Cisco IP phone 7960 User Manual

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Cisco SIP IP Phone 7960 Administrator Guide

Version 2.0

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408 526-4000 800 553-NETS (6387)

Fax: 408 526-4100

Customer Order Number: DOC-7810497= Text Part Number: 78-10497-02

THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.

THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PR ODUCT ARE S ET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.

The following information is for FCC compliance of Class A devices: This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant to part 15 of the FCC rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial environment. This equipment generates, uses, and can radiate radio-frequency energy and, if not installed and used in accordance with the instruction manual, may cause harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference, in w hich case users will be required to correct t he interference at their own expense.

The following information is for FCC complia nce of Cl ass B devices: The equi pment desc ribed in thi s manual generates and may rad iate radio-frequency energy. If it is not installed in accordance with Cisco’s installation instructions, it may cause interference with radio and television reception. This equipment has been tested and found to comply with the limits for a Class B digital device in accordance with the specifications in part 15 of the FCC rules. These specifications are designed to provide reasonable protection against such interference in a residential installation. However, there is no guarantee that interference will not occur in a particular installation.

Modifying the equipment wit hou t Cisc o’s written authori zatio n may res ult in the e quipm ent no lon ger comply ing with FCC re quirem ents for Class A or Class B digital devices. In that event, your right to use the equipm ent may be lim ited by FCC regulations, and you may be requi red to correct any interference to radio or television communicati ons at you r own expense.

You can determine whether your equipmen t is causing interf erence by turni ng it off. If the in terference s tops, it was probably caused by the Cisco equipment or one of its peripheral devices. If the equipment causes interference to radio or television reception, try to correct the interference by using one or more of the following measures:

• Turn the television or radio antenna until the interf erenc e stops.

• Move the equipment to one side or the other of the te levision or r adio.

• Move the equipment farther away from the television or radi o.

• Plug the equipment into an outlet that is on a different circuit from the tel evision or radio. (That is, make certain the equ ipment and the television or radio are on circuits controlled by different circuit br eakers or fuses.)

Modifications to this product not authori zed by Cisco Sys tems, Inc. could void the FCC app roval and negate your authori ty to operate the product. The Cisco implementation of TCP header compression is an adaptation of a pr ogr am d eveloped by the University of California, Berkeley (UCB) as

part of UCB’s public domain version of the UNIX operating system. All rights reserved. Copyright © 1981, Regents of the University of California. NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE

PROVIDED “AS IS” WITH ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE.

IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.

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All other brands, names, or trademark s mentioned in this do cument or Web site ar e the property of their re spective owner s. The use of t he word partner does not imply a partnership relat ionsh ip between Cis co and any oth er com pany. (001 0R)

Cisco SIP IP Phone 7960 Administrator Guide

About This Guide ix

CONTENTS

CHAPTER

Overview Who Should Use This Guide Objectives Organization Related Documentation Document Conventions Obtaining Documentation

Obtaining Technical Assistance

Product Overview

What is Session Initiation Protocol?

World Wide Web

Documentation CD-ROM Ordering Documenta tion

Cisco Connection O nline

xvi

Technical Assistance Center Documentation Feedback

1-1

xvi

xvii

1-1

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Components of SIP

SIP Clients SIP Servers

1-3

1-4

1-5

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Contents

CHAPTER

What is the Cisco SIP IP Phone 7960?

Supported Features Supported Protocols

Prerequisites

1-12

1-7

1-10

Cisco SIP IP Phone Con nections

Connecting to the Network Connecting to Power Using a Headset

1-14

1-15

The Cisco SIP IP Phon e with a Catalyst Switch

Getting Started with Your CiscoSIP IP Phone

Initialization Process Overview Installing the Cisco SIP IP Phone

Installation Task Summary Downloading Files to Your TFTP Server Configuring SI P Parameters

Configuring SI P Parameters via a TFTP Server

1-5

1-13 1-13

1-16

2-1

2-3

2-4

2-5

2-6

Manually Configuring the SIP Parameters

Configuring Network Parameters

Configuring Network Parameters via a DHCP Server Manually Configuring the Network Parameters

Connecting the Phone

Adjusting the Pl acement of the Cisco SIP Phone

Verifying Startup

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2-11

2-13

2-14

2-16

2-18

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CHAPTER

Using the Cisco SIP IP Phone Menu Interface Reading the Cisco SIP IP Phone Icons

2-22

Customizing the Cisco SIP IP Phone Ring Types Creating Dial Plans

Managing Cisco SIP IP Phones

Entering Configuration Mode

Unlocking Configuration Mode Locking Configu ration Mode

2-24

3-1

3-2

Modifying the Phone’s Network Settings Modifying the Phone’s SIP Settings

3-5

Modifying SIP Parameters via a TFTP Server

Modifying the Default SIP Configuration File Modifying the Phone-Specific SIP Confi guration File

Modifying the SIP Parameters Manually Setting the Date, Time, and Daylight Savings Time Erasing the Locally-Defined Settings

3-28

2-21

2-24

3-2

3-8

3-15

3-18

3-22

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Erasing the Local ly-Defined Networ k Settings

Erasing the Local ly-Defined SIP Settings Accessing Status Information

Viewing Status Messages

Viewing Networ k S tatistics

Viewing the Firmware Version

3-30

3-31

3-33

Upgrading the Cisco SIP IP Phone Firmware

3-29

3-33

Performing an Image Upgrade and Remote Reboot

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3-35

Contents

APPENDIX

SIP Compliance with RFC-2543 Information

SIP Functions SIP Methods SIP Responses

A-2

A-3

1xxResponse—Information Responses

2xxResponse—Successful Responses

3xxResponse—Redirection Responses

4xxResponse—Request Failure Responses

5xx Response—Server Failure R es po n se s

6xxResponse—Global Responses SIP Header Fields

A-10

SIP Session Description Protocol (SDP) Usage

SIP Call Flows

B-1

Call Flow Scenarios for Successful Calls

Gateway-to Cisco SIP IP Phone—Successful Call Setup and Disconnect

Gateway-to-Cisco SIP IP Phone—Succes sful Call Setup and Call Hold

A-1

A-4

A-5

A-10

A-12

B-2

B-3

B-7

Gateway to-Cisco SIP IP Phone—Successf ul Call Setup and Call

Transfer

B-11

Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold

Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Hold with Consultation

Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting

Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer witho ut

Consultation

B-31

Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer with

Consultation

B-35

Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding

(Unconditional)

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B-20

B-25

B-41

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Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding

(Busy)

B-44

Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (No

Answer)

B-48

APPENDIX

Cisco SIP IP Phone-to Cisco SIP IP Phone 3-Way Calling Call Flow Scenarios for Failed Calls

B-58

Gateway-to-Cisco SIP IP Phone—Called User is Busy

Gateway-to-Cisco SIP IP Phone—Called User Does Not Answer

Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error

Cisco SIP IP Phone-to-Cisco SIP IP Phone— Called User is Busy

B-52

B-58

B-60

B-63

B-66

Cisco SIP IP Phone-to-Cisco SIP IP Phone— Called User Does Not

Answer

Cisco SIP IP Phone-to-Cisco SIP IP Phone— Authentication Error

Technical Specifications

Physical and Operating Environment Specifications Cable Specifications Connections Specifications

Translated Safety Warnings

Installation Warning

B-68

B-70

C-1

C-3

D-1

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Product Disposal Warning

Lightning Activity Warning

D-2

D-3

SELV Circuit Warning (other versions available)

Circuit Breaker (15A) Warning

D-6

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vii

Contents

GLOSSARY

INDEX

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About This Guide

Overview

The Cisco Session Initiation Protocol (SIP) IP Phone 7960 Admini strator Guide provides information about how to setup, connect cables to, and configure a Cisco SIP IP phone 7960 ( here after refe rr ed to a s a C isco SIP IP phone) . Th e administrator guide also p rovides info rmation on how to con figure the network and SIP settings and change the settings and options of the Cisco SIP IP phone. The administrator guide also includes reference information such as Cisco SIP IP phone call flows and compliance information.

Who Should Use This Guide

Network engineers, system administrators, or telecommunic ation engineers should use this guide to learn the steps required to properly set up the Cisco SIP IP phone on the networ k.

The tasks described are considered to be administration-level tasks and are not intended for end-users of the phones. Many of the tasks involve configuring network settings which could affect the phone’s ability to function in the netw ork and require an un de rstanding of IP networ king and tele phony con cepts.

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Cisco SIP IP Phone 7960 Administrator Guide

Objectives

The Cisco SIP I P Pho ne 79 60 Adm inistrator Guide provides necessary information to get the Cisco SIP IP phone operational in a Voice-over-IP (VoIP) network.

It is not the intent of this administrator guide to provide information on how to implement a SIP VoIP network. For information on implementing a SIP VoIP network, refer to the documents listed in the “Related Documentation” section on page xi.

Organization

This administrator guide is divided into the following cha pters a nd a ppend ixes:

About This Guide

•

Chapter 1, “Product Overview” describes SIP and the Cisco SIP IP phone.

•

Chapter 2, “Getting Started with Your Cisco SIP IP Phone” describes how to install, connect, and con figure th e Cisco SI P IP ph one.

•

Chapter 3, “Managing Cisco SIP IP Phone s” de scribe s how to mo dify the Cisco SIP IP phone’s network and SIP settings, how to a ccess network and call status information, and how to u pgrad e the firmware.

•

Appendix A, “SIP Complianc e w ith R FC-25 43 Infor ma tion” p rovides reference information a bou t the SI P IP phone c omplian ce to RFC 25 43.

•

Appendix B, “SIP Call Flows” provides reference information about the SIP IP phone call flows.

•

Appendix C, “Technical Specifications” lists the physical and oper ating environment specifications, cable specifications, and connection specifications.

•

Appendix D, “Translated Safety Warnings” lists translated safety warnings that should be followed when installing an electrical device such as the SIP IP phone.

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About This Guide

Related Documentation

•

Voice over IP for the Cisco AS5300 Doc umen ts

Document Conventions

This docume nt u ses t he fo ll owing conventions:

•

Commands and keywords are in boldface font.

•

Arguments for which you supply values are in italic font.

•

Elements in square br ackets ([ ]) are optional.

•

Alternative keywords are grouped in braces and separated by vertical bars (for example, { x | y | z }).

•

Optional alternative keywords are grouped in brac kets and sepa ra ted by vertical bars (for examp le, [ x | y | z ] ).

•

Terminal sessions and information the system displays are in

•

Information you must enter is in

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boldface screen

Cisco SIP IP Phone 7960 Administrator Guide

font.

screen

font.

Document Conventions

About This Guide

Notes use the following conventions:

Note

Caution

Warning

Waarschuwing

Means reader take note. Notes contain helpful suggestions or references to material not covered in the publication.

Cautions use the following conventions:

Means read er be careful . In this situation, you m ight do something that could result in equipment damage or loss of data.

Warnings use the following conventions:

This warning symbol means danger. You are in a situation that could cause bodily injury. Before you work on any equipm ent, be aware of the hazards involved with electrical circuitry and be familiar with standard practices for preventing accidents. (To see translations of the warnings that appear in this publication, refer to the appendix, “Translated Safety Warnings.”)

Dit waarschuwingssymbool betekent gevaar. U verkeert in een situatie die lichamelijk letsel kan veroorzaken. Voordat u aan enige apparatuur gaat werken, dient u zich bewust te zijn van de bij elektrische schakelingen betrokken risico’s en dient u op de hoogte te zijn van standaard maatregelen om ongelukken te voorkomen. (Voor vertalingen van de waarschuwingen die in deze publicatie verschijnen, kunt u het aanhangsel “Translated Safety Warnings” (Vertalingen van veiligheidsvoorschriften) raadplegen.)

xii

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Document Conventions

Varoitus

Attention

Warnung

Tämä varoitusmerkki merkitsee vaaraa. Olet tilanteessa, joka voi johtaa ruumiinvammaan. Ennen kuin työskentelet minkään laitteiston parissa, ota selvää sähkökytkentöihin liittyvistä vaaroista ja tavanomaisista onnettomuuksien ehkäisykeinoista. (Tässä julkaisussa esiintyvien varoitusten käännökset löydät liitteestä "Translated Safety Warnings" (käännetyt turvallisuutta koskevat varoitukset).)

Ce symbole d’avertissement indique un danger. V ous vous trouvez dans une situation pouvant entraîner des blessures. Avant d’accéder à cet équipement, soyez conscient des dangers posés par les circuits électriques et familiarisez-vous avec les procédures courantes de prévention des accidents. Pour obtenir les traductions des mises en garde figurant dans cette publication, veuillez consulter l’annexe intitulée « Translated Safety Warnings » (Traduction des avis de sécurité).

Dieses Warnsymbol bedeutet Gefahr. Sie befinden sich in einer Situation, die zu einer Körperverletzung führen könnte. Bevor Sie mit der Arbeit an irgendeinem Gerät beginnen, seien Sie sich der mit elektrischen Stromkreisen verbundenen Gefahren und der Standardpraktiken zur Vermeidung von Unfällen bewußt. (Übersetzungen der in dieser Veröffentlichung enthaltenen Warnhinweise finden Sie im Anhang mit dem Titel “Translated Safety Warnings” (Übersetzung der Warnhinweise).)

Avvertenza

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Questo simbolo di avvertenza indica un pericolo. Si è in una situazione che può causare infortuni. Prima di lavorare su qualsiasi apparecchiatura, occorre conoscere i pericoli relativi ai circuiti elettrici ed essere al corrente delle pratiche standard per la prevenzione di incidenti. La traduzione delle avvertenze riportate in questa pubblicazione si trova nell’appendice, “Translated Safety Warnings” (Traduzione delle avvertenze di sicurezza).

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Document Conventions

About This Guide

Advarsel

Aviso

Advertencia

Dette varselsymbolet betyr fare. Du befinner deg i en situasjon som kan føre til personskade. Før du utfører arbeid på utstyr, må du være oppmerksom på de faremomentene som elektriske kretser innebærer, samt gjøre deg kjent med vanlig praksis når det gjelder å unngå ulykker. (Hvis du vil se oversettelser av de advarslene som finnes i denne publikasjonen, kan du se i vedlegget "Translated Safety Warnings" [Oversatte sikkerhetsadvarsler].)

Este símbolo de aviso indica perigo. Encontra-se numa situação que lhe poderá causar danos fisicos. Antes de começar a trabalhar com qualquer equipamento, familiarize-se com os perigos relacionados com circuitos eléctricos, e com quaisquer práticas comuns que possam prevenir possíveis acidentes. (Para ver as traduções dos avisos que constam desta publicação, consulte o apêndice “Translated Safety Warnings” - “Traduções dos Avisos de Segurança”).

Este símbolo de aviso significa peligro. Existe riesgo para su integridad física. Antes de manipular cualquier equipo, considerar los riesgos que entraña la corriente eléctrica y familiarizarse con los procedimientos estándar de prevención de accidentes. (Para ver traducciones de las advertencias que aparecen en esta publicación, consultar el apéndice titulado “Translated Safety Warnings.”)

xiv

Varning!

Denna varningssymbol signalerar fara. Du befinner dig i en situation som kan leda till personskada. Innan du utför arbete på någon utrustning måste du vara medveten om farorna med elkretsar och känna till vanligt förfarande för att förebygga skador . (Se förklaringar av de varningar som förekommer i d enna publikation i appendix "Translated Safety Warnings" [Översatta säkerhetsvarningar].)

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About This Guide

Obtaining Documentation

World Wide Web

You can access the most current Cisco documentation on the World Wide Web at http://www.cisco.com, http://www-china.cisco.com, or http://www-europe.cisco.com.

Documentation CD-ROM

Cisco documentation and additional literature are available in a CD-ROM package, which ships with your product. The Documentation CD-ROM is updated monthly. Therefore, it is probably more current than printed documentation. The CD-ROM package is available as a single unit or as an annual subscription.

Obtaining Do cu m e ntation

Ordering Documentation

Registered CCO users can order the Docu mentation CD-ROM and other Cisc o Product documentation throug h our onlin e Subscrip tion Se rv ices a t http://www.cisco.com/cgi-bin/subcat/kaojump.cgi.

Nonregistered CCO users can o rder do cum entation throug h a lo cal ac count representative by calling Cisco’s corporate headquarters (Califor nia, U SA) at 408 526-4000 or, in North Am erica , call 8 00 553 -NET S (6387).

Obtaining Technical Assistance

Cisco provides Cisco Connection On line (CC O) as a starting p oint for all technical assistance. Warranty or maintenance contract customers can use the Technical Assistance Center. All customers can submit technical feedback on Cisco documentation using the web, e-mail, a self-addr essed stamped response card included in many printed docs, or by sending mail to Cisco.

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Obtaining Technical Assistance

Cisco Connection Online

Cisco continues to revolutionize how business is done on the Internet. Cisco Connection Online is the foundation of a suite of interactive, networked services that provides immediate, open access to Cisco information and resources at anytime, from anywhere in the world. T his highly integrated Internet application is a powerful, easy-to-use tool for doing business with Cisc o.

CCO’s broad range of features a nd serv ice s helps cu stomers a nd partne rs to streamline business processes and improve productivity. Through CCO, you will find information about C isco and our n etworking solu tions, se rvices, and programs. In addition, you can resolve technical issues w ith online support services, download and test software packages, and order Cisco learning materials and merchandise. Valuable online skill assessment, training, and certification programs are also available.

Customers and partners can self-register on CCO to obtain additional personalized information and service s. Registered users may order prod ucts, check on the status o f an orde r and v i ew benefits specific to their rela tionshi ps with Cisco.

You can access CCO in the following ways:

About This Guide

•

WWW: ww w.cisco.com

•

Telnet: cco.cisco.com

•

Modem using standard connec tion rates and the fo llowing terminal settings: VT100 emulation; 8 data bits; no parity; and 1 stop bit.

–

From North America , c all 408 526 -8070

–

From Europe, call 33 1 64 46 40 82

You can e-mail questions about using CCO to cco-team@ cisco.com.

Technical Assistance Center

The Cisco Technical Assistance Ce nter (TAC) is available to warranty or maintenance co ntract custo mers w ho need te chnica l assi stan ce with a Cisco product that is under warranty or covered by a maintenance co ntract.

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About This Guide

Obtaining Technical Assistance

To display the TAC web site that includes links to technical support information and software upgr ades an d for req ues ting TAC support, use www.cisco.com/techsupport.

To contact by e-mail, use one of the following:

Language E-mail Address

English tac@cisco.com Hanzi (Chinese) chinese-tac@cisco.com Kanji (Japanese) japan-tac@cisco.com Hangul (Korean) korea-tac@cisco.com Spanish tac@cisco.com Thai thai-tac@cisco.com

In North America, TAC can be reached at 800 553-2447 or 408 526- 7209. For other telephone numbe rs a nd TAC e-mail addresses world wide, consu lt the following web site: http://www.cisco.com/warp/public/687/Directory/DirTAC.shtml.

Documentation Feedback

If you are reading Cisco product documentation on the World Wi de Web, you can submit technical comments electronically. Click Feedback in the toolbar and select Documentation. After you complete the form, click Submit to send it to Cisco.

You can e-mail your comments to bug-doc@c isco.com. T o submit your comments by mail, for your convenience many documents contain

a response card behind the front cover. Otherwise, you can mail your comments to the following address:

Cisco Systems, Inc. Document Resource C onnectio n 170 West Tasman Drive San Jose, CA 95134-9883

We appreciate and valu e your co mmen ts.

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Obtaining Technical Assistance

About This Guide

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CHAPTER

Product Overview

This chapter contains the following information about the Cisco SIP IP phone:

•

What is Session Initiation Protocol?, page 1-1

•

What is the Cisco SIP IP Ph one 7 960 ?, pa ge 1-5

•

Prerequisites, page 1-12

•

Cisco SIP IP Phone Connec tions, page 1-13

•

The Cisco SIP IP Phone with a Catalyst Switch, page 1-16

What is Session Initiation Protocol?

Session Initiation Protocol (SIP) is the Internet Engineering Task Force’s (IETF’ s) standard for multimedia conferencing over IP. SIP is an ASCII-based, application-layer control protocol (de fined in RFC 2543) that can be used to establish, maintain, and terminate calls between two or more end points.

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Like other VoI P protocols, SIP is designed to address the func tions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundar ies. Session management provides the ability to control the attributes of an end-to-end call.

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What is Session Initiation Protocol?

SIP provides the capabilities to:

•

Determine the location of the target end point—SIP supports address resolution, name mapping, and call redirection.

•

Determine the media capabilities of the target end point—Via Session Description Protocol (SDP), SIP determines the “lowest level” of common services between the end points. Confer ences are establishe d using only the media capabilities that can be supported by all end points.

•

Determine the availability of the target end point—If a call cannot be completed because the target end point is unavailable, SIP determines whether the called party is al read y on the ph one o r d id n ot answ er in the allotted number of rings. It then returns a message indicating why the target end point was unavailable.

•

Establish a session between the originating and target end point—If the call can be completed, SIP establishes a session between the end points. SIP also supports mid-call changes, such as the addition of another end point to the conference o r t h e ch an ging of a m ed ia ch arac t er isti c o r co de c.

•

Handle the transfer and termination of calls—SIP supports the transfer of calls from one end poin t to an other. During a call transfe r, SIP simply establishes a session between the transferee and a new end point (specified by the transferring party) and terminates the session between the transferee and the transferring party. At the end of a call, SIP terminates the sessions between all parties.

Chapter1 Product Overview

1-2

Conferences can consist of two or more users and can be establish ed using multicast or multiple unicast sessions.

Note

The term conference means an established session (or call) between two or more end points. In this documen t, the terms conference and call are used interchangeably.

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Chapter 1 Product Overvi ew

Components of SIP

SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can function in one of the following roles:

•

User agent client (UAC)—A client application that initiates the SIP request.

•

User agent serve r (UAS)—A server application t ha t c ontacts the user when a SIP request is received and that returns a response on behalf of the user.

T ypically, a SIP end point is capable of functioning as both a UAC and a UAS, b ut functions only as one or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that initiated the request.

From an architecture s tandpoint, the p hysic al com ponen ts of a SIP n etwork can also be grouped into two categories: clients and servers. Figure 1-1 illustrates the architecture of a SIP ne twork.

What is Session Initiation Protocol?

Note

In addition, the SIP servers can interact with other application services, such as Lightw eght Direc tory Acce ss Protoc ol (LDAP) servers, a database application, o r an extensible marku p lang uage (XML) application. These application services provide back-end services such as directory, authentication, and billing services.

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What is Session Initiation Protocol?

Figure 1-1 SIP Architecture

SIP User

gents (UA)

Chapter1 Product Overview

SIP Proxy and

Redirect Servers

SIP

SIP SIP

SIP Gateway

SIP Client s

PSTN

42870

RTP

Legacy PBX

SIP clients include:

•

Phones—Can act as either a UAS or UAC. Softphones (PCs that have phone capabilities installed) and Cisco SIP IP phones can initiate SIP requests and respond to reques ts .

•

Gateways—Provide call control. Gateways provide many services, the most common being a tra nslation f unction be twee n SIP c onfere ncing e ndpoin ts and other terminal type s. T his func tion include s transla tion be twee n transmission formats a nd be twee n com mu nications pr oced ure s. In addition, the gateway also translates between audio and video codecs and performs call setup and clearing on both the LAN side a nd the switched- circuit ne twork side.

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Chapter 1 Product Overvi ew

SIP Server s

What is the Cisco SIP IP Phone 7960?

SIP servers includ e:

•

Proxy server—The proxy server is an intermediate device that receives SIP requests from a client and then forwards the re quests on the client’s behalf. Basically, proxy servers receive SIP messages and for ward th em to the next SIP server in the network. Proxy servers can provide functions such as authentication, authorization, network access control, routing, reliable request retransmission, and security.

•

Redirect server—Receives SIP requests, strips out the address in the request, checks its address tables for any other addresse s that may be mapp ed to the one in the request, and then returns the results of the address mapping to the client. Basically, redirect servers provide the client with information about the next hop or hops that a message should take and then the clien t contacts the next hop server or UAS directly.

•

Registrar server—Processes requ ests from UACs for registration of their current location. Registrar servers are often co-located with a redirect or proxy server.

What is the Cisco SIP IP Phone 7960?

Cisco SIP IP phones 7960 s (he reafter r eferred to a s C isco SIP IP phon es ) are full-featured telephones that can be plugged directly into an IP network and used very much like a standard private branch exchange (PBX) telephone. The Cisco SIP IP phone is an IP telephony instrument that can be used in VoIP networks.

The Cisco SIP IP phone model terminals can attach to the existing in place data network infrastructure, via 10B aseT /100B ase T interfa ces on an Ethe rnet sw itch. When used with a voice-capable Ethernet switch (one that understa nds Type of Service [ToS] bits and can prioritize VoIP traffic), the phones eliminate the need for a traditional proprietary tele phone set and key system/PBX.

The Cisco SIP IP phone com plies with RFC 25 43.

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What is the Cisco SIP IP Phone 7960?

Figure 1-2 illustrates physical features of the Cisco SIP IP phone:

Figure 1-2 Cisco SIP IP Phone Physical Features

LCD

Chapter1 Product Overview

Line or speed dial

buttons

Footstand adjustment

Soft keys

" button

" On-screen

mode buttons Volume

buttons

1-6

Handset

Dialing

pad

•

LCD screen—Deskto p which displays infor mation about your Cisco SIP IP phone, such as the time, date, your phone number, caller ID, line/call status and the soft key tabs.

•

Line or speed dial buttons—Opens a new line or speed dials the number on the LCD screen.

•

Footstand adjustment—Adjusts the an gle of the p hone ba se.

•

Soft keys—Acti v ates the feature d escribed b y the text me ssage directl y abo ve on the LCD screen.

•

Information (i) button—Pr ovide s online hel p for sele cted ke ys or fea tures and network statistics about the active call. This feature will be available in a future rele ase .

Cisco SIP IP Phone 7960 Administrator Guide

Scroll

key

Function

toggles

38007

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Chapter 1 Product Overvi ew

•

On-screen mode buttons—Retrieves information about current settings, recent calls, available services, and voice mail messages.

•

Volume buttons—Adjusts the volume of the handset, headset, speaker, ringer and adjusts the brightness contrast settings on the LCD screen.

•

Function toggles—Includes these optio ns:

–

•

Scroll key—Enables you to move among different soft key options displayed on LCD screen.

•

Dialing pad—Press the dia l pa d buttons to d ial a p hon e num be r. Dial pad buttons work exactly like those on your existing telephone.

•

Handset—Lift the handset and press the dial pad numbers to place a call, review voice mail messages, answer a call, and so on.

Supported Features

What is the Cisco SIP IP Phone 7960?

Headset and speaker—Toggles these functions enabling you to answer

the phone using a headset or speakerphone.

Mute—Stops or resumes voice transmission.

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In addition to the physical features illustrated in Figure 1-2, the Cisco SIP IP ph one a lso provides the following:

•

An adjustable ring tone

•

A hearing-aid compatible handset

•

Headset compatibility

•

An integrated two-port Ethernet switch that allows the telephone and a computer to share a single Ethernet jack

•

A direct connection to a 10Bas eT or 10 0Ba seT Ethe rnet (RJ- 45) n etwor k (half- or full-duplex conn ections a re sup ported )

•

A large (4.25 x 3 in.) display with adjustable contrast

•

G.711 (u-law and a-law) and G.729a audio comp ression

•

IP address assignment—Dynamic Ho st Configuration Protocol (DHCP) client or manually configured via a local setup menu

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What is the Cisco SIP IP Phone 7960?

•

Ability to:

– – – – –

–

•

In-band dual-tone multifreque ncy (DTMF) sup port for touc h-tone dialing

•

Out-of-band DTMF signaling for codecs that do not tran sport the DTMF signaling correctly (for example, G.72 9 or G.729A)

•

Local or remote (using the SI P 183 R i nging m essage ) ca ll p rogre ss tone

•

AVT payload type negotiation

•

Network startup via DHC P a nd Trivial File Transfer Protocol (TFTP)

•

Dial plan support that enables automatic dialing and automatic generation of a secondary dial ton e

•

Current date and time support via Simple Network Time Protocol (SNTP) and time zone and daylight savings time supp ort

Chapter1 Product Overview

Configure Ethernet port mode and speed

Specify a TFTP boot dire ctory

Configure a label for phone ide ntification display p urpose s

Configure a name for caller identification purposes for each active line

on a phone

Configure a 12- or 24-hour user interfac e time disp lay

1-8

•

Call redirection information supp ort via the CC-Diversion head er

•

Third-party call control via d elayed media negotia tion. A de layed m edia negotiation is one where the Session Description Protocol ( SDP) informati on is not completely advertised in the initial call setup.

•

Support for endpoints specified as Fully Qualified Domain Names (FQDNs) in the SDP

•

Local directory configuration (save and rec all) a nd auto matic dial completion—Each time a call i s successfully made or recei v ed, the number is stored in a local dir ector y th at is mainta ined o n the pho ne . The maximu m number of entries is 32. Entries are aged-out based on their usage and age. The oldest entry called the least number of times is overwritten first. This feature cannot be programmed by the user, however, up to 20 entries can be “locked” (via the Locked soft key) so that they will never be deleted.

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Chapter 1 Product Overvi ew

•

What is the Cisco SIP IP Phone 7960?

Message Waiting Indication (via unsolicited NOTIFY)—Lights to indicate that a new voice m essage is in a subscriber’ s mailbox. If the subscriber listens to the message but does not save or delete the message, the light remains on. If a subscriber listens to the new message or messages, and saves or deletes them, the light goes off. The message waiting indicator is controlled by the voicemail server.

Speed dial to voicemail via the messages button Remote reset support (via the E vent header in NOTIFY messa ges) The following call options:

–

Call forward (network)—A llows the Cisco SIP IP phon e user to re qu est

forwarding service from the network (v ia a third party tool that en ables

this feature to be c onfigured). When a call is placed to the user ’s phone,

it is redirected to the appropriate forward destination by the SIP proxy

server.

–

Call hold—Allows the Cisco SI P IP phone u ser (use r A) to pla ce a ca ll

(from user B) on hold. When user A places user B on hold, the 2-way RTP

voice path between user A and user B is temporarily disconnected but the

call session is still connected. When user A takes user B off hold, the

2-way RTP voice path is reestablished.

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–

Call transfer—Allows the Cisco SIP IP phone user (user A) to transfer a

call from one user (u ser B ) to a nothe r use r (user C) . U ser A p lac es user

B on hold and calls use r C. If user C a ccepts th e transfe r, a session is

established between u ser B and use r C a nd the s ession betw een user A

and user B is terminated.

–

Three-way calling—Allows a “bridged” 3-way call. When a 3-way call

is established, the Cisco SIP IP phone through which the call is

established acts as a bridge, mixing the audio media for the other parties.

–

Do not disturb—Allows the user to instruct the system to intercept

incoming calls during specified periods of time when the user does not

want to be disturbed.

–

Multiple directory numbers—Allows the Cisco SIP IP phone to have up

to six directory numbers or lines.

–

Call waiting—Plays an audible tone to indicate that an incoming call is

waiting. The user can then put the existing call on-hold and accept the

other call. The user can alternat e betwee n the two calls .

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What is the Cisco SIP IP Phone 7960?

–

Chapter1 Product Overview

Direct number dialing—Allows users to initiate or receive a call using a

standard E.164 number format in a local, national, or international

format.

Direct URL dialing—Provides the ability to place a call using an email

address instead of a phone number.

Caller ID blocking—Allows the user to instruct the system to block their

phone number or e mail a ddr ess f rom p hone s th at have caller

identification capabilities.

Anonymous call blocking—Allows the user to ins truct the sy stem to

block any calls for which the identification is blocked.

Note

For information on how to use the standard telephony features and URL dialing, refer to the Getting Started

Cisco IP Phone 7960 a nd Quick Reference Cisco IP Phone 7960 documents that shippe d w ith the p hon e.

Supported Protocols

The Cisco SIP IP phone s upports the fo llowing standard pr otoc ols:

•

Domain Nam e Syst em (D NS) DNS is used in the Internet for translating names of network node s into

addresses. SIP uses DNS to resolve the host names of end points to IP addresses.

•

Dynamic Host Control Protocol (DHCP) DHCP is used to dynamically allocate and assign IP addresses. DHCP allows

you to move network devices from one subnet to another without administrative attention. If using DHCP , you can connect Cisco SIP IP phones to the network and become operational without having to manually assign an IP address and additional network parameters.

The Cisco SIP IP phone complies with the DHCP specifications documented in RFC 2131. By default, Cisco SIP IP phones are DHCP-enabled.

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Chapter 1 Product Overvi ew

•

What is the Cisco SIP IP Phone 7960?

Internet Control Message Protocol (ICMP) ICMP is a network layer Interne t protocol that enables hosts to send error or

control messages to other hosts. ICMP also provides other inform ation relevant to IP packet processing.

The Cisco SIP supports I CMP as it is docum en ted in RFC 792 . Internet Protocol (IP) IP is a network layer protocol that sends datagram packets between nodes on

the Internet. IP also provides features for addressing, type-of-service (ToS) specification, fragmentation and reassembly, and security.

The Cisco SIP IP phone s upports IP a s it is de fined in RFC 79 1. Real-Time Transport Protocol (RTP) RTP transports real-time data (such as voice data) over data networks. RTP

also the ability to obtain Quality of Service (QoS) information. The Cisco SIP IP phone s upports RTP as a media channel . Session Description Protoco l (SDP ) SDP is an ASCII-based protocol that describes multimedia sessions and their

related scheduling information.

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The Cisco SIP IP phone us es SD P for se ssion de scr iption.

•

Simple Network Time Protocol (SNTP) SNTP sychronizes computer clocks on an IP network. The Cisco SIP IP

phones use SNTP for their date and time support.

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Prerequisites

•

Prerequisites

For the Cisco SIP IP phone to su cce ssfully ope rate as a SIP e ndpoint in y our network, your network must meet the following requirements:

•

Chapter1 Product Overview

Trivial File Transfer Protocol (TFTP) TFTP allows files to be transferred from one computer to another over a

network. The Cisco SIP IP phone us es T FTP to d ownload configuration files and

software updates. User Datagram Pr otocol (UD P) UDP is a simple protocol that exchanges data packets witho ut

acknowledgments or guaranteed delivery . SIP can use UDP as the underlying transport protocol. If UDP is use d, r etran smissions are u sed to en sure reliability.

The Cisco SIP IP phone s upports UD P as it is d efined in RFC 76 8 for SIP signaling.

A working IP network is established.

1-12

For more informati on abou t configuri ng IP, refer to Cisco IOS IP and IP Routing Configuration Guide.

•

VoI P is con figured on you r Cisco r outers. For more information about configuring VoIP, refer to the Cisco IOS

Release 12.1 Multiservice Applications Configuration Guide for the appropriate access platfor m. For more informa tion about configurin g SIP VoIP, refer to the Enhancements to SIP for VoIP on Cisco Access Platforms.

•

VoI P ga teways are co nfigured fo r SIP.

•

A TFTP server is active and contains the latest Cisco SIP IP phone firmware image in its root directory.

•

A proxy server is active and conf ig ured to r eceive and forw ar d SIP messages.

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Chapter 1 Product Overvi ew

Cisco SIP IP Phone Connections

The Cisco SIP IP phone has connec tions for c on necting to th e data network, f or providing power to the phone, and for connecting a head set to the phone. Figure 1-3 illustrates the connections on the Cisco SIP IP phone.

Figure 1-3 Cisco SIP IP Phone Cable Connections

Cisco IP Phone 7960 (rear view)

Power

outlet

AC adapter

port

(DC48V)

(optional power

cable)

Cisco SIP IP Phone Connections

Headset

port

RJ-11 port

Connecting to the Network

The Cisco SIP IP phone has two RJ-45 ports that each support 10/100 Mbps halfor full-duplex Ethernet co nnec tions to exter nal devices—networ k port (la beled 10/100 SW) and ac cess port ( labe led 1 0/100 PC) . You can use either Category 3 or 5 cabling for 10 Mp bs con nection s, but use C ategory 5 f or 100 Mbps connections. On both the network port and access por t, use full- duplex mode to avoid collisions.

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Network port (10/100 SW)

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Access port (10/100 PC)

Handset

port

38006

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Cisco SIP IP Phone Connectio ns

Network Port (10/100 SW)

Use the network port to connect the phone to the network. You must use a straight-through cable on this port. The ph one can also obta in inline power from the Cisco Catalyst switch over this connection. See the “Connecting to Power” section on page 1-14 for de tails.

Access Port (10/100 PC)

Use the access port to connect a netw ork de vice, such as a compute r , to the phone. You must use a straight-through cable on this port.

Connecting to Power

The Cisco SIP IP phone can be powered by the following sources:

•

External power source—Optional Cisco AC adaptor and power cord for connecting to a standard wall receptacle.

•

WS-X6348-RJ45V 10/100 switchin g modu le—P rovides inline power to the Cisco SIP IP phone when connected to a Catalyst 3500, 4000, or 6000 family 10/100BaseTX switching mo dule.

Chapter1 Product Overview

1-14

This module sends p ower on p ins 1 & 2 and 3 & 6 .

•

WS-PWR-PANEL—Power patch panel provides power to the Cisco SIP IP phone which allows the Cisco SI P IP ph one to be con necte d to existing Catalyst 4000, 5000, and 6000 family 10/100BaseTX switc hing modules.

This module sends p ower on p ins 4 , 5, 7, a nd 8.

•

WS-X4148-RJ45V—48 por t 10 /100 Ether net w ith inline power modu le for the Catalyst 4006.

•

WS-X4095-PEM—VoIP DC Power Entry module for the Cata lyst 4 006.

•

WS-X4608-2PSU and WS-X46 08—External -48V DC power shelf common equipment for the Catalyst 4006 with two AC-to-DC PSUs and one empty bay for redundant option and the 110V 15A AC-to48V DC PSU redundant option for the power shelf

•

WS-C3524-PWR-XL-EN—C atalyst 3524-PWR X L switch

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Chapter 1 Product Overvi ew

Cisco SIP IP Phone Connections

Note

Only the network port (labeled 10/10 0 SW) supports inline power from the Cisco Catalyst switches.

For redundancy, you can use the Cisco AC adapter even if you are using inline power from the Cisco Catalyst switches. The Cisco SIP IP phone can share the power load being used from the inline power and external power source. If either the inline power or the external power goes down, the phone can switch entirely to the other power source.

T o use this redundancy feature you must set the inline power mode to auto on the Cisco Catalyst switch. Next, connect the un-powered Cisco SIP IP phone to the network. After the p hone powers up, c onne ct the extern al power s upply to the phone.

Using a Headset

The Cisco SIP IP phone supports a four or six-wire headset jack. Specifically , the Cisco SIP IP phone suppo rts the following Plantron ics he ad set mod els:

•

The Volume and Mute controls will also adjust volume to the earpiece and mute the speech path of the headset. The headset activation key is located on the front of the Cisco SIP IP phone.

Tristar Monaural Encore Monaur al H 91 Encore Binaural H101

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Note

When using a headset, an amplifier is not required. However, a coil cord is required to connect the headset to the headset port on the back of your Cisco IP Phone 7960. For inform ation on ordering compatible headsets and coil cords for the Cisco IP phone 7960, see http://cisco.getheadsets.com.

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Chapter1 Product Overview

The Cisco SIP IP Phone with a Catalyst Switch

To function in the IP telepho ny network, the C is co SIP IP phone must b e connected to a networking device, such as a Catalyst switch, to obtain network connectivity.

The Cisco SIP IP phone has an internal Ethernet switch, which enables it to switch traffic coming from the phone, ac cess p ort, and the n etwor k por t.

If a computer is connected to the access port, packets traveling to and from the computer and to and from the phone share the same p hysical link to the switch and the same port on the s witc h.

This configuration has these implications for the VLAN configuration on the network:

•

The current VLANs might be configured on an IP subnet basis, and additional IP addresses might not b e available to assign the ph one to a po rt so that it belongs to the same subnet as other devices (PC) connected to the same port.

•

Data traffic present on the VLAN supporting phones might reduce the quality of VoIP traffic.

1-16

Yo u can resolve these issues by isolating the voice traffic onto a separate VLAN on each of the ports connected to a phone. The switch port configured fo r connecting a phone would have separate VLANs co nfigured for carr ying:

•

Voic e tra ffic to and fr om the C isco SIP IP pho ne (aux iliary VLA N)

•

Data traffic to and from the PC connected to the switch through the access port of the Cisco SIP IP phone (na tive VLAN)

Isolating the phones on a separate, auxiliary VLAN increases the quality of the voice traffic and allows a large number of phones to be added to an existing network where ther e ar e not en ough I P add re sses .

For more information, refer to the documentation included with the Cisco C atalyst switch.

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CHAPTER

Getting Started with Your Cisco SIP IP Phone

This chapter explains the Cisco SIP IP phone initialization and the process that you should follow to install and c onnec t the Cisco SIP IP pho ne.

This chapter provides the following major sections:

•

Initialization Process Overview, page 2 -1

•

Installing the Cisco SIP IP Pho ne , pa ge 2-3

•

Verifying Startup, page 2-20

•

Using the Cisco SIP IP Phone Menu Interface, page 2-21

•

Reading the Cisco SIP I P Pho ne I cons, pa ge 2-22

•

Customizing the Cisco SIP IP Phone Ring Types, page 2-24

•

Creating Dial Plans, pa ge 2-24

Initialization Process Overview

The initialization process of the Cisco SIP IP phone is responsible for establishing network connectivity and fo r making the p hone opera tional in you r IP netwo rk.

Once you connect your phone to the network and to an electrical supply, the phone begins its initialization process.

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2-1

Initialization Process Overview

During the initialization process, the following events take place:

Chapter 2 Getting Started with Your Cisco SIPIP Phone

The stored image is loaded. The Cisco SIP IP phone has non-volatile Flash memory in which it stores the

firmware images, u ser-defined pre fer ence s, and pe rmane nt fac tory information about the phone.

During initialization, the phone runs a bootstrap loader that loads and executes the phone image stored in Flash memory.

The VLAN is configured. If the Cisco SIP IP phone is connected to a Catalyst switch, the switch notifies

the phone of the voice VLAN defined on the switch. The phone needs to know its VLAN membership before it can procee d with the DH CP request f or its IP settings (if using DHCP).

An IP address is acquired. If the Cisco SIP IP phone is using DH CP to obtain the IP settings, the phone

queries the DHCP server. If the phone is not using DHCP , then the phone will use IP settings th at are st ored in Fl ash memo ry.

The TFTP se rver is co n tac ted .

2-2

On the TFTP server is the latest Cisco SIP IP phone firmware image and the dual boot file (OS79XX.TXT) that enables the phone to automatically determine and initialize for the VoIP environment in which it is being installed.

If the phone is using the TFTP server to obtain its SIP parameters, there should also be a configuration file or files on the TFTP server that the phone will request and download. In the configuration file or files, SIP parameters that are required by the phone to operate in a SIP VoIP environment are defined. If the phone is not obtaining its SIP parameters via the TFTP server, the phone will use SIP settings that are stored in Flash memory.

The firmware vers io n is ve rified. If the phone is obtaining its SIP parameters via a TFTP server, the

configuration files are requested. If the phone deter mines that the ima ge defined in a configuration file differs from the imag e it ha s store d in Fla sh memory, it performs a firmware upgrad e.

When performing a firmware u pgr ad e, the pho ne downlo ad s the firmware image from the TFTP server, programs the image into Flash memory, and reboots.

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Chapter 2 Getting Started with Your Cisco SIP IPPhone

Installing the Cisco SIP IP Phone

This section contains in forma tion on how to install Cisc o SIP IP p hon es in your IP network. Before getting started, read over the information in this section carefully.

Installation Task Summary

To successfully install the Cisco SIP IP phone, you must complete the following tasks:

Download the required files from CCO to the TFTP server as described in the the “Downloading Files to Your TFTP Server” section on pag e 2-4.

If you are configuring SIP parameters via a TFTP server, create and store the configuration files as described in the “Co nfiguring SIP Parame ter s via a TFTP Server” section on page 2-6.

If you are using DCH P to co nfigure th e pho ne s’ n etwork s ettings, c onfigure the required network pa ra meter s o n y our D HCP s er ver as d escrib ed in th e “Configuring Network Parameters via a DHCP Server” section on page 2-14.

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Connect the phone to the network and to a power supply as described in the “Connecting the Phone” sec tion on page 2-16.

If you are not using DHCP to configure network param eters, manua lly configure the required network parameters as described in the “Manually Configuring the Network Parameter s” section on page 2-14 .

If you are not configurin g the SIP par ame ters via a TF TP server, manually configure the required parameters as described in the “Manually Confi guring the SIP Parameters” section on page 2-11.

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Chapter 2 Getting Started with Your Cisco SIPIP Phone

Installing the Cisco SIP IP Phone

Downloading Files to Your TFTP Server

Before installing the Cisco SIP IP phones, c opy the following files from CCO to the root directory of you r TFT P server.

File Description

OS79XX.TXT (Required) Enables the phone to automatica lly

determine and initialize for the VoIP environment in which it is being installed.

After downloading this file, you will need to use an ASCII editor to open it and specify the file name (without the file extension) of the image version that you plan to run on you r pho nes .

SIPDefaultGeneric.cnf (Optional) File in which to configure SIP parameters

intended for all phones. For more information on using the SIPDefault.cnf

file, see the “Creating the Default SIP Configuration File” section on page 2-7.

SIPConfigGeneric.cnf (Required) File which ca n be us ed as a temp late to

configure SIP parameters specific to a phone. When customized for a phone, this file must be renamed to the MAC address of the phone.

RINGLIST.DAT (Optional) Lists audio files that are the custom ring

type options for the phones. The audio files listed in the RINGLIST.DAT file must also be in the root directory of the TFTP server.

For more information on custom ring types, see the “Customizing the Cisco SIP IP Phone R ing Types” section on page 2-24.

P0S3xxyy.bin (where xx is the v ersion number and yy is the subver s i o n n umber)

dialplan.xml (Optional) North American example dial plan. syncinfo.xml (Optional) Controls the image version and associated

(Required) The C isco SIP IP pho ne firmware ima ge.

sync value to be used for re mo te re boo ts.

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Chapter 2 Getting Started with Your Cisco SIP IPPhone

Configuring SIP Parameters

Installing the Cisco SIP IP Phone

Note

This section describes how to configure the b asic SI P param eters that are required for the phone to operate in a SIP VoIP environment. For a complete list of the SIP parameters that you can configure, see the “Modifying the Phone’s SIP Settings” section on page 3-5.

The SIP parameter s are tho se pa rame te rs th at a Cisc o SIP I P ph one n eeds to operate in a SIP V oIP en vironment. You can configure SIP parameters via a TFTP server or you can manually configure the parameters on a phone-by-phone basis after connecting the phone s.

When the phone initializes, it loads the parameters stored in Flash memory . After loading the parameters stored in Flash memory, the phone requests the default configuration file from the T FTP se rv e r. If the def ault configuration file has been configured and stored in the root directory of the TFTP ser ver , th e phone re ads the parameters defined in the file, and stores those parameters that differ in Flash memory. The phone then requests its phone-specific configuration file. If the phone-specific configuration file has been c onfigured and pla ced on the TFTP server (in the root direc tory o r a subd irector y), the pho ne rea ds th e para meter s defined in the file and stores those parameters that differ in Flash memory.

Therefore, when co nfiguring SI P p aram eters, rem embe r the fo llowing:

•

Parameters defined in the default configuration file will override the values stored in Flash memory.

•

Parameters defined in the phone-spe cific configuration file will override the values specified in the default configuration file.

•

Parameters entered locally will be used by the phone until the next reboot (if a phone-specific configuration file exists).

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•

If you choose not to configure the phone via a TFTP server, you must manage the phone locally.

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Installing the Cisco SIP IP Phone

Configuring SIP Parameters via a TFTP Server

If you are co nfiguring SIP p ar ameters via a T FTP server, you must u se configuration files.

There are two co nfiguration files that you can use to define the SIP p aramete rs; the default configuration file (optional) and the phone-specific configuration file (required). If you choose to use a default configuration file, you must store the file in the root directory of your TFT P s er ver. Phone-specific configuration files can be stored in the root director y or in a subd irec tory in whic h all p hon e-spe cific configuration file s ar e s tore d .

Except for parameters used to defined the lines and users on a phone, all other SIP parameters can be defined in either the default configuration file or the phone-specific configuration file. However, for network control and maintenance purposes, we recommend that you define the parameters that you want to apply to all phones in the default configuration file (SIPDefault.cnf). Phone-specific parameters should only be defined via a phone- specific configuration file or manually configured. Phone-speci fic parameters should not be defined in the default configuration file.

Configuration File Guidelines

When modifying the default configuration file and creating the phone-specific configuration files, adhere to the following guidelines and requirements:

•

SIP parameters specified in the default configuration file (SIPDefault.cnf) will override those parameters stored in Flash memory. Parameters specifi ed in a phone-specific configuration file will override those stor ed in Flash memory and parameters specified in the default configuration file.

•

The name of each pho nes ’ ph one-sp eci fic configuration file is unique an d is based on the MAC address of the phone.

The format o f th e file n am e mu st be “ S IP X XXXYYYYZZZZ.cnf” w h er e XXXXYYYYZZZZ is the MAC address of the phone. The MAC address must be in uppercase and the extension, cnf , mus t be in lower ca se (for exam ple, SIP00503EFFD842.cnf).

Note

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The MAC address of a phone is identified on the middle sticker adhered to the base of the phone and ca n also be viewed on the Network Configuration menu.

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•

The default configuration file must be stored in the root directory of the TFTP server. The phone-specific configuration file can be stored in the root directory or in a subdirectory in which all pho ne-spe cific configuration files are located.

•

Each line in the configuration files must use the following format:

variable-name : value ; optional comments

•

Use colons to separa te variable na mes an d values.

•

Only one value can be associated with a variable.

•

The variab le and va lue can ha ve as muc h white space before or af ter them and can contain any char acte rs. However, if white spaces ar e need ed wi thin the value, the value must be enclose d in single or do uble qu ote s. If the value is enclosed i n quo tes , t he e nd qu ote mu s t be th e s a me a s t h e st ar t qu ot e.

•

After the value, you can include optional comments. Use the semicolon (;) and pound (#) delimiters to distinguish the comments.

•

Blank lines ar e allowed.

•

Comment lines are a llowed.

Installing the Cisco SIP IP Phone

•

Variable names are not case sensitive.

•

Only one variable can be set per line.

•

Distinguish the end of a line using <lf> or <cr><lf> .

•

The variable and value must be on the same line and cannot break the line.

•

Creating the Default SIP Configuration File

In the default configuration file (SIPDefault.cnf), we recommend that you define the SIP parameters that will be common to all of your phones such as the image_version parameter and call environment parameters (for example, will the phones be required to register with a proxy server and which codec will the phones use when initiating a call).

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By maintaining these parameters in the default conf iguration file, you can perform global changes, such as upgrad ing the image version, w ithout having to modify the phone-specific configuration file for e ach pho ne.

Before You Begin

•

Ensure that you have downloaded the SIPDefault.cnf file from CCO to the root directory of your T FTP server.

•

Review the guidelines and restrictions documented in the “Configuration File Guidelines” section on page 2-6.

•

For a complete list of the SIP parameters that you can configure, see the “Modifying the Phone’s SIP Settings” section on p age 3-5.

Procedure

Chapter 2 Getting Started with Your Cisco SIPIP Phone

Step 1

Step 2

Using an ASCII editor, open the SIPDefault.cnf file and define values for the following SIP global parameters:

•

image_version—(Required) Firmware vers ion th at the Cisco SI P IP phone should run.

Enter the name of the image ve rsion ( as it is re lea sed by Cisco). Do not enter the extension. Y ou cannot change the image version by changing the file name because the version is also built into the file header. Trying to change the image version by changing the f ile name will cause the fir mware to fa il when it compares the version in the header against the file name.

•

proxy1_address—(Required) IP address of the primary SIP proxy server that will be used by the phones. Enter this address in IP dotted-decimal notation.

•

tftp_cfg_dir—(Required if phone-specific configuration files are located in a subdirectory) Path to the TFTP s ubdir ector y in whic h phon e- specific configuration files ar e store d.

Save the file with the same file name, SIPDefault.cnf, to the root directory of your TFTP server.

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The following is an example of a SIP d efau lt co nfiguration file:

; sip default configuration file

Installing the Cisco SIP IP Phone

#Image Version image_version:P0S3

#Proxy server address proxy1_address: 192.168.1.1 ;

#Subdirectory config file location tftp_cfg_dir: /tftpboot/configs/sipphone

xxyy

;

Creating the Phone-Specific SIP Configurat ion File

In the phone-specific SIP configuration file, define the parameters that are specific to a phone such as the lines configured on a phone and the users defined for those lines.

Before You Begin

•

Review the guidelines and restrictions documented in the “Configuration File Guidelines” section on page 2-6.

•

Line paramete rs (t h ose id en ti fied as li nex) define a line on the phone. If you configure a line to use an e-mail address, that line can be called only using an e-mail address. Similarly , if you configure a line to use a number, that line can only be called using the number. Each line can have a different proxy configured.

•

For a complete list of the SIP parameters that you can configure, see the “Modifying the Phone’s SIP Settings” section on p age 3-5.

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Procedure

Chapter 2 Getting Started with Your Cisco SIPIP Phone

Step 1

Step 2

Using an ASCII editor, create a phone-specific configuration file for each phone that you plan to install. In the phone-specific configuration file, def ine values for the following SIP parameters (where x is a number 1 through 6):

•

linex_name—(Required) Number or e-mail address used when registering. When entering a num ber, enter the nu mber with out an y dashe s. F or e xam ple, enter 555-1212 as 5551 212 . When en terin g an e- mail a ddr ess , ente r the e-mail ID without the host name.

•

linex_authname—(Required when registration is enabled and the proxy server requires authentication) Name used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a v alue is not configured for the linex_authname parameter when registra tion is enabled, the default name is used. The default name is UNPROVISIONED.

•

linex_password—(Required when registration is enab led and the proxy requires authentication) Password used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a value is not configured for the linex_password parameter when registratio n is enabled, the default logical password is used. The default logical password is UNPROVISIONED.

Save the file to your TFTP server (in th e roo t dir ect ory or a subdir ector y containing all the phone-specific configuration files). Name the file “SIPXXXXYYYYZZZZ.cnf” where XXXXYYYYZZZZ is the MAC address of the phone. The MAC address must be in uppercase and the extension, cnf, must be in lower case (for example, SIP00503EFFD842.cnf) .

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The following is an example of a configuration file:

; phone-specific configuration file sample ; Line 1 phone number line1_name : 5551212

; Line 1 name for authentication with proxy server line1_authname : 5551212

; Line 1 authentication name password line1_password : password

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Manually Configuring the SIP Parameters

If you did not conf igure the SIP pa rameters via a TFT P server , yo u must manually configure them after you have connected the phone as descr ibed in the “Connecting the Phone” sec tion on page 2-16.

Before You Begin

•

Connect your phone as described in the “Conne cting the Phone” sectio n on page 2-16.

•

Unlock configuration mode as de scribe d in th e “Unlo ck ing C onfiguration Mode” section on page 3-2. By default, the SIP parame ters are locked to ensure that en d-us ers ca nnot mo dify settin gs that might affect their call capabilities.

•

Review the guidelines on using the Cisco SIP IP phone menus documented in the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-21.

•

When configuring the Prefe rred Co dec an d Out of Ba nd D T MF pa rame ter s, press the Change soft key until the option you desire is displayed and then press the Save soft key.

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Step 1 Step 2 Step 3 Step 4

•

After making your ch ang es, r elo ck c onfiguration m ode as de scr ibed in the “Locking Configuration Mode ” se ction o n p ag e 3-2.

•

For a complete list of the SIP parameters that you can configure, see the “Modifying the Phone’s SIP Settings” section on p age 3-5.

Procedure

Press the settings key. The Settings menu is displayed. Highlight SIP Configuration. The SIP Configuration menu is displa yed. Highlight Line 1 Settings. Press the Select soft key. The Line 1 Con figuration menu is d isplay ed.

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Step 5

Step 6 Step 7

Step 8

Highlight and press the Select soft key to configure the fo llowing param eters:

•

Name—(Required) Number or e-mail addre ss used when registering. Whe n entering a number, enter the number without a ny dashes. For examp le, enter 555-1212 as 5551212. When e nteri ng an e -mail ad dress, en ter th e e-m ail ID without the host name.

•

Authentication Name—(Required when registration is enabled) Name used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a val ue is not configured for the Authentication Name parameter when registration is enabled, the default name is used. The default name is SIPmacaddr ess wh ere macaddress is the MAC address of the phone.

•

Authentication Password—(Required when registration is enabled) P assword used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a value is not configured for the Authentication Password parameter when registration is enabled, the default logical password is used. The default logical password is SIPmaca ddress where macaddress is the MAC address of the phone.

•

Proxy Address—(Req uire d for the first line configured o n th e pho ne) I P address of the primary SIP proxy server that will be used by the phone. Enter this address in IP dotted-decimal notation.

Press the Back soft key to exit the Line 1 C onfiguration m enu. To configure additional lines on the p hon e, highlight th e next Line x Settings,

press the Select soft key and repeat Step 5 and Step 6. When done, p res s th e Save soft key to save your changes and exit the

SIP Configuration menu.

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Caution

When you have completed your changes, ensure that yo u lock the phone as described in the “Locking Configuration Mode” section on page 3-2.

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Configuring Network Parameters

Installing the Cisco SIP IP Phone

Note

This section describes how to configure the b asic n etwork parameters that are required fo r the phone to operate o n the networ k. For a complete list of the network parameters that you can configure, see the “Modifying the Phone’s Network Settings” section on page 3-2.

The network param eters inclu de those pa rame ters that m ust b e configured on a phone for the phone to oper ate in a n I P n etwork. You can configure the required network parameters via D HCP or m anua lly configure the m after you have connected the phone to a p ower supply.

The following parameters must be defined for your ph one to establish network connectivity:

•

Phone's IP address

•

Subnet mask

•

Default gateway for the subnet (use “0.0.0.0” if not required)

•

Domain name

•

DNS server IP address (use “0.0.0.0” if not required)

•

TFTP server IP address

When configuring the network parameters of an IP phone, adhere to the following guidelines:

•

Use 0.0.0.0 for unused IP addr esses.

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•

You can use 0.0.0.0 for the su bnet m ask on ly if the d efault gateway is also

0.0.0.0.

•

The TFTP server m us t h ave a non -zer o IP ad dre ss.

•

The default gateway must be on th e sam e sub net as the p hon e.

•

The default gateway can be 0.0.0.0 only if the TFTP or DNS server is on the same subnet as the phon e.

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Chapter 2 Getting Started with Your Cisco SIPIP Phone

Note

By default, DHCP is enabled on your phone. Bef ore you can manually con figure t he network parameters , you must disa ble DHCP after connecting your phone to a power supply.

Configuring Network Parameters via a DHCP Server

If you are using DH CP to c onfigure the ne twork p aram ete rs, c onfigure th e following DHCP options on your DHCP server before you connect your Cisco SIP IP phone:

•

dhcp option #50 (IP address)

•

dhcp option #1 (IP subnet mask)

•

dhcp option #3 (Default IP gateway)

•

dhcp option #15 (Domain name)

•

dhcp option #6 (DNS server IP address)

•

dhcp option #66 (TFTP ser ver IP addr ess)

Manually Configuring the Network Parameters

If you are not using DCHP to configure your network para meters, yo u must manually configure them.

2-14

Before You Begin

•

Connect your phone as described in the “Conne cting the Phone” sectio n on page 2-16.

•

Unlock configuration mode as de scribe d in th e “Unlo ck ing C onfiguration Mode” section on page 3-2. By default, the network parameters are locked to ensure that end-users canno t modify settin gs that might affect their network connectivity.

•

Review the guidelines on using the Cisco SIP IP phone menus documented in the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-21.

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•

When configuring a domain name:

–

Press the Number soft key if entering a numerical ID or press the Alpha

soft key to enter a name.

–

If entering letters, use the numbers on the dial pad associate d with a

particular letter. For example, the 2 key has the letters A, B, and C. F o r a

lower case “a”, press the 2 key once. To scroll through the available

letters and numbers, press the key repeatedly.

–

Press the << soft key to delete any mistakes.

•

After making your ch ang es, r elo ck c onfiguration m ode as de scr ibed in the “Locking Configuration Mode ” se ction o n p ag e 3-2.

•

For a complete list of the SIP parameters that you can configure, see the “Modifying the Phone’s Network Settings” section on p age 3-2.

Procedure

Installing the Cisco SIP IP Phone

Step 1 Step 2 Step 3 Step 4 Step 5 Step 6

Press the settings key. The Settings menu is displayed. Highlight Network Configuration. Press the Select soft key. The Network Configuration menu is displayed . Highlight DHCP Enabled. Press the No soft key. DHCP is now disabled. Highlight and configured each of the following parameters:

•

IP Address—IP addr ess of the phon e.

•

Subnet Mask—IP subnet m ask us ed by the ph one .

•

TFTP Server—IP address of the TFT P server from whic h the pho ne downloads its configuration files and firmware images.

•

Default Routers 1 through 5—IP addre ss of the default gateway used by the phone. Default Routers 2 through 5 are the IP addresses of the gateways that the phone will attempt to use as an alternate gateway if the primary gateway is not available.

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•

Domain Name—Name of the DNS domain in which the phone resides.

•

DNS Servers 1 through 5—IP address of the DNS server used by the phone to result names to IP address es. T he phone will attempt to use DNS Servers 2 through 5 if DN S Se rver 1 is u navailable.

Chapter 2 Getting Started with Your Cisco SIPIP Phone

Step 7

When done, press the Sa ve soft key . The phone programs the ne w information into Flash memory and resets.

Caution

When you have completed your changes, ensure that yo u lock the phone as described in the “Locking Configuration Mode” section on page 3-2.

Connecting the Phone

You must connect the phone to the network and to a power source before using it.

Before You Begin

•

Refer to Figure 2-1 for a graphical overvie w of the procedure s in this section.

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Figure 2-1 Cisco SIP IP Phone Cable Connections

Power

outlet

AC adapter

port

(DC48V)

(optional power

cable)

RJ-11 port

Installing the Cisco SIP IP Phone

Cisco IP Phone 7960 (rear view)

Headset

port

Handset

port

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Step 1

Step 2

Network port (10/100 SW)

Access port

(10/100 PC)

38006

Procedure

Connect a Category 3 or 5 straight-through Ethernet cable from the switch or hub to the network port on the phone .

See “Connecting to the Ne twork ” sec tion on pa ge 1-13 f or mo re inf orm ation on the network port.

Connect the handset and headset to their respective ports. See “Using a Headset” section on page 1-15 for more information on the headset

port.

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Step 3

Connect a Category 3 or 5 straight-through Ethern et cable from an othe r network device, such as a desktop computer, to the access port on the phone (o ptiona l).

See “Connecting to the Ne twork ” sec tion on pa ge 1-13 f or mo re inf orm ation on the access port.

Step 4

Connect the power plug to the Cisco AC Adapter port (optional). See “Connecting to Power” sec tion on p ag e 1-14 for more in forma tion.

Adjusting the Placement of the Cisco SIP Phone

The Cisco SIP IP phone includes an adjustable footstand. When placing the phone on a desktop surface, you can adjust the tilt height to several different angles in

7.5 degree increm ents f rom f lat to 60 d egrees. Al tern atively, you can mount the phone to the wall using the footsta nd or usin g th e option al lo ckin g ac cesso ry.

Adjusting Phone Placement on the Desktop

Adjust the footstand to the height that provides optimum view of the display and use of the buttons and keys.

To adjust the phone placement on the desktop:

Step 1 Step 2

Push in the footstand adjustment knob. Adjust the footstand to its desired height and release the knob.

Mounting the Phone to the Wall

You can mount the Cisco SIP IP phone on the wall using the footstand as a mounting bracket, or using the optional locking bracket. Use the following procedure to mount the phone on the wall using the standard footstand. T o use the optional locking bracket, refer to the Installing the Wall Mount Kit for the Cisco IP Phone docume nt.

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Before You Begin

•

Mounting the Cisco SIP IP phone on the wall requires some tools and equipment that are not pr ovided as standa rd e quipm ent.

Following are the tools and parts r equired for a typ ica l Cisco SIP IP phon e installation:

–

Screwdriver

–

Screws to secure the Cisco SIP IP phone to the wall

•

Refer to Figure 2-1 for a g raphic al overview of these proc edur es.

Procedure

Installing the Cisco SIP IP Phone

Step 1 Step 2 Step 3

Step 4

Step 5

Push in the footstand adjustment knob. Adjust the footstand so it is flat agai nst the ba ck of the phon e. Modify the handset rest so that the handset remains on the ear-piece rest when the

phone is vertically placed.

Remove the handset from the ea r-piece re st.

Locate the tab (hands et wa ll h ook) a t the base of the e ar-piece r est.

Slide this tab out, rotate it 180 degrees, and reinsert it.

Place the handset on the ear-piece rest.

Insert two screws into a wall stud, matching them to the two screw holes on the back of the footstand.

The keyholes fit standard phone jack mounts. Hang the phone on the wa ll.

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Verifying Startup

Chapter 2 Getting Started with Your Cisco SIPIP Phone

Figure 2-2 Adjusting the Footst and

Cisco IP Phone 7960 (rear view)

Footstand adjustment button raises and lowers adjustment plate

Verifying Startup

After the phone has power connected to it, the phone begins its startup process by cycling through these steps:

Adjustment plate installation screws holes (2)

Adjustment plate raises and lowers phone vertically

38036

2-20

These buttons flash on a nd off in sequenc e :

–

Headset

–

Mute

–

Speaker

The Cisco Systems, Inc. copyright displays on the LCD.

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These messages display as phone starts up:

–

Configuring VLAN—The phone is configuring the E ther net conne ction .

–

Configuring IP—The phone is contacting the DHCP server to obtain

network paramete rs an d the IP add ress of the TF TP s erver.

–

Requesting Configuration—The p hone is c on tact ing the TFTP se rver to

request its configuration files and compare firmware images.

–

Upgrading Software—Th e Upgrade Softwa re message displa ys only if

the phone has determine d that an im ag e upg ra de is re quired. Af ter

upgrading the image, the phone will automatically reboot to run the new

image.

The main LCD scr een ap pear s disp layin g:

–

Primary directory numbe r

–

Soft keys

If the phone successfully passes through these stages, it has started up p roperly.

Using the Cisco SIP IP Phone Menu Interface

As you configure your phone’s settings via the menu interface, f ollow these guidelines:

•

Select a parameter by pressing the down arrow to scroll to and highlight the parameter or by pressing the number that represents the parameter (located to the left of the parameter on the LCD).

•

During configuration, use * fo r do ts (pe riods) o r p ress the “.” soft key when available on the LCD.

•

Press Cancel during configuration to cancel all changes and exit a menu.

•

When configuring an SIP IP address or ID parame ter:

–

Press the Number soft key if entering a numerical value or press the

Alpha soft key to enter a name.

–

Use the buttons on the dial pad to enter a new value.

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Reading the Cisco SIP IP Phone Icons

–

If entering letters, use the numbers on the dial pad associate d with a

particular letter. For example, the 2 key has the letters A, B, and C. F o r a

lower case “a”, press the 2 key once. To scroll through the available

letters and numbers, press the key repeatedly.

–

Press the << soft key to delete any mistakes.

•

When configuring an networ k IP addr ess or I D pa ram eter:

–

Use the buttons on the dial pad to enter a new value.

–

Press the << soft key to delete any mistakes.

•

After editing a parameter, press the Validate soft key to save the value that you have entered and exit the Edit panel.

Reading the Cisco SIP IP Phone Icons

When using the Cisco SIP IP phone, a variety of icons can display on the phone’s LCD. Table 1 lists and de scri bes e ach icon that you mig ht see w hile usin g the Cisco SIP IP phone.

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Table 1 Cisco SIP IP Phone User Interface Icon Meanings

Icon Meanin g

The Cisco IP ph one 7 960 tha t yo u ar e using is runn ing SIP.

The line is configured for E.164 number dia ling and you can enter only numbers when placing the call.

The character “x” displayed to the right of the icon indicates that registration has failed.

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Table 1 Cisco SIP IP Phone User Interface Icon Meanings (continued)

Icon Meanin g

The line is configured for E.164 number dialing and ready for you to place the call. When a line is configured for E.164 number dialing, you can enter only number s when p lacing the call.

You can change to UR L dia lin g at any time while dialing on a line by pressing the more soft key and then the URL soft key.

The character “x” displayed to the right of the icon indicates that registration has failed.

The line is configured for URL dialing and you can enter both numbers and letters when placing the call.

The character “x” displayed to the right of the icon indicates that registration has failed.

The line is configured f or URL dia ling a nd ready fo r you to place the call. When a line is configured for URL dialing, you can enter both numb ers and l ette rs when plac ing th e call .

You can change to E.164 numbe r dialing a t any time while dialing on a line by pressing the more soft key and then the Number soft key.

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The character “x” displayed to the right of the icon indicates that registration has failed.

The Cisco SIP IP phone configuration mode is locked. When the phone is locked, the phone’s network or SIP settings cannot be modified.

The Cisco SIP IP phone configure mode is unlocked. When the phone is unlocked, the phone’s network or SIP settings can be modified.

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Customizing the Cisco SIP IP Phone Ring Types

The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By default, your ring type options will be those two choices. H owever, using the RINGLIST.DAT file, you can customize the ring types that are available to the Cisco SIP IP phone user s.

Step 1

Step 2

Step 3

Create a pulse code modulation (PCM) file of the desired ring types a nd st ore the PCM files in the root directory of y our TFTP ser ver. PCM files must contain no header information and comply with the following format guidelines:

•

8000 Hz sampling ra te

•

8 bits per sample

•

ulaw compression

Using a ASCII editor, open the RINGLIST .D AT file and for each of the ring types you are adding, specify the name as you want it to display on the Ring Type menu, press Tab, and then specify the filename of the ring type. For example, the format of a pointer in your RIN GLI ST.DAT file should appe ar simi lar to th e fo llowing:

Ring Type 1 ringer1.pcm

After defining pointers for eac h of the r ing ty pes yo u are ad ding, save your modifications and close the RINGLIST.DAT file.

Creating Dial Plans

Dial plans enable the Cisco SIP IP phon e to su pport a uto matic d ialing a nd automatic generation o f a se co ndary dia l ton e. I f a single dial plan is to be us ed for a system of phones, the dial plan is best specified in the default configuration file. However, you can create multiple dial plans and specify whic h phones are to use which dial plan by defining the dial_template parameter in the phone-specific configuration file. If one phone in a system of phones needs to use a different dial plan than the rest, you need to define the differing dial plan by specifying the dial_template parameter in that phone’s phone-specific configuration file.

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Creating Dial Plans

Note

Step 1 Step 2 Step 3

We recommend that you define the dial_template pa rameter in the default configuration file for maintenance and control purposes. Specify the dial_template parameter in a phone-specific configuration file only if that phone needs to use a different dial plan than is being used by the other phones in the same system.

When creating a dial plan, remember the following:

•

Dial plans must be in a n .xm l f orm at an d be stored on yo ur TF TP ser ver.

•

You must specify which dial plan a phone is to use by specifying the path to the dial plan in the dial_template parameter that you define in either the phone-specific configuration file or the default configuration. We recommend that the dial_template parameter be defined in the default configuration file unless a specific phone must u se a dial plan that differs from the one b eing used by other phones in the same system.

•

<DIALTEMPLATE> indicates the start of a template and </DIALTEMPLATE> indicates the end of a template

•

Rules are matched from start to finish with the longest matching rule tak en as the one to use. Matche s again st a period a re no t coun ted for the leng th to be the longest.

Using an ASCII editor, open a new file. Type <DIALTEMPLATE> to indicate the start of the dial plan template. For each of the numbering schemes that you wish to define, add the following

string to the template, each starting each on a separate line:

<TEMPLATE MATCH=”

pattern

” Timeout=”

sec

” User=”

type

” Rewrite=”

altstrng

”

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Where:

•

MATCH=”patter n” is the dial pattern to match. When entering the pattern, use a period (.) to match any character or use an asterisk (*) to match one or more characters. To have the phone generate a secondar y dia l tone w hen the part of the templ ate ma tche s, us e a comm a (,).

•

Timeout=”sec” is the number of seconds before a timeout will occur and the number will be dialed as entered by the user. To have the number dial immediately, specify 0.

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Creating Dial Plans

Chapter 2 Getting Started with Your Cisco SIPIP Phone

•

User=”type” is the either IP or Phone. Enter User=phone or User=IP to have the tag automatically added to the dialed number.

•

Rewrite=”altstrng” is the alternate string to be d ialed in stead of what the user enters.

Step 4

If desired, spec ify  at the end of each string where comment defines the type of plan (for example, Long Distance or Corporate Dial Plan).

Step 5

When completed, specify </DIALTEMPLATE> to indicate the end the dial plan template.

Step 6

Give the file a unique name specific to the dial plan it defines and save the file with an .xm l ex ten sio n to yo u TFT P se rver.

Step 7

If the dial plan applies to a specific phon e, a dd the path to the dia l plan (withou t specifying the file type of .xml) via the dial_template parameter in the phone specific configuration file. If the dial plan applies to a system of phones, add the path to the dial plan via th e dia l_tem plate par amete r in the d efau lt co nfiguration file. For more information on defining the dial_template parameter, see the “Modifying the Phone’s SIP Settings” section on p age 3-5.

The following is an example of a North American dial plan:

Example 2-1 Example of a PBX North American Dial Plan

<DIALTEMPLATE> <TEMPLATE MATCH="0" Timeout="1" User="Phone"/>  <TEMPLATE MATCH="9,011*" Timeout="6" User="Phone"/>  <TEMPLATE MATCH="9,0" Timeout="1" User="Phone"/>  <TEMPLATE MATCH="9,11" Timeout="0" User="Phone" Rewrite="9911"/>  <TEMPLATE MATCH="w!" Timeout="1" User="PHONE" Rewrite="9911"/> <!-- 911 when entered in

Alpha mode --> <TEMPLATE MATCH="9,.11" Timeout="0" User="Phone"/>

<TEMPLATE MATCH="9,101..............." Timeout="0" User="Phone"/> <!-- Long Distance

Service-->

<TEMPLATE MATCH="9,10.............." Timeout="0" User="Phone"/> <!-- Long Distance

Service--> <TEMPLATE MATCH="9,10*" Timeout="6" User="Phone"/>

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Managing Cisco SIP IP Phones

This chapter provides information on the following:

•

Entering Configuration Mode, page 3-1

•

Modifying the Phone’s Network Settings, page 3-2

•

Modifying the Phone’s SIP Settings, page 3-5

•

Setting the Date, Time, and Daylight Savings Time, page 3-22

•

Erasing the Locally -De fined Settings , page 3-28

•

Accessing Status Information, page 3-30

•

Upgrading the Cisc o SIP I P Pho ne Firmwar e , p ag e 3-33

Entering Configuration Mode

CHAPTER

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When you access the network c onfiguration infor matio n on y our Cisco SIP IP phone, you will notice that there is a padlock symbol located in the upper right corner of your LCD. By default, the network configuration information is locked. Before you can mod ify any of the netwo rk configuration pa ram eters, you m ust unlock the phone.

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Modifying the Phon e’s Network Settings

Unlocking Configuration Mode

To unlock the Cisco SIP IP p hone , pr es s ** #.

Chapter3 Managing Cisco SIP IP Phones

Note

You have activated the configuration mode for your phone. There is no indication an action has taken place.

If the Network Configuration or SI P Co nfiguration pa nel is d isplay ed , the loc k icon in the upper right corner of your LCD will change to an unlocked state. If you are located elsewhere in the Cisco SIP IP phone menu s, the next tim e you access the Network Configuration or the SIP Configuration panels, the lock icon will be displayed in an unlocked state.

The unlocked symbo l indic ates that you can modify the netwo rk an d SI P configuration settings.

Locking Configuration Mode

To lock the Cisco SIP IP phone when you are done modifying the settings, press **#.

If the Network Configuration or SI P Co nfiguration pa nel is d isplay ed , the loc k icon in the upper right corner of your LCD will change to a locked state. If you are located elsewhere in the Cisco SIP IP phone menus, the next time you access the Network Configuration or the SIP Configuration panels, the lock icon will be displayed in a locked state.

The unlocked symbo l indic ates that you can modify the netwo rk an d SI P configuration settings.

Modifying the Phone’s Network Settings

You can display and configure the network settings of a Cisco SIP IP phone. The network settings include info rmation suc h as the phone ’s DHCP server, MAC address, IP address , and doma in name .

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Before You Begin

When configuring network settings, remember the following:

•

Unlock configuration mode as de scribe d in th e “ Unlo cking C onfiguration Mode” section on page 3-2. By default, the network parameters are locked to ensure that end-users canno t modify settin gs that might affect their network connectivity.

•

Review the guidelines on using the Cisco SIP IP phone menus documented in the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-21.

•

After making your ch ang es, r elo ck c onfiguration m ode as de scr ibed in the “Locking Configuration Mode” se ction on p age 3-2.

Procedure

Modifying the Phone’s Network Settings

Step 1 Step 2 Step 3

Press the settings key. The Settings menu is displayed. Highlight Network Configuration. Press the Select soft key.The Network Configuration menu is displayed. The following network parameters are available on the Network Configuration

menu:

•

DHCP Server—IP address of the DH CP server f rom wh ich the ph one received its IP address and additional network settings. You cannot change the information in this field.

•

MAC Address—Factory-assigned unique 48-bit hexadec imal MAC address of the phone. You cannot change the information in this field.

•

Host Name—Unique host name assigned to the phone. The value in this field is always SIPmac where mac is the MAC address of the phone. You cannot change the information in this field.

•

Domain Name—Name of the DNS domain in which the phone resides.

•

IP Address—IP address of the phon e that was assigned by DHCP or locally configured. To edit this field, DHCP must be disabled.

•

Subnet Mask—IP subnet m ask us ed by the ph one . A subne t m ask p artitions the IP address into a network and a host identifier. To edit this field, DHCP must be disabled.

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Modifying the Phon e’s Network Settings

•

TFTP Server—IP address of the TFT P server from whic h the pho ne downloads its configuration files and firmware images. To edit this field, DHCP must be disabled.

•

DNS Servers 1 through 5—IP address of the DNS server used by the phone to result names to IP address es. T he phone will attempt to use DNS Servers 2 through 5 if DNS Server 1 is unavailable. To edit this field, DHCP must be disabled.

•

Operational VLAN Id—Unique identifier of the VLAN of which the phone is a member. This identifier is obtained through Cisco Discovery Protocol (CDP). You cannot change the infor ma tion in this field.

•

Admin. VLAN Id—Unique identifier of the VLAN to which the phone is attached. The value in this field is only used in non-Cisco switched networks. You can change the administrative VLAN used by the phone, however, if you have an administrative VLAN assigned on the Catalyst switch, that setting overrides any changes made o n the pho ne.

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•

Network Media Type—Ethernet port negotiation mode. Possible values are

–

Auto—Port is auto-negotiated.

–

Full-100—Port is configured to be a fu ll-dup lex, 100M B con ne ction.

–

Half-100—Port is configured to be a half -duplex, 100 MB conn ec tion.

–

Full-10—Port is configured to be a full-d uplex, 10M B co nne ctio n.

–

Half-10—Port is configured to be a half-duplex, 10MB connection.

The default is Auto.

•

DHCP Enabled—Whether the pho ne w ill use D HCP to c onfigure networ k settings (IP address, subnet mask, domain name, default router list, DNS server list, and TFTP address). Possible values for this field are Yes and No. By default, DHCP is enabled on the phone. To manually configure your IP settings, you must first disable DHCP.

•

DHCP Address Released—Wh ether the I P a ddre ss of the phone c an be released for reuse in the network. When you set this field to Yes, the phone sends a DHCP release message to the DHCP server and goes into a release state. The release state provide s e nough time to remove the pho ne fr om the

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network before the phone attemp ts to acqu ire another IP ad dress fro m the DHCP server. When moving the phone to a new network segment, you should first release the DHCP address.

•

Alternate TFTP—Whether to use an alter nate TFTP serv er . T his field e nables an administrator to specify the remote TFTP ser ver rather than the local one. Possible values for this parameter are Yes and No. The default is No. Wh en Yes is specified, the IP address in the TFTP Address parameter must be changed to the address of the alternate TFTP server.

•

Erase Configuration—Whether to erase all of the locally-defined settings on the phone and reset the values to the defaults. Selecting Yes will re-enable DHCP. For more information on erasing the local configuration, see the “Erasing the Locally-Defined Settings” section on page 3-28.

Modifying the Phone’s SIP Settings

Step 4

When done, press the Sa ve soft key . The phone programs the ne w information into Flash memory and resets.

Caution

When you have completed your changes, ensure that yo u lock the phone as described in the “Locking Configuration Mode” section on page 3-2.

Modifying the Phone’s SIP Settings

You can modify the SIP para meter s o f a Cisco SI P IP phone . When modifying SIP parameters, remember the following:

•

Parameters defined in the default configuration file will override the values stored in Flash memory.

•

Parameters defined in the phone-spe cific configuration file will override the values specified in the default configuration file.

•

Parameters entered locally will be used by the phone until the next reboot if a phone-specific configuration file exists.

•

If you choose not to configure the phone via a TFTP server, you must manage the phone locally.

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Modifying the Phone’s SIP Settings

Table 3-1 lists each of the SIP parameters that you can configure. In the Configuration column, the name of a parameter as you would specif y it in a configuration file is listed. In the menu column (SIP Configuration, Network Configuratio n , an d Ser vi ces) , th e nam e of the same par am ete r as i t wo uld appear on the user interface is listed. If NA appears for a parameter name in a menu column, it can c ann ot be defined via th at me nu.

Table 3-1 SIP Parameters Summary

Configuration File SIP Configuration Menu Network Configuration

Services Menu

anonymous_call_block NA NA An onymous Call Block autocomplete NA NA Auto-Complete

Numbers callerid_blocking NA NA Caller ID Blo ck dial_template NA NA NA dnd_control NA NA NA dst_auto_adjust NA NA NA dst_offset NA NA NA dst_start_day NA NA NA dst_start_day_of_week NA NA NA dst_start_month NA NA NA dst_start_time NA NA NA dst_start_week_of_monthNA NA NA

dst_stop_day NA NA NA dst_stop_day_of_week NA NA NA dst_stop_month NA NA NA dst_stop_time NA NA NA dst_stop_week_of_monthNA NA NA

dtmf_avt_payload NA NA NA dtmf_db_level NA NA NA dtmf_inband NA NA Do Not Disturb

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Modifying the Phone’s SIP Settings

Table 3-1 SIP Parameters Summary (continued)

Configuration File SIP Configuration Menu Network Configuration

Services Menu

dtmf_outofband O ut of Ba nd DTM F NA NA image_vers i on NA NA NA linex_authname

Authen tic a ti o n N a m e NA NA

(line1 to line6) linex_displayname

Display Name NA NA

(line1 to line6) linex_name

Name NA NA

(line1 to line6) linex_password

(line1 to line6) linex_shortname

Authen tication

NA NA

Password Shortname NA NA

(line1 to line6) messages_uri Messages URI NA NA network_media_type NA Network Media Type NA phone_label Phone Label NA NA preferred_codec Preferred Codec NA NA proxy_register Register with Proxy NA NA proxy1_address Proxy Addres s NA NA proxy1_port Proxy Port NA NA sip_invite_retx NA NA NA sip_retx NA NA NA sntp_mode NA NA NA sntp_server NA NA NA sync NA NA NA tftp_cfg_dir TFTP Directory NA NA time_format_24hr NA NA NA time_zone NA NA NA timer_invite_expires NA NA NA

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Modifying the Phone’s SIP Settings

Table 3-1 SIP Parameters Summary (continued)

Configuration File SIP Configuration Menu Network Configuration

timer_register_expires R egister Expires NA NA timer_t1 NA NA NA timer_t2 NA NA NA tos_media NA NA NA

Services Menu

Modifying SIP Parameters via a TFTP Server

If you have set up your phones to retrieve their SIP parameters via a TFTP server as described in the “Co nfiguring SIP Para meter s v ia a TFTP Se rver” sec tion on page 2-6, you can also modify yo ur SIP para meter s u sing the configuration files.

As explained in the “Co nfiguring SIP Parameters” section on page 2-5, there are two configuration files that you can use to define the SIP parameters; the default configuration file and the phone-specific configuration file. If used, the default configuration file must be stored in the ro ot di rector y of your TFT P se rver. The phone-specific configuration file can be stored in the root directory of the TFTP server or a subdirectory in which phone-specific configuration files are stored.

While not required, we re comm en d th at you use th e de fault configuration file to define values for SIP parameters that are common to all phones. Doing so will make controlling and maintaining your network an easier task . You can then define only those parame ters that a re s pec ific to a phon e in th e phone -spe cific configuration file. Phone-specific parameters shou ld o nly be defined in a phone-specific configuration file or manually configured. Phon e-spe cific parameters should not be d efined in the d efault configuration file.

Modifying the Default SIP Configuration File

In the default configuration file (SIPDefault.cnf), we recommend that you maintain the SIP parameters that are common to all of your phones.

By maintaining these parameters in the default configuration file, you can perform global changes, such as upgrad ing the image version, w ithout having to modify the phone-specific configuration file for e ach pho ne.

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Before You Begin

•

Ensure that you have downloaded the SIPDefault.cnf file from CCO to the root directory of your T FTP server.

•

Review the guidelines and restrictions documented in the “Configuration File Guidelines” section on pa ge 2-6.

Procedure

Modifying the Phone’s SIP Settings

Step 1

Using an ASCII editor, open the SIPDefault.cnf file and define or modify values for the following SIP parameter s as nece ssary:

•

image_version—(Required) Firmware vers ion th at the Cisco SI P IP phone should run.

Enter the name of the image version (as it is release by Cisco). Do not enter the extension. Y ou cannot change the image version by changing the file name because the version is also built into the file header. Trying to change the image version by changing the f ile name will cause the fir mware to fa il when it compares the version in the header against the file name.

•

proxy1_address—(Required) IP address of the primary SIP proxy server that will be used by the phones. Enter this address in IP dotted-decimal notation.

•

proxy1_port—(Optional) Port numbe r of the pr imar y SIP prox y server. This is the port on which the SIP client will listen for messages. The default is 5060.

•

tos_media—(Op tiona l) Type of Service (ToS) level for the media stream being used. Valid values are:

–

0 (IP_ROUTINE)

–

1 (IP_PRIORITY)

–

2 (IP_IMMEDIATE)

–

3 (IP_FLASH)

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–

4 (IP_OVERIDE)

–

5 (IP_CRITIC)

The default is 5.

•

preferred_codec—(Optional) CODEC to use when initiating a call. Valid values are g711alaw, g711ulaw, and g729a. The defa ult is g711 ulaw.

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Modifying the Phone’s SIP Settings

•

dtmf_inband—(Optional) Wh ether to detect and ge nerate in -band signaling format. Valid values are 1 (generate DTMF digits in-band) and 0 (do not generate DTMF digits in-band). The default is 1.

•

dtmf_db_level—(Optional) In-band DTMF digit tone level. Valid values are:

– – – – –

The default is 3.

•

dtmf_outofband—(Optional) Wh ether to generate the out-of-band signa ling (for tone detection o n the IP s ide of a ga teway) and if so , wh en . Th e Cisco SIP IP phone supports out-of-bound signaling via the AVT tone method. Valid values are:

– –

Chapter3 Managing Cisco SIP IP Phones

1 (6 db below nominal) 2 (3 db below nominal) 3 (nominal) 4 (3 db above nominal) 5 (6 db above nominal)

none—Do not generate D TMF digi ts out-of -band . avt—If requested by the remote side, generate DTMF digits out-of-band

(and disable in-band DTMF signaling), otherwise, do no t generate DTMF digits out-of-band.

3-10

–

avt_always—Always generate DTMF digits out-of-band . This optio n disables in-band DTMF signaling.

The default is avt.

•

dtmf_avt_payload—(Optional) Payload type for A VT packets. Possible range is 96 to 127. If the value specified exceeds 127, the phone will default to 101.

•

timer_t1—(Optional) Lowest value (in milliseconds) of the retransmission timer for SIP messages. The valid value is any positive integer. The default is 500.

•

timer_t2—(Optional) Highest value (in milliseconds) of the retransmission timer for SIP messages. The valid value is any positive integer greater than timer_t1. The default is 4000 .

•

timer_invite_expires—(Optional) The amount of time, in seconds, after which a SIP INVITE will e xpire. This va lue is used in the Expire header field. The valid value is any positiv e number, howe ver , we recommend 180 seconds. The default is 180.

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•

sip_retx—(Optional) Maximum number of times a SIP message other than an INVITE request will be retransmitted. The v alid v alue is any positi v e integer. The default is 10.

•

sip_invite_retx—(Optional) Maximum number o f times an I NVI TE r eque st will be retransmitted. The valid value is any positive inte ger. The default is 6.

•

proxy_register—(Optional) Whethe r the phone must register with a proxy server during initialization. Valid values are 0 and 1. Specify 0 to disable registration during initialization. Specify 1 to enable registration during initialization. The default is 0.

After a phone has initialized and registered with a proxy serv er, changing the value of this parameter to 0 will unregister the phone from the proxy server. To reinitiate a registration, change the value of this parameter back to 1.

Modifying the Phone’s SIP Settings

Note

If you enable registration, and authentication is required, you must sp ecify values for the linex_authname and linex_password parameters (where x is a num ber 1 throug h 6 ) in the phon e- spec ific configuration file. For information on configuring the phone-specific configuration file, see th e “M odif ying the Phone-Specific SIP Configuration File” section on page 3-15.

•

timer_register_expires—(Optional) The am ount of time , in se cond s, after which a REGISTRATION request will expire. This value is inserted into the Expire header field. The valid value is any positive number, however, we recommend 3600 sec onds. The default is 3600.

•

messages_uri—(Optional) Number to call to check voicemail. This number will be called when the Messages key is pressed.

•

Date, Time, and Daylight Savings Time parameters. See the “Setting the Date, Time, and Daylight Savings Time” section on page 3-22 section fo r more information on setting the following parameters:

–

sntp_mode—(Optional) Mode in which the phone will listen for the SNTP server.

–

sntp_server—(Optional) IP address of the SNTP server from which the phone will obtain time data.

–

time_zone—(Optional) Time zone in which the phone is located.

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–

– – –

–

– – – –

–

Chapter3 Managing Cisco SIP IP Phones

dst_offset—(Optional) Offset from the phone’s time when DST is in effect.

dst_start_month—(Optional) Month in w hich DST s tarts. dst_start_day—(Optional) Day of the month on which DST begins. dst_start_day_of_week—( Optiona l) Da y of the we ek on w hich DST

begins. dst_start_week_of_month—( Optio nal) Week of month in which D ST

begins. dst_start_time—(Optional) Time of day on which DST begins. dst_stop_month—(Optional) Mon th in which D ST ends . dst_stop_day—(Optional) Day of the m onth on w hich DS T e nds. dst_stop_day_of_week—( Optional) Day of the week on whic h DST

ends. dst_stop_week_of_month—(Op tional) Week of month in which DST

ends. dst_stop_time—(Optional) Time of day on which DST ends.

3-12

–

dst_auto_adjust—(Optional) Whether or not DST is automatically adjusted on the pho nes.

•

dnd_control—(Optional) Whethe r the Do Not Di sturb feature is enabled or disabled by default on the ph one o r wh eth er the fea tur e is pe rm ane ntly enabled. When the featu re is permanently en abled, a phone is a “call out” phone only. When the Do Not Dis turb fea ture is tu rned on , th e pho ne will block all calls placed to the phone and log those calls in the Missed Calls directory. Valid values are:

–

0—The Do Not Disturb f eat ure is off by default, but can be tur n o n an d off locally via the phone’s user interface.

–

1—The Do Not Disturb feature is on by default, but can be turned on and off locally via the phone’s user interface.

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–

2—The Do Not Disturb fe atur e is off perman en tly and canno t be tur ned on and off locally via the phone’ s user interf ace. If specifying this value, specify this parameter in the phone-specific configuration file.

–

3—The Do Not Disturb feature is on perman ently and c annot be turned on and off locally via the phone’s user interface. This setting sets the phone to be a “call out” phone only. If specif y ing this value, specify this parameter in the phone-specific co nfiguration file.

•

callerid_blocking—(Optional) Whether the Caller ID Blocking feature is enabled or disabled by default on the phon e. W hen e nable d, the p hon e w ill block its number or email address from phones that have caller identification capabilities. Valid values are:

–

0—The Caller ID Blocking fea ture is disa bled by de fault, but can be turned on and off via the phone’s user interface. When disabled, the caller identification is included in the Request-URI header field.

–

1—The Caller ID Blocking feature is enabled by default, but can be turned on and off via the ph one’s user interface . When en ab led, “Anonymous” is included in plac e of the user id en tification in th e Request-URI header field.

Modifying the Phone’s SIP Settings

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–

2—The Caller ID Blocking fea ture is disa bled p erma nently and c annot be turned on and off locally via the phone’s user interface. If specifying this value, specify this parameter in the phone-specific configuration file.

–

3—The Caller ID Blocking feature is enabled permanently and cannot be turned on and off locally via the phone’s user interface. If specifying this value, specify this parameter in the phone-specific configuration file.

•

anonymous_call_block—(Optional) Whet her the Anonymous Call Block feature is enabled or disabled by default on the phone. Valid values are:

–

0—The Anonymous Call Blocking feature is disabled by default, but can be turned on and off via the phone’s user interface. When disabled, anonymous calls will be received.

–

1—The Anonymous Call Blocking features is enabled by default, but can be turned on and off via the phone’s user interface. When enabled, anonymous calls will be rejected

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Modifying the Phone’s SIP Settings

–

•

tftp_cfg_dir—(Required if phone-specific configuration files are located in a subdirectory) Path to the TFTP s ubdir ector y in whic h phon e- specific configuration files ar e store d.

•

network_media_type—(Op tional) Et hernet port negotiation mo de. Valid values are:

– – – –

Chapter3 Managing Cisco SIP IP Phones

2—The Anonymous Call Blocking feature is disab led perman ently and cannot be turned on and off loca lly via th e pho ne’s user interface . If specifying this value, specify this parameter in the phone-specific configuration file.

3—The Anonymous Call Blocking feature is enabled permane ntly and cannot be turned on and off loca lly via th e pho ne’s user interface . If specifying this value, specify this parameter in the phone-specific configuration file.

Auto—Port is auto-negotiated. Full100—Port is configured to be a full-d uplex, 100 MB conn ection. Half100—Port is configured to be a h alf-dup lex, 1 00MB co nnection . Full10—Port is configured to be a full-duplex, 10MB connection.

3-14

–

Half10—Port is configured to be a half-duplex, 10MB conn ection.

The default is Auto.

•

autocomplete—(Optional) Whe ther to have numbers auto matically completed when dialing. Valid values are 0 (disable auto completion) or 1 (enable auto completion). The default is 1.

•

sync—Value against which to compare the value in the syncinfo.xml before performing a remote re boot. Valid value is a character string up to 32 characters lo ng.

•

time_format_24hr—Whether a 12 or 24 -hour time form at is displayed by default on the phones’ user interface. Valid values are:

–

0—The 12-hour fo rm at is displa yed by default but ca n b e c hange d to a 24-hour format v ia the pho ne ’s user interface.

–

1—The 24-hour fo rm at is displa yed by default but ca n b e c hange d to a 12-hour format v ia the pho ne ’s user interface.

–

3—The 12-hour format is displayed and cannot be changed to a 24-hour format via the ph one ’s user interface.

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Modifying the Phone’s SIP Settings

Step 2

Save the file with the same file name, SIPDefault.cnf, to the root directory of your TFTP server.

The following is an example of a SIP d efau lt co nfiguration file:

; sip default configuration file

#Image Version image_version:P0S3

#Default Codec preferred_codec :g711ulaw

#Enable Registration proxy_register :1 ;

#Registration expiration timer_register_expires :3600 ;

#Proxy address proxy1_address: 192.168.1.1 ;

xxyy

;

Modifying the Phone-Specific SIP Configuration File

In the phone-specific SIP configuration file, maintain those parameters that are specific to a phone such as the lines configured on a phone and the users defined for those lines.

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Before You Begin

•

Review the guidelines and restrictions documented in the “Configuration File Guidelines” section on pa ge 2-6.

•

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Procedure

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Step 1

•

linex_shortname—(Optional) N ame or nu mbe r a ssoci ated w ith the linex_name as you want it to display on the phone’s LCD if the linex_name length exceeds the allowable space in the display area. For example, if the linex_name value is the phone n um ber 1 11- 222-33 3-4 444 , you c an spec ify 34444 for this para mete r to have 3444 d isplay on the L CD in stead. Alternately, if the value for the linex_name parameter is the email address “username@company .com”, you can sp ecif y th e “u ser na me ” to have ju st the user name appear on the LCD instead.

This parameter is used for display-only purposes. If a value is not specified for this parameter, the value in the linex_name variable is displayed.

•

linex_authname—(Required for line 1 when registration is enabled and the proxy server requires authenticatio n) Name used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a value is not configured for the linex_authname parame ter for a line when registration is enabled, the value defined for line 1 is used. If a value is not defined for line 1, th e default line1_ au thname is UNPROVISIONED.

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•

linex_password—(Required for line 1 when registration is enabled and the proxy server requires authenticatio n) Password used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a value is not configured for the linex_password parameter for a line when registration is enabled, the value defined for line 1 is used. If a value is not defined for line 1, the de fault line1_ passwor d is UNPROVISIONED.

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•

linex_displayname—(Optional) Identification as it should a ppear for c aller identification purposes. For example, instead of jdoe@company.com displaying on phones th at have caller ID, you ca n sp ecify Joh n Doe in th is parameter to have John Doe display on the callee end instead. If a value is not specified for this parameter, nothing is used.

•

dnd_control—(Optional) Whethe r the Do Not Di sturb feature is enabled or disabled by default on the ph one o r wh eth er the fea tur e is pe rm ane ntly enabled, making the phone a “call out” phone only. When the Do Not Disturb feature is turned on, the phone will block all calls placed to the phone and log those calls in the Missed Calls directory. Valid values are:

–

0—The Do Not Disturb f eat ure is off by default, but can be tur n o n an d off locally via the phone’s user interface.

–

1—The Do Not Disturb feature is on by default, but can be turned on and off locally via the phone’s user interface.

–

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Note

This parameter is best configured in the SIPDefault.dnf file unless configuring a phone to be a “call-out” phone on ly. When configuring a phone to be a “call-out” phone, define this parameter in the phone-specific configuration file.

•

phone_label—Label to display on the top status line of the LCD. This field is for end-user display only purposes. For example, a phone’s label can display “John Doe’s phone.” Approximately up to 11 cha racters can be used whe n specifying the phone label.

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Step 2

Save the file to your TFTP server (in the roo t dir ect ory o r a subd irec tor y containing all the phone-specific configuration files). Name the file “SIPXXXXYYYYZZZZ.cnf” where XXXXYYYYZZZZ is the MAC address of the phone. The MAC address must be in uppercase and the extension, cnf, must be in lower case (for example, SIP00503EFFD842.cnf) .

The following is an example of a configuration file:

; phone-specific configuration file sample ; Line 1 phone number line1_name : 5551212

; Line 1 name for authentication with proxy server line1_authname : 5551212

; Line 1 authentication name password line1_password : password

Modifying the SIP Parameters Manually

If you did not configure the SIP parameters via a TFTP server, you can configure them manually after you have connected the phone.

Before You Begin

•

Unlock configuration mode as de scribe d in th e “ Unlo cking C onfiguration Mode” section on page 3-2. By default, the SIP parame ters are locked to ensure that en d-us ers ca nnot mo dify settin gs that might affect their call capabilities.

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•

Review the guidelines on using the Cisco SIP IP phone menus documented in the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-21.

•

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•

When configuring the Prefe rred Co dec an d Out of Ba nd D TMF pa rame ter s, press the Change soft key until the option you desire is displayed and then press the Save soft key.

•

After making your ch ang es, r elo ck c onfiguration m ode as de scr ibed in the “Locking Configuration Mode” se ction on p age 3-2.

Procedure

Modifying the Phone’s SIP Settings

Step 1 Step 2 Step 3 Step 4 Step 5

Press the settings key. The Settings menu is displayed. Highlight SIP Configuration. The SIP Configuration menu is displayed. Highlight Line 1 Settings. Press the Select soft key. The Line 1 Con figuration menu is d isplay ed. Highlight and press the Select soft key to configure the fo llowing param eters as

necessary:

•

Short Name—(Optional) Name or number associated with the line x_name as you want it to display on the p hone’s LCD if the line x_name value exceeds the display area. For example, if the linex_nam e value is th e ph one n umb er 111-222-333-4444, you c an spe cify 3 4444 f or this p ar ame ter to h ave 3444 display on the LCD instead. Alternately, if the value for the linex_name parameter is the email address “username@company.com”, you can specify the “username” to have just the user name appear on the LCD instead. This parameter is used for disp lay- only pu rpo ses.

If a value is not specified for this parameter, the value in the Name variable is displayed.

•

Authentication Name—(Required when registration is enabled) Name used by the phone for authentication if a registration is challenged by the proxy server during initialization.

•

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Authentication Password parameter when registration is enabled, the default logical password is used. The default logical password is SIPmaca ddress where macaddress is the MAC address of the phone.

•

Display Name—(Optional) Identification as it should appear for caller-identification purposes. For exa mple, instead of jdo e@co mpany.com displaying on phones th at have caller ID, you ca n sp ecify Joh n Doe in th is parameter to have John Doe display on the callee end instead. If a value is not specified for this parameter, the Name value is used.

•

Proxy Address—(Req uire d) I P a ddress of the p rima ry SI P pr oxy se rver tha t will be used by the phone. Enter this address in IP dotted-decimal notation.

•

Proxy Port—(Optional) Port number of the primary SIP proxy server. This is the port on which the SIP client will listen for messages. The default is 5060.

Step 6

Press the Back soft key exit the Line 1 Con figuration me nu.

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Step 7

Step 8

To configure additional lines on the p hon e, highlight th e next Line x Settings, press the Select soft key and repeat Step 5 and Step 6, and then continue with Step 8.

In addition to the line settings, you can highlight and press Select to configure the following parameters on the SIP Configuration menu:

•

Message URI—Number to call to check voicemail. This number will be called when the Messages key is pressed.

•

Preferred Codec—(Optional) CODEC to use when initiating a call. Valid values are g711alaw, g711ulaw, and g729a. The defa ult is g711 ulaw.

•

Out of Band DTMF—(Optional) Whether to detect and generate the out-of-band signaling (for tone detection on the IP side of a gateway) and if so, when. The Cisco SIP IP phone supports out-of-bo und signaling via the AVT tone method. Valid values are:

–

none—Do not generate D TMF digi ts out-of -band .

–

avt—If requested by the remote side, generate DTMF digits out-of-band (and disable in-band DTMF signaling), otherwise, do no t generate DTMF digits out-of-band.

–

avt_always—Always generate DTMF digits out-of-band . This optio n disables in-band DTMF signaling.

The default is avt.

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•

Register with Proxy—(Optional) Whether the phone must register with a proxy server during initialization. Valid v alues are Yes and No. Select the No soft key to disable registration during initialization. Select the Yes soft key to enable registration during initialization. The default is No.

After a phone has initialized and registered with a proxy serv er, changing the value of this parameter to No will unregister the phone from the proxy server. To reinitiate a registration, change the value of this parameter back to Yes.

Modifying the Phone’s SIP Settings

Step 9

Caution

Note

If you enable registration, and authentication is required, you must specify va lues for the Authe ntication Name and Authentication Password para meters.

•

Register Expires—(Optional) The amount of time, in seconds, after which a REGISTRATION request will expir e. Thi s value i s used the Expi re he ader field. The valid value is any positive number, however, we recommend 3600 seconds. The default is 3600.

•

TFTP Directory—(Required if phone-specific configuration files are located in a subdirectory) Path to the TFTP subdirectory in which phone-specific configuration files ar e store d.

•

Phone Label—(Optional) Label to display on the top status line of the LCD. This field is for end-user display only purposes. For example, a phone’s label can display “John Doe’s phone.”

When done, p res s th e Save soft key to save your changes and exit the SIP Configuration menu.

When you have completed your changes, ensure that yo u lock the phone as described in the “Locking Configuration Mode” section on page 3-2.

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Setting the Date, Time, and Daylight Savings Time

The current da te and tim e is sup ported on the Ci sco SIP IP pho ne via SN TP a nd is displayed on the phon e’s LCD. In addition to supporting the cur rent date a nd time, daylight savings time (DST) and time zone settings are also supported. DST can be configured to be obtained via an absolute (for example, starts on April 1 and ends on October 1) or relative (for example, starts the first Sunday in April and ends on the last day o f Octobe r) c on figuration.

We recommend that date and time-related parameters be defined in the SIPDefault.cnf file.

Before You Begin

When configuring the date, time, time zone, and DST settings, remember the following:

•

Review the guidelines and restrictions documented in the “Configuration File Guidelines” section on pa ge 2-6.

•

Determine whether you want to configure absolute DST or relative DST.

•

The SNTP parameters specify how the phone will obtain the current time from an SNTP server. Review the guidelines in Table 3-2 and Table 3-3 before configuring the SNTP para meter s:

Table 3-2 lists the actions that take place when a null value (0.0.0.0) is specified in the sntp_server parameter.

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Table 3-2 Actions Based on sntp_mode When the sntp_server Parameter is Set to a Null Value

sntp_server =0.0.0.0

Sends

Receives

sntp_mode= unicast

Nothing. No known server

with which to communicate.

Nothing. No known server

with which to communicate.

sntp_mode= multicast

Nothing. When in multicast

mode, SNTP requests are not sent.

SNTP data via the SNTP/NTP multicast address from the local network broadcast address from any server on the network.

sntp_mode= anycast

SNTP packet to the local network address.

After the first SNT P response is recei ved, the phone switches to unicast mode with the server being set as the one who first responded.

Unicast SNTP data from the SNTP server that first responded to the network broadcast request.

sntp mode= directedbroadcas t

SNTP packet to the local network address.

After the first SNT P response is received, the phone switches to multicast mode.

SNTP data from the SNTP/NTP multicast address and the lo cal network broadcast address from any server on th e network.

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Setting the Date, Time, and Daylight Savings Time

T abl e 3-3 lists the actions that take place when a valid IP address is specified in the sntp_server parameter.

Table 3-3 Actions Based on sntp_mode When the sntp_server Parameter is Set to an IP Address

sntp_server = 0.0.0.0

Sends

Receives

sntp_mode= unicast

SNTP request to the SNTP server.

SNTP response from the SNTP server an d ignores responses from other SNTP servers.

sntp_mode= multicast

Nothing. When in multicast

mode, SNTP requests are not sent.

SNTP data via the SNTP/NTP multicast address from the local network broadcast address.

sntp_mode= anycast

SNTP request to the SNTP serve r.

SNTP response from the SNTP server and ignores responses from other SNTP servers.

sntp_mode= directedbroadcast

SNTP packet to the SNTP server.

After the first SNT P response is received, the phone switches to multicast mode.

SNTP data from the SNTP/NTP multicast address and the lo cal network broadcast address and ignores responses from other SNTP servers.

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Procedure

Setting the Date, Time, and Daylight Savings Time

Step 1

Step 2

Using an ASCII editor, open the SIPDefault.cnf file and define or modify values for the following SN TP- spe cific SIP pa ra met ers as n ece ssar y:

•

sntp_mode—(Required) Mode in which the phone will listen for the SNTP server. Valid values are unicast, multicast, anycast, or directedbroadcast.

See Table 3-2 a nd Table 3-3 for an explanat ion o n h ow the se values wo rk depending on the sntp_s er ver par amete r value.

•

sntp_server—(Requi red) IP ad dress of the SN TP server from wh ich the phone will obtain time data.

See Table 3-2 a nd Table 3-3 for an explanat ion o n h ow the se values wo rk depending on the sntp_s er ver par amete r value.

•

time_zone—(Required) Time zone in which the phon e is located. Valid values are hour/minute, -hour/minute, +hour/minute, hour, -hour, +hour, PST , MST, CST, or EST.

To configure common DST settings, specify values for the following parameters:

•

dst_offset—Offset from the phone’s time when DST is in effect. When DST is over, the specified offset is no longer applied to the phone’s time. Valid values are the same as for the time_zone parameter.

•

dst_auto_adjust—Whether or not DST is automatically adjusted on the phones. Valid values are 0 (disable automatic DST adjustment) or 1 (enable automatic DST adjustment). The default is 1.

•

dst_start_month—Month in which DST starts. Valid values are January, February, March, April, May, June, July, August, September, October, November, and December or 1 through 12 with Janu ary being 1 and December being 12. When specify ing the name of a month, the value is case-sensitive and should be typed as cited in this description.

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•

dst_stop_month—Month in which DS T e nds. Valid values are January, February, March, April, May, June, July, August, September, October, November, and December or 1 through 12 with Janu ary being 1 and December being 12. When specify ing the name of a month, the value is case-sensitive and should be typed as cited in this description.

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Setting the Date, Time, and Daylight Savings Time

•

dst_start_time—Time of day on which DST begins. Valid values are hour/minute (02/00) or hour ( 14:30 ).

•

dst_stop_time—Time of day on which DST ends . Valid values are hour/minute (02/00) or hour ( 14:30 ).

Chapter3 Managing Cisco SIP IP Phones

Step 3

Step 4

To configure absolute DST, specify values for the foll owing param eters or to configure relative DST, proceed to Step 4:

•

dst_start_day—Day of the m onth on w hich DST b egins. Valid values are 1 through 31 for the days of the month or 0 when specifying

relative DST to specify that this field be ignored and that the value in the dst_start_day_of_week para meter be used in stead.

•

dst_stop_day—Day of the month on which DST ends. Valid values are 1 through 31 for the days of the month or 0 when specifying

relative DST to specify that this field be ignored and that the value in the dst_stop_day_of_week pa rameter be use d instead.

To configure relative DST, specify values for the following parameters:

•

dst_start_day_of_week—Day of the week on which DST begins. V alid values are Su nday or Sun, Mond ay or Mon, T uesday or Tu e, W ednesda y

or W ed, Thursday or Thu, Friday or Fri, Saturday or Sat, or Sunday or Sun or 1 through 7 with 1 being Sun day and 7 being Saturday. When specifying the name of the day, the value is case-sensitive and should be typed as cited in this description.

•

dst_start_week_of_month—Week of month in whic h D ST begins. Valid values are 1 through 6 and 8 with 1 being the first week and each

number thereafter being subsequ ent week s an d 8 specif ying the last week in the month regardless of which week the last week is.

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•

dst_stop_day_of_week—D ay of the wee k on which DST ends. V alid values are Su nday or Sun, Mond ay or Mon, T uesday or Tu e, W ednesda y

•

dst_stop_week_of_month—Week of month in which DST ends. Valid values are 1 through 6 and 8 with 1 being the first week and each

number thereafter being subsequ ent week s an d 8 specif ying the last week in the month regardless of which week the last week is.

Step 5

Save the file with the same file name, SIPDefault.cnf, to the root directory of your TFTP server.

The following is an example of the configuration for an absolute DST configuration:

; sip default configuration file (additional configuration text omitted)

Setting the Date, Time, and Daylight Savings Time

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time_zone : 03/00 dst_offset : 01/00 dst_start_month : April dst_start_day : 1 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 1 dst_stop_time : 02/00 dst_stop_autoadjust : 1

(additional configuration text omitted)

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Erasing the Local ly - Defined Settings

The following is an e xample of the configura tion for a relati v e DST conf iguration:

; sip default configuration file (additional configuration text omitted)

time_zone : PST dst_offset : 01/00 dst_start_month : April dst_start_day : 0 dst_start_day_of_week : Sunday dst_start_week_of_month : 1 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 02/00 dst_stop_autoadjust : 1

(additional configuration text omitted)

Erasing the Locally-Defined Settings

You can erase the locally-defined network settings and the SIP settings that have been configured in the phon e.

Erasing the Locally-Defined Network Settings

When you erase the locally-defined settings, the values are reset to the defaults.

Before You Begin

•

Unlock configuration mode as de scribe d in th e “ Unlo cking C onfiguration Mode” section on pa ge 3-2.

•

If DHCP has been disabled on a phone, clearing the phone’s settings will reenable it.

•

Select the Erase Config parameter by pressing the down arrow to scroll to and highlight the parameter or by pressin g the numbe r that repre se nts the parameter (located to the left of the parameter name on the LCD).

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Procedure

Erasing the Locally-Defined Settings

Step 1 Step 2 Step 3 Step 4 Step 5 Step 6

Press the settings key. The Settings menu is displayed. Highlight Network Configuration. Press the Select soft key. The Network Configuration settings are displayed. Highlight Erase Configuration. Press the Yes soft key. Press the Save soft key. The phone program s the new infor mation into Fla sh

memory and resets.

Erasing the Locally-Defined SIP Settings

When you erase the locally-defined SIP settings, the values are reset to the defaults.

Note

If your system has been set up to have the phones re trieve their SIP parameters via a TFTP server , you will need to edit the configuration file in which a parameter is defined to delete the parameter. When deleting a parameter, leave the variable in the file, but change its value to a null value ““ ”” or “UNPROVISIONED”. If both the variable and its v alue are r emove d, the phone will use the setting for that variable that it has stored in Flash memory.

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Step 1 Step 2 Step 3

Before You Begin

Unlock configuration mode as described in the “Unlocking Configuration Mode” section on page 3-2.

Procedure

Press the settings key. The Settings menu is displayed. Highlight SIP Configuration. Press the Select soft key. The SIP Configuration settings a re d isplay ed.

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Accessing Status I nformation

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Step 4 Step 5 Step 6 Step 7 Step 8

Highlight the parameter for which you wish to erase the setting. Press the Edit soft key. Press the << soft key to delete the current value. Press the Validate soft key to save your change and exit the Edit panel. If modifying a line para meter, press the Back soft key to exit the Line

Configuration panel.

Step 9

Press the Save soft key. The phone program s the new infor mation into Fla sh memory and resets.

Accessing Status Information

There are several types of status information that you can access via the settin gs key. The information that you can obtain via the settings key can aid in system management.

To access status information, select settings and then select Status from the Settings menu. From the Status which the following three options are available:

•

Status Messages—Displays diagnostic messages.

•

Network Status—Displays performance m essages.

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•

Firmware V ersion—Displays information about the current firmware version on the phone.

In addition to the status messages available via the Setting Status menu, you ca n also obtain status messages for a current call.

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Viewing Status Messages

T o view status messages that you can use to diagnose network problems, complete the following steps:

Accessing Sta tu s In fo rm at io n

Step 1 Step 2 Step 3 Step 4 Step 5 Step 6

Press the Settings key. The Settings menu is displayed. Highlight Status. Press the Select soft key. The Setting Status menu is disp layed. Highlight Status Messages. Press the Select soft key. The Status Messages pane l is displayed . To exit the Status Messages panel, press the Exit soft key.

Viewing Network Statistics

To view statistical information abou t the p hone an d n etwork pe rform an ce, complete the following steps:

Step 1 Step 2 Step 3 Step 4 Step 5

Press the settings key. The Settings menu is displayed. Highlight Status. Press the Select soft key. The Setting Status menu is disp layed. Highlight Network Statistics. Press the Select soft key. The Network Statistics panel is displayed.

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The following information is displayed on this panel:

•

Rcv—Number of packets received by the phone; not through the switch .

•

Xmit—Number of packets se nt by the pho ne; no t thr ough the switch.

•

REr—Number o f pa ckets re ceived by the p hon e tha t cont ain ed err ors .

•

BCast—Number of broadca st pac kets received by the pho ne.

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•

Chapter3 Managing Cisco SIP IP Phones

Phone State Message—TCP messages indicating the state of the phone. Possible mess ages are:

–

Phone Initialized—TCP conn ec tion ha s n ot gone d own since the phone was powered on.

–

Phone Closed TCP—TCP connection was closed by the phone.

–

TCP Timeout—TCP connection was c lose d b ecause of a r etry tim eout.

–

Error Code—Error mes sag es indicating un usua l reas ons the TCP connection was close d.

Elapsed Time—Length of time (in days, hours, minutes, and seconds) since the last power cycle.

Port 0 Full, 100—Indicates that the network is in a linked state and has auto-negotiated a full-duplex 10 0Mbps co nne ctio n.

Port 0 Half, 100—Indicates that the network is in a linked state and has auto-negotiated a half-duplex 100Mbps connection.

Port 0 Full, 10—Indicates that the network is in a linked state and has auto-negotiated a full-duplex 10Mbps connection.

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•

Port 0 Half, 10—Indicates that the network is in a linked state and has auto-negotiated a half-duplex 10Mbps connection.

•

Port 1 Full, 100—Indicates that the network is in a linked state and has auto-negotiated a full-duplex 10 0Mbps co nne ctio n.

•

Port 1 Half, 100—Indicates that the network is in a linked state and has auto-negotiated a half-duplex 100Mbps connection.

•

Port 1 Full, 10—Indicates that the network is in a linked state and has auto-negotiated a full-duplex 10Mbps connection.

•

Port 1 Half, 10—Indicates that the network is in a linked state and has auto-negotiated a half-duplex 10Mbps connection.

Step 6

Note

To exit the Network Statistics panel, press the Exit soft key.

T o re set the v alues displaye d on Network Statistics pane l, po wer of f and power on the phone.

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Viewing the Firmware Version

To view network statistics, complete the following steps:

Upgrading the Cisco SIP IP Phone Firmware

Step 1 Step 2 Step 3 Step 4 Step 5

Press the settings key. The Settings menu is displayed. Highlight Status. Press the Select soft key. The Setting Status menu is disp layed. Highlight Firmware Versions. Press the Select soft key. The Firmware Versions panel is displayed. The following information is displayed on this panel:

•

Application Load ID—C urr ent soft ware image on the phon e.

•

Boot Load ID—Bootstra p loade r imag e version tha t is m a nufactur ed on the phone. This image name doe s not chan ge.

Step 6

To exit the Firmware Versions panel, press the Exit soft key.

Upgrading the Cisco SIP IP Phone Firmware

There two methods that you can use to upgra de the firmware on your Cisc o SIP IP phones. You can upgrade the firmware on one phone a t a time via the phone-specific configuration or you can upgrad e the firmware on a syste m of phones using the default configuration file.

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Before You Begin

•

To upgrade the firmware on just one ph one at a tim e, y ou upg rade the image_version in the phone-specific configuration file. To upgrade the firmware on a system of phones, specify the image_version in the de fault configuration file and do not define the image_version in the ph one -spec ific configuration files.

•

Ensure that the late st versio n of the Cisc o SIP I P ph one firmware has be en copied from CCO to the root d ire ctor y of you r TFTP server.

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Procedure

Chapter3 Managing Cisco SIP IP Phones

Step 1

Step 2

Step 3

Note

Copy the binary file P0S3xxyy.bin (where xx is the version number and yy is the subversion number) from CCO to the r oot direc tory of the T FTP server.

Using a text editor, open the configuration file and update the image version specified in the image_version variable. The version name in image_version variable should match the version name (without the .bin extension) of the latest firmware that y ou down lo ad ed .

Reset each phone. The phone contacts the TFTP server and requests its configuration files. The

phone compares the image defined in the file to the image that it has stored in Flash memory. If the phone determines that the image defined in the file differs from the image in Flash memory, it downloads the image defined in the configuration file (which is stored in the root directory on the TFTP server). Once the new image has been downloaded, the phone programs that image into Flash memory and then reboots.

If you do not define the image_version parameter in the default configuration file, only phones for whic h you have upd ated th eir phone-specific configuration file with the new image version and restarted will use the latest firmware image. All other phones will use the older version until their configuration files have been updated with the new image version.

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Chapter 3 Managing Cisco SI P IP Phones

Performing an Image Upgrade and Remote Reboot

With Vers ion 2. 0 o f the Cisco SI P I P Pho ne 79 60, you ca n pe rf orm a n image upgrade and remote reboot using Notif y messag es and the synchin fo.xml file.

Note

To perform an image grade and remote reboot, a SIP proxy server and a TFTP server must exist in the phone network.

To upgrade the firmware image and pe rform a re mote r eb oot, c omp lete the following tasks:

Using an ASCII editor, open the SIPDefault.cnf file located in the root directory of your TFTP server and change the image_version parameter to the name of the latest image.

Using an ASCII editor, open the syncinfo.xml file located in the root directory of your TFTP server and specify values for the image version and sync parameter as f ollows:

Where:

–

image_version is the image version of the phone. The asterisk (*) can be used as a wildcard character.

–

sync_number is the synchronization level of the phone. The default sync level for the phone is 1. Valid values is a character-string up to 32 characters.

Send a NOTIFY message to the phone. In the Noti fy message, ensure that the an Event header equal to “check-sync” is included.

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Performing an Image Upgrade and Remote Reboot

The following i s a n exam pl e of a N oti f y m es sag e:

NOTIFY sip: Via: SIP/2.0/UDP Via: SIP/2.0/UDP From: <sip:webadim@ To: <sip: Event: check-sync Date: Mon, 10 Jul 2000 16:28:53 -0700 Call-ID: 1349882@ CSeq: 1300 NOTIFY Contact: <sip:webadmin@ Content-Length: 0

lineX_name@ipaddress

Once the remote reboot process is initiated on the phone via the Notify message, the following actions take place:

If the phone is currently in an idle state, the phone will wait 20 seconds and then contact the TFTP server for the syncinfo.xml file. If the phone is not in an idle state, the phone will wait until it is in an idle state for 20 seconds and then contact the TFTP server for the syncinfo.xml file.

The phone reads the syncinfo.xml file and perfor ms the following as appropriate:

Determines whether the current image is specified. If so, the phone proceeds to c. If no t, the pho ne p roce eds to b.

ipaddress ipaddress

ipaddress

Chapter3 Managing Cisco SIP IP Phones

:5060 SIP/2.0

:5060;branch=1

3-36

Determines whether there is a wildcard entry (*) in the image version parameter. If so, the phone proceeds to c. If not, the phone proceeds to d.

Determines if the sync value is dif ferent than what is stored on the phone. If so, the phone pr ocee ds to e. I f n ot, the ph one pro cee ds to d.

The phone does no th ing.

The phone reboots. The phone the performs a normal re boot process as de scribed in

“Initialization Process Overview” section on page 2-1, sees the new image, and upgrades to the new image with a sync value of 2.

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APPENDIX

SIP Compliance with RFC-2543 Information

This section describes how the Cisco SIP IP phone complies with the IETF definition of SIP as described in RFC 2543.

This section contains compliance information on the following:

•

SIP Functions, page A-2

•

SIP Methods, page A-2

•

SIP Responses, page A-3

•

SIP Header Fields, page A-10

•

SIP Session Description Protocol (SDP) Usage, page A-12

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A-1

SIP Functions

Function Supported?

User Agent Client (UAC) Yes User Agent Server (UAS) Yes Proxy Server Third-party only Redirect Server Third-party only

SIP Methods

Five of the six methods used by the SIP gateway are supported:

Method S uppo rted? Comments

INVITE Yes The Cisco SIP IP phone supports

Appendix A SIP Compliance with RFC-2543 Information

mid-call change s s u ch as p utt ing a ca ll on hold as signaled by a n ew INVIT E that contains an existing Ca ll- ID.

A-2

ACK Yes OPTIONS No

None.

BYE Yes CANCEL Yes REGISTER Yes The Cisco SIP IP phone supports both

user and device registration.

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Appendix A SIP Compliance with RFC-2543 Information

SIP Responses

Release 1.0 of the Cisco SIP IP phone supports the following SIP responses:

•

1xx Response—Information Response s, pa ge A-4

•

2xx Response—Successful Responses, page A-4

•

3xx Response—Redirection Responses, page A-5

•

4xx Response—Request Failure Responses, page A-5

•

5xx Response—Server Failure Responses, page A-10

•

6xx Response—Global Responses, page A-10

SIP Responses

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Appendix A SIP Compliance with RFC-2543 Information

SIP Responses

1xx Response—Information Responses

1xx Response Supported? Comments

100 Trying Yes The Cisco SIP IP phone generates this

response for an incoming INVITE. Upon receiving this response, the phone waits for a 180 Ringing, 183 Session progress,

or 200 OK response. 180 Ringing Yes None 181 Call is being

forwarded 182 Queued

See comments

183 Session Progress

2xx Response—Successful Responses

2xx Response Supported? Comments

200 OK Yes None

The Cisco SIP IP phone does not

generate these respons es, h owever, the

phone does rece ive them. The p hone

processes these responses the same way

that it processes the 100 Trying

response.

The SIP IP phone does not generate this

message. Upon receiving this response,

the phone provides early media cut

through and then waits for a 200 OK

response.

A-4

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Cisco IP phone 7960 User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual