTP) traffic use
symmetric ports. You can configure the external IP address and ports used by the NAT on the
handset’s behalf on a per-handset basis. Note that the source port in transmitted packets is the
same as the associated listening port used to receive packets.
These parameters define port and IP address changes used in NAT traversal. The port changes
will change the port used by the handset, while the IP entry simply changes the IP advertised in
the SIP signaling. This allows the use of simple NAT devices that can redirect traffic, but does
not allow for port mapping. For example, port 5432 on the NAT device can be sent to port 5432
on an internal device, but not to port 1234.
Table 10-12: Network Access Translation
Parameter
Permitted Values
Default
nat.ip1
dotted- decimal IP address
Null
IP address to advertise within SIP signaling - should match the external IP address used by the NAT device.
nat.keepalive.interval
0 to 3600
0
The keep-alive interval in seconds. Sets the interval for handsets to send a keep-alive packet to the gateway/NAT
device to keep the communication port open so that NAT can continue to function. If Null or 0, the handset will not
send out keep-alive messages.
The Microsoft Live Communications Server 2005 keep-alive feature will overrule this parameter. If you want to
deploy handsets behind a NAT and connect them to Live Communications Server, the keep-alive interval received
from the Live Communications Server must be short enough to keep the NAT port open. Once the TCP connection
is closed, the handsets stop sending keep-alive packets.
nat.mediaPortStart1
0 to 65440
0
The initially allocated RTP port. Overrules the value set for tcIpApp.port.rtp.mediaPortRangeStart.
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Parameter
Permitted Values
Default
nat.signalPort1
1024 to 65535
0
The port used for SIP signaling. Overrules voIpProt.local.port.
1
Change causes handset to restart or reboot.
Example Network Address Translation Configuration
The parameter nat.ip is the public IP that you want to advertise in SIP signaling. The IP is
120.242.6.155.
The parameter nat.mediaPortStart is the RTP used to send media. If non-Null, this
attribute will set the initially allocated RTP port and will overrule the value set in
tcpIpApp.port.rtp.mediaPortRangeStart. In the example below, the starting port is
12500 and the handset will cycle through start-port +.
The parameter nat.signalPort specifies the port that the handset will use for SIP signaling.
This parameter will overrule voIpProt.local.Port. In the example below, the handset will
use port 5070 for SIP traffic.
Use t/he nat.keepalive.interval to specify the keepalive interval in seconds. This
parameter sets the interval at which handsets will send a keepalive packet to the gateway/NAT
device. The keepalive packet keeps the communication port open so that NAT can continue to
function as initially set up. In the example below, the handset will send the keepalive every 120
seconds.
Provisioning Server System Settings
This parameter’s settings control aspects of the handset’s provisioning server system.
Table 10-13: Provisioning Parameters
Parameter
Permitted Values
Default
prov.configUploadPath
string
Null
The directory - relative to the provisioning server - where the handset uploads the current configuration file when
the user selects Upload Configuration. If set to Null, use the provisioning server directory.
1
Change causes handset to restart or reboot.
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<request/>
These settings control the handset’s behavior when a request for restart or reconfiguration is
received.
Table 10-14: Configuration Request Parameter
Parameter
Permitted Values
Default
request.delay.type1
audio, call
call
Specify when the handset should process a request for a restart or reconfiguration. If set to audio, the request will
be executed once there is no active audio on the handset — regardless of the call state. If set to call, the request
should be executed once there are no calls —in any state — on the handset.
1
Change causes handset to restart or reboot.
Security <sec/>
These parameters affects the security features of the handset.
Table 10-15: General Security Parameters
Parameter
Permitted Values
Default
sec.tagSerialNo1
0 or 1
0
If 0, the handset does not advertise its serial number (MAC address) through protocol signaling. If 1, the handset
may advertise its serial number through protocol signaling.
1
Change causes handset to restart or reboot.
<srtp/>
As per RFC 3711, you cannot turn off authentication of RTCP.
Table 10-16: SRTP Parameters
Parameter
Permitted values
Defaults
sec.srtp.answerWithNewKey1
0 or 1
1
If 0, a new key is not provided when answering a call. If 1, a new key is provided when answering a call.
sec.srtp.holdWithNewKey1
0 or 1
1
If 0, a new key is not provided when holding a call. If 1, a new key is provided when holding a call.
sec.srtp.key.lifetime1
Null, 0, positive integer
minimum 1024 or
power of 2 notation
Null
The lifetime of the master key used for the cryptographic parameter in SDP. The value specified is the number of
SRTP packets. If 0 or Null, the master key lifetime is not set. If set to a valid value (at least 1024, or a power such
as 2^10), the master key lifetime is set. When the lifetime is set, a re-invite with a new key will be sent when the
number or SRTP packets sent for an outgoing call exceeds half the value of the master key lifetime. Note: Setting
this parameter to a non-zero value may affect the performance of the handset.
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Parameter
Permitted values
Defaults
sec.srtp.mki.enabled1
0 or 1
0
The master key identifier (MKI) is an optional parameter for the cryptographic parameter in the SDP that uniquely
identifies the SRTP stream within an SRTP session. MKI is expressed as a pair of decimal numbers in the form:
|mki:mki_length| where mki is the MKI value and mki_length its length in bytes. If 1, a four-byte MKI
parameter is sent within the SDP message of the SIP INVITE / 200 OK. If 0, the MKI parameter is not sent.
sec.srtp.mki.length1
1 to 4
4
The length of the master key identifier (MKI), in bytes. Microsoft Skype for Business offers 1-byte MKIs.
sec.srtp.mki.startSessionAtOne
0 or 1
0
If set to 1, use an MKI value of 1 at the start of an SDP session. If set to 0, the MKI value will increment for each
new crypto key.
sec.srtp.resumeWithNewKey1
0 or 1
1
If 0, a key is not provided when resuming a call. If 1, a key is provided when resuming a call.
sec.srtp.simplifiedBestEffort
0 or 1
0
If 0, no SRTP is supported. If 1, negotiation of SRTP compliant with Microsoft Session Description Protocol
Version 2.0 Extensions is supported.
1
Change causes handset to restart or reboot.
<dot1x><eapollogoff/>
Table 10-17: 802.1X EAP over LAN (EAPOL) Logoff Parameters
Parameter
Permitted Values
Default
sec.dot1x.eapollogoff.enabled1
0 or 1
0
If 0, the handset will not send an EAPOL Logoff message on behalf of the disconnected supplicant. If 1, the
feature is enabled and the handset will send an EAPOL Logoff message on behalf of the disconnected supplicant
connected to the handset’s secondary (PC) port.
sec.dot1x.eapollogoff.lanlinkreset1
0 or 1
0
If 0, the handset software will not reset (recycle) the LAN port link in the application initiation stage. If 1, the LAN
port link will be reset in the application initiation stage.
1
Change causes handset to restart or reboot.
Secure Real-Time Transport Protocol
Secure Real-Time Transport Protocol (SRTP) provides a way of encrypting audio stream(s) to
avoid interception and eavesdropping on phone calls. As described in RFC 3711, both RTP and
RTCP signaling may be encrypted using an AES (advanced encryption standard) algorithm. The
parameters used to configure SRTP are shown in Table 10-18: Secure Real Time Transport
Protocol. When this feature is enabled, handsets will negotiate with the other end-point the type
of encryption and authentication to use for the session. This negotiation process is compliant
with RFC4568 —Session Description Protocol (SDP) Security Descriptions for Media Streams.
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Web Info: SRTP RFC Resources
For more information on SRTP, see RFC 3711. For the procedure describing how
two handsets set up SRTP for a call, see RFC 4568.
Authentication proves to the handset receiving the RTP/RTCP streams that the packets are
from the expected source and have not been tampered with. Encryption modifies the data in the
RTP/RTCP streams so that, if the data is captured or intercepted, it sounds like noise and
cannot be understood. Only the receiver knows the key to restore the data.
A number of session parameters have been added to enable you to turn off authentication and
encryption for RTP and RTCP streams. This is done mainly to reduce the handset’s processor
usage.
Summary
Parameter
Used to:
sec.srtp.enable
Enable SRTP
sec.srtp.offer
Include secure media in SDP of SIP INVITE
sec.srtp.offer.*
Include crypto in offered SDP
sec.srtp.require
Secure media stream required in all SIP INVITEs
sec.srtp.requireMatchingTag
Check tag in crypto parameter in SDP
sec.srtp.sessionParams.*
Specify if the handset offers and/or requires: RTP encryption, RTP
authentication, and RTCP encryption
Table 10-18: Secure Real Time Transport Protocol
Parameter
Permitted Values
Default
sec.srtp.enable1
0 or 1
1
If 0, the handset always declines SRTP offers. If 1, the handset accepts SRTP offers.
sec.srtp.offer1
0 or 1
0
If 1, the handset includes a secure media stream description along with the usual non-secure media description in
the SDP of a SIP INVITE. This parameter applies to the handset initiating (offering) a phone call. If 0, no secure
media stream is included in SDP of a SIP invite.
sec.srtp.offer.HMAC_SHA1_321
0 or 1
0
If 1, a crypto line with the AES_CM_128_HMAC_SHA1_32 crypto-suite will be included in offered SDP. If 0, the
crypto line is not included.
sec.srtp.offer.HMAC_SHA1_801
0 or 1
1
If 1, a crypto line with the AES_CM_128_HMAC_SHA1_80 crypto-suite will be included in offered SDP. If 0, the
crypto line is not included.
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Parameter
Permitted Values
Default
sec.srtp.require1
0 or 1
0
If 0, secure media streams are not required. If 1, the handset is only allowed to use secure media streams. Any
offered SIP INVITEs must include a secure media description in the SDP or the call will be rejected. For outgoing
calls, only a secure media stream description is included in the SDP of the SIP INVITE, meaning that the nonsecure media description is not included. If this parameter set to 1, sec.srtp.offer will also be set to 1,
regardless of the value in the configuration file.
sec.srtp.requireMatchingTag1
0 or 1
1
If 0, the tag values in the crypto parameter in an SDP answer are ignored. If 1, the tag values must match.
sec.srtp.sessionParams.noAuth.offer1
0 or 1
0
If 0, authentication of RTP is offered. If 1, no authentication of RTP is offered; a session description that includes
the UNAUTHENTICATED_SRTP session parameter is sent when initiating a call.
sec.srtp.sessionParams.noAuth.require1
0 or 1
0
If 0, authentication of RTP is required. If 1, no authentication of RTP is required; a call placed to a handset
configured with this parameter must offer the UNAUTHENTICATED_SRTP session parameter in its SDP. If this
parameter is set to 1, sec.srtp.sessionParams.noAuth.offer will also be set to 1, regardless of the value in
the configuration file.
sec.srtp.sessionParams.noEncrypRTCP.offer1
0 or 1
0
If 0, encryption of RTCP is offered. If 1, no encryption of RTCP is offered; a session description that includes the
UNENCRYPTED_SRTCP session parameter is sent when initiating a call.
sec.srtp.sessionParams.noEncrypRTCP.require1
0 or 1
0
If set to 0, encryption of RTCP is required. If set to 1, no encryption of RTCP is required; a call placed to a handset
configured with noAuth.require must offer the UNENCRYPTED_SRTCP session parameter in its SDP. If this
parameter is set to 1, sec.srtp.sessionParams.noEncryptRTCP.offer will also be set to 1, regardless of
the value in the configuration file.
sec.srtp.sessionParams.noEncrypRTP.offer1
0 or 1
0
If 0, encryption of RTP is offered. If 1, no encryption of RTP is offered; a session description that includes the
UNENCRYPTED_SRTP session parameter is sent when initiating a call.
sec.srtp.sessionParams.noEncrypRTP.require1
0 or 1
0
If 0, encryption of RTP is required. If 1, no encryption of RTP is required. A call placed to a handset configured with
noAuth.require must offer the UNENCRYPTED_SRTP session parameter in its SDP. If set to 1,
sec.srtp.sessionParams.noEncryptRTP.offer will also be set to 1, regardless of the value in the configuration file.
1
Change causes handset to restart or reboot.
In Example 1, the srtp_1.cfg configuration file is shown below:
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This would result in an offer (SIP INVITE with SDP) with 8 crypto attributes with the following
session parameters:
<no session parameters> UNENCRYPTED_SRTCP UNENCRYPTED_SRTP
UNAUTHENTICATED_SRTP
UNAUTHENTICATED_SRTP,UNENCRYPTED_SRTCP
UNENCRYPTED_SRTP,UNENCRYPTED_SRTCP
UNAUTHENTICATED_SRTP,UNENCRYPTED_SRTP
UNAUTHENTICATED_SRTP,UNENCRYPTED_SRTP,UNENCRYPTED_SRTCP
In the above example, the crypto attributes are ordered “most secure” to “least secure” (more
security turned off). The handset receiving this call should chose the most secure crypto it can
support based on the SRTP “require” settings and reply with it in the SDP of a 200 OK SIP
message.
In Example 2, the srtp_2.cfg configuration file is shown below:
This would result in an offer (SIP INVITE with SDP) with 4 crypto attributes with the following
session parameters:
UNENCRYPTED_SRTP UNENCRYPTED_SRTP,UNENCRYPTED_SRTCP
UNAUTHENTICATED_SRTP,UNENCRYPTED_SRTP
UNAUTHENTICATED_SRTP,UNENCRYPTED_SRTP,UNENCRYPTED_SRTCP
In the above example, every crypto includes the UNENCRYPTED_SRTP session parameter
because it is required.
If nothing compatible is offered based on the receiving handset’s STRP “require” settings, then
the call is rejected or dropped.
Server Redundancy
Server redundancy is often required in VoIP deployments to ensure continuity of phone service
if, for example, the call server needs to be taken offline for maintenance, the server fails, or the
connection between the handset and the primary server fails.
Terminology
Before you read this section, take a moment to familiarize yourself with the following definitions:
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Primary server The primary server is the highest priority server in a group of servers with an
active registration. All communications route to the primary server first, unless the handset
environment is configured otherwise.
Secondary server A secondary server backs up a primary server when the primary server
fails. A secondary server may offer the same, or lesser, functionality than the primary server.
Server redundancy This refers to the practice of employing multiple servers so that when a
primary server fails, a secondary server can take over.
Failover A type of server redundancy in which a secondary server takes over all the functions
of the primary server when the primary server fails. No handset functionality is lost when the
secondary server takes over.
Fallback In this mode, a second call server of lesser capability (such as a router or gateway
device) takes over call control to provide basic calling capability without some of the richer
features offered by the primary call server (for example, voicemail, presence, and Message
Waiting Indicator). Spectralink 84-Series handsets support configuration of multiple servers per
SIP registration for this purpose.
Re-registration on failover A redundancy requirement in which a handset must successfully
register with a server before communications can take place. If a server fails and a handset
must communicate with another server (for example, a secondary server), the handset must
register with the secondary server before communications can take place.
Failback A type of server redundancy in which a secondary server remains operational while
communications with a primary server are retried to see if the primary server is functioning
again. In certain configurations, the handset attempts to re-register with the primary server
during failback.
Register transaction A register transaction associates a handset with a particular location,
such as an IP address. A handset sends a message—called a ‘REGISTER’ message—
informing the server of its location.
Registrar or Registrar server A registrar server accepts registrations, or location
information, from handsets and places this information in a database. Every handset must
register its current location with a Registrar server before the handset can communicate with a
server.
About the Optional Failover Behaviors
Spectralink handsets rely on two server redundancy technologies: failover and fallback. Using
these technologies, multiple servers are set up so that when the primary server fails, a
secondary server can take over. In some cases, a combination of the two may be deployed.
Consult your SIP server provider for recommended methods of configuring handsets and
servers for failover configuration.
Re-register on failover only applies to servers that use the failover method.
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With failover, all servers (primary and secondary) share the same registration data. In this
scenario, secondary servers support all the features that the primary server supports.
With re-register on failover disabled—the default behavior—when a handset’s registration
request is diverted to a secondary server, the handset doesn’t have to register with the
secondary server.
A potential issue with the default failover behavior is that some servers or intermediate SIPaware devices may limit a handset’s functionality if the server hasn’t successfully processed the
handset’s registration request. With re-register on failover enabled, when a handset’s SIP
request is diverted to a secondary server, the handset will first register with the secondary
server.
Optional behaviors enhance redundancy features. These behaviors include re-registration and
recovery behaviors, as well as a behavior that controls how existing calls—calls that are
established before a server fails—are treated.

Re-registration behavior The handset must complete a new registration with the failover
server before communications can take place between the handset and the failover server.

Recovery behavior This behavior requires handsets to communicate with the server that
processed the last successful transaction, rather than always with the primary server. If this
behavior is configured, you must set up rules to determine when the primary server is tried
again (for example, whenever the handset has a new request, or after a specific period of
time). The secondary server will remain operational while the handset is trying to re-register
with the primary server (‘failback’).

Behavior for existing calls This behavior controls the handling of calls established
through the failed server after failover occurs. When this behavior is enabled, handsets
won’t communicate with failed servers that recover until failback succeeds. This helps avoid
situations in which large numbers of handsets toggle rapidly between servers when there is
an intermittent failure.
The following diagrams show how a network uses the re-registration on failover behavior. In the
diagrams, primary and secondary re-registration on failover-aware Session Border Controllers
(RRoFO-aware SBCs) are set up so that if a handset can’t communicate with the primary
RRoFO-aware SBC, the handset can attempt to register, and then communicate, with a
secondary RRoFO-aware SBC.
In the following diagram, handsets are communicating with a primary RRoFO-aware SBC that is
just about to fail.
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When the primary RRoFO-aware SBC fails, handsets can no longer communicate with it. The
handsets will attempt to register with a secondary RRoFO-aware SBC, as shown next.
If registration is successful (as shown next), handsets can communicate with the secondary
RRoFO-aware SBC, and traffic flow will continue without interruption.
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Enabling Proxy Servers
If your implementation uses proxy servers, make sure you enable the use of proxy
servers in your configuration file(s). If you’ve defined reg parameters, set
reg.x.server.y.useOutboundProxy to 1.
Note: Compatibility with Microsoft Skype for Business
The concurrent failover/fallback feature is not compatible with Microsoft Skype for
Business.
Fallback Deployments
Fallback deployments are most common in satellite office situations where the primary server is
in a distant location and the WAN connection may fail. The basic configuration uses a
reg.1.server.1.address and reg.1.server.2 address and the handset registers to both servers.
When a handset makes a call the invites are sent to server.1. When server.1 is unavailable, the
calling handset detects this condition and sends the invite to server.2. The handset continues to
attempt to register to server.1 with every call. When server.1 comes back online, invites are
once again routed to it.
In situations where server redundancy for fallback purpose is used, the following measures
should be taken to optimize the solution:

Deploy an on-site DNS server to avoid long call initiation delays that can result if the
DNS server records expire.

Avoid using too many servers as part of the redundancy configuration as each
registration will generate more traffic.

Educate users as to the features that will not be available when in fallback operating
mode.
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Failover Deployments
Failover deployments rely on a SRV (Service Record) built in the DNS server. Phones then use
the domain name as the reg.1.server.1 address, and use the NAPTR (Name Pointer) method
(the default setting in the 84-Series) for querying the DNS server. The domain contains the SRV
records which provide a list of potential hosts and connection protocols (TCP, UDP, or TLS),
ordered in the sequence you wish clients to attempt to connect.
Most configuration is done in the DNS server, not in the handset. The basic configuration uses
just one registration (reg.1).
DNS Server Unavailability
Failover redundancy can only be utilized when the configured IP server hostname resolves
(through SRV or A record) to multiple IP addresses. Some configurations are not able to take
advantage of failover redundance due to non-availability of the DNS server. In these situations,
the handsets support the ability to statically configure a set of DNS NAPTR SRV and/or A
records into the handset. See Static DNS Cache in the Special Use Cases Chapter.
Redundancy Parameters
Summary
Parameter
Used to:
reg.x.server.y.failOver.*
Specify server redundancy options including failback mode, failback timeout,
and failover registration behaviour
reg.x.auth.optimizedInFailover
Specify which server to contact if failover occurs
reg.x.outboundProxy.failOver.*
Overrule the default server redundancy options for a specific registration
The following table shows the parameters you need to define.
Table 10-19: Basic FailOver Configuration File Settings
Parameter
Permitted Values
Default
reg.x.server.y.failOver.reRegisterOn
If you use proxy servers, set this parameter instead:
reg.x.outboundProxy.failOver.reRegisterOn
0 or 1
0
When set to 1, the handset will attempt to register with (or via, for the outbound proxy scenario), the secondary
server. If the registration succeeds (a 200 OK response with valid expires), signaling will proceed with the secondary
server.
When set to 0, the handset won’t attempt to register with the secondary server, since the handset will assume that
the primary and secondary servers share registration information.
Note: When this parameter is enabled, the authOptimizedInFailover parameter is automatically enabled.
reg.x.server.y.failOver.failRegistrationOn
If you use proxy servers, set this parameter instead:
reg.x.outboundProxy.failOver.failRegistrationOn
0 or 1
0
When set to 1, and the reRegisterOn parameter is enabled, the handset will silently invalidate an existing
registration (if it exists), at the point of failing over.
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Parameter
Permitted Values
Default
When set to 0, and the reRegisterOn parameter is enabled, existing registrations will remain active. This means
that the handset will attempt failback without first attempting to register with the primary server to determine if it
has recovered.
reg.x.server.y.failOver.onlySignalWithRegistered
If you use proxy servers, set this parameter instead:
reg.x.outboundProxy.failOver.onlySignalWithRegistered
0 or 1
1
When set to 1, and the reRegisterOn and failRegistrationOn parameters are enabled, no signaling is accepted
from or sent to a server that has failed until failback is attempted or failover occurs. If the handset attempts to send
signaling associated with an existing call via an unregistered server (for example, to resume or hold a call), the call
will end. No SIP messages will be sent to the unregistered server.
When set to 0, and the reRegisterOn and failRegistrationOn parameters are enabled, signaling will be accepted
from and sent to a server that has failed (even though failback hasn’t been attempted or failover hasn’t occurred).
Note: This setting primarily affects signaling associated with existing dialogs that are RFC-mandated to
communicate with the servers through which they were established. A new dialog’s signaling will be sent through
the ‘current’ server.
reg.x.server.y.failOver.failBack.mode
If you use proxy servers, set this parameter instead:
reg.x.outboundProxy.failOver.failBack.mode
DNSTTL,
registration,
duration
registration
When set to DNSTTL, the primary server is retried after a timeout equal to the DNSTTL configured for the server
the handset is registered to (or via, for the outbound proxy scenario).
When set to registration, the primary server is retried when the current working server’s registration requires
renewal.
When set to duration, the primary server is retried after the amount of time defined by the timeout parameter (as
shown in the next row).
Note: When failback mode is set to DNSTTL or duration, re-registration with the primary server takes place only if
the handset is idle (that is, the handset has no calls or active lines). If the timeout period expires and call activity is
detected, failback will be retried every second.
reg.x.server.y.failOver.failBack.timeout
If you use proxy servers, set this parameter instead:
reg.x.outboundProxy.failOver.failBack.timeout
0, 60 - 65535
3600
When failBack.mode is set to duration, the time in seconds after failing over to the current working server before
the primary server becomes the first server to forward new requests to.
If you set a value between 1 and 59, the timeout will be 60 seconds. If you set a value of 0, the primary server
won’t be selected as the first server to forward new requests to until a failover event occurs with the current
working server.
Table 10-20: All possible reg.x parameters used to set up server redundancy
Parameter
Permitted Values
Default
reg.x.auth.optimizedInFailover
0 or 1
0
The destination of the first new SIP request when failover occurs. If 0, the SIP request is sent to the server with the
highest priority in the server list. If 1, the SIP request is sent to the server which sent the proxy authentication
request.
reg.x.outboundProxy.failOver.failBack.mode
DNSTTL, registration,
duration
registration
The mode for failover failback (overrules reg.x.server.y.failOver.failBack.mode).
DNSTTL: the handset tries the primary server again after a timeout equal to the DNS TTL configured for the
server that the handset is registered to.
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Parameter
Permitted Values
Default
registration: the handset tries the primary server again when the registration renewal signaling begins.
duration: the handset tries the primary server again after the time specified by
reg.x.outboundProxy.failOver.failBack.timeout expires.
reg.x.outboundProxy.failOver.failBack.timeout
0, 60 to 65535
3600
The time to wait (in seconds) before failback occurs (overrules
reg.x.server.y.failOver.failBack.timeout).If the fail back mode is set to Duration, the handset waits
this long after connecting to the current working server before selecting the primary server again. If 0, the handset
will not fail-back until a fail-over event occurs with the current server.
reg.x.outboundProxy.failOver.failRegistrationOn
0 or 1
0
When set to 1, and the reRegisterOn parameter is enabled, the handset will silently invalidate an existing
registration (if it exists), at the point of failing over. When set to 0, and the reRegisterOn parameter is enabled,
existing registrations will remain active. This means that the handset will attempt failback without first attempting to
register with the primary server to determine if it has recovered.
Note that reg.x.outboundProxy.failOver.RegisterOn must be enabled.
reg.x.outboundProxy.failOver.onlySignalWithRegistered
0 or 1
1
When set to 1, and the reRegisterOn and failRegistrationOn parameters are enabled, no signaling is accepted from
or sent to a server that has failed until failback is attempted or failover occurs. If the handset attempts to send
signaling associated with an existing call via an unregistered server (for example, to resume or hold a call), the call
will end. No SIP messages will be sent to the unregistered server. When set to 0, and the reRegisterOn and
failRegistrationOn parameters are enabled, signaling will be accepted from and sent to a server that has failed
(even though failback hasn’t been attempted or failover hasn’t occurred).
reg.x.outboundProxy.failOver.reRegisterOn
0 or 1
0
This parameters overrules reg.x.server.y.failOver.failBack.RegisterOn. When set to 1, the handset
will attempt to register with (or via, for the outbound proxy scenario), the secondary server. If the registration
succeeds (a 200 OK response with valid expires), signaling will proceed with the secondary server. When set to 0,
the handset won’t attempt to register with the secondary server, since the handset will assume that the primary and
secondary servers share registration information.
reg.x.server.y.failOver.failBack.mode
DNSTTL, registration,
duration
registration
The mode for failover failback (this parameter overrules voIpProt.server.x.failOver.failBack.mode):
DNSTTL – the handset tries the primary server again after a timeout equal to the DNS TTL configured for the
server that the handset is registered to.
registration – the handset tries the primary server again when the registration renewal signaling begins.
duration – the handset tries the primary server again after the time specified by
reg.x.server.y.failOver.failBack.timeout.
reg.x.server.y.failOver.failBack.timeout
0, 60 to 65535
3600
The time to wait (in seconds) before failback occurs (overrules
voIpProt.server.x.failOver.failBack.timeout).If the fail back mode is set to Duration, the handset
waits this long after connecting to the current working server before selecting the primary server again. If 0, the
handset will not fail-back until a fail-over event occurs with the current server.
reg.x.server.y.failOver.failRegistrationOn
0 or 1
0
When set to 1, and the reRegisterOn parameter is enabled, the handset will silently invalidate an existing
registration (if it exists), at the point of failing over. When set to 0, and the reRegisterOn parameter is enabled,
existing registrations will remain active. This means that the handset will attempt failback without first attempting to
register with the primary server to determine if it has recovered.
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Parameter
Permitted Values
Default
reg.x.server.y.failOver.onlySignalWithRegistered
0 or 1
1
When set to 1, and the reRegisterOn and failRegistrationOn parameters are enabled, no signaling is accepted from
or sent to a server that has failed until failback is attempted or failover occurs. If the handset attempts to send
signaling associated with an existing call via an unregistered server (for example, to resume or hold a call), the call
will end. No SIP messages will be sent to the unregistered server. When set to 0, and the reRegisterOn and
failRegistrationOn parameters are enabled, signaling will be accepted from and sent to a server that has failed
(even though failback hasn’t been attempted or failover hasn’t occurred).
reg.x.server.y.failOver.reRegisterOn
0 or 1
0
This parameter overrules the voIpProt.server.x.failOver.reRegisterOn. When set to 1, the handset will
attempt to register with (or via, for the outbound proxy scenario), the secondary server. If the registration succeeds
(a 200 OK response with valid expires), signaling will proceed with the secondary server. When set to 0, the
handset won’t attempt to register with the secondary server, since the handset will assume that the primary and
secondary servers share registration information.
reg.x.tcpFastFailover
0 or 1
0
If 1, failover occurs based on the values of reg.x.server.y.retryMaxCount or
voIpProt.server.x.retryTimeOut.
Supporting 802.1X Authentication
IEEE 802.1X is a port-based Network Access Control (PNAC). It provides an authentication
mechanism to devices trying to attach to a local area network (LAN) or a wireless local area
network (WLAN). IEEE 802.1X is based on the Extensible Authentication Protocol (EAP).
Spectralink handsets support standard IEEE 802.1X authentication. Figure 10-1 shows the
RADIUS server using the AP as the authenticator for the Spectralink 84-Series handsets.
Figure 10-1: A Typical 802.1X Network Configuration
Spectralink handsets support the following EAP authentication methods:
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 EAP-PEAPv0/MSCHAPv2 (requires CA certificates)
 EAP-FAST (optional Protected Access Credential (PAC) file, if not using in-band
provisioning)
 EAP-TLS (requires CA certificates)
To set up an EAP method that requires a Device or CA certificate, you need to configure TLS
Platform Profile 1 or TLS Platform Profile 2 to use with 802.1X. You can use the parameters in
Table 10-21: Supporting 802.1X Authentication to configure 802.1X Authentication.
For more information about certificates see Downloading Certificates to a Spectralink Phone.
Web Info: EAP Authentication Protocol
For more information, see RFC 3748, Extensible Authentication Protocol.
Summary
Parameter
Used to:
device.net.dot1x.identity
Specify the identity (username) for authentication
device.net.dot1x.password
Specify the password for authentication
device.net.dot1x.eapFastInBandProv
To enable EAP In-Band Provisioning for EAP-FAST
device.pacfile.data
Specify a PAC file for EAP-FAST (optional)
device.pacfile.password
Specify the optional password for the EAP-FAST PAC file
Table 10-21: Supporting 802.1X Authentication
Parameter
Permitted Values
Default
device.net.dot1x.anonid1
string
Null
EAP-TTLS and EAP-FAST only. The anonymous identity (user name) for 802.1X authentication.
device.net.dot1x.eapFastInBandProv1
0 or 1
0
EAP-FAST only, optional. Choose 1 to enable EAP In-Band Provisioning by server unauthenticated PAC
provisioning using anonymous Diffie-Hellman key exchange. Choose 0 to disable EAP In-Band Provisioning.
Reserved for Future Use – Choose 2 to enable EAP In-band provisioning by server authenticated PAC provisioning
using certificate based server authentication.
device.net.dot1x.identity1
string
Null
The identity (user name) for 802.1X authentication.
Specify the 802.1X authentication method, where EAP-NONE means no authentication.
device.net.dot1x.password1
string
Null
The password for 802.1X authentication. This parameter is required for all methods except EAP-TLS.
device.pacfile.data1
String
Null
EAP-FAST only, optional. The PAC file (base 64 encoded). To generate a base 64-encoded PAC file, generate the
PAC file using your authentication server and then convert it to base 64. You can convert the file to base 64 using
the following openssl commands:
$ openssl enc -base64 -in myfile -out myfile.b64
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Parameter
Permitted Values
Default
device.pacfile.password1
String
Null
EAP-FAST only, optional. The password for the PAC file.
1
Change causes handset to restart or reboot.
<tcpIpApp/>
This parameter includes:

<dhcp/>

<dns/>

<ice/>

<keepalive/>
<dhcp/>
The DHCP parameters enable you to change how the handset reacts to DHCP changes.
Table 10-22: DHCP Parameters
Parameter
Permitted Values
Default
tcpIpApp.dhcp.releaseOnLinkRecovery
0 or 1
1
If 0, no DHCP release occurs. If 1, a DHCP release is performed after the loss and recovery of the network.
<dns/>
The <dns/> parameters provide a way to set Domain Name System (DNS). However, any
values set through DHCP will have a higher priority and any values set through the <device/>
parameter in a configuration file will have a lower priority.
Table 10-23: Domain Name System (DNS) Parameters
Parameter
Permitted Values
Default
tcpIpApp.dns.server1
Dotted-decimal IP address
Null
The primary server to which the handset directs DNS queries.
tcpIpApp.dns.altServer1
Dotted-decimal IP address
Null
The secondary server to which the handset directs DNS queries.
tcpIpApp.dns.domain1
String
Null
The handset’s DNS domain.
1
Change causes handset to restart or reboot.
<ice/>
The <ice/> parameters enable you to set the STUN/TURN/ICE feature.
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Table 10-24: Ice Parameters
Parameter
Permitted Values
Default
tcpIpApp.ice.mode
Disabled, Standard,
MSOCS
Disabled
Turn SIP ICE negotiation on or off. If using Skype for Business or Microsoft Lync Server 2013 or 2010. , set to
MSOCS to enable ICE.
tcpIpApp.ice.password
String
Null
String
Null
1-65535
3478
0 or 1
1
0 or 1
0
Enter the password to authenticate to the TURN server.
tcpIpApp.ice.stun.server
Enter the IP address of the STUN server.
tcpIpApp.ice.stun.udpPort
The UDP port number of the STUN server.
tcpIpApp.ice.tcp.enabled
If 0, TCP is disabled. If 1, TCP is enabled.
tcpIpApp.ice.turn.callAdmissionControl.enabled
If 0, call admission control is disabled. If 1, call admission control is enabled
tcpIpApp.ice.turn.server
String
Null
1-65535
443
1-65535
443
String
Null
Enter the IP address of the TURN server.
tcpIpApp.ice.turn.tcpPort
The TCP port number of the TURN server.
tcpIpApp.ice.turn.udpPort
The UDP port number of the TURN server.
tcpIpApp.ice.username
Enter the user name to authenticate to the TURN server.
<keepalive/>
These parameters enable the configuration of TCP keep-alive on SIP TLS connections; the
handset can detect a failure quickly (in minutes) and attempt to re-register with the SIP call
server (or its redundant pair).
Table 10-25: TCP Keep-Alive Parameters
Parameter
Permitted Values
Default
tcpIpApp.keepalive.tcp.idleTransmitInterval
10 to 7200
30
The amount of time to wait (in seconds) before sending the keep-alive message to the call server.
Note: If this parameter is set to a value that is out of range, the default value is used.
tcpIpApp.keepalive.tcp.noResponseTransmitInterval
5 to 120
20
If no response is received to a keep-alive message, subsequent keep-alive messages are sent to the call server at
this interval (every x seconds).
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Parameter
Permitted Values
Default
tcpIpApp.keepalive.tcp.sip.tls.enable
0 or 1
0
If 0, disable TCP keep-alive for SIP signaling connections that use TLS transport. If 1, enable TCP keep-alive for
SIP signaling connections that use TLS transport.
Tones <tones/>
This parameter describes configuration items for the tone resources available in the handset. It
includes:

<DTMF/>

<chord/>
<chord/>
Chord-sets are the building blocks of sound effects that use synthesized audio rather than
sampled audio. Most call progress and ringer sound effects are synthesized. A chord-set is a
multi-frequency note with an optional on/off cadence. A chord-set can contain up to four
frequency components generated simultaneously, each with its own level.
There are three chord sets: callProg, misc, and ringer. Each chord set has different chord
names, represented by x in the following table. The chord names are as follows:
For callProg, x can be one of the following chords:

dialTone, busyTone, ringback, reorder, stutter_3, callWaiting, callWaitingLong,
howler, recWarning, stutterLong, intercom, callWaitingLong,
precedenceCallWaiting, preemption, precedenceRingback, or spare1 to spare6.
For misc, x can be one of the following chords

spare1 to spare9.
For ringer, x can be one of the following chords:

ringback, originalLow, originalHigh, or spare1 to spare19.
Table 10-26: Chord Parameters
Parameter
Permitted Values
tone.chord.callProg.x.freq.y
tone.chord.misc.x.freq.y
tone.chord.ringer.x.freq.y
0-1600
0-1600
0-1600
The frequency (in Hertz) for component y. Up to six chord-set components can be specified (y=1 to 6).
tone.chord.callProg.x.level.y
tone.chord.misc.x.level.y
tone.chord.ringer.x.level.y
-57 to 3
-57 to 3
-57 to 3
The level of component y in dBm0. Up to six chord-set components can be specified (y=1 to 6).
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Parameter
Permitted Values
tone.chord.callProg.x.onDur
tone.chord.misc.x.onDur
tone.chord.ringer.x.onDur
positive integer
positive integer
positive integer
The on duration (length of time to play each component) in milliseconds, 0=infinite.
tone.chord.callProg.x.offDur
tone.chord.misc.x.offDur
tone.chord.ringer.x.offDur
positive integer
positive integer
positive integer
The off duration (the length of silence between each chord component) in milliseconds, 0=infinite.
tone.chord.callProg.x.repeat
tone.chord.misc.x.repeat
tone.chord.ringer.x.repeat
positive integer
positive integer
positive integer
The number of times each ON/OFF cadence is repeated, 0=infinite.
Web Configuration Utility
<httpd/>
The handset contains a local Web Configuration Utility server for user and administrator
features. This can be disabled for applications where it is not needed or where it poses a
security threat. The Web server supports both basic and digest authentication. The
authentication user name and password are not configurable for this release.
Table 10-27: HTTPD (Web Server) Parameters
Parameter
Permitted Values
Default
httpd.enabled1
0 or 1
1
If 0, the HTTP server is disabled (the Web Configuration Utility will also be disabled). If 1, the server will be
enabled.
httpd.cfg.enabled1
0 or 1
1
If 0, the Web Configuration Utility is disabled. If 1, the Web Configuration Utility is enabled.
httpd.cfg.port1
1 to 65535
80
Port is 80 for HTTP servers. Care should be taken when choosing an alternate port.
httpd.cfg.secureTunnelEnabled1
0 or 1
1
If 0, the Web does not use a secure tunnel. If 1, the server connects through a secure tunnel.
httpd.cfg.secureTunnelPort1
1 to 65535
443
The port to use for communications when the secure tunnel is used.
httpd.cfg.secureTunnelRequired1
0 or 1
0
If 0, communications to the Web server do not require a secure tunnel. If 1, communications do require a secure
tunnel.
1
Change causes handset to restart or reboot.
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Chapter 11: Special Use Cases
Certain situations require special handling and more involved provisioning. This chapter
addresses these special use cases and the parameters that may be needed for a successful
deployment.
Acoustic Echo Cancellation
Your Spectralink handset uses advanced acoustic echo cancellation (AEC) for handsfree
operation using the speakerphone. The handset also supports headset echo cancellation. The
handsets use both linear and non-linear techniques to aggressively reduce echo while
permitting natural, full-duplex communication patterns.
Caution: Contact Spectralink Support Before Modifying Acoustic Echo
Cancellation Parameters
Consult Spectralink customer support before you make changes to any acoustic
echo cancellation parameters.
Audio Codecs
The following table details the audio codec support and priority for Spectralink handsets:
Table 11-1: Audio Codec Priority
Supported Audio Codecs
Priority
G.722
4
5
G.711m-law
6
G.711a-law
7
G.729a, G.729b
8
The following table summarizes the audio codecs supported on Spectralink handsets:
Table 11-2: Audio Codec Specifications
Algorithm
Reference
Raw Bit
Rate
IP Bit Rate
Sample
Rate
Default
Payload
Size
Effective
Audio
Bandwidth
G.711 u-law
RFC 1890
64 Kbps
80 Kbps
8 Ksps
20 ms
3.5 KHz
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Algorithm
Reference
Raw Bit
Rate
IP Bit Rate
Sample
Rate
Default
Payload
Size
Effective
Audio
Bandwidth
G.711 a-law
RFC 1890
64 Kbps
80 Kbps
8 Ksps
20 ms
3.5 KHz
G.711
RFC 1890
64 Kbps
80 Kbps
16 Ksps
20 ms
7 KHz
G.722.1
RFC 3047
16 Kbps
24 Kbps
32 Kbps
32 Kbps
40 Kbps
48 Kbps
16 Ksps
20 ms
7 KHz
G.729a
RFC 2833
8 Kbps
24 Kbps
8 Ksps
10 ms
8 KHz
Caution: New Zealand requirements
Re: The Optimum Packet Size for transmission through the Public Switched
Telephone Network (PSTN)
Because of the extensive delay already experienced when calling cellular and
international networks, Telecom Access Standards recommends the use of either
10 or 20mS packet length when passing packets through the PSTN. The use of
G.711 codecs and 10 or 20mS packet length is critical to maintaining delay times
which comply with PTC220 requirements for Customer Equipment (<50mS).
Note: Network Bandwidth Requirements for Encoded Voice
The network bandwidth necessary to send the encoded voice is typically 5-10%
higher than the encoded bit rate due to packetization overhead.
Use the parameters in the following table to specify the priority for audio codecs on your
Spectralink handsets.
Table 11-3: Audio Codec Priorities
Parameter
Permitted Values
Default
voice.codecPref.G711_A
voice.codecPref.G711_Mu
voice.codecPref.G722
voice.codecPref.G7221.16kbps
voice.codecPref.G7221.24kbps
voice.codecPref.G7221.32kbps
voice.codecPref.G729_AB
0 to 27
7
6
4
0
0
5
8
The priority of the codec. If 0 or Null, the codec is disabled. A value of 1 is the highest priority. If a handset does
not support a codec, it will treat the setting as if it were 0 and not offer or accept calls with that codec.
Band Steering
Band steering is a feature that allows you to configure the handsets to use a preferred band
during roaming.
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If this preference is not configured, the 84-Series handset performs inter-band roaming between
the 2.4GHz and 5GHz bands if both bands are enabled without any preference to one band
over the other. The band of the access point with the strongest signal strength, as measured by
the handset, is used.
Band steering is the use of a preferred band when selecting an AP. The handset uses the
preferred band as long as telephone performance is not degraded by staying on the preferred
band instead of using the non-preferred band.
Table 11-4: Band steering parameters
Parameter
Permitted Values
Default
preferredBandRoaming.band
noPreference
prefer2_4GHz
prefer5GHz
noPreference
The configuration parameter, preferredBandRoaming.band, selects the preferred inter-band roaming mode.
Allowable values are noPreference (the default value), prefer2_4GHz, or prefer5GHz. For inter-band
roaming to function, both 2.4GHz and 5GHz bands must be enabled using existing device.wifi.radio
parameters.
preferredBandRoaming.threshold
-65 to -40 db
-65
The 84 Series handset makes band selections based on signal strength for three different conditions; strong signal
strength, moderate signal strength, and low signal strength.
If the measured signal strength on a channel in the preferred band is above
preferredBandRoaming.threshold + 10db, the handset stays on the preferred band regardless of how much
better the signal strength on the non-preferred band may be. There is no degradation in performance as long as
the signal strength is strong so there is no reason to roam to the non-preferred band. The 84 Series handset may
still roam between APs on the preferred band. The threshold is set by preferredBandRoaming.threshold
which may be configured for RSSI signal strengths from -65db to -40db.
While the default is -65db, Spectralink recommends that users experiment with values between -65db and -55db.
The +10db is added the threshold to prevent it from ping-ponging between two APs.
On the other end of the signal strength range is the point where signal strength is so low the wireless performance
is impacted. Below this threshold, the handset uses the access point with the best signal strength regardless of
which band is available (i.e. uses the rules that existed prior to adding the band steering feature). Any user
configured preference is ignored. This threshold is -75db and is not configurable.
preferredBandRoaming.bias
0 to 10 db
0
The lower end of the signal strength range is the point where signal strength is so low the wireless performance is
impacted. Below this threshold, the handset uses the access point with the best signal strength regardless of
which band is available. Any user configured preference is ignored. This threshold is -75db and is not configurable.
When the signal strength on the preferred band is moderate, between -75db and
preferredBandRoaming.threshold + 10db, the 84 Series handset MAY use the preferred band. The handset
uses the non-preferred band if the signal strength is significantly better than the preferred band. If the signal
strength on the non-preferred band is only slightly better than on the preferred band, the 84 Series handset uses
the preferred band.
The handset software has 10db built into the roaming algorithm to prevent the telephone from constantly switching
between two APs with similar signal strength. The handset will switch from a non-preferred band AP to a preferred
band AP if the preferred band AP signal strength is greater than the signal strength of the non-preferred band AP.
But to switch back to the non-preferred band AP, the non-preferred band AP must be at least 10db better than the
preferred band AP. This amount can be increased up to 20db by using the bias parameter, thus strengthening the
preferred band criteria. The preferredBandRoaming.bias is added to the preferred band signal strength
before comparing it to the non-preferred band AP.
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As an example, imagine the following scenario:

The threshold is set to -60

The bias is set to 5

The band is set to prefer5GHz (a-band)
In this case, the following would be true:

If there is a channel on the a-band that is at -50 or greater then the phone will always
use the a-band and will never roam to the b-band. The value of -50 is the threshold + 10
or -60 + 10 = -50.

If the phone is on an AP on the a-band and the RSSI is less than -75, the phone will use
the old rules to roam to a new AP. If it finds a new a band AP with sufficient signal
strength it will use that AP, but may also roam to the b-band. At this point, it is just
looking for the best AP it can find.

If the phone is on the a-band and the candidate to roam to is also on the a-band, then
normal roaming rules will apply.

If the phone is on the a-band and the candidate to roam to is on the b-band, then the
phone shall add the bias value into the hysteresis rules. The higher the bias the more it
will try to stay on the preferred band (the a-band in this case) even if there are APs on
the b-band that have better signal strength.
No other changes are made to the 84-Series handset roaming behavior. Setting
preferredBandRoaming.band to noPreference results in the same roaming behavior as in
previous versions of software. The 84-Series handset still does not actively roam in standby (to
preserve battery life) but will use the inter-band roaming rules if the handset loses the network
and must reacquire the network.
Bridged Line Appearance
Bridged line appearance connects calls and lines to multiple handsets. With bridged line
appearance enabled, an active call displays simultaneously on multiple handsets in a group. By
default, the answering handset has sole access to the incoming call—line seize. If the
answering handset places the call on hold, that call becomes available to all handsets of that
group. All call states—active, inactive, on hold—are displayed on all handsets of a group.
Tip: Bridged Line and Shared Call Appearances are Distinct
Shared call appearances and bridged line appearances are similar signaling
methods that enable more than one handset to share the same line or registration.
The methods you use vary with the SIP call server you are using. In the
configuration files, bridged lines are configured by ‘shared line’ parameters. The
barge-in feature is not available with bridged line appearances; it is available with
shared call appearances.
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Summary
Parameter
Used to:
call.shared.disableDivert
Specify whether call diversion should be disabled by default on all shared lines
reg.x.type
Specify the per-registration line type (private or shared)
reg.x.thirdPartyName
Specify the shared line third-party name.
divert.x.sharedDisabled
Specify whether call diversion should be disabled on a specific shared line
(overrules default)
Table 11-5: Enabling Bridged Line Appearance
Parameter
Permitted Values
Default
call.shared.disableDivert1
0 or 1
1
If set to 1, the diversion feature for shared lines is disabled. Note: This feature is disabled on most call servers.
reg.x.type
private or shared
private
If set to private, use standard call signaling. If set to shared, augment call signaling with call state subscriptions and
notifications and use access control for outgoing calls.
reg.x.thirdPartyName
string address
Null
This field must match the reg.x.address value of the registration which makes up the part of a bridged line
appearance (BLA). It must be Null in all other cases.
divert.x.sharedDisabled1
0 or 1
1
If 0, call diversion features can be used on shared lines. If 1, call diversion features are disabled on shared lines.
1
Change causes handset to restart or reboot.
Local Digit Map
The handset has a local digit map feature that, when configured, will automatically call a dialed
number, eliminating the need to press the Dial or Send softkey to place outgoing calls. Note
that digit maps do not apply to on-hook dialing.
Digit maps are defined by a single string or a list of strings. If a number you dial matches any
string of a digit map, the call is automatically placed. If a number you dial matches no string—an
impossible match— you can specify the handset’s behavior. If a number ends with #, you can
specify the handset’s behavior, called trailing # behavior. You can also specify the digit map
timeout, the period of time after you dial a number that the handset will wait before the call will
be placed. The configuration syntax of the digit map is based on recommendations in section
2.1.5 of RFC 3435.
Summary
Parameter
Used to:
dialplan.applyTo*
Apply a dial plan to dialing scenarios
dialplan.digitmap
Specify the digit map to use for the dial plan
dialplan.digitmap.timeOut
Specify the timeout for each segment of the digit map
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Parameter
Used to:
dialplan.impossibleMatchHandling
Specify the behavior if an impossible dial plan match occurs
dialplan.removeEndOfDial
Specify if trailing # digits should be removed from digits sent out
dialplan.routing.emergency.x.*
Specify the details for emergency dial plan routing
dialplan.routing.server.x.*
Specify the server that will be used for routing calls
dialplan.x.*
Configure the same parameters as above for a specific registration (overrules
the global parameters above)
Understanding Digit Map Rules
The following is a list of digit map string rules. If you are using a list of strings, each string in the
list can be specified as a set of digits or timers, or as an expression which the handset will use
to find the shortest possible match.
Digit map extension letter ‘R’ indicates that certain matched strings are replaced. Using a ‘RRR’
syntax, you can replace the digits between the first two ‘R’s with the digits between the last two
‘R’s. For example, R555R604R would replace 555 with 604. Digit map timer letter ‘T’ indicates a
timer expiry. The following examples illustrate the semantics of the syntax:

R9R604Rxxxxxxx—Replaces 9 with 604
For example, a customer dials 91524810 – the digits sent to the PBX will be
6041524810

xxR601R600Rxx—When applied to 1160122 gives 1160022

R9RRxxxxxxx—Remove 9 at the beginning of the dialed number (replace 9 with
nothing)
For example, if a customer dials 914539400, the first 9 is removed when the call is
placed.

RR604Rxxxxxxx—Prepend 604 to all seven digit numbers (replace nothing with 604)
For example, if a customer dials 4539400, 604 is added to the front of the number,
so a call to 6044539400 is placed.

xR60xR600Rxxxxxxx—Replace any 60x with 600 in the middle of the dialed number
that matches
For example, if a customer dials 16092345678, a call is placed to 16002345678.

911xxx.T—A period (".") that matches any arbitrary number, including zero, of
occurrences of the preceding construct
For example:
911123 with waiting time to comply with T is a match
9111234 with waiting time to comply with T is a match
91112345 with waiting time to comply with T is a match
and the number can grow indefinitely given that pressing the next digit takes less than T.
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The following guidelines should be noted:
 The following letters are case sensitive: x, T, and R
 You must use only *, #, +, or 0-9 between the second and third R
 If a digit map does not comply with section 2.1.5 of RFC 3435, it is not included in the digit
plan as a valid map. That is, no match will be made.
 There is no limit to the number of R triplet sets in a digit map. However, a digit map that
contains less than a full number of triplet sets (for example, a total of 2Rs or 5Rs) is
considered an invalid digit map.
For example, R9Rxxxx would be invalid but RR9Rxxxx would be valid.
 If you use T in the left part of ‘RRR’ syntax, the digit map will not work. For example,
R0TR322R will not work. The T can only be applied at the end of the digit string.
The dial plan (or digit map) is not applied against Placed Call List, Voicemail, last call return,
remote control dialed numbers, or on-hook dialing.
This parameter allows the user to create a specific routing path for outgoing SIP calls
independent of other default configurations.
All characters of a digit map must conform to the rules explained in section 2.1.5 of RFC3435.
The following list defines the valid character types and characters usable within a digit map:

Digit:

Timer:

DTMF: A digit, a timer, or one of the symbols "A", "B", "C", "D", "#", or "*". Extensions
may be defined.

Wildcard: The symbol "x" which matches any digit ("0" to "9").

Range:

Subrange: Two digits separated by hyphen ("-") which matches any digit between and
including the two. The subrange construct can only be used inside a range construct,
i.e., between "[" and "]".

Position: A period (".") which matches an arbitrary number, including zero, of
occurrences of the preceding construct.

Separator: An alternation operator or pipe (“|”) is used to separate multiple digit map
strings
A digit from "0" to "9".
The symbol "T" matching a timer expiry.
One or more DTMF symbols enclosed between square brackets ("[" and "]").
Note: Using the alternation operator
When using the alternation operator—the vertical bar (|)—the elements must be
enclosed in parenthesis. To specify A or B or C, the syntax should be (A|B|C).

Miscellaneous: A comma (“,”) is used to return dial tone to the user during dialing
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The Spectralink 84-Series handset utilizes a default digit map as part of the dial plan
configuration. The following explains what each of the default digit mappings will match to:

[2-9]11 – This string will match 211, 311, 411, 511, 611, 711, 811, or 911

0T – This string will match 0 and then wait for the inter-digit timeout to expire (3 seconds
by default)

+011xxx.T – This string will be used for international dialing within the United States and
Canada. It will match 011 followed by any three digits (0-9) and then any additional
number of digits until the inter-digit timeout expires (3 seconds by default)

0[2-9]xxxxxxxxx – This string will be used for collect calling within the United States and
Canada. It will match 0 plus a digit between 2 and 9 followed by nine additional digits (09)

+1[2-9]xxxxxxxx – This string will be used for standard long distance dialing within the
United States and Canada. It will match 1 plus a digit between 2 and 9 followed by eight
additional digits (0-9)

[2-9]xxxxxxxxx – This string will be used for long distance dialing where ten digits are
allowed. This will be dependent on the local PSTN provider as to whether it will be a
valid dial string. Note that this is similar to ten digit dialing with cellular telephones. It will
match a digit between 2 and 9 followed by nine additional digits (0-9)

[2-9]xxxT – This string will be used for local dialing in situations where area codes and
local exchange codes are not required. This string may also match internal extension
based dialing depending on the digits being dialed. It will match a digit between 2 and 9
followed by three additional digits (0-9) until the inter-digit timeout expires (3 seconds by
default)
Please note that the default dial plan may not be sufficient for your particular needs. For
example, many customers require a digit be dialed to access an outside line, such as 9. In this
case they may not be able to dial a sufficient number of digits before the call attempts to
complete.

For example, a customer is attempting to dial 913035551155. With the default digit map
this would match to [2-9]xxxxxxxxx. This would mean that after the customer had dialed
9130355511 the call would attempt to complete. However, this would be an incomplete
dial string so it would fail to connect.
To remedy the issue it is recommended that customers consider their dial plan carefully. It may
be prudent to alter the default digit map as well to a more simplified format. One such example
would be to set the digit map to x.T and remove all other strings. This string would match all
possible dialed digits as it will allow for one digit (0-9) followed by any number of additional
digits (0-9). The T at the end will mean that the user will have 3 seconds, by default, between
each digit before the handset stops collecting digits and attempts to complete the call.
Many customer environments rely on complex dial plans within the PBX to handle call routing
and class of service requirements to prevent undesired dialing by users. Because of this it may
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not be necessary to further restrict the dial plan at the handset level. If this is the case then a
simplified digit map, as described, would be ideal.
Table 10-6: Dial Plan (Digit Map) Parameters
Parameter
Permitted Values
Default
dialplan.applyToCallListDial1
0 or 1
1
If 0, the dial plan does not apply to numbers dialed from the Received Call List or Missed Call List. If 1, the dial
plan is applied to numbers dialed from the received call and missed call lists, including sub-menus.
dialplan.applyToDirectoryDial1
0 or 1
0
If 0, the dial plan is not applied to numbers dialed from the directory or speed dial list. If 1, the dial plan is applied
to numbers dialed from the directory or speed dial, including auto-call contact numbers.
dialplan.applyToForward1
If 0, the dial plan does not apply to forwarded calls. If 1, the dial plan applies to forwarded calls.
dialplan.applyToTelUriDial1
0 or 1
1
If 0, the dial plan does not apply to URI dialing. If 1, the dial plan applies to URI dialing.
dialplan.applyToUserDial1
0 or 1
1
If 0, the dial plan does not apply to calls made when the user presses the Dial softkey to place a call. If 1, the dial
plan applies to calls placed using the Dial softkey.
dialplan.applyToUserSend1
0 or 1
1
If 0, the dial plan does not apply to calls placed when the user presses the Send softkey to place a call. If 1, the
dial plan applies to calls placed using the Send softkey.
dialplan.digitmap1
string compatible with the
digit map feature of MGCP
described in 2.1.5 of RFC
3435
[2-9]11|0T|
+011xxx.T|
0[2-9]xxxxxxxxx|
+1[2-9]xxxxxxxx|
[2-9]xxxxxxxxx|
[2-9]xxxT
The digit map used for the dial plan. The string is limited to 2560 bytes and 100 segments of 64 bytes; a comma is
also allowed; a comma will turn dial tone back on;’+’ is allowed as a valid digit; extension letter ‘R’ is used as
defined above. This parameter enables the handset to automatically initiate calls to numbers that match a digit
map pattern.
dialplan.digitmap.timeOut1
string of positive integers
separated by ‘|’
3 | 3 | 3 | 3 | 3| 3
Specify a timeout in seconds for each segment of digit map. After you press a key, the handset will wait this many
seconds before matching the digits to a dial plan and dialing the call. Note: If there are more digit maps than
timeout values, the default value of 3 will be used. If there are more timeout values than digit maps, the extra
timeout values are ignored.
dialplan.filterNonDigitUriUsers1
0 or 1
0
If 0, do not filter out (+) in the dial plan. If 1, filter out (+) from the dial plan.
dialplan.impossibleMatchHandling1
0, 1 or 2
0
This parameter applies to digits entered in dial mode. Users are in dial mode after going off-hook on the handset
or headset, or after pressing the New Call key. Users are not in dial mode when on-hook dialing, contact dialing, or
call list dialing.
If set to 0, the digits entered up to and including the point where an impossible match occurred are sent to the
server immediately.
If set to 1, the handset gives the reorder tone when the impossible match occurs.
If set to 2, allow user to accumulate digits and dispatch call manually with the Send softkey.
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Parameter
Permitted Values
Default
dialplan.removeEndOfDial1
0 or 1
1
SIP, secure SIP, or TEL
URI
Null
If set to 1, strip trailing # digit from digits sent out.
dialplan.routing.emergency.outboundIdentity
The identity used to identify your handset when you place an emergency call from your handset. A valid SIP,
secure SIP, or TEL URI. The string may be 10 to 25 characters in length.
dialplan.routing.emergency.x.description1
Emergency contact description
dialplan.routing.emergency.x.server.y1
Emergency server
dialplan.routing.emergency.x.value
Emergency URL values
string
x=1:Emergency, Others: Null
positive integer
x=1: 1, others: Null
SIP URL (single entry)
x=1: 911, others: Null
x is the index of the emergency entry description and y is the index of the server associated with emergency entry
x. For each emergency entry (index x), one or more server entries (indexes (x,y)) can be configured. x and y must
both use sequential numbering starting at 1.
description: The label or description for the emergency address
server.y: The index representing the server to use for emergency routing
(dialplan.routing.server.x.address where x is the index).
value: The URLs that should be watched for. When the user dials one of the URLs, the call will be directed to the
emergency server defined by address.
Note: Blind transfer for 911 (or other emergency calls) may not work if registration and emergency servers are
different entities.
dialplan.routing.server.x.address1
dotted-decimal IP address
or hostname
Null
The IP address or hostname of a SIP server that will be used for routing calls. Multiple servers can be listed
starting with x=1 to 3 for fault tolerance. Note: Blind transfer for 911 (or other emergency calls) may not work if
registration and emergency servers are different entities.
dialplan.routing.server.x.port1
1 to 65535
5060
The port of a SIP server that will be used for routing calls
dialplan.routing.server.x.transport1
DNSnaptr, TCPpreferred,
UDPOnly, TLS, TCPOnly
DNSnaptr
The dns lookup of the first server to be dialed will be used, if there is a conflict with the others.
For example, if dialplan.routing.server.1.transport="UDPOnly" and
dialplan.routing.server.2.transport = "TLS", then UDPOnly is used.
1
Change causes handset to restart or reboot.
Per-registration dial plan configuration is also supported. The descriptions for each parameter
are in the table below. The parameters listed in this table overrule the parameters in the
previous table for registration x, where x is the registration number (for example,
dialplan.x.applyToTelUriDial overrules dialplan.applyToTelUriDial for
registration x):
Table 10-7: Per-Registration Dial Plan (Digit Map) Parameters
Parameter
Permitted Values
Default
dialplan.x.applyToCallListDial1
0 or 1
1
dialplan.x.applyToDirectoryDial1
0 or 1
0
dialplan.x.applyToForward
0 or 1
0
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Parameter
Permitted Values
Default
dialplan.x.applyToTelUriDial1
0 or 1
1
dialplan.x.applyToUserDial1
0 or 1
1
dialplan.x.applyToUserSend1
0 or 1
1
dialplan.x.digitmap1
string - max number of
characters 2560
Null
dialplan.x.digitmap.timeOut1
string - max number of
characters 100
Null
dialplan.x.e911dialmask
string - max number of
characters 256
Null
dialplan.x.e911dialstring
string - max number of
characters 256
Null
dialplan.x.applyToForward
0 or 1
0
dialplan.x.impossibleMatchHandling1
0 to 2
0
dialplan.x.originaldigitmap
string - max number of
characters 2560
Null
dialplan.x.removeEndOfDial1
0 or 1
1
dialplan.x.routing.emergency.y.value1
string - max number of
characters 64
Null
dialplan.x.routing.emergency.y.server.z1
0 to 3
0 For all x, y, and z = 1 to 3
dialplan.x.routing.server.y.address1
string - max number of
characters 256
Null
dialplan.x.routing.server.y.port1
1 to 65535
5060
dialplan.x.routing.server.y.transport1
DNSnaptr,
TCPpreferred, UDPOnly,
TLS, TCPOnly
DNSnaptr
1
Change causes handset to restart or reboot.
Location Values for E.911 Services
The values you enter for these Skype for Business Server-only parameters will be used by
E.911 services.
Table 11-8
Parameter
Permitted Values
Default
locInfo.x.label
String
Null
String
Null
String
Null
Enter a label for your location.
locInfo.x.country
Enter the country the handset is located in.
locInfo.x.A1
Enter the national subdivision the handset is located in, for example, a state or province.
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Parameter
Permitted Values
Default
locInfo.x.A3
String
Null
String
Null
String
Null
Enter the city the handset is located in.
locInfo.x.PRD
Enter the leading direction of the street location.
locInfo.x.RD
The name of the road or street the handset is located on.
locInfo.x.STS
String
Null
Enter the suffix of the name used in locInfo.x.RD, for example, Street, Avenue.
locInfo.x.POD
String
Null
String
Null
Enter the trailing street direction, for example SW.
locInfo.x.HNO
Enter the street address number of the handset’s location.
locInfo.x.HNS
String
Enter a suffix for the street address used in locInfo.x.HNS, for example,
locInfo.x.LOC
Null
A
or ½.
String
Null
Enter any additional information that identifies the location.
locInfo.x.NAM
String
Null
String
Null
Enter a name for the location, for example, a
business name, an occupant, a resident.
locInfo.x.PC
Enter the postal code of the location.
Real-Time Transport Protocol Ports
You can configure the handset to filter incoming RTP packets. You can filter the packets by IP
address, or by port. For greater security, you can also configure RTP settings to reject packets
arriving from a non-negotiated IP address or from an unauthorized source. You can reject
packets that the handset receives from a non-negotiated IP address or a non-negotiated port.
You can configure the handset to enforce symmetric port operation for RTP packets. When the
source port is not set to the negotiated remote sink port, arriving packets can be rejected.
You can also fix the handset’s destination transport port to a specified value regardless of the
negotiated port. This can be useful for communicating through firewalls. When you use a fixed
transport port, all RTP traffic is sent to and arrives on that specified port. Incoming packets are
sorted by the source IP address and port, which allows multiple RTP streams to be multiplexed.
You can specify the handset’s RTP port range. Since the handset supports conferencing and
multiple RTP streams, the handset can use several ports concurrently. Consistent with RFC
1889, the next-highest odd-numbered port is used to send and receive RTP.
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The handset is compatible with RFC 1889 - RTP: A Transport Protocol for Real-Time
Applications - and the updated RFCs 3550 and 3551. Consistent with RFC 1889, the handset
treats all RTP streams as bi-directional from a control perspective and expects that both RTP
end points will negotiate the respective destination IP addresses and ports. This allows real-time
transport control protocol (RTCP) to operate correctly even with RTP media flowing in only a
single direction, or not at all.
Summary
Parameter
Used to:
tcpIpApp.port.rtp.filterByIp
Filter RTP packets by IP address
tcpIpApp.port.rtp.filterByPort
Filter RTP packets by port
tcpIpApp.port.rtp.forceSend
Force-send packets on a specified port
tcpIpApp.port.rtp.mediaPortRangeStart
Set the starting port for RTP packet port range
Table 11-9: Configuring Real-Time Transport Protocol Ports
Parameter
Permitted Values
Default
tcpIpApp.port.rtp.filterByIp1
0 or 1
1
IP addresses can be negotiated through the SDP protocols. If set to 1, the handset rejects RTP packets that arrive
from non-negotiated IP addresses.
tcpIpApp.port.rtp.filterByPort1
0 or 1
0
Ports can be negotiated through the SDP protocol. If set to 1, the handset will reject RTP packets arriving from
(sent from) a non-negotiated port.
tcpIpApp.port.rtp.forceSend1
0 to 65535
0
Send all RTP packets to, and expect all RTP packets to arrive on, this port. If 0, RTP traffic is not forced to one
port. Note: Both tcpIpApp.port.rtp.filterByIp and tcpIpApp.port.rtp.filterByPort must be set
to 1 for this to work.
tcpIpApp.port.rtp.mediaPortRangeStart1
even integer 1024 to 65440
2222
The starting port for RTP packets. Ports will be allocated from a pool starting with this port up to a value of (startport + 47).
Note: Ensure that there is no contention for port numbers. For example, do not use 5060 (default port for SIP).
1
Change causes handset to restart or reboot.
Shared Line Appearances
With the shared call appearance feature enabled, an active call displays simultaneously on
multiple handsets in a group. By default, the answering handset has sole access to the
incoming call, called line seize. You can enable another handset in the group the ability to enter
a conversation, called a barge in. If the answering handset places the call on hold, that call
becomes available to all handsets of that group. All call states of a call —active, inactive, on
hold—are displayed on all handsets of a group.
This feature is dependent on support from a SIP call server. To enable shared call appearances
on your handset, you will need to obtain a shared line address from your SIP service provider.
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Shared Call Appearance Signaling
A shared line is an address of record managed by a call server. The server allows multiple end
points to register locations against the address of record.
The handset supports shared call appearances (SCA) using the SUBSCRIBE-NOTIFY method
in the “SIP Specific Event Notification” framework (RFC 3265). The events used are:

“call-info” for call appearance state notification

“line-seize for the handset to ask to seize the line
Summary
Parameter
Used to:
reg.x.address
specify the shared line address.
reg.x.type
specify the line type as shared.
reg.x.*
specify barge-in capabilities and line-seize subscription period if using
per-registration servers. A shared line will subscribe to a server providing
call state information.
call.shared.*
disable call diversion, expose auto-holds, resume with one touch, or play
a tone if line-seize fails.
voIpProt.SIP.specialEvent.lineSeize.no
nStandard
specify standard or non-standard behavior for processing a line-seize
subscription for mutual exclusion.
divert.x.sharedDisabled
specify per-registration whether diversion should be disabled on shared
lines.
Table 11-10: Shared Call Appearances
Parameter
Permitted Values
Default
call.shared.disableDivert1
0 or 1
1
If set to 1, the diversion feature for shared lines is disabled. Note: This feature is disabled on most call servers.
call.shared.exposeAutoHolds1
0 or 1
0
If 1, a re-INVITE will be sent to the server when setting up a conference on a shared line. If 0, no re-INVITE will be
sent to the server.
call.shared.seizeFailReorder1
0 or 1
1
If set to 1, play re-order tone locally on shared line seize failure.
voIpProt.SIP.specialEvent.lineSeize.nonStandard1
0 or 1
1
If set to 1, process a 200 OK response for a line-seize event SUBSCRIBE as though a line-seize NOTIFY with
Subscription State: active header had been received,. This speeds up processing.
divert.x.sharedDisabled1
0 or 1
1
If 0, call diversion features can be used on shared lines. If 1, call diversion features are disabled on shared lines.
1
Change causes handset to restart or reboot.
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Static DNS Cache
Failover redundancy can only be used when the configured IP server hostname resolves
(through SRV or A record) to multiple IP addresses. Unfortunately, the DNS cache cannot
always be configured to take advantage of failover redundancy.
The solution is to statically configure a set of DNS NAPTR SRV and/or A records into the
handset.
When a handset is configured with a DNS server, it will behave as follows by default:
 The handset will make an initial attempt to resolve a hostname that is within the static
DNS cache. For example, a query will be made to the DNS if the handset registers with its
SIP registrar.
 If the initial DNS query returns no results for the hostname or cannot be contacted, then
the values in the static cache are used for their configured time interval.
 After the configured time interval has elapsed, a resolution attempt of the hostname will
again result in a query to the DNS.
 If a DNS query for a hostname that is in the static cache returns a result, the values from
the DNS are used and the statically cached values are ignored.
When a handset is not configured with a DNS server, it will behave as follows:
 When the handset attempts to resolve a hostname within the static DNS cache, it will
always return the results from the static cache.
Support for negative DNS caching as described in RFC 2308 is also provided to allow faster
failover when prior DNS queries have returned no results from the DNS server. For more
information, see RFC 2308.
Summary
Parameter
Used to:
reg.x.server.y.*
Specify the call server used for this registration
dns.cache.A.x.*
Specify the DNS A address, hostname, and cache time interval (ttl)
dns.cache.NAPTR.x.*
Specify the DNS NAPTR parameters, including: name, order, preference,
regexp, replacement, service, and ttl
dns.cache.SRV.x.*
Specify DNS SRV parameters, including: name, port, priority, target, ttl, and
weight
Table 11-11: Configuring the Static DNS Cache
Parameter
Permitted Values
Default
reg.x.server.y.address
dotted-decimal IP address or
hostname
Null
The IP address or host name of a SIP server that accepts registrations. If not Null, all of the parameters in this table
will overrule the parameters specified in voIpProt.server.*. Notes: If this parameter is set, it overrules even if
the DHCP server is available. If this registration is used for Microsoft Office Communications Server 2007 R2 on
Spectralink handsets, this parameter must be in the form OCShostname.OSCdomain_name.
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Parameter
Permitted Values
Default
dns.cache.A.x.address
dotted-decimal IP version 4 address
Null
valid hostname
Null
300 to 536870912 (2^29), seconds
300
IP address.
dns.cache.A.x.name
Hostname
dns.cache.A.x.ttl
The TTL describes the time period the handset will use the configured static cache record. If a dynamic network
request receives no response, this timer begins on first access of the static record and once the timer expires, the
next lookup for that record will retry a dynamic network request before falling back on the static entry and its reset
TTL timer again.
dns.cache.NAPTR.x.flags
A single character from [A-Z, 0-9]
Null
The flags to control aspects of the rewriting and interpretation of the fields in the record. Characters are casesensitive. At this time, only ‘S’, ‘A’, ‘U’, and ‘P’ are defined as flags. See RFC 2915 for details of the permitted flags.
dns.cache.NAPTR.x.name
domain name string
Null
The domain name to which this resource record refers.
dns.cache.NAPTR.x.order
0 to 65535
0
An integer specifying the order in which the NAPTR records must be processed to ensure the correct ordering of
rules.
dns.cache.NAPTR.x.preference
0 to 65535
0
A 16-bit unsigned integer that specifies the order in which NAPTR records with equal "order" values should be
processed. Low numbers are processed before high numbers.
dns.cache.NAPTR.x.regexp
string containing a substitution
expression
Null
This parameter is currently unused.
Applied to the original string held by the client. The substitution expression is applied in order to construct the next
domain name that will be looked up. The grammar of the substitution expression is given in RFC 2915.
dns.cache.NAPTR.x.replacement
domain name string with SRV prefix
Null
The next name to query for NAPTR records depending on the value of the flags field. It must be a fully qualified
domain-name.
dns.cache.NAPTR.x.service
string
Null
Specifies the service(s) available down this rewrite path. For more information, see RFC 2915.
dns.cache.NAPTR.x.ttl
300 to 536870912 (2^29), seconds
300
The TTL describes the time period the handset will use the configured static cache record. If a dynamic network
request receives no response, this timer begins on first access of the static record and once the timer expires, the
next lookup for that record will retry a dynamic network request before falling back on the static entry and its reset
TTL timer again.
dns.cache.SRV.x.name
domain name string with SRV prefix
Null
0 to 65535
0
The domain name string with SRV prefix.
dns.cache.SRV.x.port
The port on this target host of this service. For more information, see RFC 2782.
dns.cache.SRV.x.priority
0 to 65535
0
The priority of this target host. For more information, see RFC 2782.
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Parameter
Permitted Values
Default
dns.cache.SRV.x.target
domain name string
Null
The domain name of the target host. For more information, see RFC 2782.
dns.cache.SRV.x.ttl
300 to 536870912 (2^29), seconds
300
The TTL describes the time period the handset will use the configured static cache record. If a dynamic network
request receives no response, this timer begins on first access of the static record and once the timer expires, the
next lookup for that record will retry a dynamic network request before falling back on the static entry and its reset
TTL timer again.
dns.cache.SRV.x.weight
0 to 65535
0
A server selection mechanism. For more information, see RFC 2782.
Example Static DNS Cache Configuration
The following examples show you how to configure the static DNS cache.
Example 1
This example shows how to configure static DNS cache using A records IP addresses in SIP
server address fields.
When the static DNS cache is not used, the site.cfg configuration will look as follows:
When the static DNS cache is used, the site.cfg configuration will look as follows:
Note: Details of the Preceding Example
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Above addresses are presented to 84-Series software in order, for example,
dns.cache.A.1, dns.cache.A.2, and so on.
Example 2
This example shows how to configure static DNS cache where your DNS provides A records for
reg.x.server.y.address but not SRV. In this case, the static DNS cache on the handset
provides SRV records. For more information, see RFC 3263.
When the static DNS cache is not used, the site.cfg configuration will look as follows:
When the static DNS cache is used, the site.cfg configuration will look as follows:
Settings: Port Value Settings
The reg.1.server.1.port and reg.1.server.2.port values in this
example are set to null to force SRV lookups.
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Example 3
This example shows how to configure static DNS cache where your DNS provides NAPTR and
SRV records for reg.x.server.x.address.
When the static DNS cache is used, the site.cfg configuration will look as follows:
When the static DNS cache is used, the site.cfg configuration will look as follows:
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Settings: Forcing NAPTR Lookups
The reg.1.server.1.port, reg.1.server.2.port,
reg.1.server.1.transport, and reg.1.server.2.transport values in
this example are set to null to force NAPTR lookups.
Using Static DNS Cache for Redundancy
Failover redundancy can only be utilized when the configured IP server hostname resolves
(through SRV or A record) to multiple IP addresses. Unfortunately, some customer’s are unable
to configure the DNS to take advantage of failover redundancy.
The solution is to provide the ability to statically configure a set of DNS NAPTR SRV and/or A
records into the handset.
When a handset is configured with a DNS server, it will behave as follows by default:

An initial attempt to resolve a hostname that is within the static DNS cache, for example
to register with its SIP registrar, results in a query to the DNS.
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
If the initial DNS query returns no results for the hostname or cannot be contacted, then
the values in the static cache are used for their configured time interval.

After the configured time interval has elapsed, a resolution attempt of the hostname will
again result in a query to the DNS.

If a DNS query for a hostname that is in the static cache returns a result, the values from
the DNS are used and the statically cached values are ignored.
When a handset is not configured with a DNS server, it will behave as follows

An attempt to resolve a hostname that is within the static DNS cache will always return
the results from the static cache.
Support for negative DNS caching as described in RFC 2308 is also provided to allow faster
failover when prior DNS queries have returned no results from the DNS server. For more
information, go to http://tools.ietf.org/html/rfc2308 .
Configuration File Changes
Configuration changes can be performed centrally at the boot server:
Central
(boot server)
Configuration file:
sip.cfg
Specify DNS NAPTR, SRV, and A records for use when the handset
is not configured to use a DNS server.
For more information, refer to DNS Cache <dns/>.
DNS Cache <dns/>
In the tables below, a maximum of 12 entries of NAPTR, SRV, and A record can be added.
This attribute includes:

• NAPTR <NAPTR/> attribute

• SRV <SRV/>

• A <A/>
NAPTR <NAPTR/>
Attribute
Permitted Values
Default
Interpretation
dns.cache.NAPTR.x.name
domain name string
Null
The domain name to which this resource record
refers.
dns.cache.NAPTR.x.ttl
0 to 65535, seconds
300
Specifies the time interval that the resource
record may be cached before the source of the
information should again be consulted.
dns.cache.NAPTR.x.order
0 to 65535
0
A 16-bit unsigned integer specifying the order in
which the NAPTR records must be processed to
ensure the correct ordering of rules.
dns.cache.NAPTR.x.
pref erence
0 to 65535
0
A 16-bit unsigned integer that specifies the
order in which NAPTR records with equal
"order" values should be processed, low
numbers being processed before high numbers.
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Attribute
Permitted Values
Default
Interpretation
dns.cache.NAPTR.x.flags
string
Null
Flags to control aspects of the rewriting and
interpretation of the fields in the record. Flags
are single characters from the set [A-Z, 0-9].
The alphabetic characters are case insensitive.
At this time only four flags, "S", “A”, “U”, and “P”
are defined. For more information, go to
http://tools.ietf.org/html/rfc2915.
dns.cache.NAPTR.x.serv
ice
string
Null
Specifies the service(s) available down this
rewrite path. For more information, go to
http://tools.ietf.org/html/rfc2915.
dns.cache.NAPTR.x.
regexp
string
Null
A string containing a substitution expression
that is applied to the original string held by the
client in order to construct the next domain
name to lookup. The grammar of the
substitution expression is given in RFC 2915.
Note: This attribute is currently not used.
dns.cache.NAPTR.x.
replacement
domain name string
with SRV prefix
Null
The next name to query for NAPTR, SRV, or
address records depending on the value of the
flags field. It must be a fully qualified domainname.
SRV <SRV/>
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
dns.cache.SRV.x.name
domain name string
Null
The domain name string with SRV prefix.
dns.cache.SRV.x.ttl
0 to 65535, seconds
300
Specifies the time interval that the resource
record may be cached before the source of
the information should again be consulted.
dns.cache.SRV.x.priority
0 to 65535
0
The priority of this target host. For more
information, go to
http://tools.ietf.org/html/rfc2782 .
dns.cache.SRV.x.weight
0 to 65535
0
A server selection mechanism. For more
information, go to
http://tools.ietf.org/html/rfc2782 .
dns.cache.SRV.x.port
0 to 65535
0
The port on this target host of this service. For
more information, go to
http://tools.ietf.org/html/rfc2782 .
dns.cache.SRV.x.target
domain name string
Null
The domain name of the target host. For more
information, go to
http://tools.ietf.org/html/rfc2782.
A <A/>
This configuration attribute is defined as follows:
Attribute
Permitted Values
Default
Interpretation
dns.cache.A.x.name
valid hostname
Null
Hostname
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Attribute
Permitted Values
Default
Interpretation
dns.cache.A.x.ttl
0 to 65535
300
Specifies the time interval that the resource
record may be cached before the source of
the information should again be consulted.
dns.cache.A.x.address
dotted-decimal IP
version 4 address
Null
IP address that hostname
dns.cache.A.x.name maps to.
Examples
Example 1
This example shows how to configure static DNS cache using A records IP addresses in SIP
server address fields.
When the static DNS cache is not used, the configuration would look as follows:
reg.1.address="1001"
reg.1.server.1.address="172.23.0.140"
reg.1.server.1.port="5075"
reg.1.server.1.transport="UDPOnly"
reg.1.server.2.address="172.23.0.150"
reg.1.server.2.port="5075"
reg.1.server.2.transport="UDPOnly"
When the static DNS cache is used, the configuration would look as follows:
reg.1.address="1001"
reg.1.server.1.address="sipserver.example.com"
reg.1.server.1.port="5075"
reg.1.server.1.transport="UDPOnly"
reg.1.server.2.address=""
reg.1.server.2.port=""
reg.1.server.2.transport=""
dns.cache.A.1.name="sipserver.example.com"
dns.cache.A.1.ttl="3600"
dns.cache.A.1.address="172.23.0.140"
dns.cache.A.2.name="sipserver.example.com"
dns.cache.A.2.ttl="3600"
dns.cache.A.2.address="172.23.0.150"
Parameter order
Above addresses are presented to SIP application in order, for example,
dns.cache.A.1, dns.cache.A.2, and so on.
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Example 2
This example shows how to configure static DNS cache where your DNS provides A records for
server.X.address but not SRV. In this case, the static DNS cache on the handset provides
SRV records. For more information, go to http://tools.ietf.org/html/rfc3263 .
When the static DNS cache is not used, the configuration would look as follows:
reg.1.address="1002@sipserver.example.com"
reg.1.server.1.address="primary.sipserver.example.com"
reg.1.server.1.port="5075"
reg.1.server.1.transport="UDPOnly"
reg.1.server.2.address="secondary.sipserver.example.com"
reg.1.server.2.port="5075"
reg.1.server.2.transport="UDPOnly"
When the static DNS cache is used, the configuration would look as follows:
reg.1.address="1002"
reg.1.server.1.address="sipserver.example.com"
reg.1.server.1.port=""
reg.1.server.1.transport="UDPOnly"
reg.1.server.2.address=""
reg.1.server.2.port=""
reg.1.server.2.transport=""
dns.cache.SRV.1.name="_sip._udp.sipserver.example.com "
dns.cache.SRV.1.ttl= "3600"
dns.cache.SRV.1.priority="1"
dns.cache.SRV.1.weight="1"
dns.cache.SRV.1.port="5075"
dns.cache.SRV.1.target="primary.sipserver.example.com"
dns.cache.SRV.2.name="_sip._udp.sipserver.example.com "
dns.cache.SRV.2.ttl= "3600"
dns.cache.SRV.2.priority="2"
dns.cache.SRV.2.weight="1"
dns.cache.SRV.2.port="5075"
dns.cache.SRV.2.target="secondary.sipserver.example.com
Reason for Null setting
The reg.1.server.1.port and reg.1.server.2.port values in this example are set to
null to force SRV lookups.
Example 3
This example shows how to configure static DNS cache where your DNS provides NAPTR and
SRV records for server.X.address .
When the static DNS cache is not used, the configuration would look as follows:
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reg.1.address="1002@sipserver.example.com
reg.1.server.1.address="172.23.0.140"
reg.1.server.1.port="5075"
reg.1.server.1.transport="UDPOnly"
reg.1.server.2.address="172.23.0.150"
reg.1.server.2.port="5075"
reg.1.server.2.transport="UDPOnly"
When the static DNS cache is used, the configuration would look as follows:
reg.1.address="1002"
reg.1.server.1.address="sipserver.example.com"
reg.1.server.1.port=""
reg.1.server.1.transport=""
reg.1.server.2.address=""
reg.1.server.2.port=""
reg.1.server.2.transport=""
dns.cache.NAPTR.1.name="sipserver.example.com"
dns.cache.NAPTR.1.ttl= "3600"
dns.cache.NAPTR.1.order="1"
dns.cache.NAPTR.1.preference="1"
dns.cache.NAPTR.1.flag="s"
dns.cache.NAPTR.1.service=" SIP+D2U"
dns.cache.NAPTR.1.regexp=""
dns.cache.NAPTR.1.replacement="_sip._udp.sipserver.example.com"
dns.cache.SRV.1.name="_sip._udp.sipserver.example.com "
dns.cache.SRV.1.ttl= "3600"
dns.cache.SRV.1.priority="1"
dns.cache.SRV.1.weight="1"
dns.cache.SRV.1.port="5075"
dns.cache.SRV.1.target="primary.sipserver.example.com"
dns.cache.SRV.2.name="_sip._udp.sipserver.example.com "
dns.cache.SRV.2.ttl= "3600"
dns.cache.SRV.2.priority="2"
dns.cache.SRV.2.weight="1"
dns.cache.SRV.2.port="5075"
dns.cache.SRV.2.target="secondary.sipserver.example.com
dns.cache.A.1.name="primary.sipserver.example.com"
dns.cache.A.1.ttl="3600"
dns.cache.A.1.address="172.23.0.140"
dns.cache.A.2.name="secondary.sipserver.example.com"
dns.cache.A.2.ttl="3600"
dns.cache.A.2.address="172.23.0.150"
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Reason for Null setting
The reg.1.server.1.port, reg.1.server.2.port, reg.1.server.1.transport, and
reg.1.server.2.transport values in this example are set to null to force NAPTR lookups.
Voice Activity Detection
The purpose of voice activity detection (VAD) is to detect periods of silence in the transmit data
path so the handset doesn’t have to transmit unnecessary data packets for outgoing audio. This
process conserves network bandwidth. VAD must be turned on in the PBX for it to work
properly.
For compression algorithms without an inherent VAD function, such as G.711, the handset uses
the codec-independent comfort noise transmission processing specified in RFC 3389. The RFC
3389 algorithm is derived from G.711 Appendix II, which defines a comfort noise (CN) payload
format (or bit-stream) for G.711 use in packet-based, multimedia communication systems. The
handset generates CN packets — also known as Silence Insertion Descriptor (SID) frames —
and also decodes CN packets, to efficiently regenerate a facsimile of the background noise at
the remote end.
These settings control the performance of the voice activity detection (silence suppression)
feature.
Table 11-12: Voice Activity Detection (VAD) Parameters
Parameter
Permitted Values
Default
voice.vad.signalAnnexB1
0 or 1
1
If 0, there is no change to SDP. If 1, Annex B is used and a new line is added to SDP depending on the setting of
voice.vadEnable.

If voice.vadEnable is set to 1, add parameter line a=fmtp:18 annexb="yes" below a=rtpmap…
parameter line (where '18' could be replaced by another payload).

If voice.vadEnable is set to 0, add parameter line a=fmtp:18 annexb="no" below a=rtpmap…
parameter line (where '18' could be replaced by another payload).
voice.vadEnable1
0 or 1
0
If 0, voice activity detection (VAD) is disabled. If 1, VAD is enabled.
voice.vadThresh1
integer from 0 to 30
15
The threshold for determining what is active voice and what is background noise in dB. Sounds louder than this
value will be considered active voice, and sounds quieter than this threshold will be considered background noise.
This does not apply to G.729AB codec operation which has its own built-in VAD function.
1
Change causes handset to restart or reboot.
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Part IV: Troubleshooting and
Maintaining your Deployment
Part IV provides you with the information you need to troubleshoot issues with your Spectralink
84-Series handsets and for basic, advanced, audio, user and telephone security features.
Part IV consists of the following chapters:

Chapter 11: Troubleshooting Your Spectralink handsets

Chapter 12: Miscellaneous Maintenance Tasks
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Chapter 12: Troubleshooting Your
Spectralink Handsets
This chapter shows you some tools and techniques for troubleshooting Spectralink handsets
running Spectralink Software. The handset can provide feedback in the form of on-screen error
messages, status indicators, and log files for troubleshooting issues.
This chapter includes information on:

Understanding Error Message Types

Status Menu

Testing Phone Hardware

Log Files

Managing the Phone’s Memory

Testing Phone Hardware

Uploading a Phone’s Configuration

Network Diagnostics

Network Protocols and Ports Used by Spectralink handsets
This chapter also addresses handset issues, likely causes, and corrective actions. Issues are
grouped as follows:

Power and Startup Issues

Dial Pad Issues

Screen and System Access Issues

Calling Issues

Display Issues

Audio Issues

Licensed Feature Issues

Upgrading Issues
Review the latest Spectralink Software Release Notes on the support site for known problems
and possible workarounds. If a problem is not listed in this chapter or in the latest Release
Notes, contact your Certified Spectralink Reseller for support.
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Troubleshooting Flow Diagram
A troubleshooting flow diagram provides an easy to follow list of symptoms to facilitate problem
isolation, data collection and resolution.
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Understanding Error Message Types
Several types of errors can occur while the handset is booting. If an error occurs, the handset
will inform you by displaying an error message. Errors can affect how the handset boots up. If
the error is fatal, the handset will not be able to boot until the error is resolved. If the error is
recoverable, the handset will continue to boot but the handset’s configuration may change or be
incomplete.
Updater Error Messages
Most of the following errors will be logged to the handset’s boot log. However, if you are having
trouble connecting to the provisioning server, the handset will likely not be able to upload the
boot log <MAC-boot.log> file to its provisioning server for later analysis.
Failed to get boot parameters via DHCP
The handset does not have an IP address and therefore cannot boot. Check the following items:


Is the 84-Series handset connected to the WLAN?
o
Does the AP signal strength indicator, at the top left of the handset display, show
connected to the WLAN, as shown below, or does it have a red X across the AP
signal strength bars?
o
A red X across the AP signal strength bars indicates the handset is not
connected to the WLAN. In this case, ensure the handset has the proper
parameters set. These are configured either during the initial QNC/SLIC
configuration with the USB cable or thru the handset keypad interface. Ensure
the proper radio is turned on and WLAN security with the AP is turned on and
available.
Check the DHCP configuration.
o
Settings> Status> Network> TCP/IP Parameters. Is DHCP enabled or
disabled.
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
Does the handset have an IP address on the right network subnet?

Check the IP, Subnet Mask & IP Gateway. Are they as expected?

If not, check with the DHCP server administrator to ensure that
DHCP is providing the 84-Series handset with the proper settings.

If yes, are the neighbor addresses in the right subnet as
expected? Settings> Status> Network> Neighbors
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
Can the handset ping its provisioning and SIP server IP
addresses? Settings> Status> Diagnostics> Network> Ping>
Enter IP address to ping> Start
Application <file name> is not compatible with this handset!
When the Updater displays the error ‘The application is not compatible’, an application file was
downloaded from the provisioning server but cannot be installed on this handset. This issue can
usually be resolved by finding a software image that is compatible with the hardware or the
BootROM and installing it on the provisioning server. Be aware that there are various different
hardware and software dependencies.
Do other 84-Series handsets work with this version of software?
Updater has an incorrect signature
An error is detected by a .ld file just loaded to the handset. While customers load and run
General Availability (GA) code from the Spectralink.com web site, there are a few special
84-Series handsets which use special engineering code. This error means the wrong code (GA
code on engineering handset or vice versa) is being loaded on the handset. Ensure the correct
*.sip.ld code is loaded to the handset. Download the latest code from the www.spectralink.com
web site to get GA code.
Could not contact boot server using existing configuration
The handset could not contact the provisioning server, but the causes may be numerous. It may
be related to DHCP configuration, or it could be a problem with the provisioning server itself.
The handset can recover from this error so long as it previously downloaded a valid application
BootROM image and all of the necessary configuration files.

Check the provisioning boot server IP address known by the handset at Settings>
Status> Platform> Configuration. This screen shows Boot server IP and protocol.

Determine if the boot server IP address can be Pinged from the handset by going to
Settings> Status> Diagnostics> Network> Ping. Enter the IP address and press start.

Check the Boot server in the load protocol log file to see if there are any errors or
reasons why the configuration files do not load.

Ensure there are no network fire walls blocking the 84-Series handset from accessing
the boot server.
Error, application is not present!
This message indicates that the handset has no application (operating handset code) stored in
firmware or memory, that the handset could not download an application, and that the handset
cannot boot. To resolve this issue, you must download compatible Spectralink Software to the
handset using one of the supported provisioning protocols. You need to resolve the issue of
connecting the handset to the provisioning server and provide a compatible software image on
the provisioning server. This error is fatal, but recoverable. Check the following:
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
Check the boot server IP address at Settings> Status> Platform> Configuration.
Ensure the boot server is operational and available.

Check the protocol used by the handset at the above menu. Ensure the FTP/TFTP
server is running on the boot server.

Check the boot server log for the proper protocol to determine if the handset contacted
the server and asked for a sip.ld.

Check the parameter APP_FILE_PATH="sip.ld" in the 000000000000.cfg or <mac>.cfg
file to ensure a load file is specified.

Is the proper sip.ld file available at the FTP/TFTP server location/directory specified?
Spectralink Software Error Messages
The warning notification feature provides users a visual indication that one or more error
conditions exist. When the warning notification displays, users will see:

An informative message when the warning is first detected

An error icon in the status bar on the idle display, as shown next:

A persistent list of current warnings, which can be viewed from Settings> Status>
Diagnostics> Warnings
Wi-Fi: No APs Found
This message displays on Spectralink handsets if the handset is unable to connect to an
access point (AP) on the wireless network. Check the following:

Ensure the initial configuration parameters are set properly in the handset using the USB
cable or keypad to set domain, SSID, turn on the radio, set WLAN security, set DHCP or
provide a static IP and specify the boot server IP address and protocol.

Ensure an AP with the proper SSID is available
○
○
Check at Settings> Status> Diagnostics> WiFi Stats
»
Screen 2/6 General Info shows the SSID configured in the handset
»
Screen 3/6 AP List shows available APs
Check Site Survey mode at: Settings> Advanced Settings> [enter password]>
Administration Settings> Diagnostics> Run Site Survey
»
The Current Configuration screen shows SSID, Regulatory Mode and frequency
bands enabled
»
Press Start to see available APs, Channel, RSSI and SSID for the configuration
in the handset
»
Press All to see all available APs for all (any) SSIDs
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Network link is down
Since the Spectralink handsets do not have an LED indicating network LINK status like many
networking devices, link failures are indicated with a message.
The ‘Network link is down’ message will be shown on the screen whenever the handset is not in
the menu system and will persist until the link problem is resolved. Call related functions and the
softkeys and line keys are disabled when the network is down; however the menu system
works.
Check the top left corner of the 84-Series display to determine if the AP signal strength bars
have a red X across them.
If there is a red X, the handset is not connected to an AP.
Has the handset connected to the WLAN before or is this the first failure of this kind? Is the AP
operational?
Ensure the handset is configured properly for the WLAN (SSID, security, RF radio and
channels,
If there is no red X then the handset has connected to an AP.
This means the handset and AP agree for SSID, wireless security, RF radio and channels, etc.
Does the handset have an IP address?
If DHCP is enabled then it must supply an IP address to the handset.
If DHCP is disabled, then the handset will have a static IP address assigned.
Check handset for an IP address at Settings> Status> Network> TCPIP Parameters
Can it ping the boot server, SIP server or another 84-Series handset?
The Ping function is at: Settings> Status> Diagnostics> Network> Ping. Enter IP address
and press Start.
Did the handset download configuration files when it booted?
Check Settings> Status> Platform> Configuration scroll down to see file names and
parameters loaded from each one
Config file errors
Config file error: Files contain invalid params: <filename1>, <filename2>,...
Config file error: <filename> contains invalid params.
This message also appears if any configuration file contains:

More than 100 unknown parameters, or

More than 100 out-of-range values, or

More than 100 invalid values.
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Update the configuration files to use the correct parameters. See the Deployment Guide for
details.
Navigate to Settings> Status> Platform> Configuration then scroll down to see file names of
the files that loaded and number of parameters loaded from each file (Errors, Duplicates or OK).
Are common files loaded properly by other handsets?
Did the problem just start? What has changed? Have the files been revised recently?
Insufficient Bandwidth
This message displays if a Spectralink handset has a poor network connection or the AP does
not have enough bandwidth available to handle this handset.
This error could go with the Admission Control setting. Generally, the handset will require
enough bandwidth to handle (standby) control frames and in call signaling and audio frames.
When the AP has little bandwidth available because of other clients, with Admission Control
enabled, it will inform the handset there is not enough bandwidth available. The handset will
roam to find another AP, if possible. If there is no other APs available the handset is not able to
recover.
Move the handset to another area with required bandwidth or add RF signaling with new APs so
there is enough bandwidth available in this and other areas of the facility.

Does the handset work in other areas of the facility (other APs)?

Does the failure occur in one area or the same area each time?
Invalid Regulatory Domain
This message will display on Spectralink 84-Series handsets if you set the regulatory domain on
your handset to an incorrect regulatory domain for your location. If you see this message, press
the Details softkey to get additional information about the invalid setting and to find out what are
valid settings. If an invalid regulatory domain is set, the handset’s radio will be disabled. For
example, the valid regulatory domain for the US is 01; if the regulatory domain is set to 10 (New
Zealand), then this error is generated and the radio is disabled.

Set the Regulatory Domain to the proper domain number and try again.

Check Site Survey mode at: Settings> Advanced Settings> [enter password]>
Administration Settings> Diagnostics> Run Site Survey
o

The Current Configuration screen shows SSID, Regulatory Mode and frequency
bands enabled
Are there WLAN APs available with this SSID, security, frequency bands and channels?
Invalid Regulatory Domain Setting
This message will display on Spectralink 84-Series handsets if some of your handset settings
are deemed incorrect according to the regulatory domain for your location. Each domain has its
own set of restrictions such as TX power limits and sub-bands. If one of these settings is not
within the restriction limits, an error message displays with the details about which setting is
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incorrect. If an invalid regulatory domain setting is detected, the handset’s radio is not disabled,
but the restriction is enforced.
Line: Unregistered
This message displays if a line fails to register with the call server. Check the following:

Does the handset have the proper SIP server IP address? Navigate to Settings>
Status> Lines> Server-1: IP Address

Do the parameters in [the config file with extension parameters] match the SIP server for
this extension?

Are there errors in [the config file with extension parameters]? Navigate to Settings>
Status> Platform> Configuration
○
Config: files loaded to the handset
○
Web: over-ride parameters loaded
○
Local: for over-ride parameters loaded
○
[Extension parameters]: Are there errors or duplicates? Is this extension set up and
enabled in the SIP server?
Login credentials have failed. Please update them if information is incorrect.
This message displays when the user enters incorrect login credentials (Settings> Basic
Settings> Login Credentials).

Ensure the username entered matches a .cfg file which has extension parameters.
Example: User enters Username: Sallyj. A Sallyj.cfg file must exist. User enters a
password. The Sallyj.cfg file must have a password parameter which matches the
password entered by the user. Finally, the Sallyj.cfg file must have parameters which
allow the connection to the SIP server.


Check the files loaded at: Settings> Status> Platform> Configuration
○
Config: files loaded to the handset
○
Web: over-ride parameters loaded which cause a failure
○
Local: over-ride parameters loaded which cause a failure
○
[Extension parameters]: for proper settings to allow the connection. Are there errors
or duplicates?
Check the boot server and protocol log file to determine which file is asked for, if it exists
and if it is loaded to the handset.
Time/Date out of sync
This message indicates the SNTP server and services are not available or could not be
contacted. It could also be the configuration files do not specify the SNTP server.
Ensure the SNTP server is available and config files are setup properly.
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Missing files, config reverted
This message displays when errors in the configuration or a failure to download the
configuration files force the handset to revert to its previous (known) condition with a complete
set of configuration files. This will also display if the files listed in the 000000000000.cfg or
<MAC Address>.cfg file are not present on the provisioning server. Check the following:

Navigate to Settings> Status> Platform> Configuration scroll down to see files list.

Check the CONFIG_FILES= parameter in the 000000000000.cfg or <mac>.cfg file to
see if it has the proper files specified, the files are available and the boot server is
available.

Check the boot server and protocol log file to determine which file is asked for and if it is
loaded to the handset.
Network Authentication Failure
This message displays if 802.1X authentication between the Spectralink handset and WLAN AP
fails. The codes shown in the following table will display on the handset’s screen if the Details
softkey is pressed. They can also be found in the log files:
Table 11-1: Event Codes and Descriptions
Event Code
Description
Comments
1
Unknown events
This includes any event listed in
this table.
2
Mismatch in EAP Method type
Authenticating server's list of EAP methods does not
match with clients’.
30xxx
TLS Certificate failure
The TLS certificate-related failures. "xxx" when having a
non-zero value, is the standard TLS alert message
code. For example, if a bad/invalid certificate (on the
basis of its signature and/or content) is presented by the
handset, "xxx" will be 042. If the exact reason for the
certificate being invalid is not known, then the generic
certificate error code will be xxx=000.
31xxx
Server Certificate failure
Certificate presented by the server is considered invalid.
"xxx" can take the following values:
• 009 - Certificate not yet Valid
• 010 - Certificate Expired
• 011 - Certificate Revocation List
(CRL) not yet Valid
• 012 - CRL Expired
4xxx
Other TLS failures
This is due to TLS failure other than certification related
errors. The reason code (the TLS alert message code)
is represented by "xxx". For example, if the protocol
version presented by the server is not supported by the
handset, then xxx will be 70, and the EAP error code will
be 4070.
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See section 7.2 of RFC 2246 for
further TLS alert codes and error
codes.
See section 7.2 of RFC 2246 for
further TLS alert codes and error
codes.
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Status Menu
Debugging of a single 84-Series handset may be possible by examining the handset’s status
menu. Press the right arrow key, select Settings> Status, and press the OK key to view the
Status menu. Scroll to one of the Status menu items and press the OK key. Each of the menu
items are explained next.
Troubleshooting: I can’t find the Status menu on my Spectralink handset
To view the Status menu on a Spectralink handset use the left or right arrow key to
navigate to Settings> Status> OK.
Under the Platform menu, you can get details on the handset’s serial number, MAC address,
the current IP address, the Updater version, the application version, the name of the SIP.ld file
loaded, the names of the config files loaded, and the IP address and protocol of the provisioning
server along with Errors, Duplicates and OK parameters per file.
In the Network menu, you can find information about the TCP/IP Setting, DHCP, handset IP,
subnet mask, default gateway, SNTP, DNS info, , and up time and statistics on packets sent
and received since last boot. The Neighbors screen shows the MAC and IP address of neighbor
nodes in this subnet.
The Lines menu will show you details about the status of each line that has been configured on
the handset as well as Server IP address.
The Diagnostics menu offers a series sub menus to test the following:

Hardware tests to verify correct operation of the microphone, speaker, handset, keypad,
display, LED and third party headset, if present. You can also test that each of the keys
on the handset is working, and display the function assigned to each of the keys in the
configuration.

Graphs show CPU load, Network load and Memory Usage

Media Statistics shows Audio codec packet stats and Jitter

Network allows Ping tests and Trace Route tests

WiFi Stats shows six screens displaying [1/6] packet counts, [2/6]General Info (SSID,
Last data Rate, Tx Power), [3/6]AP List (up to four APs with MAC, channel and RSSI),
[4/6]Re/Association Stats, [5/6]WLAN Security stats, [6/6]EAP Info including 802.11n if
disabled.

Warnings menu shows warning messages related to handset operation, connection to
the WLAN, SIP server, etc.

The Licenses menu shows installed license information

The Location menu shows location information.
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Log Files
Spectralink handsets will log various events to files stored in the flash file system and will
periodically upload these log files to the provisioning server. The files are stored in the handset’s
home directory or a user-configurable directory. You can also configure a handset to send log
messages to a syslog server.
There is one log file for the Updater and one for the Spectralink Software. When a handset
uploads its log files, they are saved on the provisioning server with the MAC address of the
handset prepended to the file name. For example, 00907a0e360b-boot.log and
00907a0e360b-app.log are the files associated with MAC address 00907a0e360b. The
Updater (boot) log file is uploaded to the provisioning server after every reboot. The application
log file is uploaded periodically or when the local copy reaches a predetermined size.
Both log files can be uploaded on demand using a multiple key combination described in
Multiple Key Combinations. The handset uploads four files, namely, <mac>-boot.log, <mac>app.log, mac-now-boot.log, and mac-now-app.log. The -now- logs are uploaded manually
unless they are empty. Spectralink 84-Series handset use the 1, 5, and 9 dial pad keys to force
log file uploads.
The amount of logging that the handset performs can be tuned for the application to provide
more or less detail on specific components of the handset’s software. For example, if you are
troubleshooting a SIP signaling issue, you are not likely interested in DSP events. Logging
levels are adjusted in the configuration files or via the Web Configuration Utility. You should not
modify the default logging levels unless directed to by Spectralink Customer Support.
Inappropriate logging levels can cause performance issues on the handset.
In addition to logging events, the handset can be configured to automatically execute commandline instructions at specified intervals that output run-time information such as memory
utilization, task status, or network buffer contents to the log file. These techniques should only
be used in consultation with Spectralink Customer Support.
Logging Options
Each of the components of the Spectralink Software is capable of logging events of different
severity. This allows you to capture lower severity events in one part of the application, and high
severity events for other components.
The parameters for log level settings are found in the log.cfg template. Log levels range from 0
to 6 (0 for the most detailed (Debug) logging, 6 for the least detailed (fatal errors only). There
are many different log types or categories that can be adjusted to assist with the investigation of
different problems.
When testing is complete, remember to remove the special troubleshooting configuration
parameters from the configuration files.
There are other logging parameters, described next, that you may wish to modify. Changing
these parameters will not have the same impact as changing the logging levels, but you should
still understand how your changes will affect the handset and the network.
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
log.render.level—Sets the lowest level that can be logged (default=1) by any of
the log.level.change.module_name parameters.

log.render.file.size—Maximum size before log file is uploaded (default=32 kb)

log.render.file.upload.period—Frequency, in seconds, of log uploads (default
is 86400 seconds = 24 hours)

log.render.file.upload.append—Controls whether log files on the provisioning
server are overwritten or appended, not supported by all servers (default=1 so files are
appended)

log.render.file.upload.append.sizeLimit—Controls the maximum size of log
files on the provisioning server (default=512 kb)

log.render.file.upload.append.limitMode—Controls whether to stop or
delete logging when the server log reaches its maximum size (default=delete)
Scheduled Logging
Scheduled logging is a powerful tool that can help troubleshoot issues that occur after the
handset has been operating for some time.
The output of the logging parameters is written to the application log, and can be examined later
(for trend data).
The parameters for scheduled logging are found in the log.cfg template. They are
log.sched.x.
For an example of a configuration file and the resulting log file, see Figure 11-1: Scheduled
Logging Log File, shown next.
Figure 11-1: Scheduled Logging Log File
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Manual Log Upload
If you want to look at the log files without having to wait for the handset to upload them (which
could take as long as 24 hours or more), initiate an upload by pressing the correct multiple key
combination on the handset (see Multiple Key Combinations – 1 & 5 & 9).
When the log files are manually uploaded, the word now is inserted into the name of the file, for
example, 00907a0e360b-now-boot.log .
Logging Modules
Logging Modules include:
app1
httpd
push
brow
ice
sip
cfg
key
so
dot1x
pps
tickt
Additional log modules can be found in the log.cfg file.
tls
utilm
wlan
wmgr
WLAN syslog entries can be very useful in determining AP RSSI, packet retry rates, Network
Jitter, Dropped packet rates, Late packets, reasons to roam and many other elements.
Major categories of WLAN entries

AStats
○
AP MAC address
○
RSSI
○
Pay load size, release interval
○
TX packet count
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


○
RX packet count
○
Missed packet count
○
Dropped packet count
○
Jitter in ms
AThresh
○
AP MAC address
○
RSSI
○
Pay load size, release interval
○
TX packet count
○
RX packet count
○
Missed packet count
○
Dropped packet count
○
Jitter in ms
NStats
○
AP MAC address
○
AP RSSI
○
TX packet count
○
RX packet count
○
BTX (Broadcast TX) packet count
○
BRX packet count
○
MTX (Multicast TX) packet count
○
MRX packet count
○
TX Drop packet count & %
○
TX Retry packet count & %
○
RX Retry packet count & %
NThresh
○
AP MAC address
○
AP RSSI
○
TX packet count
○
RX packet count
○
BTX (Broadcast TX) packet count
○
BRX packet count
○
MTX (Multicast TX) packet count
○
MRX packet count
○
TX Drop packet count & %
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

○
TX Retry packet count & %
○
RX Retry packet count & %
Successful Handoff
○
Roam To AP MAC address
○
Roam To channel, Score, RSSI & Penalty
○
Roam From AP MAC address
○
Roam From channel, score, RSSI, Penalty & Reason code
○
Other AP (up to four total) MAC, channel, score, RSSI, penalty & reason code
○
TXPO (Tx power of old AP in dBm)
○
TXPN (TX power of new AP in dBm)
Failed Handoff
○
Roam TO AP MAC address
○
Roam To channel, score, RSSI, penalty & reason code
○
Previous AP MAC, channel, score, RSSI, penalty & original handoff reason code
This information can be used to evaluate how well the WLAN is supporting the 84-Series
handsets.
Low RSSI (too weak) will cause high retries, dropped frames, missed frames, poor audio and
poor roaming.
Web Info: Using Syslog on Spectralink handsets
For more information about syslog, see Technical Bulletin CS-14-20: Syslog on
Spectralink Handsets.
Managing the Phone’s Memory Resources
Spectralink handsets are designed to operate optimally in a variety of deployments and realworld environments. Each new software release adds new features and capabilities that require
varying degrees of the handset’s memory resources. To ensure your handsets and their
configured features operate smoothly, you will need to check that the handsets have adequate
available memory resources. If you are using a range of handset features - especially
customized or advanced features - you may need to manage handset memory resources. To
help you optimize your handset features and memory resources, Spectralink provides several
tools and troubleshooting tips.
Identifying Symptoms
When the handset memory resources start to run low, you may notice one or more of the
following symptoms:
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
The handsets reboot or freeze up.

The handsets do not download all ringtones, directory entries, backgrounds, or XML
dictionary files.

Applications running in the browser stop or do not run at all.
The next sections show you how to check your handset’s available memory and manage the
handset features to make handset memory available.
Checking the Phone’s Available Memory
You can use two methods to quickly check whether you need to manage your handset’s
memory. Before you begin checking, load and configure the features and files you want to make
available on the handset.
Using the first method, on your handset’s keypad Select Settings> Status> Diagnostics>
Graphs> Memory Usage as shown next.
Use the Memory Usage chart to check what the current Memory Usage amount is. Typically,
you want to ensure that the handset is running at less than 95 percent of its available memory.
If the handset is using more than 95 percent of its available memory, you may need to take
steps to reduce this amount. For information and tips on freeing memory on the handset, see
Managing the Phone Features.
The second method you can use to confirm whether you need to manage your handset’s
memory is to check the app log files. The app log file is enabled by default and is saved to your
provisioning server directory with the MAC address of the handset prepended to the app log file.
For example, if the MAC address of your handset is 00907a0e33b0, the app log file name will
be 00907a0e33b0-app.log.
Open the app log. If you see the message shown next you may need to manage your handset’s
memory resources.
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Figure 11-6: Application Log Error Message
Web Info: Reading the App Log Files
For more information on reading the log files see Technical Bulletin CS-14-20:
Syslog on Spectralink Handsets.
Managing the Phone Features
This section provides tips for managing the handset features to conserve handset memory
resources. This section is especially useful if you are customizing features or using several
advanced features.
If you are using a mixed deployment, such as a combination of Spectralink 84-Series handsets
and Polycom desksets, see Interoperability Guide: Spectralink 84-Series Wireless Telephones
and Polycom Desksets.
All handset features are designed to operate optimally on Spectralink handsets. The features
listed in Table 11-2: Managing the Phone Features are all customizable, advanced features that
can take up significant memory. Use the following table as a reference guide to the amount of
memory a feature can use and for tips on balancing features so that you can optimize the
handset features you want for your deployment.
Table 11-2: Managing the Phone Features
Feature
Typical Memory Size
Browser
Variable. Optimized to display three or four elements.
The browser is optimized to display three or four application elements. If you display complex pages that include
large table or images, try to display a simplified page. If the page cannot be simplified, try reducing the number of
available ringtones or display backgrounds.
Custom Display Image
15KB
The average size of Spectralink display images is 15KB. If you are using custom images, Spectralink recommends
limiting the file size to 15KB for images on the display. If your handset does not display your custom image and the
file size is less than 15KB, try reducing the number of available ringtones or display and image backgrounds.
Local Contact Directory
170 bytes per entry
Spectralink handsets are optimized to display four contact attributes to a maximum of 250 contact entries. Each
entry averages about 170 bytes of memory.
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Feature
Typical Memory Size
If you need more space for the contact directory, try reducing the number of available ringtones or image
backgrounds.
Corporate Directory
Varies by server
The Corporate Directory feature is optimized to display five contact attributes up to a maximum of eight. Because
the corporate directory entries are saved to a server, the size of each entry and the corporate directory as a whole
will vary with the server you are using. If the handset has difficulty displaying directory search results with more than
five attributes, try reducing the number of available ringtones or image backgrounds, or disable the browser.
Ringtones
16KB
Spectralink provides a number of audio files for ringtones that are designed to work correctly with the wireless
handsets. Spectralink ringtones can range in size from 30KB to 125KB. If you want to use custom ringtones,
Spectralink recommends limiting the file size to 16KB. If you want to make more room for custom ringtones, try
disabling the browser, or reduce the number of custom or image backgrounds. If you want to make room for other
features, try reducing the number of available ringtones.
Background Images
8 – 32KB
Spectralink handsets are optimized to display background images of about 50KB. If you want to display background
images having a file size of more than 50KB or make room for more images, try disabling the browser, or reduce the
number of available ringtones. If you want to make room for other features, try reducing the number and size of
available background images.
Phone Interface Language
90KB
The average size of the XML dictionary files for languages that display on the handset’s interface is about 90KB.
Some of these language files use an expanded character set that can increase the file size to 115KB. To conserve
memory resources, Spectralink recommends using only those XML language files for the languages you need.
Web Configuration Utility Interface
250KB
The average size of the languages XML dictionary files for languages that display on the Web Configuration Utility
interface is about 250KB. Some of these language files use an expanded character set that can increase the file
size to 370KB. To conserve memory resources, Spectralink recommends using only those XML language files for
the languages you need.
If you are still having difficulty freeing up sufficient space on your handsets, contact Spectralink
Voice Product Support.
Testing Phone Hardware
You can view diagnostic information from the Diagnostics menu on your handset (Settings>
Status> Diagnostics).
If you select Diagnostics> Test Hardware, you can select one of the following menu items to
perform a hardware diagnostic test:

Audio Diagnostics – test the speaker, microphone, handset, and a third party headset

Keypad Diagnostics – verify the function assigned to each keypad key

Display Diagnostics – test the LCD for faulty pixels

Brightness – test the brightness of the display

LED Diagnostics – test the LED light on your handset

Vibrate – test the vibrate option
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
Accelerometer – test using the Personal Alarms application
Uploading a Phone’s Configuration
Spectralink Software allows the upload of the files representing a handset’s current
configuration. A number of files can be uploaded to the provisioning server, one for every active
source as well as the current non-default configuration set.
You can upload the handset’s configuration from the handset’s menu or through the Web
Configuration Utility.
This is primarily a diagnostics tool to help find configuration errors.
To upload the handset’s current configuration:
1
Navigate to the Upload Configuration menu on the handset (Settings> Advanced
Settings> [enter password]> Administration Settings> Upload Configuration).
2
Choose to upload the configuration from one of All Sources, Configuration Files, Local,
Web or SIP.
3
Press the Upload softkey.
The handset uploads the configuration file to the location that you specify in
prov.configUploadPath. For example, if you select All Sources, a file
<MACaddress>-update-all.cfg is uploaded.
Network Diagnostics
Ping and traceroute are available as diagnostics tools. These diagnostics can be used for
troubleshooting network connectivity problems in the wired and wireless networks.
Both tools are accessible by pressing the Home key and left or right arrows to select Settings>
Status> Diagnostics> Network.
Enter a URL address (for example, http://www.google.com) or any IP address (for example, the
boot server IP address, SIP server IP address or any other handset’s IP address), and then
press the Enter softkey.
Note: When it is in standby, The 84-Series handset will have ping times which seem large. The
handset will go off channel, to scan for other APs, or go to sleep to save battery power. The instandby handset will not respond as quickly as a wired powered network device. When in call
the handset will respond to pings with times often less than 20ms.
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Network Protocols and Ports Used on Spectralink
Handsets
See the next table for a list of the protocols and ports currently used by the Spectralink
Software.
Table 11-3: Protocols & Ports used by Spectralink handsets
Port Number
Protocol
Outgoing
Incoming
21
FTP
Provisioning, Logs
22
SSH
Admin
23
Telnet1
Admin
53
DNS
67
DHCP
Server
UDP
68
DHCP
Client
UDP
69
TFTP
Provisioning, Logs
UDP
80
HTTP
Provisioning, Logs, Pull
Web interface, Poll
TCP
123
NTP
Time Server
UDP
389
LDAP
443
HTTPS
Provisioning, Logs
514
Syslog
Logs
636
LDAP
1719
H.3232
RAS Signaling
RAS Signaling
1720
H.3232
Signaling
Signaling
2222
RTP3
Media Packets
Media Packets
2223
RTCP3
Media Packet Statistics
Media Packet Statistics
5060
SIP
SIP signaling
SIP signaling
5061
SIP over
TLS
Secure signaling
Secure signaling
5070
SIP
SIP signaling
(Nortel CS1K)
SIP signaling
(Nortel CS1K)
7778
OCS
14394
QBC
Signaling
24800
PDC
TCP
Admin
TCP
TCP
UDP
PDC Client messages
HTTP Pull Web interface, HTTP
Push
TCP
QBC Server
TCP
PDC Server messages
TCP
1
Telnet is disabled by default.
2
RTP and RTCP can use any set of even/odd ports between 2222 and 2269. This is configurable by setting
tcpIpApp.port.rtp.mediaPortRangeStart.
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Power and Startup Issues
The following table describes possible solutions to several power and startup issues.
Table 11-4: Troubleshooting Power and Startup Issues
The handset has power issues or the handset has no power.
Determine if the problem is caused by the handset, or the battery. Do one of the following:

Verify that no lights appear on the unit when it is powered up.

Press any key to determine if the display lights up (comes out of battery save mode)

Try another battery

Try a battery from a working handset
The handset will not boot
If your handset will not boot, there may be a corrupt or invalid firmware image or an invalid
configuration on the handset:

Ensure the handset is connected to the WLAN (no red X on the AP signal strength bars)

Ensure that the provisioning server is accessible on the network and a valid software load and
valid configuration files are available.

Ensure that the handset is pointing to the provisioning server on the network. Check Settings>
Status> Platform> Configuration. Boot Server shows IP address. Check to see load protocol
configured (FTP/TFTP/HTTP/HTTPS). Check Config for files which load to the handset.

Reboot the handset.
Battery Pack power specifications
The battery pack voltage measurement triggers the low battery or very low battery alert:
< 3.625 V – Low battery
< 3.500 V – Very low battery (Critically low)
< 3.400 V – Shutdown
Syslog will display battery status with the showBatteryStat tag.
Key Pad Issues
The following table describes possible solutions to issues you may have with the key pad.
Table 11-5: Troubleshooting Key Pad Issues
The key pad does not work
If the dial pad on your handset does not respond, do the following:

Check for a response from other feature keys or from the key pad.

Place a call to the suspect handset from a known working telephone. Check for display
updates. Answer the call. Does the handset work properly in call?

Press the Settings> Status> Diagnostics> Test Hardware> Keypad Diagnostics> Verify: press
a key. Press each key to ensure each key is recognized properly.

If the keypad seems defective contact your help desk, the reseller or Spectralink support.
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Screen and System Access Issues
The following table describes possible solutions to screen and system access issues.
Table 11-6: Troubleshooting Screen and System Access Issues
There is no response from feature key presses
If your handset is not in the active state, do one of the following:

Press the keys more slowly.

Check to see whether or not the key has been mapped to a different function or disabled.

Make a call to the handset to check for inbound call display and ringing. If successful, try to
press feature keys while a call is active to access a Directory, for example.

Navigate to Settings> Status> Lines to confirm the line is actively registered to the call
server.

Reboot the handset to attempt re-registration to the call server (see Rebooting the Phone).
The display shows the message Network Link is Down

Ping the handset from another machine.

Reboot the handset to attempt re-registration to the call server (navigate to Settings> Basic
Settings> Restart Phone).
Calling Issues
The following table provides possible solutions to a number of generic calling issues.
Table 11-7: Troubleshooting Calling Issues
The handset does not ring
If there is a no ring tone, but the handset displays a visual indication when it receives an incoming call,
do the following:

Adjust the ring level using the volume up/down keys.

Check the status of handset, headset (if connected) and through the Handsfree
Speakerphone.
The line icon shows an unregistered line
If you see unregistered line icon the handset line is unregistered. Register the line and try to place a
call.
Unregistered Line Icon:
Registered Line Icon:
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Display Issues
The following table provides tips for resolving display screen issues.
Table 11-8: Troubleshooting Display Issues
There is no display or the display is incorrect
If there is no display, power may not be correctly supplied to the handset. Do one of the following:

Check that the display is illuminated. Press any key to wake up the display.

Power up the handset. Ensure the battery used is charged.

Test the display by Settings> Status> Diagnostics> Test Hardware> Display Diagnostics.
Watch the test run to see the display change. Press any key to exit the test

Use the screen capture feature to determine if the display on the handset is incorrect. See
Capturing the Phone’s Current Screen.
The display is too dark or too light
The handset contrast may be set incorrectly. To adjust the contrast, do one of the following:

Adjust the backlight intensity. Navigate to Settings> Basic Settings> Backlight Intensity
settings.

Reboot the handset to obtain the default level of contrast (see Rebooting the Phone).

Use the screen capture feature to see if the screen displays properly in the capture. See
Capturing the Phone’s Current Screen.
The time and date are flashing
If the time and date are flashing, you have disconnected the handset from the LAN or there is no SNTP
time server configured. Do one of the following (for instructions, see Time and Date Display):

Ensure the handset is connected to the WLAN and to the LAN.

Configure an SNTP server.

Disable the time and date (if you do not wish to connect your handset to a LAN or SNTP
server).
Audio Issues
The next table briefly describes possible solutions to audio issues.
Table 11-9: Troubleshooting audio issues
There is no audio on the headset
If there is no audio on your headset, the connections may not be correct. Do one of the following:

Ensure the headset is plugged into the headset jack on the handset.

Ensure the handset volume is turned up or set to a comfortable volume.
There are audio or echo issues
If you experience echo issues, investigate the issue to determine the following:
 Is the problem localized to one 84-Series handset?
 Are other 84-Series handsets experiencing audio issues?
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


Is the problem related to one area of the facility?
Check SIP server settings which may be applicable to the audio or echo issues.
See Technical Bulletin Troubleshooting Audio and Echo Issues
There is choppy/poor audio
Poor audio can be caused by several things including:
 Inadequate RSSI (RF Signaling from the AP)
 High retry rates
 High Missed packet rates
 High Dropped packet rates
 High Late packet rates
 Loss of connection to the WLAN
 Long roam times between APs
 Congestion on the network between the 84-Series handset, SIP server or other end point
Additionally…
Logging data is useful for later syslog data analysis with render =1 and WLAN = 1 along with
Call start and end times and symptom descriptions at the logging level for WLAN at 2.
Another troubleshooting technique is to use Diagnostics mode to check for many of the above
items during the call. Go to Settings> Status> Diagnostics> WiFi Stats>

[1/6] Packet Count to see Missed RX, Missed TX packets, RX Retry count and TX Retry
count. This screen shows cumulative packet counts not percentage. If any of these
counters are incrementing frequently it is a bad sign.

[2/6] General Information shows SSID, Last TX rate and TX power

[3/6] AP List shows last few bytes of the AP MAC, channel and RSSI (a lower number (45 versus -75 is a stronger signal)

[4/6] Re/Association Count shows Association count, Reassociation count, Association
fail count and Reassociation fail count

[5/6] Security shows MIC Err Count, MIC Err Last Seq , ICV Err Count and QoS Fail
Count

6[6/6] EAP Information shows EAP Err Count, Last EAP Err Code and 802.11n disabled
When in call, focus on screens1 and 3 to see how packet counts seem to be affected by low
(inadequate) RSSI
Upgrading Issues
Tip: Important!
When upgrading handset software using the Web Configuration Utility, the handset
is unable to connect to the Spectralink Hosted Server.
Occasionally, the handset is unable to connect to the Spectralink Hosted Server because:
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 The Spectralink Hosted Server is temporarily unavailable.
 There isn’t any software upgrade file for the handset to download.
 The network configuration is preventing the handset from connecting to the Spectralink
Hosted Server. Check for a firewall, ACL or another issue preventing the handset from
loading code from the provisioning server.
Note: Cannot upgrade through a Web proxy
Spectralink Software does not support internet access for software upgrades
through a Web proxy.
To troubleshoot an upgrade issue:
 Verify that new software is available for your handset.
 Verify that your network’s configuration will allow the handset to connect to
http://downloads.Spectralink.com.
 Check the web site with a computer to verify it is available and the file can download
 Try upgrading the code again
 Use a local provisioning server
If the issue persists, try manually upgrading your handset’s software. To upgrade handset
software using this method, see Setting Up the Provisioning Server.
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Chapter 13: Miscellaneous Maintenance
Tasks
This chapter shows you how to maintain the Spectralink Software. This includes:

Trusted Certificate Authority List

Encrypting Configuration Files

Multiple Key Combinations

Default Feature Key Layouts

Parsing Vendor ID Information

Product, Model, and Part Number Mapping

Capturing the Phone’s Current Screen
Encrypting Configuration Files
For security reasons, an administrator may wish to encrypt configuration files sent to the phone
from a provisioning server and files uploaded by the phone to the provisioning server. Setting up
dynamic encryption involves generating a key, downloading it to the phone, encrypting all config
files used by the phone and making these files available to the phone by referencing them in the
master config file which itself is not encrypted.
Note: Not all config files can be encrypted
Note that the master configuration file or the bootrom.ld or sip.ld software files cannot
be encrypted.
All exe files referred to in this section can be found on the 84-Series webpage under the
Downloads tab in the ConfigFileEncryption_v[x].zip. Our config file examples use the
configuration templates provided with the software.
Ensure that you securely store all files generated by this procedure.
Admin Tip: Using Cygwin or Unix?
The configuration steps and examples below assume you are using DOS. If you
are using Cygwin or Unix, make the appropriate adjustments to the code.
Configuration steps
1
Generate a key.
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Note: ConfigFileKenGen.exe requires a random seed file
If using the DOS version of the key generator you can create the “c:\.rnd” and
populate it with random data of your choosing.
If compiled under a Linux or Unix system (generally including Cygwin), it should
default to /dev/random and there will be no extra steps required. Some installations
of Cygwin or Unix do not have a /dev/random device installed. For this situation
you can create the .rnd file and populate it with random data of your choosing.
Use the configFileKeyGen.exe file to generate the key. To generate a key named
“key1.key” use this command format:
> configFileKeyGen.exe -k key1.key -d hello
The command has these components:
»
-k [the key filename]
»
-d [the key description]
Your generated key (key1) is a string that looks like this:
Crypt=1;KeyDesc=hello;Key=93750C896A35F74EF704CEA66CC89049;
The string contains three attributes:
»
The type of encryption - 128bit AES encryption [Crypt=1]
»
A description of the key [KeyDesc=hello]
»
The key itself [Key=93750C896A35F74EF704CEA66CC89049]
Admin Tip: Problems with key generation?
Try generating a key using “hello” as the key description as shown in our example.
2
Create an XML .cfg file for the key. Our example is named key1.cfg.
Use the device parameters in the next table:
Table 12-1: Encryption key parameters
Parameter
Permitted Values
Default
device.set
0 or 1
0
[the key]
Null
Set to 1 to enable encryption
device.sec.configEncryption.key
The key that is generated by the process described in this section. Note that the KeyDesc must be “hello”.
device.sec.configEncryption.key.set
0 or 1
0
Set to 1 in order to use the key set with device.sec.configEncryption.key
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Example: config file (key1.cfg) for loading the key into the handsets
Caution: Use the entire string produced by the key generator
Ensure that you include the full string (including the “Crypt” and “KeyDesc” fields)
rather than just the key. Also include the semicolon at the end of the key. The
phone will fail to read the key if you omit any part of the string.
3
Load the key in the phones. This should be done securely through a USB connection or
within a secure wireless lab. The assumption here is that this is part of the initial
deployment, all phones are available, config files are developed, and the QNC/SLIC
configuration is complete.
Whether connected through usb or through the secure LAN, the phone will find the
server and look for the master config file. Therefore the key file must be referenced in
the master config file, as shown below, and both the key.cfg and master config files must
be loaded on the provisioning server.
A master configuration file referencing the key1.cfg file
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a
Place the master config file and the key config file onto the provisioning server that
the handset is programmed (through QNC/SLIC or manually) to access.
b
Plug in (or turn on) the phone and allow it to find and load the file. Do this for all your
phones.
Once this is done the phones will be able to decrypt any files that use this key. They
will also encrypt files uploaded to the server if you configure it this way. (per step 4)
However, they do not have the encrypted config files yet so they will not be able to
make calls, etc., just yet.
c
Once the key has loaded to the phone, the key1.cfg file is no longer necessary and
the master config file must be edited to include the encrypted config files. For
greatest security you can remove these two files from the server and store them
securely for later use and configure a new master config file as described in step 6.
You can also comment out the commands or change the device.set parameter to
zero. But be aware that the master config file cannot be encrypted.
4
(Optional) Add <sec.encryption> parameters to phones’ site.cfg config file (or other
config file that all encrypted phones will access).
The <sec.encryption> parameters specify whether the phone uploads its
configuration files in an encrypted or unencrypted format. See Configuration File
Encryption for exact information.
5
Encrypt the config files used by the phones.
Use the configFileEncrypt.exe program to encrypt the config files used by the
phones.
We recommend that you rename the encrypted file so that it is obvious it is encrypted.
For example, here we rename the site.cfg file to site-cfg.enc when we encrypt it using
the encryption command. Use the same key name in the command that you just loaded
on the phones! This is the command format:
> configFileEncrypt.exe -i site.cfg -o site-cfg.enc1 -k key1.key
The command has these components:
»
-i [the un-encrypted filename]
»
-o [the encrypted filename]
»
-k [the key filename]
Run configFileEncrypt.exe on each config file used by the phones, except of
course, the master config file.
Securely store the unencrypted files, just as you stored the key file in step 3c.
Caution: If using a Linux computer to generate the encrypted files
Check the encrypted files to ensure they are not altered when you copy them from
a computer running the Linux operating system to a computer running the Microsoft
Windows operating system. See Comparing encrypted and unencrypted files.
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6
Load the encrypted config files into your master configuration file. See the next example.
A master configuration file with encrypted config files referenced:
7
The phones are now ready to deploy.
Comparing encrypted and unencrypted files
8
Run configFileEncrypt.exe on the unencrypted file with the "-d" option. This shows
the "digest" field. E.g.:
> configFileEncrypt.exe -i site.cfg -d
The command has these components:
9
○
-i [the un-encrypted filename]
○
-d [the digest field request]
View the Digest= line in the resulting file.
10
Look at the encrypted file using text editor and check the "Digest=…." field.
11
If the two fields are the same, then the encrypted and un-encrypted file are the same.
Decrypting existing configuration files
The configFileDecrypt.exe program can be used to decrypt an encrypted file as long as
the key filename is included in the command. Use the following format:
> configFileDecrypt.exe -i site-cfg.enc1 -o site.cfg -k key1.key
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The command has these components:

-i [the encrypted filename]

-o [the decrypted filename]

-k [the key filename]
Changing an existing key
Changing to a new key is a multi-step process. This is best done by gathering up all the phones
and doing them all at once. Phones that are in use will experience a period of disruption as the
key is swapped out. Use the following steps:
1
Generate a new key following step 1 above and save the file. We’ll call it key2.key.
2
Create a new .cfg file with the new key. Save it with some way of differentiating it from
the previous key config file. Use step 2 above for guidance. We will use key2.cfg.
3
Use the old key (key1) to encrypt the key2.cfg file per step 5 above.
4
Load the new key into the phone. This time you can do this wirelessly as key2.cfg is
already encrypted with key1.
a
Create a master config file that references this key2.cfg file.
b
Replace the master config file that the phone has been using with this new one.
The phone will load the key2.cfg file containing the new key by using the old key still on
the phone. The phone will then replace the old key (key1) with the new key (key2).
At this point, the phone can no longer read the config files encrypted with key1.
5
Use key2 to generate new versions of the encrypted config files. You will need to decrypt
each file with the old key and then encrypt each file with the new key. Use a new name
for them so that they can easily be differentiated from the files that use the old key. See
step 5 above.
6
In the master configuration file, replace the configuration files that used the old key with
the configuration files that use the new key. Load the master configuration file and all the
config files using the new key on the provisioning server.
7
Reboot the phones. They should now be able to read the new files.
Log messages
You can look in the app log files for logging related to the configuration files.
If a file cannot be decrypted, messages similar to this will appear in the log:
000014.938|so |*|00|Configuration files: 00907a112233-cfg.enc,
pbx_4053.enc 000014.938|so |*|00|Configuration file
"[PHONE_MAC_ADDRESS]-ext-cfg.enc1" SHA1 digest: Unknown
000014.938|so |*|00|Configuration file "site-cfg.enc1" SHA1
digest: Unknown
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In particular, note the presence of the “Unknown” as the SHA1 digest. Additionally, logs like
this may be present:
0804163651|cfg |5|00|Prm|Could not decrypt site-cfg.enc1
When the config files are properly decrypted, the messages like this should appear:
000134.739|so |*|00|Configuration files: [PHONE_MAC_ADDRESS]-extcfg.enc1, site-cfg.enc1 000134.739|so |*|00|Configuration file
"00907a112233-cfg.enc" SHA1 digest:
DB944E56E0413904353A2CD1A0FA29BF69E2AC1E
000134.739|so |*|00|Configuration file " site-cfg.enc1 " SHA1
digest: B4465B48A226DE487445A1519F0D566CDC07BBB3
In addition, messages similar to the following may be present, indicating the file was read and
parameters were parsed:
000134.739|cfg |3|00|Prm|Configuration file(s) statistics: 20
valid parameters found. 000134.739|cfg |3|00|Prm|Configuration
file(s) statistics: 14 parameter values have been used. 6
parameter values were set more than once
Troubleshooting: My phone keeps displaying an error message for my
encrypted file
If a handset downloads an encrypted file that it cannot decrypt, the action is logged,
and an error message displays. The handset will continue to do this until the
provisioning server provides an encrypted file that can be read, an unencrypted file,
or the file is removed from the master configuration file list.
Multiple Key Combinations
You can use multiple key combinations on your Spectralink handsets to reboot the handset, to
restore the handset to factory default values, or to upload log files from the handset to your
provisioning server.
Rebooting the Phone
Rebooting the handset downloads new software and new configuration files if they exist on the
provisioning server.
Timesaver: Download new configuration files without rebooting your phone
Not all configuration parameter changes require the handset to restart or reboot.
You can update your handset’s configuration by navigating to Settings> Basic
Settings> Update Configuration. If there is new software (different version) on
the provisioning server, the handset will restart or reboot to download the software.
If there are configuration file changes, your handset will only restart if it is
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necessary. Otherwise, the handset will download the new configuration files without
restarting.
You can use a multiple key combination to reboot your handset. Press and hold the 0, 1, and 3
keys simultaneously until you hear a confirmation tone (for about three seconds).
Power Tip: Quickly restarting your phone
Users can restart their handsets by pressing the Home key and selecting
Settings> Basic Settings> Restart Phone. If new Updater or Spectralink
Software is available on the provisioning server, the handset will download the
software when it restarts.
Resetting to factory defaults
Resetting the handset to factory defaults clears the flash parameters and removes log files, user
data, and cached data.
You can use a multiple key combination to reset your handset to the factory defaults. Press and
hold the 1, 3, and 5 dial pad keys simultaneously during the Updater/BootROM countdown
process until the password prompt displays.
Enter the administrator password to initiate the reset. Resetting to factory defaults will also reset
the administrator password (factory default password is 456). Spectralink recommends that you
change the administrative password from the default value.
Updating log files
Uploading the log files copies the log files from the handset to the provisioning server. The files
called <MACaddress>-now-xxx.log are created. Xxx is boot or app.
You can use a multiple key press to upload log files to your provisioning server. Press and hold
the 1, 5, and 9 dial pad keys simultaneously until you hear a confirmation tone (for about three
seconds).
Setting base profile
Setting the base profile allows for quick setup of Spectralink handsets with Microsoft Skype for
Business or Microsoft Lync Server 2013 or 2010.
You can use a multiple key combination to set the base profile on a particular Spectralink
handset. Depending on your handset model, press and hold the 1, 4, and 9 dial pad keys
simultaneously for about three seconds until you hear a confirmation tone.
A login screen displays. Enter the administrator password (default 456) to initiate the setup.
Spectralink recommends that you change the administrative password from the default value.
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Default Feature Key Layouts
The following figures and tables show the default key layouts for the Spectralink 84-Series
wireless handsets.
The illustration of the Spectralink handsets is followed by a table that shows the available
handset key functions.
Spectralink 84-Series
Table 12-2: Spectralink 8440 and 8450 handset key functions
Key ID
Function
Key ID
Function
Key ID
Function
1
Dialpad1
12
VolUp
23
ArrowRight
2
Dialpad2
13
DialpadPound
24
ArrowDown
3
Dialpad3
14
Green
25
Select
4
Dialpad4
15
DialpadStar
26
Home
5
Dialpad5
16
SoftKey1
27
Back
6
Dialpad6
17
SoftKey2
28
Red
7
Dialpad7
18
SoftKey3
29
Barcode
8
Dialpad8
19
SoftKey4
9
Dialpad 9
20
Talk
10
Dialpad0
21
ArrowUp
11
VolDown
22
ArrowLeft
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Parsing Vendor ID Information
After the handset boots, it sends a DHCP Discover packet to the DHCP server. This is found in
the Bootstrap Protocol/option ‘Vendor Class Identifier’ section of the packet and includes the
handset’s hardward ID and the BootROM version. RFC 2132 does not specify the format of this
option's data which can be defined by each vendor. To be useful, every vendor's format must be
distinguishable from every other vendor's format. To make our format uniquely identifiable, the
format follows RFC 3925, which uses the IANA Private Enterprise number to determine which
vendor's format should be used to decode the remaining data. The private enterprise number
assigned to Spectralink is 13885 (0x0000363D).
This vendor ID information is not a character string, but an array of binary data.
The steps for parsing are as follows:
1
Check for the Spectralink signature at the start of the option:
4 octet: 00 00 36 3d
2
Get the length of the entire list of sub-options:
1 octet
3
Read the field code and length of the first sub-option, 1+1 octets
4
If this is a field you want to parse, save the data.
5
Skip to the start of the next sub-option.
6
Repeat steps 3 to 5 until you have all the data or you encounter the End-of-Suboptions
code (0xFF).
For example, the following is a sample decode of a packet from an IP 601:
3c 74
○ Option 60, length of Option data (part of the DHCP spec.)
00 00 36 3d
○
Spectralink signature (always 4 octets)
6f
○ Length of Spectralink data
01 07 50 6f 6c 79 63 6f 6d
○ sub-option 1 (company), length, "Spectralink"
02 15 53 6f 75 6e 64 50 6f 69 6e 74 49 50 2d 53 50 49 50 5f 36 30 31
○ sub-option 2 (part), length, "SoundPointIP-SPIP_601"
03 10 32 33 34 35 2d 31 31 36 30 35 2d 30 30 31 2c 32
○ sub-option 3 (part number), length, "2345-11605-001,2"
04 1c 53 49 50 2f 54 69 70 2e 58 58 58 58 2f 30 38 2d 4a 75 6e 2d 30
37 20 31 30 3a 34 34
○
sub-option 4 (Application version), length, "SIP/Tip.XXXX/08-Jun-07 10:44"
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05 1d 42 52 2f 33 2e 31 2e 30 2e 58 58 58 58 2f 32 38 2d 41 70 72 2d
30 35 20 31 33 3a 33 30
○ sub-option 5 (BootROM version), length, "BR/3.1.0.XXXX/28-Apr-05
13:30"
ff
○
end of sub-options
For the Updater, sub-option 4 and sub-option 5 will contain the same string. The string is
formatted as follows:
<apptype>/<buildid>/<date+time>
where:
<apptype> can be 'BR' (BootROM) or 'SIP' (SIP Application)
Product Model Number and Hardware ID Mapping
The master configuration file can direct handset upgrades to a software image and configuration
files based on a handset model name, a firmware hardware ID, or a single handset’s MAC
address.
The hardware ID has precedence over the model name, which has precedence over the default
CONFIG_FILES parameter. For example,
CONFIG_FILES_3111-36150=001=
”phone1_3111-36150-001.cfg, sip_3111-36150-001.cfg”
will override
CONFIG_FILES_SL8450=
”phone1_SL8450.cfg, sip_SL8450.cfg”,
which will override
CONFIG_FILES=
”phone1.cfg, sip.cfg”
for a Spectralink 8450.
You can also add variables to the master configuration file that are replaced when the handset
reboots. The variables include PHONE_MODEL, PHONE_PART_NUMBER, and
PHONE_MAC_ADDRESS.
Table 12-3: Model Name and Hardware ID
Model Name
Hardware ID
SL8440
SL8450
SL8452
SL8441
SL8453
3111-36150-001
3111-36152-001
3111-36154-001
3111-67360-001
3111-67361-001
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Capturing the Phone’s Current Screen
You can capture your handset’s current screen using a Web browser.
Troubleshooting: I Can’t Take a Screen Capture of the Spectralink Site
Survey Screen
You will not be able to take screen captures of the site survey screens on the
Spectralink handsets as the network connection is disabled while site survey is
running.
To capture the handset’s current screen:
1
Modify your configuration file to enable the screen capture feature.
You will need to open your configuration file in an XML editor and add the following line:
2
Save the configuration file and update your handset’s configuration.
3
On the handset, turn on the screen capture feature from the Screen Capture menu
(Settings> Basic> Preferences> Screen Capture).
You will need to turn the screen capture on again (repeat this step) each time the
handset restarts or reboots.
4
In a Web browser, enter http://<handsetIPaddress>/captureScreen in the browser
address field. (To find your handset’s IP address, navigate to Settings> Status>
Platform> Phone.)
5
Enter the username and password, as needed.
6
The Web browser will display an image showing the handset’s current screen. The
image can be saved as a BMP or JPEG file. Right-click the image and save accordingly.
7
Refresh your browser to display the current screen on the handset.
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Part V: Appendices
Provides reference information about Ringtone Pattern Names and Sound Effects, Session
Initiation Protocol (SIP), and information about the third-party software that is included in the
84-Series software.
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Appendix A: Ringtone Pattern Names and
Sound Effects Parameters
Ringer Patterns
The following table shows the ring pattern names and their default descriptions:
Parameter
Name
Ring Type number
Ringtone Name
Description
ringer1
na
Silent Ring
Silent ring
ringer2
1
Low Trill
Long single A3 Db3 major warble
ringer3
2
Low Double Trill
Short double A3 Db3 major warble
ringer4
3
Medium Trill
Long single C3 E3 major warble
ringer5
4
Medium Double
Trill
Short double C3 E3 major warble
ringer6
5
High Trill
Long single warble 1
ringer7
6
High Double Trill
Short double warble 1
ringer8
7
Highest Trill
Long single Gb3 A4 major warble
ringer9
8
Highest Double
Trill
Short double Gb3 A4 major warble
ringer10
9
Beeble
Short double E3 major
ringer11
10
Triplet
Short triple C3 E3 G3 major ramp
ringer12
11
Ringback-style
Short double ringback
ringer13
12
Low Trill
Precedence
Long single A3 Db3 major warble Precedence
ringer14
13
Ring Splash
Splash
ringer15
14
Ring16
Sampled audio file 1
ringer16
15
Ring17
Sampled audio file 2
ringer17
16
Ring18
Sampled audio file 3
ringer18
17
Ring19
Sampled audio file 4
ringer19
18
Ring20
Sampled audio file 5
ringer20
19
Ring21
Sampled audio file 6
ringer21
20
Ring22
Sampled audio file 7
ringer22
21
Ring23
Sampled audio file 8
ringer23
22
Ring24
Sampled audio file 9
ringer24
23
Ring25
Sampled audio file 10
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Note: Using the Answer Ring Type
The auto-answer on incoming call is currently only applied if there is no other call in
progress on the handset at the time.
Note: Silent Ring
Silent ring will provide a visual indication of an incoming call, but no audio
indication.
Sampled audio files 1 to 10 all use the same built-in file unless that file has been
replaced with a downloaded file.
Ring Tones <rt/>
Ringtone is used to define a simple class of ring to be applied based on some credentials that
are usually carried within the network protocol. The ring class includes parameters such as callwaiting and ringer index, if appropriate. The ring class can use one of four types of ring that are
defined as follows:
ring Play a specified ring pattern or call waiting indication
visual Provide only a visual indication (no audio) of an incoming call, no ringer needs to be
specified
answer Provide auto-answer on an incoming call
ring-answer Provide auto-answer on an incoming call after a certain number of rings
The handset supports the following ring classes: default, visual, answerMute, autoAnswer,
ringAnswerMute, ringAutoAnswer, internal, external, emergency, precedence, splash,
and custom<y> where y is 1 to 17.
In the following table, x is the ring class name.
Sound Effects Ringtone Parameters
Parameter
Permitted Values
se.rt.enabled
0 or 1 (default)
If 0, the ringtone feature is not enabled on the handset. If 1 (default), the ringtone feature is enabled.
se.rt.modification.enabled
0 or 1 (default)
A flag to determine whether or not to allow user modification (through handset’s user interface) of the pre-defined
ringtone enabled for modification.
se.rt.<ringClass>.callWait
callWaiting, callWaitingLong,
precedenceCallWaiting
The call waiting tone to be used for this class of ring. The call waiting should match one defined in Table 14-591:
Call Progress Tone Pattern Names. The default call waiting tone is callWaiting.
se.rt.<ringClass>.name
UTF-8 encoded string
The answer mode for a ringtone. Used for identification purposes in the user interface.
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Parameter
Permitted Values
se.rt.<ringClass>.ringer
default, ringer1 to ringer24
The ringtone to be used for this class of ring. The ringer should match one of the Ringtone Pattern Names in the
above table. The default ringer is ringer2 .
se.rt.<ringClass>.timeout
1 to 60000 only relevant if the type is set to ringanswer
The duration of the ring in milliseconds before the call is auto answered. The default is 2000.
se.rt.<ringClass>.type
ring, visual, answer, ring-answer
The answer mode for a ringtone as defined in list earlier in this section.
Miscellaneous Patterns
The following table shows the miscellaneous patterns and their descriptions:
Miscellaneous Pattern Names
Miscellaneous pattern name
Description
instant message
New instant message
local hold notification
Local hold notification
message waiting
New message waiting indication
negative confirmation
Negative confirmation
positive confirmation
Positive confirmation
remote hold notification
Remote hold notification
welcome
Welcome (boot up)
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Appendix B: Session Initiation Protocol
(SIP) Information
This chapter describes the basic Session Initiation Protocol (SIP) and the protocol extensions
that the current Spectralink Software supports.
This chapter contains information on:

Basic Protocols—All the basic calling functionality described in the SIP specification is
supported. Transfer is included in the basic SIP support.

Protocol Extensions—Extensions add features to SIP that are applicable to a range of
applications, including reliable 1xx responses and session timers.
For information on supported RFCs and Internet drafts, see the following section.
This chapter also describes:

Request Support

Header Support

Response Support

Hold Implementation

Reliability of Provisional Responses

Transfer

Third party call control

SIP for Instant Messaging and Presence
RFC and Internet Draft Support
The following RFC’s and Internet drafts are supported. For more information on any of the
documents, enter the RFC number at http://www.ietf.org/rfc.html.

RFC 1321—The MD5 Message-Digest Algorithm

RFC 2327—SDP: Session Description Protocol

RFC 2387—The MIME Multipart / Related Content-type

RFC 2976—The SIP INFO Method

RFC 3261—SIP: Session Initiation Protocol (replacement for RFC 2543)

RFC 3262—Reliability of Provisional Responses in the Session Initiation Protocol (SIP)

RFC 3263—Session Initiation Protocol (SIP): Locating SIP Servers

RFC 3264—An Offer / Answer Model with the Session Description Protocol (SDP)
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
RFC 3265—Session Initiation Protocol (SIP) - Specific Event Notification

RFC 3311—The Session Initiation Protocol (SIP) UPDATE Method

RFC 3325—SIP Asserted Identity

RFC 3420—Internet Media Type message/sipfrag

RFC 3515—The Session Initiation Protocol (SIP) Refer Method

RFC 3555 — MIME Type of RTP Payload Formats

RFC 3611 — RTP Control Protocol Extended reports (RTCP XR)

RFC 3665—Session Initiation Protocol (SIP) Basic Call Flow Examples

draft-ietf-sip-cc-transfer-05.txt—SIP Call Control - Transfer

RFC 3725—Best Current Practices for Third Party Call Control (3pcc) in the Session
Initiation Protocol (SIP)

RFC 3842—A Message Summary and Message Waiting Indication Event Package for
the Session Initiation Protocol (SIP)

RFC 3856—A Presence Event Package for Session Initiation Protocol (SIP)

RFC 3891—The Session Initiation Protocol (SIP) “Replaces” Header

RFC 3892—The Session Initiation Protocol (SIP) Referred-By Mechanism

RFC 3959—The Early Session Disposition Type for the Session Initiation Protocol (SIP)

RFC 3960—Early Media and Ringing Tone Generation in the Session Initiation Protocol
(SIP)

RFC 3968—The Internet Assigned Number Authority (IANA) Header Field Parameter
Registry for the Session Initiation Protocol (SIP)

RFC 3969—The Internet Assigned Number Authority (IANA) Uniform Resource Identifier
(URI) Parameter Registry for the Session Initiation Protocol (SIP)

RFC 4028—Session Timers in the Session Initiation Protocol (SIP)

RFC 4235—An INVITE-Initiated Dialog Event Package for the Session Initiation Protocol
(SIP)

draft-levy-sip-diversion-08.txt—Diversion Indication in SIP

draft-anil-sipping-bla-02.txt—Implementing Bridged Line Appearances (BLA) Using
Session Initiation Protocol (SIP)

draft-ietf-sip-privacy-04.txt—SIP Extensions for Network-Asserted Caller Identity and
Privacy within Trusted Networks

draft-ietf-sipping-cc-conferencing-03.txt—SIP Call Control - Conferencing for User
Agents

draft-ietf-sipping-rtcp-summary-02.txt —Session Initiation Protocol Package for Voice
Quality Reporting Event
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
draft-ietf-sip-connect-reuse-04.txt—Connection Reuse in the Session Initiation Protocol
(SIP)
Request Support
The following SIP request messages are supported:
Supported SIP Request Messages
Method
Supported
Notes
REGISTER
Yes
INVITE
Yes
ACK
Yes
CANCEL
Yes
BYE
Yes
OPTIONS
Yes
SUBSCRIBE
Yes
NOTIFY
Yes
REFER
Yes
PRACK
Yes
INFO
Yes
RFC 2976, the handset does not generate INFO requests, but
will issue a final response upon receipt. No INFO message
bodies are parsed.
MESSAGE
Yes
Final response is sent upon receipt. Message bodies of type
text/plain are sent and received.
UPDATE
Yes
Header Support
The following SIP request headers are supported:
Note: Reading the following tables
In the following table, a Yes in the Supported column means the header is sent and
properly parsed.
Supported SIP Request Headers
Header
Supported
Accept
Yes
Accept-Encoding
Yes
Accept-Language
Yes
Accept-Resource-Priority
Yes
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Header
Supported
Access-Network-Info
No
Access-URL
Yes
Alert-Info
Yes
Allow
Yes
Allow-Events
Yes
Authentication-Info
Yes
Authorization
Yes
Call-ID
Yes
Call-Info
Yes
Contact
Yes
Content-Disposition
Yes
Content-Encoding
Yes
Content-Language
Yes
Content-Length
Yes
Content-Type
Yes
CSeq
Yes
Date
Yes (for missed call, not used to adjust the time of the
handset)
Diversion
Yes
Error-Info
No
Event
Yes
Expires
Yes
Flow-Timer
Yes
From
Yes
In-Reply-To
No
Join
Yes
Max-Forwards
Yes
Min-Expires
Yes
Min-SE
Yes
MIME-Version
No
Missed-Calls
Yes
ms-client-diagnostics
Yes
ms-keep-alive
Yes
ms-text-format
Yes
Organization
No
P-Asserted-Identity
Yes
P-Preferred-Identity
Yes
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Header
Supported
Priority
No
Privacy
No
Proxy-Authenticate
Yes
Proxy-Authorization
Yes
Proxy-Require
Yes
RAck
Yes
Reason
Yes
Record-Route
Yes
Refer-Sub
Yes
Refer-To
Yes
Referred-By
Yes
Referred-To
Yes
Remote-Party-ID
Yes
Replaces
Yes
Reply-To
No
Requested-By
No
Require
Yes
Resource-Priority
Yes
Response-Key
No
Retry-After
Yes
Route
Yes
RSeq
Yes
Server
Yes
Session-Expires
Yes
SIP-Etag
Yes
SIP-If-Match
Yes
Subject
Yes
Subscription-State
Yes
Supported
Yes
Timestamp
Yes
To
Yes
Unsupported
Yes
User-Agent
Yes
Via
Yes
voice-missed-call
Yes
Warning
Yes (Only warning codes 300 to 399)
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Header
Supported
WWW-Authenticate
Yes
X-Sipx-Authidentity
Yes
Response Support
The following SIP responses are supported:
Note: Reading the following tables
In the following table, a Yes in the Supported column means the header is sent and
properly parsed. The handset may not actually generate the response.
1xx Responses - Provisional
Supported 1xx SIP Responses
Response
Supported
100 Trying
Yes
180 Ringing
Yes
181 Call Is Being Forwarded
No
182 Queued
No
183 Session Progress
Yes
2xx Responses - Success
Supported 2xx SIP Responses
Response
Supported
200 OK
Yes
202 Accepted
Yes
Notes
In REFER transfer.
3xx Responses - Redirection
Supported 3xx SIP Responses
Response
Supported
300 Multiple Choices
Yes
301 Moved Permanently
Yes
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Response
Supported
302 Moved Temporarily
Yes
305 Use Proxy
No
380 Alternative Service
No
4xx Responses - Request Failure
Handling 4xx responses
All 4xx responses for which the handset does not provide specific support will be
treated the same as 400 Bad Request.
Supported 4xx SIP Responses
Response
Supported
400 Bad Request
Yes
401 Unauthorized
Yes
402 Payment Required
No
403 Forbidden
No
404 Not Found
Yes
405 Method Not Allowed
Yes
406 Not Acceptable
No
407 Proxy Authentication Required
Yes
408 Request Timeout
No
410 Gone
No
413 Request Entity Too Large
No
414 Request-URI Too Long
No
415 Unsupported Media Type
Yes
416 Unsupported URI Scheme
No
420 Bad Extension
No
421 Extension Required
No
423 Interval Too Brief
Yes
480 Temporarily Unavailable
Yes
481 Call/Transaction Does Not Exist
Yes
482 Loop Detected
Yes
483 Too Many Hops
No
484 Address Incomplete
Yes
485 Ambiguous
No
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Response
Supported
486 Busy Here
Yes
487 Request Terminated
Yes
488 Not Acceptable Here
Yes
491 Request Pending
No
493 Undecipherable
No
5xx Responses - Server Failure
Supported 5xx SIP Responses
Response
Supported
5-Server Internal Error
Yes
501 Not Implemented
Yes
502 Bad Gateway
No
503 Service Unavailable
No
504 Server Time-out
No
505 Version Not Supported
No
513 Message Too Large
No
6xx Responses - Global Failure
Supported 6xx SIP Responses
Response
Supported
600 Busy Everywhere
No
603 Decline
Yes
604 Does Not Exist Anywhere
No
606 Not Acceptable
No
Hold Implementation
The handset supports two currently accepted means of signaling hold.
The first method, no longer recommended due in part to the RTCP problems associated with it,
is to set the “c” destination addresses for the media streams in the SDP to zero, for example,
c=0.0.0.0.
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The second, and preferred, method is to signal the media directions with the “a” SDP media
attributes sendonly, recvonly, inactive, or sendrecv. The hold signaling method used by the
handset is configurable is configurable using the voIpProt.SIP.useRFC2543hold
parameter (see Call Hold), but both methods are supported when signaled by the remote end
point.
Hold Methods
Even if the handset is set to use c=0.0.0.0, it will not do so if it gets any sendrecv,
sendonly, or inactive from the server. These flags will cause it to revert to the other
hold method.
Reliability of Provisional Responses
The handset fully supports RFC 3262 - Reliability of Provisional Responses.
Transfer
The handset supports transfer using the REFER method specified in draft-ietf-sip-cc-transfer-05
and RFC 3515.
Third party call control
The handset supports the delayed media negotiations (INVITE without SDP) associated with
third-party call-control applications.
When used with an appropriate server, the User Agent Computer Supported
Telecommunications Applications (uaCSTA) feature on the handset may be used for remote
control of the handset from computer applications such as Microsoft Office Communicator.
The handset is compliant with “Using CSTA for SIP Phone User Agents (uaCSTA), ECMA
TR/087” for the Answer Call, Hold Call, and Retrieve Call functions and “Services for Computer
Supported Telecommunications Applications Phase III, ECMA – 269” for the Conference Call
function.
This feature is enabled by configuration parameters described in the SIP parameter table in the
<voIpProt/> section and Registrations and needs to be activated by a feature application key.
SIP for Instant Messaging and Presence
The handset is compatible with the Presence and Instant Messaging features of Microsoft
Windows Messenger 5.1.
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Appendix C: Open Source Information
OFFER for Source for GPL and LGPL Software
You may have received a Spectralink 84-Series Wireless Handset from Spectralink that
contains—in part—free software (software licensed in a way that allows you the freedom to run,
copy, distribute, change, and improve the software).
A complete list of all open source software included in the Spectralink 84-Series Wireless
Handset, as well as related license and copyright information, is available at
http://support.spectralink.com.
You may also obtain the same information by contacting Spectralink by regular mail or email at
the addresses listed at the bottom of this notice.
For at least three (3) years from the date of distribution of the applicable product or software, we
will give to anyone who contacts us at the contact information provided below, for a charge of no
more than our cost of physically distributing, the items listed in “Spectralink OFFER of Source
for GPL and LGPL Software”, which is available at http://support.spectralink.com.
Contact Information for Requesting Source Code
Spectralink Open Source Manager
2560 55th Street
Boulder, CO 80301
OpenSource@Spectralink.com
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Appendix D: Library of <device/>
Settings
Spectralink provides a global device.set parameter that you can enable for software
installation and changes to device parameters.
Device settings are used when the handsets are initially deployed and need to associate with an
AP in order to locate the central provisioning server. These are the parameters that are
configured by the QNC/SLIC tool or when you manually configure the handset for wireless
association. Some device parameters are in the site.cfg template.
Each <device/> parameter has a corresponding .set parameter that enables or disables the
value for that device parameter. You will need to enable the corresponding .set parameter for
each parameter you want to apply.
Settings: Each <device/> Parameter has a Corresponding .set Parameter with
One Exception
Note that each <device/> parameter has a corresponding .set parameter that
enables or disables the parameter. There is one exception to this rule: the
device.sec.TLS.customDeviceCertX.set parameter applies to both
device.sec.TLS.customDeviceCertX.publicCert and to
device.sec.TLS.customDeviceCertX.privateKey.
Caution: Use Caution When Changing Device Parameters
Use caution when changing <device/> parameters as incorrect settings may apply
the same IP address to multiple handsets.
Note that some parameters may be ignored. For example, if DHCP is enabled it will
still overrule the value set with device.net.ipAddress.
Though individual parameters are checked to see whether they are in range, the
interaction between parameters is not checked. If a parameter is out of range, an
error message will display in the log file and the parameter will not be used.
Incorrect configuration can put the handsets into a reboot loop. For example, server
A has a configuration file that specifies that server B should be used, and server B
has a configuration file that specifies that server A should be used.
To detect errors, including IP addess conflicts, Spectralink recommends that you
test the new configuration files on two handsets before initializing all handsets.
The following table outlines the three types of <device/> parameters, their permitted values,
and the default value.
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Table 0-1: Device Parameters
Parameter
Permitted Values
Default
device.set1
0 or 1
0
If set to 0, do not use any device.xxx fields to set any parameters. Set this to 0 after the initial software
installation.
If set to 1, use the device.xxx fields that have device.xxx.set=1. Set this to 1 only for the initial software
installation.
device.xxx1
string
Null
0 or 1
0
Configuration parameter.
device.xxx.set1
If set to 0, do not use the device.xxx value. If set to 1, use the device.xxx value.
For example, if device.net.ipAddress.set=1, then use the value set for device.net.ipAddress .
1
Change causes handset to restart or reboot.
The following table lists each of the <device/> parameters that you can configure.
Table 0-2: Device Parameters
Parameter
Permitted Values
Default
device.baseProfile
Generic, Lync
Generic
Null, 128 to 254
160
Choose the Base Profile that the handset will operate with.
device.dhcp.bootSrvOpt1
When the boot server is set to Custom or Custom+Option66, specify the numeric DHCP option that the handset
will look for.
device.dhcp.bootSrvOptType1
IP or String
String
The type of DHCP option in which the handset will look for its provisioning server (if
device.dhcp.bootSrvUseOpt is set to Custom). If IP, the IP address provided must specify the format of the
provisioning server. If String, the string provided must match one of the formats specified by
device.prov.serverName.
device.dhcp.bootSrvUseOpt1
Default, Custom,
Static,
CustomAndDefault
CustomAndDefault
Default The handset will look for option number 66 (string type) in the response received from the DHCP
server. The DHCP server should send address information in option 66 that matches one of the formats described
for device.prov.serverName.
Custom The handset will look for the option number specified by device.dhcp.bootSrvOpt, and the type
specified by device.dhcp.bootSrvOptType in the response received from the DHCP server.
Static The handset will use the boot server configured through the provisioning server device.prov.*
parameters.
Custom and Default The handset will use the custom option first or use Option 66 if the custom option is not
present.
device.dhcp.enabled1
0 or 1
1
Binary, ASCII
ASCII
If 0, DHCP is disabled. If 1, DHCP is enabled.
device.dhcp.option60Type1
The DHCP option 60 type. Binary: vendor-identifying information is in the format defined in RFC 3925. ASCII:
vendor-identifying information is in ASCII format.
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Parameter
Permitted Values
Default
device.dhcp.dhcpVlanDiscUseOpt1
Disabled, Fixed,
Custom
Fixed
VLAN Discovery. Disabled, no VLAN discovery through DHCP. Fixed, use predefined DHCP vendor-specific
option values of 128, 144, 157 and 191 (device.dhcp.dhcpVlanDiscOpt will be ignored). Custom, use the
number specified by device.dhcp.dhcpVlanDiscOpt.
device.dhcp.dhcpVlanDiscOpt1
128 to 254
129
The DHCP private option to use when device.dhcp.dhcpVlanDiscUseOpt is set to Custom.
device.dns.altSrvAddress1
server address
Null
The secondary server to which the handset directs Domain Name System (DNS) queries.
device.dns.domain1
string
Null
string
Null
The handset’s DNS domain.
device.dns.serverAddress1
The primary server to which the handset directs Domain Name System queries.
device.hostname1
string
Null
This parameter enables you to specify a hostname for the handset when using DHCP by adding a hostname string
to the handset’s configuration. If device. hostname.set=1, and device. hostname=Null, the DHCP client
uses Option 12 to send a predefined hostname to the DHCP registration server using
Spectralink_<MACaddress>. Note that the maximum length of the hostname string is <=255 bytes. The valid
character set is defined in RFC1035.
device.prov.maxRedunServers1
1 to 8
8
The maximum number of IP addresses that will be used from the DNS.
device.prov.password1
string
admin123
The password for the handset to log in to the provisioning server. Note that a password may not be required. Note:
If you modify this parameter, the handset will re-provision. The handset may also reboot if the configuration on the
provisioning server has changed.
device.prov.redunAttemptLimit1
1 to 10
3
The maximum number of attempts to attempt a file transfer before the transfer fails.
device.prov.redunInterAttemptDelay1
0 to 300
1
The number of seconds to wait after a file transfer fails before retrying the transfer.
device.prov.serverName
dotted-decimal IP
address, domain
name string, or
URL
Null
The IP address, domain name, or URL of the provisioning server, followed by an optional directory and optional
configuration filename. This parameter is used if DHCP is disabled (device.dhcp.enabled is 0), if the DHCP
server does not send a boot server option, or if the boot server option is static (device.dhcp.bootSrvUseOpt
is static). Note: If you modify this parameter, the handset will re-provision. The handset may also reboot if the
configuration on the provisioning server has changed.
device.prov.serverType1
FTP, TFTP, HTTP,
HTTPS, FTPS
FTP
The protocol the handset uses to connect to the provisioning server. Note: Active FTP is not supported for
BootROM version 3.0 or later. Note: Only implicit FTPS is supported.
device.prov.upgradeServer
string
Null
The Upgrade server is an alternate way of getting software updates into the handset through the Web
Configuration Utility. It is a completely different process than using a provisioning server method.
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Parameter
Permitted Values
Default
When a value is displayed in this field, it is the address/URL that has been accessed for software updates through
the Web Configuration Utility. This value is also stored in the handset’s override file on the provisioning server.
If this field in the handset menu is populated then you cannot get code onto the phone from any other method than
using the WebUI upgrade method because handset settings have highest precedence and this setting is basically
a mirror of the override file. If you want to download code into a particular handset using a provisioning server,
clear the value set by the WebUI, or edit the override file parameter to “” which will also delete the setting in the
handset.
device.prov.tagSerialNo
0 or 1
0
If 0, the handset’s serial number (MAC address) is not included in the User-Agent header of HTTPS/HTTPS
transfers and communications to the browser. If 1, the handset’s serial number is included.
device.prov.user
string
administrator
The user name required for the handset to log in to the provisioning server (if required). Note: If you modify this
parameter, the handset will re-provision. The handset may also reboot if the configuration on the provisioning
server has changed.
device.sec.configEncryption.key1
string
Null
The configuration encryption key used to encrypt configuration files. For more information, see Encrypting
Configuration Files.
device.sec.TLS.dot1x.strictCertCommonNameValidation
0 or 1
1
If set to 1, 802.1X always verifies the server certificate for commonName/SubjectAltName match with the server
hostname that the phone is trying to connect.
device.sec.TLS.customCaCert1 (TLS Platform Profile 1)
device.sec.TLS.customCaCert2 (TLS Platform Profile 2)
string, PEM format
Null
The custom certificate to use for TLS Platform Profile 1 and TLS Platform Profile 2
device.sec.TLS.profile.caCertList must be configured to use a custom certificate.
device.sec.TLS.customDeviceCert1.publicCert
device.sec.TLS.customDeviceCert2.publicCert
Enter the signed
custom device
certificate in PEM
format (X.509)
Null
device.sec.TLS.customDeviceCert1.publicKey
device.sec.TLS.customDeviceCert2.publicKey
Enter the
corresponding
signed private key
in PEM format
(X.509)
Null
device.sec.TLS.customDeviceCert1.set
device.sec.TLS.customDeviceCert2.set
0 or 1
0
Note that you use a single .set parameter to enable or disable only these two related <device/> parameters device.sec.TLS.customDeviceCertX.publicCert and
device.sec.TLS.customDeviceCertX.publicKey. All other <device/> parameters have their own
corresponding .set parameter that will enable or disable that parameter.
device.sntp.gmtOffset
-43200 to 46800
0
The GMT offset – in seconds – to use for daylight savings time, corresponding to -12 to +13 hours.
device.sntp.serverName
dotted-decimal IP
address or domain
name string
Null
The SNTP server from which the handset will obtain the current time.
device.syslog.facility
0 to 23
16
A description of what generated the log message. For more information, see section 4.1.1 or RFC 3164.
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Parameter
Permitted Values
Default
device.syslog.prependMac1
0 or 1
0
If 1, the handset’s MAC address is pre-pended to the log message sent to the syslog server.
device.syslog.renderLevel1
0 to 6
4
Specify the logging level that will display in the syslog. Note that when you choose a log level, you are including all
events of an equal or greater severity level and excluding events of a lower severity level. The logging level you
choose determines the lowest severity of events that will be logged.
0 or 1: SeverityDebug(7). 2 or 3: SeverityInformational(6). 4: SeverityError(3). 5: SeverityCritical(2). 6:
SeverityEmergency(0).
device.syslog.serverName
dotted-decimal IP
address OR
domain name
string
Null
None, UDP, TCP,
TLS
UDP
The syslog server IP address or domain name string.
device.syslog.transport
The transport protocol that the handset will use to write to the syslog server. If set to None, transmission is turned
off but the server address is preserved.
device.usbnet.dhcpServerEnabled
0 or 1
0
If 1, a DHCP Server (which gives out addresses) needs to be started, as opposed to a DHCP Client (which gets an
address).
device.usbnet.ipGateway1
String
169.254.1.1
String
255.255.0.0
0 or 1
1
String
169.254.1.2
The provisioning server IP address.
device.usbnet.subnetMask1
The handset’s subnet mask for USBNet.
device.usbnet.enabled1
If 0, USBNet is disabled. If 1, USBNet is enabled.
device.usbnet.ipAddress1
The handset’s dotted-decimal IP address on the USBNet interface.
device.wifi.noBkgScanRssi
-100 to 0
-100
This parameter allows the handset to scan during standby in order to locate an AP with a stronger signal. As the
number gets higher, the handset scans more frequently. At -100 the handset will never scan in standby. The -65 to
-75 range will give you periodic scanning and is appropriate for most installations but will need to be tweaked
according to the signal quality in the facility. At -45, the phone will scan constantly and battery life will be impaired.
device.wifi.dhcpEnabled
0 or 1
1
If 0, DHCP is disabled on the wireless interface. If 1, DHCP is enabled on the wireless interface.
device.wifi.dot11n.enabled
0 or 1
1
If 0, 802.11n support is disabled. If 1, 802.11n support is enabled.
device.wifi.enabled
0 or 1
0
If 0, the wireless interface is disabled. If 1, the wireless interface is enabled.
device.wifi.ipAddress
String
0.0.0.0
String
0.0.0.0
The IP address of the wireless interface (if not using DHCP).
device.wifi.ipGateway
The IP gateway address for the wireless interface (if not using DHCP).
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Parameter
Permitted Values
Default
device.wifi.psk.keyType
The key type: key or passphrase.
device.wifi.psk.key
The hexadecimal key or ASCII passphrase.
0 or 1
0
string
0xFF
The WPA(2) PSK key type and key. If the key type is 0, a 256-bit hexadecimal key is used. If the key type is 1, a
string of 8 to 63 ASCII characters is used as the pass code.
device.wifi.acMandatory
0 or 1
0
If 1, the handset will only connect to access points that enforce admission control or access control. If 0, the
handset access control or admission control is not necessary. When deploying both 84-Series and 87-Series
handsets in the same facility using the same Wireless LAN, Wi-Fi Multimedia Admission Control (aka access
control, AC or WMM-AC) must be disabled in any handset parameters and APs as it is not supported by 87 Series
handsets. Any parameter that requires or enforces AC must be disabled.
device.wifi.radio.band5GHz.subBand1.enable1
device.wifi.radio.band5GHz.subBand2.enable1
device.wifi.radio.band5GHz.subBand3.enable1
device.wifi.radio.band5GHz.subBand4.enable1
0 or 1
0 or 1
0 or 1
0 or 1
0
0
0
0
If 0, the 5GHz sub-band (sub band 1, 2, 3, or 4) is disabled. If 1, the sub band is enabled. Note: Regulatory
authorities (FCC North America) further subdivide the 5GHz band into multiple sub-bands (some of which are not
available in all countries). You can enable and disable individual sub-bands and set the maximum transmit power
for each. For maximum performance, you should enable the same bands and sub-bands as configured on your
wireless infrastructure, otherwise your handset will waste time looking for a signal on the unused sub-bands.
device.wifi.radio.band5GHz.subBand1.txPower1
device.wifi.radio.band5GHz.subBand2.txPower1
device.wifi.radio.band5GHz.subBand3.txPower1
device.wifi.radio.band5GHz.subBand4.txPower1
1 to 7
1 to 7
1 to 7
1 to 7
5
5
5
5
The maximum power that the handset will use to transmit in the sub-band (for sub-band 1, 2, 3, and 4). In general,
this power should match the power setting at the access point so that the coverage radius of the handset matches
that of the access point. “7” is also called “Auto” and will use the maximum permitted power setting.
device.wifi.radio.band5GHzEnable1
0 or 1
0
If 0, the 5 GHz wireless band is disabled. If 1, the 5 GHz band is enabled. Note: enable the individual sub-bands
and set the transmit power for the sub-bands by configuring device.wifi.radio.band5GHz.subBandx.
device.wifi.radio.band2_4GHzEnable1
0 or 1
0
If 0, the 2.4 GHz wireless band is disabled. If 1, the 2.4 GHz band is enabled.
device.wifi.radio.band2_4GHz.txPower1
1 to 7
5
The maximum power that the handset will use to transmit in the 2.4 GHz band. In general, this power should
match the power setting at the access point so that the coverage radius of the handset matches that of the access
point. “7” is also called “Auto” and will use the maximum permitted power setting. Note that ESTI regulations limit
the maximum setting in Europe. When domain 2 is selected, txPower will not go above 4.
device.wifi.radio.regulatoryDomain
1, 2, or 10
Null
Available values specify the regulatory domain. The supported values are 1 (North America), 2 (Europe) and 10
(Australia). If Null, no regulatory domain is selected. You must set the regulatory domain before the handsets can
be used. There is no default setting for this option and the handsets will not associate with an access point (AP)
until you specify a value.
device.wifi.securityMode1
None, WEP, WPAPSK, WPA2-PSK,
WPA2-Enterprise
None
String
SSID1
The wireless security mode.
device.wifi.ssid1
The Service Set Identifier (SSID) of the wireless network.
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Parameter
Permitted Values
Default
device.wifi.subnetMask1
String
255.0.0.0
The network mask address of the wireless interface (if not using DHCP).
device.wifi.wep.authType1
OpenSystem,
SharedKey
0
1 to 4
1
The Wi-Fi WEP authentication type.
device.wifi.wep.defaultKey1
Specifies which of the four keys from device.wifi.wep.key1 to device.wifi.wep.key4 is used.
device.wifi.wep.encryptionEnable1
0 or 1
1
If 0, WEP encryption is disabled. If 1, WEP encryption is enabled.
device.wifi.wep.keyLength1
0 or 1
0
The length of the hexadecimal WEP key. 0= 40-bits, 1= 104-bits.
device.wifi.wep.key11
device.wifi.wep.key21
device.wifi.wep.key31
device.wifi.wep.key41
String
String
String
String
0xFF
0xFF
0xFF
0xFF
The WEP hexadecimal key with a 40-bit or 104-bit length, as specified by device.wifi.wep.keyLength.
device.wifi.wpa2Ent.eapFast.inBandProv1
0 or 1
0
If 0, the PAC file is initially loaded into to the handset during configuration (called out-of-band). If 1, the PAC file is
automatically loaded form the network (called in-band).
device.wifi.wpa2Ent.method1
EAPPEAPv0/MSCHAPv2,
EAP-FAST, EAP-TLS
PEAPv0/MSCHAPv2
The Extensible Authentication Protocol (EAP) to use for 802.1X authentication.
device.wifi.wpa2Ent.password1
String
admin123
The WPA2-Enterprise password.
The password is used in EAP-PEAP but is not used in EAP-TLS.
device.wifi.wpa2Ent.roaming1
OKC, CCKM
OKC
The WPA2-Enterprise fast roaming method. If OKC, Opportunistic Key Caching (OKC) is used. If CCKM, Cisco
Centralized Key Management (CCKM) is used. The fast roaming methods allow part of the key derived from the
server to be cached in the wireless network to shorten the time it takes to renegotiate a secure handoff.
device.wifi.wpa2Ent.user1
String
administrator
The WPA2-Enterprise user name. Used for EAP-PEAP and EAP-TLS
EAP-TLS uses this parameter or the User ID field under Network Interfaces> Wi-Fi Menu> WPA2-Enterprise for
the answer to the EAP "Identity" request.
For EAP-TLS: The value that is set in the User ID field should match the identity the RADIUS server will accept,
which may vary from one RADIUS server to another. If it is necessary for the identity to match the common name
from the factory installed certificate, then User ID should be set to the MAC address with lower case letters and no
punctuation, for example "00907a0cd9fd".
1
Change causes handset to restart or reboot.
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Appendix E: Trusted Certificate
Authority List
The phone trusts the following certificate authorities by default:
Release 4.13

Actalis Authentication Root CA

Starfield Services Root Certificate Authority - G2

EE Certification Centre Root CA

Juur-SK

Atos TrustedRoot 2011

Autoridad de Certificacion Firmaprofesional CIF A62634068

Buypass Class 2 CA 1

Buypass Class 2 Root CA

Buypass Class 3 Root CA

CA Disig

CA Disig Root R1

CA Disig Root R2

Chambers of Commerce Root

Chambers of Commerce Root - 2008

Global Chambersign Root

Global Chambersign Root - 2008

AC RaÃz Certicámara S.A.

Certigna

Certinomis - Autorité Racine

Certinomis - Root CA

certSIGN ROOT CA

CFCA EV ROOT

China Internet Network Information Center EV Certificates Root

CNNIC ROOT

ePKI Root Certification Authority

AAA Certificate Services

AddTrust Class 1 CA Root

AddTrust External CA Root

AddTrust Public CA Root

AddTrust Qualified CA Root

COMODO Certification Authority

COMODO ECC Certification Authority

COMODO RSA Certification Authority

Secure Certificate Services

Trusted Certificate Services
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
USERTrust ECC Certification Authority

USERTrust RSA Certification Authority

UTN-USERFirst-Client Authentication and Email

UTN-USERFirst-Hardware

UTN-USERFirst-Object

ComSign CA

ComSign Secured CA

EC-ACC

SecureSign RootCA11

D-TRUST Root Class 3 CA 2 2009

D-TRUST Root Class 3 CA 2 EV 2009

S-TRUST Authentication and Encryption Root CA 2005:PN

S-TRUST Universal Root CA

TC TrustCenter Class 3 CA II

Baltimore CyberTrust Root

Cybertrust Global Root

DigiCert Assured ID Root CA

DigiCert Assured ID Root G2

DigiCert Assured ID Root G3

DigiCert Global Root CA

DigiCert Global Root G2

DigiCert Global Root G3

DigiCert High Assurance EV Root CA

DigiCert Trusted Root G4

Class 2 Primary CA

E-Tugra Certification Authority

EBG Elektronik Sertifika Hizmet Sağlayıcısı

ACEDICOM Root

Entrust Root Certification Authority

Entrust Root Certification Authority - EC1

Entrust Root Certification Authority - G2

Entrust.net Certification Authority (2048)

GlobalSign

GlobalSign

GlobalSign ECC Root CA - R4

GlobalSign ECC Root CA - R5

GlobalSign Root CA

Go Daddy Class 2 CA

Go Daddy Root Certificate Authority - G2

Starfield Class 2 CA

Starfield Root Certificate Authority - G2

IGC/A

Hongkong Post Root CA 1
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Spectralink 84-Series Series Wireless Telephones Administration Guide

ApplicationCA - Japanese Government

ACCVRAIZ1

Root CA Generalitat Valenciana

Taiwan GRCA

Staat der Nederlanden EV Root CA

Staat der Nederlanden Root CA

Staat der Nederlanden Root CA - G2

Staat der Nederlanden Root CA - G3

TÜBİTAK UEKAE Kök Sertifika Hizmet Sağlayıcısı - Sürüm 3

Hellenic Academic and Research Institutions RootCA 2011

DST ACES CA X6

DST Root CA X3

IdenTrust Commercial Root CA 1

IdenTrust Public Sector Root CA 1

Izenpe.com

Microsec e-Szigno Root CA

Microsec e-Szigno Root CA 2009

NetLock Arany (Class Gold) FÅ‘tanúsÃtvány

NetLock Expressz (Class C) Tanusitvanykiado

NetLock Kozjegyzoi (Class A) Tanusitvanykiado

NetLock Minositett Kozjegyzoi (Class QA) Tanusitvanykiado

NetLock Uzleti (Class B) Tanusitvanykiado

PSCProcert

QuoVadis Root CA 1 G3

QuoVadis Root CA 2

QuoVadis Root CA 2 G3

QuoVadis Root CA 3

QuoVadis Root CA 3 G3

QuoVadis Root Certification Authority

RSA Security 2048 v3

Security Communication EV RootCA1

Security Communication Root CA

Security Communication RootCA2

StartCom Certification Authority

StartCom Certification Authority

StartCom Certification Authority G2

Swisscom Root CA 1

Swisscom Root CA 2

Swisscom Root EV CA 2

SwissSign Gold CA - G2

SwissSign Platinum CA - G2

SwissSign Silver CA - G2

Equifax Secure CA
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
Equifax Secure eBusiness CA-1

Equifax Secure Global eBusiness CA-1

GeoTrust Global CA

GeoTrust Global CA 2

GeoTrust Primary Certification Authority

GeoTrust Primary Certification Authority - G2

GeoTrust Primary Certification Authority - G3

GeoTrust Universal CA

GeoTrust Universal CA 2

thawte Primary Root CA

thawte Primary Root CA - G2

thawte Primary Root CA - G3

VeriSign Class 1 Public PCA

VeriSign Class 1 Public PCA - G3

VeriSign Class 1 Public PCA – G2

VeriSign Class 2 Public PCA - G3

VeriSign Class 2 Public PCA – G2

VeriSign Class 3 Public PCA

VeriSign Class 3 Public PCA - MD2

VeriSign Class 3 Public PCA – G2

VeriSign Class 3 Public Primary Certification Authority - G3

VeriSign Class 3 Public Primary Certification Authority - G4

VeriSign Class 3 Public Primary Certification Authority - G5

VeriSign Universal Root Certification Authority

Deutsche Telekom Root CA 2

T-TeleSec GlobalRoot Class 2

T-TeleSec GlobalRoot Class 3

TWCA Global Root CA

TWCA Root Certification Authority

Sonera Class1 CA

Sonera Class2 CA

TeliaSonera Root CA v1

AffirmTrust Commercial

AffirmTrust Networking

AffirmTrust Premium

AffirmTrust Premium ECC

Trustis FPS Root CA

Secure Global CA

SecureTrust CA

XRamp Global Certification Authority

TÜRKTRUST Elektronik Sertifika Hizmet Sağlayıcısı

TÜRKTRUST Elektronik Sertifika Hizmet Sağlayıcısı H5

TÜRKTRUST Elektronik Sertifika Hizmet Sağlayıcısı H6
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
Certum CA

Certum Trusted Network CA

Visa eCommerce Root

Network Solutions Certificate Authority

WellsSecure Public Root Certificate Authority

OISTE WISeKey Global Root GA CA

OISTE WISeKey Global Root GB CA

CA WoSign ECC Root

CA æ²ƒé€šæ ¹è¯•ä¹¦

Certification Authority of WoSign

Certification Authority of WoSign G2
Release 4.12

ACCVRAIZ1

ACEDICOM Root

Actalis Authentication Root CA

AddTrust External Root

AddTrust Low-Value Services Root

AddTrust Public Services Root

AddTrust Qualified Certificates Root

AffirmTrust Commercial

AffirmTrust Networking

AffirmTrust Premium

AffirmTrust Premium ECC

ApplicationCA - Japanese Government

Atos TrustedRoot 2011

A-Trust-nQual-03

Autoridad de Certificacion Firmaprofesional CIF A62634068

Baltimore CyberTrust Root

Buypass Class 2 CA 1

Buypass Class 2 Root CA

Buypass Class 3 CA 1

Buypass Class 3 Root CA

CA Disig

CA Disig Root R1

CA Disig Root R2

Camerfirma Chambers of Commerce Root

Camerfirma Global Chambersign Root

Certigna

Certinomis - Autorité Racine

Certplus Class 2 Primary CA

certSIGN ROOT CA
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
Certum Root CA

Certum Trusted Network CA

CFCA EV ROOT

Chambers of Commerce Root - 2008

China Internet Network Information Center EV Certificates Root

CNNIC ROOT

Comodo AAA Services root

COMODO Certification Authority

COMODO ECC Certification Authority

COMODO RSA Certification Authority

Comodo Secure Services root

Comodo Trusted Services root

ComSign Secured CA

Cybertrust Global Root

Deutsche Telekom Root CA 2

DigiCert Assured ID Root CA

DigiCert Assured ID Root G2

DigiCert Assured ID Root G3

DigiCert Global Root CA

DigiCert Global Root G2

DigiCert Global Root G3

DigiCert High Assurance EV Root CA

DigiCert Trusted Root G4

DST ACES CA X6

DST Root CA X3

D-TRUST Root Class 3 CA 2 2009

D-TRUST Root Class 3 CA 2 EV 2009

EBG Elektronik Sertifika Hizmet Sa\xC4\x9Flay\xc4\xb1\x63\xc4\xb1s\xc4\xb1

EC-ACC

EE Certification Centre Root CA

Entrust Root Certification Authority

Entrust Root Certification Authority - EC1

Entrust Root Certification Authority - G2

Entrust.net Premium 2048 Secure Server CA

ePKI Root Certification Authority

Equifax Secure CA

E-Tugra Certification Authority

GeoTrust Global CA

GeoTrust Global CA 2

GeoTrust Primary Certification Authority

GeoTrust Primary Certification Authority - G2

GeoTrust Primary Certification Authority - G3

GeoTrust Universal CA
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Spectralink 84-Series Series Wireless Telephones Administration Guide

GeoTrust Universal CA 2

Global Chambersign Root - 2008

GlobalSign ECC Root CA - R4

GlobalSign ECC Root CA - R5

GlobalSign Root CA

GlobalSign Root CA - R2

GlobalSign Root CA - R3

Go Daddy Class 2 CA

Go Daddy Root Certificate Authority - G2

Hellenic Academic and Research Institutions RootCA 2011

Hongkong Post Root CA 1

IdenTrust Commercial Root CA 1

IdenTrust Public Sector Root CA 1

IGC/A

Izenpe.com

Juur-SK

Microsec e-Szigno Root CA

Microsec e-Szigno Root CA 2009

NetLock Arany (Class Gold) FÅ‘tanúsÃtvány

NetLock Notary (Class A) Root

Network Solutions Certificate Authority

OISTE WISeKey Global Root GA CA

PSCProcert

QuoVadis Root CA

QuoVadis Root CA 1 G3

QuoVadis Root CA 2

QuoVadis Root CA 2 G3

QuoVadis Root CA 3

QuoVadis Root CA 3 G3

Root CA Generalitat Valenciana

RSA Security 2048 v3

Secure Global CA

SecureSign RootCA11

SecureTrust CA

Security Communication EV RootCA1

Security Communication Root CA

Security Communication RootCA2

Sonera Class 2 Root CA

Staat der Nederlanden EV Root CA

Staat der Nederlanden Root CA

Staat der Nederlanden Root CA - G2

Staat der Nederlanden Root CA - G3

Starfield Class 2 CA
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Spectralink 84-Series Series Wireless Telephones Administration Guide

Starfield Root Certificate Authority - G2

Starfield Services Root Certificate Authority - G2

StartCom Certification Authority

StartCom Certification Authority

StartCom Certification Authority G2

Swisscom Root CA 1

Swisscom Root CA 2

Swisscom Root EV CA 2

SwissSign Gold CA - G2


SwissSign Silver CA - G2
T\xc3\x9c\x42\xC4\xB0TAK UEKAE K\xC3\xB6k Sertifika Hizmet
Sa\xC4\x9Flay\xc4\xb1\x63\xc4\xb1s\xc4\xb1 - S\xC3\xBCr\xC3\xBCm 3

Taiwan GRCA

TC TrustCenter Class 2 CA II

TC TrustCenter Universal CA I

TeliaSonera Root CA v1

thawte Primary Root CA

thawte Primary Root CA - G2

thawte Primary Root CA - G3

Trustis FPS Root CA

T-TeleSec GlobalRoot Class 2

T-TeleSec GlobalRoot Class 3

TURKTRUST Certificate Services Provider Root 1

TURKTRUST Certificate Services Provider Root 2

TURKTRUST Certificate Services Provider Root 2007

TWCA Global Root CA

TWCA Root Certification Authority

USERTrust ECC Certification Authority

USERTrust RSA Certification Authority

UTN DATACorp SGC Root CA

UTN USERFirst Hardware Root CA

Verisign Class 3 Public Primary Certification Authority - G3

VeriSign Class 3 Public Primary Certification Authority - G4

VeriSign Class 3 Public Primary Certification Authority - G5

Verisign Class 4 Public Primary Certification Authority - G3

VeriSign Universal Root Certification Authority

Visa eCommerce Root

WellsSecure Public Root Certificate Authority

WoSign

WoSign China

XRamp Global CA Root
Pre-release 4.12
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Spectralink 84-Series Series Wireless Telephones Administration Guide

AAA Certificate Services by COMODO

ABAecom (sub., Am. Bankers Assn.) Root CA

Add Trust Class1 CA Root by COMODO

Add Trust External CA Root by COMODO

Add Trust Public CA Root by COMODO

Add Trust Qualified CA Root by COMODO

ANX Network CA by DST

American Express CA

American Express Global CA

BelSign Object Publishing CA

BelSign Secure Server CA

COMODO CA Limited

COMODO Certificate Authority

Deutsche Telekom AG Root CA

Digital Signature Trust Co. Global CA 1

Digital Signature Trust Co. Global CA 2

Digital Signature Trust Co. Global CA 3

Digital Signature Trust Co. Global CA 4

Entrust Worldwide by DST

Entrust.net Premium 2048 Secure Server CA

Entrust.net Secure Personal CA

Entrust.net Secure Server CA

Equifax Premium CA

Equifax Secure CA

Equifax Secure eBusiness CA 1

Equifax Secure eBusiness CA 2

Equifax Secure Global eBusiness CA 1

GeoTrust Primary Certification Authority

GeoTrust Global CA

GeoTrust Global CA 2

GeoTrust Universal CA

GeoTrust Universal CA 2

GTE CyberTrust Global Root
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Spectralink 84-Series Series Wireless Telephones Administration Guide

GTE CyberTrust Japan Root CA

GTE CyberTrust Japan Secure Server CA

GTE CyberTrust Root 2

GTE CyberTrust Root 3

GTE CyberTrust Root 4

GTE CyberTrust Root 5

GTE CyberTrust Root CA

GlobalSign Partners CA

GlobalSign Primary Class 1 CA

GlobalSign Primary Class 2 CA

GlobalSign Primary Class 3 CA

GlobalSign Root CA

Go Daddy Class 2 Certification Authority Root Certificate

Go Daddy Class 2 Certification Authority Root Certificate – G2

National Retail Federation by DST

RSA 2048 v3 Root CA

Secure Certificate Services by COMODO

TC TrustCenter, Germany, Class 1 CA

TC TrustCenter, Germany, Class 2 CA

TC TrustCenter, Germany, Class 3 CA

TC TrustCenter, Germany, Class 4 CA

Thawte Personal Basic CA

Thawte Personal Freemail CA

Thawte Personal Premium CA

Thawte Premium Server CA

Thawte Server CA

Thawte Universal CA Root

Trusted Certificate Services by COMODO

UTN-DATA Corp SGC by COMODO

UTN-USER First-Client Authentication and Email by COMODO

UTN-USER First-Hardware by COMODO

UTN-USER First-Object by COMODO

UPS Document Exchange by DST
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Spectralink 84-Series Series Wireless Telephones Administration Guide

ValiCert Class 1 VA

ValiCert Class 2 VA

ValiCert Class 3 VA

Verisign 2048 Root CA

VeriSign Class 4 Primary CA

Verisign Class 1 Public Primary Certification Authority

Verisign Class 1 Public Primary Certification Authority - G2

Verisign Class 1 Public Primary Certification Authority - G3

Verisign Class 2 Public Primary Certification Authority

Verisign Class 2 Public Primary Certification Authority - G2

Verisign Class 2 Public Primary Certification Authority - G3

Verisign Class 3 Public Primary Certification Authority

Verisign Class 3 Public Primary Certification Authority - G2

Verisign Class 3 Public Primary Certification Authority - G3

Versign Class 3 Public Primary Certification Authority – G5

Verisign Class 4 Public Primary Certification Authority - G2

Verisign Class 4 Public Primary Certification Authority - G3

Verisign/RSA Commercial CA

Verisign/RSA Secure Server CA

Windows Root Update by COMODO
Troubleshooting: My Certificate Authority is Not Listed
Spectralink endeavors to maintain a built-in list of the most commonly used CA
Certificates. Due to memory constraints, we cannot ensure a complete set of
certificates. If you are using a certificate from a commercial Certificate Authority not
in the list above, contact Spectralink Support for help. At this point, you can use the
Custom Certificate method to load your particular CA certificate into the phone.
Refer to Using Custom Certificates on Spectralink handsets (Technical Bulletin
17877).
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Appendix F: Spectralink Certificates
Spectralink CA certificates can be obtained from:
http://pki.spectralink.com/aia/Spectralink%20Issuing%20CA.crt
http://pki.spectralink.com/aia/Spectralink%20Root%20CA.crt
http://pki.spectralink.com/aia/Spectralink%20Issuing%20CA%20BLCAI01.crt
END OF DOCUMENT
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