Avaya 9600 User Manual

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Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Administrator Guide Release 2.0

16-601944

Issue 2

December 2007

Page 2

Notice

While reasonable efforts were made to ensure that the infor mation in this document was complete and accurate at the time of printing, Avaya Inc. can assume no liability for any errors. Changes and corrections to the information in this document may be incorporated in future releases.

For full legal page information, please see the complete document, A vaya Legal Page for Hardware Documentation, Document number 03-600759.

To locate this document on our Web site, simply go to

http://www.avaya.com/support

the search box. Documentation disclaimer

Avaya Inc. is not responsible for any modifications, addition s, or deletions to the original published version of this documentation unless such modifications, additions, or deletions were performed by Avaya. Customer and/or End User agree to indemnify and hold harmless Avaya, Avaya's agents, servants and employees against all claims, lawsuits, demands and judgments arising out of, or in connection with, subsequent modifications, additions or deletions to this documentation to the extent made by the Customer or End User.

Link disclaimer

Avaya Inc. is not responsible for the contents or reliability of any linked Web sites referenced elsewhere within this documentation, and Avaya does not necessarily endorse the products, services, or informa tion described or o ff ered within them. We cannot guarantee that these links will work all of the time and we have no control over the availability of the linked pages.

Warranty

Avaya Inc. provides a limited warranty on this product. Refer to your sales agreement to establish the terms of the limited warran ty. In addition, Avaya’s standard warranty language, as well as information regarding support for this product, while under warranty, is available through the following Web site:

http://www.avaya.com/support

Except where expressly stated otherwise, the Product is protected by copyrigh t and other laws respecting proprietary rights. Unauthorized reproduction, transfer, and or use can be a criminal, as well as a civil, offense un der the applicable law.

Avaya support

Avaya provides a telephone number for you to use to report pro blems or t o ask questions about your product. The support telephone number is 1-800-242-2121 in the United States. For additional support telephone numbers, see the Avaya Web site:

http://www.avaya.com/support

Software License

USE OR INSTALLATION OF THE PRODUCT INDICATES THE END USER’S ACCEPTANCE OF THE TERMS SET FORTH HEREIN AND THE GENERAL LICENSE TERMS AVAIL ABLE ON T HE AVAYA WEBSITE AT

http://support.avaya.com/LicenseInfo/

YOU DO NOT WISH TO BE BOUND BY THESE TERMS, YOU MUST RETURN THE PRODUCT(S) TO THE POINT OF PURCHASE WITHIN TEN (10) DAYS OF DELIVERY FOR A REFUND OR CREDIT.

Avaya grants End User a license within the scope of the license types described below. The applicable number of licenses and units of capacity for which the license is granted will be one (1), unless a different number of licenses or units of capacity is specified in the Documentation or other materials available to End User. “Designated Processor” means a single stand-alone computing device. “Server” means a Designated Processor that hosts a software application to be accessed by multiple users. “Soft w are” means the computer programs in object code, originally licensed by Avaya and ultimately utilized by End User, whether as stand-alone Products or pre-installed on Hardware. “Hardware” means the standard hardware Products, originally sold by Avaya and ultimately utili zed by End User.

License Type(s):

Designated System(s) License (DS). End User may install and use each copy of the Software on only one Designated Processor, unless a different number of Designated Processors is indicated in the Documentation or other mat erials available to End User. Avaya may require the Designated Processor(s) to be identified by type, serial number, feature key, location or other specific designation, or to be provided by End User to Avaya through elect roni c mean s established by Avaya specifically for this purpose.

and search for the document number in

(“GENERAL LICENSE TERMS”). IF

Third-party Components

Certain software programs or portions thereof included in the Product may contain software distributed under third party agreements (“Third Party Components”), which may contain terms that expand or limit rights to use certain portions of the Product (“Third Party Terms”). Information identifying Third Party Components and the Third Party Terms that apply to them is available on Avaya’s Web site at:

http://support.avaya.com/ThirdPartyLicense/

Interference

Using a cell, mobile, or GSM telephone, or a two-way radio in close proximity to an Avaya IP Telephone might cause interference.

Security

See http://support.avaya.com/security vulnerabilities in Avaya products. See http://support.avaya.com latest software patches and upgrades. For information about secure configuration of equipment and mitigation of toll fraud threats, see the Avaya Toll Fraud and Security Handbook at http://support.avaya.com

to locate and/or report known

to locate the

Page 3

Chapter 1: Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

About This Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7

Major Differences Between 9600 Series SIP IP and 9600 Series H.323 IP Telephones 8

Features & Functions Supported by H.323 and Not Supported by SIP: . . . . 9

Change History . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

What’s New in SIP Software Release 2.0. . . . . . . . . . . . . . . . . . . . . . . 10

Document Organization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

Other Documentation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

Chapter 2: Administration Overview and Requirements . . . . . . . . . 15

9600 Series IP Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15

Parameter Data Precedence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18

The Administrative Process. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

Administrative Checklist . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19

Telephone Initialization Process . . . . . . . . . . . . . . . . . . . . . . . . . . . 21

Step 1: Telephone to Network . . . . . . . . . . . . . . . . . . . . . . . . . . 21

Step 2: Telephone to LLDP-Enabled Network . . . . . . . . . . . . . . . . . . 21

Step 3: Telephone to DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . 22

Step 4: Telephone and File Server . . . . . . . . . . . . . . . . . . . . . . . . 22

Step 5: Telephone and the SES Server. . . . . . . . . . . . . . . . . . . . . . 22

Error Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23

Chapter 3: Network Requirements . . . . . . . . . . . . . . . . . . . . . 25

Network Assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

Hardware Requirements. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

Server Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26

HTTP/HTTPS Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Network Time Protocol (NTP) Server. . . . . . . . . . . . . . . . . . . . . . . 27

Required Network Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Other Network Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

SNMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

Registration and Authentication . . . . . . . . . . . . . . . . . . . . . . . . . 28

Reliability and Performance. . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

IEEE 802.1D and 802.1Q. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

Network Audio Quality Display on 9600 Series SIP IP Telephones . . . . . . 30

SIP Station Number Portability . . . . . . . . . . . . . . . . . . . . . . . . . . 30

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Contents

TCP/UDP Port Utilization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31

Security. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

Registration and Authentication . . . . . . . . . . . . . . . . . . . . . . . 35

Chapter 4: Communication Manager Administration . . . . . . . . . . . 37

Call Server Requirements. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

Switch Compatibility. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37

Communication Manager Administrative Requirements . . . . . . . . . . . . . . 37

System-Level Preparation Tasks . . . . . . . . . . . . . . . . . . . . . . . . . 38

SIP Trunk Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Call Routing Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

IP Interface and Addresses . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

UDP Port Selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39

RSVP and RTCP/SRTCP. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

IEEE 802.1D and 802.1Q. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

NAT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

DIFFSERV . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40

Voice Mail Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41

Auto Hold. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41

Call Transfer Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41

Conferencing Call Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . 42

Telephone Administration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42

CM/SIP IP Telephone Configuration Requirements . . . . . . . . . . . . . . . . . 44

Administering Stations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48

Administering Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49

Chapter 5: SIP Enablement Services (SES) Administration . . . . . . . 51

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51

Using the Web Browser to Configure SES. . . . . . . . . . . . . . . . . . . . . . 51

Chapter 6: Server Administration . . . . . . . . . . . . . . . . . . . . . 53

Software Checklist. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53

DHCP and File Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53

DHCP Server Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54

Configuring DHCP for 9600 Series SIP IP Telephones . . . . . . . . . . . . . 54

DHCP Generic Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56

4 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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Windows NT 4.0 DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . . . 59

Verifying the Installation of the DHCP Server . . . . . . . . . . . . . . . . 59

Creating a DHCP Scope for the IP Telephones . . . . . . . . . . . . . . . 60

Editing Custom Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

Adding the DHCP Option . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

Activating the Leases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62

Verifying Your Configuration . . . . . . . . . . . . . . . . . . . . . . . . . 62

Windows 2000 DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . . . . 63

Verifying the Installation of the DHCP Server . . . . . . . . . . . . . . . . 63

Adding DHCP Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Activating the New Scope. . . . . . . . . . . . . . . . . . . . . . . . . . . 65

HTTP Generic Setup. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66

Chapter 7: Telephone Software and Binary Files . . . . . . . . . . . . . 67

General Download Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67

Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67

9600 Series SIP IP Telephone Scripts and Binary Files. . . . . . . . . . . . . . . 68

Choosing the Right Binary File and Upgrade Script File . . . . . . . . . . . . 68

Upgrade Script File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69

Settings File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69

Contents of the Settings File . . . . . . . . . . . . . . . . . . . . . . . . . . . 70

The GROUP System Value . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72

Contents

Chapter 8: Administering Telephone Options . . . . . . . . . . . . . . . 73

Administering Options for the 9600 Series SIP IP Telephones. . . . . . . . . . . 73

VLAN Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94

VLAN Tagging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94

VLAN Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95

VLAN Default Value and Priority Tagging . . . . . . . . . . . . . . . . . . . . 95

VLAN Separation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96

DNS Addressing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98

IEEE 802.1X . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98

802.1X Pass-Through and Proxy Logoff . . . . . . . . . . . . . . . . . . . . . 99

802.1X Supplicant Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . 99

Link Layer Discovery Protocol (LLDP) . . . . . . . . . . . . . . . . . . . . . . . . 101

Visiting User Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105

Emergency Number Administration . . . . . . . . . . . . . . . . . . . . . . . . . 106

Local Administrative (Craft) Options Using the Telephone Dialpad . . . . . . . . 107

Language Selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107

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Contents

Enhanced Local Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108

Enhanced Local Dialing Requirements . . . . . . . . . . . . . . . . . . . . . 110

Chapter 9: Administering Applications and Options . . . . . . . . . . . 111

Customizing Telephone Applications and Options . . . . . . . . . . . . . . . . . 111

Avaya “A” Menu Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . 112

Administering Standard Avaya Menu Entries . . . . . . . . . . . . . . . . . . 112

Administering the WML Browser . . . . . . . . . . . . . . . . . . . . . . . . . 112

Appendix A: Glossary of Terms . . . . . . . . . . . . . . . . . . . . . . 115

Appendix B: Related Documentation . . . . . . . . . . . . . . . . . . . 119

IETF Documents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

ITU Documents. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119

ISO/IEC, ANSI/IEEE Documents . . . . . . . . . . . . . . . . . . . . . . . . . 119

Appendix C: Sample Station Forms . . . . . . . . . . . . . . . . . . . . 121

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135

6 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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Chapter 1: Introduction

About This Guide

This guide is for personnel who administer Avaya Communication Manager, DHCP, HTTP/ HTTPS servers for 9600 Series SIP IP Telephones, a Local Area Network (LAN), SIP Enablement Services (SES) or a Network Time server.

The 9600 Series IP Telephones use Internet Protocol (IP) technology with Ethernet line interfaces and support both SIP and H.323 protocols. The 9600 Series IP Telephones provide support for DHCP, HTTP, and HTTPS over IPv4/UDP, which enhance the administration and servicing of the telephones. These telephones use DHCP to ob t ain dynamic IP Add resses, and HTTPS or HTTP to download new versions of software or customized settings for the telephones.

Important:

Important: This document covers administration for 9600 Series SIP IP Telephones only . For

administration for 9600 Series IP Telephones using the H.323 protocol, see the

Avaya one-X™ Deskphone Edition for 9600 Series IP Telephones Administrator Guide (Document Number 16-300698), available at: www.avaya.com/support

This document does not cover administration for Avaya Distributed Office. Full documentation for Avaya Distributed Office is available on the Avaya support Web site, www.avaya.com/support

Avaya does not provide product support for many of the products mentioned in this document. Take care to ensure that there is adequate technical support available for servers used with any 9600 Series IP and/or SIP IP Telephone system. If the servers are not functioning correctly, the 9600 Series IP Telephones might not operate correctly.

Issue 2 December 2007 7

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Introduction

Major Differences Between 9600 Series SIP IP and 9600 Series H.323 IP Telephones

Review this section if your administrative environment includes both SIP and H.323 signaling protocols for 9600 Series IP Telephones.

General IP Telephony - Two major protocols handle Voice over IP (VoIP) signaling, Session Initiation Protocol (SIP) and H.323. The two protocols provide connection control and call progress signaling, but in very different ways. These protocols can be used simultaneously over the same network, but in general, no endpoint supports both p rotocols at the same time. Neither protocol is necessarily superior, but each offers some unique advantages. SIP telephones, for example, do not require centralized call servers, and can route telephone calls when a URL identifies the destination. H.323 telephones leverage the call server’s presence into the potential availability of hundreds of telephone-related features that a standalone SIP telephone cannot provide.

Signaling - 96xx Series IP Telephones ship from the factory with H.323 signaling. To use the SIP protocol, applicable H.323 96xx Series IP Telephones must be appropriately converted and configured. See the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide for detailed conversion/configuration information.

Avaya Communication Manager Release - 9600 Series SIP IP Telephones are supported only by Communication Manager Release 4.0 and greater. SIP telephones use Avaya OPS (Outbound SIP Proxy) features on the “trunk” side of Avaya Communication Manager whereas the H.323 (IP) telephones are supported on the “line” side of the Communication Manager. When a 9600 Series SIP IP Telephone is running under Communication Manager Release 5.0, an additional feature, Extend Call, is available.

Required Servers - SIP 9600 Series IP Telephones use two [additional] servers that H.323 telephones do not:

- SIP Proxy server - provided by SIP Enablement Services (SES) software, and

- Network Time server - which controls time-related parameters.

These servers are not necessarily separate hardware units.

Features & Functions supported by H.323 9600 Series IP Telephones, Not SIP - Button modules are not currently supported by 9600 SIP IP Telephones.

Backup/Restore - 9600 Series (H.323) IP Telephones use HTTP to store backup files. 9600

Series SIP IP Telephones use the Personal Profile Manager (PPM) functionality within SIP Enablement Services (SES) for backup and restore functions.

Settings File & System Parameters - Both SIP and H.323 9600 Series IP Telephones (and 4600 Series IP Telephones) use the same settings file. Some of the same system parameters are used, however, numerous SIP-specific parameters support SIP operation only. In H.323 9600 Series IP Telephones, the parameters OPSTAT and APPSTAT control all user interface functions, whereas 9600 Series SIP IP Telephones use a separate parameter (for example ENABLE_CONTACTS, ENABLE_CALLLOG) for each user interface function.

Supported by

8 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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Change History

Language Support - SIP telephones support the same languages as H.323 telephones, with

the exception of Hebrew. SIP does not support Hebrew or the English Large Text Font for any language. Further, all SIP language files have .xml file extensions whereas H.323 language files have .txt file extensions.

SNMP & MIBs - Although both SIP and H.323 telephones support SNMP v2c and have custom Management Information Bases (MIBs), the MIBs are formatted somewhat differently.

RSVP & VMON (VMM) - 9600 Series SIP IP Telephones do not use RSVP (Resource ReSerVation Protocol) software to provide real-time monitoring and historical data of audio quality for VoIP calls. 9600 Series SIP IP Telephones do support Avaya Voice over IP (VoIP) Monitoring Manager (VMON, now called VMM). 9600 Series IP T elephones use both RSVP and VMON.

QoS - Unlike H.323 telephones, 9600 Series SIP IP Telephones do not use Avaya Communication Manager to set Quality of Service (QoS). The SIP IP telephones use the parameters L2QAUD, L2QSIG, DSCP AUD, and DSCPSIG (described in Table 11:

9600 Series

SIP IP Telephones Customizeable System Parameters).

NAT - 9600 Series SIP IP Telephones do not support Network Address T ranslation (NAT); 9600 Series IP (H.323) Telephones do support NAT.

Features & Functions Supported by H.323 and Not Supported by SIP:

SIP Software Release 2.0 SIP Software Release 1.0

● Calltype Digit Conversion

● RSVP

● Remote Ping & Trace Route

● SBM24 Button Modules

● Push (Top Line, web page, and/or audio)

● Link Layer Discovery Protocol (LLDP)

● GigE (Gigabit Ethernet)

● Calltype Digit Conversion

● IEEE 802.1X

● RSVP

● VMON

● Remote Ping & Trace Route

● Web browser

● SBM24 Button Modules

● Push (Top Line, web page, and/or audio)

● Autodial feature buttons

Change History

Issue 1 This document was issued for the first time in May 2007 to support the first release of

9600 Series SIP IP Telephones.

Issue 2 This is the current version of the document, revised and issued in December, 2007 to

support SIP IP Software Release 2.0. This release provides the 9600 SIP IP Telephones with similar functionality to their H323 9600 IP Telephone counterparts, despite their signaling protocol differences. Release 2.0 introduces several new functions, new configuration parameters, and adds telephone models 9630G and 9640G. What’s New in

SIP Software Release 2.0 describes this release in more detail.

Issue 2 December 2007 9

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Introduction

What’s New in SIP Software Release 2.0

New material in this issue to support SIP Release 2.0 software includes: New GigE Models Support SIP - This release extends SIP capability to two additional

telephones, the 9630G and 9640G. Both models provide built-in Gigabyte Ethernet (GigE) support, but are otherwise identical to their 9630 and 9640 SIP IP telephone counterparts.

Language Support - 9600 Series SIP IP Telephones now support 13 languages. See

Language Selection

Emergency Button - Administrators can now program an “Emergency” number using the new

PHNEMERGNUM

logged into the telephone from which they are calling for assistance. For more information, see

Emergency Number Administration

Administration Enhancements - SIP Software Release 2.0 supports functionality introduced on Avaya Communication Manager Release 5.0 and SIP Enablement Services (SES) Release

5.0.

Visiting User Support - Visiting use r su pport allows users to e asily move be tween g eograp hic locations while retaining their telephone extension and settings. 9600 Series SIP IP Tele phones can be provisioned through the settings file VU_MODE modes:

on page 107 for more information.

parameter. Users can dial the Emergency Number whether or not they are

configuration parameter to one of three

● No Visiting User - the telephone operates “normally” and has no user interface impact for

normal operation. The telephone can be forced to a “registered Inactive” state when a visiting user registers elsewhere.

● Optional Visiting User - the telephone prompts the user at registration time if they are

visiting or not.

● Forced Visiting User - the telephone allows only visiting user registrations.

For more information, see Visiting User Administration

Link Layer Discovery Protocol (LLDP) - 9600 Series SIP IP Telephones now support link layer discovery protocol. See Link Layer Discovery Protocol (LLDP)

for information.

802.1X - 9600 Series SIP IP Telephones now support IEEE standard 802.1X for increased

security. The new configuration parameter DOT1X defines the 802.1X operational mode. The new parameter DOT1XSTAT enables/disables 802.1X. The new parameter DOT1XEAPS specifies the authentication method to use with 802.1X. These parameters can be set through the settings file or on a per-phone basis using a local Craft procedure.

Support for Non-Avaya (Third Party) Environments - Several parameters, most notably ENABLE_AVAYA_ENVIRONMENT, have been added to cover operation for either:

● an Avaya environment, which provisions SIP/AST features and uses Personal Profile

Manager (PPM) for download and backup/restore, or

● a non-Avaya mode, which complies with 3rd party standard SIP proxy with provision for

SIPPING 19 feature.

10 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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What’s New in SIP Software Release 2.0

WML Applications/Browser - 9600 Series SIP IP Telephones now provide access to WML

applications via a WML Browser, as described in Chapter 9:

Administering Applications and

Options.

New, Revised, and Deleted Configuration Parameters - The following configuration parameters have been added for this release and are linked to the table that describes them in detail:

● CALL_TRANSFER_ MODE

● CALLFWDADDR

● CALLFWDDELAY

● CALLFWDSTAT

● CNAPORT

● CNASRVR

● CONFIG_SERVER_ SECURE_MODE

● COVERAGEADDR

● DIALPLAN

● DOT1X

● DOT1XEAPS

● DOT1XSTAT

● ENABLE_AVAYA_ ENVIRONMENT

● INTER_DIGIT_TIMEOUT (replaces INTER_DIGIT_DIALING_TIMEOUT_DURATION)

● LAST_LOGIN_STATUS (system-set only)

● LLDP_ENABLED

● MWISRVR

● NO_DIGITS_TIMEOUT

● PHNEMERGNUM

● PHNNUMOFSA

● POE_CONS_SUPPORT

● PRESENCE_SERVER

● PROVIDE_EDITED_ DIALING

● PROVIDE_EXCHANGE_CALENDAR

● PROVIDE_EXCHANGE_CONTACTS

● QKLOGINSTAT

● RTCPCONT

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Introduction

● RTCPMON

● RTCPMONPORT

● SIP_MODE

● SIPCONFERENCECONTINUE

● TLSSRVRID

● VU_MODE

● VU_TIMER

● WMLEXCEPT

● WMLHOME

● WMLIDLETIME

● WMLIDLEURI

● WMLPORT

● WMLPROXY

The following parameters have been modified or renamed:

● Parameters PHYxDUPLEX and PHYxSPEED were combined. PHY1SPEED has been

renamed to PHY1_OPERATIONAL_MODE. This parameter now includes the current duplex mode. PHY2SPEED has been renamed to PHY2_OPERATIONAL_MODE. This parameter now includes the current duplex mode.

● The OUTBOUND_SUBSCRIPTION_REQUEST_DURATION default value has been

changed from 17280000 to 86400 seconds. This parameter can now be set through the settings file.

● The dimensions for SNTP_SYNC_INTERVAL and

SNTP_SYNC_RANDOMIZATION_INTERVAL have changed from seconds to minutes.

● EXCHANGE_CONTACTS_ENABLED has been renamed to

USE_EXCHANGE_CONTACTS.

● EXCHANGE_CALENDAR_ENABLED has been renamed to

USE_EXCHANGE_CALENDAR.

● The default value definition of ENABLE_G726 has changed from 0 to 1.

● The default values and sidetone definitions of the audio parameters AUDIOSTHD and

AUDIOSTHS

● WAIT_FOR_REGISTRATION_TIMER can now be set through the settings file.

have been modified.

The following configuration parameters are no longer valid and have been removed:

● PHY1DUPLEX

● PHY2DUPLEX

● INTER_DIGIT_DIALING_TIMEOUT_DURATION

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Document Organization

The guide contains the following sections:

Document Organization

Chapter 1: Chapter 2: Administration

Overview and Requirements Chapter 3:

Requirements Chapter 4: Communication

Manager Administration Chapter 5: SIP Enablement

Services (SES) Administration Chapter 6: Server Administration Describes DHCP and HTTP/HTTPS administration for the 9600

Chapter 7: and Binary Files

Chapter 8: Administering Telephone Options

Chapter 9: Applications and Options

Appendix A: Glossary of Terms Provides a glossary of terms used in this document or which can be

Introduction Provides an overview of this document.

Provides an overview of the administrative process and describes general hardware, software, and operational requirements.

Network

Telephone Software

Administering

Describes administrative requirements for your Local Area Network.

Describes how to administer Avaya Communication Manager to operate with 9600 Series SIP IP Telephones.

Covers SIP Enablement Services (SES) configuration for 9600 Series SIP IP Telephones.

Series IP Telephones. Describes telephone software, covers software downloads, and

provides information about the configuration file. Describes how to use file parameters and options to administer

9600 Series SIP IP Telephones. Covers backup and restoration of telephone data. Also describes how to use local procedures to customize a single telephone from the dialpad.

Describes customizeable application-specific parameters, to provide administrative control of telephone functions and options.

applicable to 9600 Series SIP IP Telephones.

Appendix B: Related Documentation

Appendix C: Sample Station Forms

Provides references to Web sites with external documents that relate to telephony in general, and can provide additional information about specific aspects of the telephones.

Provides examples of Avaya Communication Manager forms related to system wide and individual telephone administration.

9600 Series IP Telephones

The 9600 Series IP Telephones currently support the H.323 signaling protocol and the SIP signaling protocol.

The H.323 standard provides for real time audio, vide o, and data communications transmission over a packet network. An H.323 telephone protocol stack comprises several protocols:

● H.225 for registration, admission, status (RAS), and call signaling,

● H.245 for control signaling,

● Real Time Transfer Protocol (RTP) and Secure Real Time Transfer Protocol (SRTP)

● Real Time Control Protocol (RTCP) and Secure Real Time Control Protocol (SRTCP)

SIP was developed by the IETF. Like H.323, SIP provides for real time audio, video, and data communications transmission over a packet network. SIP uses various messages, or methods, to provide:

● Registration (REGISTER),

● Call signaling (INVITE, BYE)

● Control signaling (SUBSCRIBE, NOTIFY)

The 9600 Series SIP IP Telephones support Media Encryption (SRTP) and use built-in Avaya SIP Certificates for trust management. Trust management involves downloadin g certificates for additional trusted Certificate Authorities (CA) and the policy management of those CAs. Identity management is handled by Simple Certificate Enrollment Protocol (SCEP) with phone certificates and private keys.

The 9600 Series IP Telephones are loaded with either H.323 or SIP software as part of initial script file administration and initialization during installation. Post-installation, sof tware upgrades automatically download using the proper signaling protocol.

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Administration Overview and Requirements

The parameters under which the 9600 Series SIP IP Telephones need to operate are summarized as follows:

● Telephone Administration on the Communication Manager (CM) call server, as covered in

Chapter 4:

● Administration on SIP Enablement Services (SES), as covered in Chapter 5: SIP

Communication Manager Administration.

Enablement Services (SES) Administration.

● IP Address management for the telephone, as covered in Chapter 6: Server

Administration for dynamic addressing. For static addressing, see the Avaya one-X™

Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide.

● Tagging Control and VLAN administration for the telephone, if appropriate, as covered in

Chapter 8:

● Quality of Service (QoS) administration for the telephone, if appropriate. QoS is covered in

QoS

● Protocol administration, for example, Simple Network Management Control (SNMP) and

Administering Telephone Options.

on page 29 and QoS on page 40.

Link Layer Discovery Protocol (LLDP).

● Interface administration for the telephone, as appropriate. Administer the telephone to

LAN interface using the PHY1 parameter described in Chapter 3: Administer the telephone to PC interface using the PHY2 parameter described in “Interface Control” in the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP

Telephones Installation and Maintenance Guide.

Network Requirements.

● Application-specific telephone administration, if appropriate, as described in Chapter

8: Administering Telephone Options. An example of application-specific data is

Web-specific information required for the optional Web browser application.

Table 1

indicates that you can administer system configuration parameters in a variety of ways

and use a variety of administrative mechanisms like:

● Maintaining the information on the call server.

● Manually entering the information by means of the telephone dialpad.

● Administering the DHCP server.

● Editing the configuration file on the applicable HTTP or HTTPS file server.

● User modification of certain parameters, when given administrative permission to do so.

Note:

Note: Not all parameters can be administered on all administrative mechanisms.

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Table 1: Administration Alternatives and Options for 9600 Series SIP IP Telephones

Administrative

Parameter(s)

Mechanisms

For More Information See:

9600 Series IP Telephones

Telephone Administration

Avaya Communication Manager and SES

IP Addresses DHCP

(strongly

Chapter 4: Chapter 6: B: Related Documentation.

DHCP and File Servers on page 53, and especially DHCP Server Administration

Communication Manager Administration, Server Administration, and Appendix

on page 54.

recommended) Settings file Chapter 7: Telephone Software and Binary Files and

Manual administration at the telephone

LLDP

Tagging and VLAN LLDP

DHCP

Chapter 8:

“St atic Addressing Inst allation” in the Avaya one-X™

Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.

Link Layer Discovery Protocol (LLDP) Link Layer Discovery Protocol (LLDP)

DHCP Server Administration

Administering Telephone Options.

on page 101. on page 101.

on page 54, and

Chapter 8: Administering Telephone Options.

Settings file DHCP and File Servers

on page 53 and

Chapter 8: Administering Telephone Options.

Manual administration at the telephone

Network Time Server (NTS)

DHCP Settings file

Quality of Service Settings file Chapter 8:

“St atic Addressing Inst allation” in the Avaya one-X™

Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.

DHCP Server Administration on page 54 and Network Time Protocol (NTP) Server on page 27.

Administering Telephone Options.

Interface DHCP DHCP and File Servers on page 53, and Chapter

7: Telephone Software and Binary Files.

Settings file (strongly

DHCP and File Servers on page 53, and Chapter 7: Telephone Software and Binary Files.

recommended) LLDP

Link Layer Discovery Protocol (LLDP) on page 101.

Application specific parameters

Manual administration at the telephone

“Secondary Ethernet Interface Enable/Disable” in the

Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.

DHCP DHCP and File Servers on page 53, and especially

DHCP Server Administration

on page 54. Also,

Chapter 8: Administering Telephone Options.

Settings file (strongly recommended)

DHCP and File Servers

on page 53, and especially

HTTP Generic Setup on page 66. Also, Chapter 8:

Administering Telephone Options.

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Administration Overview and Requirements

General information about administering DHCP servers is covered in DHCP and File

Servers on page 53, and more specifically, DHCP Server Administration on page 54. General

information about administering HTTP servers is covered in DHCP and File Servers specifically, HTTP Generic Setup

. Once you are familiar with that material, you can administer

telephone options as described in Chapter 8:

Parameter Data Precedence

As shown in Table 1: Administration Alternatives and Options for 9600 Series SIP IP

Telephones, you can administer a given parameter in a number of ways. If a given parameter is

administered through different mechanisms, the last server to provide the parameter has precedence. The precedence, from lowest to highest, is:

1. LLDP

2. DHCP

3. Settings file

, and more

Administering Telephone Options.

4. Personal Profile Manager (PPM),

Note:

Note: Exception: In the case of the parameter SIPDOMAIN, the settings file has a

higher precedence than PPM.

5. Manual administration, unless the system parameter USE_DHCP is set to 1 (Get IP Address automatically by DHCP), or backup file data obtained through PPM.

For example, if the SIP outbound proxy server address is defined to have the precedence information so that the value retrieved from DHCP server has a lower precedence than the value retrieved from the settings file, and the value retrieved from the settings file is higher than the value retrieved from PPM, then the following determination occurs:

● If the most recent value the telephone has is from DHCP and new server address

information is retrieved from the settings file, the telephone will use the new value from the settings file.

● If later on, the telephone receives a new server address value from PPM, it will not use this

value because PPM’s precedence as a data source for the server address is lower than the current value (which came from the settings file).

● If the server to which a specific telephone points is changed manually using the Craft

ADDR procedure, that value now takes precedence over the previous value.

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Note:

Note: The only exception to this sequence is in the case of VLAN IDs. In the case of

VLAN IDs, LLDP settings of VLAN IDs are the absolute authority. Then the usual sequence applies. For the L2QVLAN and L2Q system values, LLDP settings of VLAN IDs are the absolute authority only if the LLDP task receives the VLAN IDs before DHCP, and the DHCP client of the telephone is activated at all. If the LLDP task receives the VLAN IDs after DHCP negotiation, several criteria must be successful before the telephone accepts VLAN IDs from LLDP. For more information, see Link Layer Discovery Protocol (LLDP)

The Administrative Process

The following list depicts administration for a typical 9600 Series SIP IP Telephone network. Your own configuration might differ depending on the servers and system you have in place.

1. Avaya Communication Manager (4.0 or greater) administered for 9600 Series IP Telephones. All 9600 Series SIP IP Telephones must be administered with the 4620SIP station type.

The Administrative Process

on page 101.

2. SES (SIP Enablement Services) administered.

3. LAN and applicable servers (file servers, Network Time server) administered to accept the telephones.

4. Telephone software downloaded from the Avaya support site.

5. 46xxsettings file updated with site-specific and SIP-specific information, as applicable.

6. 9600 Series Telephones installed. For more information, see the Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.

7. Individual 9600 Series IP Telephones updated using Craft procedures, as applicable. For more information, see “Local Administrative Procedures” in the Avaya one-X™ Deskphone Edition for 9600 SIP P Telephones Installation and Maintenance Guide.

Administrative Checklist

Use the following checklist as a guide to system and LAN administrator responsibilities. This high-level list helps ensure that all telephone system prerequisites and requirements are met prior to telephone installation.

Note:

Note: One person might function as both the system administrator and the LAN

administrator in some environments.

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Administration Overview and Requirements

Table 2: Administrative Checklist

Task Description For More Information See:

Network Requirements Assessment

Administer Avaya Communication Manager

Administer the Proxy Server

DHCP server installation

Administer DHCP application

Administer Network Time Server

HTTP/HTTPS server installation

Determine that network hardware is in place and can handle telephone system requirements.

Verify that the call server is licensed and is administered for Voice over IP (VoIP).

Verify the individual telephones are administered as desired.

Administer for SIP Enablement Services (SES).

Install a DHCP application on at least one new or existing PC on the LAN.

Add IP telephone administration to DHCP application.

Set value(s) for Simple Network Time Protocol (SNTP)

Install an HTTP/HTTPS application on at least one new or existing PC on the LAN.

Chapter 3:

Network Requirements.

Chapter 4: Communication Manager Administration.

Chapter 4:

Communication Manager

Administration.

Installing and Administering SIP Enablement Services (03-600768),

available on the Avaya support Web site,

http://www.avaya.com/support.

Vendor-provided instructions.

DHCP Server Administration

Chapter 6: Server Administration.

Option 42 under DHCP Generic

Setup.

Vendor-provided instructions.

Binary file(s), script file, and settings file installation on HTTP/ HTTPS server

Modify settings file as needed

Download the files from the Avaya support site.

Edit the settings file as necessary for your environment, using your

http://www.avaya.com/support Chapter 7: Telephone Software and

Binary Files. Chapter 7:

Binary Files.

own tools.

Administer telephones locally as applicable

As a Group: The GROUP System Value on

page 72 and the Avaya one-X™

Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.

Individually: The applicable Craft Local

Procedures in the Avaya one-X™

Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.

20 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

Telephone Software and

1 of 2

Page 21

Telephone Initialization Process

Table 2: Administrative Checklist (continued)

Task Description For More Information See:

Installation of telephones in the network

Allow user to modify Options, if applicable

Set the following parameters in the settings file:

ENABLE_CALL_LOG ENABLE_CONTACTS ENABLE_MODIFY_CONTACTS ENABLE_PRESENCE

PROVIDE_OPTIONS_SCREEN PROVIDE_NETWORKINFO_SCR

EEN PROVIDE_LOGOUT

Telephone Initialization Process

These steps offer a high-level description of the information exchanged when the telephone initializes and registers. This description assumes that all equipment is properly administered ahead of time. This description can help you understand how the 9600 Series SIP IP Telephones relate to the routers and servers in your network.

Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.

9600 Series SIP IP Telephones Customizeable System Parameters.

2 of 2

Step 1: Telephone to Network

The telephone is appropriately installed and powered. After a short initialization process, the telephone identifies the LAN speed and sends a message out into the network, identifying itself and requesting further information. A router on the network re ceives and relays this message to the appropriate DHCP server.

Step 2: Telephone to LLDP-Enabled Network

An LLDP-enabled network provides information to the telephone, as described in Link Layer

Discovery Protocol (LLDP) on page 101. Among other data passed to the telephone is the IP

Address of the HTTP or HTTPS server.

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Administration Overview and Requirements

Step 3: Telephone to DHCP Server

The DHCP server provides information to the telephone, as described in DHCP and File

Servers on page 53. Among other data passed to the telephone is the IP Address of the HTTP

or HTTPS server.

Step 4: Telephone and File Server

The 9600 Series IP Telephones can download script files, binary files, certificates, language files, and settings files from either an HTTP or HTTPS server. The telephone queries the file server, which transmits a script file to the telephone. This script file, at a minimum, tells the telephone which binary file the telephone must use. The binary file is the software that has the telephony functionality.

The telephone uses the script file to determine if it has the proper binary file. If the telephone determines the proper binary file is missing, the telephone requests an binary file download from the file server. The file server the n downloads the file and conducts some checks to ensure that the file was downloaded properly. If the telephone determines it already has the proper file, the telephone proceeds as described in the next paragraph without downloading the binary file again.

The telephone checks and loads the binary file, then uses the script file to look for a settin gs file, if appropriate. The optional settings file can contain settings you have administered for any or all of the 9600 Series SIP IP Telephones in your network. For more information about this download process and settings file, see Chapter 7:

Step 5: Telephone and the SES Server

In this step, the telephone might prompt the user for an extension and password. The telephone uses that information to exchange a series of messages with SES, which in turn communicates with Avaya Communication Manager (CM). For a new installation and for full service, the user can enter the telephone extension and the SES password. For a restart of an existing installation, this information is already stored on the telephone, but the user might have to confirm the information. The telephone and SES and SES and CM exchange more messaging. The expected result is that the telephone is appropriately registered and CM call server data such as feature button assignments are downloaded.

For more information about the installation process, see the Ava ya one-X™ Deskphone Editio n for 9600 SIP IP Telephones Installation and Maintenance Guide.

Telephone Software and Binary Files.

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Error Conditions

Assuming proper administration, most of the problems reported by telephone users are likely to be LAN-based. Quality of Service, server administration, and other issues can impact user perception of IP telephone performance.

The Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide covers possible operational problems that might be encountered after successful 9600 Series SIP IP Telephone installation. The User Guides for a specific tele phone model also contain guidance for users having problems with specific IP telephone applications.

Error Conditions

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Administration Overview and Requirements

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Chapter 3: Network Requirements

Network Assessment

Perform a network assessment to ensure that the network will have the capacity for the expected data and voice traffic, and that it can support for all applications:

● SIP,

● DHCP,

● HTTP/HTTPS, and

● Jitter buffers

Also, QoS support is required to run VoIP on your configuration. For more information, see

Appendix B: Related Documentation

and DSCPSIG in Table 11:

9600 Series SIP IP Telephones Customizeable System

Parameters.

and the QoS parameters L2QAUD, L2QSIG, DSCPAUD,

Hardware Requirements

To operate properly, you need:

● Category 5e cables designed to the IEEE 802.3af-2003 standard, for LAN powering,

● TN2602 IP Media Processor circuit pack. Sites with a TN2302 IP Media Processor circuit

pack are strongly encouraged to install a TN2602 circuit p ack to benefit from the increased capacity.

● TN799C or D Control-LAN (C-LAN) circuit pack.

Important:

Important: IP telephone firmware Release 1.0 or greater requires TN799C V3 or greater

C-LAN circuit pack(s). For more information, see the Communication Manager Software and Firmware Compatibility Matrix on the Avaya support Web site

http://www.avaya.com/support

To ensure that the appropriate circuit pack(s) are administered on your Communication Manager call server, see Chapter 4: information about hardware requirements in general, see the Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.

Communication Manager Administration. For more

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Network Requirements

Server Requirements

Four server types can be configured for the 9600 Series IP Telephones:

● DHCP server

● HTTP or HTTPS server

● SIP Proxy or Registration server

● Network Time Protocol server for SNTP

Note:

Note: 9600 Series SIP IP Telephones need SIP Enablement Services (SES) to work

properly. The SIP Proxy and Registration servers reside on the SES server. Avaya Communication Manager (CM) is considered a “feature server” behind SES that provides Outboard Proxy SIP (OPS) features. SIP software Release 2.0 supports both SES 4.X and 5.X, but when the corresponding server is running SES 4.X, the telephones assume only those features compatible with SES 4.X.

While the servers listed provide different functions that relate to the 9600 Se ries IP Telephones, they are not necessarily different boxes. For example, DHCP provides network information whereas HTTP provides configuration and application file management, yet both functions can co-exist on one hardware unit. Any standards-based server is recommended.

For parameters related to Avaya Communication Manager information, see Chapter

4: Communication Manager Administration. For parameters related to DHCP and file servers,

see Chapter 6:

Important:

Important: The telephones obtain important information from the script files on the server(s)

and depend on the binary file for software upgrades. If these servers are unavailable when the telephones reset, the telephones will not operate properly. Some features might not be available. To restore them you need to reset the telephone(s) when the file server is available.

DHCP Server

Avaya recommends that a DHCP server and application be installed and that static addressing be avoided. Install the DHCP server and application as described in DHCP and File Servers page 53.

Server Administration.

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HTTP/HTTPS Server

Administer the HTTP or HTTPS file server and application as described in HTTP Generic

Setup on page 66.

Network Time Protocol (NTP) Server

SIP IP telephones require NTP server support to set the time and date, used in system log time stamps and other time/date functions. The NTP server is typically needed by one or more servers within the enterprise. Administration of the NTP server is beyond the scope of this document.

Required Network Information

Before you administer DHCP and HTTP/HTTPS, as applicable, complete the information in

Table 3

configuration, complete Table 3

. If you have more than one router, HTTP/TLS server and subnetwork mask in your

for each DHCP server.

Required Network Information

The 9600 Series SIP IP Telephones support specifying a list of IP Addresses for a gateway/ router and the HTTP/HTTPS server . Each list can contain up to 255 tot al ASCII characters, with IP Addresses separated by commas with no intervening spaces. Depending on the specific DHCP application, only 127 characters might be supported.

When specifying IP Addresses for the file server, use either dotted decimal format (“xxx.xxx.xxx.xxx”) or DNS names. If you use DNS, the system value DOMAIN is appended to the IP Addresses you specify. If DOMAIN is null, the DNS names must be fully qualified, in accordance with IETF RFCs 1034 and 1035. For more information about DNS, see DHCP

Generic Setup on page 56 and DNS Addressing on page 98.

Table 3: Required Network Information Before Installation - Per DHCP Server

1. Gateway (router) IP Address(es)

2. HTTP server IP Address(es)

3. Subnetwork mask

4. HTTP server file path (HTTPDIR)

5. Telephone IP Address range

From: To:

6. DNS server address(es) If applicable.

7. HTTPS server address(es) If applicable.

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Network Requirements

The default file server file path is the “root” directory used for al l tra nsfe r s by the server. All files are uploaded to or downloaded from this default directory. In configurations where the upgrade script and binary files are in the default directory, do not use item 4 in Table 3

As the LAN or System Administrator, you are also responsible for:

● Administering the DHCP server as described in Chapter 6: Server Administration.

● Editing the configuration file on the applicable HTTP or HTTPS file server, as covered in

9600 Series SIP IP Telephone Scripts and Binary Files

Other Network Considerations

SNMP

The 9600 Series SIP IP Telephones are fully compatible with SNMPv2c and with Structure of Management Information Version 2 (SMIv2). The telephones respond correctly to queries from entities that comply with earlier versions of SNMP, such as SNMPv1. “Fully compatible” means that the telephones respond to queries directed either at the MIB-II or the read-only Custom MIB. Read-only means that the values therein cannot be changed externally by means of network management tools.

You can restrict which IP Addresses the telephone accepts SNMP queries from. You can also customize your community string with system values SNMPADD and SNMPSTRING, respectively. For more information, see Chapter 6:

Series SIP IP Telephones Customizeable System Parameters.

Note:

Note: SNMP is disabled by default. Administrators must initiate SNMP by setting the

SNMPADD and SNMPSTRING system values appropriately.

For more information about SNMP and MIBs, see the IETF Web site listed in

Appendix B: Related Documentation

. The Avaya Custom MIB for the 9600 Series SIP IP Telephones is available for download in *.txt format on the Avaya support Web site at

http://www.avaya.com/support

Registration and Authentication

A 9600 Series SIP IP Telephone requires an outboard proxy SIP (OPS) extension on Avaya Communication Manager and a login and password on the SES Server to register and authenticate it. Registration is described in the Initialization process, in Step

the SES Server on page 22. For further information, see Installing and Administering SIP

Enablement Services R 4.0 (03-600766), available on the Avaya support Web site,

http://www.avaya.com/support

Server Administration and Table 11: 9600

5: Telephone and

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Reliability and Performance

All 9600 Series SIP IP Telephones respond to a ping or traceroute message sent from Avaya Communication Manager or any other network source. The telephones do not originate a ping or traceroute. The 9600 Series SIP IP Telephones offer and support “remote ping” and “remote traceroute.” The switch can instruct the telephone to originate a ping or a traceroute to a specified IP Address. The telephone carries out that instruction and sends a message to the switch indicating the results. For more information, see your switch administration documentation.

If applicable, the telephones test whether the network Ethernet switch port supports IEEE

802.1D/q tagged frames by ARPing the router with a tagged frame. For more information, see

VLAN Considerations

on page 94. If your LAN environment includes Virtual LANs (VLANs),

your router must respond to ARPs for VLAN tagging to work properly.

QoS

Other Network Considerations

For more information about the extent to which your network can support any or all of the QoS initiatives, see your LAN equipment documentation. See QoS for the 9600 Series SIP IP Telephones.

All 9600 Series SIP IP Telephones provide some detail about network audio quality. For more information see, Network Audio Quality Display on 9600 Series SIP IP Telephones

IEEE 802.1D and 802.1Q

For more information about IEEE 802.1D and IEEE 802.1Q and the 9600 Series SIP IP Telephones, see IEEE 802.1D and 802.1Q Three bits of the 802.1Q tag are reserved for ide ntifying p acket priority to allow any one of eight priorities to be assigned to a specific packet.

● 7: Network management traffic

● 6: Voice traffic with less than 10ms latency

● 5: Voice traffic with less than 100ms latency

● 4: “Controlled-load” traffic for critical data applications

● 3: Traffic meriting “extra-effort” by the network for prompt delivery, for example, executive

e-mail

● 2: Reserved for future use

● 0: The default priority for traffic meriting the “best-effort” for prompt delivery of the network.

● 1: Background traffic such as bulk data transfers and backups

on page 40 for QoS implications

on page 30.

on page 40 and VLAN Considerations on page 94.

Note:

Note: Priority 0 is a higher priority than Priority 1.

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Network Requirements

Network Audio Quality Display on 9600 Series SIP IP Telephones

All 9600 Series SIP IP Telephones give the user an opportunity to monitor network audio performance while on a call. For more information, see the telephone user guide.

While on a call, the telephones display network audio quality parameters in real-time, as shown in Table 4

Table 4: Parameters in Real-Time

Parameter Possible Values

Received Audio Coding G.711, G.722, G.726A, or G.729. Packet Loss No data or a percentage. Late and out-of-sequence packet s are

Packetization Delay No data or an integer number of milliseconds. The number

counted as lost if they are discarded. Packets are not counted as lost until a subsequent packet is received and the loss confirmed by the RTP sequence number.

reflects the amount of audio data in each RTP packet.

One-way Network Delay No data or an integer number of milliseconds. The number is

one-half the value RTCP or SR TCP computes for the round-trip delay.

Network Jitter Compensation Delay

The implication for LAN administration depends on the values the user reports and the specific nature of your LAN, like topology, loading, and QoS administration. This information gives the user an idea of how network conditions affect the audio quality of the current call. Avaya assumes you have more detailed tools available for LAN troubleshooting.

No data or an integer number of milliseconds reporting the average delay introduced by the jitter buffer of the telephone.

SIP Station Number Portability

The 9600 Series SIP IP Telephones provide station number portability. On startup or a reboot, the telephone attempts to establish communication with its home Personal Profile Manager (PPM)/SIP Enablement Services (SES) server based on the User Name and Password.

Assume a situation where the company has multiple locations in London and New York, all sharing a corporate IP network. Users want to take their telephone functionality from their offices in London to their New York office. When users start up their telephones in the new location and enter their credentials, the local SES/PPM server usually routes them to the local call server. With proper administration of the local SES/PPM server, the telephone knows to try its home SES/PPM server, the one in London. The user can then be automatically registered with the London SES/PPM server.

30 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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TCP/UDP Port Utilization

The 9600 Series SIP IP Telephones use a variety of protocols, particularly TCP (Transmission Control Protocol), UDP (User Datagram Protocol), and TLS (Transport Layer Security) to communicate with other equipment in the network. Part of this communication identifies which TCP or UDP ports each piece of equipment uses to support each protocol and each task within the protocol. For additional TCP/UDP port utilization information as it applies to Avaya Communication Manager, see UDP Port Selection

Depending on your network, you might need to know what ports or ranges are used in the operation of 9600 Series IP Telephones. Knowing these ports or ranges helps you administer your networking infrastructure.

Note:

Note: In many cases, the ports used are the ones called for by IETF or other standards

bodies. Many of the explanations in Table options settings. For more information about parameters and settings, see

Administering Options for the 9600 Series SIP IP Telephones

Table 5: Received Packets (Destination = SIP IP Telephone)

Other Network Considerations

on page 39.

and Table refer to configuration parameters or

Destination Port Source Port Use UDP or TCP?

The number used in the

Any Received DNS messages UDP

Source Port field of the DNS

query sent by the telephone

The number used in the

Source Port field of the

Any Packets received by the

telephone’s HTTP client

packets sent by the

telephone’s HTTP client The number used in the

Source Port field of the TLS/

SSL packets sent by the

Any TLS/SSL packets received

by the telephone’s HTTP client

telephone’s HTTP client

68 Any Received DHCP messages UDP

The number used in the

Any Received SNTP messages UDP

Source Port field of the SNTP

query sent by the telephone

161 Any Received SNMP messages UDP

50000 Any Received CNA test request

messages

TCP

UDP

1 of 2

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Network Requirements

Table 5: Received Packets (Destination = SIP IP Telephone) (continued)

Destination Port Source Port Use UDP or TCP?

The number used in the

Source Port field of

registration messages sent by

the telephone’s CNA Agent

PORT AUD or the port number

reserved for CNA RTP tests

PORTAUD + 1 (if PORTAUD is even) or PORTAUD – 1 (if

PORTAUD is odd) or the port

number reserved for CNA

RTP tests plus or minus one,

as for PORTAUD, above

If signaling is initiated by the

telephone = the number used

in the Source Port field of the signaling packets sent by the

telephone

If signaling is initiated by the

server = System-Specific

Any Received CNA registration

messages

Any Received RTP and SRTP

packets

Any Received RTCP and

SRTCP packets

Any Received signaling protocol

packets

TCP

UDP

UDP/TCP

2 of 2

Table 6: Transmitted Packets (Source = SIP IP Telephone)

Destination Port Source Port Use UDP or TCP?

53 Any unused

Transmitted DNS messages UDP

port number

67 68 Transmitted DHCP

messages

80 unless explicitly specified

otherwise (i.e. in a URL)

123 Any unused

The number used in the

Source Port field of the SNMP

Any unused port number

Packets transmitted by the

telephone’s HTTP client

Transmitted SNTP

port number

messages

161 Transmitted SNMP

messages

query packet received by the

telephone

443 unless explicitly specified

otherwise (i.e. in a URL)

Any unused port number

TLS/SSL packets

transmitted by the

telephone’s HTTP client

UDP

TCP

UDP

TCP

1 of 3

32 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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Other Network Considerations

Table 6: Transmitted Packets (Source = SIP IP Telephone) (continued)

Destination Port Source Port Use UDP or TCP?

514 Any unused

CNAPORT Any

The port number specified in

the test request message

System-specific Any unused

FEPORT or the port number

specified in a CNA RTP test

request

FEPORT + 1 (if FEPORT is

even) or FEPORT -1 (if

FEPORT is odd) or the port

number specified in a CNA

RTP test request plus or minus

one, as with FEPORT above

Transmitted Syslog

port number

messages

Transmitted CNA

otherwise

registration messages

unused port

number

50000 Transmitted CNA test result s

messages

Transmitted signaling

port number

PORTAUD, which must

protocol packets

Transmitted R TP and SR TP

packets

be in the

range

specified by

the

RTP_PORT

_LOW and

RTP_PORT

_RANGE

parameters

or the port

number

reserved for

CNA RTP

tests

PORTAUD +

1 (if

PORTAUD

RTCP and SRTCP packets transmitted to the far-end of the audio connection

is even) or

PORT AUD –

1 (if

PORTAUD

is odd) or

the port number

reserved for

CNA RTP

tests plus or

minus one,

as for

PORTAUD,

above

UDP

TCP

UDP

TCP

UDP

2 of 3

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Network Requirements

Table 6: Transmitted Packets (Source = SIP IP Telephone) (continued)

Destination Port Source Port Use UDP or TCP?

Security

For information about toll fraud, see the respective call server documents on the Ava ya support Web site. The 9600 Series SIP IP Telephones cannot guarantee resistance to all Denial of Service attacks. However, there are checks and protections to resist such attacks while maintaining appropriate service to legitimate users.

9600 Series SIP IP Telephones support Transport Layer Security (TLS) for signaling and for secure communications (SRTP). This standard allows the telephone to establish a secure connection to a HTTPS server, in which the upgrade and settings file can reside. This setup adds security over another alternative.

RTCPMONPORT PORTAUD +

1 (if PORTAUD is even) or

PORT AUD –

1 (if PORTAUD

is odd)

System-specific Any unused

port number

RTCP packets transmitted to an RTCP monitor

Transmitted signaling protocol packets

UDP

3 of 3

Communications between the 9600 Series SIP IP telephone and the Personal Profile Manager (PPM) can also be secured by setting the CONFIG_SERVER_SECURE_MODE parameter.

You also have a variety of optional capabilities to restrict or remove how crucial network information is displayed or used. These capabilities are covered in more detail in

Chapter 6:

● Depending on the SIGSIGNAL parameter, supporting signaling channel encryption while

Server Administration and include:

registering, and when registered, with appropriately administered Avaya Communication Manager.

● Restricting the response of the 9600 Series SIP IP T elephones to SNMP queries to only IP

Addresses on a list you specify.

● Specifying an SNMP community string for all SNMP messages the telephone sends.

● Restricting dialpad access to Craft Local Procedures to experienced installers and

technicians and requiring password entry to access Craft procedures.

● Restricting the end user’s ability to use a telephone Options application to view network

data.

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Registration and Authentication

A 9600 Series SIP IP Telephone requires an off-PBX station (OPS) extension on Avaya Communication Manager and a login and password on the SES Server to register and authenticate it. For more information, see the current version of your call server administration manual.

Other Network Considerations

Issue 2 December 2007 35

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Network Requirements

36 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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Chapter 4: Communication Manager Administration

Call Server Requirements

Avaya Communication Manager (CM) extends advanced telephony features to SIP telephones via Outboard Proxy SIP (OPS) support. This feature set offers enhanced calling features in advance of SIP protocol definitions and telephone implementations.

Before you perform administration tasks, ensure that the proper hardware is in place, and your call server software is compatible with the 9600 Series SIP IP Telephones. Avaya recommends the latest CM software and the latest SIP IP telephone firmware.

Switch Compatibility

As of SIP Release S1.0, 9600 Series IP Telephones are supported by Avaya Communication Manager (CM) Release 4.0 and later. Be sure to administer 9600 Series SIP IP Telephones as 4620SIP telephones on Avaya Communication Manager.

Note:

Note: The 9620 only supports a total of 12 call appearances and administered feature

buttons. The 9630/9630G and 9640/9640G can be administered for a total of 24 call appearances and feature buttons.

For specific administration instructions about the 9600 Series SIP IP Telephones, see

Administering Stations

on page 48.

Communication Manager Administrative Requirements

There are several initial CM provisioning tasks that must be performed before administering SIP users. These tasks are described in SIP Support in Avaya Communication Manager Ru nning on Avaya S8XXX Server s (Document Number 555-245-206), the latest release of which is Issue 8, January 2008. The tasks to administer Communication Manager for SIP Enablement Services (SES) and fall into three categories:

● system-level preparation,

● SIP trunk administration, and

● call routing administration

The sections that follow describe each of these tasks.

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Communication Manager Administration

System-Level Preparation Tasks

The system-level preparation tasks include:

● Setting the SIP Trunk capacity on the System Capacity screen.

● Verifying that the IP Trunks field is set to y on the System-Parameters Customer-Options

screen page 4.

● Verifying that the Maximum Administered SIP Trunks are set correctly on the System

Parameters Customer-Options screen page 2.

● Setting the OPS SIP station capacity on the System Parameters Customer Options screen

page 1.

● Setting the IP Node name for SES on the IP Node Names screen.

● Entering the IP Address and host name for the administered SES server on the IP Address

Mapping screen.

● Setting the Authoritative Domain on the IP Network Region screen.

● Setting the intra- and inter-region IP-IP Direct Audio to yes on the IP Network Region

screen.

● Setting the Signaling Group on the Signaling Group screen page 1.

SIP Trunk Administration

SIP trunk administration tasks include:

● Setting the SIP Intercept Treatment and Trunk-to-Trunk Transfer on the System

Parameters Features screen page 1.

● Administering Trunk Groups on the Trunk Group screens (pages 1 through 4).

● Assigning public unknown numbering data on the Numbering - Public/Unknown

Numbering screen.

● Assigning a SIP phone Set description on Configuration Set screen.

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Call Routing Administration

Call routing administration includes:

● Administering Feature Access Codes (FACs) on the Feature Access Code screen.

● Administering the ARS Digit Analysis Table on the ARS Digit Analysis Table screen.

● Administering the Route Pattern on the Route Pattern screen.

● Adding the Route Pattern to the Numbering - Public/Unknown Numbering screen.

● Administering the Proxy Selection Route Pattern on the Locations screen.

● Allowing the system to identify the location of a caller who dials a 911 emergency call fro m

a SIP endpoint on the IP Network Map screen.

The Administrator Guide for Avaya Communication Manager (Document Number 03-300509) provides detailed instructions for administering an IP telephone system on Avaya Communication Manager . See Chapter 3 “Managing Telephones,” which describes the process of adding new telephones. Also, you can locate pertinent screen illustrations and field descriptions in Chapter 19 “Screen References” of that guide. You can find this document on the Avaya support Web site.

Communication Manager Administrative Requirements

IP Interface and Addresses

Follow these general guidelines:

● Define the IP interfaces for each C-LAN and Media processor circuit pack on the switch

that uses the IP Interfaces screen. For more information, see Administration for Network Connectivity for Avaya Communication Manager (Document 555-233-504).

● On the Customer Options form, verify that the IP Stations field is set to “y” (Yes). If it is

not, contact your Avaya sales representative.

UDP Port Selection

The 9600 Series SIP IP Telephones use an even-numbered port, selected from the interval 4000 to 10000. The telephones cannot be administered from the Avaya Communication Manager Network Region form to support UDP port selection.

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Communication Manager Administration

RSVP and RTCP/SRTCP

Avaya SIP IP Telephones support the RTP/SRTP Control Protocol (RTCP/SRTCP). The 9600 Series SIP IP Telephones do not support RSVP (Resource ReSerVation Protocol).

QoS

The 9600 Series SIP IP Telephones support both IEEE 802.1D/Q and DiffServ. Other network-based QoS initiatives such as UDP port selection do not require support by the telephones. However, the initiatives contribute to improved QoS for the entire network.

IEEE 802.1D and 802.1Q

The 9600 Series IP Telephones can simultaneously support receipt of packets using, or not using, 802.1Q parameters. To support IEEE 802.1D/Q, you can administer 9600 Series SIP IP Telephones by the value of the following configuration parameters:

● L2Q,

● L2QVLAN,

● L2QAUD, and

● L2QSIG.

NAT

9600 Series SIP IP Telephones do not support Network Address Translation (NAT) interworking.

DIFFSERV

Type o f Service bits 0-5 (also called the Dif ferentiated Services Code Point) are set to the binary equivalent of the decimal number represented by the value of the following configuration parameters:

● DSCPAUD for transmitted audio (RTP, RTCP, SRTP and SRTCP) packets;

● DSCPSIG for transmitted system-specific signaling packets;

● Zero for all other transmitted packets (e.g., DHCP, DNS, HTTP, SNMP, etc.).

Received DSCP information will be ignored.

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Voice Mail Integration

9600 Series SIP IP Telephones use the settings file to configure the Messages button by setting the system parameter MSGNUM

● a standard telephone number the telephone should dial to access your voice mail system,

such as AUDIX or Octel.

● a Feature Access Code (FAC) that allows users to transfer an active call directly to voice

mail. FACs are supported only for QSIG-integrated voice mail systems like AUDIX or Octel. QSIG is an enhanced signaling system that allows the voice mail system and Avaya Communication Manager Automated Call Processing (ACP) to exchange information.

When the user presses the Messages button on the telephone, that number or FAC is automatically dialed, giving the user one-touch access to voice mail.

The settings file specifies the telephone number to be dialed automatically when the user presses this button. The command is:

SET MSGNUM 1234

Voice Mail Integration

to any dialable string. MSGNUM examples are:

where 1234 is the Voice Mail extension (CM hunt group or VDN). For more information, see Table 11

Auto Hold

9600 Series SIP IP Telephones always provide auto hold, regardless of whether or not the Auto Hold parameter is administered on the IP Network System Parameters form.

Call Transfer Considerations

Unlike 9600 H.323 IP Telephones, the 9600 Series SIP IP Telephones transfer operation is controlled locally by the telephone and is not affected by the settings Abort Transfer?, Transfer Upon Hang-up and Toggle Swap, on page 7 of the system-parameters features screen.

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Communication Manager Administration

Conferencing Call Considerations

Unlike 9600 H.323 IP Telephones, the 9600 Series SIP IP Telephones conference operation is controlled locally by the phone and is not affected by the settings Abort Conference Upon Hang-up, No Dial Tone Conferencing, Select Line Conferencing and Toggle Swap, on page 7 of the system-parameters features screen.

Telephone Administration

Table 7 summarizes the calling features available on 9600 Series SIP IP Telephones. Some

features are supported locally at the telephone, while others are only available with Avaya SIP Enablement Services and Communication Manager with OPS.

The features shown in Table 7 CM-provisioned feature button. Communication Manager automatically handles many other standard calling features via OPS such as call coverage, trunk selection using Automatic Alternate Routing (AAR), or Automatic Route Selection (ARS), Class Of Service/Class Of Restriction (COS/COR), and voice messaging. Details on feature operation and administration can be found in the Avaya Extension to Cellular and OPS Installa tion and Administration Guide (Document Number 210-100-500). The Avaya SIP solution configures all SIP telephones in Communication Manager as OPS.

Table 7: 9600 Series SIP IP Telephone Release S2.0 Feature Support

Feature

3-Way Conferencing X

6-way Conference Bridge

Automatic Call Back/ Cancel

Call Forward All Calls

can be invoked at the phone either directly or by selecting a

Delivered by

Telephone

(non-Avaya

environment)

CM Feature Button/FNU

FNU

Avaya Communication

Manager

Call Forward Busy/ Don’t Answer

42 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

(non-Avaya

environment)

FNU

1 of 3

Page 43

Telephone Administration

Table 7: 9600 Series SIP IP Telephone Release S2.0 Feature Support (continued)

Feature

Call Forward Deactivation

Call Forward Unconditional

Call Hold Call Management -

incoming, outgoing call screening

Call Park and Unpark Call Pick-Up Group Call Pickup Directed Call Pickup Extended

Group

Delivered by

Telephone

(non-Avaya

environment)

CM Feature Button/FNU

FNU

FNU FNU FNU FNU

Avaya Communication

Manager

Calling Party Number Block/Unblock

Consultation Hold Directed Call Pick-Up Distinctive Alerting EC500 Enable EC500 Disable Extend Call for EC500

Extended Group Call Pickup

Find Me Group Call Pickup

FNU

X X FNU X

FNU FNU FNU Av ailable with CM Release

5.0

FNU

2 of 3

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Communication Manager Administration

Table 7: 9600 Series SIP IP Telephone Release S2.0 Feature Support (continued)

Feature

Last Number Dialed (Redial)

Malicious Call Trace Message Waiting

Indication Music on Hold One Touch Recording Priority Call Send All Calls Enable/

Disable Transfer - attended

(non-Avaya environment)

Transfer - unattended (one-button transfer) (non-Avaya environment

Delivered by

Telephone

No FNU

CM Feature Button/FNU

FNU FNU

Avaya Communication

Manager

Transfer to Voice Mail Whisper Page

FNU

CM/SIP IP Telephone Configuration Requirements

This section refers to Communication Manager (CM) administration on the Switch Administration Terminal (SAT) or by Avaya Site Administration. The system wide CM form and the particular page that needs to be administered for each feature are provided . These features, which already exist, are not required but are recommended because they optimize the telephone user interface. CM 4.0 or greater is required. For sample Station and other pertinent forms, see Appendix C:

44 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

Sample Statio n Forms.

3 of 3

Page 45

CM/SIP IP Telephone Configuration Requirements

Table 8: CM/SIP Configuration Requirements

Task/Form Command Field(s) Value(s)

IP Network Region RTCP Report

Period (secs)

SIP telephones have a fixed reporting period. Note that this parameter is only displayed if "Use Default Server Parameters?" is set to "n".

IP Network Region Authoritative

Domain

Make sure that the Authoritative Domain is set to the same value as SIP Domain for Solution.

Off-PBX Telephones Station Mapping

change off-pbx-station mapping xxxx

Bridged call items on this form MUST be “none” or “orig.” In CM Release 5.0, default is “none.”

Feature - Related System Parameters (page 1)

change system-parameters features

Music/Tone on Hold

This CM setting controls the music on hold capability for all endpoints, including SIP telephones.

Feature - Related System Parameters (page 4)

change system-parameters features

Directed Call Pickup

Extended Group Call Pickup

This CM setting controls the availability of directed call pickup.

This CM setting allows a user to answer calls that were directed to another call pickup group.

Feature - Related System Parameters (page 17)

Define the dial plan formats on the Dialplan Analysis Table form

change system-parameters features

change dialplan analysis

Whisper Page Tone Given To

This CM setting controls who hears the whisper page.

Call Type Includes all telephone

extensions and OPS Feature Name Extensions (FNEs). To define the FNEs for the OPS features listed in Table FAC must also be specified for the corresponding feature. In a sample configuration, telephone extensions are five digits in length and begin with 3 or 4, FNEs are five digits beginning with 7, and the access codes have various formats as indicated with the Call Type of “fac.”

Define the access codes corresponding to the OPS FNEs on the

change feature-access-codes

Various fields on pages 1-5 of the

form Feature Access Code form

, a

1 of 4

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Communication Manager Administration

Table 8: CM/SIP Configuration Requirements (continued)

Task/Form Command Field(s) Value(s)

After defining the FACs, define the FNEs not provisioned by CM feature buttons using the

change off-pbx-telephone feature-nameextensions

Used to support both OPS and Extension to Cellular.

command Set the appropriate

change cos Varied y (Yes) or n (No) service permissions to support OPS features on the Class of Service form

Enable applicable calling features on the Class of Restriction form

change cor Varied To use the Call Pickup

feature, the Can Use Directed Call Pickup and Can Be Picked Up By Call Pickup fields must be set to "y" for the affected stations. Note that Page 3 can be used to implement a form of centralized call screening for groups of stations and trunks

Add a station for each SIP phone to be supported using the Station form (page 1)

add station xxxxxx

(where xxxxxx

represents the

extension number)

Extension Assign the same extension

as the CM call server extension administered in SIP Enablement Services. See

Chapter 5: SIP Enablement Services (SES) Administration for SES

configuration information. (Station) Type Use 9620 or 9630. Port System-populated. Coverage Path For voice messaging or other

hunt group, if available. COS and COR Same values as administered

in the previous COS & COR

section(s). Name The person associated with

the telephone. This name

should match what is entered

for name in the Avaya SES

proxy configuration. Message Lamp

Ext

Enter the extension of the

station you want to track with

the message waiting lamp.

(Usually the same extension

initially entered on the S t ation

form.)

2 of 4

46 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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CM/SIP IP Telephone Configuration Requirements

Table 8: CM/SIP Configuration Requirements (continued)

Task/Form Command Field(s) Value(s)

Continue adding station information for the SIP phone using the Station form (page 2)

add station xxxxxx (where xxxxxx represents the extension number)

Bridged Call Alerting

Set to "y" if the extension for

this SIP telephone will have a

"bridged" appearance defined

on another non-SIP

telephone. Note that no other

attributes of the bridged

appearance feature apply to

SIP telephones (e.g. off-hook

indication, bridge-on, etc.). Restrict Last

Appearance

By default, the last call

appearance is reserved for

outgoing calls from a phone.

On stations with only three (3)

call appearances, set the field

to "n" for proper SIP

conference and transfer

operation. In this mode, all

call appearances are

available for making or

receiving calls. AUDIX Name Enter the name of the voice

messaging system

administered for this system. Coverage After

Forwarding

This field, with a default of "s"

for system, governs whether

an unanswered forwarded

call is given CM coverage

treatment. Per Station CPN

Send Calling Number?

If CM is configured to always

send Caller ID, you can

individually block certain

stations by setting this field to

"n". This field also needs to

be set to "n" if you want to

use the "Calling Number

nblock" FNE.

Continue adding station button assignments for the SIP telephone using the Station form (page 4)

BUTTON ASSIGNMENTS

1. call-appr

2. call-appr etc.

Fill in the number of call

appearances ("call-appr"

buttons) to be supported for

this telephone. Use the

following guidelines to

determine the correct

number:

To support certain transfer

and conference scenarios,

the minimum number of

"call-appr" buttons should be

3 of 4

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Communication Manager Administration

Table 8: CM/SIP Configuration Requirements (continued)

Task/Form Command Field(s) Value(s)

Stations With Off-PBX Telephone Integration form (page 1)

Stations With Off-PBX Telephone Integration form (page 2)

change off-pbx-telephone station-mapping

xxxxxx where xxxxxx represents

the extension number of the station being configured

change off-pbx-telephone station-mapping

xxxxxx where xxxxxx represents

the extension number of the station being configured

Station Extension

Application Dial Prefix Phone Number Trunk Selection Configuration

Set Call Limit Change the call limit to match

Use to map the

Communication Manager

extension to the same SIP

Enablement Services call

server extension. The

Application is "OPS." Enter

the other appropriate field

values, for example, the

Trunk Selection value

indicates the SIP trunk group.

The Configuration Set value

can reference a set that has

the default settings in

Communication Manager.

the number of "call-appr"

entries in the Add Station

form.

4 of 4

Administering Stations

This section refers to Communication Manager (CM) administration on the Switch Administration Terminal (SAT) or by Avaya Site Administration. Administer the following items on the Station form, sample screens of which are provided in Figure 1 recommends setting the features covered in this section because they optimize the user interface.

48 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

through Figure 4. Avaya

Page 49

Administering Features

The following buttons can be administered for a 9600 Series SIP IP Telephone, unless otherwise noted:

Administrable Station Features

Feature Administration Notes Audix One-Touch

Recording Auto Callback Autodial Bridged Call Appearances Busy Indicator Call Appearances Call Forward (all) Leave the Ext: field blank, as the telephone does not support

Call Forwarding (busy/ don’t answer)

Call Park Call Unpark This (SES) feature will show up automatically without

Call Pickup CPN Block CPN Unblock Directed Call Pickup EC500 EC500 Extend Call Extended Call Pickup This (SES) feature will show up automatically without

MCT Activation This (SES) feature will show up automatically without

Priority Call Send All Calls Leave the Ext: field blank, as the telephone does not support

Transfer-to-Voicemail This (SES) feature will show up automatically without

Whisper Page

Administering Stations

3rd party call forwarding. Leave the Ext: field blank, as the telephone does not support

3rd party call forwarding.

administration. Regardless of CM St ation screen administration, these features will show on the Features menu automatically, but not on a telephone button.

3rd party send all calls. administration. Regardless of CM St ation screen administration,

these features will show on the Features menu automatically, but not on a telephone button.

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Communication Manager Administration

For additional information about administering Avaya Communication Manager for 9600 Series SIP IP Telephones, see the following Avaya documents, available on the Avaya Support Web site:

● Administrator Guide for Avaya Communication Manager (Document 03-300509).

● Feature Description and Implementation for Avaya Communication Manager (Document

555-245-770).

50 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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Chapter 5: SIP Enablement Services (SES)

Administration

Introduction

SIP Enablement Services (SES) software resides on the SIP Proxy server and provides most of the features and functionality to SIP telephones.This chapter describes using the SES Web browser to configure SES for use with 9600 Series SIP IP Telephones.

Avaya provides a Web browser to simplify SES administration.

Using the Web Browser to Configure SES

Follow this configuration procedure.

1. Set the browser URL to http://IP-address/admin Avaya SIP Enablement Services Edge or Edge/Home Server.

2. Log in as the administrator “admin” and when prompted, enter the password. The main administration screen displays after login.

Note:

Note: This example administers station 34071 as a SIP endpoint using a 9630

telephone.

3. Click on Launch Administration Web Interface. The SIP Enablement Services Web interface screen displays.

4. Click Add under the Users heading on the left side menu. The Add User screen displays.

5. Complete all required fields, indicated by asterisks *.

6. Enter a handle in the Primary Handle field. The Primary Handle must be all numeric.

7. Set the Host field to the DNS host name of the Avaya SIP Enablement Services Home or Home/Edge server to which the telephone will register.

8. Check the Add Media Server Extension checkbox and click Add. The confirmation screen displays.

, where IP-address is the IP Address of the

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SIP Enablement Services (SES) Administration

9. Click Continue. The Add Media Server Extension page displays.

10. In the Extension field, enter the same extension you entered on page 1 of the Communication Manager Station form. This step links the extension recorded in Avaya Communication Manager to the extension recorded in SES. (See Feature Description and Implementation for Avaya Communication Manager Document Number 555-245-205 for information about Station form entries if necessary).

11. Click Add. Since the user is being added to Avaya SES Home, the Communication Manager (CM) call

server corresponding to the SIP trunk between the CM server and SES Home is selected. The confirmation page displays.

12. Click Continue.

13. Repeat Steps 4 - 11 for each SIP telephone.

14. When you finish configuring all applicable telephones, click Update on the left side menu. This link appears on the current page whenever updates are outstanding, and can be selected at any time to save the administration performed to that point.

52 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

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Chapter 6: Server Administration

Software Checklist

Ensure that you own licenses to use the DHCP, HTTP, and HTTPS server software.

Note:

Note: You can install the DHCP and HTTP server software on the same machine.

Important:

Important: The firmware in the 9600 Series SIP IP Telephones reserves IP Addresses of the

form 192.168.2.x for internal communications. The telephone(s) improperly use addresses you specify if they are of that form.

DHCP and File Servers

Dynamic Host Configuration Protocol (DHCP) minimizes maintenance for a 9600 Series SIP IP Telephone network by removing the need to individually assign and maintain IP Addresses and other parameters for each telephone on the network.

The DHCP server provides the following information to the 9600 Series SIP IP Telephones:

● IP Address of the 9600 Series SIP IP Telephone(s)

● IP Address of the HTTP or HTTPS server

● IP Address of the NTP (Network Time Protocol) server (using Option 42)

● The subnet mask

● IP Address of the router

● DNS Server IP Address

Administer the LAN so each SIP IP telephone can access a DHCP server that contains the IP Addresses and subnet mask.

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Important:

Important: An IP telephone cannot function without an IP Address. The failure of a DHCP

server at boot time leaves all the affected telephones unusable. A user can manually assign an IP Address to an IP telephone. When the DHCP server finally returns, the telephone never looks for a DHCP server unless the static IP data is unassigned manually. In addition, manual entry of IP data is an error-prone process.

Avaya recommends that:

● A minimum of two DHCP servers be available for reliability.

● A DHCP server be available when the IP telephone reboots.

● A DHCP server be available at remote sites if WAN failures isolate IP telephones from the

central site DHCP server(s).

The file server provides the 9600 Series SIP IP Telephone with a script file and, if appropriate, new or updated binary software. See Step

Telephone Initialization Process

customize telephone parameters for your specific environment. For more information, see

Chapter 8:

Administering Telephone Options.

4: Telephone and File Server on page 22 under

. In addition, you can edit an associated settings file to

DHCP Server Administration

This document concentrates on the simplest case of the single LAN segment. Information provided here can be used for more complex LAN configurations.

Important:

Important: Before you start, understand your current network configuration. An improper

installation will cause network failures or reduce the reliability and performance of your network.

Configuring DHCP for 9600 Series SIP IP Telephones

To administer DHCP option 242, make a copy of an existing option 176 for your 46xx IP Telephones. You can then either:

● leave any parameters the 9600 Series SIP IP Telephones do not support for setting via

DHCP in option 242 to be ignored, or

● delete unused or unsupported 9600 IP Series T elephone parameters to shorten the DHCP

message length.

Only the following parameters can be set in the DHCP site-specific option for 96xx telephones, although most of them can be set in a 46xxsettings.txt file as well.

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Table 9: Parameters Set by DHCP

Parameter Description

HTTPDIR Specifies the path name to prepend to all file names used in HTTP GET

operations during startup. (0 to 127 ASCII characte rs, no spaces.) The command is “SET HTTPDIR myhttpdir”. The path (relative to the root of the HTTP file server) where 96xx telephone files are stored. If an Avaya file server is used to download configuration files over TLS, but a different server is used to download software files via HTTP, set the path of the Avaya server in the DHCP site-specific option, and set HTTPDIR again in the 46xxsettings.txt file with the appropriate path for the second server. HTTPDIR is the path for all HTTP operations except

for BRURI. HTTPPORT Destination port for HTTP requests (0-65535, default is 80). HTTPSRVR IP Address(es) or DNS name(s) of HTTP file server(s) used for file download

(settings file, language files, code) during startup. The files are digitally signed, so

TLS is not required for security. ICMPDU Controls the extent to which ICMP Destination Unreachable messages are sent in

response to messages sent to closed ports so as not to reveal information to

potential hackers. The default is 1 (sends Destination Unreachable messages fo r

closed ports used by traceroute). ICMPRED Controls whether ICMP Redirect messages are processed. The default is 0

(redirect messages are not processed). L2Q 802.1Q tagging mode. The default is 0 (automatic). L2QVLAN VLAN ID of the voice VLAN. The default is 0. LOGSRVR Syslog server IP or DNS address. MTU_SIZE Maximum transmission unit size. Used to accommodate older Ethernet switches

that cannot support the longer maximum frame length of tagged frames (since

802.1Q adds 4 octets to the frame). PHY1STAT Controls the Ethernet line interface speed. The default is 1 (auto-negotiate). PHY2STAT Controls the secondary Ethernet interface speed. The default is 1

(auto-negotiate). PROCPSWD Security string used to access local procedures. The default is 27238. PROCSTAT Controls whether local procedures are enabled. The default is 0 (enabled). SIPPROXYSRVR SIP proxy/registrar server IP or DNS address. (0 to 255 characters; zero or one IP

Address in dotted decimal or DNS name format, separated by commas without

any intervening spaces.) The default is null. SNTPSRVR List of SNTP server IP or DNS address(es) u.sed to retrieve date and time via

SNTP TLSDIR Used as path name that is prepended to all file names used in HTTPS get

operations during initialization (0-127 character string). TLSPORT Destination TCP port used for requests to https server (0-65535). The default is

443.

TLSSRVR IP Address(es) or DNS name(s) of Avaya file server(s) used to download

configuration files.

Note: Transport Layer Security is used to authenticate the server. VLANTEST Number of seconds to wait for a DHCPOFFER on a non-zero VLAN. The default

is 60 seconds.

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DHCP Generic Setup

This document is limited to describing a generic administration that works with the 9600 Series SIP IP Telephones. Three DHCP software alternatives are common to Windows operating systems:

● Windows NT

● Windows 2000

● Windows 2003

Any other DHCP application might work. It is the responsibility of the customer to install and configure the DHCP server correctly.

DHCP server setup involves:

1. Installing the DHCP server software according to vendor instructions.

2. Configuring the DHCP server with:

● IP Addresses available for the 9600 Series SIP IP Telephones.

4.0 DHCP Server

DHCP Server

● The following DHCP options:

- Option 1 - Subnet mask. As described in Table 3

, item 3.

- Option 3 - Gateway (router) IP Address(es). As described in Table 3

, item 1. If using more than one address, the total list can contain up to 127 total ASCII characters. You must separate IP Addresses with commas with no intervening spaces.

- Option 6 - DNS server(s) address list. If using more than one address, the total list can contain up to 255 total ASCII characters. Y ou must separate IP Addresses with commas with no intervening sp aces. At least one address in Option 6 must be a valid, non zero, dotted decimal address.

- Option 12 - Host Name. V alue is AVohhhhhh, where: o has one of t he following values base d on the OID (first three octets) of the telephone’s MAC a ddress: “A” if the OID is 00-04-0D, “B” if the OID is 00-1B-4F, (SIP software Release 2.0+), “E” if the OID is 00-09-6E, “L” if the OID is 00-60-1D, “T” if the OID is 00-07-3B, (SIP software Release R2.0+) a nd “X” if the OID is anything else, and where hhhhhh are ASCII characters for the hexadecimal representation of the last three octets of the telephone’s MAC address.

- Option 15 - DNS Domain Name. This string contains the domain name to be used when DNS names in system parameters are resolved into IP Addresses. This domain name is appended to the DNS name before the 9600 IP Telephone attempts to resolve the DNS address. Option 15 is necessary if you want to use a DNS name for the HTTP server. Otherwise, you can specify a DOMAIN as part of customizing HTTP as indicated in

DNS Addressing

on page 98.

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- Option 42 - SNTP Server. This option specifies a list of IP Addresses indicating NTP servers available to the telephone. List servers in the order of preference.The minimum length is 4, and the length must be a multiple of 4.

- Option 51 - DHCP lease time. If this option is not received, the DHCPOFFER is not be accepted. Avaya recommends a lease time of six weeks or greater. If this option has a value of FFFFFFFF hex, the IP Address lease is assumed to be infinite as per RFC 2131, Section 3.3, so that renewal and rebinding procedures are not necessary even if Options 58 and 59 are received. Expired leases cause Avaya IP Telephones to reboot. Avaya recommends providing enough leases so an IP Address for an IP telephone does not change if it is briefly taken offline.

Note:

Note: Regarding Option 51: The DHCP standard states that when a DHCP lease

expires, the device should immediately cease using its assigned IP Address. If the network has problems and the only DHCP server is centralized, the server is not accessible to the given telephone. In this case the telephone is not usable until the server can be reached. Avaya recommends that once assigned an IP Address, the telephone continues using that address after the DHCP lease expires, until a conflict with another device is detected. As Table 11:

9600 Series SIP IP Telephones Customizeable System Parameters indicates, the system

parameter DHCPSTD allows an administrator to specify that the telephone will either: a). Comply with the DHCP standard by setting DHCPSTD to “1”, or b). Continue to use its IP Address after the DHCP lease expires by setting DHCPSTD to “0.” The latter case is the default. If the default is invoked, after the DHCP lease expires the telephone sends an ARP Request for its own IP Address every five seconds. The request continues either forever, or until the telephone receives an ARP Reply. After receiving an ARP Reply, the telephone d isp lays an error message, sets its IP Address to 0.0.0.0, and attempts to contact the DHCP server again.

- Option 52 - Overload Option, if desired. If this option is received in a message, the telephone interprets the sname and file fields in accordance with IETF RFC 2132, Section 9.3, listed in Appendix B: Related Documentation

- Option 53 - DHCP message type. Value is 1 (DHCPDISCOVER) or 3 (DHCPREQUEST).

- Option 55 - Parameter Request List. Acceptable values are:

1 (subnet mask), 3 (router IP Address[es]) 6 (domain name server IP Address[es]) 7 (log server) 15 (domain name) 26 (Interface MTU) 42 (NTP servers) SSON (site-specific option number)

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- Option 57 - Maximum DHCP message size.

- Option 58 - DHCP lease renew time. If not received or if this value is greater than that for Option 51, the d efault value of T1 (renewal timer) is used as per IETF RFC 2131, Section 4.5, listed in Related

Documentation.

- Option 59 - DHCP lease rebind time. If not received or if this value is greater than that for Option 51, the d efault value of T2 (rebinding timer) is used as per RFC 2131, Section 4.5

The 9600 Series IP Telephones do not support Regular Expression Matching, and therefore, do not use wildcards. For more information, see Administering Options for the 9600 Series SIP IP

Telephones on page 73.

In configurations where the upgrade script and binary files are in the default directory on the HTTP server, do not use the HTTPDIR=<path>.

Avaya recommends that you administer DHCP servers to deliver only the options specified in this document. Administering additional, unexpected options might have unexpected results, including causing the IP telephone to ignore the DHCP server.

The SIP Proxy server name and HTTP server name must each be no more than 32 characters in length.

Examples of good DNS administration include:

- Option 6: “aaa.aaa.aaa.aaa”

- Option 15: “dnsexample.yourco.com,zzz.zzz.zzz.zzz”

- Option 42: "aaa.aaa.aaa.aaa" Depending on the DHCP application you choose, be aware that the application most

likely does not immediately recycle expired DHCP leases. An expired lease might remain reserved for the original client a day or more. For example, Windows NT DHCP reserves expired leases for about one day. This reservation period protects a lease for a short time. If the client and the DHCP server are in two different time zones, the clocks of the computers are not in sync, or the client is not on the network when the lease expires, there is time to correct the situation.

The following example shows the implication of having a reservation period: Assume two IP Addresses, therefore two possible DHCP leases. Assume three IP telephones, two of which are using the two available IP Addresses. When the lease for the first two telephones expires, the third telephone cannot get a lease until the reservation period expires. Even if the other two telephones are removed from the network, the third telephone remains without a lease until the reservation period expires.

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In Table 10, the 9600 Series IP Telephone sets the system values to the DHCPACK message field values shown.

Table 10: DHCPACK Setting of System Values

System Value Set to

DHCP lease time Option #51 (if received). DHCP lease renew time Option #58 (if received). DHCP lease rebind time Option #59 (if received). DOMAIN Option #15 (if received). DNSSRVR Option #6 (if received, which might be a list of IP Addresses). HTTPSRVR The siaddr field, if that field is non-zero. IPADD The yiaddr field. LOGSRVR Option #7 (if received). MTU_SIZE Option #26. NETMASK Option #1 (if received). ROUTER Option #3 (if received, which might be a list of IP Addresses). SNTPSRVR Option #42.

Windows NT 4.0 DHCP Server

Verifying the Installation of the DHCP Server

Use the following procedure to verify whether the DHCP server is installed.

1. Select Start-->Settings-->Control Panel.

2. Double-click the Network icon.

3. Verify that Microsoft DHCP Server is listed as one of the Network Services on the Services tab.

4. If it is listed, continue with the next section. If it is not listed, install the DHCP server.

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Creating a DHCP Scope for the IP Telephones

Use the following procedure to create a DHCP scope for the IP telephones.

1. Select Start-->Programs-->Admin Tools-->DHCP Manager.

2. Expand Local Machine in the DHCP Servers window by double clicking it until the + sign changes to a - sign.

3. Select Scope-->Create.

4. Using information recorded in Table 3:

Required Network Information Before Installation -

Per DHCP Server:

Define the Telephone IP Address Range. Set the Subnet Mask. To exclude any IP Addresses you do not want assigned to IP telephones within the Start

and End addresses range:

a. In the Exclusion Range Start Address field, enter the first IP Address in the range that

you want to exclude.

b. In the Exclusion Range End Address field, enter the last IP Address in the range that

you want to exclude. c. Click the Add button. d. Repeat steps a. through c. for each IP Address range to be excluded.

Note:

Note: Avaya recommends that you provision the 9600 Series IP Telephones with

sequential IP Addresses. Also do not mix 9600 Series IP Telephones and PCs in the same scope.

5. Under Lease Duration, select the Limited To option and set the lease duration to the maximum.

6. Enter a sensible name for the Name field, such as “CM IP Telephones," where CM would represent Avaya Communication Manager.

7. Click OK.

A dialog box prompts you: Activate the new scope now?

8. Click No.

Note:

Note: Activate the scope only after setting all options.

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Editing Custom Options

Use the following procedure to edit custom options.

1. Highlight the newly created scope.

2. Select DHCP Options-->Defaults in the menu.

3. Click the New button.

4. In the Add Option T ype dialog box, enter an appropriate cu stom option name, for example, “9600OPTION.”

5. Change the Data Type Byte value to String.

6. Enter 242 in the Identifier field.

7. Click the OK button.

The DHCP Options menu displays.

8. Select the Option Name for 242 and set the value string.

9. Click the OK button.

10. For the Option Name field, select 003 Router from the drop-down list.

DHCP Server Administration

11. Click Edit Array.

12. Enter the Gateway IP Address recorded in Table 3:

Installation - Per DHCP Server for the New IP Address field.

13. Select Add and then OK.

Adding the DHCP Option

Use the following procedure to add the DHCP option.

1. Highlight the scope you just created.

2. Select Scope under DHCP Options.

3. Select the 242 option that you created from the Unused Options list.

4. Click the Add button.

5. Select option 003 from the Unused Options list.

6. Click the Add button.

7. Click the OK button.

8. Select the Global parameter under DHCP Options.

9. Select the 242 option that you created from the Unused Options list.

Required Network Information Before

10. Click the Add button.

11. Click the OK button.

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Activating the Leases

Use the following procedure to activate the leases.

● Click Activate under the Scope menu.

The light-bulb icon for the scope lights.

Verifying Your Configuration

This section describes how to verify that the 96XXOPTIONs are correctly configured for the Windows NT

4.0 DHCP server.

Verify the Default Option, 242 96XXOPTION

1. Select Start-->Programs-->Admin Tools-->DHCP Manager.

2. Expand Local Machine in the DHCP servers window by double clicking until the + sign changes to a - sign.

3. In the DHCP servers frame, click the scope for the IP telephone.

4. Select Defaults from the DHCP_Options menu.

5. In the Option Name pull-down list, select 242 96XXOPTION.

6. Verify that the Value String box contains the correct string from DHCP Server

Administration.

If not, update the string and click the OK button twice.

Verify the Scope Option, 242 96XXOPTION

1. Select Scope under DHCP OPTIONS.

2. In the Active Options: scroll list, click 242 96XXOPTION.

3. Click the Value button.

4. Verify that the Value String box contains the correct string from DHCP Generic Setup page 56.

If not, update the string and click the OK button.

Verify the Global Option, 242 96XXOPTION

1. Select Global under DHCP OPTIONS.

2. In the Active Options: scroll list, click 242 96XXOPTION.

3. Click the Value button.

4. Verify that the Value String box contains the correct value from DHCP Generic Setup page 56. If not, update the string and click the OK button.

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Windows 2000 DHCP Server

Verifying the Installation of the DHCP Server

Use the following procedure to verify whether the DHCP server is installed.

1. Select Start-->Program-->Administrative Tools-->Computer Management.

2. Under Services and Applications in the Computer Management tree, find DHCP.

3. If DHCP is not installed, install the DHCP server. Otherwise, proceed directly to Creating

and Configuring a DHCP Scope for instructions on server configuration.

Creating and Configuring a DHCP Scope

Use the following procedure to create and configure a DHCP scope.

1. Select Start-->Programs-->Administrative Tools-->DHCP.

2. In the console tree, click the DHCP server to which you want to add the DHCP scope for the IP telephones. This is usually the name of your DHCP server machine.

3. Select Action-->New Scope from the menu.

DHCP Server Administration

Windows displays the New Scope Wizard to guide you through rest of the setup.

4. Click the Next button.

The Scope Name dialog box displays.

5. In the Name field, enter a name for the scope such as “CM IP Telephones” (where CM would represent Avaya Communication Manager), then enter a brief comment in the Description field.

6. When you finish Steps 1 - 5, click the Next button.

The IP Address Range dialog box displays.

7. Define the range of IP Addresses used by the IP telephones listed in Table 3:

Required Network Information Before Installation - Per DHCP Server. The Start IP Address is the

first IP Address available to the IP telephones. The End IP Address is the last IP Address available to the IP telephones.

Note:

Note: Avaya recommends not mixing 9600 Series IP Telephones and PCs in the same

scope.

8. Define the subnet mask in one of two ways:

● The number of bits of an IP Address to use for the network/subnet IDs.

● The subnet mask IP Address.

Enter only one of these values. When you finish, click the Next button. The Add Exclusions dialog box displays.

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9. Exclude any IP Addresses in the range specified in the previous step that you do not want assigned to an IP telephone.

a. In the Start Address field under Exclusion Range, enter the first IP Address in the

range you want to exclude.

b. In the End Address field under Exclusion Range, enter the last IP Address in the

range you want to exclude. c. Click the Add button. d. Repeat steps a. through c. for each IP Address range that you want to exclude.

Note:

Note: You can add additional exclusion ranges later by right clicking the Address Pool

under the newly created scope and selecting the New Exclusion Range option. Click the Next button after you enter all the exclusions. The Lease Duration dialog box displays.

10. For all telephones that obtain their IP Addresses from the server, enter 30 days in the Lease Duration field. This is the duration after which the IP Address for the device expires and which the device needs to renew.

11. Click the Next button. The Configure DHCP Options dialog box displays.

12. Click the No, I will activate this scope later button. The Router (Default Gateway) dialog box displays.

13. For each router or default gateway, enter the IP Address and click the Add button. When you are done, click the Next button. The Completing the New Scope Wizard dialog box displays.

14. Click the Finish button. The new scope appears under your server in the DHCP tree. The scope is not yet active

and does not assign IP Addresses.

15. Highlight the newly created scope and select Action-->Properties from the menu.

16. Under Lease duration for DHCP clients, select Unlimited and then click the OK button.

CAUTION:

CAUTION: IP Address leases are kept active for varying periods of time. To avoid having

calls terminated suddenly, make the lease duration unlimited.

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Adding DHCP Options

Use the following procedure to add DHCP options to the scope you created in the previous procedure.

1. On the DHCP window, right-click the Scope Options fold er under the scope you created in the last procedure.

A drop-down menu displays.

2. In the left pane of the DHCP window, right click the DHCP Server name, then click Set

Predefined Options....

3. Under Predefined Options and Values, click Add.

4. In the Option Type Name field, enter any appropriate name, for example, “Avaya IP Telephones.”

5. Change the Data Type to String.

6. In the Code field, enter 242, then click the OK button twice.

The Predefined Options and Values dialog box closes, leaving the DHCP dialog box enabled.

DHCP Server Administration

7. Expand the newly created scope to reveal its Scope Options.

8. Click Scope Options and select Action-->Configure Options from the menu.

9. In the General tab page, under the Available Options, check the Option 242 checkbox.

10. In the Data Entry box, enter the DHCP IP telephone option string as described in

DHCP Generic Setup

Note:

Note: You can enter the text string directly on the right side of the Data Entry box under

the ASCII label.

11. From the list in Available Options, check option 003 Router.

12. Enter the gateway (router) IP Address from the IP Address field of Table 3:

Network Information Before Installation - Per DHCP Server.

13. Click the Add button.

14. Click the OK button.

Activating the New Scope

Use the following procedure to activate the new scope.

1. In the DHCP console tree, click the IP Telephone Scope you just created.

on page 56.

Required

2. From the Action menu, select Activate. The small red down arrow over the scope icon disappears, indicating tha t the scope was

activated.

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HTTP Generic Setup

You can store the same binary file, script file, and settings file on an HTTP server as you can on a TFTP server. TFTP is not supported for 9600 Series SIP IP Telephones. With proper administration, the telephone seeks out and uses that material. Some functio nality might be lost by a reset if the HTTP server is unavailable. For more information, see DHCP and File

Servers on page 53.

Important:

Important: The files defined by HTTP server configuration must be accessible from all IP

telephones invoking those files. Ensure that the file names match the names in the upgrade script, including case, since UNIX systems are case-sensitive.

Note:

Note: Use any HTTP application you want. Commonly used HTTP applications include

Apache

Important:

Important: To set up an HTTP server:

and Microsoft® IIS™.

● Install the HTTP server application.

● Administer the system parameter HTTPSRVR to the address of the HTTP server.

Include this parameter in DHCP Option 242, or the appropriate SSON Option.

● Download the upgrade script file and binary file(s) from the Avaya Web site

http://www.avaya.com/support Contents of the Settings File

to the HTTP server. For more information, see

on page 70.

Note:

Note: Many LINUX servers distinguish between upper and lower case names. Ensure

that you specify the settings file name accurately, as well as the names and values of the data within the file.

If you choose to enhance the security of your HTTP environment by using Transport Layer Security (TLS), you also need to:

● Install the TLS server application.

● Administer the system parameter TLSSRVR to the address(es) of the Avaya HTTP server.

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Chapter 7: Telephone Software and Binary Files

General Download Process

The 9600 Series SIP IP Telephones download script files, binary files, and settings files from either an HTTP or HTTPS server. The HTTPS server applies only if the server supports Transport Layer Security (TLS) encryption.

Note:

Note: The script files, binary files, and settings files discussed in this chapter are

identical for HTTP and HTTPS servers. The generic term “file server” refers to both “HTTP server” and “HTTPS server.”

The file downloading process is the same for both servers, except that when you use an HTTPS server, a TLS server is contacted first. The telephone queries the file server, which transmits a script file to the telephone. The script file tells the telephone which binary file the telephone must use. The binary file is the software that has the telephony functionality, and is easily updated for future enhancements. In a newly installed telephone, the binary file might be missing. In a previously installed telephone, the binary file might not be the proper one. In both cases, the telephone requests a download of the proper binary file from the file server. The file server downloads the file and conducts some checks to ensure that the file was downloaded properly. If the telephone determines it already has the proper file, the telephone proceeds to the next step without downloading the binary file again.

After checking and loading the binary file, the 9600 Series SIP IP Telephone, if appropriate, uses the script file to look for a settings file. The settings file contains options you have administered for any or all of the IP Telephones in your network. For more information about the settings file, see Contents of the Settings File

Software

As part of installation, a con version from H.323 to SIP signaling protocol is done as described in "Converting Software on 9600 Series IP Telephones" of the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide. When the telephone is first plugged in, a software download from an HTTP or HTTPS server start s to give the phone its proper functionality.

For software upgrades, SIP Enablement Services (SES) provides the capability for a remote reboot of the 9600 Series SIP IP Telephones. As a result, the telephone automatically starts reboot procedures. If new software is available on the server , the telephone downloads it as p art of the reboot process. The Avaya one-X™ Deskphone Edition for 9600 IP Telephones Installation and Maintenance Guide covers upgrades to a previously installed telephone and related information.

on page 70.

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Telephone Software and Binary Files

9600 Series SIP IP Telephone Scripts and Binary Files

Choosing the Right Binary File and Upgrade Script File

The software releases containin g the files needed to operate the 9 600 Series IP Telephones are bundled together. You download this self-extracting executable file to your file server from the Avaya support Web site at: http://www.avaya.com/support and unzipped format. You must select one of two software “bundles" to download the latest software, depending on whether your telephone environment is primarily SIP-centric or H.323-centric.

Each bundle contains:

● An upgrade script file, 96xxupgrade.txt, which allows you to upgrade to new software

releases and new functionality without having to replace SIP IP telephones. The upgrade script tells the telephone whether a software upgrade is needed. All Avaya IP Telephones attempt to read this file whenever they reset. The upgrade script file is also use d to point to the settings file. An alternate file may be included, depending on which software bundle you download.

. The file is available in both zipped

● Binary files for all current 9600 Series SIP IP Telephones.

● Other useful information such as a ReadMe file and the latest binary code.

In addition to the upgrade script, binary files and Read Me file you need the latest binary code the Avaya SIP IP Telephones use, which is part of the sof tware bundle you choose for your site. All these files are in self-extracting executable file comes in both zipped and unzipped format.

When the majority of your IP telephones are SIP-based, select the software bundle identified as “SIP” from the Avaya Support Web site. The binary files in this SIP software bundle are the same as in the H.323 bundle. The difference is a modified upgrade script file that assumes SIP is the default protocol for your 9600 Series IP Telephones, and that H.323 is the exception. For more information on SIP-centric environments, see "Converting Software on 9600 Series IP Telephones" in the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide.

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Upgrade Script File

An upgrade script file named 96xxupgrade.txt, tells the IP telephone whether the telephone needs to upgrade software. The 9600 Series SIP IP Telephones attempt to read this file whenever they reset. The upgrade script file also points to the settings file.

You download the upgrade script file, sometimes called the “script file,” from

http://www.avaya.com/support

customer-definable options. All files must reside in the same directory. An "alternate" upgrade script is also included, designed for environments that will support both

the H323 and SIP modes of operation. For such environments, the file needs to be edited in those sections having headings of “H.323 EDIT INSTRUCTIONS.” Specific instructions are provided in the Readme file that accompanies each software bundle. Once these changes are made, the alternate file should be renamed to “96xxupgrade.txt” and placed in the HTTP download directory. The HTTP download directory holds the telephone backup and application binaries the telephone will download. Renaming the alternate file causes any “96xxupgrade.txt” files residing in that directory to be overwritten.

Note:

Note: Avaya recommends that the settings file have the extension *.txt. The Avaya IP

Telephones can operate without this file. You can also change these settings with DHCP or, in some cases, from the dialpad of the telephone.

9600 Series SIP IP Telephone Scripts and Binary Files

. This file allows the telephone to use default settings for

Settings File

The settings file contains the option settings you need to customize the Avaya IP Telephones for your enterprise.

Note:

Note: Use one settings file for all your Avaya IP Telephones. The settings file includes

the 9600 Series SIP IP Telephones covered in this document. The settings file also includes 9600 Series (H.323) IP Telephones, 4600 Series IP Telephones, and 1600 Series IP Telephones as covered in their respective administrator guides.

The settings file can include any of five types of statements, one per line:

● Comments, which are statements with a “#” character in the first column.

● Tags, which are comments that have exactly one space character after the initial #,

followed by a text string with no spaces.

● Goto commands, of the form GOTO tag. Goto commands cause the telephone to

continue interpreting the configuration file at the next line after a # tag statement. If no such statement exists, the rest of the configuration file is ignored.

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● Conditionals, of the form IF $name SEQ string GOTO tag. Conditionals cause the Goto

command to be processed if the value of name is a case-insensitive equivalent to string. If no such name exists, the entire conditional is ignored. The only system values that can be used in a conditional statement are: BOOTNAME, GROUP, and SIG.

● SET commands, of the form SET parameter_name value. Invalid values cause the

specified value to be ignored for the associated parameter_name so the default or previously administered value is retained. All values must be text strings, even if th e value itself is numeric, a dotted decimal IP Address, and so on.

Note:

Note: Enclose all data in quotation marks for proper interpretation.

The upgrade script file Avaya provides includes a line that tell the telephone to GET 46xxsettings.txt. This lines causes the telephone to use HTTP to attempt to download the file specified in the GET command. If the file is obtained, its contents are interpreted as an additional script file. That is how your settings are changed from the default settings. If the file cannot be obtained, the telephone continues processing the upgrade script file.

If the configuration file is successfully obtained but does not include any setting changes the telephone stops using HTTP. This happens when you initially download the script file template from the Avaya support Web site, before you make any changes. When the configuration file contains no setting changes, the telephone does not go back to the upgrade script file.

Avaya recommends that you do not alter the upgrade script file. If Avaya changes the upgrade script file in the future, any changes you have made will be lost. Avaya recommends that you use the 46xxsettings file to customize your settings instead. However, you can change the settings file name, if desired, as long as you also edit the corresponding GET command in the upgrade script file.

For more information on customizing your settings file, see Contents of the Settings File

Contents of the Settings File

After checking the software, the 9600 Series IP Telephone looks for a 46xxsettings file. This file is where you identify non-default option settings, application-specific parameters, and so on. You can download a template for this file from the Avaya support Web site. An examp le of what the file might look like follows.

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Note:

Note: The following is intended only as a simple example. Your settings will vary from

the settings shown. This sample assumes specification of a DNS Server, identifying SIP-specific settings, and setting the time/date.

SET DNSSRVR ”dnsexample.yourco.com” SET SIPPROXYSRVR 192.168.1.110 SET SIPSIGNAL “1” (TCP) SET ENABLE_PRESENCE “1” (show presence icons) SET SIPDOMAIN ”domain name” SET SNTPSRVR 192.168.1.111 SET GMTOFFSET “-5:00” SET DSTOFFSET “1” SET DSTSTART “2SunMar2L” (second Sunday in March at 2 am Local time) SET DSTSTOP “1SunNov2L” (first Sunday in November at 2 am Local time)

Note that the DSTSTART and DSTSTOP parameters reflect the new 2007 Daylight Savings Time values for North America

See Chapter 8:

Administering Telephone Options for details about specific values. You

need only specify settings that vary from defaults, although specifying defaults is harmless.

VLAN separation controls whether or not traffic received on the secondary Ethernet interface is forwarded on the voice VLAN and whether network traffic received on the data VLAN is forwarded to the telephone. Add commands to the 46xxsettings.txt file to enable VLAN separation. The following example assumes the data VLAN ID is “yyy” and the data traffic priority is “z”:

SET VLANSEP 1 SET PHY2VLAN yyy SET PHY2PRIO z

Note:

Note: Also configure the network switch so that 802.1Q tags are not removed from

frames forwarded to the telephone.

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The GROUP System Value

Y ou might have dif ferent communities of users, all of which have the same telephone model, but which require different administered settings. For example, you might want to group users by time zones or work activities.

Use the GROUP system value for this purpose:

1. identify which telephones are associated with which group, and designate a number for each group. The number can be any integer from 0 to 999, with 0 as the default, meaning your largest group is assigned as Group 0.

2. At each non-default telephone, instruct the installer or user to invoke the GROUP Craft Local procedure as specified in the Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide and specify which GROUP number to use. The GROUP System value can only be set on a phone-by-phone basis.

3. Once the GROUP assignments are in place, edit the configuration file to allow each telephone of the appropriate group to download its proper settings.

Here is an example of a settings file with associates in different groups at the same location:

IF $GROUP SEQ 1 goto GROUP1

IF $GROUP SEQ 2 goto GROUP2

{specify settings unique to Group 0}

goto END

# GROUP1

{specify settings unique to Group 1}

goto END

# GROUP2

{specify settings unique to Group 2}

# END

{specify settings common to all Groups}

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Administering Options for the 9600 Series SIP IP Telephones

This chapter explains how to change parameters by means of the DHCP or HTTP servers. In all cases, you are setting a system parameter in the telephone to a desired value. Table 11

● the parameter names,

● their default values,

● the valid ranges for those values, and

● a description of each one.

Table 11 is a comprehensive list of all the parameters you can configure. However, you do not have to set every parameter. In most cases, you will include only those parameters in the settings file that are specific to your own environment and let the telephones use the default values for the remaining ones. At a minimum, be sure to set these important SIP-related parameters: SIPPROXYSRVR, SIPDOMAIN, SNTPSRVR, SIPSIGNAL, ENABLE_PRESENCE, GMTOFFSET, DSTOFFSET, DSTSTART, and DSTSTOP.

lists:

For DHCP, the DHCP Option sets these parameters to the desired values as discussed in

DHCP and File Servers

values in the script file. For more information, see Contents of the Settings File Avaya recommends that you administer options on the 9600 Series SIP IP Telephones using

script files. Some DHCP applications have limits on the amount of user-specified information. The administration required can exceed those limits for the more full-featured telephone models.

You might choose to completely disable the capability to enter or change option settings from the dialpad. You can set the system value, PROCPSWD, as part of standard DHCP/HTTP administration. If PROCPSWD is non-null and consists of 1 to 7 digits, a user ca nnot invoke any local options without first entering the PROCPSWD value on the Craft Access Code Entry screen. For more information on craft options, see the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide.

on page 53. For HTTP, the parameters in Table 11 are set to desired

on page 70.

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Important:

Important: PROCPSWD is likely stored on the server “in the clear” and is sent to the

telephone in the clear. Therefore, do not consider PROCPSWD as a high-security technique to inhibit a sophisticated user from obtaining access to local procedures.

Administering PROCPSWD limits access to all local procedures, including VIEW. VIEW is a read-only Craft option that allows review of the current telephone settings.

Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters

Parameter Name Default

Description and Value Range

Value

AGCHAND 1 Automatic Gain Control status for handset. Values are

0=disabled, 1=enabled.

AGCHEAD 1 Automatic Gain Control status for headset. Values are

0=disabled, 1=enabled.

AGCSPKR 1 Automatic Gain Control status for speaker. Values are

0=disabled, 1=enabled.

AUDASYS 3 Globally controls audible alerting. Values range from 0

through 3. Value 0 or 2=audible alerting off. Value 1 or 3=audible alerting on.

AUDIOENV 0 Audio environment selection index. Values range from 0

through 191.

AUDIOSTHD 0 Headset sidetone setting. Values are:

0 = Nominal 1 = -3dB below nominal

2 = -9dB below nominal 3 = -15dB below nominal 4 = -30dB below nominal (essentially no sidetone) 5 = 10dB above nominal.

AUDIOSTHS 0 Handset sidetone setting. Values are:

0 = Nominal 1 = -3dB below nominal

2 = -9dB below nominal 3 = -15dB below nominal 4 = -30dB below nominal (essentially no sidetone) 5 = 10dB above nominal.

AUTH 0 Authentication flag for settings file download. Values are:

0=secure setting file download is not required 1=secure setting file download is required

BAKLIGHTOFF 120 Number of minutes without display activity to wait before

turning off the backlight. Values range from zero (never turn off) through 999 minutes (16.65 hours).

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

CALL_TRANSFER_ MODE

0 When ENABLE_AVAYA_ENVIRONMENT=0, this

parameter indicates how transfers are performed: 0 = attended transfer 1 = unattended transfer

CALLFWDADDR " " (Null) The URI to which calls are forwarded in 3rd party

(non-Avaya) environments only.

CALLFWDDELAY 1 Third-party (non-Avaya) environments only. Specifies the

number of ring cycles generated at the phone before the call is forwarded to the Call Forwarding Address, if call forwarding on "No answer" is selected in 3rd party environments. Valid number of ringing cycles are 0-20.

CALLFWDST AT 0 Third-party (non-Avaya) environments only. Specifies the

sum of the allowed Call Forwarding permissions. This parameter controls which of the Call Forwarding Feature Buttons are made visible and active for the user in 3rd party environments. Valid values are:

0 = no Call Forwarding permitted. 1 = Call Forward Unconditional only permitted. 2 = Call Forward Busy only permitted. 4 = Call Forward No Answer only permitted. Others = sum of Call Forward types permitted.

CNAPORT 50002 Transport-layer port number to be used for registration to

CNA server for network analysis. Valid range is 0-65535.

CNASRVR " " (Null) List of CNA server IP or DNS address(es). Used to

connect to CNA server for network analysis (in case of several entries first address always first, etc.). Format is 0 to 255 characters: zero or more IP addresses in dotted decimal or DNS name format, separated by commas without any intervening spaces. Currently set to a maximum of 5 servers.

CNGLABEL 1 Indicates whether end user can personalize button

labels. Valid values are: 0=User cannot change button labels 1=User has ability to change button labels

CONFIG_ SERVER “” (Null) Address of Avaya configuration server (currently, this

parameter, when used, is set to the PPM server address). Format is dotted decimal or DNS format, separated by commas, with no spaces (0-255 ASCII characters, including commas, optionally followed by colon and port number).

CONFIG_SERVER_ SECURE_MODE

0 Indicates whether or not secure communication via

HTTPS is required to access the configuration server. 0 = Use HTTP. 1 = Use HTTPS. 2 = Use HTTPS if the SIP transport type is TLS,

otherwise use HTTP.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

COUNTRY USA Country of operation for specific dial tone generation. COVERAGEADDR " " (Null) The URI to which call coverage is sent to in 3rd party

(non-Avaya) environments only.

CURRENT_SKIN " " (Null) Defines if a custom skin is currently selected (non-empty

string) or built-in default skin is used (empty string or not set). If a custom skin is selected (non-empty string), this value points to the corresponding skin resource definition (i.e. contains a label as defined in "SKINS" configuration parameter). Can also be set by the end user via Avaya Menu Screen & Sounds option.

DATEFORMAT %m/%d/%y Formatting string defining how to display the date in the

top line and the call log.

DAYLIGHT_SAVING_ SETTING_MODE

2 Controls daylight saving setting. Values are:

0=daylight saving time is deactivated (no offset to local time)

1=daylight saving time is activated (offset to local time as configured in "DSTOFFSET")

2=the device switches automatically to daylight saving time and back according to the contents of "DSTSTART" and "DSTSTOP"

DHCPSTD 0 DHCP Standard lease violation flag. Indicates whether to

keep the IP Address if there is no response to lease renewal. If set to “1” (No) the telephone strictly follows the DHCP standard with respect to giving up IP Addresses when the DHCP lease expires. If set to “0” (Yes) the telephone continues using the IP Address until it detects reset or a conflict (see DHCP Generic Setup

DIALPLAN " " (Null) Dial plan (in "non-PPM" format) Used to identify the end

of dialing information to accelerate dialing. Valid value is 0 to 1023 characters that define the dial plan.

DNSSRVR 0.0.0.0 Text string containing the IP Address of zero or more

DNS servers, in dotted-decimal format, separated by commas with no intervening spaces (0-255 ASCII characters, including commas).

DOMAIN " " (Null) Text string containing the domain name to be used when

DNS names in system values are resolved into IP Addresses. Valid values are 0-255 ASCII characters.

DOT1X 0 Defines the telephone's operational mode for IEEE

802.1X.Valid values are: 0 = Unicast Supplicant operation only , with PAE multicast pass-through, but without proxy Logoff.

1= Unicast Supplicant operation only, with PAE multicast pass-through and proxy Logoff.

2= Unicast or multicast Supplicant operation, without PAE multicast pass-through or proxy Logoff.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

DOT1XEAPS MD5 Specifies the EAP authentication method(s) to be used

with IEEE 802.1X. Comma-separated list of key words defining EAP methods. In SIP Release 2.0, this value is restricted to a single EAP method. V alid values are either "MD5" or "TLS".

DOT1XSTAT 1 IEEE 802.1X status. Enables/disables IEEE 802.1X

function and, if enabled, additionally defines reaction on received multicast or unicast EAPOL messages. Valid values are: 0 = Supplicant operation disabled.

1 = Supplicant operation enabled, but responds only to received unicast EAPOL messages.

2 = Supplicant operation enabled, responds to received unicast and multicast EAPOL messages.

DSCPAUD 46 Differentiated Services Code Point for audio. Values

range from 0 to 63.

DSCPSIG 34 Differentiated Services Code Point for signaling. Values

range from 0 to 63.

DSTOFFSET 1 Used for daylight saving time calculation in hours. V alues

range from 0 to 2.

DSTSTART 2Sun

Mar2L

Used to identify start date for automatic change to Daylight Saving Time. Default string length with a format of either odddmmmht or Dmmmht, where:

o = one character representing an ordinal adjective of "1" (first), "2" (second), "3" (third), "4" (fourth) or "L" (last)

ddd = 3 characters containing the English abbreviation for the day of the week

mmm = 3 characters containing the English abbreviation for the month

h = one numeric digit representing the time to make the adjustment, exactly on the hour at hAM (0h00 in military format), where valid values of h are "0" through "9"

t = one character representing the time zone relative to the adjustment where "L" is local time and U is universal time

D = one or two ASCII digits representing the date of the month from "1" or "01" to "31", or the character "L", which means the last day of the month)

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

DSTSTOP 1SunNov2L Used to identify stop date for automatic change to

Daylight Saving Time. Default string length with a format of either odddmmmht or Dmmmht, where:

o = one character representing an ordinal adjective of "1" (first), "2" (second), "3" (third), "4" (fourth) or "L" (last)

ddd = 3 characters containing the English abbreviation for the day of the week

mmm = 3 characters containing the English abbreviation for the month

h = one numeric digit representing the time to make the adjustment, exactly on the hour at hAM (0h00 in military format), where valid values of h are "0" through "9"

t = one character representing the time zone relative to the adjustment where "L" is local time and U is universal time

D = one or two ASCII digits representing the date of the month from "1" or "01" to "31", or the character "L", which means the last day of the month)

DTMF_PAYLOAD_TYPE 120 RTP dynamic payload used for RFC 2833 signaling.

Range is 96 to 127.

ENABLE_AVAYA_ ENVIRONMENT

1 Determines whether the phone operates in a mode to

comply with 3rd party standard SIP proxy (provision of SIPPING 19 feature) or the Avaya environment mode (provision of SIP/AST features and use of PPM for download and backup/restore). Valid values are: 0=Non-Avaya environment; 1=Avaya environment.

ENABLE_CALL_LOG 1 Enable or disable complete Call Log application. If

disabled no calls are logged, screens related to Call Log are not displayed to user, and menu items of User Interface to set Call Log options are not displayed. Values are 0=disabled; 1=enabled.

ENABLE_CONTACTS 1 Enable or disable complete Contact application. If

disabled no contacts are downloaded during initialization from PPM, screens related to Contacts application are not displayed to user, and menu items of the User Interface to set Contacts options are hidden. Values are 0=disabled; 1=enabled.

ENABLE_EARLY_MEDIA 1 Flag that indicates if SIP early is enabled. If enabled and

18x progress message includes early SDP, Spark uses that information to open a VoIP channel to the far-end before the call is answered. Values are 0=disabled; 1=enabled.

ENABLE_G71 1A 1 Enable or disable G711A codec cap ability of the phone. If

the parameter is set to 1, the phone includes G711A capability in an outbound INVITE request, and accepts G711A when received in an incoming INVITE request. Values are 0=disabled; 1=enabled.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

ENABLE_G711U 1 Enable or disable G71 1U codec capability of the phone. If

the parameter is set to 1, the phone includes G711U capability in an outbound INVITE request, and accepts G711U when received in an incoming INVITE request. Values are 0=disabled; 1=enabled.

ENABLE_G722 0 Enable or disable G722 capability of the telephone. If the

parameter is set to 1, the phone includes G722 capability in an outbound INVITE request, and accepts G722 when received in an incoming INVITE request. If set to 0, processing of G722 as a capability is disabled.

Values are 0=disabled, off; 1=enabled, on.

ENABLE_G726 1 Enable or disable G726 capability of the telephone. If the

parameter is set to 1, the telephone includes G726 capability in an outbound INVITE request, and accepts G726 when received in an incoming INVITE request. Values are 0=disabled, off; 1=enabled, on.

ENABLE_G729 1 Enable or disable G729A codec capability of the phone.

Values are: 0=G729A disabled 1=The phone includes G729(A) without Annex B support

capability in an outbound INVITE request, and accepts G729 when received in an incoming INVITE request.

2=The phone includes G729(A) with Annex B support capability in an outbound INVITE request, and accepts G729 when received in an incoming INVITE request.

ENABLE_MODIFY_ CONTACTS

1 Enable or disable the ability to modify contacts if the

Contact application is enabled. Values are 0=disabled; 1=enabled.

ENABLE_MULTIPLE_ CONTACTS_WARNING

1 Activate/deactivate multiple contacts warning. Depending

on current value, a warning message is displayed explaining to the user there are multiple devices registered on user's behalf and that this can cause service disruption. Values: 0 = warning disabled, 1 = warning enabled.

ENABLE_PRESENCE 0 Enable or disable complete Presence functionality. If

disabled, Presence icons do not show in Contacts or Call History Lists, Presence is not displayed to the user, incoming Presence updates are ignored, and menu items of User Interface to set Presence options are not displayed (if available). Values are 0=disabled, off; 1=enabled, on.

ENABLE_REDIAL 1 Enable or disable complete Redial functionality. If

disabled pressing the redial button has no effect and the redial softkeys and menu items are not displayed. V alues are 0=disabled; 1=enabled.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

ENABLE_REDIAL_LIST 1 Enables or disables the capability to redial out of a list of

recently dialed numbers instead of performing last number redial. V alues are 0=disabled (last number re dial only is offered to the user); 1=enabled (user can select either last number redial or redial from a list).

ENHDIALST A T 1 Enhanced Dialing St atus. V alid range is 0 to 2. If set to “0”

the feature is turned off. If set to “1” it is partially enabled (dialing rules do not apply for dialing from Contacts). If set to “2”, the Enhanced Local Dialing

feature is fully enabled (dialing rules also apply for dialing from Contacts). Note that If CTDC_SUPPORT is enabled, Enhanced Local Dialing is automatically disabled, independent of the actual setting of ENHDIALSTAT. If CTDC_SUPPORT is disabled, Enhanced Local Dialing is processed as defined by ENHDIALSTAT.

EXCHANGE_SERVER_ LIST

" " (Null) Used to connect to Microsoft Exchange™ server to

access calendar data. Zero to 255 characters: zero or more IP Addresses in dotted decimal or DNS name format, separated by commas without any intervening spaces.

EXCHANGE_USER_ DOMAIN

FAILED_SESSION_ REMOVAL_TIMER

" " (Null) String of 0 to 255 characters representing user domain

for Microsoft Exchange™ Server.

30 Timer to automatically remove a failed call session.

Range in seconds is 5 to 999.

G726_PAYLOAD_TYPE 110 RTP dynamic payload used for G.726. Range is 96

to 127.

GMTOFFSET 0:00 Offset used to calculate time from GMT reference time.

Default string length positive or negative number of hours and minutes less than 13 hours.

GROUP 0 Specific user group as tested in configuration files. Valid

values are 0 to 999.

HEADSYS 1 Headset operational mode. One ASCII numeric digit.

Valid values are: 0 or 2=General Operation, where a disconnect message

returns the telephone to an idle state. 1 or 3=Call Center Operation, where a disconnect

message does not change the state of the telephone.

HTTPDIR " " (Null) HTTP server directory path. The path name prepended to

all file names used in HTTP and HTTPS get operations during initialization. Value: 0-127 ASCII characters, no spaces. Null is a valid value. Leading or trailing slashes are not required. The command syntax is “GET HTTPDIR myhttpdir” where “myhttpdir” is your HTTP server path. HTTPDIR is the path for all HTTP operations.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

HTTPEXCEPTION DOMAINS

" " (Null) Domains to be excluded for SCEP. String representing

zero or one domains in a URL of 0 to 255 characters in dotted decimal or DNS name format with multiple domains delimited by commas.

HTTPPORT 80 Destination TCP port used for requests to the HTTP

server during initialization. Range is 0 - 65535. Note: For SIP Release 1.0, there should be no need to set this parameter to values other than default value.

HTTPPROXY " " (Null) Zero or one IP or DNS address of the HTTP server for

SCEP. 0 to 255 characters in dotted decimal or DNS name format followed by a colon and port number. The colon and port number are optional. If this parameter is not null, this (proxy) transport address is used to set up the HTTP connection as the transport protocol for SCEP.

HTTPSRVR 0.0.0.0 List of IP Address(es) or DNS Name(s) of HTTP file

server(s) used to download telephone files. HTTP server addresses can be in dotted decimal or DNS format, and must be separated by commas (0-255 ASCII characters, including commas).

ICMPDU 1 Controls whether ICMP Destination Unreachable

messages will be processed. Values are: 0=DU messages not transmitted 1= DU messages not transmitted in response to specific

events 2= DU message with code 2 will be transmitted in case of

specific events

ICMPRED 0 Controls whether ICMP Redirect messages will be

processed. Values are: 0 = Redirect messages will neither be transmitted nor

received Redirect messages will be supported 1 = Redirect messages will not be transmitted, but

received Redirect messages will be supported per RFC 1122

INTER_DIGIT_TIMEOUT 5 This is the timeout that takes place when user stops

inputting digits. The timeout is treated as digit collection completion, and when it occurs, the application sends out an invite. Range in seconds of 1 to 10.

IPADD 0.0.0.0 IP Address of the telephone. Range is 7 to 15 ASCII

characters (less than the default string length) defining one IP Address in dotted-decimal format.

L2Q 0 Requests 802.1Q tagging mode (auto/on/off). Values are:

0 = auto 1 = on 2 = off

L2QAUD 6 Layer 2 audio priority value. Range from 0 to 7. L2QSIG 6 Layer 2 signaling priority value. Range from 0 to 7.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

L2QVLAN n/a 802.1Q VLAN Identifier (0 to 4094). Null (" ") is not a valid

value and the value cannot contain spaces. This parameter is preserved in RAM which survives reset and stored to flash (as L2QVLAN_INIT) only upon successful registration. This value is initialized from L2QVLAN_INIT after power-up. This value will not be initialized from L2QVLAN_INIT after reset, but can be modified using th e ADDR craft procedure.

LANG0STAT 1 This flag defines, whether or not the built-in English is

offered to the user as selectable item in the language selection UI menu. At least one other language file must be downloaded, before "not offering" built-in English. Values are 0=not offered; 1=selectable.

LANGUAGES " " (Null) List of links to language files to be downloaded.

Substrings are delimited by commas. Maximum length is 1023 characters. Each substring shall follow one of the these naming rules:

A substring is identical to a file name without any prefix specifying the path or server: The files are downloaded from the same source as the setting file(s).

A substring can provide a prefix to the file name, which specifies the relative path ("./" for next higher directory level) from the directory the settings file(s) has been downloaded to the directory the language file shall be download.

A substring specifies the completed URL to the language file including protocol identifier ("http://" or "https://"), server and path.

LLDP_ENABLED 2 Flag to enable/disable LLDP (Link Layer Discovery

Protocol). Valid values are: 0 = disabled; the telephone will not support LLDP.

1 = enabled; the telephone will support LLDP. 2 = auto; the telephone will support LLDP, but the

transmission of LLDP frames will not begin until or unless an LLDP frame is received.

LOCAL_LOG_LEVEL 3 Numerical code of severity level. Store entrie s to the local

event log, if event occurs with a severity level whose numerical code is equal to or less than the LOCAL_LOG_LEVEL value. Values are: 0 (emergencies), 1 (alerts), 2 (critical), 3 (errors), 4 (warning), 5 (notice), 6 (informational), 7 (debug).

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

LOG_CATEGORY Comma-separated list of keywords in standard string

format representing logging categories (SW modules or functions to be included in lower level logging). Logging implementation blocks all traces at level "Warning" or lower, unless the category corresponding to a given trace is enabled. If the LOCAL_LOG_LEVEL is set to "Warning" or lower, this parameter would enable low-level traces from the adaptors or manager as indicated. Applies to all logging mechanisms (syslog and local log). Example: "ALSIP, SESSION" enables debug level traces from the ALSIP adaptor and Session manager.

LOGOS " " (Null) List of custom logo definitions used as background on

display. Each logo tuple is delimited by commas. Each logo tuple contains logo label (verbatim label displayed on the screen) and logo URL. Logo label and URL are separated from one another by a '='. String maximum of 1023 characters.

LOGSRVR " " (Null) Syslog server IP or DNS address. 0 to 255 characters:

zero or one IP Addresses in dotted decimal or DNS name format.

MEDIAENCRYPTION 9 This parameter sets the cryptosuite and session

parameters for SRTP. The parameter can have one or two of the following nine values (separated by commas without any intervening spaces):

1=aescm128-hmac80 2=aescm128-hmac32 3=aescm128-hmac80-unauth 4=aescm128-hmac32-unauth 5=aescm128-hmac80-unenc 6=aescm128-hmac32-unenc 7=aescm128-hmac80-unenc-unauth 8=aescm128-hmac32-unenc-unauth 9=none

MSGNUM " " (Null) Voice mail system telephone/extension number.

Specifies the numb er to be dialed automatically when the telephone user presses the Message button.

MTU_SIZE 1500 Maximum T ransmission Unit size. Range is 1496 or 1500

only octets.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

MUSICSRVR List of Music-on-Hold Server IP or DNS address(es).

Used to retrieve "Music on hold" for audio streams of sessions put on hold (in case of several entries first address always first, etc.). In some third-party proxy environments the SIP proxy/registrar might be different from the Music-on-Hold server. In this case, the Music-on-Hold server is set via this parameter. If both functions are provided by the same server, it is not necessary to set MUSICSRVR and the SIP proxy server is used for Music-on-Hold. Zero to 255 characters: zero or more IP addresses in dotted decimal or DNS name format, separated by commas without any intervening spaces. if operating in a non-Avaya environment, this value is set via a SET command in the settings file, otherwise the address of SIP Proxy server is used.

MWISRVR " " (Null) List of Message Waiting Indicator Event Server IP or

DNS address(es). Used to register for MWI event notifications (in case of several entries first address always first, etc.). In some third-party proxy environments the SIP proxy/registrar may be different than the MWI server. In this case, the MWI server is set via this parameter. If both functions are provided by the same server, it is not necessary to set MWISRVR. The SIP proxy server is then used for MWI indications. Zero to 255 characters: zero or more IP addresses in dotted decimal or DNS name format, separated by commas without any intervening spaces. if operating in a non-Avaya environment, this value is set via a SET command in the settings file, otherwise the address of SIP Proxy server is used.

MYCERTCAID CAIdentifier Certificate Authority Identifier. String identifying whether

the endpoints can work with another certificate authority.

MYCERTCN $SERIALNO Common name (CN) for SUBJECT in SCEP certificate

request. Values are: $SERIALNO = the phone's serial number is included as

CN parameter in the SUBJECT of a certificate request. $MACADDR = the phone's MAC address is included as

CN parameter in the SUBJECT in the certificate request.

MYCERTDN " " (Null) Common part of SUBJECT in SCEP certificate request.

String which defines the part of SUBJECT in a certificate request (including Organizational Unit, Organization, Location, State, Country), of 0 to 255 characters, starting with / and separating items with /.

MYCERTKEYLEN 1024 Private Key length in range of 1024 to 2048. MYCERTRENEW 90 Threshold to renew certificate (given as percentage of

device certificate's Validity Object). Range is 1 to 99.

MYCERTURL " " (Null) URL of SCEP server. String representing zero or one URI

starting with "http://", 0 to 255 characters.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

MYCERTWAIT 1 Flag defining phone's behavior when performing

certificate enrollment. Values are: 0=wait until a certificate or a denial is received or a

pending notification is received 1=periodical check in the background

NETMASK 0.0.0.0 IP subnet mask. Range is 7 to 15 ASCII characters

defining one IP Address in dotted-decimal format.

NO_DIGITS_TIMEOUT 20 Number of seconds of delay after going "off-hook" or

getting secondary dial tone before phone automatically plays a warning tone and does not accept dial input any longer. Range in seconds is 1 to 60.

OUTBOUND_ SUBSCRIPTION_ REQUEST_DURATION

86400 Number of seconds used in initial SUBSCRIBE

messages. This is the suggested duration value of the telephone, which might be lowered by the server, depending on the server configuration. Range is 60-31536000. Note that the default value is equal to one day and the maximum value represents one year.

PHNEMERGNUM " " (Null) The number dialed when the Emerg softkey is pressed,

or when a pop-up screen for making an emergency call is confirmed.

PHNCC 1 Telephone country code. The administered international

country code for the location by the algorithm that dials calls from the incoming Call Log or from Web pages. Range: 1-3 digits, from “1” to “999.”

PHNDPLENGTH 5 Internal extension telephone number length. Specifies

the number of digits associated with internal extension numbers by the algorithm that dials calls from the incoming Call Log or from Web pages. Range: 1 or 2 digits, from “3” to “13.”

PHNIC 011 Telephone international access code. The maximum

number of digits, if any, dialed to access public network international trunks by the algorithm that dials calls from the incoming Call Log or from Web pages. Range: 0-4 digits.

PHNLD 1 Telephone long distance access code. The digit, if any,

dialed to access public network long distance trunks. Range: 1 digit (0 to 9) or " " (Null). Needed to for "Enhanced Local Dialing Algorithm".

PHNLDLENGTH 10 Length of national telephone number. The number of

digits in the longest possible national telephone number. Range: 5 to 15. Needed to for "Enhanced Local Dialing Algorithm".

PHNOL 9 Outside line access code. The character(s) dialed,

including # and *, if any, to access public network local trunks. Range: 0-2 dialable numeric digits, including " " (Null).

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

PHNNUMOFSA " " (Null) When ENABLE_AVAYA_ENVIRONMENT=0, this value

sets the number of Session Appearances.

PHY1STAT 1 Ethernet line interface setting (1=auto-negotiate,

2=10Mbps half-duplex, 3=10Mbps full-duplex, 4=100Mbps half-duplex, 5=100Mbps full-duplex, and 6=1000Mbps full-duplex if supported by the hardware).

PHY2PRIO 0 Layer 2 priority value for frames received on or forwarded

to the secondary Ethernet interface. Set this parameter only when VLAN separation is "1" (enabled). Values are from 0-7 and correspond to the drop-down menu selection.

PHY2STAT 1 Secondary Ethernet interface setting

(0=Secondary Ethernet interface off/disabled, 1=auto-negotiate, 2=10Mbps half-duplex, 3=10Mbps full-duplex, 4=100Mbps half-duplex, 5=100Mbps full-duplex), and, for post-Release S1.0 use, 6=1000Mbps full-duplex (if supported by the hardware).

PHY2VLAN 0 VLAN identifier used by frames received on or forwarded

to the secondary Ethernet interface. Set this parameter only when VLAN separation is “1” (enabled). Value is 1-4 ASCII numeric digits from “0” to “4094.” Null is not a valid value, nor can the value contain spaces.

POE_CONS_SUPPORT 1 Flag to activate Power over Ethernet conservation mode.

Valid values are: 0 = the telephone does not support power conservation mode.

1 = the telephone indicates support of power conservation mode by transmission of LLDP frames with appropriate indication in Avaya/Extreme proprietary PoE Conservation Support Level TL V. The telephone supports power conservation mode, if requested by reception of an LLDP frame with Avaya/Extreme proprietary PoE Conservation Level Request.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

PRESENCE_SERVER " " (Null) List of Presence Server IP or DNS address(es). This

value is used to access the server for presence indications (in case of several entries first address always first, etc.). In some environments the SIP proxy/ registrar may be different than the presence server. In this case, the presence server is set via this parameter . If both functions are provided by the same server, it is not necessary to set PRESENCE_SERVER - the SIP proxy server is accessed for server-based presence indications. Zero 0 to 255 characters: zero or more IP addresses in dotted decimal or DNS name format, separated by commas without any intervening spaces. When operating in a non-Avaya environment, this value is "set via a SET command in the settings file. If this value is not set, the SIP Proxy server address is used.

When not set via settings file, this value is retrieved via PPM.

PROCPSWD 27238 Text string containing the local (dialpad) procedure

password (Null or 1-7 ASCII digits). If set, password must be entered immediately after accessing the Craft Access Code Entry screen, either during initialization or when Mute (or Contacts for the 9610) is pressed to access a craft procedure. Intended to facilitate restricted access to local procedures even when command sequences are known. Password is viewable, not hidden.

PROCSTAT 0 Controls access to local (dialpad) administrative

procedures. Values are: 0 = Full access to craft local procedures

1 = restricted access to craft local procedures

PROVIDE_EDITED_ DIALING

2 Controls whether edited dialing is allowed and whether

on-hook dialing is disabled. Valid values are: 0 = Disable edit dialing. "Dialing Options" is not displayed to the user so the user cannot change edit dialing; the telephone defaults to on-hook dialing.

1 = Disable on-hook dialing and do not display "Dialing Options" to the user so the user cannot change edit dialing; the telephone defaults to edit dialing.

2 = Display "Dialing Options" to allow user to change from on-hook to edit dialing. This is the default.

3 = Display "Dialing Options" to allow user to change from edit dialing to on-hook dialing; the telephone defaults to edit dialing.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

PROVIDE_EXCHANGE_ CALENDAR

1 Flag to define whether or not menu item(s) for MS

Exchange® Calendar integration are provided to the user. Values are 0=off, 1=on. If disabled, the menu item in Options&Settings sub-menu to select access to MS Exchange® Calendar is hidden to the user. If PROVIDE_EXCHANGE_CONTACTS is also disabled the complete sub-branch for MS Exchange® integration is hidden. Values are: 0=Off; 1=On.

PROVIDE_EXCHANGE_ CONTACTS

0 Flag to define whether or not menu item(s) for MS

Exchange® Contacts integration are provided to user. If disabled, the menu item in “Options & Settings” sub-menu to select access to MS Exchange® Contacts is hidden. If PROVIDE_EXCHANGE_CALENDAR is also disabled the complete subbranch for MS Exchange® integration is hidden. Values are: 0=Off; 1=On.

PROVIDE_ LOGOUT 1 Flag to define whether or not logout function is provided

to user. If disabled and phone is operating in user mode, hide "Logout" item in option menu. Values are: 0=off; 1=on

PROVIDE_ NETWORKINFO_ SCREEN

1 Flag to define whether or not "Network Information"

menu is provided to user. If disabled and phone is operating in user mode, hide complete "Network Information". Values are: 0=off; 1=on

PROVIDE_OPTIONS_ SCREEN

1 Flag to define whether or not "Options & Settings" menu

is provided to user. If disabled and phone is operating in user mode, hide complete "Option & Settings" menu tree. Values are: 0=off; 1=on

PROVIDE_TRANSFER_ TYPE

0 Flag to determine whether user can select a Transfer

Type (Attended/Unattended) via the Avaya A Menu Call Settings options. Applies to 3rd party environments only. V alues are: 0=user ca nnot select a transfer type, transfer type not shown; 1=user can select a transfer type, transfer type shown.

QKLOGINSTAT 0 Quick login status indicator. Specifies whether a

password must always be entered manually, when the telephone is in a "registered and inactive" state (another telephone is used to take over a primary extension e.g. SIP visiting User). Valid values are:

0 = manual password entry is mandatory. 1 = quick-login is enabled; a "quick-login" is possible by

pressing the Continue softkey on the login screen to accept the current password value.

REGISTERWAIT 3600 Number of seconds for next re-registration to SIP server.

Range in second 10 - 1 000 000 000.

ROUTER 0.0.0.0 Address(es) of default router(s) / gateway(s) in the IP

network. Range is 7-127 characters defining one or more IP Addresses in dotted decimal format, separated by commas without any intervening spaces.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

RTCPCONT 1 Enables/disables the RTCP in parallel to RTP audio

streams. Values are 0=RTCP disabled, 1=RTCP enabled.

RTCPMON " " (Null) RTCP Monitor IP or DNS address to be used as

destination for RTCP monitoring. Zero to 255 characters: zero or one IP addresses in dotted decimal or DNS name format. Note that this value is only set via SET command in settings file if operating in a NON-Avaya environment,

otherwise this value is retrieved via PPM.

RTCPMONPORT 5005 RTCP monitor port number. TCP/UDP port to be used as

destination port for RTCP monitoring. Valid range is 0-65535. Note that this value is only set via SET command in settings file if operating in a NON-Avaya environment, otherwise this value is retrieved via PPM.

RTP_PORT_LOW 5004 Specifies lower limit of a port range to be used by RTP/

RTCP or SRTP/SRTCP connections, for example, to adapt to firewall traversal policies. Values: 1024-65503.

RTP_PORT_RANGE 40 Specifies the width of the port range to be used by RTP/

RTCP or SRTP/SRTCP connections, for example, to adapt to firewall traversal policies. The upper limit is calculated by the value of RTP_PORT_LOW plus the value of RTP_PORT_RANGE, taking into consideration the overall limit of 65535. Values: 32-64511.

SCREENSA VERON 240 Number of idle time minutes after which the screen saver

is turned on. Valid values range from zero (disabled) to 999 minutes (16.65 hours).

SEND_DTMF_ TYPE 2 Defines whether DTMF tones are send in-band (regular

audio) or out-band (negotiation and transmission of DTMF according to RFC 2833, with fallback to send in-band DTMF tones, if far end does not support RFC2833). Values are 1=in-band DTMF; 2=RFC2833 procedure.

SIG 0 Parameter to allow to download during start-up the

specific configuration sets for H323 or SIP endpoints. Valid values are:

0=Default 1=H323 2=SIP

SIG_PORT_LOW 1024 Lower limit of port range for signaling to support by the

phone. Values range from 1024 to 65503.

SIG_PORT_RANGE 64511 Port range for signaling to support by the phone. Values

range from 32 to 64511.

SIP_MODE 0 Determines whether the telephone uses a proxy to

receive incoming calls or can receive calls directly from another telephone. Values are: 0=proxy mode, 1=peer-to-peer mode.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

SIP_PORT_SECURE 5061 Default SIP port (for secure message transfer via TLS).

Values range from 1024 - 65535.

SIPCONFERENCECONTI NUE

0 When the ENABLE_AVAYA_ENVIRONMENT parameter

is 0 (non-Avaya environment) and the telephone initiating the conference ends the call, the other parties will be dropped unless SIPCONFERENCECONTINUE is set to 1 (continue conference call without initiator). If this parameter is set to 0, the capability is turned off and the phone ends the conference when the initiator hangs up.

SIPDOMAIN " " (Null) SIP domain name for registration. 0 to 255 characters:

string representing domain name.

SIPPORT 5060 Default SIP port (for non-secure message transfer only).

Values range from 1024 - 65535.

SIPREGISTRAR " " (Null) List of SIP registrar server IP or DNS address(es).

Server(s) used to address SIP registrations, if operating in proxy mode. In case of several entries, the first address always first, etc. In some third-party environments the SIP proxy and SIP registrar may be different servers. In this case, the SIP registrar will be set using SIPREGISTRAR. If both functions are provided by the same server, it is not necessary to set the SIPREGISTRAR (i.e., this value will remain null). Zero to 255 characters: zero or more IP addresses in dotted decimal or DNS name format, separated by commas without any intervening spaces. Only set via SET command in settings file if you are operating in a non-Avaya environment. In an Avaya environment, this value is not applicable, because this is always identical to SIPPROXYSRVR.

SIPROXYSRVR " " (Null) SIP proxy/registrar server IP or DNS address. Zero or

one IP Address. Format is dotted decimal or DNS format, separated by commas, with no spaces (0-255 ASCII characters, including commas).

SIPSIGNAL 2 SIP signaling transport protocol. Values are:

0=UDP 1=TCP 2=TLS over TCP

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

SKINS " " (Null) Applicable to the SIP 9640 IP Telephone only.

Represents a list of skin information tuples. Each skin information is a pair of {skin label, skin URL} data. Each skin tuple is delimited by commas. Each skin tuple contains skin label (verbatim label displayed on the screen) and skin URL. Skin label and URL are separated by a ' ='. The URL may be specified in an absolute or relative path format (“./” for next higher directory level in relative path format; origin is the directory specified by HTTPDIR or TLSDIR depending on download via http or https). String maximum is 1023 characters. Example: Yankees (Color)=http://svn.avaya.com/drop/skins/ yankees_color/boohisscolor.xml

SNMPADD " " (Null) Text string containing zero or more allowable source IP

Addresses for SNMP queries, in dotted decimal or DNS format, separated by commas, with up to 255 total ASCII characters including commas and no intervening spaces.

SNMPSTRING " " (Null) Text string containing the SNMP community name string

(up to 32 ASCII characters, no spaces).

SNTPSRVR " " (Null) Used to retrieve date and time via SNTP (in case of

several entries first address always first, etc.). Zero to 255 characters: zero or more IP Addresses in dotted decimal or DNS name format, separated by commas without any intervening spaces.

SPEAKERSTAT 2 Limits the hands-free audio operation mode. Valid values

are: 0=no speakerphone allowed 1=one-way speakerphone operation allowed (monitor) 2=two-way speakerphone operation allowed

SUBSCRIBE_SECURITY 2 Controls the use of SIP and SIPS subscriptions. Valid

values are 0 - 2.

SUPPORT_GIGABIT 0 Flag indicating whether the telephone supports GigE

(Gigabit Ethernet). Valid values are: 0=Telephone does not support GigE 1=Telephone supports GigE

SYSTEM_ LANGUAGE " " (Null) System Default Language definition. String representing

a file name (shall be identical to one of the file names received via LANGUAGES parameter or null).

TCP_KEEP_ALIVE_ INTERVAL

10 Time interval (number of seconds) after which TCP

keep-alive packets are re-transmitted. The interval is started by the system TCP/IP stack (when TCP keep-alive is enabled with specified time intervals). Values are 5-60 seconds.

TCP_KEEP_ALIVE_ STATUS

1 Indicates whether TCP/IP keep-alive should be enabled

at the system. Values are 0=TCP keep alive disabled, 1=TCP keep alive enabled.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

TCP_KEEP_ALIVE_TIME 60 This time interval is the time 9600 Series SIP IP

Telephones will wait before sending out a TCP keep-alive message (TCP ACK message) to the far-end. The time is controlled by the system's TCP/IP stack. The timer is restarted after application level data (for example, a SIP message) is sent over the socket. When the system is idle, this keep-alive time expires and results in sending a TCP ACK (keep-alive) packet. Valid values are 10-3600 (seconds).

TIMEFORMAT 0 Display time according to defined format in the top line

and in the call log. Values are: 0=am/pm format 1=24h format

TLSDIR " " (Null) Path name for https downloads. Character string of 0 to

127 characters representing a directory name or path to directory.

TLSPORT 443 Destination TCP port used for requests to https server

during initialization. Values: 0-65535.

TLSSRVRID 1 Flag to indicate if TLS server identification is required.

Valid values are: 0 = no certificate match necessary; TLS/SSL connection will be established anyway.

1 = certificate match required; TLS/SSL connection will only be established if the server's identity matches the server's certificate.

TRUSTCERTS " " (Null) File names of certificates to be used for authentication.

List of file names separated by commas (0 to 1024 characters).

USE_EXCHANGE_ CALENDAR

USE_QUAD_ZEROS_ FOR_HOLD

0 Activate/deactivate usage of calendar on Microsoft

Exchange™ Server. Values are: 0=disabled, 1=enabled.

0 Flag that indicates whether a= directional attributes or

0.0.0.0 IP Address is used in the SDP to signal hold operation. 0=use “a= directional attributes”, 1=use quad zeros.

VLANSEP 1 Enables or disables VLAN separation. Controls whether

frames to/from the secondary Ethernet interface receive IEEE 802.1Q tagging treatment. The tagging treatment enables frames to be forwarded based on their tags in a manner separate from telephone frames. If tags are not changed, no tag-based forwarding is employed. Values are: 1=On/Enabled, 2= Off/Disabled. This parameter is used with several related parameters. For more information, see VLAN Separation

on page 96.

VLANTEST 60 Number of seconds to wait for a DHCPOFFER when

using a non-zero VLAN ID (1-3 ASCII digits, from “0” to “999”).

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

VU_MODE 0 Visiting User mode. Determines, if and how the

telephone supports Visiting User capabilities: 0 = Off; the telephone operates "normally" and "Visiting

User" has no essential impact for normal operation. 1 = Optional; the telephone prompts the user at

registration time if they are Visiting or Not. 2 = Forced; the telephone only allows Visiting User

registrations.

VU_TIMER 36000 To Be Determined WAIT_FOR_

REGISTRATION_TIMER

32 Time in seconds the SIP application will wait for a register

response message. If no message is received, registration is retried. Range is 1-60 (seconds).

WMLEXCEPT " " (Null) Exceptions domains for the WML browser pro xy server . If

WMLPROXY is resolved and WMLEXCEPT is null, the HTTP proxy server defined by WMLPROXY is used for all transactions of the WML browser application. If WMLEXCEPT is not null, the HTTP proxy server is only used for the URLs whose domains are not on the WMLEXCEPT list. Format is zero or more strings in DNS format, separated by commas without any intervening spaces.

WMLHOME " " (Null) Home page for WML browser. If this parameter is null,

the telephone will not display the browser option under the "A" Avaya Menu. If non-null the URL specified is retrieved via HTTP and rendered in the Web page display area, when the WML browser application is initially accessed. Value is zero or one URL.

WMLIDLETIME 10 Number of minutes of inactivity until the Web browser will

display the idle URL. When the Web idle timer reaches the number of minutes equal to this parameter, the telephone sends an HTTP GET for the URI specified by WMLIDLEURI. Valid value is 1-999. Note that the web idle timer starts only when access to the WML browser is provided by an application line under the "A" Avaya Menu and the parameter WMLIDLEURI is non-null.

WMLIDLEURI " " (Null) URL of web page displayed after idle timer expires. Note

that the web idle timer will only be started when access to the WML browser is provided by an application line under the "A" Avaya Menu and the parameter WMLIDLEURI is non-null. Value is zero or one URL.

WMLPORT 8080 TCP port number to be used to access the HTTP proxy

server by the WML browser application (if defined by WMLPROXY). Valid value is 0 - 65535.

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Table 11: 9600 Series SIP IP Telephones Customizeable System Parameters (continued)

Parameter Name Default

Description and Value Range

Value

WMLPROXY " " (Null) Address of WML proxy server. WMLPROXY is used as

the HTTP proxy server by the WML browser application. If WMLPROXY is null, or if WMLPROXY cannot be

resolved into a valid IP address, an HTTP proxy server is not used. Value is zero or one IP address in dotted decimal or DNS name format. Note that WMLPROXY defines the HTTP proxy server for WML browser application and HTTPPROXY to perform SCEP certificate enrollment.

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Note:

Note: Table 11 applies to all 9600 Series SIP IP Telephones. Certain 9600 SIP IP

Telephones might have additional, optional information that you can administer. For more information, see Chapter 8:

Administering Telephone Options.

VLAN Considerations

This section contains information on how to administer 9600 Series SIP IP Telephones to minimize registration time and maximize performance in a Virtual LAN (VLAN) environment. If your LAN environment does not include VLANs, set the system parameter L2Q to 2 (off) to ensure correct operation.

VLAN Tagging

IEEE 802.1Q tagging (VLAN) is a useful method of managing VoIP traffic in your LAN. Avaya recommends that you establish a voice VLAN, set L2QVLAN to that VLAN, and provide voice traffic with priority over other traffic. You can set VLAN tagging manually, by DHCP, or in the 46xxsettings.txt file.

If VLAN tagging is enabled (L2Q= 0 or 1), the 9600 Series SIP IP Telephones set the VLAN ID to L2QVLAN, and the VLAN priority for packets from the telephone to L2QAUD for audio packets and L2QSIG for signalling packets. The default value (6) for these parameters is the recommended value for voice traffic in IEEE 802.1D.

Regardless of the tagging setting, a 9600 Series SIP IP Telephone will always transmit packets from the telephone at absolute priority over packets from secondary Ethernet. The priority settings are useful only if the downstream equipment is administered to give the voice VLAN priority.

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VLAN Detection

The Avaya IP Telephones support automatic detection of the condition where the L2QVLAN setting is incorrect. When VLAN tagging is enabled (L2Q= 0 or 1) initially the 9600 Series SIP IP Telephone transmits DHCP messages with IEEE 802.1Q tagging and the VLAN set to L2QVLAN. The telephones will continue to do this for VLANTEST seconds.

● If the VLANTEST timer expires and L2Q=1, the telephone sets L2QVLAN=0 and transmits

DHCP messages with the default VLAN (0).

● If the VLANTEST timer expires and L2Q=0, the telephone sets L2QVLAN=0 and transmits

DHCP messages without tagging.

● If VLANTEST is 0, the timer will never expire.

Note:

Note: Regardless of the setting of L2Q, VLANTEST, or L2QVLAN, you must have

DHCP administered so that the telephone will get a response to a DHCPDISCOVER when it makes that request on the default (0) VLAN.

VLAN Considerations

After VLANTEST expires, if a 9600 Series SIP IP Telephone receives a non-zero L2QVLAN value, the telephone will release the IP Address and send DHCPDISCOVER on that VLAN. Any other release will require a manual reset before the telephone will attempt to use a VLAN on which VLANTEST has expired. See the Reset procedure in Chapter 3 of the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide.

The telephone ignores any VLAN ID administered on the Communication Manager call server.

VLAN Default Value and Priority Tagging

The system value L2QVLAN is initially set to “0” and identifies the 802.1Q VLAN Identifier. This default value indicates “priority tagging” as defined in IEEE 802.IQ Section 9.3.2.3. Priority tagging specifies that your network closet Ethernet switch automatically insert the switch port default VLAN without changing the user priority of the frame (cf. IEEE 802.1D and 802.1Q).

The VLAN ID = 0 (zero) is used to associate priority-tagged frames to the port/native VLAN of the ingress port of the switch. But some switches do not understand a VLAN ID of zero and require frames tagged with a non-zero VLAN ID.

If you do not want the default VLAN to be used for voice traffic:

● Ensure that the switch configuration lets frames tagged by the 9600 Series SIP IP

Telephone through without overwriting or removing them.

● Set the system value L2QVLAN to the VLAN ID appropriate for your voice LAN.

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Another system value you can administer is VLANTEST. VLANTEST defines the number of seconds the 9600 IP Series Telephone waits for a DHCPOFFER message when using a non-zero VLAN ID. The VLANTEST default is “60” seconds. Using VLANTEST ensures that the telephone returns to the default VLAN if an invalid VLAN ID is administered or if the phone moves to a port where the L2QVLAN value is invalid. The default value is long, allowing for the scenario that a major power interruption is causing the phones to restart. Always allow time for network routers, the DHCP servers, etc. to be returned to service. If the telephone restarts for any reason and the VLANTEST time limit expires, the telephone assumes the administered VLAN ID is invalid. The telephone then initiates registration with the default VLAN ID.

Setting VLANTEST to “0” has the special meaning of telling the phone to use a non-zero VLAN indefinitely to attempt DHCP. In other words, the telephone does not return to the default VLAN.

Note:

Note: If the telephone returns to the default VLAN but must be put back on the

L2QVLAN VLAN ID, you must Reset the telephone. See the Reset procedure in the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide.

VLAN Separation

VLAN separation is available to control priority tagging from the device on the secondary Ethernet, typically PC data. The following system parameters control VLAN separation:

● VLANSEP - enables (1) or disables (0) VLAN separation.

● PHY2VLAN - provides the VLAN ID for tagged frames received on the secondary Ethernet

interface.

● PHY2PRIO - the layer 2 priority value to be used for tagged frames received on the

secondary Ethernet interface.

Table 12

provides several VLAN separation guidelines.

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Table 12: VLAN Separation Rules

If Then

VLANSEP is “0”, OR the telephone is

not tagging frames,

Frames received on the secondary Ethernet interface will not be changed before forwarding. For example, tagging is not added or removed and the VLAN ID

OR the telephone is tagging frames with a VLAN ID equal to PHY2VLAN.

and tagged frames priority are not changed. The Ethernet switch forwarding logic determines that frames received on the Ethernet line interface are forwarded to the secondary Ethernet interface or to the telephone without regard to specific VLAN IDs or the existence of tags.

VLANSEP is “1” (On/Enabled) All tagged frames received on the secondary Ethernet

interface are changed before forwarding to make the VLAN ID equal to the PHY2VLAN value and the priority value equal to the PHY2PRIO value.

Untagged frames received on the secondary Ethernet interface are not changed before forwarding.

Tagged frames with a VLAN ID of zero (priority-tagged frames) will either be:

- forwarded without being changed (preferred), or

- changed before they are forwarded such that the VLAN ID of the forwarded frame is equal to the PHY2VLAN value and the priority value is equal to the PHY2PRIO value.

VLANSEP is “1” (On/Enabled)

AND the telephone is not tagging frames,

The Ethernet switch forwarding logic determines that frames received on the Ethernet line interface are forwarded to the secondary Ethernet interface or to

OR if the telephone is tagging frames with a

the telephone without regard to specific VLAN IDs or the existence of tags.

VLAN ID equal to PHY2VLAN,

Frames received on the secondary Ethernet interface will not be changed before forwarding. In other words, tagging is not added or removed, and the VLAN ID and priority of tagged frames is not changed.

Tagged frames received on the Ethernet line interface will only be forwarded to the secondary Ethernet interface if the VLAN ID equals PHY2VLAN.

Tagged frames received on the Ethernet line interface

VLANSEP is “1” (On/Enabled)

OR if the PHY2VLAN value is zero.

AND the telephone is tagging frames with a VLAN ID not equal to PHY2VLAN,

will only be forwarded to the telephone if the VLAN ID AND the PHY2VLAN value is not zero.

equals the VLAN ID used by the telephone.

Untagged frames will continue to be forwarded or not

forwarded as determined by the Ethernet switch

forwarding logic.

Tagged frames with a VLAN ID of zero

(priority-tagged frames) will either be:

- forwarded to the secondary Ethernet interface or the

telephone as determined by the forwarding logic of

the Ethernet switch (preferred), or

- dropped.

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DNS Addressing

The 9600 Series SIP IP Telephones support DNS addresses and dotted decimal addresses. The telephone attempts to resolve a non-ASCII-encoded dotted decimal IP Address by checking the contents of DHCP Option 6. See DHCP Generic Setup At least one address in Option 6 must be a valid, non-zero, dotted decimal address, otherwise, DNS fails. The text string for the DOMAIN system parameter (Option 15, Table 11 to the address(es) in Option 6 before the telephone attempts DNS address resolution. If Option 6 contains a list of DNS addresses, those addresses are queried in the order given if no response is received from previous addresses on the list. As an alternative to administering DNS by DHCP, you can specify the DNS server and/or Domain name in the HTTP script file. But first SET the DNSSRVR and DOMAIN values so you can use those names later in the script.

Note:

Note: Administer Options 6 and 15 appropriately with DNS servers and Domain names

respectively.

on page 56 for information.

) is appended

IEEE 802.1X

Certain 9600 Series SIP IP Telephones support the IEEE 802.1X standard for pass-through and Supplicant operation but only if the value of the configuration parameter DOT1XSTAT is “1” (the default, meaning supplicant operation is enabled, and the telephone responds only to received unicast EAPOL messages) or “2” (supplicant operation enabled, and telephone responds to received unicast and multicast EAPOL messages). If DOT1XSTAT has any other value, supplicant operation will not be supported. The system parameter DOT1X determines how the telephones handle 802.1X multicast packets and proxy logoff, as follows:

● When DOT1X = 0 (the default), the telephone forwards 802.1X multicast packets from the

Authenticator to the PC attached to the telephone and forwards multicast p ackets from the attached PC to the Authenticator (multicast pass-through). Proxy Logoff is not supported.

● When DOT1X = 1, the telephone supports the same multicast pass-through as when

DOT1X=0. Proxy Logoff is supported.

● When DOT1X = 2, the telephone forwards multicast packets from the Authenticator only to

the telephone, ignoring multicast packets from the attached PC (no multicast pass-through). Proxy Logoff is not supported.

Regardless of the DOT1X setting, the telephone always properly directs unicast packets from the Authenticator to the telephone or its attached PC, as dictated by the MAC address in the packet.

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802.1X Pass-Through and Proxy Logoff

9600 Series SIP IP Telephones with a secondary Ethernet interface support pass-through of

802.1X packets to and from an attached PC. This enables an attached PC running 802.1X supplicant software to be authenticated by an Ethernet data switch.

The SIP IP Telephones support two pass-through modes:

● pass-through and

● pass-through with proxy logoff.

The DOT1X parameter setting controls the pass-through mode. In Proxy Logoff mode (DOT1X=1), when the secondary Ethernet interface loses link integ rity, the telephone sends an

802.1X EAPOL-Logoff message on the Ethernet line interface to the data switch on behalf of the attached PC. The message alerts the switch that the device is no longer present. Proxy logoff occurs only after at least one EAPOL frame with the Port Access Entity (PAE) group multicast address as the destination MAC address was received on the secondary Ethernet interface. The destination MAC address of the proxy EAPOL-Logoff frame is the PAE group multicast address. The source MAC address of the proxy EAPOL-Logoff frame is the same as the source MAC address of the last frame received on the secondary Ethernet interface that had the PAE group multicast address as the destination MAC address.

IEEE 802.1X

Note:

Note: When DOT1X = 0 or 2, the Proxy Logoff function is not supported.

802.1X Supplicant Operation

9600 SIP IP Telephones that support Supplicant operation also support Extensible Authentication Protocol (EAP), but only with the MD5-Challenge authentication method as specified in IETF RFC 3748 [8.5-33a] or with TLS.

If an EAP method in the configuration parameter DOT1XEAPS requires the authentication of a digital certificate, the standard authentication requirements apply, including matching the TLSSRVRID with that on the certificate.

If an EAP response requires an identity or a password, the values of the DOT1XID and DOT1XPSWD parameters will be used unless a new identity and/or password has been entered by the user via an 802.1X User Input interrupt screen, in which case the new values entered by the user will be used instead. The ID and password are not overwritten by telephone software downloads. For all EAP methods, if the Supplicant is unauthenticated, an 802.1X Waiting interrupt screen is displayed when a response is transmitted, unless an 802.1X User Input interrupt screen is already being displayed.

If an EAP-Failure frame is received after transmitting a response that contains an identity or a password, an 802.1X User Input interrupt screen is displayed, unless an 802.1X User Input interrupt screen is already being displayed. If an EAP-Failure frame is received after

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transmitting a response that did not contain an identity or a password, an 802.1X Failure interrupt screen is displayed.

When a telephone is installed for the first time and 802.1x is in effect, the dynamic address process prompts the installer to enter the Supplicant identity and password. The IP telephone does not accept null value passwords. See “Dynamic Addressing Process” in the Avaya

one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide. The telephone stores 802.1X credentials when successful authentication is achieved.

Post-installation authentication attempts occur using the stored 802.1X credentials, without prompting the user for ID and password entry.

An IP telephone can support several different 802.1X authentication scenarios, depending on the capabilities of the Ethernet data switch to which it is connected. Some switches may authenticate only a single device per switch port. This is known as single-supplicant or port-based operation. These switches typically send multicast 802.1X packets to authenticating devices.

These switches support the following three scenarios:

● Standalone telephone (Telephone Only Authenticates) - When the telephone is

configured for Supplicant Mode (DOT1X=2), the telephone can support authentication from the switch.

● Telephone with attached PC (Telephone Only Authenticates) - When the telephone is

configured for Supplicant Mode (DOT1X=2), the telephone can support authentication from the switch. The attached PC in this scenario gains access to the network without being authenticated.

● Telephone with attached PC (PC Only Authenticates) - When the telephone is

configured for Pass-Through Mode or Pass-Through Mode with Logoff (DOT1X=0 or 1), an attached PC running 802.1X supplicant software can be authenticated by the data switch. The telephone in this scenario gains access to the network without being authenticated.

Some switches support authentication of multiple devices connected through a single switch port. This is known as multi-supplicant or MAC-based operation. These switches typically send unicast 802.1X packets to authenticating devices. These switches support the following two scenarios:

● Standalone telephone (Telephone Only Authenticates) - When the telephone is

configured for Supplicant Mode (DOT1X=2), the telephone can support authentication from the switch. When DOT1X is "0" or "1" the telephone is unable to authenticate with the switch.

● Telephone and PC Dual Authentication - Both the telepho ne and the conn ected PC can

support 802.1X authentication from the switch. The telephone may be configured for Pass-Through Mode or Pass-Through Mode with Logoff (DOT1X=0 or 1). The attached PC must be running 802.1X supplicant software.

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