Avaya SIP Firmware Release 2.0 for IP Phone 1140E User Manual

SIP Firmware Release 2.0 for IP Phone 1140E Administration
NN43113-300
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Document status: Standard Document version: 03.09 Document date: 15 September 2008
Copyright © 2008, Nortel Networks All Rights Reserved.
LEGAL NOTICE While the information in this document is believed to be accurate and reliable, except as otherwise expressly agreed
to in writing NORTEL PROVIDES THIS DOCUMENT AS IS WITHOUT WARRANTY OR CONDITION OF ANY KIND, EITHER EXPRESS OR IMPLIED. The information and/or products described in this document are subject to change without notice.
Sourced in Canada Nortel, the Nortel Logo, the Globemark, SL-1, Meridian 1, and Succession are trademarks of Nortel Networks. All other trademarks are the property of their respective owners.
Contents
New in this release 7
Server Supplied User Credentials 7 Expansion module for the IP Phone 1100 Series when connected to IP Phone 1100
IEEE 802.1ab Link Layer Discovery support 7 IP Phone 1140E with SIP firmware PRACK and UPDATE support feature 8 RFC 3581 8 RTP port configuration 9 Voice Quality Monitoring 9 Distinctive ringing and Call waiting 9 Shared Call Appearance 10 Transfer to Voice Mail 10 Other changes 10
How to Get Help 13
Getting Help from the Nortel Web site 13 Getting Help over the telephone from a Nortel Solutions Center 13 Getting Help from a specialist using an Express Routing Code 14 Getting Help through a Nortel distributor or reseller 14
3
series SIP Client 7
Revision history 10
Introduction to this guide 15
Subject 15 Intended audience 15 Acronyms 15 Related publications 17
Overview 19
Introduction 19
SIP overview 19 IP Phone 1140E with SIP Firmware 19 Related documentation 20 Installation overview 21
Supported SIP proxy servers 24
Before installation 25
Introduction 25
SIP Firmware Release 2.0 for IP Phone 1140E Administration
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4 Contents
Preinstallation 25
Configure the provisioning server 29
How provisioning works 29 Download the SIP Firmware to the provisioning server 30 Create the SIP provisioning file on the provisioning server 30
Setting the default language on the IP Phone 1140E to French 35
Create the device configuration file on the provisioning server 36
Device configuration commands 40
Create the set configuration file on the provisioning server 59
Set configuration commands 60
Create the Dialing Plan file on the provisioning server 61
Dialing function description 63
Downloadable WAV files 67
Configure the DHCP server 69
Partial DHCP mode 69 Full DHCP mode 70
Configure the DHCP server to support Full DHCP mode 71 DHCP VLAN Auto Discovery 73
Physically install the IP Phone 1140E 75 Upgrade and convert the IP Phone 1140E firmware 79
Introduction 79 Upgrade the SIP Firmware on the IP Phone 1140E 79
Download the SIP Firmware to the provisioning server 79 Modify the SIP provisioning file 80
Upgrade to the minimum UNIStim Firmware 81
Identify the current version of UNIStim Firmware 81 Upgrade UNIStim Firmware to the minimum required UNIStim Firmware 84
Convert UNIStim Firmware to SIP Firmware on the IP Phone 1140E 87
Voice Quality Monitoring 91
Feature overview 91 VQMon set-up 91 Server set-up 92 How VQMon works 92
End of call report 93 Session interval report 93 Alert interval report 93
Device Settings on the IP Phone 1140E with SIP Firmware 95
Introduction 95
802.1x (EAP) Port-based network access control 98
Configure the device settings 111
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Contents 5
Multiple Appearance Directory Number (Single Call
Arrangement) 123
Vertical services 123
Privacy 124
Feature dependencies and restrictions 124
Emergency 911 - Operator control of disconnect 125
Overview 125 Location information 125 Dialing plan configuration 126 Characteristics of E911 calls 126
IP Phone 1140E restrictions 127
Service package restrictions 127 Distinctive Ringing feature 127
NAT firewall traversal 129 Three-port switch and VLAN functionality 131
System overview 131
SIP messages supported by the IP Phone 1140E 135
SIP methods 135 SIP responses 136
1xx Response--Information Responses 136 2xx Response--Successful responses 137 3xx Response--Redirection responses 137 4xx Response--Request failure responses 138 5xx Response--Server failure responses 140
6xx Response--Global responses 141 Default error handling 141 SIP header fields 141 Session description protocol usage 144 SDP and Call Hold 145 Transport layer protocols 145 SIP security authentication 145 SIP DTMF Digit transport 146 Supported subscriptions 146 Supported instant messaging 146
Maintenance and troubleshooting 149
Convert SIP Firmware to UNIStim Firmware 149 IP Phone 1140E diagnostics 150
Index 153
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6 Contents
SIP Firmware Release 2.0 for IP Phone 1140E Administration
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New in this release

The following changes are introduced for SIP Firmware Release 2.0 for IP Phone 1140E Administration (NN43113-300).

Server Supplied User Credentials

This release adds support for downloading a users login credentials from a central repository. The login credentials are associated with the MAC address of the set. This feature simplifies installation of the IP Phone by allowing it to automatically log into the SIP proxy server without end user intervention.
To provision this feature see "Set configuration commands" (page 60).

Expansion module for the IP Phone 1100 Series when connected to IP Phone 1100 series SIP Client

An expansion module providing 18 additional feature keys, can be attached to a Nortel IP Phone 1140E with SIP firmware. Up to three expansion modules can be added to the phone. The feature keys in the expansion module provide the same functionality as the feature keys on the IP Phone.
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Additionally, this feature adds the ability to automatically populate the extra feature keys, using the friends list or the address book as sources.
For more information on the Expansion module refer to, SIP Firmware for IP Phone 1100 Series Expansion Module User Guide (NN43110-301).
To provision this feature see "Feature configuration commands " (page 42).

IEEE 802.1ab Link Layer Discovery support

Discovery protocols provide a mechanism to identify devices attached to a network. Popular network-management systems use automated discovery to obtain the topology of a network. These applications detect adds and removals, provide Layer 3 information, and group the attached devices into IP subnets.
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IEEE has developed 802.1ab Link Layer Discovery Protocol (LLDP), a standard for discovering the physical topology between neighboring devices.
802.1ab LLDP defines a standard method for Ethernet network devices such as switches, routers and IP Phones to advertise information about themselves to other nodes on the network and store the information they discover in a MIB.
This feature implements 802.1ab and its IEEE 802.1/802.3 and TIA-MED organizationally specific extensions.
To provision this feature see "802.1ab Link Layer Discovery Protocol"
(page 99) and Procedure 13 "Provisioning the Device Settings parameters" (page 111). The parameters in Table 18 "TLV formats" (page 100) can be
configured on the IP Phone.
IP Phone 1140E with SIP firmware PRACK and UPDATE support feature
The combined support of PRACK and UPDATE allows the IP Phone 1140E with SIP firmware to provide reliability to provisional responses, and the ability to update session parameters during call setup as well as after the initial invite has received a final response. The combination of reliable provisional responses (PRACK) and the ability to change session parameters before call establishment (using UPDATE) will improve the IP Phone 1140E interactions with some PSTN networks where the parameters of a session may need updating before the call is established.
PRACK is used for adding reliability to provisional responses (such as 180 Ringing) in SIP messaging. The PRACK request plays the same role as ACK, but for provisional responses. [RFC 3262]
UPDATE allows a client to update parameters of a session (such as the set of media streams and their codecs) but has no impact on the state of a dialog. In that sense, it is like a re-INVITE, but unlike re-INVITE, it can be sent before the initial INVITE has been completed. This makes it very useful for updating session parameters within early dialogs. [RFC 3311]
To provision this feature see "Feature configuration commands " (page 42).

RFC 3581

This feature implements an extension to SIP for Symmetric Response Routing for the IP Phone 1140E with SIP firmware. This extension permits the conduction of SIP dialogs through a Symmetric Network Address Translator (NAT) using UDP.
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This allows the phone to work from behind and/or in front of a symmetrical NAT with servers and/or clients that support RFC 3581. In particular, it enhances the capability of the IP Phone 1140E with SIP firmware to interoperate with other proxies and to work in any network configuration. For this feature to work properly, the receiving end device must support RFC 3581.
To provision this feature see "Feature configuration commands " (page 42).
RTP port configuration
This RTP port configuration feature provides options to set or change RTP ports in the Nortel IP Phone 1140E with SIP firmware. RTP port configuration is available only through the provisioning server.
To provision this feature see "Feature configuration commands " (page 42).

Voice Quality Monitoring

When Voice Quality Monitoring (VQMon) is enabled, the IP Phone 1140E with SIP firmware gathers statistics regarding the quality metrics of the current call and sends reports to the server at regular intervals. The call-related statistics contain condensed information about the SIP Session Description Protocol (SDP), the Call ID, the local and remote address, voice quality-related statistics, Zulu times for start-time and the time the report was sent. The voice quality-related statistics include jitter, packet loss, delay, burst gap loss, listening R-factor, R-LQ, R-CQ, MOS-LQ and MOS-CQ. The reports can be used for QoS monitoring. A third party server is required to collect and organize the data.
Distinctive ringing and Call waiting 9
End of call reports are sent with the Zulu time indicating start of session and end of session; session interval reports are sent with interval defined in the configuration using time stamps, and alert reports are sent with current time stamps with a limited frequency of 1 report per 5 seconds if any alert violation persists.
Release 2.0 supports interworking with SQLMediator 1.1. To provision this feature see "VQMon configuration commands" (page 57).

Distinctive ringing and Call waiting

The Distinctive ringing feature permits users to distinguish between different types of call actions by playing a different ringing pattern. The request for a specific ringing pattern comes from the server at the time the call is being established. The Nortel IP Phone 1140E with SIP firmware does not request the playing of distinctive ringing from other parties. The ringing patterns to be used follow the North American standards which also include call waiting
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10 New in this release
tones for the times when the receiving end is already engaged in a call session. The predefined ringing pattern identifiers available on the phone accommodate usage by the BroadWorks servers.
To provision this feature see "Feature configuration commands " (page 42).

Shared Call Appearance

The Shared Call Appearance feature allows a given line to be configured with multiple locations, essentially allowing multiple endpoints to login to the system while using the same external number. Any one of these locations can be used to originate or receive calls. Any call appears to the other party to be originating from or terminating to the same number, regardless of the location initiating or receiving the call.
The Shared Call Appearance feature allows a user to pick up a call that was put on hold by another user of the same group and it allows a user to join an active call of another user in the group.
The Shared Call Appearance feature is supported on BroadWorks servers. To provision this feature see "Feature configuration commands " (page 42).

Transfer to Voice Mail

The transfer to Voice Mail feature enables a soft key to transfer an incoming call, or a call waiting, to a voice mail box on the IP Phone 1140E with SIP firmware.
The transfer to Voice Mail feature is supported on MCS 5100 servers only. To provision this feature see "Feature configuration commands " (page 42).

Other changes

For a detailed history of past releases of this document, see "Revision
history" (page 10).

Revision history

September 2008
Standard 03.09. This document is up-issued to reflect changes in technical content.
July 2008
Standard 03.08. This document is up-issued to reflect changes in technical content.
June 2008
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Other changes 11
Standard 03.07. This document is up-issued to support SIP Firmware for IP Phone 1140E Release 2.0.
February 2008
Standard 02.04. This document is up-issued to include editing changes.
November 2007
Standard 02.03. This document is up-issued to reflect changes in the technical information in response to change requests. A note was added stating that NAT_SIGNALLING is required for networks that use STUN or SIP_PING for NAT traversal. The commands FORCE_OCT_END and OCT_END were changed to FORCE_OCT_ENDDIAL and DISABLE_OCT_ENDDIAL respectively.
October 2007
Standard 02.02. This document is up-issued to ensure that the value of
802.1P_MEDIA parameter in the QoS and ToS commands must be set to
-1 to be able to have a voice path.
July 2007
Standard 02.01. This document is up-issued to support SIP Firmware for IP Phone 1140E Maintenance Release 1.1.
March 2007
Standard 01.02. This document is up-issued to incorporate an addition to the Maintenance and Troubleshooting section.
January 2007
Standard 01.01. This is a new document created to support the IP Phone 1140E with SIP Firmware.
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How to Get Help

This section explains how to get help for Nortel products and services.

Getting Help from the Nortel Web site

The best way to technical support for Nortel products is from the Nortel Technical support Web site:
ww.nortel.com/support
w
This site provides quick access to software, documentation, bulletins, and tools to address issues with Nortel products. More specifically, the site enables you to:
download software, documentation, and product bulletins
search the Technical Support Web site and the Nortel Knowledge Base for answers to technical issues
sign up for automatic notification of new software and documentation
for Nortel equipment
13
open and manage technical support cases

Getting Help over the telephone from a Nortel Solutions Center

If you do not find the information you require on the Nortel Technical Support Web site, and have a Nortel support contract, you can also get help over the telephone from a Nortel Solutions Center.
In North America, call 1-800-4Nortel (1-800-466-7835). Outside North America, go to the following Web site to obtain the telephone
number for your region:
ww.nortel.com/callus
w
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14 How to Get Help

Getting Help from a specialist using an Express Routing Code

To access some Nortel Technical Solutions Centers, you can use an Express Routing Code (ERC) to quickly route your call to a specialist in your Nortel product or service. To locate the ERC for your product or service, go
ww.nortel.com/erc
to w

Getting Help through a Nortel distributor or reseller

If you purchased a service contract for your Nortel product from a distributor or reseller, contact the technical support staff for that distributor or reseller.
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Introduction to this guide

Subject

SIP Firmware Release 2.0 for IP Phone 1140E Administration
(NN43113-300) describes how to install, configure, and provision the IP Phone 1140E for use on a SIP network.

Intended audience

This administration guide is intended for system administrators of the Nortel IP Phone 1140E with a basic understanding of SIP. This guide is not intended for end users of the IP Phone 1140E. Many of the tasks outlined in the guide influence the function of the IP Phone 1140E on the network and require an understanding of telephony and Internet Protocol (IP) networking.

Acronyms

This guide uses the following acronyms:

Table 1 Acronyms used

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AAA Authentication, Authorization, and Accounting ALG Application Layer Gateway DHCP Dynamic Host Configuration Protocol DNS Domain Name System DRegex Digit Regular Expression DSCP Differentiated Services Code Point EAP Extensible Authentication Protocol ERE Extended Regular Expressions FQDN Fully Qualified Domain Name FTP File Transfer Protocol GARP Gratuitous Address Resolution Protocol GUI Graphical User Interface HTTP Hyper Text Transfer Protocol
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16 Introduction to this guide
IETF Internet Engineering Task Force ISDN Integrated Services Digital Network IM Instant Message IP Internet Protocol IPCM Internet Protocol Client Manager ITU-T Telecommunications Standardization sector of the International
Telecommunications Union LAN Local Area Network MAC Media Access Control MADN Multiple Appearance Directory Number MAS Media Application Server NAT Network Address Translator NetConfig Configuration screens available after an IP Phone resets NDU Network Diagnostic Utility PDT Problem Determination Tool PEC Product Engineering Code POE Power Over Ethernet POSIX Portable Operating System Interface PRACK Provisional Acknowledgement PSTN Public Switched Telephone Network PVQMon Proactive Voice Quality Monitoring QoS Quality of Service RADIUS Remote Authentication Dial In User Service RTCP Real-time Control Protocol RTCP XR RTP Control Protocol Extended Reports RTP Real-time Transfer Protocol SCA Single Call Arrangement
Shared Call Appearance SDP Session Description Protocol SIMPLE SIP for Instant Messaging and Presence Leveraging Extensions SIP Session Initiation Protocol SMTP Simple Mail Transfer Protocol STUN Simple Traversal of UDP through NAT devices TFTP Trivial File Transport Protocol TPS Terminal Proxy Server
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TTL Time-to-live UDP User Datagram Protocol UFTP UNIStim File Transfer Protocol UNIStim Unified Network IP Stimulus Protocol VoIP Voice over IP VLAN ID Virtual Local Area Network Identification VLAN IP Virtual Local Area Network Internet Protocol VQMon Voice Quality Monitoring

Related publications

Other publications related to the SIP Firmware Release 2.0 for IP Phone 1140E Administration are:
SIP Firmware Release 2.0 for IP Phone 1140E Quick Reference Card
(NN43113-102)
SIP Firmware Release 2.0 for IP Phone 1140E User Guide
(NN43113-101)
Related publications 17
Broadsoft Partner Configuration Guide Nortel SIP Firmware for 1120E / 1140E IP Phones (NN43121-300)
Expansion Module for IP Phones 1140E and 1120E (SIP Firmware) User Guide (NN43110-301)
Nortel IP Phone 1100 Series product bulletins on w
ww.nortel.com/sup-
port.
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18 Introduction to this guide
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Overview

Introduction

This chapter describes the hardware and firmware features of the Nortel IP Phone 1140E and provides a brief overview of Session Initiation Protocol (SIP).

SIP overview

Session Initiation Protocol (SIP) is a signaling protocol used for establishing multimedia sessions in an Internet Protocol (IP) network.
SIP is a text-based protocol similar to HTTP and SMTP. With the introduction of SIP to IP Phones, telephony integrates easily with other Internet services. SIP allows the convergence of voice and multimedia.

IP Phone 1140E with SIP Firmware

The IP Phone 1140E connects to an IP network using an Ethernet connection. All voice and signaling information is converted into IP packets and sent across the network.
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The IP Phone 1140E can be ordered with UNIStim Firmware installed or with SIP Firmware installed. UNIStim Firmware and SIP Firmware use the same hardware, but the model number of an IP Phone 1140E with UNIStim Firmware is different from the model number of an IP Phone 1140E with SIP Firmware.
If you have an IP Phone 1140E with UNIStim Firmware, you can convert the firmware to SIP Firmware. To download the most recent version of SIP Firmware, see "Download the SIP Firmware to the provisioning server"
(page 30).
This guide explains how to:
configure the provisioning server and the DHCP server. Note: The provisioning server is where the firmware and the configuration files for the IP Phone 1140E reside. This is not the IP Client Manager (IPCM) of the call server.
convert an IP Phone 1140E with UNIStim Firmware to an IP Phone 1140E with SIP Firmware
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20 Overview
provision the Device Settings parameters on the IP Phone 1140E with SIP Firmware
Note: Converting the firmware on an IP Phone 1140E from UNIStim Firmware to SIP Firmware overwrites the UNIStim Firmware. The IP Phone 1140E cannot operate in both modes simultaneously. A switch from UNIStim to SIP Firmware or SIP to UNIStim Firmware requires a firmware reload.
The following figure shows the main components of the IP Phone 1140E with SIP Firmware.

Figure 1 IP Phone 1140E with SIP Firmware

Related documentation

The SIP Firmware Release 2.0 for IP Phone 1140E User Guide (NN43113-101) tells the end user how to use the IP Phone 1140E, including how to:
use the Context-sensitive soft keys and Navigation key cluster
enter text
use the address book
access and use the call inbox and call outbox
configure and use instant messaging
receive, identify, answer, redirect, decline, or ignore an incoming call
operate hold, three-way calling, call transfer, and call park
use other features such as speed dial, call forward, do not disturb, and setting up conference calls
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configure Bluetooth headset operation (IP Phone 1140E only) Note: All features are not available with all call servers.
The IP Phone 1140E Getting Started Card included in the box with the IP Phone 1140E also contains useful information and explains how to:
connect the AC power adapter
control the volume when answering a call
make a call using the handset
make a call with the headset or using handsfree
use hold and mute
set the contrast
set the language

Installation overview

To install the IP Phone 1140E with SIP Firmware, three basic steps are required.
Introduction 21

Procedure 1 Installation overview

Step Action 1
Configure the provisioning server and, optionally, the DHCP server. The function of the provisioning server is to provide configuration options to every IP Phone 1140E throughout the network. TheDHCP server can be configured to provide basic network-configuration data or a more comprehensive set of network-configuration data for the IP Phone 1140E with SIP Firmware.
2 3
Load SIP Firmware on the IP Phone 1140E. Configure the initial network-configuration parameters on the IP
Phone 1140E with SIP Firmware.
—End—
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22 Overview

Figure 2 Installation of IP Phone 1140E with SIP Firmware, page 1 of 2

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Figure 3 Installation of IP Phone 1140E with SIP Firmware, page 2 of 2

Introduction 23
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24 Overview

Supported SIP proxy servers

Call server Call server release Minimum SIP Firmware on
the IP Phone 1140E
MCS 5100
MCS 5100
MCS 5100 with CS 1000
MCS 5100 with CS 1000
MCS 5100 with CS 1000
CS 1500
3.5 1.0
4.0 1.0
3.5
4.5
3.5
5.0
4.0
5.0
1.0 1.1
CS 2000 SN09U
CS 2000 SN10
CS 2100 SE09.1
1.0
1.1
1.1
1.1
1.1
1.1
CS 2100 SE10
BroadSoft
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Before installation

Introduction

This chapter features a checklist of tasks you must complete before you install SIP Firmware on the IP Phone 1140E.

Preinstallation

Complete the following checklist.

Procedure 2 Preinstallation checklist

Step Action 1 Read and become familiar with the SIP Firmware Release 2.0
for IP Phone 1140E User Guide (NN43113-101) and the SIP Firmware Release 2.0 for IP Phone 1140E Quick Reference Card
(NN43113-102).
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2
Ensure there is one IP Phone 1140E boxed package for each IP Phone 1140E being installed.
3
Table 2 IP Phone 1140E box contents
Item
IP Phone 1140E Graphite with icon keys without PS (SIP) (RoHS)
IP Phone 1140E Graphite with English keys without PS (SIP) (RoHS)
Accessories
Handset, Charcoal NTYS09AA70 Handset cord, Charcoal NTYS10AA70
Copyright © 2008, Nortel Networks
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Ensure that the IP Phone 1140E box includes:
Order number
NTYS05CAE6 N0142353
NTYS05DAE6 N0142354
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Footstand kit, Charcoal NTYS11AA70 Telephone number label and lens kit NTYS12AA
2.3 m (7 ft) CAT5 Ethernet cable NTYS13AA
The IP Phone 1140E can be powered either by Power Over Ethernet (POE) or powered through an external power supply. Order the external power supply separately.
WARNING
Do not use the AC power adapter, if you are connected to a Power over the Ethernet (PoE) connection. Only use the AC power adapter when you do not have a Power over the Ethernet connection.
Table 3 1100 series phones separably orderable parts
CPC code PEC code
N0146475 NTYS17BAE6 IP Phone Global Power Supply (2000, 1100, 1200) (RoHS) N0089603 NTYS14AAE6 Standard IEC Cable - North America (RoHS) A0781922 NTTK15AA Standard IEC Cable – Australia / NZ (Note: RoHS not
N0114986 NTTK16ABE6 Standard IEC Cable – Europe N0109787 NTTK17ABE6 Standard IEC Cable – Switzerland N0109881 NTTK18ABE6 Standard IEC Cable – UK N010978 NTTK22ABE6 Standard IEC Cable – Denmark A0814961 A0814961 Standard IEC Cable - Argentina (Note: RoHS not required) N0118951 NTTK26AAE6 Standard IEC Cable - Japan
Product description
required)
CAUTION
TheIP Phone 1140E must be plugged into a10/100-BaseT Ethernet jack. Severe damage occurs if this IP Phone 1140E is plugged into an ISDN connection.
4 Ensure that the location meets the network requirements:
SIP Firmware Release 2.0 for IP Phone 1140E Administration
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a DNS server and a DHCP server with DHCP relay agents installed, configured, and running. Using DHCP and DNS servers with CS 2000 network is recommended but not mandatory.
An Ethernet connection to a network with an appropriate SIP proxy server.
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Preinstallation 27
One of the following file servers used as a Provisioning server: — TFTP server — FTP server — HTTP server
Only a TFTP server can be used for an initial UNIStim-to-SIP Firmware conversion. An IP Phone 1140E with SIP Firmware can operate with a TFTP, FTP, or HTTP file server.
—End—
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Configure the provisioning server
ATTENTION
If you have UNIStim Firmware on your IP Phone 1140E, the firmware must be converted from UNIStim to SIP before you proceed with the following instructions. See the chapter "Upgrade and convert the IP Phone 1140E firmware" (page
79) for instructions on how to convert the firmware on an IP Phone 1140E from
UNIStim to SIP.
If the IP Phone 1140E is installed with SIP Firmware, further SIP Firmware upgrades can be done with a TFTP, an FTP, or an HTTP server.

How provisioning works

Provisioning is performed without interaction with the call server. The IP Phone 1140E with SIP Firmware connects directly with the provisioning server in order to retrieve firmware files and configuration files. In this case, the provisioning server is not to be confused with the IP Client manager on the call server. The methods of provisioning are:
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Automatic provisioning at power-up: After the IP Phone 1140E powers up or is reset, it checks the provisioning server for the latest files.
Provisioning through user interaction: The end user can manually check for updates by pressing the Services Context-sensitive soft key and selecting SystemSystem . In the menu select Check for Updates.
Automatic provisioning at a preconfigured time: The IP Phone 1140E with SIP Firmware checks for updates every 24 hours, at a time specified by a parameter in the device configuration file.
The following steps are taken when provisioning updates occur. The IP Phone 1140E with SIP Firmware:
1. connects to the provisioning server
2. retrievesthe provisioning file (1140eSIP.cfg) from the provisioning server
3. reads and acts upon the content of the provisioning file and decides whether any other file is needed, based on a set of rules. If files need to be downloaded to the IP Phone 1140E, a new file transfer session starts
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30 Configure the provisioning server
for each file to be downloaded. The provisioning file (1140eSIP.cfg) can contain commands that prompt for confirmation before a file is downloaded.

Download the SIP Firmware to the provisioning server

To download the SIP Firmware, follow the steps in the next procedure.

Procedure 3 Downloading SIP Firmware for IP Phone 1140E from the Nortel Web site

Step Action 1
2
Go to w Log on to the Nortel Web site with a valid Nortel User ID and
ww.nortel.com/support.
Password. The Technical Support page appears.
3 4 5
Enter IP Phone 1140E in the Knowledge and Solution Engine box. Select Software in the All types scroll down menu. Press the gray arrow at the end of the Knowledge and Solution
Engine box to obtain the Search Results.
6
From the Search Results, select the appropriate version of the SIP Firmware for the IP Phone 1140E, for example, SIP IP Phone 1140E
Release 0625C39D26.bin.
7
Place the selected firmware on the provisioning server.
—End—
Create the SIP provisioning file on the provisioning server
The provisioning file is downloaded from the provisioning server to the IP Phone 1140E every time the IP Phone 1140E checks for updates. The provisioning file is a clear text file that has the naming convention 1140eSIP.cfg. The following is an example of a provisioning file:
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Figure 4 Sample provisioning file
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Table 4 Provisioning file supported sections
[DEVICE_CONFIG] Device configuration file [LANGUAGE] Downloadable language files (more than one can be specified in
each section) [FW] Firmware image [DIALING_PLAN] Dialing plan [TONES] Downloadable tones (.wav files) [USER_CONFIG] Set specific configuration file
Provisioning is performed using the commands in the 1140eSIP.cfg configuration file. The configuration file can have multiple sections.
Note: The maximum length of a line item in the configuration file is 80 characters. If a line item with more than 80 characters is encountered when parsing the configuration file, the remaining portion of the file following that line item will be ignored.
’#’ is used to indicate a comment. Anything preceeding a ’#’ is a comment.
Each section in the configuration file defines rules for different file types. A section starts with a [SECTION NAME] to specify rules for each file type. For example: [FW].
A section is a mandatory field. Parsing of download rules for each file type starts with finding this key word. Currently, the following sections are supported by the IP Phone 1140E with SIP Firmware:
[FW] section for the firmware load download
[DEVICE_CONFIG] section for the device configuration
[DIALING_PLAN] section for the dialing plan files
[LANGUAGE] section for the language prompts files
[TONES] section for the downloadable tone files
[USER_CONFIG] section for set specific configuration file
Firmware [FW] image files originate from Nortel only and are authenticated during firmware download. If the FW authentication fails, the IP Phone 1140E displays an error message and continues operation with the existing FW image.
Device configuration files are used to set various parameters in the IP Phone 1140E.
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Dialing plan files are used for configuring dialing patterns and the format of originated URIs in the SIP message.
Note: Some call servers do not yet support SIP URI dialing.
Language files are simple text files containing all text prompts used by the IP Phone 1140E. Language files are used for the localization of the IP Phone 1140E without firmware upgrade. Each language file has a header that contains a firmware load version with which this file is associated. Language files are signed by Nortel and are authenticated by the firmware for security reasons.
Tones files are standard in the Telecommunications Standardization sector of the International Telecommunications Union (ITU-T). The set supports custom tone files. The tone files must be WAV files with the following specification: A-law or u-law (8.0 kHz, 8-bit, mono or 16.0 kHz, 16 bit mono). The WAV files can be created and downloaded to the IP Phone 1140E. These files are not authenticated by the IP Phone 1140E.
Set specific configuration files support customizing the set on a per set/user level. Parameters in the device configuration file can be overwritten with a set specific configuration file.
Mandatory keywords in the Provisioning file are:
VERSION [xxxxxx], where xxxxxx is a six- to ten-digit number
representing the version of the file on the server. The version of the module is specified in this field. The command is used for version comparison in AUTO mode. VERSION is mandatory for all sections. In the FW section, the firmware version of the load located on the provisioning server must be entered in this field. For all other sections, VERSION is just a counter that can be incremented if it is necessary to download a new file version.
CAUTION
The version number is stored permanently on the IP Phone 1140E until a higher version number is downloaded or until the version number is deleted using the Srvcs, System, Erase User Data menu selection on the IP Phone 1140E. This prevents a new file from being downloaded if a lower version number is used in the Provisioning File.
DOWNLOAD_MODE [AUTO | FORCED] defines whether the version
is checked. This command is optional. If this command is not present, AUTO mode is used as the default.
AUTO - This mode compares the version of the module from the
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VERSION field and the version of the module version stored in the FLASH memory of the IP Phone 1140E. The file download is initiated
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only if the version specified is higher than the current version stored in the IP Phone 1140E. If the version is not applicable, as in the case of language files, the date of the file must be used for the decision.
FORCED - This mode forces the firmware download process.
FORCED can be used for firmware downgrade procedures. Note: In FORCED or AUTO DOWNLOAD_MODE, the version
number is overwritten with each firmware download.
FILENAME [filename] specifies the file name to be downloaded for
this section. For the language and tone section, the use of multiple filenames is allowed.
CAUTION
The version number stored in the FLASH is permanent until a higher number is downloaded from the Provisioning file or you selectSrvcs, System, Erase User Data on the IP Phone 1140E.
Optional keywords in the Provisioning file are:
PROMPT [YES | NO] is used to indicate if the IP Phone 1140E should
prompt the user for an update before the operation is performed. This command is optional with the default set to NO.
YES - enables the prompt — NO - disables the prompt
PROTOCOL [TFTP | FTP | HTTP] defines the protocol used to download
the file. The IP Phone 1140E with SIP Firmware supports TFTP, FTP and HTTP protocols for file download. This command is optional. If it is not present, the default protocol TFTP is used.
ATTENTION
When using the TFTP protocol to transfer the firmware image, the average round trip time must be < 75 ms. The IP Phone will timeout and abort the firmware download if the total download time is > 10 minutes.
If the average round trip time will be >75 ms, use the FTP or HTTP protocol to transfer the firmware image.
If using FTP or HTTP, SRV_USER_NAME and SRV_USER_PASS are also key words. These commands specify the credential used to login to the file server for file download. If not present, the protocol default
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credential is used (no credential for TFTP and HTTP and anonymous with no password for FTP).
SERVER_IP [address] allows the IP Phone 1140E to connect to the
specified IP address or name of the server for which the file can be downloaded. If the address is not specified, the SERVER_IP that is used is the same SERVER_IP that is used to download the provisioning file.
DELETE_FILES [YES | NO], if present, erases the language and tone files stored in the IP Phone 1140E before new files are downloaded. Otherwise, new files with different names are added without erasing existing files. This command is optional. Note that there is a hard limit of 5 language files and 5 tone files that can be stored in the IP Phone 1140E. When the limits are exceeded, no new file can be accepted for download.
YES - erases the existing language and tone files — NO - does not erase existing language and tone files
SRV_USER_NAME [username] - If the protocol is FTP or HTTP, this
keyword specifies the user name to log on to the server.
SRV_USER_PASS [password] - If the protocol is FTP or HTTP, this
keyword specifies the password to log on to the server.
The downloading of these files is initiated when an IP Phone 1140E is powered on, when an automatic check for updates is invoked, or when you select Srvcs, System, Erase User Data. Any of these actions causes the IP Phone 1140E to contact the provisioning server and attempt to read the Provisioning file. A Soft Reset (Srvcs, System, Reset Phone) does not cause the IP Phone 1140E to retrieve the Provisioning file.

Setting the default language on the IP Phone 1140E to French

To set the default language on a new IP Phone 1140E, or an IP Phone 1140E that has not been logged into by an end user, include the following in the [DEVICE_CONFIG] and [LANGUAGE] sections of the 1140eSIP.cfg configuration file.
[DEVICE_CONFIG] DOWNLOAD_MODE AUTO VERSION 000002 FILENAME DeviceConfig.dat
[LANGUAGE] DOWNLOAD_MODE AUTO VERSION 0000000001 FILENAME French_d24.lng
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The DeviceConfig.cfg file should contain the following. DeviceConfig.cfg
DEF_LANG French_d24 On a new IP Phone 1140E, the language will switch to French after
downloading and processing the configuration files. The login menu will be in French. On a subsequent bootup, the login menu and all boot messages will be in French.
For a new user login, the set will create a new user profile. All menus remain in French. When a new user is created, the default language used is obtained from the DeviceConfig setting and stored as a user preference. After which the user preference for language is always used.
If a user has already logged in and either defaulted or chosen English as the user language preference, changing the configuration files will not affect the users language display.
Create the device configuration file on the provisioning server
After the IP Phone 1140E downloads the provisioning file, the IP Phone 1140E reads the [DEVICE_CONFIG] section and is directed to download the device configuration file.
The device configuration file is a clear text file and the naming convention is defined by the administrator. See the FILENAME keyword in the [DEVICE_CONFIG] section of the SIP provisioning file.
The following is an example of a device configuration file.
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Figure 5 Sample device configuration file
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The next table provides a summary of the commands that can be used in the device configuration file. The exact syntax of each command follows the table.
Table 5 Device configuration commands
Server and network configuration commands
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SIP_DOMAIN1 SIP_DOMAIN2 SIP_DOMAIN3 SIP_DOMAIN4 SIP_DOMAIN5 SERVER_IP1_1
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SERVER_PORT3_2 SERVER_PORT4_1 SERVER_PORT4_2 SERVER_PORT5_1 SERVER_PORT5_2 SERVER_RETRIES1
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Feature configuration commands
SERVER_IP1_2 SERVER_IP2_1 SERVER_IP2_2 SERVER_IP3_1 SERVER_IP3_2 SERVER_IP4_1 SERVER_IP4_2 SERVER_IP5_1 SERVER_IP5_2 SERVER_PORT1_1 SERVER_PORT1_2 SERVER_PORT2_1 SERVER_PORT2_2 SERVER_PORT3_1 VMAIL VMAIL_DELAY AUTOLOGIN_ENABLE
SERVER_RETRIES2 SERVER_RETRIES3 SERVER_RETRIES4 SERVER_RETRIES5 DNS_DOMAIN DEF_USERS DEF_USER1 DEF_USER2 DEF_USER3 DEF_USER4 DEF_USER5 UPDATE_USERS SIP_PING
MAX_INBOX_ENTRIES MAX_OUTBOX_ENTRIES MAX_REJECTREASONS
Feature configuration commands(continued)
AUTO_UPDATE AUTO_UPDATE_TIME AUTO_UPDATE_TIME_
RANGE TRANSFER_TYPE REDIRECT_TYPE ENABLE_PRACK
PROXY_CHECKING ENABLE_BT AUTH_METHOD BANNER FORCE_BANNER DST_ENABLED TIMEZONE_OFFSET FORCE_TIME_ZONE IM_MODE
MAX_CALLSUBJECT MAX_PRESENCENOTE DEF_LANG MAX_IM_ENTRIES MAX_ADDR_BOOK_ENTRIES ADDR_BOOK_MODE DEF_AUDIO_QUALITY HOLD_TYPE ENABLE_3WAY_CALL DISABLE_PRIVACY_UI DISABLE_OCT_ENDDIAL FORCE_OCT_ENDDIAL SNTP_ENABLE SNTP_SERVER MADN_TIMER MADN_DIALOG
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QoS and ToS commands
Tone configuration commands
NAT configuration commands
IM_NOTIFY DEF_DISPLAY_IM CALL_WAITING DISTINCTIVE_RINGING USE_RPORT TOVM_SOFTKEY_ENABLE TOVM_VOICEMAIL_ALIAS TOVM_VOICEMAIL_PARAM MAX_RING_TIME ENABLE_UPDATE DSCP_CONTROL
802.1P_CONTROL DSCP_MEDIA DIAL_TONE RINGING_TONE BUSY_TONE NAT_SIGNALLING NAT_MEDIA
DEFAULT_CFWD_NOTIFY FORCE_CFWD_NOTIFY RTP_MIN_PORT RTP_MAX_PORT SCA_HOLD_BEHAVIOR SCA_APPEARANCES SCA_BROADWORKS EXP_MODULE_ENABLE FORCE_REBOOT PROMPT_ON_LOCATION_OTHER
802.1P_MEDIA DSCP_DATA
802.1P_DATA FASTBUSY_TONE CONGESTION_TONE
STUN_SERVER_IP1 STUN_SERVER_IP2
Voice Quality Monitoring (VQMon) configuration commands
System commands
NAT_TTL
VQMON_PUBLISH VQMON_PUBLISH_IP LISTENING_R_ENABLE LISTENING_R_WARN LISTENING_R_EXCE PACKET_LOSS_ENABLE PACKET_LOSS_WARN
ADMIN_PASSWORD
STUN_SERVER_PORT1 STUN_SERVER_PORT2 PACKET_LOSS_EXCE JITTER_ENABLE JITTER_WARN JITTER_EXCE DELAY_ENABLE DELAY_WARN DELAY_EXCE SESSION_RPT_EN SESSION_RPT_INT
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Device configuration commands
CAUTION
The syntax of the device configuration file is case sensitive. Verify that the commands entered follow the case defined in this document.
Parameters in the device configuration file with empty values are not allowed and will cause write failure.
Server and network configuration commands
SIP_DOMAIN[x] [domain_name] preconfigures the proxy domain
name for all servers. The same configuration can be done through the domain configuration menu on the IP Phone 1140E.
x - the number of the SIP domain number from 1 to 5. — domain_name - the proxy domain name for all servers.
ATTENTION
Note: SIP_DOMAIN[x] is provisioned after user logout.
SERVER_IP[x]_[y]_ip_address] configures the primary and secondary
IP address for each domain, two proxies for each domain. — x - the domain number from 1 to 5. — y - the corresponding primary and secondary IP addresses. y=1
indicates the primary address and y=2 indicates the secondary address.
ip_address - the IP address of the SIP proxy server.
SERVER_PORT[x]_[y] [port_number] configures the signaling ports
for each proxy. — x - the domain number. — y - the corresponding primary and secondary IP addresses. y=1
indicates the primary address and y=2 indicates the secondary address.
port_number - the SIP proxy signaling port (default is 5060).
SERVER_RETRIES[x] [number_of_retries] confirms the number of
retries for each domain. The default number of retries is 3. — x - the domain number from 1 to 5.
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number_of_retries - the number of retry attempts to connect to
the proxy server.
DNS_DOMAIN [domain] is the DNS domain of the IP Phone 1140E.
DEF_USERS[x] [user_name] allows you to enter the default user name
for all domains. When the device configuration file gets downloaded, the default user name is used when logging in.
x - the domain number from 1 to 5. — user_name - the default user name.
UPDATE_USERS [YES | NO] affects the default user names stored in
the IP Phone 1140E. If this flag is set to YES, the default user names are overwritten each time a new device configuration file is downloaded.
YES - the default user names are overwritten each time a new
device configuration file is downloaded.
NO - the default user names are not overwritten each time a new
device configuration file is downloaded.
SIP_PING [YES | NO] The SIP_PING configuration value is used to
maintain server heartbeat detection and to keep a firewall pinhole open. When used for server heartbeat detection, the IP Phone 1140E
periodically pings the SIP Proxy and awaits a response. When three attempts to ping the SIP Proxy fail, the IP Phone 1140E begins a failover process and attempts to connect to the next configured SIP Proxy IP in the same domain.
When a NAT TRAVERSAL method is selected, the SIP_PING configuration value also helps keep a firewall pinhole open.
ATTENTION
Decide carefully whether SIP_PING usage is appropriate for your environment. Even when SIP_PING is not used for NAT TRAVERSAL, it is highly likely that you must keep SIP_PING enabled for server heartbeat detection.
If the IP Phone 1140E is behind a firewall, it is very likely that you must keep SIP_PING enabled, unless an alternate method of keeping the firewall pinhole open is used.
The default value is yes if not specified in the device configuration file. If SIP_PING is changed in the Device configuration file, the IP Phone 1140E must be rebooted for the change to take effect.
YES - enables pinging
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NO - disables pinging
Feature configuration commands
TOVM_SOFTKEY_ENABLE [YES | NO]
The transfer to Voice Mail feature is supported on MCS 5100 servers. — YES - enables the toVM soft key on the phone. — NO - disables the toVM soft key on the phone.
TOVM_VOICEMAIL_ALIAS <string> customizes the user ID of the
SIP URI of the voice mail system. The transfer to Voice Mail feature is supported on MCS 5100 servers.
TOVM_VOICEMAIL_PARAM<string> customizes the parameter name
of the SIP URI of the voice mail system. The transfer to Voice Mail feature is supported on MCS 5100 servers.
SCA_APPEARANCES sets the maximum number of appearances that
will be used for outgoing calls by the Shared Call Appearance (SCA) group. The valid range for this parameter is 2 to 24. The default value is
12. The SCA feature is supported on BroadWorks servers.
SCA_BROADWORKS [YES | NO]
The SCA feature is supported on BroadWorks servers. — YES - activates the Broadsoft Shared Call Appearance (SCA)
feature.
NO - deactivates the Broadsoft Shared Call Appearance (SCA)
feature. This is the default option.
SCA_HOLD_BEHAVIOR [PRIVATE | PUBLIC] sets the default behavior
of the hold button when user determined behavior does not exist. When a user creates a new profile the default behavior is taken from this setting. After the creation of a new profile this configuration setting is not used. The default option is PUBLIC. The SCA feature is supported on BroadWorks servers.
RTP_MIN_PORT
The minimum RTP port value is an integer between 1024 and 65535, exclusive of the restricted SIP ports between 5059 and 5080. The default value is 50000.
RTP_MAX_PORT
The maximum RTP port value is an integer between 1024 and 65535, exclusive of the restricted SIP ports between 5059 and 5080. The default value is 50100.
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Note 1: The RTP port configuration parameters must satisfy the constraints that (RTP_MAX_PORT - RTP_MIN_PORT) is greater than or equal to 100 and less than 1000.
Note 2: If there is a provisioning error, RTP_MIN_PORT is reset to the default value of 50000 and RTP_MAX_PORT is reset to the default value of 50100. An error message will be logged. The SystemConfig file stores 50000 and 50100, rather than the erroneous configuration values, to indicate that the configuration attempt has been rejected.
CALL_WAITING [SPEAKER | STREAM]
SPEAKER - the call waiting tone is played on the phone speaker.
This is the default option.
STREAM - the call waiting tone is injected into the stream played
on the transducer in use for the active call
DISTINCTIVE_RINGING [YES | NO]
YES - turns on the distinctive ringing feature. This is the default
option.
NO - turns off the distinctive ringing feature.
USE_RPORT [YES | NO]
YES - allows the phone to work from behind and/or in front of a
symmetrical NAT with servers and/or clients that support RFC 3581.
NO - disables implementation of support for RFC 3581. This is the
default option. Note: To provision USE_RPORT, the IP Phone 1140 must be
rebooted after the device configuration file is updated. To force a hard reboot after the device configuration file is updated set FORCE_REBOOT YES.
EXP_MODULE_ENABLE [YES | NO]
YES - the set will detect and enable an expansion module. — NO - the set will not detect an expansion module. This is the default
option.
MAX_RING_TIME [x] - an integer between 30 and 600 that sets the
number of seconds for incoming calls to ring before ignoring them. The default value is 120.
ENABLE_UPDATE [YES | NO]
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YES - enables UPDATE message support and adds “UPDATE” to
ALLOW header. This is the default option.
NO - disables UPDATE message support.
Note: ENABLE_UPDATE is provisioned after user logout.
FORCE_REBOOT [YES | NO]
YES - forces hard reboot after device configuration update. — NO - does not force hard reboot after device configuration update.
This is the default option. Note: In order for FORCE_REBOOT to reboot the phone the
VERSION of the device configuration file must be incremented, even if DOWNLOAD_MODE is set to FORCED.
PROMPT_ON_LOCATION_OTHER [YES | NO]
YES - prompt the user to select new location if location “other” was
previously selected.
NO - do not prompt the user to select new location if location “other”
was previously selected. This is the default option.
VMAIL [vmail_number] is the voice mail address which can be the
URI or the DN number of the voice mail server. This command takes a string as a parameter. This is the default link for a new user profile only. Individual users can customize the link through Prefs, Message Options, Voice Mail Settings. This command has no effect on the user profiles after it is created.
vmail_number - the number or URI of the voicemail server.
VMAIL_DELAY[x] is a delay, configured in milliseconds, between when the voice mail server answers the call and the start of dialing the voice mail user ID. The default value is 1000ms.
x - the delay in milliseconds
AUTOLOGIN_ENABLE [YES | NO | USE_AUTOLOGIN_ID] or [1 | 0 | 2] controls whether the set attempts to automatically login to the proxy
server. — YES - turns on the auto login feature. — NO - turns off the auto login feature. — USE_AUTOLOGIN_ID - enables the auto login id feature using
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specified in AUTOLOGIN_PASSWD_KEY01 to register and authenticate. Both userid and password must be specified.
The AUTOLOGIN_ID_KEY01 and AUTOLOGIN_PASSWD_KEY01 parameters are defined in the set specific configuration file.
Note: When using this setting, the user will be prevented from logging out of the phone.
or — 1 - turns on the auto login feature. — 0 - turns off the auto login feature. — 2 - enables the auto login id feature using the userid specified
in AUTOLOGIN_ID_KEY01 and the password specified in AUTOLOGIN_PASSWD_KEY01 to register and authenticate. Both userid and password must be specified.
The AUTOLOGIN_ID_KEY01 and AUTOLOGIN_PASSWD_KEY01 parameters are defined in the set specific configuration file.
Note: When using this setting, the user will be prevented from logging out of the phone.
Note: If auto login id is enabled in the set specific configuration file, it is recommended that AUTOLOGIN_ENABLE be set to either Yes/No or 1/0 in the device configuration file. This recommendation facilitates migrating a set that uses the set specific configuration file to not using the set specific configuration file. The migration to just using the device configuration file can be done by deleting the set specific configuration file. If the device configuration file does not have the matching parameters in the set specific configuration file, the set will continue to use the previously assigned settings after the set specific configuration file is deleted. This recommendation applies to other parameters in the set specific configuration file.
AUTO_UPDATE [YES | NO] is a command to enable or disable the automatic updating of the IP Phone 1140E with SIP Firmware configuration files from the provisioning server. Enabling this command causes the IP Phone 1140E with SIP Firmware to check for updates once every day. The default is disabled.
YES - turns on the AUTO_UPDATE feature. — NO - turns off the AUTO_UPDATE feature.
AUTO_UPDATE_TIME [x] is the actual time in seconds, starting from midnight, before an automatic update occurs. Each IP Phone 1140E adds random numbers to the time specified by this command so every
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IP Phone 1140E does not try to access the provisioning server at the same time. By default the automatic update feature is disabled (see AUTO_UPDATE_RANGE).
x - the time after midnight that the automatic update occurs.
AUTO_UPDATE_TIME_RANGE [x] is the range in hours, from the
AUTO_UPDATE_TIME whereby an IP Phone 1140E checks for updates from the server. The default range is 1 hour.
x-the range in hours when the IP Phone checks for updates from
the server. The range can be from 1 to 6 hours.
TRANSFER_TYPE [MCS | STANDARD] is used to configure the IP
Phone 1140E to activate Nortel conference server-assisted attended transfers, instead of the industry standard method of attended transfers. The default setting is MCS.
MCS - the typical attended transfer used by Nortel proxies. MCS
uses a conference server to do the attended transfer.
STANDARD - the standard method of a transfer. This method does
not involve a conference server.
REDIRECT_TYPE [MCS | RFC3261] is a command used to select
different protocols for set redirection. The default setting is MCS. — MCS - when the IP Phone 1140E receives either 301 (moved
permanently) or 302 (moved temporarily) during registration, it is assumed the IP Phone 1140E is moved to a new MCS 5100 system (proxy+registrar) and all subsequent messages are sent to the new address.
RFC3261 - the IP Phone 1140E assumes that, if during registration,
a 301 (moved permanently) is received, the message contains a new registrar address. The IP Phone 1140E tries to register to the registrar using the existing proxy.
ENABLE_PRACK [YES | NO] PRACK is utilized to make some SIP
messages reliable and requires that an ACK be sent with many SIP messages. ENABLE_PRACK is often utilized to verify that early media is being received. See RFC 3262 for details.
Note 1: ENABLE_PRACK must be set to NO when connected to the MCS 5100 Release 3.5 system.
Note 2: ENABLE_PRACK is provisioned after user logout.
NO - disables PRACK and is the default value. — YES - enables PRACK.
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PROXY_CHECKING [YES | NO] enables and disables extra security
checking when i
ncoming requests are sent to the IP Phone 1140E. The IP Phone 1140E with SIP Firmware always sends requests through an outgoing proxy. However, it is possible, through this configuration, to be able to accept an incoming request directly or through an incoming proxy.
YES - means that the request must come directly from the proxy
server. YES is the default to enable proxy checking.
NO - means the request can be sent directly to the IP Phone 1140E.
(NO is only suitable in a few situations).
ENABLE_BT [YES | NO] is a flag to enable and disable Bluetooth
support in the IP Phone 1140E. — YES - enables Bluetooth. — NO - disables Bluetooth. The default is NO.
Note that the IP Phone 1140E hardware does not support Bluetooth and this command is ignored.
DEF_AUDIO_QUALITY [Low | Medium | High] is a command used for
setting the default audio quality used for each new call. Audio quality can be changed when the call is active. If this command is not present in the configuration file, the IP Phone 1140E uses High quality as its default value. The possible parameters for this command are High, Medium, and Low. If any other parameter is entered or these commands are misspelled, the IP Phone 1140E uses High as the default setting.
The following codecs are used for each selection: — Low - G729 ptime 30. — Medium - G711 ptime 30. — High - G711 ptime 20.
AUTH_METHOD [AUTH | AUTH_INT] is used to configure the SIP
authentication method. — AUTH - only authenticates (username/password)AUTH_INT - authentication plus integrity checking (an MD5 hash of
BANNER [banner_text] preconfigures the banner on the IP Phone 1140E. Use a text string to set the banner. For example, BANNER ABC Company sets the banner to ABC Company. The text string can have a maximum of 24 characters.
banner_text - an ASCII string displayed on the screen of the IP
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FORCE_BANNER [YES | NO] is set by the system administrator
through the configuration file. If FORCE_BANNER is set to YES, the banner from the configuration file is reloaded each time the IP Phone 1140E powers up, even if the user changes the banner manually.
YES - causes the banner set by the administrator to override any
banner set by the user.
NO - allows the user to set the banner.
DST_ENABLED [YES | NO] enables and disables the daylight saving
time mechanism. The time received from the server is GMT and is converted to the proper timezone by the IP Phone 1140E. If the Daylight Saving Time feature is enabled, the IP Phone 1140E automatically calculates the DST time at the appropriate date and converts the time to and from DST. The calculations used are based on the new rules applicable to DST in 2007. The IP Phone 1140E is set up to use the North American DST scheme.
YES - enables daylight savings time. — NO - disables daylight savings time.
TIMEZONE_OFFSET [x] is used to set the current time zone offset from
GMT in seconds. TIMEZONE_OFFSET takes a number as a parameter. For example, TIMEZONE_OFFSET -25200 sets the time zone offset to MST, which is GMT-7 (-7*3600 = -25200 seconds).
Table 6 Time zone offset
Location Time zone offset (seconds)
(GMT-10:00) Hawaii (GMT-09:00) Alaska (GMT-08:00) Pacific time (US and Canada) (GMT-07:00) Mountain time (US and Canada) (GMT-06:00) Central time (US and Canada) (GMT-05:00) Eastern time (US and Canada) (GMT-04:00) Atlantic time (US and Canada) (GMT-03:00) Brasilia, Buenos Aires (GMT+00:00) Greenwich, Dublin, Lisbon, London (GMT+01:00) Amsterdam, Berlin, Rome, Stockholm,
Madrid, Paris
-36000
-32400
-28800
-25200
-21600
-18000
-14400
-10800 0 3600
(GMT+02:00) Athens, Istanbul, Cairo, Helsinki, Jerusalem
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(GMT+03:00) Moscow, St. Petersburg (GMT+05:30) Bombay, Calcutta, Madras, New Delhi (GMT+08:00) Beijing, Chongqing, Hong Kong,
Singapore, Taipei (GMT+09:00) Osaka, Sapporo, Tokyo, Seoul (GMT+10:00) Canberra, Melbourne, Sydney (GMT+12:00) Auckland, Wellington
FORCE_TIME_ZONE [YES | NO] allows you to force the timezone
offset on each user’s IP Phone 1140E. The default is NO. — YES - forces the IP Phone to use the TIMEZONE_OFFSET specified
in the device configuration file.
NO - uses the value stored in the user preferences.
IM_MODE [ENCRYPTED | TEXT | SIMPLE | DISABLED] is used to
configure the mode of instant messaging (IM). The default setting is ENCRYPTED.
ENCRYPTED - instant messages are sent encrypted.TEXT - instant messages are sent as text.
10800 18000 28800
32400 36000 43200
SIMPLE - instant messages are sent using SIP for Instant Messaging
and Presence Leveraging Extensions (SIMPLE) protocol.
DISABLED - instant messaging is turned off and no instant
messages can sent or received.
IM_NOTIFY [YES | NO] is used to turn on or off the Blue LED indicator
upon receipt of an instant message. — YES - the Blue LED functions when an instant message is received. — NO - the Blue LED does not function when an instant message is
received.
Note: If IM_NOTIFY is disabled, the Blue LED continues to operate for other features.
DEF_DISPLAY_IM [YES | NO] enables or disables the display of instant
messages (IM). The default setting is NO. — YES - enables display of IMs. — NO - disables display of IMs.
MAX_INBOX_ENTRIES [x] is used to restrict the maximum number
of inbox entries and takes a number as a parameter. For example,
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MAX_INBOX_ENTRIES 100 limits the number of entries in the inbox to
100. The default limit is 100. — x - the maximum number of in box entries.
MAX_OUTBOX_ENTRIES [x] is used to restrict the maximum number
of outbox entries and takes a number as a parameter. For example, MAX_OUTBOX_ENTRIES 100 limits the number of entries in the outbox to 100. The default limit is 100.
x - the maximum number of outbox entries.
MAX_REJECTREASONS [x] is used to restrict the maximum number of
Call Decline Reasons (Prefs, Feature Options, Call Decline Reasons) and takes a number as a parameter. The default limit is 20.
x - the maximum number of reject reasons.
MAX_CALLSUBJECT [x] is used to restrict the maximum number
of call subjects (Prefs, Feature Options, Call Subject) and takes a number as a parameter. The default limit is 20.
x - the maximum number of call subject reasons.
MAX_PRESENCENOTE [x] is used to restrict the maximum number
of presence notes and takes a number as a parameter. The default limit is 20.
x - the maximum number of presence notes that an IP Phone 1140E
can receive.
DEF_LANG [language] is a command used for setting the default
language. Select one of the supported languages from the language list downloaded. Note that the corresponding language file must be downloaded and stored in the IP Phone 1140E through the [LANGUAGE] section in Provisioning. If the language file is not stored in the IP Phone 1140E, the default language English is used.
language - the default language used.
MAX_IM_ENTRIES [x] is used to set the maximum number of instant
message (IM) entries and takes a number as a parameter. Once the maximum number is reached, the oldest IM is deleted without any user notification. The default limit is 999.
x - the maximum number of instant messages.
MAX_ADDR_BOOK_ENTRIES [x] is used to set the maximum number
of entries in the address book and takes a number as a parameter. The default limit is 100.
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x - the maximum number of address book entries.
ADDR_BOOK_MODE [NETWORK | LOCAL | BOTH] is a command
to choose the address book that is used to search for other users. The default setting is NETWORK.
NETWORK - downloads the user’s address book from the network
(MCS). New address book entries are uploaded to the network.
LOCAL - creates a user address book and stores it locally on the IP
Phone
BOTH - attempts to download a network address book and keep a
copy on the IP Phone. If a network address book is available, the IP Phone 1140E functions as if NETWORK mode has been selected.
HOLD_TYPE [RFC2543 | RFC3261] is used to select the protocol to
hold a call. The default setting is RFC3261. — RFC2543 - RFC 2543 is a standard protocol of the Internet
Engineering Task Force (IETF).
RFC3261 - RFC3261 is a standard protocol of the IETF.
ENABLE_3WAY_CALL [YES | NO] is a flag to enable or disable local
phone-based three-way calling for three-party conferences. — YES - enables local (phone-based) three-way calling for three-party
conferences. YES is the default.
NO - disables local (phone-based) three-way calling.
DISABLE_PRIVACY_UI [YES | NO] is a flag to disable the privacy
setting in UI menus. Disabling the privacy setting in UI menus disables the user ability to set privacy options.
YES - disables the privacy setting in the UI menus. — NO - enables the privacy setting in the UI menus. NO is the default.
DISABLE_OCT_ENDDIAL [YES | NO] is a flag used to configure the
pound (#) key. The default setting is YES. — YES - the pound (#) key initiates dialing when pressed after a
telephone number is entered.
NO - the pound (#) key functions as any other digit or character
on the dial pad typically used in networks that use vertical service codes or access codes.
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FORCE_OCT_ENDDIAL [YES | NO] is a flag used to override attempts
to change the function of the pound (#) key on the Graphical User Interface (GUI). The default setting is NO.
YES - overrides attempts to change the function of the pound (#)
key on the GUI.
NO - does not override a change of the function of the pound (#)
key on the GUI.
SNTP_ENABLE [YES | NO] allows the IP Phone 1140E to obtain the
time and date from an NTP server. The default is NO. The IP Phone 1140E updates the time once every 24 hours from the
NTP server. If the IP Phone 1140E cannot contact the server, the IP Phone 1140E tries every 15 minutes up to a maximum of 6 attempts, and then hourly attempts are made. If SNTP_ENABLE is set to NO the IP Phone 1140E tries to retrieve the time and date from the SIP proxy server. However, not all SIP proxy servers support this method of retrieving the time and date.
YES - enables NTP. — NO - disables NTP.
SNTP_SERVER [ip_address] is the IP address or FQDN of the NTP
server that provides the time and date to the IP Phone 1140E. If this is not specified, the IP Phone does not generate any NTP requests.
ip_address - the IP address of the NTP server in either Fully
Qualified Domain Name (FQDN) or non-FQDN format.
MADN_TIMER [x] is used to set the MADN polling timer interval (the
interval at which the IP Phone 1140E attempts to determine the MADN group of the logged-in user). The minimum value for the polling interval is 900 seconds (15 minutes). The default value is 1800.
x - the time delay (in seconds) between queries to find the MADN
group DN of a user. The minimum value 900.
MADN_DIALOG [YES |NO] is used to set the SIP URI or the GROUP
DN for the subscription to the dialog event. The default value is NO. — YES - subscribes to the dialog event using the SIP URI of the user. — NO - subscribes to the dialog event using the group of the user.
Note: The SIP URI is used to subscribe for MADN support for CS
2000.
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DEFAULT_CFWD_NOTIFY [YES | NO] is used to configure the "ring
splash" which occurs when either local call forwarding (MCS 5100) or network-based call forwarding have been enabled (CS 2000, CS 2100). If this configuration value is enabled, the IP Phone 1140E plays an abbreviated ring tone to remind the user that a call has been forwarded. This configuration value only effects users when their user profile is first created, unless the FORCE_CFWD_NOTIFY flag is also used. The default setting is NO
YES - a brief ring splash plays when a call is forwarded. — NO - the ring splash is not be played.
FORCE_CFWD_NOTIFY [YES | NO ] allows the administrator to force
the behavior of the DEFAULT_CFWD_NOTIFY value on all users who login to the IP Phone 1140E. The default setting is NO.
YES - the DEFAULT_CFWD_NOTIFY configuration value is forced
into effect for the user.
NO - the configuration value is not be forced into effect for the user.
QoS and ToS commands
DSCP_CONTROL [x] is a value entered in decimal format between -1
and 63. If the value is -1, the DSCP value is picked up by the Service Package. The default value is 40.
x - a value from -1 to 63 indicating the DSCP value.
802.1P_CONTROL [x] is a value entered in decimal format between -1
and 7 representing the 802.1P value in the SIP signaling packets. If the value is -1, the 802.1P value is retrieved from the Service Package. The default value is 6.
x - the value from -1 to 7 indicating the 802.1P value.
DSCP_MEDIA [x] is a value entered in decimal format between -1 and
63 representing the DSCP value in the Real-time Transfer Protocol packets. If the value is -1, the DSCP value is retrieved from the Service Package. The default value is 44.
x - a value from -1 to 63 indicating the DSCP value.
802.1P_MEDIA [x] is a value entered in decimal format between -1 and
7 representing the 802.1P value in the IP Phone 1140 Media (RTP) packets. If the value is -1 then the 802.1P value is retrieved from the Service Package is the 802.1 setting for media Real-time Transport Protocol (RTP). The default value is -1.
x - a value from -1 to 7 indicating the 802.1P value.
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DSCP_DATA [x] is a value entered in decimal format between -1 and
63 representing the DSCP value in the provisioning packets. If the value is -1, the DSCP value is retrieved from the Service Package. The default value is 40.
x - a value from -1 to 63 indicating the DSCP value.
802.1P_DATA [x] is a value entered in decimal format between -1 and 7
representing the 802.1P value in the provisioning packets. If the value is -1, the 802.1P value is retrieved from the Service Package. The default value is 6.
x - a value from -1 to 7 indicating the 802.1P value.
Tone configuration commands
DIAL_TONE [frequency1 | frequency2 | on_time | off_time] is used to
select the tone advising the caller that the exchange is ready to receive call information and invites the user to start sending call information. You can select the country specific tone. The default tone is the North American tone.
frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates
a continuous tone.
off_time - the duration when no tone is played.
The following are examples of DIAL_TONE:
1. 350,440;-1 (350 and 440 Hz continuous tone)
RINGING_TONE [frequency1 | frequency2 | on_time | off_time] is
used to select the tone advising the caller that a connection is made and a calling signal is applied to a telephone number or service point. You can select the country specific tone. The default tone is the North American tone.
frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates
a continuous tone.
off_time - the duration when no tone is played. The following are examples of RINGING_TONE:
1. 440,480; 2000,4000 (440 and 480 Hz with 2 seconds on, 4 seconds
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BUSY_TONE [frequency1 | frequency2 | on_time | off_time] is used
to select the tone advising the caller that the telephone number is busy. You can select the country specific tone. The default tone is the North American tone.
frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates
a continuous tone.
off_time - the duration when no tone is played.
FASTBUSY_TONE [frequency1 | frequency2 | on_time | off_time] is
used to select the tone advising the caller that the telephone number is busy. It is fast in cadence or frequency. You can select the country specific tone. The default tone is the North American tone.
frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates
a continuous tone.
off_time - the duration when no tone is played.
CONGESTION_TONE [frequency1 | frequency2 | on_time | off_time]
is used to select the tone advising the caller that the groups of lines or switching equipment necessary for setting up the required call, or for the use of a specific service, are temporarily engaged. You can select the country specific tone. The default tone is the North American tone.
frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates
a continuous tone.
off_time - the duration when no tone is played. The IP Phone 1140E supports using WAV files to replace the ringtone
Frequency/Cadence pattern. For a system-wide setting, the country default values can be used.
NAT configuration commands
NAT_SIGNALLING [NONE | SIP_PING | STUN] indicates the type
of protocol used for NAT traversal in the signaling port. The IP Phone 1140E with SIP Firmware supports two methods of NAT traversal of the signaling path: SIP_PING and STUN.
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NONE - If the value is not set to None, this parameter overrides the
value of the parameter SIP_PING in the device configuration file.
SIP_PING - a Nortel proprietary NAT traversal protocol. Note that
SIP_PING only supports NAT traversal in the signaling port.
STUN - the most common NAT traversal method.
NAT_MEDIA [NONE | STUN] indicates the type of protocol used for
NAT traversal in the media ports. The default is NONE. — NONE - is the default and disables NAT_MEDIA. — STUN - the most common NAT traversal protocol for the media (RTP
and Real-time Control Protocol [RTCP]) port.
x - is the binding lifetime in seconds.
NAT_TTL [x] is used for future development. Currently, the default value is 2 minutes (120 seconds) and SIP IP Phones 1120E/1140E do not process or use the value defined in NAT_TTL [x]. SIP IP Phones 1120E/1140E always ping the ports at regular intervals of 60 seconds regardless of the NAT_TTL value.
ATTENTION
STUN_SERVER_IP1[ip_address] NAT traversal using STUN protocol
requires a STUN server in the public internet. Two STUN server IPs can be provisioned.
ip_address - is the IP address of STUN server 1.
STUN_SERVER IP2[ip_address] NAT traversal using STUN protocol
requires a STUN server in the public internet. Two STUN servers IPs can be provisioned.
ip_address - is the IP address of STUN server 2.
STUN_SERVER_PORT1[port_number] is the port number used
corresponding to STUN_SERVER_IP1. The default port number is
3478. — port_number - is the port number.
STUN_SERVER_PORT2[port_number] is the port number used
corresponding to STUN_SERVER_IP2. The default port number is
3478. — port_number - is the port number.
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VQMon configuration commands
It is important to read "How VQMon works" (page 92) before configuring the VQMON parameters.
VQMON_PUBLISH [YES | NO] is the command that is used to enable
or disable the publish message containing the voice quality monitoring metrics sent to the Proactive Voice Quality Monitoring (PVQMoN) collecting server.
YES - enables VQMoN. — NO - disables VQMoN. NO is the default.
VQMON_PUBLISH_IP [xxx.xxx.xxx.xxx] is used to set the IP address
of the PVQMoN server that collects voice quality monitoring metrics from the publish message.
This IP address is used within the report only.
LISTENING_R_ENABLE [YES | NO] is used to enable or disable the
alerts based on the Listening R Minor and Major Thresholds. The default value is vocoder-dependent using a scale from 1 (lowest quality) to 100 (highest quality). Currently default values are used based on VOCODER on a per call basis as summarized below.
YES - enables the sending of the alert report based on the Listening
R Value.
NO - disables the sending of the alert report based on the Listening
R Value.
VOCODER_G711_ULAW VOCODER_G711_ULAWPLP
VOCODER_G723 VOCODER_FLAG_G723_RATE_53
VOCODER_FLAG_G723_RATE_63 VOCODER_G729
VOCODER_G722 VOCODER_PCM16 VOCODER_PCM8 vqmonVocoderTypeUnknown
LISTENING_R_WARN [xx] is the threshold to send a report on Listening R less than [xx]. The default value is 70. Using 0 will reset it to default based on far end VOCODER.
xx - is an INTEGER value used as threshold.
LISTENING_R_WARN = 80 LISTENING_R_EXCE = 70
LISTENING_R_WARN = 60 LISTENING_R_EXCE = 50
LISTENING_R_WARN = 70 (default if not configured and unknown type) LISTENING_R_EXCE = 60
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LISTENING_R_EXCE [xx] is the threshold to send a report on Listening
R less than [xx]. The default value is 60. Using 0 will reset it to default based on far end VOCODER.
xx - is an INTEGER value used as threshold.
PACKET_LOSS_ENABLE [YES | NO] is used to enable or disable the
alerts based on the packet loss thresholds. Packet loss is the fraction of RTP data packets from the source lost since the beginning of reception. The value is an integer scaled by 256. The range is 1 to 25600.
YES - enables the sending of alert report based on the packet loss — NO - disables the sending of alert report based on the packet loss
PACKET_LOSS_WARN [xx] is the threshold to send a report on Packet
Loss greater than [xx]. The default is 256 (1%). Using 0 will reset the threshold to default.
xx - is an INTEGER value scaled by 256 that is used as threshold.
The range is 1 to 25600.
PACKET_LOSS_EXCE [xx] is the threshold to send a report on Packet
Loss greater than [xx]. The default is 1280 (5%). Using 0 will reset the threshold to default.
xx - is an INTEGER value scaled by 256 that is used as threshold.
The range is 1 to 25600.
JITTER_ENABLE [YES | NO] is used to enable or disable alerts based
on the inter-arrival Jitter on incoming RTP packets inter-arrival time. The value is represented in 1/65536 of a second.
YES - enables the sending of alert report based on jitter detection — NO - disables the sending of alert report based on jitter detection
JITTER_WARN [xx] is the threshold to send a report on Inter-arrival Jitter greater than [xx]. 1 second is broken up into 65535 (0xffff hex) parts. [xx] / 65535 is the threshold in seconds. The default is 3276 (50 ms). Using 0 will reset the threshold to default.
xx - is an INTEGER value used as threshold
JITTER_EXCE [xx] is the threshold to send a report on Inter-arrival
Jitter greater than [xx]. 1 second is broken up into 65535 (0xffff hex) parts. [xx] / 65535 is the threshold in seconds. The default is 32760 (500 ms). Using 0 will reset the threshold to default.
xx - is an INTEGER value used as threshold
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DELAY_ENABLE [YES | NO] is used to enable or disable the alerts
based on the excessive delay detection. This is the one-way delay (including system delay) for the call, measured in milliseconds.
YES - enables Excessive delay detection. — NO - disables Excessive delay detection.
DELAY_WARN [xx] is the threshold to give warning on Excessive Delay greater than [xx]. The default is 150 ms. Using 0 will reset the threshold to default.
xx - is an INTEGER value used as a threshold measured in 1/1000
of a second.
DELAY_EXCE [xx] is the threshold to report unacceptable Excessive
Delay greater than [xx]. The default is 175 ms. Using 0 will reset the threshold to default.
xx - is an INTEGER value used as a threshold measured in 1/1000
of a second.
SESSION_RPT_EN [YES | NO] is used to enable or disable periodic
VQMon session reports. The default is disabled. Both session report enable and session report interval must be set if the
phone firmware has been upgraded to SIP Release 2.0. Otherwise, the SESSION_RPT_INT default of 60 seconds will be used automatically.
YES - enables periodic VQMon session reports. — NO - disables periodic VQMon session reports. Default is NO.
SESSION_RPT_INT [xx] is used to specify the interval for the periodic
VQMon session report in seconds. The minimum acceptable value is 60 seconds. The maximum acceptable value is 600 seconds. The default is 60 seconds.
xx - is an INTEGER value in seconds.
System commands
ADMIN_PASSWORD [password] is used to change the default
administrator password of the IP Phone 1140E that is used for unlocking network menus. The default is 26567*738.
password - the administrator password.
Create the set configuration file on the provisioning server
If the IP Phone 1140E encounters a [USER_CONFIG] section while parsing the 1140eSIP.cfg configuration file, the set will download its set specific configuration file SIP<mac id>.cfg.
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Set specific configuration files support customizing the set on a per set/user level. Parameters in the device configuration file can be overwritten with a set specific configuration file.
Most of the parameters in the set configuration file are saved on the set. Removing a parameter from the set configuration file does not change the parameters saved on a configured set. If a parameter is set only in the set specific configuration file, removing the set specific configuration file will not clear the setting.
Note: If the 1140eSIP.cfg configuration file contains a [USER_CONFIG] section, it is recommended that DOWNLOAD_MODE be set to FORCED. This is a global setting for all sets used to determine if the mac id file should be read. Alternatively, if the user wishes to use DOWNLOAD_MODE set to AUTO, when a change is made to any mac id file the version number should be incremented so that all phones read the file.
Table 7 "Set configuration commands" (page 60) provides a summary of the
commands that can be used in the set configuration file. The syntax of each command is summarized in "Set configuration commands" (page 60).
Table 7 Set configuration commands
Auto login
Set configuration commands
AUTOLOGIN_ID_KEY01 [* | xx] This parameter is located within the
set specific configuration file. This is the id the set will use to register and authenticate. The default user id "user1" will be used, if an id is not supplied and the set has not previously logged in.
* - indicates that the set should use its mac address (lower case)
as the user id
xx - an ASCII string that corresponds to the user id.
Note: To provision AUTOLOGIN_ID_KEY01, the IP Phone 1140 must be rebooted after the set configuration file is updated. To force a hard reboot after the set configuration file is updated set FORCE_REBOOT YES in the device configuration file.
AUTOLOGIN_PASSWD_KEY01 This parameter is located within the
set specific configuration file. There is not a default password. If this is
AUTOLOGIN_ID_KEY01 AUTOLOGIN_PASSWD_KEY01
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blank and AUTOLOGIN_ENABLE is set to USE_AUTOLOGIN_ID (or 2) in the device configuration file, the set will not login.
Note: To provision AUTOLOGIN_PASSWD_KEY01, the IP Phone 1140 must be rebooted after the set configuration file is updated. To force a hard reboot after the set configuration file is updated set FORCE_REBOOT YES in the device configuration file.
Create the Dialing Plan file on the provisioning server
A dialing plan essentially describes the number and pattern of digits that a user dials to reach a particular telephone number. Access codes, area codes, specialized codes, and combinations of the number of digits dialed are all part of a dialing plan.
The purpose of the dialing plan is so that the end user does not have to press the send or pound key (#) to have the IP Phone 1140E with SIP Firmware send the initial message to start the call.
Dialing a telephone number on an IP Phone that supports SIP can be different than dialing a number from a traditional telephone. SIP signaling is communicated through a SIP URI to get to the far end. For example, you can key in the SIP address, jsmith@yourcompany.com to reach John Smith. When the IP Phone 1140E with SIP Firmware receives this address, the dialing plan is bypassed and the IP Phone 1140E simply uses the SIP URI to send a SIP INVITE to jsmith@yourcompany.com (INVITE sip: jsmith@yourcompany.com).
Entering a SIP URI address, however, is inconvenient for an IP Phone with SIP Firmware unless a USB keyboard is attached. Also, the user must explicitly press the send key (or use some method to indicate the end of the URI) to indicate the completion of the SIP address. This is not something that the user is accustomed to in a traditional PBX environment.
The alternative is to use a URI where numbers are used to reach the far end. Using different access codes, the IP Phone with SIP Firmware translates the digits entered into something that the server can understand and remaps the number entered into different URIs. Some of the numbers are mapped as intercom calls, some numbers are mapped as local Public Switched Telephone Network (PSTN) calls, and some numbers are mapped as public long-distant calls.
The issue is that until the IP Phone itself can determine the type of call, no SIP INVITE message is sent. This is where the dialing plan comes into effect. The call type is determined by the dialing plan. Based on the rules defined in the dialing plan, once a match has been identified, the IP Phone 1140E with SIP Firmware sends the invite without the need to press the send key. This behavior closely matches the traditional PBX operation.
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The IP Phone 1140E with SIP Firmware design places no restriction in the format of the SIP URI. The dialing plan is a scheme to match the user experience with traditional PBX operation. It does not restrict the type of URI that the user can use.
The IP Phone 1140E with SIP Firmware uses a dialing plan to recognize a call as an emergency call when it sends an INVITE. The dialing plan can have multiple emergency numbers. See the chapter "Emergency 911 -
Operator control of disconnect" (page 125) for information on the handling
of Emergency 911 calls by the IP Phone 1140E with SIP firmware. The following is an example of a dialing plan.

Figure 6 Sample dialing plan

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Dialing function description

Dialing plan
As most telephone users are used to dialing digits to indicate the address of the destination, there is a need to specify the rule by which digits are transformed into a URI. The IP Phone 1140E with SIP Firmware dialing plan contains two sections delimited by two percent signs (%%).

Figure 7 Sample dialing plan declarations section

In the declaration section, the administrator can define the variables. The variables must start with a dollar ($) sign, followed by a number or a character, such as $1 or $a. There are two variables that are reserved by system. They are as follows:
Create the Dialing Plan file on the provisioning server 63
$$ : used for the collected digits if they match the pattern $t : default timer There must be a domain name defined and the domain name can be
represented by any variable. In Figure 6 "Sample dialing plan" (page 62), the domain name is represented by $n.
The variable definitions take the form:
Figure 8 Sample dialing plan variable definitions
For example: $1="nortel.com" $2="Nortel"
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$3="." $4="com" $5="Nortel.com" $t=10000 (default timer is 10 seconds) $a=Nortel.com
The second section of dialing plan contains the digit map. The digit map section has three subsections that are divided by a separator of two ampersands (&&).

Figure 9 Sample dialing plan digit map section

The first part of a dialing plan contains a pattern defined with DRegex, which is used for matching the dialed number. The patterns are separated by the pipe (|) sign. The second part contains the result string used in the dial step. The third part defines the parameters used by UA in dialing action.
The following parameter is currently defined: t=xxxx: After this timer expires, the number entered is automatically dialed.
The timer starts after the first digit is entered and after it expires, the collected digits are automatically dialed out. xxxx is a decimal number in msec. The default timer is used when t is not specified in the digit map.
For example: X{4} && sip:$$; phone-context=nortel.com;user=phone && t=7000 When the user presses any 4 digits, such as 4567, the following SIP URIs
are generated because of the translation rule: Sip:4567; phone-context=nortel.com;user=phone. The timeout of stopping
the collection of digits is 7 seconds.
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The pound sign (#) at the end of the digit map causes the IP Phone 1140E to dial the matched dialing plan immediately.
DRegex
The Digit Regular Expression (DRegex) syntax is a telephony-oriented mapping of Portable Operating System Interface (POSIX) Extended Regular Expressions (ERE). Users must take care not to confuse the DRegex syntax with POSI EREs as they are not identical. In particular, there are many features of POSIX EREs that DRegex does not support. The dialing plan uses DRegex instead of ERE. The following rules demonstrate the use of DRegex.
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Figure 10 DRegex rules

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Downloadable WAV files
It is possible to customize the ring tones on the IP Phone 1140E with SIP Firmware. Up to five special ring tones can be downloaded from the provisioning server and stored on the IP Phone 1140E. The end user can select which ring tone they would like to implement.
In order to download these special files, the files must reside on the provisioning server and be specified in the SIP provisioning file. For more information, see "Download the SIP Firmware to the provisioning server"
(page 30). The WAV files have a maximum size of 512 KB each for the IP
Phone 1140E. The file format is restricted to ITU-T A-law or u-law (8.0 kHz, 8-bit, mono
or 16.0 kHz, 16 bit mono). After the WAV files are downloaded to the IP Phone 1140E, the WAV file
names appear in Pref, Audio, Tones, Ring Pattern(1 to 8 are standard ring tones, and 9 and above are WAV ring tones) and the WAV ring tones can then be selected to replace the standard ring tones.
Downloadable WAV files 67
For further information about downloadable WAV files, see the SIP Firmware Release 2.0 for IP Phone 1140E User Guide (NN43113-101).
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Configure the DHCP server
Use DHCP to provide partial Device Settings or full Device Settings, as well as to automatically provide Voice VLAN IDs.
Table 8 Characteristics of Partial DHCP mode and Full DHCP mode
Partial DHCP mode Partial DHCP mode retrieves the following network
parameters from the DHCP server: IP address, subnet mask, and default gateway configuration for the IP Phone 1140E.
Full DHCP mode As well as those parameters retrieved in Partial DHCP
mode, Full DHCP also retrieves the following from the DHCP server: DNS server and Provisioning Server.
Auto VLAN Discovery Auto VLAN Discovery retrieves the Voice VLAN ID for
the IP Phone 1140E with SIP Firmware.

Partial DHCP mode

After the IP Phone 1140E with SIP Firmware is configured to operate in Partial DHCP mode, the DHCP server does not need any special configuration to support an IP Phone 1140E with SIP Firmware. The IP Phone 1140E with SIP Firmware receives the following Device Settings parameters from the DHCP server in the following table:
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Table 9 DHCP options in Partial DHCP mode
Parameter requested by the
IP Phone 1140E
Subnet mask The subnet mask of the IP
Phone 1140E.
Router and gateways IP address of the default
gateway of the IP Phone 1140E.
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Description
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DHCP server option
1
3
70 Configure the DHCP server
Lease time Implementation varies
according to DHCP server configuration.
Renewal time Implementation varies
according to DHCP server configuration.
Rebinding interval Implementation varies
according to DHCP server configuration.
Partial DHCP is the response from a DHCP server that is not configured to recognize the Nortel-specific Vendor Class Identifier in option 60 (Nortel-SIP-Phone-A). After the IP Phone 1140E is configured to operate in Partial DHCP mode, the DHCP server needs no special configuration to support an IP Phone 1140E. Using Partial DHCP, an IP Phone 1140E can obtain its IP address, subnet mask, and gateway IP address. The remainder of the configuration information is manually entered on the IP Phone 1140E. This includes the DNS server IP and the Provisioning server address and protocol.

Full DHCP mode

The DHCP server requires special configuration in Full DHCP mode. The IP Phone 1140E with SIP Firmware obtains Device Settings parameters from specially configured DHCP servers.
51
58
59
The IP Phone 1140E with SIP Firmware requests the following Device Settings parameters from the DHCP server:
IP address configuration for the IP Phone 1140E
subnet mask for the IP Phone 1140E IP address
default gateway for the IP Phone 1140E subnet
DNS server
Provisioning Server
It is also possible to obtain the DNS IP automatically using Full DHCP mode without any special configuration of the DHCP server. This means that there is no need to configure the Nortel-specific Vendor Class Identifier (Option
60) on the DHCP server. However, the provisioning server address needs
to be manually defined within the Device Settings menu. This configuration requires toggling to Partial DHCP mode to define the provisioning server address and toggling back to Full DHCP mode after this is complete.
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SIP IP Phone class identifier
A Nortel-SIP-Phone-aware DHCP server is needed for Full DHCP mode. For Full DHCP mode, the DHCP server requires special configuration.
After the DHCP server is configured to recognize the IP Phone 1140E with SIP Firmware as a unique IP Phone, the DHCP server can treat the IP Phone 1140E differently than other DHCP IP Phones. An IP Phone-aware DHCP server can automatically configure Nortel IP Phones by sending all information that the IP Phone requires.
The IP Phone 1140E and the DHCP server communicate using a unique class identifier. After the IP Phone 1140E first sends the DHCP DISCOVER, it includes the Nortel-SIP-Phone-A ASCII string within the Vendor Class Identifier (Option 60). The DHCP server recognizes this special Vendor Class Identifier (Option 60) and sends back OFFER, which also includes the same Vendor Class Identifier. This makes it possible to notify the IP Phone 1140E with SIP Firmware that the server is IP Phone-aware, and that it is safe to accept the offer from the server.
Full DHCP mode 71
Every Nortel IP Phone 1140E with SIP Firmware fills in the Vendor Class ID option of the DHCPDISCOVER and DHCPREQUEST messages with the null-terminated, ASCII-encoded string Nortel-SIP-Phone-A, where A identifies the version number of the information format of the IP Phone.
The Class Identifier Nortel-SIP-Phone-A must be unique in the DHCP server domain.
The unique DHCP configuration is required to allow the DHCP server to respond with a unique Option 66 parameter to the IP Phone 1140E with SIP Firmware.
Note: The DHCP standard defines Option 66 as the bootp server address in a string. The meaning of the bootp server address is extended in Nortel IP Phone 1140E with SIP Firmware to include the provisioning server address. The string in the DHCP offer for Option 66 can be
the numeric IP address or name of the TFTP server or the URI (if
FTP or HTTP protocol is used) of the provisioning server in the form of
<protocol>://<provisioning server URL>
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For example: http://mydomain.com/SIP_phone
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If provisioning server authentication is required, the user credential must be embedded in the URI in the form of
<protocol>://[<userid>:<password>@]<provisioning server URL>[:port][/path]
For example, ftp://www.mydomain.com/ABC or ftp://myuserid:mypass@ftp.mydo-
main.com:21/ABC
Requested Device Settings parameters
Using Full DHCP mode, a SIP IP Phone-aware DHCP server can automatically configure Nortel SIP IP Phones by requesting a list of configuration parameters. The IP Phone 1140E uses DHCP to request and receive the information.
The following table lists the Device Settings parameters requested by the IP Phone 1140E in the Parameter Request List option (Option Code 55) in the DHCPDISCOVER and DHCPREQUEST messages. The DHCPOFFER and the DHCPACK reply messages from the DHCP server must contain the options in the following table.
Table 10 DHCP options in Full DHCP mode
Parameter requested by the
IP Phone 1140E
Subnet mask This is the subnet mask of the
IP Phone 1140E
Router and gateways IP address of the default
gateway of the IP Phone 1140E.
DNS server DNS Server address; only the
first one from the list is used.
Broadcast address This is the Broadcast address
of the subnet. The IP Phone 1140E automatically calculates the Broadcast address if it is not provided.
DNS domain Implementation varies
according to DHCP server.
Lease time Implementation varies
according to DHCP server.
Renewal time Implementation varies
according to DHCP server.
Description
DHCP server option
1
3
6
28
15
51
58
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Rebinding interval Implementation varies
according to DHCP server.
Provisioning server Used for delivering the
provisioning server IP address. This parameter can contain either IP address or a URL to the provisioning server orfolder.

DHCP VLAN Auto Discovery

Configuring a server for Voice VLAN Discovery is optional. This configuration is done in addition to any configuration done for Full DHCP or Partial DHCP and it is required only when you are configuring the VLAN Auto Discovery in the Device Settings menu on the IP Phone 1140E.
Auto Voice VLAN Discovery is only possible in Full and Partial DHCP mode. VLAN Auto Discovery configuration is useful in a network with separate
Data VLAN for traffic (commonly used for PC-to-PC communication) and Voice VLAN for VoIP traffic with different priorities.
The VLAN Auto Discovery is a two-step process:
1. DISCOVER is sent to the DHCP server for available Voice VLAN IDs. The DHCP server sends an OFFER with the available Voice VLAN IDs. If the Data VLAN ID has been manually provisioned in the Device Settings of the IP Phone 1140E, DHCP DISCOVER is tagged with the Data VLAN ID; otherwise, it is untagged.
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66
2. DISCOVER is sent to the DHCP server for all of the DHCP required parameters. However, this DISCOVER is tagged with the Voice VLAN obtained in step 1.
DHCP VLAN Auto Discovery requires a Nortel-SIP-Phone-aware DHCP server. All DHCP requests carry the Vendor Class Identifier, Nortel-SIP-Phone-A, to allow the DHCP server to identify that the requests are coming from an IP Phone 1140E with SIP Firmware.
DHCP Auto Discovery returns Voice VLAN IDs. The DHCP protocol provides no standard option for VLAN ID requests. Separate DHCP vendor-specific entry is needed for DHCP VLAN Auto Discovery to convey the VLAN information to the IP Phone. DHCP VLAN Auto Discovery uses one of the reserved for site-specific use DHCP options for VLAN list retrieval. At least one of the following Nortel site-specific options must be returned by the DHCP server as part of each DHCPOFFER and DHCPACK message for the IP Phone to accept these messages as valid; 43, 128, 131, 144, 157, 188, 191, 205, 219, 223, 232, 247, 251.
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After multiple VLAN IDs are returned from the DHCP server, the IP Phone 1140E tries to connect to each of the VLANs, following the order in which VLAN IDs are specified in the DHCP option.
The format of the field for DHCP VLAN Auto Discovery is: Type, length, and data, described in the following sections.
Type (1 octet)
To avoid the possibility of option types already being used by different vendors, there are fourteen options types supported by the telephone. They are: 128, 131, 144, 157, 188, 191, 205, 219, 223, 232, 247, 251, 247, and
251. Select one value from the type byte list for the DHCPOFFER response.
Nortel recommends using 232, 247, or 251. DHCP option numbers less than 224 are reclaimed by the IETF (RFC 3942). Future changes in the DHCP protocol can force the telephone to stop sending these option requests. Currently, IP Phone 1140E with SIP Firmware does support the remaining listed options to maintain backward compatibility.
Length (1 octet)
The Length value is variable. Count only the number of octets in the data field.
Data (variable number of octets)
ASCII based format: VLAN-A:XXX+YYY+ZZZ. where, VLAN– A: uniquely identifies this as the Nortel DHCP VLAN discovery request. Each VLAN ID is followed by a plus (+) sign if there are more VLAN IDs or a period (.) to terminate the string.
There are a maximum of 10 VLAN IDs that can be configured in the current version.
Once IP Phone 1140E with SIP Firmware receives the DHCP offer containing the site-specific Voice VLAN option, the next DHCPDISCOVERY message is tagged with the Voice VLAN ID.
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Physically install the IP Phone 1140E

Complete instructions to physically install the IP Phone 1140E, including detailed figures and applicable warnings, are given in the document SIP Firmware Release 2.0 for IP Phone 1140E User Guide (NN43113-101).
The steps for installing the IP Phone 1140E are summarized in the following procedure.

Procedure 4 Installing the IP Phone 1140E

Step Action
75
1
Remove the stand cover. Pull upward on the center catch and removethe stand cover. The cable routing tracks are now accessible.
2
Connect the AC power adapter (optional). Connect the adapter to the AC adapter jack in the bottom of the IP Phone 1140E. Form a small bend in the cable, and then thread the adapter cord through the channels in the stand.
3
Install the handset. Connect the end of the handset cable with the short straight section into the handset. Connect the end of the handset cable with the long straight section to the back of the IP Phone 1140E, using the RJ-9 handset jack. Form a small bend in the cable, and then thread the handset cord through the channels in the stand so that it exits behind the handset on the right side, in the handset cord exit in the stand base.
4
Install the headset (optional). If installing a headset, plug the connector into the RJ-9 headset jack on the back of the IP Phone 1140E, and thread the headset cord along with the handset cord through the channels in the stand, so that the headset cord exits the channel.
5
Install the Ethernet cable. Connect one end of the supplied Ethernet cable to the back of the IP Phone 1140E using the RJ-45 connector and thread the network cable through the channel.
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6
7
8
9
10
Install the Ethernet cable connecting the PC to the IP Phone 1140E (optional). If connecting PC Ethernet through the IP Phone 1140E, connect one end of the PC Ethernet cable to the IP Phone 1140E using the RJ-45 connector and thread it through the channel. Connect the other end to the LAN connector on the back of the PC.
Install additional cables. If applicable, plug in optional USB devices. Connect the Ethernet cable to the LAN Ethernet connection. If using an AC power adapter, plug the adapter into an AC outlet.
Wall-mount the IP Phone 1140E (optional). The IP Phone 1140E can be mounted either by: (method A) using the mounting holes on the bottom of the IP Phone 1140E stand, or (method B) using a traditional-style wall-mount box with RJ-45 connector and 15-cm (6-inch) RJ-45 cord (not provided).
Replace the stand cover. Ensure that all cables are neatly routed and press the stand cover into place until a click is heard.
Put the IP Phone 1140E in the wall-mount position (optional). If the IP Phone 1140E is to be mounted on the wall, put it in the wall-mount position by holding the tilt lever and pressing the IP Phone 1140E towards the base until the IP Phone 1140E is parallel with the base. Release the tilt lever and continue to push the IP Phone 1140E towards the base until an audible click is heard. Ensure the IP Phone 1140E is securely locked in position.
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The following figure shows the connections on the IP Phone 1140E.

Figure 11 IP Phone 1140E connections

Physically install the IP Phone 1140E 77
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Upgrade and convert the IP Phone 1140E firmware

Introduction

This chapter describes how to upgrade an IP Phone 1140E with UNIStim firmware to SIP firmware.
In order to upgrade an IP Phone 1140E with UNIStim firmware, first determine if you have the minimum UNIStim firmware release on the IP Phone (0625C39). If your IP Phone 1140E is installed with the minimum version of UNIStim firmware, proceed to the section "Convert UNIStim
Firmware to SIP Firmware on the IP Phone 1140E" (page 87). If your IP
Phone 1140E is not installed with the minimum version of UNIStim firmware, proceed to the section "Upgrade to the minimum UNIStim Firmware" (page
81).
To convert the firmware on the IP Phone 1140E from SIP to UNIStim, see the section "Maintenance and troubleshooting" (page 149).
79

Upgrade the SIP Firmware on the IP Phone 1140E

Use the following procedures to upgrade existing SIP Firmware to new SIP Firmware on the IP Phone 1140E.

Download the SIP Firmware to the provisioning server

To download the SIP Firmware, follow the steps in the next procedure.

Procedure 5 Downloading SIP Firmware for the IP Phone 1140E from the Nortel Web site

Step Action 1
2
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Go to w Log on to the Nortel Web site with a valid Nortel User ID and
Password. The Technical Support page appears.
ww.nortel.com/support.
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3 4
Enter IP Phone 1140E in the Knowledge and Solution Engine box. Select Software in the All types scroll down menu.
5 Press the gray arrow at the end of the Knowledge and Solution
Engine box to obtain the Search Results.
6
From the Search Results, select the appropriate version of the SIP Firmware for the IP Phone 1140E, for example, SIP IP Phone 1140E
Release 0625C39D26.bin.
7
Place the selected firmware on the provisioning server.
Modify the SIP provisioning file
Use the following procedure to modify the SIP provisioning file, which exists on the provisioning server.
Procedure 6 Modifying the SIP provisioning file
Step Action
—End—
1
Under the firmware [FW] section of the SIP Provisioning file, increase the VERSION number (for example 06A5C39d26).
2
Under the firmware [FW] section of the SIP Provisioning file, modify the FILENAME of the new file you want to upload to the IP Phone 1140E.
3
Invoke the upgrade mechanism. Use one of the next three methods to invoke a firmware upgrade on
the IP Phone 1140E with SIP Firmware.
1. Power off and power on the IP Phone 1140E.
2. Select Services, System, Check For Updates on the IP Phone 1140E.
3. Allow for an automatic check for updates to occur. (See AUTO_UPDATE under "Feature configuration commands " (page
42)).
Any of these actions causes the IP Phone 1140E to contact the provisioning server and attempt to read the Provisioning file. A Soft Reset (Srvcs, System, Reset Phone) does not cause the IP Phone 1140E to retrieve the Provisioning file and hence does not cause a firmware upgrade.
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—End—

Upgrade to the minimum UNIStim Firmware

The IP Phone 1140E can be ordered with UNIStim Firmware installed or with SIP Firmware installed. You can convert the firmware on an IP Phone 1140E from UNIStim to SIP. To successfully convert the firmware from UNIStim to SIP, the UNIStim Firmware version on your IP Phone 1140E must be 0625C39 or higher.

Identify the current version of UNIStim Firmware

For a new IP Phone 1140E, follow Procedure 7 "Checking the UNIStim
Firmware version on a new IP Phone 1140E" (page 81) to determine the
version number of the UNIStim Firmware on an IP Phone 1140E. For an IP Phone 1140E already in use, follow Procedure 8 "Checking the
UNIStim Firmware version on an IP Phone 1140E already in use" (page
82) to determine the version number of UNIStim Firmware on an IP Phone
1140E.

Procedure 7 Checking the UNIStim Firmware version on a new IP Phone 1140E

Step Action 1
After assembling the IP Phone 1140E and turning it on, the display on the IP Phone 1140E goes through the following sequence:
Nortel splash screen appears
Nortel sonic sound plays
Nortel banner appears
Following the Nortel banner, the firmware version appears in the display (F/W version).
2
Note the UNIStim Firmware version number and write it down. Compare the version number to the minimum-required UNIStim Firmware version (0625C39).
UNIStim Firmware version names contain numbers and letters. Use the last three characters in a version to compare the version of UNIStim on an IP Phone 1140E (0625C39) with the minimum required version for the upgrade. Note that C23 is greater than C39 and C1B is less than C39.
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If the version number is equal to or higher than 0625C39, go to the section "Convert UNIStim Firmware to SIP Firmware on the IP
Phone 1140E" (page 87).
If the number is lower than 0625C39, go to the section "Upgrade
UNIStim Firmware to the minimum required UNIStim Firmware" (page 84) and follow the instructions to upgrade an IP Phone 1140E
to the minimum-required version of UNIStim Firmware before a conversion to SIP Firmware.

Procedure 8 Checking the UNIStim Firmware version on an IP Phone 1140E already in use

Step Action
—End—
1
Press the Globe/Services key on the IP Phone 1140E twice quickly.
If the admin password prompt appears, enter the password 26567*738.
The Local Tools menu appears:
Table 11 Local Tools menu
1. Preferences
2. Local Diagnostics
3. Network Configuration
4. Lock Menu
To make a selection, press the number associated with the menu item, or use the Navigation key cluster.
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to scroll through the menu items. Press the Select key to select the highlighted menu item.
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Table 12 Using the Navigation key cluster to navigate in the Local Tools menu
Key Action
Down Moves highlight down
Up Moves highlight up
Right Selected current menu item
Left Closes menu
Select key (center of cluster)
To close this menu, use the Quit key.
2
Select 2. Local Diagnostics in the Local Tools menu by pressing the Select key in the Navigation key cluster or by pressing the number 2.
3
Select IP Set and DHCP Information by pressing the Select key in the Navigation key cluster or by pressing the number 2.
4 Use the down arrow in the Navigation key cluster to scroll down
the menu to Firmware Version.
5
Note the UNIStim Firmware version number and write it down. Compare the version number to the minimum-required UNIStim
Firmware version (0625C39). UNIStim Firmware version names contain numbers and letters.
Use the last three characters in a version to compare the version of UNIStim on an IP Phone 1140E (0625C39) with the minimum required version for the upgrade. Note that C23 is greater than C39 and C1B is less than C39.
Selects current menu item
If the version number is equal to or higher than 0625C39, go to the section
"Convert UNIStim Firmware to SIP Firmware on the IP Phone 1140E" (page 87).
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84 Upgrade and convert the IP Phone 1140E firmware
If the number is lower than 0625C39, go to the section "Upgrade UNIStim
Firmware to the minimum required UNIStim Firmware" (page 84) and follow
the instructions to upgrade an IP Phone 1140E to the minimum-required version of UNIStim Firmware before you convert to SIP Firmware.

Upgrade UNIStim Firmware to the minimum required UNIStim Firmware

Use either of the following two methods to upgrade UNIStim Firmware.
1. UFTP download initiated by the server if the server supports this method of upgrading UNIStim Firmware. Refer to the appropriate documentation for your call server for instructions on using this method.
2. TFTP download on bootup.
If necessary, use the following procedure to configure the TFTP server.
Procedure 9 Configuring the TFTP server
Step Action 1
The IP Phone 1140E always executes the TFTP download at bootup if a TFTP IP address is configured on the IP Phone 1140E after being initiated by the telephony call server.
2
Go to the TFTP server and create the 1140e.cfg provisioning file. The 1140e.cfg provisioning file is a clear text file. Create the provisioning file as shown in the next table.
Table 13 Sample 1140e.cfg provisioning file
[FW] DOWNLOAD_MODE FORCED VERSION 0625C23 FILENAME 0625C23.bin
This configuration file forces the firmware download of 0625C23.bin.
3 Download and copy the firmware to the TFTP server directory.
To download the UNIStim Firmware for the IP Phone 1140E from the Nortel Web site:
SIP Firmware Release 2.0 for IP Phone 1140E Administration
Copyright © 2008, Nortel Networks
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1. Go to w
ww.nortel.com/support.
2. Log on to the Nortel Web site with a valid Nortel User ID and Password.
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The Technical Support page appears.
3. Enter IP Phone 1140E in the Knowledge and Solution Engine box.
4. Select SOFTWARE in the ALL TYPES scroll down menu.
5. Press the gray arrow at the end of the Knowledge and Solution Engine box to obtain the Search Results.
6. From the Search Results, select the appropriate version of the UNIStim Firmware for the IP Phone 1140E, for example, IP
Phone 1140E Release 0625C23
7. Place the selected firmware in the correct directory on the provisioning server.
4
In the IP Phone 1140E Network Configuration menu, change the TFTP server address and enter the correct TFTP server address.
This can be the provisioning server as defined in the chapter
"Configure the provisioning server" (page 29).
5
Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1140E.
The IP Phone 1140E downloads the firmware file. The display shows [FW] reading
If the download is successful, the display shows [FW] writing and the blue LED flashes.
After the firmware image is downloaded to the IP Phone 1140E, the display shows [FW] finished..., the blue LED stops flashing, and the IP Phone 1140E resets.
The IP Phone 1140E registers to the TPS with the new firmware version.
If the upgrade is unsuccessful, see the chapter "Maintenance and
troubleshooting" (page 149) in the section Download failures.
—End—
Follow the next procedure to download the minimum required version of UNIStim Firmware automatically through TFTP on bootup.
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Procedure 10 Downloading UNIStim Firmware automatically through TFTP on bootup

Step Action 1
Press the Globe/Services key on the IP Phone 1140E twice quickly. If the admin password prompt appears, enter the password
26567*738
The Local Tools menu appears:
Table 14 Local Tools menu
1. Preferences
2. Local Diagnostics
3. Device Settings
4. Lock Menu
2
Select 3. Device Settings from the Local Tools menu. The Device Settings screen appears.
3 If you are using DHCP, select Full or Partial. If you are manually
configuring the IP address, netmask, and gateway address, select No.
4
If the DHCP option is set to Full or Partial, the IP address is automatically obtained.
5
6
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Configure the TFTP IP address within the IP Phone 1140E Device Settings menu.
This can be the provisioning server as defined in the chapter
"Configure the provisioning server" (page 29).
Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1140E.
The IP Phone 1140E downloads the firmware file. The display shows [FW] reading
If the download is successful, the display shows [FW] writing and the blue LED flashes.
After the firmware image is downloaded to the IP Phone 1140E, the display shows [FW] finished... the blue LED stops flashing, and the IP Phone 1140E resets.
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If the upgrade is unsuccessful, see the chapter "Maintenance and
troubleshooting" (page 149) in the section Download failures.
—End—

Convert UNIStim Firmware to SIP Firmware on the IP Phone 1140E

The IP Phone 1140E can be ordered with UNIStim Firmware installed or with SIP Firmware installed. If an IP Phone 1140E is installed with UNIStim Firmware, it runs with SIP Firmware only if the firmware is converted from UNIStim to SIP. If the procedure to determine the UNIStim version number is complete, and, if necessary, the procedure to upgrade the UNIStim Firmware is complete, an IP Phone 1140E can be converted from UNIStim Firmware to SIP Firmware.
Compare the version number to the minimum required UNIStim Firmware version (0625C39).
UNIStim Firmware version names contain numbers and letters. Use the last three characters in a version to compare the version of UNIStim on an IP Phone 1140E (0625C39) with the minimum required version for the upgrade. Note that C23 is greater than C39 and C1B is less than C39.
The conversion must be performed using TFTP.
WARNING
The TFTP download and upgrade of the Flash memory on the IP Phone 1140E may take a significant amount of time (possibly up to 10 minutes). Do not unplug or reboot the IP Phone 1140E during the process.
The next procedure explains how to download the SIP Firmware from the Nortel Web site.

Procedure 11 Downloading SIP Firmware for the IP Phone 1140E from the Nortel Web site

Step Action 1
Go to w
2 Log on to the Nortel Web site with a valid Nortel User ID and
Password. The Technical Support page appears.
3
Enter IP Phone 1140E in the Knowledge and Solution Engine box.
ww.nortel.com/support.
4
SIP Firmware Release 2.0 for IP Phone 1140E Administration
Copyright © 2008, Nortel Networks
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Select SOFTWARE in the ALL TYPES scroll down menu.
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5
Press the gray arrow at the end of the Knowledge and Solution Engine box to obtain the Search Results.
6
From the Search Results, select the appropriate version of the SIP Firmware for the IP Phone 1140E, for example, IP Phone 1140E
Release 0625C39D26.bin.
7
Place the selected firmware on the provisioning server.
—End—
Follow the next procedure to convert the UNIStim Firmware to SIP Firmware on the IP Phone 1140E.

Procedure 12 Converting UNIStim Firmware to SIP Firmware using TFTP

Step Action 1
2
Run the TFTP server (for example Tftpd32.exe). Place firmware and configuration files in the folder of the TFTP
server (for example 1140e.img F/W file and 1140e.cfg file) that contains the following lines:
Table 15 Sample 1140e.cfg configuration file
[FW] DOWNLOAD_MODE AUTO VERSION 06C25D26.bin FILENAME 1140e.img
3
Configure the IP Phone 1140E Device Settings TFTP IP address to the IP address where your TFTP server is running.
After you are finished the configuration, the IP Phone reboots and sends a request to the TFTP server.
4
Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1140E.
The following messages display on the IP Phone 1140E as the IP Phone 1140E cycles through the conversion process, one after the other:
1. [FW] Reading...
2. [FW] Writing...
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3. [FW] Finished...
The IP Phone 1140E then boots up with SIP Firmware. If the conversion is unsuccessful, see the chapter "Maintenance
and troubleshooting" (page 149).
1. TFTP file transfer takes approximately 15 seconds.
2. File writing takes 2.5 minutes. The IP Phone 1140E displays the message [FW] writingand the blue Data Waiting LED flashes.
3. After the new SIP Firmware writing is finished, the blue LED stops flashing and the telephone displays [FW] finished and then reboots.
4. The first time the SIP Firmware boots, the SIP Firmware performs a Flash File System conversion that takes 2.5 minutes.
—End—
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Voice Quality Monitoring

Feature overview

Proactive Voice Quality Monitoring (PVQMon or VQMon) allows the IP Phone 1140E with SIP Firmware to report voice quality statistics to a server in the network. The IP Phone 1140E with SIP Firmware collects various voice quality statistics, for example, packet loss, and sends the voice quality statistics to the server at regular intervals during a call. A subset of these statistics is also available for the user to view on the IP Phone 1140E by selecting the Audio softkey and then the Monitor Audio Quality menu
item.

VQMon set-up

Configure the following parameters on the IP Phone 1140E with SIP Firmware to connect to the server and send the PVQMon statistics.
1. Enable the feature. To enable the feature, configure the VQMON_PUBLISH parameter in the device configuration file (see
"VQMon configuration commands" (page 57)).
91
2. Configure the IP address of the PVQMon server. Configure the IP address of the PVQMon server in either of the following settings:
a. Set VQMON_PUBLISH_IP through the device configuration file (see
b. Set PVQMon IP in Device Settings (see Table 51 "PVQMon IP
3. Configure the remainder of the VQMon parameters in the device configuration file (see "VQMon configuration commands" (page 57)). These parameters provide threshold information to the IP Phone 1140E with SIP Firmware. A report is sent to the server when these thresholds are exceeded.
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"VQMon configuration commands" (page 57)).
configuration" (page 121))
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92 Voice Quality Monitoring

Server set-up

The IP Phone 1140E with SIP Firmware works with Telchemy server software. The name of the software is SQmediator and is available through Telchemy (w
1.0.

How VQMon works

The IP Phone 1140E with SIP Firmware gathers statistics about the current call when VQMon is enabled. Statistics are also gathered regarding the quality metrics of the current call. The call-related statistics contain condensed information about the SIP Session Description Protocol (SDP), the Call ID, the local and remote address, voice quality-related statistics, Zulu times for start-time and the time the report was sent.
The voice quality-related statistics include jitter, packet loss, delay, burst gap loss, listening R-factor, R-LQ, R-CQ, MOS-LQ and MOS-CQ. See Table
16 "Glossary of RTCP XR metrics" (page 92). More information on each
of these metrics is provided in RFC3611 “RTP Control Protocol Extended Reports (RTCP XR)”.
ww.telchemy.com). The minimum version required is release
When the IP Phone 1140E detects that a particular voice quality metric has exceeded a threshold (defined in the device configuration file), the IP Phone sends a message to the server indicating that there is an issue. If the issue persists then the IP Phone reports another message indicating that there is an exceeded value at regular intervals. This happens continuously until the voice quality metric falls below the threshold value. As well, the IP Phone can send regular reports of the voice quality at time intervals defined in the device configuration file.
Table 16 Glossary of RTCP XR metrics
Burst A period of high packet losses and / or discards.
A burst is calculated in milliseconds.
Conversational R-factor Voice quality metric based on burst packet loss
and vocoder selection.
Delay One way delay which includes end-to-end delay,
jitter buffer delay and packetization delay. Delay is calculated in milliseconds.
Inter-arrival jitter The variation in packet arrival times due to
transmission (routing, queuing delay) through the network. Jitter is calculated in milliseconds.
Listening R-factor Voice quality metric based on burst packet loss,
transmission delay and burst loss.
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How VQMon works 93
MIU Media Information Unit. MIU is a concept from
VQMon. An MIU can be any size down to a 10 millisecond (8 sample) block. An MIU means a frame in the i200x implementation.
MOS Mean Opinion Score. A subjective measurement
of the voice quality of a voice call.
MOS_CQ The VQMon conversational quality MOS score
calculated for a call channel.
MOS_LQ The VQMon listening quality MOS score
calculated for a call channel.
Packet loss rate The percentage of total packets loss versus
packets received.
R-factor A measurement of voice quality based on
network impairments including burst packet loss, delay and encoding/decoding algorithm selection.

End of call report

The SIP phone sends a report using VQMON Publish message to the proxy. The proxy redirects the publish ID described within the report. End of call report is always generated if VQMON is enabled SIP phones do not negotiate or exchange messages with the device defined using PUBLISH_IP options.

Session interval report

The SIP phone can send voice quality reports at time intervals defined in the device configuration file. The minimum and default time interval is 60 seconds. If SIP phones send session interval reports more frequently, then a threshold violation has occurred.

Alert interval report

When a SIP phone detects that a voice quality metric has exceeded a threshold, the SIP phone initiates a timer which sends a message to the server every 5 seconds. When all voice quality metrics fall below the threshold values, the SIP phone stops sending VQMON Publish messages with the report. The alert interval report does not differ from the session interval reports or end of call reports.
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Device Settings on the IP Phone 1140E with SIP Firmware

ATTENTION
An IP Phone 1140E with SIP Firmware displays different menus than the menus displayed on an IP Phone 1140E with UNIStim Firmware.

Introduction

This chapter details how to configure the Device Settings parameters on the IP Phone 1140E with SIP Firmware. This includes items such as the IP address, the subnet mask, and the gateway IP address of the IP Phone 1140E.
The Device Settings parameters are listed in the following table in the order they appear on the Device Settings menu of the IP Phone 1140E with SIP Firmware. Read the section at the end of this chapter that explains how to provision the Device Settings parameters. If you are familiar with the Device Settings parameters, skip the last section in the chapter and proceed to the provisioning instructions.
95
Table 17 Parameters in Device Settings menu
Enable 802.1x (EAP) Device ID Password Enable 802.1ab (LLDP) DHCP
SET IP Net Mask
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Gateway DNS IP1 DNS IP2 Ntwk Port Speed
Auto 10BT 100BT
Ntwk Port Duplex
Auto Force Full
Force Half Disable Voice 802.1Q Voice VLAN
No VLAN
DHCP
LLDP VLAN Name
LLDP MED
Manual VLAN Filter Ctrl Priority bits
Auto
0-7 Media Priority bits
Auto
0-7 Disable PC Port PC Port Speed
Auto
10BT
100BT PC Port Duplex
Auto
Disable Data 802.1Q
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Force Full
Force Half
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Data VLAN
No VLAN
LLDP VLAN NAME
Manual (value
from 1 to 4094) Data Priority bits
Auto
0-7 PC-Port Untag all Cached IP Ignore GARP Provisioning
Server URL
Protocol
Introduction 97
TFTP FTP
PVQMon IP NAT Traversal
HTTP Device ID Password
NAT Signal
None
SIP Ping
STUN NAT Media
None
STUN NAT TTL STUN S1 IP STUN S2 IP
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802.1x (EAP) Port-based network access control

Extensible Authentication Protocol (EAP) supports multiple authentication methods and represents a technology framework that facilitates the adoption of Authentication, Authorization, and Accounting (AAA) schemes, such as Remote Authentication Dial In User Service (RADIUS). RADIUS is defined in RFC 2865. The IP Phone 1140E with SIP Firmware supports only the MD5 authentication method.
802.1x defines the following three roles:
1. Supplicant—an IP Phone that requires access to the network to use network services.
2. Authenticator—the network entry point to which the supplicant physically connects (typically a Layer 2/3 switch). The authenticator acts as the proxy between the supplicant and the authentication server. The authenticator controls access to the network based on the authentication status of the supplicant.
3. Authentication server—performs authentication of the supplicant.
Enable and disable Network-level authentication through the EAP configuration menu.
Authorization
If 802.1x is configured and the IP Phone 1140E is physically connected to the network, the IP Phone 1140E (supplicant) initiates 802.1x authentication by contacting the Layer 2/3 switch (authenticator). The IP Phone 1140E also initiates 802.1x authentication after the Ethernet connection (network interface only) is restored following a network link failure.
However, if the IP Phone 1140E resets, it assumes the Layer 2 link has remained in service and is authenticated.
The IP Phone 1140E fails to authorize if the DeviceID and the IP Phone 1140E passwords do not match the DeviceID and IP Phone 1140E passwords provisioned on the RADIUS Server. The Layer 2 switch (authenticator) locks out the IP Phone 1140E and network access is denied. If this happens during reauthorization, all telephone services are lost. The connected PC operates as normal.
Device ID
The Device ID is for use with the 802.1x (EAP) protocol. If the 802.1x (EAP) is not used, then there is no prompt to enter the Device ID.
Password
The Password is for use with the 802.1x (EAP) protocol. If the 802.1x (EAP) is not used, there is no prompt to enter the Password.
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Introduction 99
802.1ab Link Layer Discovery Protocol
802.1ab Link Layer Discovery Protocol (LLDP) is a standard for discovering
the physical topology between neighboring devices. 802.1ab LLDP defines a standard method for Ethernet network devices, such as switches, routers, and IP Phones to advertise information about themselves to other nodes on the network and to store the information they discover in a Management Information Base (MIB).
802.1ab (LLDP) takes advantage of the VLAN Name and Network Policy
TLVs, and provides an automatic configuration of the phone network policy parameters. Key parameters, such as VLAN ID, L2 priority, and DSCP values are received from the switch and are automatically configured in the phone.
802.1ab Link Layer Discovery Protocol (LLDP) provides the following
functionality
Periodic transmission of advertisements containing device information, device capabilities and media specific configuration information to neighbors attached to the same network.
Reception of LLDP advertisements from its neighbors.
Implementation of behavioral requirements specified by Link Layer Discovery Protocol Media Endpoint Discovery (LLDP-MED).
Storage of received data in local data structures, for example, in MIB modules.
TLVs
The information fields in each MIB are contained in a Link Layer Discovery Protocol Data Unit (LLDPDU) as a sequence of short, variable-length, information elements known as TLVs that each include type, length, and value fields. Each LLDPDU includes several mandatory TLVs plus optional TLVs. Optional TLVs may be inserted in any order.
The IP Phone supports both the transmit and receive LLDP mode.
Transmit direction
An LLDPDU transmitted by the IP Phone supports the following TLVs
1. Chassis ID
2. Port ID
3. Time To Live
4. End of LLDPPDU
5. Port Description
6. System Description
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7. System Capabilities
8. Port VLAN ID
9. Port And Protocol VLAN ID
10. VLAN Name
11. Protocol Identity
12. MAC/PHY Configuration Status
13. Power Via MDI
14. Link Aggregation
15. Maximum Frame Size
16. LLDP-MED Capabilities
17. Network Policy
18. Extended Power-via MDI
19. Inventory Firmware Revision
20. Inventory Manufacturer Name
21. Inventory Model Name
Receive direction
The IP Phone expects to receive the following TLVs
1. Chassis ID
2. Port ID
3. Time To Live
4. End of LLDPPDU
5. System Capabilities
6. VLAN Name
7. MAC/PHY Configuration Status
8. LLDP-MED Capabilities
9. Network Policy
10. Location Identification
Table 18 TLV formats
TLV Fields
Chassis ID Length = 6
Chassis Subtype = 5 [IP Address] Chassis ID = IP Phone IP Address
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