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

SIP Firmware Release 2.0 for IP Phone 1140E Administration

NN43113-300

Document status: Standard Document version: 03.09 Document date: 15 September 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

Server Supplied User Credentials 7 Expansion module for the IP Phone 1100 Series when connected to IP Phone 1100

IEEE 802.1ab Link Layer Discovery support 7 IP Phone 1140E with SIP ﬁrmware PRACK and UPDATE support feature 8 RFC 3581 8 RTP port conﬁguration 9 Voice Quality Monitoring 9 Distinctive ringing and Call waiting 9 Shared Call Appearance 10 Transfer to Voice Mail 10 Other changes 10

How to Get Help 13

Getting Help from the Nortel Web site 13 Getting Help over the telephone from a Nortel Solutions Center 13 Getting Help from a specialist using an Express Routing Code 14 Getting Help through a Nortel distributor or reseller 14

series SIP Client 7

Revision history 10

Introduction to this guide 15

Subject 15 Intended audience 15 Acronyms 15 Related publications 17

Overview 19

Introduction 19

SIP overview 19 IP Phone 1140E with SIP Firmware 19 Related documentation 20 Installation overview 21

Supported SIP proxy servers 24

Before installation 25

Introduction 25

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4 Contents

Preinstallation 25

Conﬁgure the provisioning server 29

How provisioning works 29 Download the SIP Firmware to the provisioning server 30 Create the SIP provisioning ﬁle on the provisioning server 30

Setting the default language on the IP Phone 1140E to French 35

Create the device conﬁguration ﬁle on the provisioning server 36

Device conﬁguration commands 40

Create the set conﬁguration ﬁle on the provisioning server 59

Set conﬁguration commands 60

Create the Dialing Plan ﬁle on the provisioning server 61

Dialing function description 63

Downloadable WAV ﬁles 67

Conﬁgure the DHCP server 69

Partial DHCP mode 69 Full DHCP mode 70

Conﬁgure the DHCP server to support Full DHCP mode 71 DHCP VLAN Auto Discovery 73

Physically install the IP Phone 1140E 75 Upgrade and convert the IP Phone 1140E ﬁrmware 79

Introduction 79 Upgrade the SIP Firmware on the IP Phone 1140E 79

Download the SIP Firmware to the provisioning server 79 Modify the SIP provisioning ﬁle 80

Upgrade to the minimum UNIStim Firmware 81

Identify the current version of UNIStim Firmware 81 Upgrade UNIStim Firmware to the minimum required UNIStim Firmware 84

Convert UNIStim Firmware to SIP Firmware on the IP Phone 1140E 87

Voice Quality Monitoring 91

Feature overview 91 VQMon set-up 91 Server set-up 92 How VQMon works 92

End of call report 93 Session interval report 93 Alert interval report 93

Device Settings on the IP Phone 1140E with SIP Firmware 95

Introduction 95

802.1x (EAP) Port-based network access control 98

Conﬁgure the device settings 111

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Contents 5

Multiple Appearance Directory Number (Single Call

Arrangement) 123

Vertical services 123

Privacy 124

Feature dependencies and restrictions 124

Emergency 911 - Operator control of disconnect 125

Overview 125 Location information 125 Dialing plan conﬁguration 126 Characteristics of E911 calls 126

IP Phone 1140E restrictions 127

Service package restrictions 127 Distinctive Ringing feature 127

NAT ﬁrewall traversal 129 Three-port switch and VLAN functionality 131

System overview 131

SIP messages supported by the IP Phone 1140E 135

SIP methods 135 SIP responses 136

1xx Response--Information Responses 136 2xx Response--Successful responses 137 3xx Response--Redirection responses 137 4xx Response--Request failure responses 138 5xx Response--Server failure responses 140

6xx Response--Global responses 141 Default error handling 141 SIP header ﬁelds 141 Session description protocol usage 144 SDP and Call Hold 145 Transport layer protocols 145 SIP security authentication 145 SIP DTMF Digit transport 146 Supported subscriptions 146 Supported instant messaging 146

Maintenance and troubleshooting 149

Convert SIP Firmware to UNIStim Firmware 149 IP Phone 1140E diagnostics 150

Index 153

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6 Contents

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New in this release

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

Server Supplied User Credentials

This release adds support for downloading a users login credentials from a central repository. The login credentials are associated with the MAC address of the set. This feature simpliﬁes installation of the IP Phone by allowing it to automatically log into the SIP proxy server without end user intervention.

To provision this feature see "Set conﬁguration commands" (page 60).

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

An expansion module providing 18 additional feature keys, can be attached to a Nortel IP Phone 1140E with SIP ﬁrmware. Up to three expansion modules can be added to the phone. The feature keys in the expansion module provide the same functionality as the feature keys on the IP Phone.

Additionally, this feature adds the ability to automatically populate the extra feature keys, using the friends list or the address book as sources.

For more information on the Expansion module refer to, SIP Firmware for IP Phone 1100 Series Expansion Module User Guide (NN43110-301).

To provision this feature see "Feature conﬁguration commands " (page 42).

IEEE 802.1ab Link Layer Discovery support

Discovery protocols provide a mechanism to identify devices attached to a network. Popular network-management systems use automated discovery to obtain the topology of a network. These applications detect adds and removals, provide Layer 3 information, and group the attached devices into IP subnets.

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8 New in this release

IEEE has developed 802.1ab Link Layer Discovery Protocol (LLDP), a standard for discovering the physical topology between neighboring devices.

802.1ab LLDP deﬁnes a standard method for Ethernet network devices such as switches, routers and IP Phones to advertise information about themselves to other nodes on the network and store the information they discover in a MIB.

This feature implements 802.1ab and its IEEE 802.1/802.3 and TIA-MED organizationally speciﬁc extensions.

To provision this feature see "802.1ab Link Layer Discovery Protocol"

(page 99) and Procedure 13 "Provisioning the Device Settings parameters" (page 111). The parameters in Table 18 "TLV formats" (page 100) can be

conﬁgured on the IP Phone.

IP Phone 1140E with SIP ﬁrmware PRACK and UPDATE support feature

The combined support of PRACK and UPDATE allows the IP Phone 1140E with SIP ﬁrmware to provide reliability to provisional responses, and the ability to update session parameters during call setup as well as after the initial invite has received a ﬁnal response. The combination of reliable provisional responses (PRACK) and the ability to change session parameters before call establishment (using UPDATE) will improve the IP Phone 1140E interactions with some PSTN networks where the parameters of a session may need updating before the call is established.

PRACK is used for adding reliability to provisional responses (such as 180 Ringing) in SIP messaging. The PRACK request plays the same role as ACK, but for provisional responses. [RFC 3262]

UPDATE allows a client to update parameters of a session (such as the set of media streams and their codecs) but has no impact on the state of a dialog. In that sense, it is like a re-INVITE, but unlike re-INVITE, it can be sent before the initial INVITE has been completed. This makes it very useful for updating session parameters within early dialogs. [RFC 3311]

To provision this feature see "Feature conﬁguration commands " (page 42).

RFC 3581

This feature implements an extension to SIP for Symmetric Response Routing for the IP Phone 1140E with SIP ﬁrmware. This extension permits the conduction of SIP dialogs through a Symmetric Network Address Translator (NAT) using UDP.

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This allows the phone to work from behind and/or in front of a symmetrical NAT with servers and/or clients that support RFC 3581. In particular, it enhances the capability of the IP Phone 1140E with SIP ﬁrmware to interoperate with other proxies and to work in any network conﬁguration. For this feature to work properly, the receiving end device must support RFC 3581.

To provision this feature see "Feature conﬁguration commands " (page 42).

RTP port conﬁguration

This RTP port conﬁguration feature provides options to set or change RTP ports in the Nortel IP Phone 1140E with SIP ﬁrmware. RTP port conﬁguration is available only through the provisioning server.

To provision this feature see "Feature conﬁguration commands " (page 42).

Voice Quality Monitoring

When Voice Quality Monitoring (VQMon) is enabled, the IP Phone 1140E with SIP ﬁrmware gathers statistics regarding the quality metrics of the current call and sends reports to the server at regular intervals. The call-related statistics contain condensed information about the SIP Session Description Protocol (SDP), the Call ID, the local and remote address, voice quality-related statistics, Zulu times for start-time and the time the report was sent. The voice quality-related statistics include jitter, packet loss, delay, burst gap loss, listening R-factor, R-LQ, R-CQ, MOS-LQ and MOS-CQ. The reports can be used for QoS monitoring. A third party server is required to collect and organize the data.

Distinctive ringing and Call waiting 9

End of call reports are sent with the Zulu time indicating start of session and end of session; session interval reports are sent with interval deﬁned in the conﬁguration using time stamps, and alert reports are sent with current time stamps with a limited frequency of 1 report per 5 seconds if any alert violation persists.

Release 2.0 supports interworking with SQLMediator 1.1. To provision this feature see "VQMon conﬁguration commands" (page 57).

Distinctive ringing and Call waiting

The Distinctive ringing feature permits users to distinguish between different types of call actions by playing a different ringing pattern. The request for a speciﬁc ringing pattern comes from the server at the time the call is being established. The Nortel IP Phone 1140E with SIP ﬁrmware does not request the playing of distinctive ringing from other parties. The ringing patterns to be used follow the North American standards which also include call waiting

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10 New in this release

tones for the times when the receiving end is already engaged in a call session. The predeﬁned ringing pattern identiﬁers available on the phone accommodate usage by the BroadWorks servers.

To provision this feature see "Feature conﬁguration commands " (page 42).

Shared Call Appearance

The Shared Call Appearance feature allows a given line to be conﬁgured with multiple locations, essentially allowing multiple endpoints to login to the system while using the same external number. Any one of these locations can be used to originate or receive calls. Any call appears to the other party to be originating from or terminating to the same number, regardless of the location initiating or receiving the call.

The Shared Call Appearance feature allows a user to pick up a call that was put on hold by another user of the same group and it allows a user to join an active call of another user in the group.

The Shared Call Appearance feature is supported on BroadWorks servers. To provision this feature see "Feature conﬁguration commands " (page 42).

Transfer to Voice Mail

The transfer to Voice Mail feature enables a soft key to transfer an incoming call, or a call waiting, to a voice mail box on the IP Phone 1140E with SIP ﬁrmware.

The transfer to Voice Mail feature is supported on MCS 5100 servers only. To provision this feature see "Feature conﬁguration commands " (page 42).

Other changes

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

history" (page 10).

Revision history

September 2008

Standard 03.09. This document is up-issued to reﬂect changes in technical content.

July 2008

Standard 03.08. This document is up-issued to reﬂect changes in technical content.

June 2008

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Other changes 11

Standard 03.07. This document is up-issued to support SIP Firmware for IP Phone 1140E Release 2.0.

February 2008

Standard 02.04. This document is up-issued to include editing changes.

November 2007

Standard 02.03. This document is up-issued to reﬂect changes in the technical information in response to change requests. A note was added stating that NAT_SIGNALLING is required for networks that use STUN or SIP_PING for NAT traversal. The commands FORCE_OCT_END and OCT_END were changed to FORCE_OCT_ENDDIAL and DISABLE_OCT_ENDDIAL respectively.

October 2007

Standard 02.02. This document is up-issued to ensure that the value of

802.1P_MEDIA parameter in the QoS and ToS commands must be set to

-1 to be able to have a voice path.

July 2007

Standard 02.01. This document is up-issued to support SIP Firmware for IP Phone 1140E Maintenance Release 1.1.

March 2007

Standard 01.02. This document is up-issued to incorporate an addition to the Maintenance and Troubleshooting section.

January 2007

Standard 01.01. This is a new document created to support the IP Phone 1140E with SIP Firmware.

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12 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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14 How to Get Help

Getting Help from a specialist using an Express Routing Code

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

ww.nortel.com/erc

to w

Getting Help through a Nortel distributor or reseller

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

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Introduction to this guide

Subject

SIP Firmware Release 2.0 for IP Phone 1140E Administration

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

Intended audience

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

Acronyms

This guide uses the following acronyms:

Table 1 Acronyms used

AAA Authentication, Authorization, and Accounting ALG Application Layer Gateway DHCP Dynamic Host Configuration Protocol DNS Domain Name System DRegex Digit Regular Expression DSCP Differentiated Services Code Point EAP Extensible Authentication Protocol ERE Extended Regular Expressions FQDN Fully Qualified Domain Name FTP File Transfer Protocol GARP Gratuitous Address Resolution Protocol GUI Graphical User Interface HTTP Hyper Text Transfer Protocol

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16 Introduction to this guide

IETF Internet Engineering Task Force ISDN Integrated Services Digital Network IM Instant Message IP Internet Protocol IPCM Internet Protocol Client Manager ITU-T Telecommunications Standardization sector of the International

Telecommunications Union LAN Local Area Network MAC Media Access Control MADN Multiple Appearance Directory Number MAS Media Application Server NAT Network Address Translator NetConfig Configuration screens available after an IP Phone resets NDU Network Diagnostic Utility PDT Problem Determination Tool PEC Product Engineering Code POE Power Over Ethernet POSIX Portable Operating System Interface PRACK Provisional Acknowledgement PSTN Public Switched Telephone Network PVQMon Proactive Voice Quality Monitoring QoS Quality of Service RADIUS Remote Authentication Dial In User Service RTCP Real-time Control Protocol RTCP XR RTP Control Protocol Extended Reports RTP Real-time Transfer Protocol SCA Single Call Arrangement

Shared Call Appearance SDP Session Description Protocol SIMPLE SIP for Instant Messaging and Presence Leveraging Extensions SIP Session Initiation Protocol SMTP Simple Mail Transfer Protocol STUN Simple Traversal of UDP through NAT devices TFTP Trivial File Transport Protocol TPS Terminal Proxy Server

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TTL Time-to-live UDP User Datagram Protocol UFTP UNIStim File Transfer Protocol UNIStim Unified Network IP Stimulus Protocol VoIP Voice over IP VLAN ID Virtual Local Area Network Identification VLAN IP Virtual Local Area Network Internet Protocol VQMon Voice Quality Monitoring

Related publications

Other publications related to the SIP Firmware Release 2.0 for IP Phone 1140E Administration are:

•

SIP Firmware Release 2.0 for IP Phone 1140E Quick Reference Card

(NN43113-102)

•

SIP Firmware Release 2.0 for IP Phone 1140E User Guide

(NN43113-101)

Related publications 17

•

Broadsoft Partner Conﬁguration Guide Nortel SIP Firmware for 1120E / 1140E IP Phones (NN43121-300)

•

Expansion Module for IP Phones 1140E and 1120E (SIP Firmware) User Guide (NN43110-301)

•

Nortel IP Phone 1100 Series product bulletins on w

ww.nortel.com/sup-

port.

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18 Introduction to this guide

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Overview

Introduction

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

SIP overview

Session Initiation Protocol (SIP) is a signaling protocol used for establishing multimedia sessions in an Internet Protocol (IP) network.

SIP is a text-based protocol similar to HTTP and SMTP. With the introduction of SIP to IP Phones, telephony integrates easily with other Internet services. SIP allows the convergence of voice and multimedia.

IP Phone 1140E with SIP Firmware

The IP Phone 1140E connects to an IP network using an Ethernet connection. All voice and signaling information is converted into IP packets and sent across the network.

The IP Phone 1140E can be ordered with UNIStim Firmware installed or with SIP Firmware installed. UNIStim Firmware and SIP Firmware use the same hardware, but the model number of an IP Phone 1140E with UNIStim Firmware is different from the model number of an IP Phone 1140E with SIP Firmware.

If you have an IP Phone 1140E with UNIStim Firmware, you can convert the ﬁrmware to SIP Firmware. To download the most recent version of SIP Firmware, see "Download the SIP Firmware to the provisioning server"

(page 30).

This guide explains how to:

•

conﬁgure the provisioning server and the DHCP server. Note: The provisioning server is where the ﬁrmware and the conﬁguration ﬁles for the IP Phone 1140E reside. This is not the IP Client Manager (IPCM) of the call server.

•

convert an IP Phone 1140E with UNIStim Firmware to an IP Phone 1140E with SIP Firmware

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20 Overview

•

provision the Device Settings parameters on the IP Phone 1140E with SIP Firmware

Note: Converting the ﬁrmware on an IP Phone 1140E from UNIStim Firmware to SIP Firmware overwrites the UNIStim Firmware. The IP Phone 1140E cannot operate in both modes simultaneously. A switch from UNIStim to SIP Firmware or SIP to UNIStim Firmware requires a ﬁrmware reload.

The following ﬁgure shows the main components of the IP Phone 1140E with SIP Firmware.

Figure 1 IP Phone 1140E with SIP Firmware

Installation overview

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

Introduction 21

Procedure 1 Installation overview

Step Action 1

Conﬁgure the provisioning server and, optionally, the DHCP server. The function of the provisioning server is to provide conﬁguration options to every IP Phone 1140E throughout the network. TheDHCP server can be conﬁgured to provide basic network-conﬁguration data or a more comprehensive set of network-conﬁguration data for the IP Phone 1140E with SIP Firmware.

2 3

Load SIP Firmware on the IP Phone 1140E. Conﬁgure the initial network-conﬁguration parameters on the IP

Phone 1140E with SIP Firmware.

—End—

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22 Overview

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

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

Introduction 23

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24 Overview

Supported SIP proxy servers

Call server Call server release Minimum SIP Firmware on

the IP Phone 1140E

MCS 5100

MCS 5100 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 1140E.

Preinstallation

Complete the following checklist.

Procedure 2 Preinstallation checklist

Step Action 1 Read and become familiar with the SIP Firmware Release 2.0

for IP Phone 1140E User Guide (NN43113-101) and the SIP Firmware Release 2.0 for IP Phone 1140E Quick Reference Card

(NN43113-102).

Ensure there is one IP Phone 1140E boxed package for each IP Phone 1140E being installed.

Table 2 IP Phone 1140E box contents

Item

IP Phone 1140E Graphite with icon keys without PS (SIP) (RoHS)

IP Phone 1140E Graphite with English keys without PS (SIP) (RoHS)

Accessories

Handset, Charcoal NTYS09AA70 Handset cord, Charcoal NTYS10AA70

Ensure that the IP Phone 1140E box includes:

Order number

NTYS05CAE6 N0142353

NTYS05DAE6 N0142354

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26 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 1140E can be powered either by Power Over Ethernet (POE) or powered through an external power supply. Order the external power supply separately.

WARNING

Do not use the AC power adapter, if you are connected to a Power over the Ethernet (PoE) connection. Only use the AC power adapter when you do not have a Power over the Ethernet connection.

Table 3 1100 series phones separably orderable parts

CPC code PEC code

N0146475 NTYS17BAE6 IP Phone Global Power Supply (2000, 1100, 1200) (RoHS) N0089603 NTYS14AAE6 Standard IEC Cable - North America (RoHS) A0781922 NTTK15AA Standard IEC Cable – Australia / NZ (Note: RoHS not

N0114986 NTTK16ABE6 Standard IEC Cable – Europe N0109787 NTTK17ABE6 Standard IEC Cable – Switzerland N0109881 NTTK18ABE6 Standard IEC Cable – UK N010978 NTTK22ABE6 Standard IEC Cable – Denmark A0814961 A0814961 Standard IEC Cable - Argentina (Note: RoHS not required) N0118951 NTTK26AAE6 Standard IEC Cable - Japan

Product description

required)

CAUTION

TheIP Phone 1140E must be plugged into a10/100-BaseT Ethernet jack. Severe damage occurs if this IP Phone 1140E is plugged into an ISDN connection.

4 Ensure that the location meets the network requirements:

SIP Firmware Release 2.0 for IP Phone 1140E Administration

•

a DNS server and a DHCP server with DHCP relay agents installed, conﬁgured, and running. Using DHCP and DNS servers with CS 2000 network is recommended but not mandatory.

•

An Ethernet connection to a network with an appropriate SIP proxy server.

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Preinstallation 27

•

One of the following ﬁle servers used as a Provisioning server: — TFTP server — FTP server — HTTP server

Only a TFTP server can be used for an initial UNIStim-to-SIP Firmware conversion. An IP Phone 1140E with SIP Firmware can operate with a TFTP, FTP, or HTTP ﬁle server.

—End—

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28 Before installation

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Conﬁgure the provisioning server

ATTENTION

If you have UNIStim Firmware on your IP Phone 1140E, the ﬁrmware must be converted from UNIStim to SIP before you proceed with the following instructions. See the chapter "Upgrade and convert the IP Phone 1140E ﬁrmware" (page

79) for instructions on how to convert the ﬁrmware on an IP Phone 1140E from

UNIStim to SIP.

If the IP Phone 1140E is installed with SIP Firmware, further SIP Firmware upgrades can be done with a TFTP, an FTP, or an HTTP server.

How provisioning works

Provisioning is performed without interaction with the call server. The IP Phone 1140E with SIP Firmware connects directly with the provisioning server in order to retrieve ﬁrmware ﬁles and conﬁguration ﬁles. In this case, the provisioning server is not to be confused with the IP Client manager on the call server. The methods of provisioning are:

•

Automatic provisioning at power-up: After the IP Phone 1140E powers up or is reset, it checks the provisioning server for the latest ﬁles.

•

Provisioning through user interaction: The end user can manually check for updates by pressing the Services Context-sensitive soft key and selecting SystemSystem . In the menu select Check for Updates.

•

Automatic provisioning at a preconﬁgured time: The IP Phone 1140E with SIP Firmware checks for updates every 24 hours, at a time speciﬁed by a parameter in the device conﬁguration ﬁle.

The following steps are taken when provisioning updates occur. The IP Phone 1140E with SIP Firmware:

1. connects to the provisioning server

2. retrievesthe provisioning ﬁle (1140eSIP.cfg) from the provisioning server

3. reads and acts upon the content of the provisioning ﬁle and decides whether any other ﬁle is needed, based on a set of rules. If ﬁles need to be downloaded to the IP Phone 1140E, a new ﬁle transfer session starts

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30 Conﬁgure the provisioning server

for each ﬁle to be downloaded. The provisioning ﬁle (1140eSIP.cfg) can contain commands that prompt for conﬁrmation before a ﬁle is downloaded.

Download the SIP Firmware to the provisioning server

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

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

Step Action 1

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

ww.nortel.com/support.

Password. The Technical Support page appears.

3 4 5

Enter IP Phone 1140E in the Knowledge and Solution Engine box. Select Software in the All types scroll down menu. Press the gray arrow at the end of the Knowledge and Solution

Engine box to obtain the Search Results.

From the Search Results, select the appropriate version of the SIP Firmware for the IP Phone 1140E, for example, SIP IP Phone 1140E

Release 0625C39D26.bin.

Place the selected ﬁrmware on the provisioning server.

—End—

Create the SIP provisioning ﬁle on the provisioning server

The provisioning ﬁle is downloaded from the provisioning server to the IP Phone 1140E every time the IP Phone 1140E checks for updates. The provisioning ﬁle is a clear text ﬁle that has the naming convention 1140eSIP.cfg. The following is an example of a provisioning ﬁle:

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Figure 4 Sample provisioning ﬁle

Create the SIP provisioning ﬁle on the provisioning server 31

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32 Conﬁgure the provisioning server

Table 4 Provisioning ﬁle supported sections

[DEVICE_CONFIG] Device configuration file [LANGUAGE] Downloadable language files (more than one can be specified in

each section) [FW] Firmware image [DIALING_PLAN] Dialing plan [TONES] Downloadable tones (.wav files) [USER_CONFIG] Set specific configuration file

Provisioning is performed using the commands in the 1140eSIP.cfg conﬁguration ﬁle. The conﬁguration ﬁle can have multiple sections.

Note: The maximum length of a line item in the conﬁguration ﬁle is 80 characters. If a line item with more than 80 characters is encountered when parsing the conﬁguration ﬁle, the remaining portion of the ﬁle following that line item will be ignored.

’#’ is used to indicate a comment. Anything preceeding a ’#’ is a comment.

Each section in the conﬁguration ﬁle deﬁnes rules for different ﬁle types. A section starts with a [SECTION NAME] to specify rules for each ﬁle type. For example: [FW].

A section is a mandatory ﬁeld. Parsing of download rules for each ﬁle type starts with ﬁnding this key word. Currently, the following sections are supported by the IP Phone 1140E with SIP Firmware:

•

[FW] section for the ﬁrmware load download

•

[DEVICE_CONFIG] section for the device conﬁguration

•

[DIALING_PLAN] section for the dialing plan ﬁles

•

[LANGUAGE] section for the language prompts ﬁles

•

[TONES] section for the downloadable tone ﬁles

•

[USER_CONFIG] section for set speciﬁc conﬁguration ﬁle

Firmware [FW] image ﬁles originate from Nortel only and are authenticated during ﬁrmware download. If the FW authentication fails, the IP Phone 1140E displays an error message and continues operation with the existing FW image.

Device conﬁguration ﬁles are used to set various parameters in the IP Phone 1140E.

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Dialing plan ﬁles are used for conﬁguring dialing patterns and the format of originated URIs in the SIP message.

Note: Some call servers do not yet support SIP URI dialing.

Language ﬁles are simple text ﬁles containing all text prompts used by the IP Phone 1140E. Language ﬁles are used for the localization of the IP Phone 1140E without ﬁrmware upgrade. Each language ﬁle has a header that contains a ﬁrmware load version with which this ﬁle is associated. Language ﬁles are signed by Nortel and are authenticated by the ﬁrmware for security reasons.

Tones ﬁles are standard in the Telecommunications Standardization sector of the International Telecommunications Union (ITU-T). The set supports custom tone ﬁles. The tone ﬁles must be WAV ﬁles with the following speciﬁcation: A-law or u-law (8.0 kHz, 8-bit, mono or 16.0 kHz, 16 bit mono). The WAV ﬁles can be created and downloaded to the IP Phone 1140E. These ﬁles are not authenticated by the IP Phone 1140E.

Set speciﬁc conﬁguration ﬁles support customizing the set on a per set/user level. Parameters in the device conﬁguration ﬁle can be overwritten with a set speciﬁc conﬁguration ﬁle.

Mandatory keywords in the Provisioning ﬁle are:

•

VERSION [xxxxxx], where xxxxxx is a six- to ten-digit number

representing the version of the ﬁle on the server. The version of the module is speciﬁed in this ﬁeld. The command is used for version comparison in AUTO mode. VERSION is mandatory for all sections. In the FW section, the ﬁrmware version of the load located on the provisioning server must be entered in this ﬁeld. For all other sections, VERSION is just a counter that can be incremented if it is necessary to download a new ﬁle version.

CAUTION

The version number is stored permanently on the IP Phone 1140E until a higher version number is downloaded or until the version number is deleted using the Srvcs, System, Erase User Data menu selection on the IP Phone 1140E. This prevents a new ﬁle from being downloaded if a lower version number is used in the Provisioning File.

•

DOWNLOAD_MODE [AUTO | FORCED] deﬁnes whether the version

is checked. This command is optional. If this command is not present, AUTO mode is used as the default.

— AUTO - This mode compares the version of the module from the

SIP Firmware Release 2.0 for IP Phone 1140E Administration

VERSION ﬁeld and the version of the module version stored in the FLASH memory of the IP Phone 1140E. The ﬁle download is initiated

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34 Conﬁgure the provisioning server

only if the version speciﬁed is higher than the current version stored in the IP Phone 1140E. If the version is not applicable, as in the case of language ﬁles, the date of the ﬁle must be used for the decision.

— FORCED - This mode forces the ﬁrmware download process.

FORCED can be used for ﬁrmware downgrade procedures. Note: In FORCED or AUTO DOWNLOAD_MODE, the version

number is overwritten with each ﬁrmware download.

•

FILENAME [ﬁlename] speciﬁes the ﬁle name to be downloaded for

this section. For the language and tone section, the use of multiple ﬁlenames is allowed.

CAUTION

The version number stored in the FLASH is permanent until a higher number is downloaded from the Provisioning ﬁle or you selectSrvcs, System, Erase User Data on the IP Phone 1140E.

Optional keywords in the Provisioning ﬁle are:

•

PROMPT [YES | NO] is used to indicate if the IP Phone 1140E should

prompt the user for an update before the operation is performed. This command is optional with the default set to NO.

— YES - enables the prompt — NO - disables the prompt

•

PROTOCOL [TFTP | FTP | HTTP] deﬁnes the protocol used to download

the ﬁle. The IP Phone 1140E with SIP Firmware supports TFTP, FTP and HTTP protocols for ﬁle download. This command is optional. If it is not present, the default protocol TFTP is used.

ATTENTION

When using the TFTP protocol to transfer the ﬁrmware image, the average round trip time must be < 75 ms. The IP Phone will timeout and abort the ﬁrmware download if the total download time is > 10 minutes.

If the average round trip time will be >75 ms, use the FTP or HTTP protocol to transfer the ﬁrmware image.

If using FTP or HTTP, SRV_USER_NAME and SRV_USER_PASS are also key words. These commands specify the credential used to login to the ﬁle server for ﬁle download. If not present, the protocol default

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

•

SERVER_IP [address] allows the IP Phone 1140E to connect to the

speciﬁed IP address or name of the server for which the ﬁle can be downloaded. If the address is not speciﬁed, the SERVER_IP that is used is the same SERVER_IP that is used to download the provisioning ﬁle.

• DELETE_FILES [YES | NO], if present, erases the language and tone ﬁles stored in the IP Phone 1140E before new ﬁles are downloaded. Otherwise, new ﬁles with different names are added without erasing existing ﬁles. This command is optional. Note that there is a hard limit of 5 language ﬁles and 5 tone ﬁles that can be stored in the IP Phone 1140E. When the limits are exceeded, no new ﬁle can be accepted for download.

— YES - erases the existing language and tone ﬁles — NO - does not erase existing language and tone ﬁles

•

SRV_USER_NAME [username] - If the protocol is FTP or HTTP, this

keyword speciﬁes the user name to log on to the server.

•

SRV_USER_PASS [password] - If the protocol is FTP or HTTP, this

keyword speciﬁes the password to log on to the server.

The downloading of these ﬁles is initiated when an IP Phone 1140E is powered on, when an automatic check for updates is invoked, or when you select Srvcs, System, Erase User Data. Any of these actions causes the IP Phone 1140E to contact the provisioning server and attempt to read the Provisioning ﬁle. A Soft Reset (Srvcs, System, Reset Phone) does not cause the IP Phone 1140E to retrieve the Provisioning ﬁle.

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

To set the default language on a new IP Phone 1140E, or an IP Phone 1140E that has not been logged into by an end user, include the following in the [DEVICE_CONFIG] and [LANGUAGE] sections of the 1140eSIP.cfg conﬁguration ﬁle.

[DEVICE_CONFIG] DOWNLOAD_MODE AUTO VERSION 000002 FILENAME DeviceConﬁg.dat

[LANGUAGE] DOWNLOAD_MODE AUTO VERSION 0000000001 FILENAME French_d24.lng

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The DeviceConﬁg.cfg ﬁle should contain the following. DeviceConﬁg.cfg

DEF_LANG French_d24 On a new IP Phone 1140E, the language will switch to French after

downloading and processing the conﬁguration ﬁles. The login menu will be in French. On a subsequent bootup, the login menu and all boot messages will be in French.

For a new user login, the set will create a new user proﬁle. All menus remain in French. When a new user is created, the default language used is obtained from the DeviceConﬁg setting and stored as a user preference. After which the user preference for language is always used.

If a user has already logged in and either defaulted or chosen English as the user language preference, changing the conﬁguration ﬁles will not affect the users language display.

Create the device conﬁguration ﬁle on the provisioning server

After the IP Phone 1140E downloads the provisioning ﬁle, the IP Phone 1140E reads the [DEVICE_CONFIG] section and is directed to download the device conﬁguration ﬁle.

The device conﬁguration ﬁle is a clear text ﬁle and the naming convention is deﬁned by the administrator. See the FILENAME keyword in the [DEVICE_CONFIG] section of the SIP provisioning ﬁle.

The following is an example of a device conﬁguration ﬁle.

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Figure 5 Sample device conﬁguration ﬁle

Create the device conﬁguration ﬁle on the provisioning server 37

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

SIP Firmware Release 2.0 for IP Phone 1140E Administration

SERVER_PORT3_2 SERVER_PORT4_1 SERVER_PORT4_2 SERVER_PORT5_1 SERVER_PORT5_2 SERVER_RETRIES1

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Feature configuration commands

SERVER_IP1_2 SERVER_IP2_1 SERVER_IP2_2 SERVER_IP3_1 SERVER_IP3_2 SERVER_IP4_1 SERVER_IP4_2 SERVER_IP5_1 SERVER_IP5_2 SERVER_PORT1_1 SERVER_PORT1_2 SERVER_PORT2_1 SERVER_PORT2_2 SERVER_PORT3_1 VMAIL VMAIL_DELAY AUTOLOGIN_ENABLE

SERVER_RETRIES2 SERVER_RETRIES3 SERVER_RETRIES4 SERVER_RETRIES5 DNS_DOMAIN DEF_USERS DEF_USER1 DEF_USER2 DEF_USER3 DEF_USER4 DEF_USER5 UPDATE_USERS SIP_PING

MAX_INBOX_ENTRIES MAX_OUTBOX_ENTRIES MAX_REJECTREASONS

Feature configuration commands(continued)

AUTO_UPDATE AUTO_UPDATE_TIME AUTO_UPDATE_TIME_

RANGE TRANSFER_TYPE REDIRECT_TYPE ENABLE_PRACK

PROXY_CHECKING ENABLE_BT AUTH_METHOD BANNER FORCE_BANNER DST_ENABLED TIMEZONE_OFFSET FORCE_TIME_ZONE IM_MODE

MAX_CALLSUBJECT MAX_PRESENCENOTE DEF_LANG MAX_IM_ENTRIES MAX_ADDR_BOOK_ENTRIES ADDR_BOOK_MODE DEF_AUDIO_QUALITY HOLD_TYPE ENABLE_3WAY_CALL DISABLE_PRIVACY_UI DISABLE_OCT_ENDDIAL FORCE_OCT_ENDDIAL SNTP_ENABLE SNTP_SERVER MADN_TIMER MADN_DIALOG

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QoS and ToS commands

Tone configuration commands

NAT configuration commands

IM_NOTIFY DEF_DISPLAY_IM CALL_WAITING DISTINCTIVE_RINGING USE_RPORT TOVM_SOFTKEY_ENABLE TOVM_VOICEMAIL_ALIAS TOVM_VOICEMAIL_PARAM MAX_RING_TIME ENABLE_UPDATE DSCP_CONTROL

802.1P_CONTROL DSCP_MEDIA DIAL_TONE RINGING_TONE BUSY_TONE NAT_SIGNALLING NAT_MEDIA

DEFAULT_CFWD_NOTIFY FORCE_CFWD_NOTIFY RTP_MIN_PORT RTP_MAX_PORT SCA_HOLD_BEHAVIOR SCA_APPEARANCES SCA_BROADWORKS EXP_MODULE_ENABLE FORCE_REBOOT PROMPT_ON_LOCATION_OTHER

802.1P_MEDIA DSCP_DATA

802.1P_DATA FASTBUSY_TONE CONGESTION_TONE

STUN_SERVER_IP1 STUN_SERVER_IP2

Voice Quality Monitoring (VQMon) configuration commands

System commands

NAT_TTL

VQMON_PUBLISH VQMON_PUBLISH_IP LISTENING_R_ENABLE LISTENING_R_WARN LISTENING_R_EXCE PACKET_LOSS_ENABLE PACKET_LOSS_WARN

ADMIN_PASSWORD

STUN_SERVER_PORT1 STUN_SERVER_PORT2 PACKET_LOSS_EXCE JITTER_ENABLE JITTER_WARN JITTER_EXCE DELAY_ENABLE DELAY_WARN DELAY_EXCE SESSION_RPT_EN SESSION_RPT_INT

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Device conﬁguration commands

CAUTION

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

Parameters in the device conﬁguration ﬁle with empty values are not allowed and will cause write failure.

Server and network conﬁguration commands

•

SIP_DOMAIN[x] [domain_name] preconﬁgures the proxy domain

name for all servers. The same conﬁguration can be done through the domain conﬁguration menu on the IP Phone 1140E.

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

ATTENTION

Note: SIP_DOMAIN[x] is provisioned after user logout.

•

SERVER_IP[x]_[y]_ip_address] conﬁgures the primary and secondary

IP address for each domain, two proxies for each domain. — x - the domain number from 1 to 5. — y - the corresponding primary and secondary IP addresses. y=1

indicates the primary address and y=2 indicates the secondary address.

— ip_address - the IP address of the SIP proxy server.

•

SERVER_PORT[x]_[y] [port_number] conﬁgures the signaling ports

for each proxy. — x - the domain number. — y - the corresponding primary and secondary IP addresses. y=1

indicates the primary address and y=2 indicates the secondary address.

— port_number - the SIP proxy signaling port (default is 5060).

•

SERVER_RETRIES[x] [number_of_retries] conﬁrms the number of

retries for each domain. The default number of retries is 3. — x - the domain number from 1 to 5.

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— number_of_retries - the number of retry attempts to connect to

the proxy server.

•

DNS_DOMAIN [domain] is the DNS domain of the IP Phone 1140E.

•

DEF_USERS[x] [user_name] allows you to enter the default user name

for all domains. When the device conﬁguration ﬁle gets downloaded, the default user name is used when logging in.

— x - the domain number from 1 to 5. — user_name - the default user name.

•

UPDATE_USERS [YES | NO] affects the default user names stored in

the IP Phone 1140E. If this ﬂag is set to YES, the default user names are overwritten each time a new device conﬁguration ﬁle is downloaded.

— YES - the default user names are overwritten each time a new

device conﬁguration ﬁle is downloaded.

— NO - the default user names are not overwritten each time a new

device conﬁguration ﬁle is downloaded.

•

SIP_PING [YES | NO] The SIP_PING conﬁguration value is used to

maintain server heartbeat detection and to keep a ﬁrewall pinhole open. When used for server heartbeat detection, the IP Phone 1140E

periodically pings the SIP Proxy and awaits a response. When three attempts to ping the SIP Proxy fail, the IP Phone 1140E begins a failover process and attempts to connect to the next conﬁgured SIP Proxy IP in the same domain.

When a NAT TRAVERSAL method is selected, the SIP_PING conﬁguration value also helps keep a ﬁrewall pinhole open.

ATTENTION

Decide carefully whether SIP_PING usage is appropriate for your environment. Even when SIP_PING is not used for NAT TRAVERSAL, it is highly likely that you must keep SIP_PING enabled for server heartbeat detection.

If the IP Phone 1140E is behind a ﬁrewall, it is very likely that you must keep SIP_PING enabled, unless an alternate method of keeping the ﬁrewall pinhole open is used.

The default value is yes if not speciﬁed in the device conﬁguration ﬁle. If SIP_PING is changed in the Device conﬁguration ﬁle, the IP Phone 1140E must be rebooted for the change to take effect.

— YES - enables pinging

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— NO - disables pinging

Feature conﬁguration commands

• TOVM_SOFTKEY_ENABLE [YES | NO]

The transfer to Voice Mail feature is supported on MCS 5100 servers. — YES - enables the toVM soft key on the phone. — NO - disables the toVM soft key on the phone.

•

TOVM_VOICEMAIL_ALIAS <string> customizes the user ID of the

SIP URI of the voice mail system. The transfer to Voice Mail feature is supported on MCS 5100 servers.

•

TOVM_VOICEMAIL_PARAM<string> customizes the parameter name

of the SIP URI of the voice mail system. The transfer to Voice Mail feature is supported on MCS 5100 servers.

•

SCA_APPEARANCES sets the maximum number of appearances that

will be used for outgoing calls by the Shared Call Appearance (SCA) group. The valid range for this parameter is 2 to 24. The default value is

12. The SCA feature is supported on BroadWorks servers.

• SCA_BROADWORKS [YES | NO]

The SCA feature is supported on BroadWorks servers. — YES - activates the Broadsoft Shared Call Appearance (SCA)

feature.

— NO - deactivates the Broadsoft Shared Call Appearance (SCA)

feature. This is the default option.

•

SCA_HOLD_BEHAVIOR [PRIVATE | PUBLIC] sets the default behavior

of the hold button when user determined behavior does not exist. When a user creates a new proﬁle the default behavior is taken from this setting. After the creation of a new proﬁle this conﬁguration setting is not used. The default option is PUBLIC. The SCA feature is supported on BroadWorks servers.

• RTP_MIN_PORT

The minimum RTP port value is an integer between 1024 and 65535, exclusive of the restricted SIP ports between 5059 and 5080. The default value is 50000.

• RTP_MAX_PORT

The maximum RTP port value is an integer between 1024 and 65535, exclusive of the restricted SIP ports between 5059 and 5080. The default value is 50100.

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Create the device conﬁguration ﬁle on the provisioning server 43

Note 1: The RTP port conﬁguration parameters must satisfy the constraints that (RTP_MAX_PORT - RTP_MIN_PORT) is greater than or equal to 100 and less than 1000.

Note 2: If there is a provisioning error, RTP_MIN_PORT is reset to the default value of 50000 and RTP_MAX_PORT is reset to the default value of 50100. An error message will be logged. The SystemConﬁg ﬁle stores 50000 and 50100, rather than the erroneous conﬁguration values, to indicate that the conﬁguration attempt has been rejected.

• CALL_WAITING [SPEAKER | STREAM]

— SPEAKER - the call waiting tone is played on the phone speaker.

This is the default option.

— STREAM - the call waiting tone is injected into the stream played

on the transducer in use for the active call

• DISTINCTIVE_RINGING [YES | NO]

— YES - turns on the distinctive ringing feature. This is the default

option.

— NO - turns off the distinctive ringing feature.

• USE_RPORT [YES | NO]

— YES - allows the phone to work from behind and/or in front of a

symmetrical NAT with servers and/or clients that support RFC 3581.

— NO - disables implementation of support for RFC 3581. This is the

default option. Note: To provision USE_RPORT, the IP Phone 1140 must be

rebooted after the device conﬁguration ﬁle is updated. To force a hard reboot after the device conﬁguration ﬁle is updated set FORCE_REBOOT YES.

• EXP_MODULE_ENABLE [YES | NO]

— YES - the set will detect and enable an expansion module. — NO - the set will not detect an expansion module. This is the default

option.

•

MAX_RING_TIME [x] - an integer between 30 and 600 that sets the

number of seconds for incoming calls to ring before ignoring them. The default value is 120.

• ENABLE_UPDATE [YES | NO]

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— YES - enables UPDATE message support and adds “UPDATE” to

ALLOW header. This is the default option.

— NO - disables UPDATE message support.

Note: ENABLE_UPDATE is provisioned after user logout.

• FORCE_REBOOT [YES | NO]

— YES - forces hard reboot after device conﬁguration update. — NO - does not force hard reboot after device conﬁguration update.

This is the default option. Note: In order for FORCE_REBOOT to reboot the phone the

VERSION of the device conﬁguration ﬁle must be incremented, even if DOWNLOAD_MODE is set to FORCED.

• PROMPT_ON_LOCATION_OTHER [YES | NO]

— YES - prompt the user to select new location if location “other” was

previously selected.

— NO - do not prompt the user to select new location if location “other”

was previously selected. This is the default option.

•

VMAIL [vmail_number] is the voice mail address which can be the

URI or the DN number of the voice mail server. This command takes a string as a parameter. This is the default link for a new user proﬁle only. Individual users can customize the link through Prefs, Message Options, Voice Mail Settings. This command has no effect on the user proﬁles after it is created.

— vmail_number - the number or URI of the voicemail server.

• VMAIL_DELAY[x] is a delay, conﬁgured in milliseconds, between when the voice mail server answers the call and the start of dialing the voice mail user ID. The default value is 1000ms.

— x - the delay in milliseconds

• AUTOLOGIN_ENABLE [YES | NO | USE_AUTOLOGIN_ID] or [1 | 0 | 2] controls whether the set attempts to automatically login to the proxy

server. — YES - turns on the auto login feature. — NO - turns off the auto login feature. — USE_AUTOLOGIN_ID - enables the auto login id feature using

SIP Firmware Release 2.0 for IP Phone 1140E Administration

the userid speciﬁed in AUTOLOGIN_ID_KEY01 and the password

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Create the device conﬁguration ﬁle on the provisioning server 45

speciﬁed in AUTOLOGIN_PASSWD_KEY01 to register and authenticate. Both userid and password must be speciﬁed.

The AUTOLOGIN_ID_KEY01 and AUTOLOGIN_PASSWD_KEY01 parameters are deﬁned in the set speciﬁc conﬁguration ﬁle.

Note: When using this setting, the user will be prevented from logging out of the phone.

or — 1 - turns on the auto login feature. — 0 - turns off the auto login feature. — 2 - enables the auto login id feature using the userid speciﬁed

in AUTOLOGIN_ID_KEY01 and the password speciﬁed in AUTOLOGIN_PASSWD_KEY01 to register and authenticate. Both userid and password must be speciﬁed.

The AUTOLOGIN_ID_KEY01 and AUTOLOGIN_PASSWD_KEY01 parameters are deﬁned in the set speciﬁc conﬁguration ﬁle.

Note: When using this setting, the user will be prevented from logging out of the phone.

Note: If auto login id is enabled in the set speciﬁc conﬁguration ﬁle, it is recommended that AUTOLOGIN_ENABLE be set to either Yes/No or 1/0 in the device conﬁguration ﬁle. This recommendation facilitates migrating a set that uses the set speciﬁc conﬁguration ﬁle to not using the set speciﬁc conﬁguration ﬁle. The migration to just using the device conﬁguration ﬁle can be done by deleting the set speciﬁc conﬁguration ﬁle. If the device conﬁguration ﬁle does not have the matching parameters in the set speciﬁc conﬁguration ﬁle, the set will continue to use the previously assigned settings after the set speciﬁc conﬁguration ﬁle is deleted. This recommendation applies to other parameters in the set speciﬁc conﬁguration ﬁle.

• AUTO_UPDATE [YES | NO] is a command to enable or disable the automatic updating of the IP Phone 1140E with SIP Firmware conﬁguration ﬁles from the provisioning server. Enabling this command causes the IP Phone 1140E with SIP Firmware to check for updates once every day. The default is disabled.

— YES - turns on the AUTO_UPDATE feature. — NO - turns off the AUTO_UPDATE feature.

• AUTO_UPDATE_TIME [x] is the actual time in seconds, starting from midnight, before an automatic update occurs. Each IP Phone 1140E adds random numbers to the time speciﬁed by this command so every

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46 Conﬁgure the provisioning server

IP Phone 1140E does not try to access the provisioning server at the same time. By default the automatic update feature is disabled (see AUTO_UPDATE_RANGE).

— x - the time after midnight that the automatic update occurs.

•

AUTO_UPDATE_TIME_RANGE [x] is the range in hours, from the

AUTO_UPDATE_TIME whereby an IP Phone 1140E checks for updates from the server. The default range is 1 hour.

— x-the range in hours when the IP Phone checks for updates from

the server. The range can be from 1 to 6 hours.

•

TRANSFER_TYPE [MCS | STANDARD] is used to conﬁgure the IP

Phone 1140E to activate Nortel conference server-assisted attended transfers, instead of the industry standard method of attended transfers. The default setting is MCS.

— MCS - the typical attended transfer used by Nortel proxies. MCS

uses a conference server to do the attended transfer.

— STANDARD - the standard method of a transfer. This method does

not involve a conference server.

•

REDIRECT_TYPE [MCS | RFC3261] is a command used to select

different protocols for set redirection. The default setting is MCS. — MCS - when the IP Phone 1140E receives either 301 (moved

permanently) or 302 (moved temporarily) during registration, it is assumed the IP Phone 1140E is moved to a new MCS 5100 system (proxy+registrar) and all subsequent messages are sent to the new address.

— RFC3261 - the IP Phone 1140E assumes that, if during registration,

a 301 (moved permanently) is received, the message contains a new registrar address. The IP Phone 1140E tries to register to the registrar using the existing proxy.

•

ENABLE_PRACK [YES | NO] PRACK is utilized to make some SIP

messages reliable and requires that an ACK be sent with many SIP messages. ENABLE_PRACK is often utilized to verify that early media is being received. See RFC 3262 for details.

Note 1: ENABLE_PRACK must be set to NO when connected to the MCS 5100 Release 3.5 system.

Note 2: ENABLE_PRACK is provisioned after user logout.

— NO - disables PRACK and is the default value. — YES - enables PRACK.

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•

PROXY_CHECKING [YES | NO] enables and disables extra security

checking when i

ncoming requests are sent to the IP Phone 1140E. The IP Phone 1140E with SIP Firmware always sends requests through an outgoing proxy. However, it is possible, through this conﬁguration, to be able to accept an incoming request directly or through an incoming proxy.

— YES - means that the request must come directly from the proxy

server. YES is the default to enable proxy checking.

— NO - means the request can be sent directly to the IP Phone 1140E.

(NO is only suitable in a few situations).

•

ENABLE_BT [YES | NO] is a ﬂag to enable and disable Bluetooth

support in the IP Phone 1140E. — YES - enables Bluetooth. — NO - disables Bluetooth. The default is NO.

Note that the IP Phone 1140E hardware does not support Bluetooth and this command is ignored.

•

DEF_AUDIO_QUALITY [Low | Medium | High] is a command used for

setting the default audio quality used for each new call. Audio quality can be changed when the call is active. If this command is not present in the conﬁguration ﬁle, the IP Phone 1140E uses High quality as its default value. The possible parameters for this command are High, Medium, and Low. If any other parameter is entered or these commands are misspelled, the IP Phone 1140E uses High as the default setting.

The following codecs are used for each selection: — Low - G729 ptime 30. — Medium - G711 ptime 30. — High - G711 ptime 20.

•

AUTH_METHOD [AUTH | AUTH_INT] is used to conﬁgure the SIP

authentication method. — AUTH - only authenticates (username/password) — AUTH_INT - authentication plus integrity checking (an MD5 hash of

• BANNER [banner_text] preconﬁgures the banner on the IP Phone 1140E. Use a text string to set the banner. For example, BANNER ABC Company sets the banner to ABC Company. The text string can have a maximum of 24 characters.

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

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•

FORCE_BANNER [YES | NO] is set by the system administrator

through the conﬁguration ﬁle. If FORCE_BANNER is set to YES, the banner from the conﬁguration ﬁle is reloaded each time the IP Phone 1140E powers up, even if the user changes the banner manually.

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

banner set by the user.

— NO - allows the user to set the banner.

•

DST_ENABLED [YES | NO] enables and disables the daylight saving

time mechanism. The time received from the server is GMT and is converted to the proper timezone by the IP Phone 1140E. If the Daylight Saving Time feature is enabled, the IP Phone 1140E automatically calculates the DST time at the appropriate date and converts the time to and from DST. The calculations used are based on the new rules applicable to DST in 2007. The IP Phone 1140E is set up to use the North American DST scheme.

— YES - enables daylight savings time. — NO - disables daylight savings time.

•

TIMEZONE_OFFSET [x] is used to set the current time zone offset from

GMT in seconds. TIMEZONE_OFFSET takes a number as a parameter. For example, TIMEZONE_OFFSET -25200 sets the time zone offset to MST, which is GMT-7 (-7*3600 = -25200 seconds).

Table 6 Time zone offset

Location Time zone offset (seconds)

(GMT-10:00) Hawaii (GMT-09:00) Alaska (GMT-08:00) Pacific time (US and Canada) (GMT-07:00) Mountain time (US and Canada) (GMT-06:00) Central time (US and Canada) (GMT-05:00) Eastern time (US and Canada) (GMT-04:00) Atlantic time (US and Canada) (GMT-03:00) Brasilia, Buenos Aires (GMT+00:00) Greenwich, Dublin, Lisbon, London (GMT+01:00) Amsterdam, Berlin, Rome, Stockholm,

Madrid, Paris

-36000

-32400

-28800

-25200

-21600

-18000

-14400

-10800 0 3600

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

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(GMT+03:00) Moscow, St. Petersburg (GMT+05:30) Bombay, Calcutta, Madras, New Delhi (GMT+08:00) Beijing, Chongqing, Hong Kong,

Singapore, Taipei (GMT+09:00) Osaka, Sapporo, Tokyo, Seoul (GMT+10:00) Canberra, Melbourne, Sydney (GMT+12:00) Auckland, Wellington

•

FORCE_TIME_ZONE [YES | NO] allows you to force the timezone

offset on each user’s IP Phone 1140E. The default is NO. — YES - forces the IP Phone to use the TIMEZONE_OFFSET speciﬁed

in the device conﬁguration ﬁle.

— NO - uses the value stored in the user preferences.

•

IM_MODE [ENCRYPTED | TEXT | SIMPLE | DISABLED] is used to

conﬁgure the mode of instant messaging (IM). The default setting is ENCRYPTED.

— ENCRYPTED - instant messages are sent encrypted. — TEXT - instant messages are sent as text.

10800 18000 28800

32400 36000 43200

— SIMPLE - instant messages are sent using SIP for Instant Messaging

and Presence Leveraging Extensions (SIMPLE) protocol.

— DISABLED - instant messaging is turned off and no instant

messages can sent or received.

•

IM_NOTIFY [YES | NO] is used to turn on or off the Blue LED indicator

upon receipt of an instant message. — YES - the Blue LED functions when an instant message is received. — NO - the Blue LED does not function when an instant message is

received.

Note: If IM_NOTIFY is disabled, the Blue LED continues to operate for other features.

•

DEF_DISPLAY_IM [YES | NO] enables or disables the display of instant

messages (IM). The default setting is NO. — YES - enables display of IMs. — NO - disables display of IMs.

•

MAX_INBOX_ENTRIES [x] is used to restrict the maximum number

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

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MAX_INBOX_ENTRIES 100 limits the number of entries in the inbox to

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

•

MAX_OUTBOX_ENTRIES [x] is used to restrict the maximum number

of outbox entries and takes a number as a parameter. For example, MAX_OUTBOX_ENTRIES 100 limits the number of entries in the outbox to 100. The default limit is 100.

— x - the maximum number of outbox entries.

•

MAX_REJECTREASONS [x] is used to restrict the maximum number of

Call Decline Reasons (Prefs, Feature Options, Call Decline Reasons) and takes a number as a parameter. The default limit is 20.

— x - the maximum number of reject reasons.

•

MAX_CALLSUBJECT [x] is used to restrict the maximum number

of call subjects (Prefs, Feature Options, Call Subject) and takes a number as a parameter. The default limit is 20.

— x - the maximum number of call subject reasons.

•

MAX_PRESENCENOTE [x] is used to restrict the maximum number

of presence notes and takes a number as a parameter. The default limit is 20.

— x - the maximum number of presence notes that an IP Phone 1140E

can receive.

•

DEF_LANG [language] is a command used for setting the default

language. Select one of the supported languages from the language list downloaded. Note that the corresponding language ﬁle must be downloaded and stored in the IP Phone 1140E through the [LANGUAGE] section in Provisioning. If the language ﬁle is not stored in the IP Phone 1140E, the default language English is used.

— language - the default language used.

•

MAX_IM_ENTRIES [x] is used to set the maximum number of instant

message (IM) entries and takes a number as a parameter. Once the maximum number is reached, the oldest IM is deleted without any user notiﬁcation. The default limit is 999.

— x - the maximum number of instant messages.

•

MAX_ADDR_BOOK_ENTRIES [x] is used to set the maximum number

of entries in the address book and takes a number as a parameter. The default limit is 100.

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— x - the maximum number of address book entries.

•

ADDR_BOOK_MODE [NETWORK | LOCAL | BOTH] is a command

to choose the address book that is used to search for other users. The default setting is NETWORK.

— NETWORK - downloads the user’s address book from the network

(MCS). New address book entries are uploaded to the network.

— LOCAL - creates a user address book and stores it locally on the IP

Phone

— BOTH - attempts to download a network address book and keep a

copy on the IP Phone. If a network address book is available, the IP Phone 1140E functions as if NETWORK mode has been selected.

•

HOLD_TYPE [RFC2543 | RFC3261] is used to select the protocol to

hold a call. The default setting is RFC3261. — RFC2543 - RFC 2543 is a standard protocol of the Internet

Engineering Task Force (IETF).

— RFC3261 - RFC3261 is a standard protocol of the IETF.

•

ENABLE_3WAY_CALL [YES | NO] is a ﬂag to enable or disable local

phone-based three-way calling for three-party conferences. — YES - enables local (phone-based) three-way calling for three-party

conferences. YES is the default.

— NO - disables local (phone-based) three-way calling.

•

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

setting in UI menus. Disabling the privacy setting in UI menus disables the user ability to set privacy options.

— YES - disables the privacy setting in the UI menus. — NO - enables the privacy setting in the UI menus. NO is the default.

•

DISABLE_OCT_ENDDIAL [YES | NO] is a ﬂag used to conﬁgure the

pound (#) key. The default setting is YES. — YES - the pound (#) key initiates dialing when pressed after a

telephone number is entered.

— NO - the pound (#) key functions as any other digit or character

on the dial pad typically used in networks that use vertical service codes or access codes.

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•

FORCE_OCT_ENDDIAL [YES | NO] is a ﬂag used to override attempts

to change the function of the pound (#) key on the Graphical User Interface (GUI). The default setting is NO.

— YES - overrides attempts to change the function of the pound (#)

key on the GUI.

— NO - does not override a change of the function of the pound (#)

key on the GUI.

•

SNTP_ENABLE [YES | NO] allows the IP Phone 1140E to obtain the

time and date from an NTP server. The default is NO. The IP Phone 1140E updates the time once every 24 hours from the

NTP server. If the IP Phone 1140E cannot contact the server, the IP Phone 1140E tries every 15 minutes up to a maximum of 6 attempts, and then hourly attempts are made. If SNTP_ENABLE is set to NO the IP Phone 1140E tries to retrieve the time and date from the SIP proxy server. However, not all SIP proxy servers support this method of retrieving the time and date.

— YES - enables NTP. — NO - disables NTP.

•

SNTP_SERVER [ip_address] is the IP address or FQDN of the NTP

server that provides the time and date to the IP Phone 1140E. If this is not speciﬁed, the IP Phone does not generate any NTP requests.

— ip_address - the IP address of the NTP server in either Fully

Qualiﬁed Domain Name (FQDN) or non-FQDN format.

•

MADN_TIMER [x] is used to set the MADN polling timer interval (the

interval at which the IP Phone 1140E attempts to determine the MADN group of the logged-in user). The minimum value for the polling interval is 900 seconds (15 minutes). The default value is 1800.

— x - the time delay (in seconds) between queries to ﬁnd the MADN

group DN of a user. The minimum value 900.

•

MADN_DIALOG [YES |NO] is used to set the SIP URI or the GROUP

DN for the subscription to the dialog event. The default value is NO. — YES - subscribes to the dialog event using the SIP URI of the user. — NO - subscribes to the dialog event using the group of the user.

Note: The SIP URI is used to subscribe for MADN support for CS

2000.

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•

DEFAULT_CFWD_NOTIFY [YES | NO] is used to conﬁgure the "ring

splash" which occurs when either local call forwarding (MCS 5100) or network-based call forwarding have been enabled (CS 2000, CS 2100). If this conﬁguration value is enabled, the IP Phone 1140E plays an abbreviated ring tone to remind the user that a call has been forwarded. This conﬁguration value only effects users when their user proﬁle is ﬁrst created, unless the FORCE_CFWD_NOTIFY ﬂag is also used. The default setting is NO

— YES - a brief ring splash plays when a call is forwarded. — NO - the ring splash is not be played.

•

FORCE_CFWD_NOTIFY [YES | NO ] allows the administrator to force

the behavior of the DEFAULT_CFWD_NOTIFY value on all users who login to the IP Phone 1140E. The default setting is NO.

— YES - the DEFAULT_CFWD_NOTIFY conﬁguration value is forced

into effect for the user.

— NO - the conﬁguration value is not be forced into effect for the user.

QoS and ToS commands

•

DSCP_CONTROL [x] is a value entered in decimal format between -1

and 63. If the value is -1, the DSCP value is picked up by the Service Package. The default value is 40.

— x - a value from -1 to 63 indicating the DSCP value.

•

802.1P_CONTROL [x] is a value entered in decimal format between -1

and 7 representing the 802.1P value in the SIP signaling packets. If the value is -1, the 802.1P value is retrieved from the Service Package. The default value is 6.

— x - the value from -1 to 7 indicating the 802.1P value.

•

DSCP_MEDIA [x] is a value entered in decimal format between -1 and

63 representing the DSCP value in the Real-time Transfer Protocol packets. If the value is -1, the DSCP value is retrieved from the Service Package. The default value is 44.

— x - a value from -1 to 63 indicating the DSCP value.

•

802.1P_MEDIA [x] is a value entered in decimal format between -1 and

7 representing the 802.1P value in the IP Phone 1140 Media (RTP) packets. If the value is -1 then the 802.1P value is retrieved from the Service Package is the 802.1 setting for media Real-time Transport Protocol (RTP). The default value is -1.

— x - a value from -1 to 7 indicating the 802.1P value.

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•

DSCP_DATA [x] is a value entered in decimal format between -1 and

63 representing the DSCP value in the provisioning packets. If the value is -1, the DSCP value is retrieved from the Service Package. The default value is 40.

— x - a value from -1 to 63 indicating the DSCP value.

•

802.1P_DATA [x] is a value entered in decimal format between -1 and 7

representing the 802.1P value in the provisioning packets. If the value is -1, the 802.1P value is retrieved from the Service Package. The default value is 6.

— x - a value from -1 to 7 indicating the 802.1P value.

Tone conﬁguration commands

•

DIAL_TONE [frequency1 | frequency2 | on_time | off_time] is used to

select the tone advising the caller that the exchange is ready to receive call information and invites the user to start sending call information. You can select the country speciﬁc tone. The default tone is the North American tone.

— frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates

a continuous tone.

— off_time - the duration when no tone is played.

The following are examples of DIAL_TONE:

1. 350,440;-1 (350 and 440 Hz continuous tone)

•

RINGING_TONE [frequency1 | frequency2 | on_time | off_time] is

used to select the tone advising the caller that a connection is made and a calling signal is applied to a telephone number or service point. You can select the country speciﬁc tone. The default tone is the North American tone.

— frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates

a continuous tone.

— off_time - the duration when no tone is played. The following are examples of RINGING_TONE:

1. 440,480; 2000,4000 (440 and 480 Hz with 2 seconds on, 4 seconds

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•

BUSY_TONE [frequency1 | frequency2 | on_time | off_time] is used

to select the tone advising the caller that the telephone number is busy. You can select the country speciﬁc tone. The default tone is the North American tone.

— frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates

a continuous tone.

— off_time - the duration when no tone is played.

•

FASTBUSY_TONE [frequency1 | frequency2 | on_time | off_time] is

used to select the tone advising the caller that the telephone number is busy. It is fast in cadence or frequency. You can select the country speciﬁc tone. The default tone is the North American tone.

— frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates

a continuous tone.

— off_time - the duration when no tone is played.

• CONGESTION_TONE [frequency1 | frequency2 | on_time | off_time]

is used to select the tone advising the caller that the groups of lines or switching equipment necessary for setting up the required call, or for the use of a speciﬁc service, are temporarily engaged. You can select the country speciﬁc tone. The default tone is the North American tone.

— frequency1 - the frequency of tone 1. — frequency2 - the frequency of tone 2. — on_time - the duration of the tone when it is on. A -1 indicates

a continuous tone.

— off_time - the duration when no tone is played. The IP Phone 1140E supports using WAV ﬁles to replace the ringtone

Frequency/Cadence pattern. For a system-wide setting, the country default values can be used.

NAT conﬁguration commands

•

NAT_SIGNALLING [NONE | SIP_PING | STUN] indicates the type

of protocol used for NAT traversal in the signaling port. The IP Phone 1140E with SIP Firmware supports two methods of NAT traversal of the signaling path: SIP_PING and STUN.

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— NONE - If the value is not set to None, this parameter overrides the

value of the parameter SIP_PING in the device conﬁguration ﬁle.

— SIP_PING - a Nortel proprietary NAT traversal protocol. Note that

SIP_PING only supports NAT traversal in the signaling port.

— STUN - the most common NAT traversal method.

•

NAT_MEDIA [NONE | STUN] indicates the type of protocol used for

NAT traversal in the media ports. The default is NONE. — NONE - is the default and disables NAT_MEDIA. — STUN - the most common NAT traversal protocol for the media (RTP

and Real-time Control Protocol [RTCP]) port.

— x - is the binding lifetime in seconds.

NAT_TTL [x] is used for future development. Currently, the default value is 2 minutes (120 seconds) and SIP IP Phones 1120E/1140E do not process or use the value deﬁned in NAT_TTL [x]. SIP IP Phones 1120E/1140E always ping the ports at regular intervals of 60 seconds regardless of the NAT_TTL value.

ATTENTION

•

STUN_SERVER_IP1[ip_address] NAT traversal using STUN protocol

requires a STUN server in the public internet. Two STUN server IPs can be provisioned.

— ip_address - is the IP address of STUN server 1.

•

STUN_SERVER IP2[ip_address] NAT traversal using STUN protocol

requires a STUN server in the public internet. Two STUN servers IPs can be provisioned.

— ip_address - is the IP address of STUN server 2.

•

STUN_SERVER_PORT1[port_number] is the port number used

corresponding to STUN_SERVER_IP1. The default port number is

3478. — port_number - is the port number.

•

STUN_SERVER_PORT2[port_number] is the port number used

corresponding to STUN_SERVER_IP2. The default port number is

3478. — port_number - is the port number.

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VQMon conﬁguration commands

It is important to read "How VQMon works" (page 92) before conﬁguring the VQMON parameters.

•

VQMON_PUBLISH [YES | NO] is the command that is used to enable

or disable the publish message containing the voice quality monitoring metrics sent to the Proactive Voice Quality Monitoring (PVQMoN) collecting server.

— YES - enables VQMoN. — NO - disables VQMoN. NO is the default.

•

VQMON_PUBLISH_IP [xxx.xxx.xxx.xxx] is used to set the IP address

of the PVQMoN server that collects voice quality monitoring metrics from the publish message.

This IP address is used within the report only.

•

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

alerts based on the Listening R Minor and Major Thresholds. The default value is vocoder-dependent using a scale from 1 (lowest quality) to 100 (highest quality). Currently default values are used based on VOCODER on a per call basis as summarized below.

— YES - enables the sending of the alert report based on the Listening

R Value.

— NO - disables the sending of the alert report based on the Listening

R Value.

VOCODER_G711_ULAW VOCODER_G711_ULAWPLP

VOCODER_G723 VOCODER_FLAG_G723_RATE_53

VOCODER_FLAG_G723_RATE_63 VOCODER_G729

VOCODER_G722 VOCODER_PCM16 VOCODER_PCM8 vqmonVocoderTypeUnknown

• LISTENING_R_WARN [xx] is the threshold to send a report on Listening R less than [xx]. The default value is 70. Using 0 will reset it to default based on far end VOCODER.

— xx - is an INTEGER value used as threshold.

LISTENING_R_WARN = 80 LISTENING_R_EXCE = 70

LISTENING_R_WARN = 60 LISTENING_R_EXCE = 50

LISTENING_R_WARN = 70 (default if not configured and unknown type) LISTENING_R_EXCE = 60

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•

LISTENING_R_EXCE [xx] is the threshold to send a report on Listening

R less than [xx]. The default value is 60. Using 0 will reset it to default based on far end VOCODER.

— xx - is an INTEGER value used as threshold.

•

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

alerts based on the packet loss thresholds. Packet loss is the fraction of RTP data packets from the source lost since the beginning of reception. The value is an integer scaled by 256. The range is 1 to 25600.

— YES - enables the sending of alert report based on the packet loss — NO - disables the sending of alert report based on the packet loss

•

PACKET_LOSS_WARN [xx] is the threshold to send a report on Packet

Loss greater than [xx]. The default is 256 (1%). Using 0 will reset the threshold to default.

— xx - is an INTEGER value scaled by 256 that is used as threshold.

The range is 1 to 25600.

•

PACKET_LOSS_EXCE [xx] is the threshold to send a report on Packet

Loss greater than [xx]. The default is 1280 (5%). Using 0 will reset the threshold to default.

— xx - is an INTEGER value scaled by 256 that is used as threshold.

The range is 1 to 25600.

•

JITTER_ENABLE [YES | NO] is used to enable or disable alerts based

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

— YES - enables the sending of alert report based on jitter detection — NO - disables the sending of alert report based on jitter detection

• JITTER_WARN [xx] is the threshold to send a report on Inter-arrival Jitter greater than [xx]. 1 second is broken up into 65535 (0xffff hex) parts. [xx] / 65535 is the threshold in seconds. The default is 3276 (50 ms). Using 0 will reset the threshold to default.

— xx - is an INTEGER value used as threshold

•

JITTER_EXCE [xx] is the threshold to send a report on Inter-arrival

Jitter greater than [xx]. 1 second is broken up into 65535 (0xffff hex) parts. [xx] / 65535 is the threshold in seconds. The default is 32760 (500 ms). Using 0 will reset the threshold to default.

— xx - is an INTEGER value used as threshold

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•

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

based on the excessive delay detection. This is the one-way delay (including system delay) for the call, measured in milliseconds.

— YES - enables Excessive delay detection. — NO - disables Excessive delay detection.

• DELAY_WARN [xx] is the threshold to give warning on Excessive Delay greater than [xx]. The default is 150 ms. Using 0 will reset the threshold to default.

— xx - is an INTEGER value used as a threshold measured in 1/1000

of a second.

•

DELAY_EXCE [xx] is the threshold to report unacceptable Excessive

Delay greater than [xx]. The default is 175 ms. Using 0 will reset the threshold to default.

— xx - is an INTEGER value used as a threshold measured in 1/1000

of a second.

•

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

VQMon session reports. The default is disabled. Both session report enable and session report interval must be set if the

phone ﬁrmware has been upgraded to SIP Release 2.0. Otherwise, the SESSION_RPT_INT default of 60 seconds will be used automatically.

— YES - enables periodic VQMon session reports. — NO - disables periodic VQMon session reports. Default is NO.

•

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

VQMon session report in seconds. The minimum acceptable value is 60 seconds. The maximum acceptable value is 600 seconds. The default is 60 seconds.

— xx - is an INTEGER value in seconds.

System commands

•

ADMIN_PASSWORD [password] is used to change the default

administrator password of the IP Phone 1140E that is used for unlocking network menus. The default is 26567*738.

— password - the administrator password.

Create the set conﬁguration ﬁle on the provisioning server

If the IP Phone 1140E encounters a [USER_CONFIG] section while parsing the 1140eSIP.cfg conﬁguration ﬁle, the set will download its set speciﬁc conﬁguration ﬁle SIP<mac id>.cfg.

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Set speciﬁc conﬁguration ﬁles support customizing the set on a per set/user level. Parameters in the device conﬁguration ﬁle can be overwritten with a set speciﬁc conﬁguration ﬁle.

Most of the parameters in the set conﬁguration ﬁle are saved on the set. Removing a parameter from the set conﬁguration ﬁle does not change the parameters saved on a conﬁgured set. If a parameter is set only in the set speciﬁc conﬁguration ﬁle, removing the set speciﬁc conﬁguration ﬁle will not clear the setting.

Note: If the 1140eSIP.cfg conﬁguration ﬁle contains a [USER_CONFIG] section, it is recommended that DOWNLOAD_MODE be set to FORCED. This is a global setting for all sets used to determine if the mac id ﬁle should be read. Alternatively, if the user wishes to use DOWNLOAD_MODE set to AUTO, when a change is made to any mac id ﬁle the version number should be incremented so that all phones read the ﬁle.

Table 7 "Set conﬁguration commands" (page 60) provides a summary of the

commands that can be used in the set conﬁguration ﬁle. The syntax of each command is summarized in "Set conﬁguration commands" (page 60).

Table 7 Set conﬁguration commands

Auto login

Set conﬁguration commands

•

AUTOLOGIN_ID_KEY01 [* | xx] This parameter is located within the

set speciﬁc conﬁguration ﬁle. This is the id the set will use to register and authenticate. The default user id "user1" will be used, if an id is not supplied and the set has not previously logged in.

— * - indicates that the set should use its mac address (lower case)

as the user id

— xx - an ASCII string that corresponds to the user id.

Note: To provision AUTOLOGIN_ID_KEY01, the IP Phone 1140 must be rebooted after the set conﬁguration ﬁle is updated. To force a hard reboot after the set conﬁguration ﬁle is updated set FORCE_REBOOT YES in the device conﬁguration ﬁle.

•

AUTOLOGIN_PASSWD_KEY01 This parameter is located within the

set speciﬁc conﬁguration ﬁle. There is not a default password. If this is

AUTOLOGIN_ID_KEY01 AUTOLOGIN_PASSWD_KEY01

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Create the Dialing Plan ﬁle on the provisioning server 61

blank and AUTOLOGIN_ENABLE is set to USE_AUTOLOGIN_ID (or 2) in the device conﬁguration ﬁle, the set will not login.

Note: To provision AUTOLOGIN_PASSWD_KEY01, the IP Phone 1140 must be rebooted after the set conﬁguration ﬁle is updated. To force a hard reboot after the set conﬁguration ﬁle is updated set FORCE_REBOOT YES in the device conﬁguration ﬁle.

Create the Dialing Plan ﬁle on the provisioning server

A dialing plan essentially describes the number and pattern of digits that a user dials to reach a particular telephone number. Access codes, area codes, specialized codes, and combinations of the number of digits dialed are all part of a dialing plan.

The purpose of the dialing plan is so that the end user does not have to press the send or pound key (#) to have the IP Phone 1140E with SIP Firmware send the initial message to start the call.

Dialing a telephone number on an IP Phone that supports SIP can be different than dialing a number from a traditional telephone. SIP signaling is communicated through a SIP URI to get to the far end. For example, you can key in the SIP address, jsmith@yourcompany.com to reach John Smith. When the IP Phone 1140E with SIP Firmware receives this address, the dialing plan is bypassed and the IP Phone 1140E simply uses the SIP URI to send a SIP INVITE to jsmith@yourcompany.com (INVITE sip: jsmith@yourcompany.com).

Entering a SIP URI address, however, is inconvenient for an IP Phone with SIP Firmware unless a USB keyboard is attached. Also, the user must explicitly press the send key (or use some method to indicate the end of the URI) to indicate the completion of the SIP address. This is not something that the user is accustomed to in a traditional PBX environment.

The alternative is to use a URI where numbers are used to reach the far end. Using different access codes, the IP Phone with SIP Firmware translates the digits entered into something that the server can understand and remaps the number entered into different URIs. Some of the numbers are mapped as intercom calls, some numbers are mapped as local Public Switched Telephone Network (PSTN) calls, and some numbers are mapped as public long-distant calls.

The issue is that until the IP Phone itself can determine the type of call, no SIP INVITE message is sent. This is where the dialing plan comes into effect. The call type is determined by the dialing plan. Based on the rules deﬁned in the dialing plan, once a match has been identiﬁed, the IP Phone 1140E with SIP Firmware sends the invite without the need to press the send key. This behavior closely matches the traditional PBX operation.

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62 Conﬁgure the provisioning server

The IP Phone 1140E with SIP Firmware design places no restriction in the format of the SIP URI. The dialing plan is a scheme to match the user experience with traditional PBX operation. It does not restrict the type of URI that the user can use.

The IP Phone 1140E with SIP Firmware uses a dialing plan to recognize a call as an emergency call when it sends an INVITE. The dialing plan can have multiple emergency numbers. See the chapter "Emergency 911 -

Operator control of disconnect" (page 125) for information on the handling

of Emergency 911 calls by the IP Phone 1140E with SIP ﬁrmware. The following is an example of a dialing plan.

Figure 6 Sample dialing plan

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Dialing function description

Dialing plan

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

Figure 7 Sample dialing plan declarations section

In the declaration section, the administrator can deﬁne the variables. The variables must start with a dollar ($) sign, followed by a number or a character, such as $1 or $a. There are two variables that are reserved by system. They are as follows:

Create the Dialing Plan ﬁle on the provisioning server 63

$$ : used for the collected digits if they match the pattern $t : default timer There must be a domain name deﬁned and the domain name can be

represented by any variable. In Figure 6 "Sample dialing plan" (page 62), the domain name is represented by $n.

The variable deﬁnitions take the form:

Figure 8 Sample dialing plan variable deﬁnitions

For example: $1="nortel.com" $2="Nortel"

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64 Conﬁgure the provisioning server

$3="." $4="com" $5="Nortel.com" $t=10000 (default timer is 10 seconds) $a=Nortel.com

The second section of dialing plan contains the digit map. The digit map section has three subsections that are divided by a separator of two ampersands (&&).

Figure 9 Sample dialing plan digit map section

The ﬁrst part of a dialing plan contains a pattern deﬁned with DRegex, which is used for matching the dialed number. The patterns are separated by the pipe (|) sign. The second part contains the result string used in the dial step. The third part deﬁnes the parameters used by UA in dialing action.

The following parameter is currently deﬁned: t=xxxx: After this timer expires, the number entered is automatically dialed.

The timer starts after the ﬁrst digit is entered and after it expires, the collected digits are automatically dialed out. xxxx is a decimal number in msec. The default timer is used when t is not speciﬁed in the digit map.

For example: X{4} && sip:$$; phone-context=nortel.com;user=phone && t=7000 When the user presses any 4 digits, such as 4567, the following SIP URIs

are generated because of the translation rule: Sip:4567; phone-context=nortel.com;user=phone. The timeout of stopping

the collection of digits is 7 seconds.

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The pound sign (#) at the end of the digit map causes the IP Phone 1140E to dial the matched dialing plan immediately.

DRegex

The Digit Regular Expression (DRegex) syntax is a telephony-oriented mapping of Portable Operating System Interface (POSIX) Extended Regular Expressions (ERE). Users must take care not to confuse the DRegex syntax with POSI EREs as they are not identical. In particular, there are many features of POSIX EREs that DRegex does not support. The dialing plan uses DRegex instead of ERE. The following rules demonstrate the use of DRegex.

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66 Conﬁgure the provisioning server

Figure 10 DRegex rules

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Downloadable WAV ﬁles

It is possible to customize the ring tones on the IP Phone 1140E with SIP Firmware. Up to ﬁve special ring tones can be downloaded from the provisioning server and stored on the IP Phone 1140E. The end user can select which ring tone they would like to implement.

In order to download these special ﬁles, the ﬁles must reside on the provisioning server and be speciﬁed in the SIP provisioning ﬁle. For more information, see "Download the SIP Firmware to the provisioning server"

(page 30). The WAV ﬁles have a maximum size of 512 KB each for the IP

Phone 1140E. The ﬁle format is restricted to ITU-T A-law or u-law (8.0 kHz, 8-bit, mono

or 16.0 kHz, 16 bit mono). After the WAV ﬁles are downloaded to the IP Phone 1140E, the WAV ﬁle

names appear in Pref, Audio, Tones, Ring Pattern(1 to 8 are standard ring tones, and 9 and above are WAV ring tones) and the WAV ring tones can then be selected to replace the standard ring tones.

Downloadable WAV ﬁles 67

For further information about downloadable WAV ﬁles, see the SIP Firmware Release 2.0 for IP Phone 1140E User Guide (NN43113-101).

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Conﬁgure the DHCP server

Use DHCP to provide partial Device Settings or full Device Settings, as well as to automatically provide Voice VLAN IDs.

Table 8 Characteristics of Partial DHCP mode and Full DHCP mode

Partial DHCP mode Partial DHCP mode retrieves the following network

parameters from the DHCP server: IP address, subnet mask, and default gateway configuration for the IP Phone 1140E.

Full DHCP mode As well as those parameters retrieved in Partial DHCP

mode, Full DHCP also retrieves the following from the DHCP server: DNS server and Provisioning Server.

Auto VLAN Discovery Auto VLAN Discovery retrieves the Voice VLAN ID for

the IP Phone 1140E with SIP Firmware.

Partial DHCP mode

After the IP Phone 1140E with SIP Firmware is conﬁgured to operate in Partial DHCP mode, the DHCP server does not need any special conﬁguration to support an IP Phone 1140E with SIP Firmware. The IP Phone 1140E with SIP Firmware receives the following Device Settings parameters from the DHCP server in the following table:

Table 9 DHCP options in Partial DHCP mode

Parameter requested by the

IP Phone 1140E

Subnet mask The subnet mask of the IP

Phone 1140E.

Router and gateways IP address of the default

gateway of the IP Phone 1140E.

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70 Conﬁgure the DHCP server

Lease time Implementation varies

according to DHCP server configuration.

Renewal time Implementation varies

according to DHCP server configuration.

Rebinding interval Implementation varies

according to DHCP server configuration.

Partial DHCP is the response from a DHCP server that is not conﬁgured to recognize the Nortel-speciﬁc Vendor Class Identiﬁer in option 60 (Nortel-SIP-Phone-A). After the IP Phone 1140E is conﬁgured to operate in Partial DHCP mode, the DHCP server needs no special conﬁguration to support an IP Phone 1140E. Using Partial DHCP, an IP Phone 1140E can obtain its IP address, subnet mask, and gateway IP address. The remainder of the conﬁguration information is manually entered on the IP Phone 1140E. This includes the DNS server IP and the Provisioning server address and protocol.

Full DHCP mode

The DHCP server requires special conﬁguration in Full DHCP mode. The IP Phone 1140E with SIP Firmware obtains Device Settings parameters from specially conﬁgured DHCP servers.

The IP Phone 1140E with SIP Firmware requests the following Device Settings parameters from the DHCP server:

• IP address conﬁguration for the IP Phone 1140E

•

subnet mask for the IP Phone 1140E IP address

•

default gateway for the IP Phone 1140E subnet

• DNS server

•

Provisioning Server

It is also possible to obtain the DNS IP automatically using Full DHCP mode without any special conﬁguration of the DHCP server. This means that there is no need to conﬁgure the Nortel-speciﬁc Vendor Class Identiﬁer (Option

60) on the DHCP server. However, the provisioning server address needs

to be manually deﬁned within the Device Settings menu. This conﬁguration requires toggling to Partial DHCP mode to deﬁne the provisioning server address and toggling back to Full DHCP mode after this is complete.

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Conﬁgure the DHCP server to support Full DHCP mode

SIP IP Phone class identiﬁer

A Nortel-SIP-Phone-aware DHCP server is needed for Full DHCP mode. For Full DHCP mode, the DHCP server requires special conﬁguration.

After the DHCP server is conﬁgured to recognize the IP Phone 1140E with SIP Firmware as a unique IP Phone, the DHCP server can treat the IP Phone 1140E differently than other DHCP IP Phones. An IP Phone-aware DHCP server can automatically conﬁgure Nortel IP Phones by sending all information that the IP Phone requires.

The IP Phone 1140E and the DHCP server communicate using a unique class identiﬁer. After the IP Phone 1140E ﬁrst sends the DHCP DISCOVER, it includes the Nortel-SIP-Phone-A ASCII string within the Vendor Class Identiﬁer (Option 60). The DHCP server recognizes this special Vendor Class Identiﬁer (Option 60) and sends back OFFER, which also includes the same Vendor Class Identiﬁer. This makes it possible to notify the IP Phone 1140E with SIP Firmware that the server is IP Phone-aware, and that it is safe to accept the offer from the server.

Full DHCP mode 71

Every Nortel IP Phone 1140E with SIP Firmware ﬁlls in the Vendor Class ID option of the DHCPDISCOVER and DHCPREQUEST messages with the null-terminated, ASCII-encoded string Nortel-SIP-Phone-A, where A identiﬁes the version number of the information format of the IP Phone.

The Class Identiﬁer Nortel-SIP-Phone-A must be unique in the DHCP server domain.

The unique DHCP conﬁguration is required to allow the DHCP server to respond with a unique Option 66 parameter to the IP Phone 1140E with SIP Firmware.

Note: The DHCP standard deﬁnes Option 66 as the bootp server address in a string. The meaning of the bootp server address is extended in Nortel IP Phone 1140E with SIP Firmware to include the provisioning server address. The string in the DHCP offer for Option 66 can be

• the numeric IP address or name of the TFTP server or the URI (if

FTP or HTTP protocol is used) of the provisioning server in the form of

SIP Firmware Release 2.0 for IP Phone 1140E Administration

For example: http://mydomain.com/SIP_phone

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If provisioning server authentication is required, the user credential must be embedded in the URI in the form of

<protocol>://[<userid>:<password>@]<provisioning server URL>[:port][/path]

For example, ftp://www.mydomain.com/ABC or ftp://myuserid:mypass@ftp.mydo-

main.com:21/ABC

Requested Device Settings parameters

Using Full DHCP mode, a SIP IP Phone-aware DHCP server can automatically conﬁgure Nortel SIP IP Phones by requesting a list of conﬁguration parameters. The IP Phone 1140E uses DHCP to request and receive the information.

The following table lists the Device Settings parameters requested by the IP Phone 1140E in the Parameter Request List option (Option Code 55) in the DHCPDISCOVER and DHCPREQUEST messages. The DHCPOFFER and the DHCPACK reply messages from the DHCP server must contain the options in the following table.

Table 10 DHCP options in Full DHCP mode

Parameter requested by the

IP Phone 1140E

Subnet mask This is the subnet mask of the

IP Phone 1140E

Router and gateways IP address of the default

gateway of the IP Phone 1140E.

DNS server DNS Server address; only the

first one from the list is used.

Broadcast address This is the Broadcast address

of the subnet. The IP Phone 1140E automatically calculates the Broadcast address if it is not provided.

DNS domain Implementation varies

according to DHCP server.

Lease time Implementation varies

according to DHCP server.

Renewal time Implementation varies

according to DHCP server.

Description

DHCP server option

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Full DHCP mode 73

Rebinding interval Implementation varies

according to DHCP server.

Provisioning server Used for delivering the

provisioning server IP address. This parameter can contain either IP address or a URL to the provisioning server orfolder.

DHCP VLAN Auto Discovery

Conﬁguring a server for Voice VLAN Discovery is optional. This conﬁguration is done in addition to any conﬁguration done for Full DHCP or Partial DHCP and it is required only when you are conﬁguring the VLAN Auto Discovery in the Device Settings menu on the IP Phone 1140E.

Auto Voice VLAN Discovery is only possible in Full and Partial DHCP mode. VLAN Auto Discovery conﬁguration is useful in a network with separate

Data VLAN for trafﬁc (commonly used for PC-to-PC communication) and Voice VLAN for VoIP trafﬁc with different priorities.

The VLAN Auto Discovery is a two-step process:

1. DISCOVER is sent to the DHCP server for available Voice VLAN IDs. The DHCP server sends an OFFER with the available Voice VLAN IDs. If the Data VLAN ID has been manually provisioned in the Device Settings of the IP Phone 1140E, DHCP DISCOVER is tagged with the Data VLAN ID; otherwise, it is untagged.

2. DISCOVER is sent to the DHCP server for all of the DHCP required parameters. However, this DISCOVER is tagged with the Voice VLAN obtained in step 1.

DHCP VLAN Auto Discovery requires a Nortel-SIP-Phone-aware DHCP server. All DHCP requests carry the Vendor Class Identiﬁer, Nortel-SIP-Phone-A, to allow the DHCP server to identify that the requests are coming from an IP Phone 1140E with SIP Firmware.

DHCP Auto Discovery returns Voice VLAN IDs. The DHCP protocol provides no standard option for VLAN ID requests. Separate DHCP vendor-speciﬁc entry is needed for DHCP VLAN Auto Discovery to convey the VLAN information to the IP Phone. DHCP VLAN Auto Discovery uses one of the reserved for site-speciﬁc use DHCP options for VLAN list retrieval. At least one of the following Nortel site-speciﬁc options must be returned by the DHCP server as part of each DHCPOFFER and DHCPACK message for the IP Phone to accept these messages as valid; 43, 128, 131, 144, 157, 188, 191, 205, 219, 223, 232, 247, 251.

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74 Conﬁgure the DHCP server

After multiple VLAN IDs are returned from the DHCP server, the IP Phone 1140E tries to connect to each of the VLANs, following the order in which VLAN IDs are speciﬁed in the DHCP option.

The format of the ﬁeld for DHCP VLAN Auto Discovery is: Type, length, and data, described in the following sections.

Type (1 octet)

To avoid the possibility of option types already being used by different vendors, there are fourteen options types supported by the telephone. They are: 128, 131, 144, 157, 188, 191, 205, 219, 223, 232, 247, 251, 247, and

251. Select one value from the type byte list for the DHCPOFFER response.

Nortel recommends using 232, 247, or 251. DHCP option numbers less than 224 are reclaimed by the IETF (RFC 3942). Future changes in the DHCP protocol can force the telephone to stop sending these option requests. Currently, IP Phone 1140E with SIP Firmware does support the remaining listed options to maintain backward compatibility.

Length (1 octet)

The Length value is variable. Count only the number of octets in the data ﬁeld.

Data (variable number of octets)

ASCII based format: VLAN-A:XXX+YYY+ZZZ. where, VLAN– A: uniquely identiﬁes this as the Nortel DHCP VLAN discovery request. Each VLAN ID is followed by a plus (+) sign if there are more VLAN IDs or a period (.) to terminate the string.

There are a maximum of 10 VLAN IDs that can be conﬁgured in the current version.

Once IP Phone 1140E with SIP Firmware receives the DHCP offer containing the site-speciﬁc Voice VLAN option, the next DHCPDISCOVERY message is tagged with the Voice VLAN ID.

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Physically install the IP Phone 1140E

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

The steps for installing the IP Phone 1140E are summarized in the following procedure.

Procedure 4 Installing the IP Phone 1140E

Step Action

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 1140E. 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 1140E, using the RJ-9 handset jack. Form a small bend in the cable, and then thread the handset cord through the channels in the stand so that it exits behind the handset on the right side, in the handset cord exit in the stand base.

Install the headset (optional). If installing a headset, plug the connector into the RJ-9 headset jack on the back of the IP Phone 1140E, and thread the headset cord along with the handset cord through the channels in the stand, so that the headset cord exits the channel.

Install the Ethernet cable. Connect one end of the supplied Ethernet cable to the back of the IP Phone 1140E using the RJ-45 connector and thread the network cable through the channel.

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Install the Ethernet cable connecting the PC to the IP Phone 1140E (optional). If connecting PC Ethernet through the IP Phone 1140E, connect one end of the PC Ethernet cable to the IP Phone 1140E using the RJ-45 connector and thread it through the channel. Connect the other end to the LAN connector on the back of the PC.

Install additional cables. If applicable, plug in optional USB devices. Connect the Ethernet cable to the LAN Ethernet connection. If using an AC power adapter, plug the adapter into an AC outlet.

Wall-mount the IP Phone 1140E (optional). The IP Phone 1140E can be mounted either by: (method A) using the mounting holes on the bottom of the IP Phone 1140E stand, or (method B) using a traditional-style wall-mount box with RJ-45 connector and 15-cm (6-inch) RJ-45 cord (not provided).

Replace the stand cover. Ensure that all cables are neatly routed and press the stand cover into place until a click is heard.

Put the IP Phone 1140E in the wall-mount position (optional). If the IP Phone 1140E is to be mounted on the wall, put it in the wall-mount position by holding the tilt lever and pressing the IP Phone 1140E towards the base until the IP Phone 1140E is parallel with the base. Release the tilt lever and continue to push the IP Phone 1140E towards the base until an audible click is heard. Ensure the IP Phone 1140E is securely locked in position.

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—End—

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The following ﬁgure shows the connections on the IP Phone 1140E.

Figure 11 IP Phone 1140E connections

Physically install the IP Phone 1140E 77

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Upgrade and convert the IP Phone 1140E ﬁrmware

Introduction

This chapter describes how to upgrade an IP Phone 1140E with UNIStim ﬁrmware to SIP ﬁrmware.

In order to upgrade an IP Phone 1140E with UNIStim ﬁrmware, ﬁrst determine if you have the minimum UNIStim ﬁrmware release on the IP Phone (0625C39). If your IP Phone 1140E is installed with the minimum version of UNIStim ﬁrmware, proceed to the section "Convert UNIStim

Firmware to SIP Firmware on the IP Phone 1140E" (page 87). If your IP

Phone 1140E is not installed with the minimum version of UNIStim ﬁrmware, proceed to the section "Upgrade to the minimum UNIStim Firmware" (page

81).

To convert the ﬁrmware on the IP Phone 1140E from SIP to UNIStim, see the section "Maintenance and troubleshooting" (page 149).

Upgrade the SIP Firmware on the IP Phone 1140E

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

Download the SIP Firmware to the provisioning server

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

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

Step Action 1

SIP Firmware Release 2.0 for IP Phone 1140E 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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80 Upgrade and convert the IP Phone 1140E ﬁrmware

3 4

Enter IP Phone 1140E in the Knowledge and Solution Engine box. Select Software in the All types scroll down menu.

5 Press the gray arrow at the end of the Knowledge and Solution

Engine box to obtain the Search Results.

From the Search Results, select the appropriate version of the SIP Firmware for the IP Phone 1140E, for example, SIP IP Phone 1140E

Release 0625C39D26.bin.

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 06A5C39d26).

Under the ﬁrmware [FW] section of the SIP Provisioning ﬁle, modify the FILENAME of the new ﬁle you want to upload to the IP Phone 1140E.

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

the IP Phone 1140E with SIP Firmware.

1. Power off and power on the IP Phone 1140E.

2. Select Services, System, Check For Updates on the IP Phone 1140E.

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

42)).

Any of these actions causes the IP Phone 1140E to contact the provisioning server and attempt to read the Provisioning ﬁle. A Soft Reset (Srvcs, System, Reset Phone) does not cause the IP Phone 1140E to retrieve the Provisioning ﬁle and hence does not cause a ﬁrmware upgrade.

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Upgrade to the minimum UNIStim Firmware 81

—End—

Upgrade to the minimum UNIStim Firmware

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

Identify the current version of UNIStim Firmware

For a new IP Phone 1140E, follow Procedure 7 "Checking the UNIStim

Firmware version on a new IP Phone 1140E" (page 81) to determine the

version number of the UNIStim Firmware on an IP Phone 1140E. For an IP Phone 1140E already in use, follow Procedure 8 "Checking the

UNIStim Firmware version on an IP Phone 1140E already in use" (page

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

1140E.

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

Step Action 1

After assembling the IP Phone 1140E and turning it on, the display on the IP Phone 1140E goes through the following sequence:

•

Nortel splash screen appears

•

Nortel sonic sound plays

•

Nortel banner appears

Following the Nortel banner, the ﬁrmware version appears in the display (F/W version).

Note the UNIStim Firmware version number and write it down. Compare the version number to the minimum-required UNIStim Firmware version (0625C39).

UNIStim Firmware version names contain numbers and letters. Use the last three characters in a version to compare the version of UNIStim on an IP Phone 1140E (0625C39) with the minimum required version for the upgrade. Note that C23 is greater than C39 and C1B is less than C39.

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If the version number is equal to or higher than 0625C39, go to the section "Convert UNIStim Firmware to SIP Firmware on the IP

Phone 1140E" (page 87).

If the number is lower than 0625C39, go to the section "Upgrade

UNIStim Firmware to the minimum required UNIStim Firmware" (page 84) and follow the instructions to upgrade an IP Phone 1140E

to the minimum-required version of UNIStim Firmware before a conversion to SIP Firmware.

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

Step Action

—End—

Press the Globe/Services key on the IP Phone 1140E twice quickly.

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

The Local Tools menu appears:

Table 11 Local Tools menu

1. Preferences

2. Local Diagnostics

3. Network Configuration

4. Lock Menu

To make a selection, press the number associated with the menu item, or use the Navigation key cluster.

SIP Firmware Release 2.0 for IP Phone 1140E Administration

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

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Table 12 Using the Navigation key cluster to navigate in the Local Tools menu

Key Action

Down Moves highlight down

Up Moves highlight up

Right Selected current menu item

Left Closes menu

Select key (center of cluster)

To close this menu, use the Quit key.

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 (0625C39). UNIStim Firmware version names contain numbers and letters.

Use the last three characters in a version to compare the version of UNIStim on an IP Phone 1140E (0625C39) with the minimum required version for the upgrade. Note that C23 is greater than C39 and C1B is less than C39.

Selects current menu item

If the version number is equal to or higher than 0625C39, go to the section

"Convert UNIStim Firmware to SIP Firmware on the IP Phone 1140E" (page 87).

SIP Firmware Release 2.0 for IP Phone 1140E Administration

—End—

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84 Upgrade and convert the IP Phone 1140E ﬁrmware

If the number is lower than 0625C39, go to the section "Upgrade UNIStim

Firmware to the minimum required UNIStim Firmware" (page 84) and follow

the instructions to upgrade an IP Phone 1140E to the minimum-required version of UNIStim Firmware before you convert to SIP Firmware.

Upgrade UNIStim Firmware to the minimum required UNIStim Firmware

Use either of the following two methods to upgrade UNIStim Firmware.

1. UFTP download initiated by the server if the server supports this method of upgrading UNIStim Firmware. Refer to the appropriate documentation for your call server for instructions on using this method.

2. TFTP download on bootup.

If necessary, use the following procedure to conﬁgure the TFTP server.

Procedure 9 Conﬁguring the TFTP server

Step Action 1

The IP Phone 1140E always executes the TFTP download at bootup if a TFTP IP address is conﬁgured on the IP Phone 1140E after being initiated by the telephony call server.

Go to the TFTP server and create the 1140e.cfg provisioning ﬁle. The 1140e.cfg provisioning ﬁle is a clear text ﬁle. Create the provisioning ﬁle as shown in the next table.

Table 13 Sample 1140e.cfg provisioning ﬁle

[FW] DOWNLOAD_MODE FORCED VERSION 0625C23 FILENAME 0625C23.bin

This conﬁguration ﬁle forces the ﬁrmware download of 0625C23.bin.

3 Download and copy the ﬁrmware to the TFTP server directory.

To download the UNIStim Firmware for the IP Phone 1140E from the Nortel Web site:

SIP Firmware Release 2.0 for IP Phone 1140E Administration

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 85

The Technical Support page appears.

3. Enter IP Phone 1140E in the Knowledge and Solution Engine box.

4. Select SOFTWARE in the ALL TYPES scroll down menu.

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

6. From the Search Results, select the appropriate version of the UNIStim Firmware for the IP Phone 1140E, for example, IP

Phone 1140E Release 0625C23

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

In the IP Phone 1140E Network Conﬁguration menu, change the TFTP server address and enter the correct TFTP server address.

This can be the provisioning server as deﬁned in the chapter

"Conﬁgure the provisioning server" (page 29).

Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1140E.

The IP Phone 1140E downloads the ﬁrmware ﬁle. The display shows [FW] reading…

If the download is successful, the display shows [FW] writing… and the blue LED ﬂashes.

After the ﬁrmware image is downloaded to the IP Phone 1140E, the display shows [FW] ﬁnished..., the blue LED stops ﬂashing, and the IP Phone 1140E resets.

The IP Phone 1140E registers to the TPS with the new ﬁrmware version.

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

troubleshooting" (page 149) in the section Download failures.

—End—

Follow the next procedure to download the minimum required version of UNIStim Firmware automatically through TFTP on bootup.

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Procedure 10 Downloading UNIStim Firmware automatically through TFTP on bootup

Step Action 1

Press the Globe/Services key on the IP Phone 1140E twice quickly. If the admin password prompt appears, enter the password

26567*738

The Local Tools menu appears:

Table 14 Local Tools menu

1. Preferences

2. Local Diagnostics

3. Device Settings

4. Lock Menu

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

3 If you are using DHCP, select Full or Partial. If you are manually

conﬁguring the IP address, netmask, and gateway address, select No.

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

SIP Firmware Release 2.0 for IP Phone 1140E Administration

Conﬁgure the TFTP IP address within the IP Phone 1140E Device Settings menu.

This can be the provisioning server as deﬁned in the chapter

"Conﬁgure the provisioning server" (page 29).

Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1140E.

The IP Phone 1140E downloads the ﬁrmware ﬁle. The display shows [FW] reading…

If the download is successful, the display shows [FW] writing… and the blue LED ﬂashes.

After the ﬁrmware image is downloaded to the IP Phone 1140E, the display shows [FW] ﬁnished... the blue LED stops ﬂashing, and the IP Phone 1140E resets.

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Convert UNIStim Firmware to SIP Firmware on the IP Phone 1140E 87

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

troubleshooting" (page 149) in the section Download failures.

—End—

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

The IP Phone 1140E can be ordered with UNIStim Firmware installed or with SIP Firmware installed. If an IP Phone 1140E is installed with UNIStim Firmware, it runs with SIP Firmware only if the ﬁrmware is converted from UNIStim to SIP. If the procedure to determine the UNIStim version number is complete, and, if necessary, the procedure to upgrade the UNIStim Firmware is complete, an IP Phone 1140E can be converted from UNIStim Firmware to SIP Firmware.

Compare the version number to the minimum required UNIStim Firmware version (0625C39).

The conversion must be performed using TFTP.

WARNING

The TFTP download and upgrade of the Flash memory on the IP Phone 1140E may take a signiﬁcant amount of time (possibly up to 10 minutes). Do not unplug or reboot the IP Phone 1140E during the process.

The next procedure explains how to download the SIP Firmware from the Nortel Web site.

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

Step Action 1

Go to w

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

Password. The Technical Support page appears.

Enter IP Phone 1140E in the Knowledge and Solution Engine box.

ww.nortel.com/support.

SIP Firmware Release 2.0 for IP Phone 1140E Administration

Select SOFTWARE in the ALL TYPES scroll down menu.

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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 1140E, for example, IP Phone 1140E

Release 0625C39D26.bin.

Place the selected ﬁrmware on the provisioning server.

—End—

Follow the next procedure to convert the UNIStim Firmware to SIP Firmware on the IP Phone 1140E.

Procedure 12 Converting UNIStim Firmware to SIP Firmware using TFTP

Step Action 1

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

server (for example 1140e.img F/W ﬁle and 1140e.cfg ﬁle) that contains the following lines:

Table 15 Sample 1140e.cfg conﬁguration ﬁle

[FW] DOWNLOAD_MODE AUTO VERSION 06C25D26.bin FILENAME 1140e.img

Conﬁgure the IP Phone 1140E Device Settings TFTP IP address to the IP address where your TFTP server is running.

After you are ﬁnished the conﬁguration, the IP Phone reboots and sends a request to the TFTP server.

Select the Apply&Reset Context-sensitive soft key to save the settings and reset the IP Phone 1140E.

The following messages display on the IP Phone 1140E as the IP Phone 1140E cycles through the conversion process, one after the other:

1. [FW] Reading...

2. [FW] Writing...

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Convert UNIStim Firmware to SIP Firmware on the IP Phone 1140E 89

3. [FW] Finished...

The IP Phone 1140E then boots up with SIP Firmware. If the conversion is unsuccessful, see the chapter "Maintenance

and troubleshooting" (page 149).

1. TFTP file transfer takes approximately 15 seconds.

2. File writing takes 2.5 minutes. The IP Phone 1140E displays the message [FW] 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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Voice Quality Monitoring

Feature overview

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

item.

VQMon set-up

Conﬁgure the following parameters on the IP Phone 1140E with SIP Firmware to connect to the server and send the PVQMon statistics.

1. Enable the feature. To enable the feature, conﬁgure the VQMON_PUBLISH parameter in the device conﬁguration ﬁle (see

"VQMon conﬁguration commands" (page 57)).

2. Conﬁgure the IP address of the PVQMon server. Conﬁgure the IP address of the PVQMon server in either of the following settings:

a. Set VQMON_PUBLISH_IP through the device conﬁguration ﬁle (see

b. Set PVQMon IP in Device Settings (see Table 51 "PVQMon IP

3. Conﬁgure the remainder of the VQMon parameters in the device conﬁguration ﬁle (see "VQMon conﬁguration commands" (page 57)). These parameters provide threshold information to the IP Phone 1140E with SIP Firmware. A report is sent to the server when these thresholds are exceeded.

SIP Firmware Release 2.0 for IP Phone 1140E Administration

"VQMon conﬁguration commands" (page 57)).

conﬁguration" (page 121))

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92 Voice Quality Monitoring

Server set-up

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

1.0.

How VQMon works

The IP Phone 1140E with SIP Firmware gathers statistics about the current call when VQMon is enabled. Statistics are also gathered regarding the quality metrics of the current call. The call-related statistics contain condensed information about the SIP Session Description Protocol (SDP), the Call ID, the local and remote address, voice quality-related statistics, Zulu times for start-time and the time the report was sent.

The voice quality-related statistics include jitter, packet loss, delay, burst gap loss, listening R-factor, R-LQ, R-CQ, MOS-LQ and MOS-CQ. See Table

16 "Glossary of RTCP XR metrics" (page 92). More information on each

of these metrics is provided in RFC3611 “RTP Control Protocol Extended Reports (RTCP XR)”.

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

When the IP Phone 1140E detects that a particular voice quality metric has exceeded a threshold (deﬁned in the device conﬁguration ﬁle), the IP Phone sends a message to the server indicating that there is an issue. If the issue persists then the IP Phone reports another message indicating that there is an exceeded value at regular intervals. This happens continuously until the voice quality metric falls below the threshold value. As well, the IP Phone can send regular reports of the voice quality at time intervals deﬁned in the device conﬁguration ﬁle.

Table 16 Glossary of RTCP XR metrics

Burst A period of high packet losses and / or discards.

A burst is calculated in milliseconds.

Conversational R-factor Voice quality metric based on burst packet loss

and vocoder selection.

Delay One way delay which includes end-to-end delay,

jitter buffer delay and packetization delay. Delay is calculated in milliseconds.

Inter-arrival jitter The variation in packet arrival times due to

transmission (routing, queuing delay) through the network. Jitter is calculated in milliseconds.

Listening R-factor Voice quality metric based on burst packet loss,

transmission delay and burst loss.

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How VQMon works 93

MIU Media Information Unit. MIU is a concept from

VQMon. An MIU can be any size down to a 10 millisecond (8 sample) block. An MIU means a frame in the i200x implementation.

MOS Mean Opinion Score. A subjective measurement

of the voice quality of a voice call.

MOS_CQ The VQMon conversational quality MOS score

calculated for a call channel.

MOS_LQ The VQMon listening quality MOS score

calculated for a call channel.

Packet loss rate The percentage of total packets loss versus

packets received.

R-factor A measurement of voice quality based on

network impairments including burst packet loss, delay and encoding/decoding algorithm selection.

End of call report

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

Session interval report

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

Alert interval report

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

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94 Voice Quality Monitoring

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Device Settings on the IP Phone 1140E with SIP Firmware

ATTENTION

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

Introduction

This chapter details how to conﬁgure the Device Settings parameters on the IP Phone 1140E with SIP Firmware. This includes items such as the IP address, the subnet mask, and the gateway IP address of the IP Phone 1140E.

The Device Settings parameters are listed in the following table in the order they appear on the Device Settings menu of the IP Phone 1140E with SIP Firmware. Read the section at the end of this chapter that explains how to provision the Device Settings parameters. If you are familiar with the Device Settings parameters, skip the last section in the chapter and proceed to the provisioning instructions.

Table 17 Parameters in Device Settings menu

Enable 802.1x (EAP) Device ID Password Enable 802.1ab (LLDP) DHCP

SET IP Net Mask

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No Partial Full

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96 Device Settings on the IP Phone 1140E with SIP Firmware

Gateway DNS IP1 DNS IP2 Ntwk Port Speed

Auto 10BT 100BT

Ntwk Port Duplex

Auto Force Full

Force Half Disable Voice 802.1Q Voice VLAN

No VLAN

DHCP

LLDP VLAN Name

LLDP MED

Manual VLAN Filter Ctrl Priority bits

Auto

0-7 Media Priority bits

Auto

0-7 Disable PC Port PC Port Speed

Auto

10BT

100BT PC Port Duplex

Auto

Disable Data 802.1Q

SIP Firmware Release 2.0 for IP Phone 1140E Administration

Force Full

Force Half

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Data VLAN

No VLAN

LLDP VLAN NAME

Manual (value

from 1 to 4094) Data Priority bits

Auto

0-7 PC-Port Untag all Cached IP Ignore GARP Provisioning

Server URL

Protocol

Introduction 97

TFTP FTP

PVQMon IP NAT Traversal

HTTP Device ID Password

NAT Signal

None

SIP Ping

STUN NAT Media

None

STUN NAT TTL STUN S1 IP STUN S2 IP

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98 Device Settings on the IP Phone 1140E with SIP Firmware

802.1x (EAP) Port-based network access control

Extensible Authentication Protocol (EAP) supports multiple authentication methods and represents a technology framework that facilitates the adoption of Authentication, Authorization, and Accounting (AAA) schemes, such as Remote Authentication Dial In User Service (RADIUS). RADIUS is deﬁned in RFC 2865. The IP Phone 1140E with SIP Firmware supports only the MD5 authentication method.

802.1x deﬁnes the following three roles:

1. Supplicant—an IP Phone that requires access to the network to use network services.

2. Authenticator—the network entry point to which the supplicant physically connects (typically a Layer 2/3 switch). The authenticator acts as the proxy between the supplicant and the authentication server. The authenticator controls access to the network based on the authentication status of the supplicant.

3. Authentication server—performs authentication of the supplicant.

Enable and disable Network-level authentication through the EAP conﬁguration menu.

Authorization

If 802.1x is conﬁgured and the IP Phone 1140E is physically connected to the network, the IP Phone 1140E (supplicant) initiates 802.1x authentication by contacting the Layer 2/3 switch (authenticator). The IP Phone 1140E also initiates 802.1x authentication after the Ethernet connection (network interface only) is restored following a network link failure.

However, if the IP Phone 1140E resets, it assumes the Layer 2 link has remained in service and is authenticated.

The IP Phone 1140E fails to authorize if the DeviceID and the IP Phone 1140E passwords do not match the DeviceID and IP Phone 1140E passwords provisioned on the RADIUS Server. The Layer 2 switch (authenticator) locks out the IP Phone 1140E and network access is denied. If this happens during reauthorization, all telephone services are lost. The connected PC operates as normal.

Device ID

The Device ID is for use with the 802.1x (EAP) protocol. If the 802.1x (EAP) is not used, then there is no prompt to enter the Device ID.

Password

The Password is for use with the 802.1x (EAP) protocol. If the 802.1x (EAP) is not used, there is no prompt to enter the Password.

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Introduction 99

802.1ab Link Layer Discovery Protocol

802.1ab Link Layer Discovery Protocol (LLDP) is a standard for discovering

the physical topology between neighboring devices. 802.1ab LLDP deﬁnes a standard method for Ethernet network devices, such as switches, routers, and IP Phones to advertise information about themselves to other nodes on the network and to store the information they discover in a Management Information Base (MIB).

802.1ab (LLDP) takes advantage of the VLAN Name and Network Policy

TLVs, and provides an automatic conﬁguration of the phone network policy parameters. Key parameters, such as VLAN ID, L2 priority, and DSCP values are received from the switch and are automatically conﬁgured in the phone.

802.1ab Link Layer Discovery Protocol (LLDP) provides the following

functionality

•

Periodic transmission of advertisements containing device information, device capabilities and media speciﬁc conﬁguration information to neighbors attached to the same network.

•

Reception of LLDP advertisements from its neighbors.

•

Implementation of behavioral requirements speciﬁed by Link Layer Discovery Protocol Media Endpoint Discovery (LLDP-MED).

•

Storage of received data in local data structures, for example, in MIB modules.

TLVs

The information ﬁelds in each MIB are contained in a Link Layer Discovery Protocol Data Unit (LLDPDU) as a sequence of short, variable-length, information elements known as TLVs that each include type, length, and value ﬁelds. Each LLDPDU includes several mandatory TLVs plus optional TLVs. Optional TLVs may be inserted in any order.

The IP Phone supports both the transmit and receive LLDP mode.

Transmit direction

An LLDPDU transmitted by the IP Phone supports the following TLVs

1. Chassis ID

2. Port ID

3. Time To Live

4. End of LLDPPDU

5. Port Description

6. System Description

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100 Device Settings on the IP Phone 1140E with SIP Firmware

7. System Capabilities

8. Port VLAN ID

9. Port And Protocol VLAN ID

10. VLAN Name

11. Protocol Identity

12. MAC/PHY Conﬁguration Status

13. Power Via MDI

14. Link Aggregation

15. Maximum Frame Size

16. LLDP-MED Capabilities

17. Network Policy

18. Extended Power-via MDI

19. Inventory Firmware Revision

20. Inventory Manufacturer Name

21. Inventory Model Name

Receive direction

The IP Phone expects to receive the following TLVs

1. Chassis ID

2. Port ID

3. Time To Live

4. End of LLDPPDU

5. System Capabilities

6. VLAN Name

7. MAC/PHY Conﬁguration Status

8. LLDP-MED Capabilities

9. Network Policy

10. Location Identiﬁcation

Table 18 TLV formats

TLV Fields

Chassis ID Length = 6

Chassis Subtype = 5 [IP Address] Chassis ID = IP Phone IP Address

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

Specifications and Main Features

Frequently Asked Questions

User Manual

New in this release

Server Supplied User Credentials

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

IEEE 802.1ab Link Layer Discovery support

RFC 3581

Voice Quality Monitoring

Distinctive ringing and Call waiting

Shared Call Appearance

Transfer to Voice Mail

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 1140E with SIP Firmware

Figure 1 IP Phone 1140E with SIP Firmware

Related documentation

Installation overview

Procedure 1 Installation overview

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

Figure 3 Installation of IP Phone 1140E 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 IP Phone 1140E from the Nortel Web site

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

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 1140E

Procedure 4 Installing the IP Phone 1140E

Figure 11 IP Phone 1140E connections

Introduction

Upgrade the SIP Firmware on the IP Phone 1140E

Download the SIP Firmware to the provisioning server

Procedure 5 Downloading SIP Firmware for the IP Phone 1140E 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 1140E

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

Procedure 11 Downloading SIP Firmware for the IP Phone 1140E 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

End of call report

Session interval report

Alert interval report

Device Settings on the IP Phone 1140E with SIP Firmware

Introduction

802.1x (EAP) Port-based network access control