Avaya SIP Firmware for IP Phone 1120E User Manual

SIP Firmware Release 1.1 for IP Phone 1120E Administration

NN43112-300

Document status: Standard Document version: 02.03 Document date: 27 May 2008

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

Other changes 7

How to Get Help 9

Getting Help from the Nortel Web site 9 Getting Help over the telephone from a Nortel Solutions Center 9 Getting Help from a specialist using an Express Routing Code 10 Getting Help through a Nortel distributor or reseller 10

Introduction to this guide 11

Subject 11 Intended audience 11 Acronyms 11 Related publications 13

Overview 15

Introduction 15

Supported SIP proxy servers 20

Revision history 7

SIP overview 15 IP Phone 1120E with SIP Firmware 15 Related documentation 16 Installation overview 17

Before installation 21

Introduction 21 Preinstallation 21

Conﬁgure the provisioning server 25

How provisioning works 25 Download the SIP Firmware to the provisioning server 26 Create the SIP provisioning ﬁle on the provisioning server 26 Create the Device Conﬁguration ﬁle on the provisioning server 31

Device conﬁguration commands 34

Create the Dialing Plan ﬁle on the provisioning server 50

Dialing function description 52

Downloadable WAV ﬁles 56

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4 Contents

Conﬁgure the DHCP server 57

Partial DHCP mode 57 Full DHCP mode 58

Conﬁgure the DHCP server to support Full DHCP mode 59 DHCP VLAN Auto Discovery 61

Physically install the IP Phone 1120E 63 Upgrade and convert the IP Phone 1120E ﬁrmware 67

Introduction 67 Upgrade the SIP Firmware on the IP Phone 1120E 67

Download the SIP Firmware to the provisioning server 67 Modify the SIP provisioning ﬁle 68

Upgrade to the Minimum UNIStim Firmware 69

Identify the current version of UNIStim Firmware 69 Upgrade UNIStim Firmware to the minimum required UNIStim Firmware 72

Convert UNIStim Firmware to SIP Firmware on the IP Phone 1120E 75

Voice Quality Monitoring 79

Feature overview 79 VQMon set-up 79 Server set-up 79 How VQMon works 80

Device Settings on the IP Phone 1120E with SIP Firmware 81

Introduction 81

802.1x (EAP) Port-based network access control 82 Authorization 83 Device ID 83 Password 83 DHCP 83 NO DHCP mode 83 SET IP 84 Net Mask 84 Gateway 84 DNS IP 84 Voice VLAN 84 VLAN Filter 85 Disable PC Port 85 Data VLAN 86 PC-Port Untag all 86 Duplex 87 Ignore GARP 87 Provisioning 87 PVQMon IP 88

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Contents 5

NAT Traversal 88

Conﬁgure the device settings 89

Multiple Appearance Directory Number (Single Call

Arrangement) 97

Vertical services 97

Privacy 98

Feature dependencies and restrictions 98

Emergency 911 - Operator control of disconnect 99

Overview 99 Location information 99 Dialing plan conﬁguration 100 Characteristics of E911 calls 100

IP Phone 1120E restrictions 101

Service package restrictions 101 Distinctive Ringing feature 101

NAT ﬁrewall traversal 103 Three-port switch and VLAN functionality 105

System overview 105

SIP messages supported by the IP Phone 1120E 109

SIP methods 109 SIP responses 110

1xx Response--Information Responses 110 2xx Response--Successful responses 111 3xx Response--Redirection responses 111 4xx Response--Request failure responses 112 5xx Response--Server failure responses 114 6xx Response--Global responses 115

Default error handling 115 SIP header ﬁelds 115 Session description protocol usage 118 SDP and Call Hold 119 Transport layer protocols 119 SIP security authentication 119 SIP DTMF Digit transport 120 Supported subscriptions 120 Supported instant messaging 120

Maintenance and troubleshooting 123

Convert SIP Firmware to UNIStim Firmware 123 IP Phone 1120E diagnostics 124

Index 127

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New in this release

Other changes

For a detailed history of past releases of this document, see "Revision

history" (page 7).

Revision history

May 2008

Standard 02.03. This document is up-issued to reﬂect changes in technical content.

November 2007

Standard 02.02. 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. A change was also made 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 1120E 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 1120E with SIP Firmware.

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8 New in this release

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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

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

•

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

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10 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 to

ww.nortel.com/erc

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 1.1 for IP Phone 1120E Administration

(NN43112-300) describes how to install, conﬁgure, and provision the IP Phone 1120E for use on a SIP network.

Intended audience

This administration guide is intended for system administrators of the Nortel IP Phone 1120E with a basic understanding of SIP. This guide is not intended for end users of the IP Phone 1120E. Many of the tasks outlined in the guide inﬂuence the function of the IP Phone 1120E on the network and require an understanding of telephony and IP networking.

Acronyms

This guide uses the following acronyms:

Table 1 Acronyms used

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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12 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 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 TTL Time-to-live

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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 1.1 for IP Phone 1120E Administration are:

•

SIP Firmware Release 1.1 for IP Phone 1120E Quick Reference Card

(NN43112-105)

•

SIP Firmware Release 1.1 for IP Phone 1120E User Guide

(NN43112-104)

Related publications 13

•

Nortel IP Phone 1100 Series Product bulletins on w

port

ww.nortel.com/sup-

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14 Introduction to this guide

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Overview

Introduction

This chapter describes the hardware and ﬁrmware features of the Nortel IP Phone 1120E 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 1120E with SIP Firmware

The IP Phone 1120E connects to an IP network using an Ethernet connection. All voice and signaling information is converted into IP packets and sent across the network.

The IP Phone 1120E 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 1120E with UNIStim Firmware is different from the model number of an IP Phone 1120E with SIP Firmware.

If you have an IP Phone 1120E with UNIStim Firmware, you can convert the ﬁrmware to SIP Firmware. The most recent version of SIP Firmware can be obtained from Nortel and installed on the IP Phone 1120E.

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 1120E reside. This is not the IP Client Manager (IPCM) of the call server.

•

convert an IP Phone 1120E with UNIStim Firmware to an IP Phone 1120E with SIP Firmware

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16 Overview

•

provision the Device Settings parameters on the IP Phone 1120E with SIP Firmware

Note: Converting the ﬁrmware on an IP Phone 1120E from UNIStim Firmware to SIP Firmware overwrites the UNIStim Firmware. The IP Phone 1120E 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 1120E with SIP Firmware.

Figure 1 IP Phone 1120E with SIP Firmware

Installation overview

To install the IP Phone 1120E with SIP Firmware, three basic steps are required.

Procedure 1 Installation overview

Step Action 1

2 3

Conﬁgure the provisioning server and, optionally, the DHCP server. The function of the provisioning server is to provide conﬁguration options to every IP Phone 1120E 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 1120E with SIP Firmware.

Load SIP Firmware on the IP Phone 1120E. Conﬁgure the initial network-conﬁguration parameters on the IP

Phone 1120E with SIP Firmware.

SIP Firmware Release 1.1 for IP Phone 1120E Administration

—End—

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18 Overview

Figure 2 Installation of IP Phone 1120E with SIP Firmware, page 1 of 2

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Figure 3 Installation of IP Phone 1120E with SIP Firmware, page 2 of 2

Introduction 19

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20 Overview

Supported SIP proxy servers

Call server Call server release Minimum SIP Firmware on

the IP Phone 1120E

MCS 5100

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

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 1120E.

Preinstallation

Complete the following checklist.

Procedure 2 Preinstallation checklist

Step Action 1 Read and become familiar with the SIP Firmware Release 1.1

for IP Phone 1120E User Guide (NN43112-104) and the SIP Firmware Release 1.1 for IP Phone 1120E Quick Reference Card

(NN43112-105).

Ensure there is one IP Phone 1120E boxed package for each IP Phone 1120E being installed.

Table 2 IP Phone 1120E box contents

Item

IP Phone 1120E Graphite with icon keys without PS (SIP) (RoHS)

IP Phone 1120E Graphite with English keys without PS (SIP) (RoHS)

Accessories

Handset, Charcoal NTYS09AA70 Handset cord, Charcoal NTYS10AA70

Ensure that the IP Phone 1120E box includes:

Order number

NTYS03CAE6 N0142351

NTYS03DAE6 N0142352

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22 Before installation

Footstand kit, Charcoal NTYS11AA70 Telephone number label and lens kit NTYS12AA

2.3 m (7 ft) CAT5 Ethernet cable NTYS13AA

The IP Phone 1120E can be powered either by Power Over Ethernet (POE) or powered through an external power supply. Order the external power supply separately.

Table 3 1100 series phones separably orderable parts

Item Power adapter contents include:

Global power supply (optional) N0023000

AC cord package includes one of the following:

1.8 m (5.9 ft), 10 amp, IEC320-C13 North America

3 m (9.9 ft),125 VAC 13 amp, NA power cord, NEMA North America, Middle East, Taiwan, Philippines, Thailand, and Japan

2.4 m (8 ft), 240 VAC 10 amp, ANZ power cord AS-3, Australia, New Zealand

250 VAC, Option 11C Standard European power cord, Other EMEA, Kenya, Korea, Thailand, Indonesia, Vietnam, India, Pakistan

3 m (9.9 ft) 125 VAC, Option 11C Swiss power cord Switzerland

240 VAC, Option 11C UK power cord Hong Kong, Ireland, United Kingdom, Singapore, Malaysia, Bangladesh, Brunei, Sri Lanka

3 m (9.9 ft), 125 VAC, Option 11C Denmark power cord Denmark

Order number

NTYS14AA

NTTK14AB

NTTK15AA

NTTK16AB

NTTK17AB

NTTK18AB

NTTK22AB

SIP Firmware Release 1.1 for IP Phone 1120E Administration

CAUTION

TheIP Phone 1120E must be plugged into a10/100-BaseT Ethernet jack. Severe damage occurs if this IP Phone 1120E is plugged into an ISDN connection.

Ensure that the location meets the network requirements:

•

a DNS server and a DHCP server with DHCP relay agents installed, conﬁgured, and running. It is possible to run the IP

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Preinstallation 23

Phone 1120E without these servers in MCS 5100 Release

3.5, but Nortel recommends using these servers to aid in the deployment and conﬁguration of the IP Phone 1120E with SIP Firmware. 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.

•

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 1120E with SIP Firmware can operate with a TFTP, FTP, or HTTP ﬁle server.

—End—

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24 Before installation

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Conﬁgure the provisioning server

ATTENTION

If you have UNIStim Firmware on your IP Phone 1120E, 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 1120E ﬁrmware" (page

67) for instructions on how to convert the ﬁrmware on an IP Phone 1120E from

UNIStim to SIP.

If the IP Phone 1120E 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 1120E 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:

•

Automatic provisioning at power-up: After the IP Phone 1120E 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 System. In the System menu select Check for Updates.

•

Automatic provisioning at a preconﬁgured time: The IP Phone 1120E 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 1120E with SIP Firmware:

1. connects to the provisioning server

2. retrievesthe provisioning ﬁle (1120eSIP.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 1120E, a new ﬁle transfer session starts

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26 Conﬁgure the provisioning server

for each ﬁle to be downloaded. The provisioning ﬁle (1120eSIP.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 the IP Phone 1120E from the Nortel Web site

Step Action 1

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 1120E 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.

From the Search Results, select the appropriate version of the SIP Firmware for the IP Phone 1120E, for example, SIP IP Phone 1120E

Release 0625C1HD26.bin.

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 1120E every time the IP Phone 1120E checks for updates. The provisioning ﬁle is a clear text ﬁle that has the naming convention 1120eSIP.cfg. The following is an example of a provisioning ﬁle:

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Figure 4 Sample provisioning ﬁle

Create the SIP provisioning ﬁle on the provisioning server 27

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)

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28 Conﬁgure the provisioning server

Provisioning is performed using the commands in the 1120eSIP.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 1120E 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

Firmware [FW] image ﬁles originate from Nortel only and are authenticated during ﬁrmware download. If the FW authentication fails, the IP Phone 1120E 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 1120E.

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 1120E. Language ﬁles are used for the localization of the IP Phone 1120E 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

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Create the SIP provisioning ﬁle on the provisioning server 29

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 1120E. These ﬁles are not authenticated by the IP Phone 1120E.

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 1120E 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 1120E. 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

VERSION ﬁeld and the version of the module version stored in the FLASH memory of the IP Phone 1120E. The ﬁle download is initiated only if the version speciﬁed is higher than the current version stored in the IP Phone 1120E. If the version is not applicable, as in the case of language ﬁles, the date of the ﬁle must be used for the decision.

CAUTION

The version number stored in the FLASH is permanent until a higher number is downloaded from the Provisioning ﬁle or the "Servcs -> System -> Erase User Data-> All Files" is selected on the IP Phone 1120E.

— 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.

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30 Conﬁgure the provisioning server

•

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.

Optional keywords in the Provisioning ﬁle are:

•

PROMPT [YES | NO] is used to indicate if the IP Phone 1120E 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 1120E 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.

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 credential is used (no credential for TFTP and HTTP and anonymous with no password for FTP).

•

SERVER_IP [address] allows the IP Phone 1120E 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 1120E 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 1120E. 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 1120E is powered on, when an automatic check for updates is invoked, or when a "Services ->System -> Check For Updates" is performed. Any of these

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Create the Device Conﬁguration ﬁle on the provisioning server 31

actions causes the IP Phone 1120E to contact the provisioning server and attempt to read the Provisioning ﬁle. ASoft Reset (Srvcs -> System -> Reset Phone) does not cause the IP Phone 1120E to retrieve the Provisioning ﬁle.

Create the Device Conﬁguration ﬁle on the provisioning server

After the IP Phone 1120E downloads the provisioning ﬁle, the IP Phone 1120E 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.

Figure 5 Sample device conﬁguration ﬁle

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32 Conﬁgure the provisioning server

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

SIP_DOMAIN1 SIP_DOMAIN2 SIP_DOMAIN3 SIP_DOMAIN4 SIP_DOMAIN5 SERVER_IP1_1 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_PORT3_2 SERVER_PORT4_1 SERVER_PORT4_2 SERVER_PORT5_1 SERVER_PORT5_2 SERVER_RETRIES1 SERVER_RETRIES2 SERVER_RETRIES3 SERVER_RETRIES4 SERVER_RETRIES5 DNS_DOMAIN DEF_USERS DEF_USER1 DEF_USER2 DEF_USER3 DEF_USER4 DEF_USER5

Feature configuration commands

SIP Firmware Release 1.1 for IP Phone 1120E Administration

SERVER_PORT2_1 SERVER_PORT2_2 SERVER_PORT3_1 VMAIL VMAIL_DELAY AUTOLOGIN_ENABLE AUTO_UPDATE AUTO_UPDATE_TIME AUTO_UPDATE_TIME_

RANGE TRANSFER_TYPE REDIRECT_TYPE

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UPDATE_USERS SIP_PING

MAX_INBOX_ENTRIES MAX_OUTBOX_ENTRIES MAX_REJECTREASONS MAX_CALLSUBJECT MAX_PRESENCENOTE DEF_LANG MAX_IM_ENTRIES MAX_ADDR_BOOK_ENTRIES ADDR_BOOK_MODE

Create the Device Conﬁguration ﬁle on the provisioning server 33

ENABLE_PRACK DEF_AUDIO_QUALITY

Feature configuration commands (continued)

QoS and ToS commands

Tone configuration commands

PROXY_CHECKING AUTH_METHOD BANNER FORCE_BANNER DST_ENABLED TIMEZONE_OFFSET FORCE_TIME_ZONE IM_MODE IM_NOTIFY DEF_DISPLAY_IM

DSCP_CONTROL

802.1P_CONTROL DSCP_MEDIA DIAL_TONE RINGING_TONE BUSY_TONE

HOLD_TYPE ENABLE_3WAY_CALL DISABLE_PRIVACY_UI DISABLE_OCT_ENDDIAL FORCE_OCT_ENDDIAL SNTP_ENABLE SNTP_SERVER MADN_TIMER MADN_DIALOG DEFAULT_CFWD_NOTIFY FORCE_CFWD_NOTIFY

802.1P_MEDIA DSCP_DATA

802.1P_DATA FASTBUSY_TONE CONGESTION_TONE

NAT configuration commands

Voice Quality Monitoring (VQMon) configuration commands

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NAT_SIGNALLING NAT_MEDIA NAT_TTL

VQMON_PUBLISH VQMON_PUBLISH_IP LISTENING_R_ENABLE LISTENING_R_WARN LISTENING_R_EXCE PACKET_LOSS_ENABLE PACKET_LOSS_WARN

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STUN_SERVER_IP1 STUN_SERVER_IP2 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

34 Conﬁgure the provisioning server

Voice Quality Monitoring (VQMon) configuration commands

System commands

Device conﬁguration commands

VQMON_PUBLISH VQMON_PUBLISH_IP LISTENING_R_ENABLE LISTENING_R_WARN LISTENING_R_EXCE PACKET_LOSS_ENABLE PACKET_LOSS_WARN

ADMIN_PASSWORD

PACKET_LOSS_EXCE JITTER_ENABLE JITTER_WARN JITTER_EXCE DELAY_ENABLE DELAY_WARN DELAY_EXCE SESSION_RPT_EN SESSION_RPT_INT

CAUTION

The syntax of the device conﬁguration ﬁle is case sensitive. Verify that the commands entered follow the case deﬁned in this document.

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 1120E.

— x - the number of the SIP domain number from 1 to 5. — domain_name - the proxy domain name for all servers.

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

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Create the Device Conﬁguration ﬁle on the provisioning server 35

— 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. — 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 1120E.

•

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 1120E. 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 1120E

periodically pings the SIP Proxy and awaits a response. When three attempts to ping the SIP Proxy fail, the IP Phone 1120E 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.

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36 Conﬁgure the provisioning server

If the IP Phone 1120E 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 1120E must be rebooted for the change to take effect.

— YES - enables pinging — NO - disables pinging

Feature conﬁguration commands

•

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, set 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] allows the IP Phone to automatically

login after the IP Phone has been restarted, either by the software or by external sources, such as a power failure. A user can be logged out due to the proxy server going down or during a ﬁrmware upgrade.

— YES - turns on the auto login feature. — NO - turns off the auto login feature.

• AUTO_UPDATE [YES | NO] is a command to enable or disable the automatic updating of the IP Phone 1120E with SIP Firmware conﬁguration ﬁles from the provisioning server. Enabling this command causes the IP Phone 1120E 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 1120E adds random numbers to the time speciﬁed by this command so every IP Phone 1120E does not try to access the provisioning server at the

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Create the Device Conﬁguration ﬁle on the provisioning server 37

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 1120E 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 1120E 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 1120E receives either 301 (moved

permanently) or 302 (moved temporarily) during registration, it is assumed the IP Phone 1120E is moved to a new MCS 5100 system (proxy+registrar) and all subsequent messages are sent to the new address.

— RFC3261 - the IP Phone 1120E assumes that, if during registration,

a 301 (moved permanently) is received, the message contains a new registrar address. The IP Phone 1120E 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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38 Conﬁgure the provisioning server

•

PROXY_CHECKING [YES | NO] enables and disables extra security

checking when i IP Phone 1120E 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 1120E.

(NO is only suitable in a few situations).

•

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 1120E 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 1120E uses High as the default setting.

The following codecs are used for each selection:

ncoming requests are sent to the IP Phone 1120E. The

— 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

the entity is also computed and checked).

•

BANNER [banner_text] preconﬁgures the banner on the IP Phone

1120E. 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 12 characters.

— banner_text - an ASCII string displayed on the screen of the IP

Phone 1120E with SIP Firmware.

•

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 1120E powers up, even if the user changes the banner manually.

— YES - causes the banner set by the administrator to override any

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Table 6 Time zone offset

Create the Device Conﬁguration ﬁle on the provisioning server 39

— 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 1120E. If the Daylight Saving Time feature is enabled, the IP Phone 1120E 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 1120E 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).

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 (GMT+02:00) Athens, Istanbul, Cairo, Helsinki,

Jerusalem (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

-36000

-32400

-28800

-25200

-21600

-18000

-14400

-10800 0 3600

7200

10800 18000 28800

32400

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40 Conﬁgure the provisioning server

(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 1120E. 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. — 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.

36000 43200

•

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, MAX_INBOX_ENTRIES 100 limits the number of entries in the inbox to

100. The default limit is 100. — x - the maximum number of inbox entries.

•

MAX_OUTBOX_ENTRIES [x] is used to restrict the maximum number

of outbox entries and takes a number as a parameter. For example,

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Create the Device Conﬁguration ﬁle on the provisioning server 41

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 1120E

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 1120E through the [LANGUAGE] section in Provisioning. If the language ﬁle is not stored in the IP Phone 1120E, 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.

— 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.

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42 Conﬁgure the provisioning server

— 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 1120E 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.

• 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_PRIVACY_UI [YES | NO] is a ﬂag to disable the privacy

setting in UI menus. — 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.

•

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 (#)

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Create the Device Conﬁguration ﬁle on the provisioning server 43

•

SNTP_ENABLE [YES | NO] allows the IP Phone 1120E to obtain the

time and date from an NTP server. The default is NO. The IP Phone 1120E updates the time once every 24 hours from the

NTP server. If the IP Phone 1120E cannot contact the server, the IP Phone 1120E 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 1120E 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 1120E. 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 1120E 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.

•

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 1120E 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.

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44 Conﬁgure the provisioning server

— 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 1120E. 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.

•

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

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Create the Device Conﬁguration ﬁle on the provisioning server 45

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 - is 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 - is 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

off)

•

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.

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— 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 1120E 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_SIGNALING [NONE | SIP_PING | STUN] indicates the type of

protocol used for NAT traversal in the signaling port. The IP Phone 1120E with SIP Firmware supports two methods of NAT traversal of the signaling path: SIP_PING and STUN.

— 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.

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•

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.

• NAT_TTL[x] is used to set the NAT Time-to-live (TTL), the binding lifetime in seconds. This command causes the IP Phone 1120E to ping the ports at regular intervals shorter than the TTL to prevent the NAT from tearing down the ports. The default is 2 minutes (120 seconds).

— x - is the binding lifetime in seconds.

•

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 - 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 - 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.

VQMon conﬁguration commands

•

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.

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•

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.

•

LISTENING_R_ENABLE [YES | NO] is used to enable or disable the

detection of the Listening R Value to measure voice quality based on burst packet loss, transmission delay, and burst loss frequency. LISTENING-R_ENABLE is represented using a scale from 0 (lowest quality) to 100 (highest quality).

— YES - enables the sending of the Listening R Value. — NO - disables the sending of the Listening R Value.

•

LISTENING_R_WARN [xx] is a threshold to report a warning on the

Listening R Value. The default value is vocoder-dependent and is 80. — xx - a value between 1 and 100, used to signal the warning

•

LISTENING_R_EXCE [xx] is the threshold to report an unacceptable

Listening R Value. The default value is vocoder-dependent and is 60. — xx - a value between 1 and 100, used to signal the warning

•

PACKET_LOSS_ENABLE [YES | NO] is a command that enables or

disables packet loss detection. 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 0 to 25600.

— YES - enables packet loss detection — NO - disables packet loss detection

•

PACKET_LOSS_WARN [xx] is the threshold to report a warning on

Packet Loss. The default is 256 (1%). — xx - a value between 1 and 25600.

•

PACKET_LOSS_EXCE [xx] is the threshold to report a warning on

unacceptable Packet Loss. The default is 1280 (5%). — xx - a value between 1 and 25600.

•

JITTER_ENABLE [YES | NO] is used to enable or disable inter-arrival

Jitter detection on incoming RTP packets inter-arrival time. The value is represented in 1/65536 of a second.

— YES - enables jitter detection — NO - disables jitter detection

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•

JITTER_WARN [xx] is the threshold to report a warning on Inter-arrival

Jitter. The default is 3276 (50ms). — xx - any integer value that is 1/65536 of a second.

•

JITTER_EXCE [xx] is the threshold to report unacceptable Inter-arrival

Jitter. The default is 32768 (500ms). — xx - any integer value that is 1/65536 of a second.

•

DELAY_ENABLE [YES | NO] is used to enable or disable 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. The default is 150ms. — xx - any integer value that is 1/65536 of a second.

•

DELAY_EXCE [xx] is the threshold to report unacceptable Excessive

Delay. The default is 175ms. — xx - any integer value that is 1/65536 of a second.

•

SESSION_RPT_EN [YES | NO] is used to enable or disable periodic

VQMon session reports. The default is disabled. — YES - enables periodic VQMon session reports. — NO - disables periodic VQMon session reports.

•

SESSION_RPT_INT [xx] is used to specify the interval for the periodic

VQMon session report in seconds. The default is 60 seconds. — xx - the value in seconds.

System commands

•

ADMIN_PASSWORD [password] is used to change the default

administrator password of the IP Phone 1120E that is used for unlocking network menus. The default is 26567*738.

— password - the administrator password.

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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 # key to have the IP Phone 1120E 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 1120E with SIP Firmware receives this address, the dialing plan is bypassed and the IP Phone 1120E 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 1120E with SIP Firmware sends the invite without the need to press the send key. This behavior closely matches the traditional PBX operation.

The IP Phone 1120E 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.

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The IP Phone 1120E 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 99) for information on the handling of

Emergency 911 calls by the IP Phone 1120E with SIP ﬁrmware. The following is an example of a dialing plan.

Figure 6 Sample dialing plan

Create the Dialing Plan ﬁle on the provisioning server 51

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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 1120E 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:

$$ : 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 51), 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

Create the Dialing Plan ﬁle on the provisioning server 53

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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54 Conﬁgure the provisioning server

The "#" character at the end of the digit map causes the IP Phone 1120E 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

Create the Dialing Plan ﬁle on the provisioning server 55

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Downloadable WAV ﬁles

It is possible to customize the ring tones on the IP Phone 1120E with SIP Firmware. Up to ﬁve special ring tones can be downloaded from the provisioning server and stored on the IP Phone 1120E. 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. (See

"Download the SIP Firmware to the provisioning server" (page 26).) The

WAV ﬁles have a maximum sizeof 128 Kbytes each forthe IP Phone 1120E. 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 1120E, 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.

For further information about downloadable WAV ﬁles, see the SIP Firmware Release 1.1 for IP Phone 1120E User Guide (NN43112-104).

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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 7 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 1120E.

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 1120E with SIP Firmware.

Partial DHCP mode

After the IP Phone 1120E 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 1120E with SIP Firmware. The IP Phone 1120E with SIP Firmware receives the following Device Settings parameters from the DHCP server in the following table:

Table 8 DHCP options in Partial DHCP mode

Parameter requested by the

IP Phone 1120E

Subnet mask The subnet mask of the IP

Phone 1120E.

Router and gateways IP address of the default

gateway of the IP Phone 1120E.

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DHCP server option

58 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 1120E is conﬁgured to operate in Partial DHCP mode, the DHCP server needs no special conﬁguration to support an IP Phone 1120E. Using Partial DHCP, an IP Phone 1120E can obtain its IP address, subnet mask, and gateway IP address. The remainder of the conﬁguration information is manually entered on the IP Phone 1120E. 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 1120E with SIP Firmware obtains Device Settings parameters from specially conﬁgured DHCP servers.

The IP Phone 1120E with SIP Firmware requests the following Device Settings parameters from the DHCP server:

• IP address conﬁguration for the IP Phone 1120E

•

subnet mask for the IP Phone 1120E IP address

•

default gateway for the IP Phone 1120E 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 1120E with SIP Firmware as a unique client, the DHCP server can treat the IP Phone 1120E differently than other DHCP clients. An IP Phone-aware DHCP server can automatically conﬁgure the Nortel IP Phone 1120E by sending all information that the IP Phone requires.

The IP Phone 1120E and the DHCP server communicate using a unique class identiﬁer. After the IP Phone 1120E 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 1120E 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 59

Every Nortel IP Phone 1120E 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 1120E.

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 1120E 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 1120E 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

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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 1120E uses DHCP to request and receive the information.

The following table lists the Device Settings parameters requested by the IP Phone 1120E 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 9 DHCP options in Full DHCP mode

Parameter requested by the

IP Phone 1120E

Subnet mask This is the subnet mask of the

IP Phone 1120E

Router and gateways IP address of the default

gateway of the IP Phone 1120E.

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 1120E 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

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Full DHCP mode 61

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 1120E.

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 1120E, DHCP DISCOVER is tagged with the Data VLAN ID; otherwise, it is untagged.)

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 1120E 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 1120E 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 1120E 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 the IP Phone 1120E 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 1120E

Complete instructions to physically install the IP Phone 1120E, including detailed ﬁgures and applicable warnings, are given in the document SIP Firmware Release 1.1 for IP Phone 1120E User Guide (NN43112-104).

The steps for installing the IP Phone 1120E are summarized in the following procedure.

Procedure 4 Installing the IP Phone 1120E

Step Action

Remove the stand cover. Pull upward on the center catch and removethe stand cover. The cable routing tracks are now accessible.

Connect the AC power adapter (optional). Connect the adapter to the AC adapter jack in the bottom of the IP Phone 1120E. Form a small bend in the cable, and then thread the adapter cord through the channels in the stand.

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 1120E, 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.

Install the headset (optional). If installing a headset, plug the connector into the RJ-9 headset jack on the back of the IP Phone 1120E, and thread the headset cord along with the handset cord through the channels in the stand, so that the headset cord exits the channel.

Install the Ethernet cable. Connect one end of the supplied Ethernet cable to the back of the IP Phone 1120E using the RJ-45 connector and thread the network cable through the channel.

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64 Physically install the IP Phone 1120E

Install the Ethernet cable connecting the PC to the IP Phone 1120E (optional). If connecting PC Ethernet through the IP Phone 1120E, connect one end of the PC Ethernet cable to the IP Phone 1120E 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 1120E (optional). The IP Phone 1120E can be mounted either by: (method A) using the mounting holes on the bottom of the IP Phone 1120E 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 1120E in the wall-mount position (optional). If the IP Phone 1120E is to be mounted on the wall, put it in the wall-mount position by holding the tilt lever and pressing the IP Phone 1120E towards the base until the IP Phone 1120E is parallel with the base. Release the tilt lever and continue to push the IP Phone 1120E towards the base until an audible click is heard. Ensure the IP Phone 1120E is securely locked in position.

SIP Firmware Release 1.1 for IP Phone 1120E Administration

—End—

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The following ﬁgure shows the connections on the IP Phone 1120E.

Figure 11 IP Phone 1120E connections

Physically install the IP Phone 1120E 65

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66 Physically install the IP Phone 1120E

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Upgrade and convert the IP Phone 1120E ﬁrmware

Introduction

This chapter describes how to upgrade an IP Phone 1120E with UNIStim ﬁrmware to SIP ﬁrmware.

In order to upgrade an IP Phone 1120E with UNIStim ﬁrmware, ﬁrst determine if you have the minimum UNIStim ﬁrmware release on the IP Phone (0625C1H). If your IP Phone 1120E is installed with the minimum version of UNIStim ﬁrmware, proceed to the section"Convert UNIStim

Firmware to SIP Firmware on the IP Phone 1120E" (page 75). If your IP

Phone 1120E is not installed with the minimum version of UNIStim ﬁrmware, proceed to the section "Upgrade to the Minimum UNIStim Firmware" (page

69).

To convert the ﬁrmware on the IP Phone 1120E from SIP to UNIStim, see the section "Maintenance and troubleshooting" (page 123).

Upgrade the SIP Firmware on the IP Phone 1120E

Use the following procedures to upgrade existing SIP Firmware to new SIP Firmware on the IP Phone 1120E.

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 1120E from the Nortel Web site

Step Action 1

SIP Firmware Release 1.1 for IP Phone 1120E Administration

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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68 Upgrade and convert the IP Phone 1120E ﬁrmware

3 4

Enter IP Phone 1120E 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.

From the Search Results, select the appropriate version of the SIP Firmware for the IP Phone 1120E, for example, SIP IP Phone 1120E

Release 0625C1HD26.bin.

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—

Under the ﬁrmware [FW] section of the SIP Provisioning ﬁle, increase the VERSION number (for example 06A5C1Hd26).

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 1120E.

Invoke the upgrade mechanism. Use one of the next three methods to invoke a ﬁrmware upgrade on

the IP Phone 1120E with SIP Firmware.

1. Power off and power on the IP Phone 1120E.

2. Perform a "Services ->System -> Check For Updates" on the IP Phone 1120E.

3. Allow for an automatic check for updates to occur. (See AUTO_UPDATE under "Feature conﬁguration commands" (page

36).

Any of these actions causes the IP Phone 1120E 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 1120E to retrieve the Provisioning ﬁle and hence does not cause a ﬁrmware upgrade.

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Upgrade to the Minimum UNIStim Firmware 69

—End—

Upgrade to the Minimum UNIStim Firmware

The IP Phone 1120E can be ordered with UNIStim Firmware installed or with SIP Firmware installed. You can convert the ﬁrmware on an IP Phone 1120E from UNIStim to SIP. To successfully convert the ﬁrmware from UNIStim to SIP, the UNIStim Firmware version on your IP Phone 1120E must be 0625C1H or higher.

Identify the current version of UNIStim Firmware

For a new IP Phone 1120E, follow Procedure 7 "Checking the UNIStim

Firmware version on a new IP Phone 1120E" (page 69) to determine the

version number of the UNIStim Firmware on an IP Phone 1120E. For an IP Phone 1120E already in use, follow Procedure 8 "Checking the

UNIStim Firmware version on an IP Phone 1120E already in use" (page

70) to determine the version number of UNIStim Firmware on an IP Phone

1120E.

Procedure 7 Checking the UNIStim Firmware version on a new IP Phone 1120E

Step Action 1

After assembling the IP Phone 1120E and turning it on, the display on the IP Phone 1120E 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).

Note the UNIStim Firmware version number and write it down. Compare the version number to the minimum-required UNIStim Firmware version (0625C1H).

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 1120E (0625C1H) with the minimum required version for the upgrade. Note that C23 is greater than C1H and C1B is less than C1H.

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70 Upgrade and convert the IP Phone 1120E ﬁrmware

If the version number is equal to or higher than 0625C1H, go to the section "Convert UNIStim Firmware to SIP Firmware on the IP

Phone 1120E" (page 75).

If the number is lower than 0625C1H, go to the section "Upgrade

UNIStim Firmware to the minimum required UNIStim Firmware" (page 72) and follow the instructions to upgrade an IP Phone 1120E

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 1120E already in use

Step Action

—End—

Press the Globe/Services key on the IP Phone 1120E twice quickly.

If the admin password prompt appears, enter the password 26567*738.

The Local Tools menu appears:

Table 10 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.

SIP Firmware Release 1.1 for IP Phone 1120E Administration

to scroll through the menu items. Press the Select key to select the highlighted menu item.

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Upgrade to the Minimum UNIStim Firmware 71

Table 11 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.

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.

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.

Note the UNIStim Firmware version number and write it down. Compare the version number to the minimum-required UNIStim

Firmware version (0625C1H). 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 1120E (0625C1H) with the minimum required version for the upgrade. Note that C23 is greater than C1H and C1B is less than C1H.

Selects current menu item

If the version number is equal to or higher than 0625C1H, go to the section

"Convert UNIStim Firmware to SIP Firmware on the IP Phone 1120E" (page 75).

SIP Firmware Release 1.1 for IP Phone 1120E Administration

—End—

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72 Upgrade and convert the IP Phone 1120E ﬁrmware

If the number is lower than 0625C1H, go to the section "Upgrade UNIStim

Firmware to the minimum required UNIStim Firmware" (page 72) and follow

the instructions to upgrade an IP Phone 1120E 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 1120E always executes the TFTP download at bootup if a TFTP IP address is conﬁgured on the IP Phone 1120E after being initiated by the telephony call server.

Go to the TFTP server and create the 1120e.cfg provisioning ﬁle. The 1120e.cfg provisioning ﬁle is a clear text ﬁle. Create the provisioning ﬁle as shown in the next table.

Table 12 Sample 1120e.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 1120E from the Nortel Web site:

SIP Firmware Release 1.1 for IP Phone 1120E Administration

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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Upgrade to the Minimum UNIStim Firmware 73

The Technical Support page appears.

3. Enter IP Phone 1120E 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 1120E, for example, IP

Phone 1120E Release 0625C23

7. Place the selected ﬁrmware in the correct directory on the provisioning server.

In the IP Phone 1120E 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 25).

Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1120E.

The IP Phone 1120E 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 1120E, the display shows [FW] ﬁnished..., the blue LED stops ﬂashing, and the IP Phone 1120E resets.

The IP Phone 1120E registers to the TPS with the new ﬁrmware version.

If the upgrade is unsuccessful, see the chapter "Maintenance and

troubleshooting" (page 123) 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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74 Upgrade and convert the IP Phone 1120E ﬁrmware

Procedure 10 Downloading UNIStim Firmware automatically through TFTP on bootup

Step Action 1

Press the Globe/Services key on the IP Phone 1120E twice quickly. If the admin password prompt appears, enter the password

26567*738

The Local Tools menu appears:

Table 13 Local Tools menu

1. Preferences

2. Local Diagnostics

3. Device Settings

4. Lock Menu

Select 3. Device Settings from the Local Tools menu. The Device Settings screen appears.

If you are using DHCP, select Full or Partial. If you are manually conﬁguring the IP address, netmask, and gateway address, select No.

If the DHCP option is set to Full or Partial, the IP address is automatically obtained.

SIP Firmware Release 1.1 for IP Phone 1120E Administration

Conﬁgure the TFTP IP address within the IP Phone 1120E Device Settings menu.

This can be the provisioning server as deﬁned in the chapter

"Conﬁgure the provisioning server" (page 25).

Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1120E.

The IP Phone 1120E 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 1120E, the display shows [FW] ﬁnished... the blue LED stops ﬂashing, and the IP Phone 1120E resets.

If the upgrade is unsuccessful, see the chapter "Maintenance and

troubleshooting" (page 123) in the section Download failures.

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Convert UNIStim Firmware to SIP Firmware on the IP Phone 1120E 75

—End—

Convert UNIStim Firmware to SIP Firmware on the IP Phone 1120E

The IP Phone 1120E can be ordered with UNIStim Firmware installed or with SIP Firmware installed. If an IP Phone 1120E 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 1120E can be converted from UNIStim Firmware to SIP Firmware.

Compare the version number to the minimum required UNIStim Firmware version (0625C1H).

WARNING

The TFTP download and upgrade of the Flash memory on the IP Phone 1120E may take a signiﬁcant amount of time (possibly up to 10 minutes). Do not unplug or reboot the IP Phone 1120E 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 1120E from the Nortel Web site

Step Action 1

Go to w Log on to the Nortel Web site with a valid Nortel User ID and

Password. The Technical Support page appears.

3 4

Enter IP Phone 1120E in the Knowledge and Solution Engine box. Select SOFTWARE in the ALL TYPES scroll down menu.

ww.nortel.com/support.

SIP Firmware Release 1.1 for IP Phone 1120E Administration

Press the gray arrow at the end of the Knowledge and Solution Engine box to obtain the Search Results.

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76 Upgrade and convert the IP Phone 1120E ﬁrmware

From the Search Results, select the appropriate version of the SIP Firmware for the IP Phone 1120E, for example, IP Phone 1120E

Release 0625C1HD26.bin.

Place the selected ﬁrmware on the provisioning server.

Follow the next procedure to convert the UNIStim Firmware to SIP Firmware on the IP Phone 1120E.

Procedure 12 Converting UNIStim Firmware to SIP Firmware using TFTP

Step Action 1

Run the TFTP server (for example Tftpd32.exe). Place ﬁrmware and conﬁguration ﬁles in the folder of the TFTP

server (for example 1120e.img F/W ﬁle and 1120e.cfg ﬁle) that contains the following lines:

Table 14 Sample 1120e.cfg conﬁguration ﬁle

—End—

[FW] DOWNLOAD_MODE AUTO VERSION 06C25D26.bin FILENAME 1120e.img

Conﬁgure the IP Phone 1120E 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.

Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1120E.

The following messages display on the IP Phone 1120E as the IP Phone 1120E cycles through the conversion process, one after the other:

1. [FW] Reading...

2. [FW] Writing...

3. [FW] Finished...

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Convert UNIStim Firmware to SIP Firmware on the IP Phone 1120E 77

The IP Phone 1120E then boots up with SIP Firmware. If the conversion is unsuccessful, see the chapter "Maintenance

and troubleshooting" (page 123).

1. TFTP file transfer takes approximately 15 seconds.

2. File writing takes 2.5 minutes. The IP Phone 1120E displays the message [FW] writing… and 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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78 Upgrade and convert the IP Phone 1120E ﬁrmware

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Voice Quality Monitoring

Feature overview

Proactive Voice Quality Monitoring (PVQMon or VQMon) allows the IP Phone 1120E with SIP Firmware to report voice quality statistics to a server in the network. The IP Phone 1120E 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 1120E by selecting the Audio softkey and then the Monitor Audio Quality menu

item.

VQMon set-up

Conﬁgure the following parameters on the IP Phone 1120E 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).

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 PVQMon IP conﬁguration)

3. Conﬁgure the remainder of the VQMon parameters in the device conﬁguration ﬁle (see VQMon conﬁguration commands). These parameters provide threshold information to the IP Phone 1120E with SIP Firmware. A report is sent to the server when these thresholds are exceeded.

Server set-up

The IP Phone 1120E with SIP Firmware works with Telchemy server software. The name of the software is SQmediator and is available through Telchemy (w

1.0.

SIP Firmware Release 1.1 for IP Phone 1120E Administration

VQMon conﬁguration commands).

ww.telchemy.com). The minimum version required is release

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80 Voice Quality Monitoring

How VQMon works

When the IP Phone 1120E 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.

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Device Settings on the IP Phone 1120E with SIP Firmware

ATTENTION

An IP Phone 1120E with SIP Firmware displays different menus than the menus displayed on an IP Phone 1120E with UNIStim Firmware.

Introduction

This chapter details how to conﬁgure the Device Settings parameters on the IP Phone 1120E with SIP Firmware. This includes items such as the IP address, the subnet mask, and the gateway IP address of the IP Phone 1120E.

The Device Settings parameters are listed in the following table in the order they appear on the Device Settings menu of the IP Phone 1120E 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.

Table 15 Parameters in Device Settings menu

Enable 802.1x (EAP) Device ID Password DHCP SET IP Net Mask Gateway DNS IP Voice VLAN VLAN Filter

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82 Device Settings on the IP Phone 1120E with SIP Firmware

Disable PC Port Data VLAN PC-Port Untag all Duplex Ignore GARP Provisioning

Server URL Protocol Device ID

Password PVQMon IP NAT Traversal

NAT Signal

NAT Media

NAT TTL

STUN S1 IP

STUN S2 IP

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 1120E 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.

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Authorization

If 802.1x is conﬁgured and the IP Phone 1120E is physically connected to the network, the IP Phone 1120E (supplicant) initiates 802.1x authentication by contacting the Layer 2/3 switch (authenticator). The IP Phone 1120E also initiates 802.1x authentication after the Ethernet connection (network interface only) is restored following a network link failure.

However, if the IP Phone 1120E resets, it assumes the Layer 2 link has remained in service and is authenticated.

The IP Phone 1120E fails to authorize if the DeviceID and the IP Phone 1120E passwords do not match the DeviceID and IP Phone 1120E passwords provisioned on the RADIUS Server. The Layer 2 switch (authenticator) locks out the IP Phone 1120E 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.

802.1x (EAP) Port-based network access control 83

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.

DHCP

There are three methods of provisioning DHCP.

1. No DHCP (Manual conﬁguration): All the Device Settings parameters are conﬁgured manually on the IP Phone 1120E with SIP Firmware.

2. Partial DHCP: The IP Phone 1120E with SIP Firmware is conﬁgured to get a standard set of Device Settings parameters from the DHCP server.

3. Full DHCP: The IP Phone 1120E with SIP Firmware is conﬁgured to get as many Device Settings parameters as possible from the DHCP server.

NO DHCP mode

No DHCP mode is also known as Manual Device Settings. In this mode, the IP Phone 1120E does not need a DHCP server because no DHCP requests are sent during startup. All necessary parameters must be conﬁgured manually or through the device conﬁguration ﬁle.

The minimum following parameters must be conﬁgured to achieve normal operation:

•

SET IP

•

Net Mask

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84 Device Settings on the IP Phone 1120E with SIP Firmware

•

Gateway

•

DNS IP

•

Provisioning server IP, protocol, and, if user authentication is required to access the provisioning server, the user credentials

Note: TFTP protocol does not require user authentication. Device ID and password are ignored. FTP protocol requires user authentication and the default Device ID is anonymous with no password.

HTTP protocol can operate with or without user authentication. If no authentication is required, make sure to clear the Device ID and password ﬁelds in the conﬁguration dialogue.

SET IP

Select SET IP in the Device Settings menu to conﬁgure the IP address.

Net Mask

Enter a subnet mask In the Net Mask ﬁeld.

Gateway

Enter an IP address of the local gateway in the Gateway ﬁeld.

DNS IP

Conﬁgure the Domain Name Server (DNS) IP.

Voice VLAN

Conﬁgure the Voice VLAN.

Table 16 Telephony Port (incoming)

Voice VLAN Setting

No (default setting) All telephony traffic that is transmitted on the

Auto (Not available if DHCP is set to No) Enable Automatic VLAN Discovery for Voice

Result

telephony port is forwarded untagged

VLAN. Initial DHCP request is forwarded based on the Data VLAN policy (for example, untagged if Data VLAN is disabled, or tagged with the Data VLAN ID if Data VLAN is enabled) to obtain Voice VLAN IDs. Subsequent DHCP requests using the acquired Voice VLAN IDs and the resolution of a Voice VLAN determine the final Voice VLAN ID used for signaling and the media traffic on the telephony port. Note: This means that if a Data VLAN ID is configured in the IP Phone 1120E and the Voice VLAN is set to Auto (Automatic Voice VLAN Discovery), the DHCP

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4095

802.1x (EAP) Port-based network access control 85

server to provide Voice VLAN information must be on the Data VLAN. If the DHCP server is not on the Data VLAN, the IP Phone 1120E fails to receive the DHCP information and fails to register. In addition, another DHCP server must reside in the Voice VLAN to provide the network parameters required. For more information, see the section on DHCP Auto VLAN Discovery.

All telephony traffic that is transmitted on the telephony port has an 802.1q header appended and its Voice VLAN ID is set to the value manually configured here. Place the cursor over the 4095 and press the enter NAV key to modify this value to the desired Voice VLAN tag. Note that the value 4095 is not a valid VLAN ID. This is intentional to ensure a proper VLAN ID is entered.

VLAN Filter

Conﬁgure the VLAN Filter (not available if Voice VLAN is set to No).

Table 17 VLAN Filter

VLAN Filter Setting

Result

Enabled (box is checked) Traffic is forwarded to the IP Phone 1120E port,

based on a review of the MAC address of the packet as well as the 802.1q tag value. Traffic is forwarded through the IP Phone 1120E port, (to the network stack of the telephone), only if the packet matches the MAC address of the IP Phone 1120E and contains the Voice VLAN tag. If the Automatic VLAN Discovery feature is used, the filter is adjusted dynamically as the IP Phone 1120E validates the VLAN suggested by DHCP.

Disabled The VLAN filter is disabled by default. If the

VLAN filter is disabled, traffic is forwarded to the IP Phone 1120E port based only on a review of the MAC address of the packet. The IP Phone 1120E accepts traffic addressed to the MAC address of the IP Phone 1120E as well as any broadcast or multicast packets from any VLAN.

Disable PC Port

With a disabled PC Port, the IP Phone 1120E with SIP Firmware does not receive or send packets to or from the PC port. No device can use the PC port to connect to the network.

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86 Device Settings on the IP Phone 1120E with SIP Firmware

Data VLAN

Conﬁgure the Data VLAN (the VLAN applicable to the PC port). The behavior of the PC port is summarized in the following table.

Table 18 PC Port (incoming trafﬁc from PC)

Data VLAN Result

No All traffic received on the PC port is forwarded

based on the MAC address. The packets are not modified in any way.

4095 (Note that the value 4095 is not a valid VLAN ID. This is intentional to ensure a proper VLAN ID is entered.)

Table 19 PC Port (outgoing)

Data VLAN Result

No All traffic received on the network port and

4095 (Note that the value 4095 is not a valid VLAN ID. This is intentional to ensure a proper VLAN ID is entered.)

All untagged packets received from the PC have an 802.1q header appended and the VLAN ID is set to the value that is manually provisioned in this field. Any packet arriving on the PC port that is already tagged with matching VLAN ID is forwarded as is. Packets tagged with different VLAN IDs are dropped.

telephony port is forwarded to the PC port based on MAC address only. The packets are not modified in any way.

Traffic is forwarded to the PC port based on a review of the MAC address of the packet, as well as the value configured in the Data VLAN field. Traffic is forwarded out the PC port (to the PC) only if the packets contain the Data VLAN tag. Untagged traffic and traffic with a VLAN tag other than the Data VLAN are dropped.

PC-Port Untag all

Conﬁgure PC-Port Untag-All, which conﬁgures PC-Port VLAN Tag Stripping. When enabled, all outgoing trafﬁc to the PC has the VLAN tag removed.

When disabled, packets are sent to the PC as is if Data VLAN is set to No or if Data VLAN is enabled and the VLAN tag matched. Packets with different VLAN ID are discarded.

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Duplex

802.1x (EAP) Port-based network access control 87

In the Network menu, Auto Negotiation mode is the default setting for initial startup. Typically, the IP Phone 1120E is connected to a network that supports Auto Negotiation, and the IP Phone 1120E selects the best speed and duplex mode available. There is no intervention required under normal operation.

ATTENTION

Nortel recommends that Auto Negotiation mode is used on the network and the IP Phone 1120E. Use Full Duplex mode only when the entire network is running in Full Duplex mode. Otherwise, a duplex mismatch results.

Note: Changing the speed, or the duplex mode, or both on the IP Phone 1120E changes both the LAN Ethernet port and PC Ethernet Port interfaces.

If the IP Phone is connected to a network conﬁgured for Full Duplex mode only, the IP Phone 1120E cannot automatically negotiate the proper conﬁguration. Therefore, in this instance, to allow the IP Phone 1120E to work at the optimum speed and duplex mode, Full Duplex mode must be enabled.

Ignore GARP

Gratuitous Address Resolution Protocol (GARP) Protection prevents the IP Phone 1120E with SIP Firmware from GARP spoof attacks on the network.

The IP Phone 1120E with SIP Firmware provides the ability to ignore GARP messages.

Provisioning

The next four menu items apply to conﬁguring the provisioning parameters.

Server URL

Enter the numeric IP address, name, or URL of the provisioning server in the Server URL box.

Protocol

Select the protocol used to access the provisioning server, either TFTP, FTP, or HTTP.

Device ID

Enter the Device ID (User ID) used by the provisioning server for authentication of the IP Phone 1120E with SIP Firmware.

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Password

Enter the password used by the provisioning server for authentication of the IP Phone 1120E with SIP Firmware.

PVQMon IP

If the Proactive Voice Quality Monitor (PVQMon) server is available on the network, enter the PVQMon IP address.

NAT Traversal

The next ﬁve menu items apply to conﬁguring the NAT Traversal Method if the IP Phone 1120E with SIP Firmware is connected to the network through a Network Address Translation device or a ﬁrewall.

NAT Signal

The IP Phone 1120E with SIP Firmware supports two methods of NAT traversal of signaling path: SIP_PING and STUN.

The default NAT traversal method is None. SIP_PING is a legacy Nortel Proprietary protocol for NAT Traversal for SIP

signaling only. STUN is an Internet standard for NAT traversal. If the value for NAT traversal is not set to None, this parameter overrides

the value of the parameter SIP_PING speciﬁed by the device conﬁguration ﬁle for NAT_SIGNALLING.

If the value for NAT traversal is set to None, the value of SIP_PING, if speciﬁed by the device conﬁguration ﬁle for NAT_SIGNALLING, is used instead.

NAT_SIGNALLING is required for networks that use STUN or SIP_PING for NAT traversal.

NAT Media

The IP Phone 1120E with SIP Firmware supports STUN for media path the NAT traversal. The NAT Media feature can be disabled by setting the NAT_Media ﬁeld in Device Settings menu to NONE.

Note: STUN protocol cannot coexist with Application Layer Gateway (ALG), Media Portals, or RTP Proxy servers. If STUN is selected, ensure none of these devices are conﬁgured in the SIP Proxy server.

NAT TTL

Enter the Time to Live (TTL) in seconds for the NATport if STUN is enabled. The Nortel IP Phone 1120E with SIP Firmware pings the open ports on an interval less than the TTL to prevent the NAT from tearing down the ports.

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STUN S1 IP

Enter an IP address for the STUN S1 IP device. The STUN server must reside in the public Internet for STUN protocol to be effective.

STUN S2 IP

Enter an IP address for the STUN S2 IP device. The STUN server must reside in the public Internet for STUN protocol to be effective.

Conﬁgure the device settings

This section describes how to provision the Device Settings parameters regardless of DHCP mode (Full DHCP, Partial DHCP, or No DHCP). Parameters that are not needed in Full or Partial DHCP mode are grayed out on the IP Phone 1140E screen and cannot be modiﬁed.

Follow the next procedure to provision the Device Settings parameters on the IP Phone 1140E with SIP Firmware.

Procedure 13 Provisioning the Device Settings parameters

Step Action

Conﬁgure the device settings 89

Press the Globe/Services key on the IP Phone 1140E quickly twice. The Network menu appears:

Table 20 Network menu

1. Server Settings

2. Device Settings

3. Diagnostics

4. Lock

Table 21 Using the Navigation key cluster to navigate in the Network menu

Key Action

Down Moves highlight down Up Moves highlight up Right Selected current menu item Left Closes menu

To navigate in the Network menu, use the Navigation key cluster:

Select key (center of cluster)

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Selects current menu item

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Menu items can also be accessed through the IP Phone 1140E keypad. Press the number (1-4) corresponding to a menu item to highlight that menu item. Press the Select key to select the highlighted menu item.

To close this menu, use the Quit key.

3 In the Network, choose 2. Device Settings. 4 5

Table 22 Using the Navigation key cluster to navigate in the Device Settings menu

Key Action

Down Opens list or moves highlight down list. Up Moves highlight up list. Right Selects the next item or moves the cursor right in

Left Selects the previous item or deletes a character

Select key (center of cluster)

Enter the admin password. Press the OK Context-sensitive soft key. Use the Navigation key cluster to edit an item:

an edit item.

in the edit field. Selects the highlighted item in the combo box or

ends edit mode.

In Edit mode, the ﬁrst ﬁeld of the item is highlighted. Press the Apply&Reset Context-sensitive soft key to save settings in the Device Settings menu after all the necessary changes are made. This action resets the IP Phone 1140E.

Press the Return Context-sensitive soft key to exit the Device Settings menu without saving any changes.

Table 23 EAP conﬁguration

Enable 802.1x (EAP): Device ID: Password:

SIP Firmware Release 1.1 for IP Phone 1120E Administration

Navigate to the ﬁrst item in the Device Settings menu to conﬁgure EAP.

A check mark appears in the check box if the item is active. If the Enable 802.1x (EAP) check box is checked, ﬁll in the Device

ID and the Password.

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Conﬁgure the device settings 91

Table 24 DHCP conﬁguration

DHCP : No

Conﬁgure DHCP in the Device Settings menu. Use the right arrow in the Navigation key cluster to highlight the

DHCP box. Press the Select key to access a list of DHCP mode choices:

•

Partial

•

Full

Scroll through the list with the Up/Down arrows and select the appropriate parameter. Press the Select key to select No.

No Partial Full

Select SET IP in the Device Settings menu to conﬁgure the IP address.

Table 25 SET IP conﬁguration

SET IP : 0.0.0.0

If DHCP is disabled, conﬁgure Net Mask and Gateway. Enter a subnet mask in the Net Mask ﬁeld and enter the IP address of the local gateway in the Gateway ﬁeld.

Table 26 Net Mask and Gateway conﬁguration

Net Mask : 0.0.0.0 Gateway : 0.0.0.0

Table 27 DNS conﬁguration

DNS IP : 0.0.0.0

To conﬁgure the DNS, enter the IP address of the local DNS server.

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92 Device Settings on the IP Phone 1120E with SIP Firmware

12 13

Choose one of the following for Voice VLAN: No, Auto or 4095. Enable or disable the VLAN Filter. Press the Select key to enable

the VLAN Filter. A check mark in the VLAN Filter box indicates that the VLAN Filter

is enabled.

Table 28 VLAN Filter conﬁguration

VLAN Filter :

Disable PC Port conﬁguration. Press the Select key to turn off the PC port on the IP Phone 1140E.

A check mark in the Disable PC Port box indicates the PC port is disabled.

Disable PC Port can be used in an environment where administrators do not want users accessing the data network through the built-in PC port.

Table 29 Disable PC Port conﬁguration

Disable PC Port :

15 Conﬁgure the Data VLAN.

It is possible to Enable or disable separate VLAN for anything other than voice and signaling.

Table 30 Data VLAN Conﬁguration in Device Settings menu

Data VLAN : No

No 4095

Conﬁgure PC-Port Untag-All, which conﬁgures PC-Port VLAN Tag Stripping. Check the check box by pressing the Select key in the Navigation key cluster to enable this item.

If the check box is not checked, tag stripping is disabled and the packet is sent to the PC port unmodiﬁed. If the check box is checked, tag stripping is enabled and the 802.1q header is removed (assuming one exists) from the packet before it is forwarded through the PC port.

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If Data VLAN is enabled in step 15, PC-Port Untag All is enabled and the check box is checked by default. In this case, the outgoing tag is stripped. Override the default by pressing the Select key to remove the checkmark.

If Data VLAN is disabled, PC-Port Untag All is disabled and the check box is not checked by default. In this case, the ingress tag is not stripped. The default can be overridden by pressing the Select key, which places a checkmark in the box.

Table 31 PC-Port Untag-All Conﬁguration in Device Settings menu

PC-Port Untag all :

Conﬁgure the device settings 93

Conﬁgure the Ethernet port mode. The Ethernet port mode is conﬁgured by the items in the Device

Settings menu Duplex and Ignore GARP.

• Duplex: Duplex enables or disables Ethernet speed and mode negotiation. Select Auto unless the Ethernet port speed must be set to 10 MB/s or 100 MB/s. Select 10BT Full for 10 MB/s and 100BT Full for 100 MB/s.

•

Ignore GARP: GARP requests handling. The default can be overridden by pressing the Select key, which places a check mark in the box.

Table 32 Ethernet port mode conﬁguration

Duplex : Auto

Auto 10BT Full 100BT Full

Ignore GARP:

Enter Provisioning Parameters: Server URL, Protocol, Device ID, and Password.

SIP Firmware Release 1.1 for IP Phone 1120E Administration

The next four tables show menu items in the Device Settings menu that apply to conﬁguring the Provisioning parameters.

•

Server URL: Enter the URL of the provisioning server in the Server URL box.

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94 Device Settings on the IP Phone 1120E with SIP Firmware

Table 33 Server URL conﬁguration

Provisioning

Server URL:

Table 34 Protocol conﬁguration

Protocol:

TFTP TFTP FTP HTTP

0.0.0.0

•

Protocol: Select either TFTP, FTP, or HTTP used for provisioning in the Protocol box.

•

Device ID: Enter the Device ID used by the provisioning server for authentication of the IP Phone 1140E.

Note 1: TFTP does not require a Device ID. If FTP or HTTP are used, enter the information for a Device ID

Note 2: Enter your User ID as the Device ID.

Table 35 Device ID conﬁguration

Device ID : XXXXXXXXX

•

Password: Enter the Password used by the provisioning server for authentication of the IP Phone 1140E. Note: TFTP does not require a Password. FTP and HTTP do require a password.

Table 36 Password conﬁguration

Password : XXXXXXXXX

Conﬁgure the PVQMon IP address.

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Table 37 PVQMon IP conﬁguration

Conﬁgure the device settings 95

If this server is available on the network, enter the PVQMon IP address.

0. 0. 0. 0

Conﬁgure the NAT Traversal Method. Use one of the following

PVQMon IP :

methods to traverse Network Address Translation (NAT) devices for the IP Phone 1140E with SIP Firmware.

The next ﬁve tables show menu items in the Device Settings menu that apply to conﬁguring the NAT Traversal Method.

• NAT Signal

Select either None, SIP_PING, or STUN to deﬁne the NAT traversal mode for SIP signaling.

Table 38 NAT Signal conﬁguration

NAT Traversal

NAT Signal : STUN

None SIP Ping STUN

Table 39 NAT media conﬁguration

NAT Media :

STUN None STUN

SIP Firmware Release 1.1 for IP Phone 1120E Administration

• NAT Media

Select either None or STUN to deﬁne NAT traversal mode for signaling.

•

NAT TTL:

Enter the Time to Live (TTL) in seconds for the NAT traversal.

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96 Device Settings on the IP Phone 1120E with SIP Firmware

Table 40 NAT TTL conﬁguration

NAT TTL :

000

Table 41 STUN S1 IP conﬁguration

STUN S1 IP :

00. 00. 00. 00

Table 42 STUN S2 IP conﬁguration

STUN S2 IP :

00. 00. 00. 00

•

STUN S1 IP: Enter an IP address for the STUN S1 IP device.

•

STUN S2 IP:

Enter an IP address for the STUN S2 IP device.

—End—

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Multiple Appearance Directory Number (Single Call Arrangement)

Note: Only users of the Communication Server 2000 (CS 2000) and Communication Server 2100 (CS 2100) have access to the Multiple Appearance Directory Number (MADN) Single Call Arrangement (SCA) feature at this time.

The Multiple Appearance Directory Number (MADN) allows a Directory Number (DN) to appear on more than one IP Phone 1120E with SIP Firmware. The MADN with Single Call Arrangement (SCA) feature allows multiple telephones to appear as a single line to a caller. Any one of the IP Phone 1120E telephones in a group with MADN can initiate or answer a call, but only one call can be active at any given time. Any other user in the group can join the active call by picking up the handset of the IP Phone 1120E with SIP Firmware.

With the MADN SCA feature conﬁgured on multiple telephones of different registered SIP users, the telephones share one single DN. An incoming call to this DN causes all the telephones in the group to ring.

Any user of an IP Phone 1120E with the MADN SCA feature can put a call on hold or can prevent others from joining in the active call.

If a user’s group is active (as seen by line icon being off-hook) and the user picks up the handset, the user is automatically joined to an ongoing MADN call (unless the server restricts this feature for privacy or other factors).

Vertical services

Vertical services are CS 2000 and CS 2100 features that can be activated or deactivated by dialing a deﬁned code, for example, Privacy. Even though no more than one active session can be established for the MADN SCA group, members of the group can still enter certain vertical services.

Currently, the available vertical service is Privacy.

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98 Multiple Appearance Directory Number (Single Call Arrangement)

Privacy

A user can activate the privacy service by putting the current session on hold and dialing the privacy code. The CS 2000 connects the IP Phone 1120E to the Media Application Server (MAS) to hear a conﬁrmation for its request and terminates the session. The user takes the original session off hold.

Privacy access codes

The privacy access codes are: PRV, PRLA, PRLC. For example: PRV = 191 PRLA = 192 PRLC = 193 If the initial state of the MADN group is nonprivate , the PRV access code is used to toggle between privacy on and privacy off. If the initial state of the MADN group is private, the PRLA access code allows bridging and PRLC closes it.

Feature dependencies and restrictions

The minimum release to support MADN SCA feature is 1.1. Multi-User login is not supported.

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Emergency 911 - Operator control of disconnect

Overview

For Emergency 911 services to be effective, it is vital that a system maintains a connection with the 911 operator until the user provides all necessary information. In traditional 911 telephony, if a 911 call is disconnected before the necessary information is provided, mechanisms are in place to allow the 911 operator to ring back the calling party. In VoIP calls, 911 support is an evolving standard.

If the user of an IP Phone 1120E attempts to disconnect a 911 call after the 911 operator has answered, the IP Phone 1120E switches to hands-free mode. Once the IP Phone 1120E is in hands-free mode, the 911 call can only be disconnected by the operator.

Emergency 911 calls originate on the IP Phone 1120E and are completed by the Call Server. The Call Server communicates with the 911 network or 911 systems for routing, call control, and location information. Although the IP Phone 1120E allows users to enter location information, this location information is not used by all Call Servers. Some Call Servers derive the location information based on the number and location databases. Characteristics of Emergency 911 calls and limitations of Emergency 911 calls using the IP Phone 1120E are given below.

Location information

EffectiveEmergency 911 services also rely on accurate location information. The IP Phone 1120E supports the inclusion of the x-nt-location header in SIP messages that provide location information to a Nortel call server. The location information is selected by the user during registration, and must be correctly provisioned at the call server level (see appropriate NTP speciﬁc to your call server).

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100 Emergency 911 - Operator control of disconnect

Dialing plan conﬁguration

To allow operator control of disconnect during a 911 call, the IP Phone 1120E must identify a 911 call as soon as a 911 call is initiated. The IP Phone 1120E uses an emergency ﬂag in the dialing plan to identify a 911 call. When the dialing plan detects that an emergency number is dialed, it automatically switches to operator controlled disconnect mode when the call is answered. The dialing plan can have multiple emergency numbers.

Examples of an emergency ﬂag in the dialing plan

(911)|(911)# &&sip:$$@$n;user=phone&&emergency (4357)|(4357)# &sip:$$@$n;user=phone&&emergency

Characteristics of E911 calls

During an E911 call, the user:

•

cannot make outgoing calls

•

is not notiﬁed of incoming calls and cannot accept incoming calls. Incoming calls receive a busy signal

•

cannot transfer, join, or conference the E911 call, place the E911 call on hold, or park the E911 call

•

cannot receive an incoming call

•

cannot auto-retrieve a parked call and auto-retrieval of parked calls is not displayed

•

cannot disconnect the E911 call. Only an E911 center or operator can disconnect the emergency call

•

cannot change Audio Quality

•

can reply to IM popups, which are operational during an E911 call

During an E911 call the keys function as follows:

•

the Services, Inbox, Outbox, and Address-Book keys are all disabled

•

a right click of the mouse does not show the services menu

•

the conspicuous keys are disabled

•

the mute key and hold key are disabled

•

the increase and decrease volume keys remain functional.

•

the feature keys are visible and all except the speed-dial keys are functional.

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Avaya SIP Firmware for IP Phone 1120E User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual

New in this release

Other changes

Revision history

How to Get Help

Getting Help from the Nortel Web site

Getting Help over the telephone from a Nortel Solutions Center

Getting Help from a specialist using an Express Routing Code

Getting Help through a Nortel distributor or reseller

Introduction to this guide

Subject

Intended audience

Acronyms

Table 1 Acronyms used

Related publications

Overview

Introduction

SIP overview

IP Phone 1120E with SIP Firmware

Figure 1 IP Phone 1120E with SIP Firmware

Related documentation

Installation overview

Procedure 1 Installation overview

Figure 2 Installation of IP Phone 1120E with SIP Firmware, page 1 of 2

Figure 3 Installation of IP Phone 1120E with SIP Firmware, page 2 of 2

Supported SIP proxy servers

Before installation

Introduction

Preinstallation

Procedure 2 Preinstallation checklist

How provisioning works

Download the SIP Firmware to the provisioning server

Procedure 3 Downloading SIP Firmware for the IP Phone 1120E from the Nortel Web site

Figure 6 Sample dialing plan

Dialing function description

Figure 7 Sample dialing plan declarations section

Figure 9 Sample dialing plan digit map section

Figure 10 DRegex rules

Partial DHCP mode

Full DHCP mode

DHCP VLAN Auto Discovery

Physically install the IP Phone 1120E

Procedure 4 Installing the IP Phone 1120E

Figure 11 IP Phone 1120E connections

Introduction

Upgrade the SIP Firmware on the IP Phone 1120E

Download the SIP Firmware to the provisioning server

Procedure 5 Downloading SIP Firmware for the IP Phone 1120E from the Nortel Web site

Upgrade to the Minimum UNIStim Firmware

Identify the current version of UNIStim Firmware

Procedure 7 Checking the UNIStim Firmware version on a new IP Phone 1120E

Procedure 8 Checking the UNIStim Firmware version on an IP Phone 1120E already in use

Upgrade UNIStim Firmware to the minimum required UNIStim Firmware

Procedure 10 Downloading UNIStim Firmware automatically through TFTP on bootup

Convert UNIStim Firmware to SIP Firmware on the IP Phone 1120E

Procedure 11 Downloading SIP Firmware for the IP Phone 1120E from the Nortel Web site

Procedure 12 Converting UNIStim Firmware to SIP Firmware using TFTP

Voice Quality Monitoring

Feature overview

VQMon set-up

Server set-up

How VQMon works

Device Settings on the IP Phone 1120E with SIP Firmware

Introduction

802.1x (EAP) Port-based network access control

Authorization

Device ID

Password

DHCP

NO DHCP mode

Gateway

DNS IP

SET IP

Net Mask

Voice VLAN

VLAN Filter

Disable PC Port