Specifications | Cisco Systems 78-10450-02 IP Phone User Manual

Cisco IP Phone Models 7905G and
7912G Administrator Guide (SIP)
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Text Part Number: OL-4277-01
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Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
Copyright © 2003 Cisco Systems, Inc. All rights reserved.
C O N T E N T S
Preface xi
Overview xi
Audience xi
Organization xii
Related Documentation xiii
Obtaining Documentation xiii
World Wide Web xiii
Documentation CD-ROM xiii
Ordering Documentation xiv
Documentation Feedback xiv
Obtaining Technical Assistance xiv
Cisco Connection Online xv
Technical Assistance Center xv
Obtaining Additional Publications and Information xvi
Document Conventions xvii
CHAPTER
1
Overview of the Cisco IP Phone Models 7905G and 7912G 1-1
Hardware Overview 1-1
Software Features 1-4
Protocols 1-4
Basic Services 1-5
Supplemental Services 1-5
SIP-Specific Services 1-6
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Contents
SIP Overview 1-7
SIP Functions 1-7
SIP Components 1-8
CHAPTER
2
Installing the Cisco IP Phone Models 7905G and 7912G 2-1
Prerequisites 2-1
Safety 2-2
Connecting to the Network 2-4
Providing Power to the Cisco IP Phone 2-5
Power Source Design 2-5
Redundancy Feature 2-6
Installing the Cisco IP Phone Models 7905G and 7912G 2-6
Physical Installation of the Cisco IP Phone Models 7905G and 7912G 2-7
Mounting the Phone to the Wall 2-12
Basic Configuration of the Cisco IP Phone Models 7905G and 7912G 2-13
Verifying Installation 2-15
CHAPTER
3
Configuring the Cisco IP Phone Models 7905G and 7912G for SIP 3-1
Overview of Configuration Methods 3-1
Default Boot Load Behavior 3-2
Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation 3-4
Basic Configuration Steps in a TFTP Server Environment 3-6
Minimum Configuration Settings for the Cisco IP Phone Models 7905G and
7912G 3-9
Making Settings with a DHCP Server 3-9
Making Settings Manually 3-10
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CHAPTER
4
Basic and Additional SIP Services 4-1
Important Basic SIP Services 4-2
Required Parameters 4-2
Establishing Authentication 4-4
Configuring Refresh Interval 4-4
Additional SIP Services 4-4
Setting Up and Placing a Call Without Using a SIP Proxy 4-10
Complete Reference of all Cisco IP Phone SIP Services 4-12
CHAPTER
5
Configuring the Cisco IP Phone Models 7905G and 7912G Using the Phone
Menus 5-1
Configuration Menus on the Cisco IP Phone 5-2
Displaying a Configuration Menu 5-3
Unlocking and Locking Configuration Parameters 5-4
Editing Parameter Values 5-5
Network Configuration Menu 5-6
SIP Configuration Menu 5-14
Model Information Menu 5-17
Status Menu 5-18
CHAPTER
6
Configuring the Cisco IP Phone Models 7905G and 7912G Using a TFTP
Server 6-1
Using Profiles with TFTP Configuration 6-2
Setting Up the TFTP Server with Cisco IP Phone Software 6-4
Configuring the Cisco IP Phone to Obtain its Configuration File from the TFTP
Server 6-5
Using a DHCP Server 6-5
Without Using a DHCP Server 6-8
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Creating or Updating a Profile 6-9
Profile Conventions 6-10
Creating a Phone-Specific Profile 6-11
Creating a Default Profile 6-13
Using Encryption 6-15
Refreshing or Resetting the Cisco IP Phone 6-17
Refreshing the Cisco IP Phone 6-17
Resetting the Cisco IP Phone 6-17
CHAPTER
7
Configuring the Cisco IP Phone Models 7905G and 7912G Using Web
Pages 7-1
Accessing a Phone’s Web Page 7-2
Viewing Statistical and Operational Information 7-4
Device Information 7-4
Network Configuration 7-5
Network Statistics 7-6
CHAPTER
8
Profile and Web Page Parameters and Defaults 8-1
Network Parameters 8-2
SIP Parameters 8-9
Call Preference Parameters 8-13
Tone Parameters 8-21
Audio Parameters 8-24
Profile Parameters 8-26
Parameter Details 8-27
BusyTone, CallWaitTone, DialTone, DialTone2, and RingBackTone Parameter
Details 8-28
ReorderTone Parameter Details 8-30
ConnectMode Detail 8-36
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DialPlan Parameter Detail 8-38
Upgradecode Parameter Detail 8-41
Upgradelogo Parameter Detail 8-43
CHAPTER
9
Troubleshooting the Cisco IP Phone Models 7905G and 7912G 9-1
General Troubleshooting Tips 9-1
Resolving Problems 9-2
Contacting TAC 9-5
Debugging 9-5
APPENDIX
A
Additional End User Features on the Cisco IP Phone Models 7905G and
7912G A-1
Call Preferences Menu A-1
Softkeys A-6
Other Features A-7
APPENDIX
B
Phone Menu, Web Page, and Profile Equivalent Parameters B-1
APPENDIX
C
SIP Call Flows C-1
Supported SIP Request Methods C-1
Call Flow Scenarios for Successful Calls C-2
Cisco IP Phone-to-SIP Server—Registration without Authentication C-2
Cisco IP Phone-to-SIP Server—Registration with Authentication C-4
Cisco IP Phone-to-Cisco IP Phone—Basic SIP to SIP Call without
Authentication C-7
Cisco IP Phone-to-Cisco IP Phone—Basic SIP to SIP Call with
Authentication C-15
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Contents
APPENDIX
D
Specifications for the Cisco IP Phone Models 7905G and 7912G D-1
Physical Specifications D-2
Electrical Specifications D-2
Environmental Specifications D-3
Physical Interfaces D-3
Software Specifications D-3
GLOSSARY
INDEX
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Preface
Overview
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP) describes
how to install and configure the Cisco IP Phone models 7905G and 7912G for use
on a Session Initiation protocol (SIP) network. It includes configuration steps for
network parameters, standard services, and supplemental services.
This guide does not cover information that is related to the implementation of a
SIP Voice over IP (VoIP) network.
Audience
This guide is intended for service providers and network administrators who
administer VoIP services using the Cisco IP Phone models 7905G and 7912G and
who have a fundamental understanding of SIP. The tasks described in this guide
are not intended for end users of the phones. Many of these tasks affect the ability
of a phone to function on the network and require an understanding of IP
networking and telephony concepts.
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Organization
Organization
This manual is organized as follows:
Chapter 1, “Overview of the
Cisco IP Phone Models 7905G and 7912G”
Describes the hardware and software features of the
phone and provides a brief overview of SIP
Chapter 2, “Installing the Cisco IP Phone
Models 7905G and 7912G”
Provides procedures for installing the phone at an
end user location
Chapter 3, “Configuring the
Cisco IP Phone Models 7905G and 7912G for
SIP”
Provides an overview of how to configure the phone
to operate with the SIP signaling image
Chapter 4, “Basic and Additional SIP Services” Provides information about basic and additional SIP
services that the phone supports
Chapter 5, “Configuring the Cisco IP Phone
Models 7905G and 7912G Using the Phone
Menus”
Explains how to use menus on the phone to
configure or view a variety of network, SIP, and
phone settings
Chapter 6, “Configuring the Cisco IP Phone
Models 7905G and 7912G Using a TFTP
Server”
Explains how to configure a phone using a Trivial
File Transfer Protocol (TFTP) server
Chapter 7, “Configuring the
Cisco IP Phone Models 7905G and 7912G
Using Web Pages”
Describes how to use a phone’s web page to
configure parameters and obtain information
Chapter 8, “Profile and Web Page Parameters
and Defaults”
Describes the parameters that you can use to
configure a phone through a profile or through a
phone’s web page
Chapter 9, “Troubleshooting the
Cisco IP Phone Models 7905G and 7912G”
Provides guidance for diagnosing and correcting
installation and configuration errors
Appendix A, “Additional End User Features on Describes end-user features that are not explained in
the Cisco IP Phone Models 7905G and 7912G” the User Guide for the phone
Appendix B, “Phone Menu, Web Page, and
Profile Equivalent Parameters”
Lists the configurable options on the phone menus
and their equivalent parameters in the phone
configuration web pages and profile
Appendix C, “SIP Call Flows”
Describes some basic call flows for the phone
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Preface
Related Documentation
Appendix D, “Specifications for the
Cisco IP Phone Models 7905G and 7912G”
Provides physical, electrical, environmental, and
software specifications for the phone
Glossary
Explains terms that may be new to you
Related Documentation
For more information, refer to the following documents, which are available at
this URL:
http://www.cisco.com/en/US/products/hw/phones/index.html
•
Cisco IP Phone 7905G/7912G User Guide for Cisco CallManager
•
Regulatory Compliance and Safety Information for the Cisco IP Phone 7900
Series
Obtaining Documentation
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Preface
Obtaining Technical Assistance
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Obtaining Technical Assistance
Cisco Connection Online
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Obtaining Additional Publications and Information
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Cisco Press publishes a wide range of networking publications. Cisco
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Internetworking Troubleshooting Guide, and the Internetworking Design
Guide. For current Cisco Press titles and other information, go to Cisco Press
online at this URL:
http://www.ciscopress.com
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•
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http://www.cisco.com/go/packet
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Document Conventions
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italic font
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Alternative keywords are grouped in braces and separated by vertical bars.
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Document Conventions
Convention
Description
[x|y|z]
Optional alternative keywords are grouped in brackets and separated by
vertical bars.
string
A nonquoted set of characters. Do not use quotation marks around the string
or the string will include the quotation marks.
screen
font
Terminal sessions and information the system displays are in screen font.
boldface screen
font
Information you must enter is in boldface
screen
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italic screen font
Arguments for which you supply values are in italic screen font.
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The symbol ^ represents the key labeled Control—for example, the key
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< >
Nonprinting characters, such as passwords are in angle brackets.
Note
Caution
Means reader take note. Notes contain helpful suggestions or references to
material not covered in the publication.
Means reader be careful. In this situation, you might do something that could
result in equipment damage or loss of data.
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Document Conventions
Warnings use the following conventions:
Warning
IMPORTANT SAFETY INSTRUCTIONS
This warning symbol means danger. You are in a situation that could cause
bodily injury. Before you work on any equipment, be aware of the hazards
involved with electrical circuitry and be familiar with standard practices for
preventing accidents. Use the statement number provided at the end of each
warning to locate its translation in the translated safety warnings that
accompanied this device. Statement 1071
SAVE THESE INSTRUCTIONS
Waarschuwing
BELANGRIJKE VEILIGHEIDSINSTRUCTIES
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 de standaard praktijken
om ongelukken te voorkomen. Gebruik het nummer van de verklaring
onderaan de waarschuwing als u een vertaling van de waarschuwing die bij
het apparaat wordt geleverd, wilt raadplegen.
BEWAAR DEZE INSTRUCTIES
Varoitus
TÄRKEITÄ TURVALLISUUSOHJEITA
Tämä varoitusmerkki merkitsee vaaraa. Tilanne voi aiheuttaa ruumiillisia
vammoja. Ennen kuin käsittelet laitteistoa, huomioi sähköpiirien
käsittelemiseen liittyvät riskit ja tutustu onnettomuuksien yleisiin
ehkäisytapoihin. Turvallisuusvaroitusten käännökset löytyvät laitteen
mukana toimitettujen käännettyjen turvallisuusvaroitusten joukosta
varoitusten lopussa näkyvien lausuntonumeroiden avulla.
SÄILYTÄ NÄMÄ OHJEET
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Document Conventions
Attention
IMPORTANTES INFORMATIONS DE SÉCURITÉ
Ce symbole d'avertissement indique un danger. Vous vous trouvez dans une
situation pouvant entraîner des blessures ou des dommages corporels. Avant
de travailler sur un équipement, soyez conscient des dangers liés aux circuits
électriques et familiarisez-vous avec les procédures couramment utilisées
pour éviter les accidents. Pour prendre connaissance des traductions des
avertissements figurant dans les consignes de sécurité traduites qui
accompagnent cet appareil, référez-vous au numéro de l'instruction situé à la
fin de chaque avertissement.
CONSERVEZ CES INFORMATIONS
Warnung
WICHTIGE SICHERHEITSHINWEISE
Dieses Warnsymbol bedeutet Gefahr. Sie befinden sich in einer Situation, die
zu Verletzungen führen kann. Machen Sie sich vor der Arbeit mit Geräten mit
den Gefahren elektrischer Schaltungen und den üblichen Verfahren zur
Vorbeugung vor Unfällen vertraut. Suchen Sie mit der am Ende jeder Warnung
angegebenen Anweisungsnummer nach der jeweiligen Übersetzung in den
übersetzten Sicherheitshinweisen, die zusammen mit diesem Gerät
ausgeliefert wurden.
BEWAHREN SIE DIESE HINWEISE GUT AUF.
Avvertenza
IMPORTANTI ISTRUZIONI SULLA SICUREZZA
Questo simbolo di avvertenza indica un pericolo. La situazione potrebbe
causare infortuni alle persone. Prima di intervenire su qualsiasi
apparecchiatura, occorre essere al corrente dei pericoli relativi ai circuiti
elettrici e conoscere le procedure standard per la prevenzione di incidenti.
Utilizzare il numero di istruzione presente alla fine di ciascuna avvertenza per
individuare le traduzioni delle avvertenze riportate in questo documento.
CONSERVARE QUESTE ISTRUZIONI
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Document Conventions
Advarsel
VIKTIGE SIKKERHETSINSTRUKSJONER
Dette advarselssymbolet betyr fare. Du er i en situasjon som kan føre til skade
på person. Før du begynner å arbeide med noe av utstyret, må du være
oppmerksom på farene forbundet med elektriske kretser, og kjenne til
standardprosedyrer for å forhindre ulykker. Bruk nummeret i slutten av hver
advarsel for å finne oversettelsen i de oversatte sikkerhetsadvarslene som
fulgte med denne enheten.
TA VARE PÅ DISSE INSTRUKSJONENE
Aviso
INSTRUÇÕES IMPORTANTES DE SEGURANÇA
Este símbolo de aviso significa perigo. Você está em uma situação que poderá
ser causadora de lesões corporais. Antes de iniciar a utilização de qualquer
equipamento, tenha conhecimento dos perigos envolvidos no manuseio de
circuitos elétricos e familiarize-se com as práticas habituais de prevenção de
acidentes. Utilize o número da instrução fornecido ao final de cada aviso para
localizar sua tradução nos avisos de segurança traduzidos que acompanham
este dispositivo.
GUARDE ESTAS INSTRUÇÕES
¡Advertencia!
INSTRUCCIONES IMPORTANTES DE SEGURIDAD
Este símbolo de aviso indica peligro. Existe riesgo para su integridad física.
Antes de manipular cualquier equipo, considere los riesgos de la corriente
eléctrica y familiarícese con los procedimientos estándar de prevención de
accidentes. Al final de cada advertencia encontrará el número que le ayudará
a encontrar el texto traducido en el apartado de traducciones que acompaña
a este dispositivo.
GUARDE ESTAS INSTRUCCIONES
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Document Conventions
Varning!
VIKTIGA SÄKERHETSANVISNINGAR
Denna varningssignal 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 vanliga förfaranden för att
förebygga olyckor. Använd det nummer som finns i slutet av varje varning för
att hitta dess översättning i de översatta säkerhetsvarningar som medföljer
denna anordning.
SPARA DESSA ANVISNINGAR
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1
Overview of the
Cisco IP Phone Models 7905G and
7912G
This chapter describes the hardware and software features of the
Cisco IP Phone models 7905G and 7912G and provides a brief overview of the
Session Initiation Protocol (SIP).
This chapter includes the following sections:
•
Hardware Overview, page 1-1
•
Software Features, page 1-4
•
SIP Overview, page 1-7
Hardware Overview
The Cisco IP Phone models 7905G and 7912G are cost-effective, entry-level,
single-line telephones that operate on IP-based telephony networks. They can
function as desk units or be mounted on a wall. The phones are easily installed by
end-users.
Figure 1-1 shows the main components of the Cisco IP Phone models 7905G and
7912G.
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Figure 1-1
Cisco IP Phone Models 7905G and 7912G Features
1
LCD screen
2
Cisco IP Phone Indicates the Cisco IP Phone series to which the phone
series type
belongs.
3
Softkeys
Displays features such as the time, date, your phone
number, caller ID, call status, and softkey tabs.
Enable you to engage any of the functions displayed on
the corresponding LCD screen tabs. Softkeys point to
feature options displayed along the bottom of the LCD
screen. Softkey functions change depending on the status
of your phone (for example, if the phone is active or
idle).
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4
Navigation
button
Enables you to scroll through text and select features
displayed on the LCD screen. Also provides access to
speed dial numbers when there are no text or features to
scroll through.
5
Menu button
Provides access to phone services and settings such as
messages, directories, contrast, ring sound, network
configuration, and status information.
6
Hold button
Puts a current call on hold or takes a call off hold.
7
Dial pad
Works exactly like the dial pad on a traditional
telephone.
8
Volume button
Increases or decreases volume for the handset and
speaker. Also controls the ringer volume (if on-hook).
9
Handset
Functions like a traditional handset. The light strip at the
top of the handset blinks when the phone rings and
remains lit to indicate a new voice mail message
(depending on your message system).
10
Footstand
Allows the phone to stand at a convenient angle on a desk
or table.
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Software Features
Software Features
The Cisco IP Phone models 7905G and 7912G support the following software
features:
•
Protocols, page 1-4
•
Basic Services, page 1-5
•
Supplemental Services, page 1-5
•
SIP-Specific Services, page 1-6
Protocols
The Cisco IP Phone models 7905G and 7912G support the following protocols:
•
Domain Name System (DNS)
•
Dynamic Host Configuration Protocol (DHCP)
•
Internet Control Message Protocol (ICMP)
•
Internet Protocol (IP)
•
Network Time Protocol (NTP)
•
Real-Time Transport Protocol (RTP)
•
Session Initiation Protocol (SIP)
•
Transmission Control Protocol (TCP)
•
Trivial File Transfer Protocol (TFTP)
•
User Datagram Protocol (UDP)
Related Topics
•
Basic Services, page 1-5
•
Supplemental Services, page 1-5
•
SIP-Specific Services, page 1-6
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Basic Services
The Cisco IP Phone models 7905G and 7912G provide the following basic
services:
•
Configuration using LCD screen menus, TFTP server, or web browser
•
IP address assignment—DHCP or manual static IP address
•
Configurable Type of Service (ToS) bit for Quality of Service (QoS)
•
Configurable tone (busy tone, call waiting tone, dial tone, secondary dial
tone, reorder tone, ringback tone)
•
Dial plan support
•
Remote reset support
•
G.711 u-law, G.711 A-law, and G.729A support
•
Voice-activity-detection (VAD) and comfort-noise-generation (CNG)
support for G.711 u-law and G.711 A-law, and support for G.729 Annex B
•
Dynamic jitter buffer for voice packets
Related Topics
•
Protocols, page 1-4
•
Supplemental Services, page 1-5
•
SIP-Specific Services, page 1-6
Supplemental Services
The Cisco IP Phone models 7905G and 7912G support the following
supplemental services:
•
Call waiting
•
Calling line ID presentation/rejection (CLIP/CLIR)
•
Three-way calling
•
Conferencing bridge—up to three parties
•
Call hold
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•
Call transfer—attended and unattended
•
Message-waiting indicator
Related Topics
•
Protocols, page 1-4
•
Basic Services, page 1-5
•
SIP-Specific Services, page 1-6
SIP-Specific Services
The Cisco IP Phone models 7905G and 7912G support the following SIP-specific
services:
•
Direct IP dialing in addition to proxy-routed calls.
•
Configurable periodic registration with a SIP proxy server.
•
Authentication support for all SIP methods.
•
Configurable NAT support.
•
Configurable outbound proxy support.
•
Local support for call return (or call back), call forwarding, and call transfer.
•
FQDN for SIP proxy and outbound proxy.
•
Uses the same configurable MediaPort to transmit and receive RTP audio.
•
Out-of-Band DTMF via RFC2833 (also known as AVT tones).
•
UDP for SIP signaling.
•
Message-waiting indication (MWI) provided via the telephone-handset LED
lamp.
•
Distinctive ringing support via the SIP Alert-Info header.
Related Topics
•
Protocols, page 1-4
•
Basic Services, page 1-5
•
Supplemental Services, page 1-5
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SIP Overview
SIP Overview
Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF)
standard for real-time calls and conferencing over Internet Protocol (IP). SIP is an
ASCII-based, application-layer control protocol that can be used to establish,
maintain, and terminate calls between two or more endpoints.
Like other VoIP protocols, SIP provides signaling and session management
within a packet telephony network. Signaling allows call information to be carried
across network boundaries. Session management controls the attributes of an
end-to-end call.
This section includes these topics:
•
SIP Functions, page 1-7
•
SIP Components, page 1-8
SIP Functions
SIP does the following:
•
Determines the location of the target endpoint—SIP supports address
resolution, name mapping, and call redirection.
•
Determines the media capabilities of the target endpoint—Via Session
Description Protocol (SDP), SIP determines the lowest level of common
services between endpoints. Conferences are established using only the
media capabilities that can be supported by all endpoints.
•
Determines the availability of the target endpoint—If a call cannot be
completed because the target endpoint is unavailable, SIP determines
whether the called party is already on the phone or did not answer in the
allotted number of rings. It then returns a message indicating why the target
endpoint was unavailable.
•
Establishes a session between the originating and target endpoint—If the call
can be completed, SIP establishes a session between the endpoints. SIP also
supports mid-call changes such as adding another endpoint to the conference
and changing media characteristic or codec.
•
Transfers and terminates calls—SIP supports the transfer of calls from one
endpoint to another. During call transfer, SIP simply establishes a session
between the transferee and a new endpoint (specified by the transferring
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party) and terminates the session between the transferee and the transferring
party. At the end of a call, SIP terminates sessions between all parties.
Conferences can consist of two or more users and can be established using
multicast or multiple unicast sessions.
Related Topic
•
SIP Components, page 1-8
SIP Components
SIP is a peer-to-peer protocol. The peers in a session are called user agents (UAs).
A user agent can function in either of two roles:
•
User-agent client (UAC)—A client application that initiates the SIP request.
•
User-agent server (UAS)—A server application that contacts the user when a
SIP request is received and returns a response on behalf of the user.
SIP servers can interact with other back-end application services such as
Lightweight Directory Access Protocol (LDAP) servers, database
applications, and extensible markup language (XML) applications that
provide back-end services such as directory, authentication, and billing
services.
Typically, a SIP endpoint is capable of functioning as either UAC or UAS (but
not both simultaneously) during a transaction. Whether it functions as UAC or
UAS depends on the UA that initiated the request.
Figure 1-2 shows the architecture of a SIP network.
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Figure 1-2
SIP Architecture
SIP proxy and
redirect servers
SIP
SIP
SIP
P user agents
SIP gateway
PSTN
Legacy PBX
72342
RTP
Related Topic
•
SIP Functions, page 1-7
•
SIP Clients, page 1-9
•
SIP Servers, page 1-10
SIP Clients
SIP clients include the following:
•
Telephones—Act as either UAS or UAC. The Cisco IP Phone models 7905G
and 7912G can initiate SIP requests and respond to requests.
•
Gateways—Provide call control. Gateways provide many services, the most
common being translation between SIP conferencing endpoints of
transmission format, communications procedures, and codecs. Other
functions include call setup and clearing on both the LAN side and the
switched-circuit network side.
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SIP Servers
SIP servers include the following:
•
Proxy servers—Receive SIP requests from a client and forward them 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 servers—Receive SIP requests, strip out the address in the request,
check its address tables for any other addresses that may be mapped to the one
in the request, and return the results of the address mapping to the client.
Basically, redirect servers provide the client with information about the next
one or more hops that a message should take and then the client contacts the
next hop server or UAS directly.
•
Registrar servers—Process requests from UACs for registration of their
current location. Registrar servers are often colocated with redirect or proxy
servers.
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2
Installing the Cisco IP Phone Models
7905G and 7912G
This chapter provides information about installing the Cisco IP Phone models
7905G and 7912G at the end user location. It includes the following sections:
•
Prerequisites, page 2-1
•
Safety, page 2-2
•
Connecting to the Network, page 2-4
•
Providing Power to the Cisco IP Phone, page 2-5
•
Installing the Cisco IP Phone Models 7905G and 7912G, page 2-6
•
Verifying Installation, page 2-15
Prerequisites
The Cisco IP Phone models 7905G and 7912G act as endpoints on an IP
telephony network, and have the following network requirements:
•
An Ethernet connection to a network with SIP clients and servers, as required
•
A Trivial File Transfer Protocol (TFTP) server for storing IP phone profiles,
if desired
•
A Dynamic Host Configuration Protocol (DHCP) server, if desired
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Safety
•
A Call Control System
– Proxy server—There must be a device running RFC 2543 SIP-compliant
software
– Voice packet gateway—Required if you are connecting to the Public
Switched Telephone Network (PSTN)
Note
If you use a firewall, Cisco recommends that it be a Cisco PIX firewall, Version
5 or later.
Related Topics
•
Safety, page 2-2
•
Connecting to the Network, page 2-4
•
Providing Power to the Cisco IP Phone, page 2-5
•
Installing the Cisco IP Phone Models 7905G and 7912G, page 2-6
•
Verifying Installation, page 2-15
Safety
To ensure general safety, follow these guidelines:
Warning
•
Do not open or disassemble this product.
•
Do not get this product wet or pour liquids into this device.
•
Do not perform any action that creates a potential hazard to people or makes
the equipment unsafe.
Ultimate disposal of this product should be handled according to all national
laws and regulations. To see translations of the warnings that appear in this
publication, refer to the Regulatory Compliance and Safety Information
document that accompanied this device.
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Safety
Warning
Read the installation instructions before you connect the system to its power
source. To see translations of the warnings that appear in this publication, refer
to the Regulatory Compliance and Safety Information document that
accompanied this device.
Warning
The plug-socket combination must be accessible at all times because it serves
as the main disconnecting device. To see translations of the warnings that
appear in this publication, refer to the Regulatory Compliance and Safety
Information document that accompanied this device.
Warning
Do not work on the system or connect or disconnect cables during periods of
lightning activity. To see translations of the warnings that appear in this
publication, refer to the Regulatory Compliance and Safety Information
document that accompanied this device.
Warning
This equipment is to be installed and maintained by service personnel only as
defined by AS/NZS 3260 Clause 1.2.14.3 Service Personnel. To see translations
of the warnings that appear in this publication, refer to the Regulatory
Compliance and Safety Information document that accompanied this device.
Warning
This product relies on the building’s installation for short-circuit (overcurrent)
protection. Ensure that a fuse or circuit breaker no larger than 120 VAC, 15A U.S.
(240 VAC, 10A international) is used on the phase conductors (all
current-carrying conductors). To see translations of the warnings that appear
in this publication, refer to the Regulatory Compliance and Safety Information
document that accompanied this device.
Warning
The device is designed to work with TN power systems. To see translations of
the warnings that appear in this publication, refer to the Regulatory Compliance
and Safety Information document that accompanied this device.
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Connecting to the Network
Related Topics
•
Connecting to the Network, page 2-4
•
Providing Power to the Cisco IP Phone, page 2-5
•
Installing the Cisco IP Phone Models 7905G and 7912G, page 2-6
•
Verifying Installation, page 2-15
Connecting to the Network
The Cisco IP Phone 7912G has two RJ-45 ports labelled “10/100 SW” and
“10/100 PC.” Each port supports 10/100 Mbps half- or full-duplex connections to
external devices. The Cisco IP Phone 7905G has one RJ-45 port labeled
“10 BASET.” This port supports 10 Mbps half- or full-duplex connections to
external devices. You can use Category 3 or 5 cabling for 10-Mbps connections,
but you must use Category 5 cabling for 100 Mbps connections. Cisco
recommends that you use full-duplex mode on all ports (set at the switch) to avoid
collisions.
Use the 10BASET port on the Cisco IP Phone 7905G or the 10/100 SW port on
the Cisco IP Phone 7912G to connect a phone to the network. You must use a
straight-through cable on these ports. A phone can also obtain inline power over
these connections.
Use the 10/100 PC port on the Cisco IP Phone 7912G to connect a network device,
such as a computer, to the phone. You must use a straight-through cable on this
port.
Figure 2-1 shows the connection ports on the back of the Cisco IP Phone 7905G.
Figure 2-2 shows the connection ports on the back of the Cisco IP Phone 7912G.
Related Topics
•
Safety, page 2-2
•
Providing Power to the Cisco IP Phone, page 2-5
•
Installing the Cisco IP Phone Models 7905G and 7912G, page 2-6
•
Verifying Installation, page 2-15
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Providing Power to the Cisco IP Phone
Providing Power to the Cisco IP Phone
The Cisco IP Phone can be powered by the following sources:
•
External power—optional Cisco AC adapter and power cord for connecting
to a standard wall receptacle.
•
Inline power—provided to the Cisco IP Phone when connected to a
Cisco Catalyst switch that is capable of providing inline power.
Note
•
Only the network port supports inline power from the
Cisco Catalyst switches. The network port is labelled
“10 BASET” on the Cisco IP Phone 7905G and “10/100 SW” on
the Cisco IP Phone 7912G.
WS-PWR-PANEL—power patch panel that allows the Cisco IP Phone to be
connected to existing Catalyst 4000, 5000, and 6000 family 10/100BaseTX
switching modules.
This module sends power on pins 4, 5, 7, & 8, which are not used for Ethernet
signaling. The power patch panel attempts to verify that the attached device
is a Cisco IP Phone before providing power.
Related Topics
•
Safety, page 2-2
•
Connecting to the Network, page 2-4
•
Installing the Cisco IP Phone Models 7905G and 7912G, page 2-6
•
Power Source Design, page 2-5
•
Redundancy Feature, page 2-6
•
Verifying Installation, page 2-15
Power Source Design
The phone and switch automatically determine which power source the phone
uses. If the power switches to a different source, the phone user will experience
different results based on which power source is being used by the phone.
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Use the following information to choose a power source for the phone:
•
If you plug a phone into the optional power supply before plugging it into the
network, the phone is powered by the power supply.
•
If you then unplug the phone from the power supply, the phone resets. If the
switch port is configured for 10/100 Mbps, the switch recognizes the loss of
power and brings the phone back up.
•
If the switch port is configured for 10 Mbps only, then you must unplug the
network connection and plug it back into the phone for the switch to
recognize the phone’s loss of power.
•
If, however, you plugged the network connection into the phone before you
plugged in the power cord, the phone receives power through the switch, and
unplugging the power cord will not bring down the phone. If the switch
reboots, the phone will then be powered by the power cord.
Redundancy Feature
For redundancy, you can use the Cisco AC adapter even if you are using inline
power from the Cisco Catalyst switches. The Cisco 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.
To use this redundancy feature:
1.
Set the inline power mode to auto on the Cisco Catalyst switch.
2.
Connect the unpowered Cisco IP Phone to the network.
3.
Connect the external power supply to the phone after the phone powers up.
Installing the Cisco IP Phone Models 7905G and
7912G
You must install and configure the Cisco IP Phone before it will operate.
Installation consists of making power and network connections. Configuration
consists of setting values for various parameters.
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Most frequently, the configuration parameter-value pairs for each phone are
contained in a configuration file known as a profile that is stored on a TFTP server
on the network. When the phone powers up, it contacts the TFTP server and
downloads its profile. To do so, however, the phone must first receive a basic
configuration that instructs it to download its profile from the TFTP server and
informs it of the server location. This basic configuration can be preprogrammed
on the phone before it is shipped to the end user, or it can be performed by the end
user when the phone is installed.
For more information about configuring a phone, see Chapter 3, “Configuring the
Cisco IP Phone Models 7905G and 7912G for SIP.” For detailed information
about profiles, see Chapter 6, “Configuring the Cisco IP Phone Models 7905G
and 7912G Using a TFTP Server.”
The following sections describe how to install the Cisco IP Phone models 7905G
and 7912G:
•
Physical Installation of the Cisco IP Phone Models 7905G and 7912G,
page 2-7
•
Mounting the Phone to the Wall, page 2-12
•
Basic Configuration of the Cisco IP Phone Models 7905G and 7912G,
page 2-13
Related Topic
•
Safety, page 2-2
Physical Installation of the Cisco IP Phone Models 7905G and
7912G
To install the Cisco IP Phone models 7905G and 7912G, perform the following
steps.
See Figure 2-1 as you install the Cisco IP Phone 7905G, or see Figure 2-2 as you
install the Cisco IP Phone 7912G.
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Warning
To avoid electric shock, do not connect safety extra-low voltage (SELV) circuits
to telephone-network voltage (TNV) circuits. LAN ports contain SELV circuits,
and WAN ports contain TNV circuits. Some LAN and WAN ports both use RJ-45
connectors. Use caution when connecting cables. To see translations of the
warnings that appear in this publication, refer to the Regulatory Compliance
and Safety Information document that accompanied this device.
Caution
Do not cover or block the air vents on the back side of the Cisco IP Phone.
Overheating can cause permanent damage to the unit.
Caution
If you plan to power your phone locally (instead of receiving power through the
Ethernet connection), use only a Cisco 48 volt power supply designed to work
with a Cisco IP Phone.
Note
Do not use hooded cables with the Cisco IP Phone models 7905G and 7912G
because they can cause the phone to rock.
Procedure
Step 1
Use a Category 3 or 5 straight-through Ethernet cable to connect the 10BASET
port on the Cisco IP Phone 7905G or the 10/100 SW port on the Cisco IP Phone
7912G to an Ethernet port.
Step 2
Connect the handset to the handset port.
Ensure that the end of the cord with the longer uncoiled section is connected to
the body of the phone.
Step 3
Connect a Category 3 or 5 Ethernet cable to the 10/100 PC port on the back of the
Cisco IP Phone 7912G (optional).
Use a crossover Ethernet cable to connect from the access port to another Ethernet
device (such as a router or desktop computer) without using a hub. Otherwise, use
a straight-through Ethernet cable.
The Cisco IP Phone 7905G does not have a 10/100 PC port.
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Step 4
If the phone will not receive power through the Ethernet connection, connect the
power supply plug to the DC adapter port on the back of the phone. Then use the
power cable to connect the power supply to a standard power outlet.
After you connect a Cisco IP Phone to its power source, a startup process begins.
After several minutes, the LCD displays a neutral “ready” screen. The details on
the screen might vary, but a ready screen typically displays the date and time,
extension number, and available softkeys. Startup is complete and the phone is
ready to use.
Softkeys point to feature options displayed along the bottom of the LCD screen.
Softkeys are flexible—they change depending on the status of the phone. For
example, one set of softkey options appears when the handset is picked up, and
another set appears when the phone is not in use.
If the phone was preprogrammed with its basic configuration, it begins to
automatically download its complete profile from the TFTP server. the initial
preloaded boot load firmware image is upgraded to a firmware image that
supports a specific signalling protocol.
You or the end user must complete the steps in the “Basic Configuration of the
Cisco IP Phone Models 7905G and 7912G” section on page 2-13 before using the
phone.
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Figure 2-1
Cisco IP Phone 7905G Cable Connections
3
1
5
4
91637
2
1
Network port (10BASE-T)
Connects the phone to the Ethernet port in
your workspace
2
Handset port
Connects the handset to the phone
3
DC adapter port (48VDC)
Connects the power supply plug to the phone
4
Power supply with DC
output connector (if
included)
Connects to the AC power cable
5
AC power cable with wall
socket plug (if included)
Connects the power supply to a standard
power outlet
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Figure 2-2
Cisco IP Phone 7912G Cable Connections
4
1
6
2
5
91638
3
1
Network port (10BASE-T,
100BASE-T)
Connects the phone to the Ethernet port in
your workspace.
2
Access port (10BASE-T,
100BASE-T)
Connects the phone to the Ethernet port in a
desktop computer
3
Handset port
Connects the handset to the phone
4
DC adapter port (48VDC)
Connects the power supply plug to the phone
5
Power supply with DC
output connector (if
included)
Connects to the AC power cable
6
AC power cable with wall
socket plug (if included)
Connects the power supply to a standard
power outlet
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Related Topics
•
Mounting the Phone to the Wall, page 2-12
•
Basic Configuration of the Cisco IP Phone Models 7905G and 7912G,
page 2-13
Mounting the Phone to the Wall
You can mount the Cisco IP Phone on the wall by removing the footstand and
using the mounting bracket on the back of the phone, or you can use special
brackets available in a Cisco IP Phone wall mount kit. (A wall mount kit must be
ordered separately from the phone.) If you attach the Cisco IP Phone to the wall
using the standard footstand and not the wall mount kit, you need to supply the
following tools and parts:
•
Screwdriver
•
Screws to secure the Cisco IP phone to the wall
To mount the phone on the wall using its mounting bracket, follow these steps:
Procedure
Step 1
Remove the footstand from the phone.
Step 2
Modify the handset rest so that the handset remains on the ear-piece rest when the
phone is vertically placed.
a.
Remove the handset from the ear-piece rest.
b.
Locate the tab (handset wall hook) at the base of the ear-piece rest.
c.
Slide this tab out, rotate it 180 degrees (left to right) to expose a lip on which
the handset catches when the phone is vertical, and reinsert this tab.
d.
Place the handset on the ear-piece rest.
For an illustrated procedure of modifying the handset rest, refer to Installing the
Wall Mount Kit for the Cisco IP Phone.
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Step 3
Insert two screws into a wall stud or into wall anchors, matching the screws to the
two screw holes on the back of the phone.
The keyholes fit standard phone jack mounts.
Step 4
Hang the phone on the wall.
Basic Configuration of the Cisco IP Phone Models 7905G and
7912G
The basic configuration of a Cisco IP Phone provides the phone with the minimal
information that it needs to download its complete operational configuration
profile from a TFTP server. This information includes a value to enable TFTP
downloads, a value for the URL for the TFTP server that stores the phone’s
complete configuration file (also known as a profile), and a value for the
encryption key.
The following steps explain how to perform this basic configuration for the
Cisco IP Phone.As administrator, you can preconfigure a phone before it is
shipped to the end user. Then, after the end user connects a phone as described in
the “Physical Installation of the Cisco IP Phone Models 7905G and 7912G”
section on page 2-7, the phone automatically downloads the configuration profile
and the latest phone software from the TFTP server. The phone is ready for use
within seconds.
If preconfiguration is not possible, the end user can perform these steps after
connecting the phone.
Procedure
Step 1
Press the Menu button to access the Services menu.
Step 2
In the Services menu, use the Navigation button to select Settings, and then press
the Select softkey.
Step 3
In the Settings menu, use the Navigation button to select Network
Configuration, and then press the Select softkey.
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Step 4
Unlock network configuration settings.
For instructions, see the “Unlocking and Locking Configuration Parameters”
section on page 5-4.
Step 5
In the Network Configuration menu, use the Navigation button to select
Alternate TFTP, and then press the Yes softkey.
Step 6
In the Network Configuration menu, use the Navigation button to select TFTP
Server, and then press the Edit softkey.
The current value for the IP address of the TFTP server appears in the Current
Value and the New Value fields.
Step 7
Enter a new IP address in the New Value field.
Use the arrow softkey (<<) to delete existing characters. Use the * key on the
keypad to enter a period (.).
Note
When the cursor moves over a character, the character is deleted and a
new character must be entered. You cannot cancel individual character
deletions, but you can use the Cancel softkey to back out of the screen
without saving your changes.
Step 8
Press the Validate softkey.
Step 9
In the Network Configuration menu, use the Navigation button to select Profile
Encrypt Key, and then press the Edit softkey.
Step 10
Enter the encryption key using the keys on the keypad, and then press the
Validate softkey.
For information about entering alpha-numeric characters, see the “Editing
Parameter Values” section on page 5-5.
Step 11
Press the Save softkey.
Step 12
Press the Back softkey and then press the Exit softkey.
The phone downloads its configuration profile and the latest phone software from
the TFTP server.
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Verifying Installation
Verifying Installation
If you hear a dial tone when you lift the handset, you have installed the
Cisco IP Phone properly. If you experience any problems, see Chapter 9,
“Troubleshooting the Cisco IP Phone Models 7905G and 7912G.”
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Verifying Installation
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3
Configuring the
Cisco IP Phone Models 7905G and
7912G for SIP
This chapter provides an overview of configuring the Cisco IP Phone to operate
with the SIP firmware image.
This chapter includes the following sections:
•
Overview of Configuration Methods, page 3-1
•
Default Boot Load Behavior, page 3-2
•
Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation,
page 3-4
•
Basic Configuration Steps in a TFTP Server Environment, page 3-6
•
Minimum Configuration Settings for the Cisco IP Phone Models 7905G and
7912G, page 3-9
Overview of Configuration Methods
You can configure the Cisco IP Phone for SIP by using any of these methods:
•
TFTP server—Recommended method for deploying a large number of
Cisco IP Phones. This method allows you to set up a unique configuration file
for each phone or to set up a configuration file that is common to all
Cisco IP phones. These configuration files are called profiles. The
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Default Boot Load Behavior
Cisco IP Phone can automatically download its latest profile from the TFTP
server when the phone powers up, is refreshed or reset, or when the specified
TFTP query interval expires.
For more information, see Chapter 6, “Configuring the Cisco IP Phone
Models 7905G and 7912G Using a TFTP Server.”
•
Phone configuration menus—Manual method for configuring an individual
phone. Each Cisco IP Phone includes menus for configuring a variety of
network and SIP settings.
For more information, see Chapter 5, “Configuring the Cisco IP Phone
Models 7905G and 7912G Using the Phone Menus.”
•
Phone web page—Convenient if you plan to deploy a small number of
Cisco IP Phones in your network. To use this method, the phone must first obtain
IP connectivity
For more information, see Chapter 7, “Configuring the
Cisco IP Phone Models 7905G and 7912G Using Web Pages.”
Related Topics
•
Default Boot Load Behavior, page 3-2
•
Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation,
page 3-4
•
Basic Configuration Steps in a TFTP Server Environment, page 3-6
•
Minimum Configuration Settings for the Cisco IP Phone Models 7905G and
7912G, page 3-9
Default Boot Load Behavior
Before you configure the Cisco IP Phone, you need to know how the default boot
load process works. Once you understand this process, you will be able to
configure the phone by following the instructions provided in this section and in
the sections that follow.
It is important to understand the boot load process because if your network
environment is not set up to follow this default behavior, you will need to make
the applicable configuration changes. For example, by default, the Cisco IP Phone
attempts to contact a DHCP server for the necessary IP addresses to achieve
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Default Boot Load Behavior
network connectivity. However, if your network does not use a DHCP server, you
must manually configure the IP address of the phone, the subnet mask used by the
phone, and the network gateway (default router) IP address for the phone.
All Cisco IP Phones are shipped with a bootload image. However, because this
image is not a fully-functional signaling image, the image must be upgraded. The
image is designed to be automatically upgraded by a properly configured TFTP
server. To configure the Cisco IP Phone to automatically upgrade to the latest
signaling image, see the “Upgradecode Parameter Detail” section on page 8-41.
In addition, the Cisco IP Phone obtains its configuration file during the bootload
process.
Here is a summary of the default Cisco IP Phone behavior during its boot-up
process:
1.
Note
2.
Note
3.
The Cisco IP Phone uses the Cisco Discovery Protocol (CDP) to discover
which VLAN to enter. If the phone receives a VLAN ID response from the
network switch, the phone enters that VLAN and adds 802.1Q VLAN tags to
its IP packets. If the phone does not receive a response with a VLAN ID from
the network switch, the phone assumes it is not operating in a VLAN
environment and does not perform VLAN tagging on its packets.
If your network environment is not set up to handle this default behavior,
make the necessary configuration changes by referring to the “Specifying
a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation” section
on page 3-4.
The phone contacts the DHCP server to request its own IP address.
If your network does not use a DHCP server, you must manually
configure the IP address of the phone, the subnet mask used by the phone,
and the network gateway (default router) IP address for the phone. For
more information, see the “Minimum Configuration Settings for the
Cisco IP Phone Models 7905G and 7912G” section on page 3-9.
The phone also requests the IP address of the TFTP server from the DHCP
server.
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Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation
4.
The phone downloads a profile from the TFTP server. To do so, it requests
the following profiles from the TFTP server, in the order shown, until a file
exists. Then it downloads that file. In the file names, xxxxxxxxxxxx is the
hexadecimal representation of the phone’s MAC address. (The first three files
are not used for SIP. They are included for compatibility reasons.)
– SEPxxxxxxxxxxxx.cfg.xml
– XMLDefault.cnf.xml
– SEPDefault.cnf
– ldxxxxxxxxxxxx (for the Cisco IP Phone 7905G) or gkxxxxxxxxxxxx (for
the Cisco IP Phone 7912G)
– lddefault.cfg (for the Cisco IP Phone 7905G) or gkdefault.cfg (for the
Cisco IP Phone 7912G)
Related Topics
•
Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation,
page 3-4
•
Basic Configuration Steps in a TFTP Server Environment, page 3-6
•
Minimum Configuration Settings for the Cisco IP Phone Models 7905G and
7912G, page 3-9
Specifying a Preconfigured VLAN ID or Disabling
VLAN IP Encapsulation
If you want the Cisco IP Phone to use a preconfigured VLAN ID instead of using
the Cisco Discovery Protocol to locate a VLAN, or if you want to disable VLAN
IP encapsulation, you may need to configure the following parameters. (For more
information about these parameters, see the “Network Parameters” section on
page 8-2.) See Table 3-1 for information about which VLAN-related parameters
and bits to configure depending on your network environment.
•
OpFlags parameter:
– Bit 4—Enable the use of user-specified voice VLAN ID
– Bit 5—Disable VLAN IP encapsulation
– Bit 6—Disable CDP discovery
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Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation
•
VLANSetting parameter:
– Bits 0-2—Specify VLAN CoS bit value (802.1Q priority) for signaling
packets
– Bits 3-5—Specify VLAN CoS bit value (802.1Q priority) for audio packets
– Bits 18-29—User-specified 802.1Q VLAN ID
Table 3-1
VLAN-Related Features and Corresponding Configuration
Parameters
Feature
VLANSetting
OpFlags Bit 4 OpFlags Bit 5 OpFlags Bit 6 Bits 18-29
Static VLAN
1
0
1
VLAN ID
CDP-acquired
VLAN
0
0
0
Not applicable1
No VLAN
—
1
—
Not applicable1
No CDP
—
—
1
Not applicable1
No CDP and no
VLAN
0
1
1
Not applicable1
1. This variable is not applicable to the feature and the setting of this variable does not affect the feature.
Example
The following procedure shows you how to configure the OpFlags and
VLANSetting parameters to allow the phone to use a user-specified VLAN ID. In
this example, the voice VLAN ID is 115 (in decimal format).
Step 1
Set bits 4-6 of the OpFlags parameter to 1, 0, and 1, respectively. This setting
translates to the following bitmap:
xxxx xxxx xxxx xxxx xxxx xxxx x101 xxxx
The remaining bits of the OpFlags parameter, using all default values, make up
the following bitmap representation:
0000 0000 0000 0000 0000 0000 0xxx 0010
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Therefore, the resulting value of the OpFlags parameter becomes the following
bitmap representation:
0000 0000 0000 0000 0000 0000 0101 0010
In hexadecimal format, this value is 0x00000052.
Step 2
Set bits 18-29 of the VLANSetting parameter to voice VLAN ID 115. This setting
translates to the following bitmap
xx00 0001 1100 11xx xxxx xxxx xxxx xxxx
where 000001110011 is the binary representation of the decimal value 115.
The remaining bits of the VLANSetting parameter, using all default values, make
up the following representation:
00xx xxxx xxxx xx00 0000 0000 0010 1011
Therefore, the resulting value of the VLANSetting parameter becomes the
following bitmap representation:
0000 0001 1100 1100 0000 0000 0010 1011
In hexadecimal format, this value is 0x01cc002b.
Related Topics
•
Default Boot Load Behavior, page 3-2
•
Basic Configuration Steps in a TFTP Server Environment, page 3-6
•
Minimum Configuration Settings for the Cisco IP Phone Models 7905G and
7912G, page 3-9
Basic Configuration Steps in a TFTP Server
Environment
The following table shows the basic steps for upgrading the firmware image for a
Cisco IP Phone, configuring the phone, and making it operational in a typical SIP
environment, which includes a TFTP server.
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Basic Configuration Steps in a TFTP Server Environment
Procedure
Reference
Step 1
Download the desired phone release software See the “Setting Up the TFTP
zip file from the Cisco website and store it on Server with Cisco IP Phone
the TFTP server.
Software” section on page 6-4.
Step 2
Follow these basic steps to create a unique
profile, which actually entails creating two
files:
1.
Create a phone profile that contains
parameters that are common to all
Cisco IP Phones in your network.
2.
Create a unique profile text file that
contains parameters that are specific to a
Cisco IP Phone.
See Chapter 6, “Configuring the
Cisco IP Phone Models 7905G
and 7912G Using a TFTP
Server.”
Make sure to use an Include parameter
in the unique profile to pull in values
from the common configuration file.
3.
Convert the unique configuration file to
binary format.
4.
Place the unique binary configuration
file on the TFTP server.
Step 3
See the “Creating a Default
Optionally, create a default profile called
lddefault.cfg for the Cisco IP Phone 7905G Profile” section on page 6-13.
or gkdefault.cfg for the Cisco IP Phone
7912G, which the phone will download from
the TFTP server only if the unique profile
called ldxxxxxxxxxxxx or gkxxxxxxxxxxxx
does not exist on the TFTP server.
Step 4
Configure the upgradecode parameter so that See the “Upgradecode Parameter
Detail” section on page 8-41.
the phone will obtain the correct signaling
image from the TFTP server when the phone
powers up.
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Step 5
Configuring the Cisco IP Phone Models 7905G and 7912G for SIP
Procedure
Reference
Configure the desired interval for the phone
to contact the TFTP server to check for a
configuration-file update or an upgrade of
the signaling image file.
See the “Network Parameters”
section on page 8-2.
Use the CfgInterval parameter to set this
interval.
Step 6
Configure the method by which the phone
will locate the TFTP server at boot up time.
See the“Configuring the
Cisco IP Phone to Obtain its
Configuration File from the
TFTP Server” section on
page 6-5.
Step 7
Power up the phone.
—
Step 8
See the “Refreshing or Resetting
If you make configuration changes to the
phone or if you upgrade the signaling image the Cisco IP Phone” section on
page 6-17.
on the TFTP server, you can refresh the
phone so that these changes take effect
immediately. Otherwise, these changes will
take effect when the specified interval
(CfgInterval parameter value) for the TFTP
query expires.
Table 4-1 on page 4-2 contains a list of all required SIP parameters. These
parameters must be properly configured for the Cisco IP Phone to work.
For descriptions of important Cisco IP Phone SIP services that you can configure,
and references to their configuration parameters, see the “Important Basic SIP
Services” section on page 4-2 and the “Additional SIP Services” section on
page 4-4.
Table 4-3 on page 4-12 lists, in alphabetical order, various features that you can
configure for the Cisco IP Phone.
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Minimum Configuration Settings for the Cisco IP Phone Models 7905G and 7912G
Related Topics
•
Default Boot Load Behavior, page 3-2
•
Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation,
page 3-4
•
Minimum Configuration Settings for the Cisco IP Phone Models 7905G and
7912G, page 3-9
Minimum Configuration Settings for the
Cisco IP Phone Models 7905G and 7912G
The Cisco IP Phone requires the following minimum settings for network
connectivity:
•
IP address
•
Subnet mask
•
Network gateway (default router) IP address
These settings can be configured using any of the methods described in the
following sections:
•
Making Settings with a DHCP Server, page 3-9
•
Making Settings Manually, page 3-10
Related Topics
•
Default Boot Load Behavior, page 3-2
•
Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation,
page 3-4
•
Basic Configuration Steps in a TFTP Server Environment, page 3-6
Making Settings with a DHCP Server
If your Cisco IP Phone network contains a DHCP server, the Cisco IP Phone
automatically learns its IP address, subnet mask, and network gateway from the
DHCP server when the phone starts up.
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Minimum Configuration Settings for the Cisco IP Phone Models 7905G and 7912G
Making Settings Manually
If your network does not use a DHCP server, you must manually specify the IP
address of the phone, the subnet mask used by the phone, and the network gateway
(default router) IP address for the phone. You can specify this information using
the phone’s Network Configuration menu, using a profile, or using a phone’s web
page.
When you manually specify these values, you must disable DHCP by using the
DHCP Enabled parameter on the Network Configuration menu or the Dhcp
parameter in a profile or web page.
To manually specify the IP address, subnet mask, and default router IP address,
use these parameters:
Setting
Parameter on Network
Configuration Menu
Parameter in Profile or on
Web Page
IP address of a phone
IP Address
StaticIP
Subnet mask used by
the phone
Subnet Mask
StaticNetMask
Network gateway
(default router) IP
address used by the
phone
Default Router
StaticRoute
For detailed information about configuring menus on the phone, see Chapter 5,
“Configuring the Cisco IP Phone Models 7905G and 7912G Using the Phone
Menus.”
For detailed information about configuring through profiles, see Chapter 6,
“Configuring the Cisco IP Phone Models 7905G and 7912G Using a TFTP
Server.”
For detailed information about configuring through a phone’s web page, see
Chapter 7, “Configuring the Cisco IP Phone Models 7905G and 7912G Using
Web Pages.”
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4
Basic and Additional SIP Services
This chapter provides information about configuring SIP services that are
supported on the Cisco IP Phone models 7905G and 7912G. For detailed
information about any parameter mentioned in the following sections, see
Chapter 8, “Profile and Web Page Parameters and Defaults.”
This chapter includes these topics:
•
Important Basic SIP Services, page 4-2—Describes parameters that you must
configure for the Cisco IP Phone to function in a SIP environment.
•
Additional SIP Services, page 4-4—Contains information about additional,
commonly used SIP features, with references to the parameters for
configuring these services.
•
Complete Reference of all Cisco IP Phone SIP Services, page 4-12—
Contains a complete listing of Cisco IP Phone services that are supported for
SIP, and includes cross references to the parameters for configuring these
services. Includes services not described in the sections about the basic SIP
services and the commonly used additional SIP services.
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Basic and Additional SIP Services
Important Basic SIP Services
Important Basic SIP Services
This section provides descriptions and cross references for configuring required
SIP parameters and for configuring other important basic SIP services. This
section includes these topics:
•
Required Parameters, page 4-2
•
Establishing Authentication, page 4-4
•
Configuring Refresh Interval, page 4-4
Required Parameters
Table 4-1 shows the SIP parameters that you must supply for a Cisco IP Phone.
The Parameter fields in this table show each parameter name as it appears on a
phone’s web page or in its profile (“web page or profile”), or on the phone’s
Network Configuration menu or SIP Configuration menu (“on phone menu”).
For more detailed explanations of these parameters, see Chapter 5, “Configuring
the Cisco IP Phone Models 7905G and 7912G Using the Phone Menus,” and
Chapter 8, “Profile and Web Page Parameters and Defaults.” These chapters also
contain information about other configuration parameters.
Table 4-1
Required SIP Parameters and Defaults
Minimum
Value
Maximum Default
Value
Value
Number of seconds between registration
renewal.
1
86400
3600
MAXRedirect
(web page or profile)
Maximum number of times to try
redirection.
0
10
5
SIPRegOn
(web page or profile)
Enable SIP registration.
0
1
0
Parameter
Description
SIPRegInterval
(web page or profile)
Register Expires
(on phone menu)
Register with Proxy
(on phone menu)
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Table 4-1
Required SIP Parameters and Defaults (continued)
Parameter
Minimum
Value
Description
NATIP
(web page or profile)
WAN address of the attached router/NAT;
—
currently only used to support SIP behind a
NAT WAN IP Address NAT.
(on phone menu)
SIPPort
(web page or profile)
Maximum Default
Value
Value
—
0.0.0.0
Port to listen for incoming SIP requests.
1
65535
5060
Base port to receive RTP media.
1
65535
16384
Outbound proxy server IP address or fully
qualified domain name (FQDN) for all
outbound SIP requests.
—
—
0
—
—
0
Local SIP Port
(on phone menu)
MediaPort
(web page or profile)
Local RTP Port
(on phone menu)
OutBoundProxy
(web page or profile)
Outbound Proxy
(on phone menu)
Proxy
(web page or profile)
All SIP requests are sent to OutBoundProxy,
when configured, instead of to the
configured Proxy.
SIP proxy server IP address or fully
qualified domain name (FQDN).
SIP Proxy
(on phone menu)
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Basic and Additional SIP Services
Additional SIP Services
Establishing Authentication
The UseLoginID parameter lets you select the user ID for authentication of the
phone to the remote end as follows:
•
If UseLoginID is set to 0, the user ID (set with the UID parameter) is used
with a user-supplied password (set with the PWD parameter) for
authentication.
•
If UseLoginID is set to 1, you must supply a login ID (set with the LoginID
parameter) and a password (set with the PWD parameter) for authentication.
Configuring Refresh Interval
When the value specified with the CfgInterval parameter is reached, the
Cisco IP Phone attempts to refresh its configuration file from the TFTP server.
(You can manually perform a refresh before the scheduled refresh as explained in
the “Refreshing the Cisco IP Phone” section on page 6-17.)
When a Cisco IP Phone contacts the TFTP server, it also checks to see if a
firmware image has been placed on the TFTP server. If such an image exists, the
phone will download this image.
Additional SIP Services
Table 4-2 describes additional SIP services and provides references to additional
information. These services are listed alphabetically.
Note
Many of the descriptions in Table 4-2 refer to specific parameters that you can set
in a profile or from a phone’s web page. For the corresponding parameter in the
phone configuration menus, see Appendix B, “Phone Menu, Web Page, and
Profile Equivalent Parameters.”
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Additional SIP Services
Table 4-2
Additional SIP Services
SIP Service
Description
Reference
Advanced audio
configuration.
The TOS parameter (which specifies the default IP
precedence of UDP packets) and AudioMode
parameter (which specifies the audio operating mode)
allow you to tune audio configuration.
See the “SIP Parameters”
section on page 8-9 and
the “Audio Parameters”
section on page 8-24.
Call preferences
See the “Call Preference
The GUI Show Mask or CallPrefGuiShow specify
Parameters” section on
which parameters, if any, appear on the Call
Preferences menu on the phone. The GUI Set Mask or page 8-13.
CallPrefGuiSet parameters specify which parameters
on the Call Preferences menu on the phone can be
changed by an end user. You can use these parameters
to offer specific features on a subscription basis
Comfort noise
See the “Audio
When silence suppression is turned on in G.711, the
Cisco IP Phone calculates and transmits its noise level Parameters” section on
to the far end to enable the remote endpoint to generate page 8-24.
the appropriate amount of comfort noise. This feature
provides the remote user with a similar experience to
that of a PSTN call and prevents silent gaps when
neither party is talking.
You enable or disable silence suppression with bit 0 in
the AudioMode parameter.
Debugging
diagnostics
Dial plan
You can use the following parameters to troubleshoot
operation issues:
•
NPrintf—Specifies the IP address and port where
debug information is sent.
•
TraceFlags—Turns on specific trace features.
See the “Network
Parameters” section on
page 8-2.
You can use the DialPlan parameter to set specific dial See the “SIP Parameters”
plan rules and timeout values. Many of these values are section on page 8-9.
determined on a country-by-country basis.
See the “Network
Disabling access To prevent tampering and unauthorized access to the
to the web
Cisco IP Phone configuration, you can use the OpFlags Parameters” section on
interface
parameter to prevent configuring the phone through the page 8-2.
web.
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Additional SIP Services
Table 4-2
Additional SIP Services (continued)
SIP Service
Description
Distinctive
ringing support
If an incoming INVITE contains an Alert-Info header —
with the name Bellcore-dr1, Bellcore-dr2,
Bellcore-dr3, Bellcore-dr4, or Bellcore-dr5, the phone
will play the ring pattern rn, where n = 1, 2, 3, 4, or 5 as
specified in Bellcore GR506. If the Alert-Info header is
not present or if the ring pattern name is unknown, the
phone will play the default ring pattern Bellcore-dr1.
DNS SRV
support
The Cisco IP Phone supports DNS SRV lookup for the See the “SIP Parameters”
section on page 8-9.
SIP proxy server. If the Proxy parameter value is a
domain name, the phone performs a DNS SRV lookup
for the SIP proxy server. A DNS SRV lookup results in
one of the following conditions:
Hardware
information
•
Zero host is returned or DNS SRV lookup failed.
The phone then performs a regular DNS A-record
lookup for the given name.
•
One host is returned. The single host is used as the
primary proxy farm.
•
Two or more hosts are returned. The two hosts with
the highest priorities are used as the primary and
backup proxy farms.
Reference
You can view Cisco IP Phone hardware information on See the “Model
the phone’s Model Information menu and on the Device Information Menu”
Information area in a phone’s web page.
section on page 5-17 and
the “Device Information”
section on page 7-4.
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Additional SIP Services
Table 4-2
Additional SIP Services (continued)
SIP Service
Description
Reference
NAT gateway
Network Address Translation (NAT) supports port
mapping and forwarding to standard default SIP
signaling port 5060 and media base port 16384, or other
ports as configured for the Cisco IP Phone. Media ports
are evenly numbered from the base port. NAT must
support multiple port mappings. The phone can use up
to four media ports to handle conference calls on both
lines. For example, if media base port 16384 is used for
one call, the next call uses port 16386 and other calls
will use ports 16388 and 16390. (Routers such as
D-Link, WinRoute, and WinProxy may not route
correctly if both caller and callee are behind the same
NAT.)
See the “Network
Parameters” section on
page 8-2 and the “SIP
Parameters” section on
page 8-9.
To configure the phone to work in a NAT environment,
make the following settings:
•
Use the StaticRoute parameter to specify the LAN
IP address of the NAT through which the phone
will communicate.
•
Use the NATIP parameter to specify the WAN IP
address of the NAT through which all external SIP
user agents will communicate.
•
Optionally use the SIPPort parameter to specify a
new port for SIP messages.
•
Optionally use the MediaPort parameter to specify
a new base port for RTP media.
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Additional SIP Services
Table 4-2
Additional SIP Services (continued)
SIP Service
Description
Reference
NAT/PAT
translation
See the “SIP Parameters”
To maintain Network Address Translation/Port
section on page 8-9.
Address Translation (NAT/PAT) for a session, the
Cisco IP Phone can be configured to periodically send
a dummy UDP packet to a server (the phone does not
expect any response from the server).
To specify this configuration, use these parameters:
Network timing
•
NatTimer—Bits 0 to 11 are for specifying the
retransmission period.
•
NatServer—Specifies the server to which the
dummy packet is sent.
You can fine tune your network timing by making the
following settings:
•
Use the TimeZone parameter to specify the
time-stamp for incoming calls (offset from
Greenwich Mean Time).
•
Use the NTPIP parameter to specify the IP address
of the Network Time Protocol (NTP) server. NTP
is a protocol built on top of TCP that ensures
accurate local time-keeping with reference to radio
and atomic clocks located on the Internet.
•
Use the AltNTPIP parameter to specify an
alternate NTP server IP address.
•
Use the ConnectMode parameter to specify the
connection mode of the SIP protocol.
See the “Network
Parameters” section on
page 8-2 and the “Audio
Parameters” section on
page 8-24.
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Additional SIP Services
Table 4-2
Additional SIP Services (continued)
SIP Service
Description
Reference
Proxy
Redundancy
support
See the “SIP Parameters”
The Cisco IP phone supports next hop server
section on page 8-9.
redundancy when a fully qualified domain name
(FQDN) is specified for the Proxy or Outbound Proxy
parameter. When sending a SIP request, the phone
performs a DNS SRV records query for the domain
name. The phone selects two records from the result
and identifies them as Primary Proxy Farm (PPF) and
Alternate Proxy Farm (APF). Each farm may contain
up to five IP addresses. If a DNS SRV record does not
exist, the phone performs a DNS A records query for
the domain name, which is now considered the Primary
Proxy Farm with no Alternate Proxy Farm. The phone
accepts up to five IP addresses from the query result.
The phone retries with the alternate next hop server for
the following SIP requests: REGISTER, INVITE,
CANCEL, BYE, REFER, and NOTIFY. If sending one
of these requests results in a 5xx response, no response,
or ARP timeout, the phone attempts the same request
with an alternate destination IP address derived from
the next hop domain name. If both primary and
alternate farms exist, the phone attempts the primary
farm first and then the alternate farm. The phone will
attempt all the IP addresses in one farm before
switching to the other farm. Once the phone finds a
responding next hop server, it remembers the farm and
IP address of the server and keeps using it for the given
domain name until it fails or expires. If the request still
fails after all the possible IP addresses are attempted,
the phone considers the request as failed.
If a server in the alternate proxy farm is active at the
moment, the phone can be configured to retry the
primary proxy farm after the number of seconds
specified by the AltProxyTimeout parameter. If this
value is 0, the phone remains with the alternate proxy
farm until all the IP addresses in the farm are down
before attempting the primary proxy farm again.
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Additional SIP Services
Table 4-2
Additional SIP Services (continued)
SIP Service
Description
Reference
Progress tones
Values for the following parameters must be
determined based on the country in which the
Cisco IP Phone is located:
See the “Tone
Parameters” section on
page 8-21.
•
DialTone
•
DialTone2
•
BusyTone
•
ReorderTone
•
RingBackTone
•
CallWaitTone
Receiver-tagged
VIA header
You can use the ConnectMode parameter to disable or See the “Audio
enable the processing of the received = parameter in the Parameters” section on
Via header. This feature is disabled by default.
page 8-24.
Setting up and
placing a call
without using a
SIP proxy
You can user the Cisco IP Phone to make a direct
IP-to-IP calls without using a SIP proxy.
See the “Setting Up and
Placing a Call Without
Using a SIP Proxy”
section on page 4-10.
Setting Up and Placing a Call Without Using a SIP Proxy
The Cisco IP Phone models 7905G and 7912G support direct IP-to-IP calls
without using a SIP proxy. When a call is placed, the phone sends the INVITE
request directly to the remote user agent and expects the usual 100/180/200
responses from the user agent.
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Additional SIP Services
To configure a Cisco IP Phone for calls that do not use a SIP proxy, set the
following parameters to the values shown:
Parameter on SIP
Configuration Menu
Parameter in Profile
or on Web Page
Setting
SIP Proxy
Proxy
0 (zero) or No
User ID
UID
Telephone number of the
phone
Register with Proxy
SipRegOn
0 (zero) or No (disables SIP
registration with a SIP proxy
server)
To place an IP call, dial the telephone number and the IP address of the remote
user agent using this format:
phone_number**IPaddress
Use the star (*) key on the telephone keypad to represent the dot (.) in an IP
address. If you are using on-hook dialing, press the Dial softkey to terminate the
dial string and place the call. If you are using off-hook dialing, press the pound
(#) key on the telephone keypad to terminate the dial string and place the call
Note
URL dialing is not supported.
For example, to place a call to a user agent with an ID of 408-555-1212 at IP
address 192.168.1.100, enter the following string on your telephone keypad:
4085551212**192*168*1*100
Related Topics
•
Chapter 5, “Configuring the Cisco IP Phone Models 7905G and 7912G Using
the Phone Menus,”
•
Chapter 6, “Configuring the Cisco IP Phone Models 7905G and 7912G Using
a TFTP Server,”
•
Chapter 7, “Configuring the Cisco IP Phone Models 7905G and 7912G
Using Web Pages.”
•
Chapter 8, “Profile and Web Page Parameters and Defaults.”
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Complete Reference of all Cisco IP Phone SIP Services
Complete Reference of all Cisco IP Phone SIP
Services
Table 4-3 lists all configurable features for the Cisco IP Phone using SIP. It also
lists the names of the parameters used to configure the features.
Table 4-3
Configurable Features and Related Parameters
Configurable Feature
Related Parameter in Profile Related Parameter on
or on Web Page
Phone Menu
802.1Q packet tagging
VLANSetting
Admin. VLAN Id
Backup proxy timeout
AltProxyTimeOut
Backup Proxy Timeout
Call forward
CallForwardNumber
(in profile)
Call Forward Number
Call Forward Number
(on web page)
Call forwarding—Maximum times allowed MAXRedirect
—
Call Preferences
Parameters on Call
Preference menu
GUI Show Mask and
GUI Set Mask
(on web page)
CallPrefGuiShow and
CallPrefGuiSet
(in profile)
Call waiting tone period
SigTimer
—
Cisco Discovery Protocol
OpFlags
—
Configuration update interval
CfgInterval
—
Debug messages—configuring host
NPrintf
—
DialPlan
DialPlan
—
Primary domain name server (DNS)
DNS1IP
DNS Server 1
DTMF signaling method
AudioMode
—
Encryption
EncryptKey
Profile Encrypt Key
IDs for phone lines
UID
User ID
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Complete Reference of all Cisco IP Phone SIP Services
Table 4-3
Configurable Features and Related Parameters (continued)
Configurable Feature
Related Parameter in Profile Related Parameter on
or on Web Page
Phone Menu
IP-like address in dial plan
IPDialPlan
—
Login ID
LoginID
Login ID
NTP IP address
NATIP
NAT WAN IP Address
Outbound proxy
OutBoundProxy
Outbound Proxy
Password for registration and
authentication
PWD
Password
Primary NTP server
NTPIP
NTP Server 1
Received= tag processing enable/disable
ConnectMode
—
Receiving-audio codec preference
RxCodec
—
Refresh phone using web server
OpFlags
—
REGISTER action types
ConnectMode
—
Registration removal
ConnectMode
—
Reset phone using web server
OpFlags
—
Retransmission interval for NAT server
NatTimer
—
Ringback tone—send to caller
ConnectMode
—
Ring-cadence pattern
RingOnOffTime
—
RTP media port
MediaPort
Local RTP Port
RTP packet size
NumTxFrames
—
RTP statistics
TraceFlags
—
Secondary domain name server
DNS2IP
DNS Server 2
Secondary NTP server
AltNTPIP
NTP Server 2
Server to which to send dummy packets to
maintain port mappings
NatServer
—
Silence compression
AudioMode
—
SIP proxy address
Proxy
SIP Proxy
SIP registration renewal interval
SIPRegInterval
Register Expires
SIP registration enable/disable
SIPRegOn
Register with Proxy
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Complete Reference of all Cisco IP Phone SIP Services
Table 4-3
Configurable Features and Related Parameters (continued)
Configurable Feature
Related Parameter in Profile Related Parameter on
or on Web Page
Phone Menu
SIP request sending and receiving port
SIPPort
Local SIP Port
Static network router probing
OpFlags
—
TFTP file—not using internally generated
name
OpFlags
—
Timeout values
SigTimer
—
Time zone offset
TimeZone
Time Zone
Tones: BusyTone, CallWaitTone,
DialTone, DialTone2, ReorderTone, and
RingBackTone parameters
Tone Parameters
—
Tracing
TraceFlags
—
Transmitting-audio codec preference
TxCodec
—
UDP packet default IP precedence
TOS
—
VLAN encapsulation
OpFlags
—
VLAN mode
OpFlags
—
WAN address of NAT
NATIP
NAT WAN IP Address
Web configuration—disallowing
OpFlags
—
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5
Configuring the Cisco IP Phone
Models 7905G and 7912G Using the
Phone Menus
The Cisco IP Phone includes configurable network and SIP parameters, which
you might need to modify before the phone is accessible and functional for your
users.
You can configure and view many parameters through configuration and status
menus on the phone, as described in this chapter. You can also view and configure
most parameters through web pages and all parameters through profiles. For more
information, see Chapter 6, “Configuring the Cisco IP Phone Models 7905G and
7912G Using a TFTP Server,” and Chapter 7, “Configuring the
Cisco IP Phone Models 7905G and 7912G Using Web Pages.”
Appendix B, “Phone Menu, Web Page, and Profile Equivalent Parameters,” shows
the configurable parameters on the phone menus and their equivalent web page
and profile parameters.
Note
If you configure a phone using a TFTP server, parameter values in a profile will
overwrite values that you set from the phone menus for the same parameters.
This chapter includes the following sections:
•
Configuration Menus on the Cisco IP Phone, page 5-2
•
Network Configuration Menu, page 5-6
•
SIP Configuration Menu, page 5-14
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Configuration Menus on the Cisco IP Phone
•
Model Information Menu, page 5-17
•
Status Menu, page 5-18
Configuration Menus on the Cisco IP Phone
The Cisco IP Phone models 7905G and 7912G include the following
configuration, information, and status menus. You access each of these menus
from the Settings menu on the phone.
•
Network Configuration menu—Lets you view and configure a variety of
network parameters. For more information, see the “Network Configuration
Menu” section on page 5-6.
•
SIP Configuration menu—Lets you configure variety of SIP signaling
parameters. For more information, see the “SIP Configuration Menu” section
on page 5-14.
•
Model Information menu—Displays hardware and software information
about phone. For more information, see the “Model Information Menu”
section on page 5-17.
•
Status menu—Provides options for displaying status messages, network
statistics, and firmware versions. For more information, see the “Status
Menu” section on page 5-18.
•
Call Preferences menu—Controls various user features on the phone. For
more information, see the “Call Preferences Menu” section on page A-1.
Before you can change parameter values on the Network Configuration menu and
the SIP Configuration menu, you must unlock the parameters that appear on these
menus. See the “Unlocking and Locking Configuration Parameters” section on
page 5-4 for instructions.
For information about the keys you can use to edit or change parameters, see the
“Editing Parameter Values” section on page 5-5.
You can also change many of the network configuration and SIP configuration
parameters from the phone’s profile or web page. For more information, see
Chapter 6, “Configuring the Cisco IP Phone Models 7905G and 7912G Using a
TFTP Server,” and Chapter 7, “Configuring the Cisco IP Phone Models 7905G
and 7912G Using Web Pages.”
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Configuration Menus on the Cisco IP Phone
This section includes the following topics:
•
Displaying a Configuration Menu, page 5-3
•
Unlocking and Locking Configuration Parameters, page 5-4
•
Editing Parameter Values, page 5-5
Displaying a Configuration Menu
To display a configuration menu, follow these steps:
Procedure
Step 1
Press the Menu button to access the Services menu.
Step 2
In the Services menu, use the Navigation button to select Settings, and then press
the Select softkey.
Tip
Step 3
You can access a menu option directly by pressing on the keypad the
number that appears next to the option.
In the Settings menu, use the Navigation button to select the desired menu, and
then press the Select softkey.
To exit a configuration menu and return to the main LCD screen, press the Back
softkey until the Exit softkey appears, and then press the Exit softkey.
Related Topics
•
Unlocking and Locking Configuration Parameters, page 5-4
•
Editing Parameter Values, page 5-5
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Configuration Menus on the Cisco IP Phone
Unlocking and Locking Configuration Parameters
Configuration parameters in the Network Configuration menu and the SIP
Configuration menu on a phone are locked by default to prevent users from
making changes that could affect the operation of a phone.
When parameters are inaccessible for modification, a locked padlock icon appears
on the Network Configuration menu and the SIP Configuration menu. When
parameters are unlocked and accessible for modification, an unlocked padlock
icon appears on these menus, as shown below.
You must unlock the network configuration parameters or the SIP configuration
parameters before you can configure them. If a password has been specified for
the phone, you must enter the password when unlocking configuration
parameters. (For more information about passwords, see the explanation of
UIPassword in Table 8-6 on page 8-26.)
Network configuration parameters and SIP configuration parameters are locked
again automatically when you exit the Settings menu.
To unlock network configuration and SIP configuration parameters on the
Cisco IP Phone, follow these steps:
Procedure
Step 1
Press the Menu button.
Step 2
Use the Navigation button to select Settings, and then press the Select softkey.
Step 3
Use the Navigation button to select Network Configuration or SIP
Configuration, and then press the Select softkey.
Step 4
From the Network Configuration menu or the SIP Configuration menu, press **#.
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Configuration Menus on the Cisco IP Phone
Step 5
If your phone displays “Enter Admin Password,” enter your password and then
press the Enter softkey.
See the “Editing Parameter Values” section on page 5-5 for information about
entering numbers or letters.
When parameters are unlocked, the icon in the upper-right portion of your LCD
appears as an unlocked padlock.
Related Topic
•
Editing Parameter Values, page 5-5
Editing Parameter Values
When you edit the value of a configuration parameter, follow these guidelines:
Note
•
Use the keys on the dial pad to enter characters (such as numbers, letters, and
periods).
•
If a parameter value can accept either numbers or letters, the Number / Alpha
toggle softkey will be available. Press the Number softkey to enter numbers
using the dial pad. Press the Alpha softkey to enter letters using the dial pad.
•
To enter letters using the dial pad, use a corresponding number key. Press the
key one or more times to display a particular letter. For example, press the 2
key once for “a,” twice quickly for “b,” and three times quickly for “c.” After
you pause, the cursor automatically advances so you can enter the next letter.
•
To enter a period (for example, in an IP address), press the * key on the dial
pad.
•
Press the << softkey if you make a mistake. This softkey deletes the character
to the left of the cursor.
•
Press the Cancel softkey before pressing the Validate softkey or the Save
softkey to discard any changes you have made.
After you change and save some parameter values, the phone may reset
automatically.
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Network Configuration Menu
Related Topic
•
Unlocking and Locking Configuration Parameters, page 5-4
Network Configuration Menu
The Network Configuration menu lets you configure a variety of network
parameters. Table 5-1 describes these parameters and explains how to change
them.
Note
Table 5-1
The Edit, Yes, or No softkeys for changing network configuration parameters
appear only if parameters are unlocked as described in the “Unlocking and
Locking Configuration Parameters” section on page 5-4. For information about
the keys you can use to edit parameters, see the “Editing Parameter Values”
section on page 5-5.
Network Configuration Menu Parameters
Parameter
Description
To Change
DHCP Server
Display only—cannot configure.
IP address of the Dynamic Host
Configuration Protocol (DHCP)
server from which the phone
obtains information such as its IP
address, its subnet mask, and the IP
addresses of the network gateway
and the DNS, NTP, and TFTP
servers.
MAC Address
Unique Media Access Control
(MAC) address of the phone.
Display only—cannot configure.
Host Name
Unique host name of the phone.
Display only—cannot configure.
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Network Configuration Menu
Table 5-1
Network Configuration Menu Parameters (continued)
Parameter
Description
Domain Name
Network domain in which the
phone resides. You can change this
value if the Alternate Domain
parameter is set to Yes.
IP Address
To Change
Internet Protocol (IP) address of
the phone. You can use this
parameter to assign the IP address
if DHCP does not provide it.
If you assign an IP address with
this parameter, you must also
assign a subnet mask and network
gateway (default router). See the
Subnet Mask and the Default
Router parameters in this table.
Subnet Mask
Subnet mask used by the phone.
You can use this parameter to
assign a subnet mask if DHCP does
not provide it.
If you assign a Subnet Mask with
this parameter, you must also
assign an IP address and a network
gateway (default router) for the
phone. See the IP Address and the
Default Router parameters in this
table.
1.
Unlock configuration parameters.
2.
Set the Alternate Domain parameter to
Yes.
3.
Scroll to the Domain Name parameter,
press the Edit softkey, and then enter a
new domain name.
4.
Press the Validate softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Set the DHCP Enabled parameter to No.
3.
Scroll to the IP Address parameter, press
the Edit softkey, and then enter a new IP
Address.
4.
Press the Validate softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Set the DHCP Enabled parameter to No.
3.
Scroll to the Subnet Mask parameter,
press the Edit softkey, and then enter a
new subnet mask.
4.
Press the Validate softkey, and then press
the Save softkey.
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Network Configuration Menu
Table 5-1
Network Configuration Menu Parameters (continued)
Parameter
Description
TFTP Server
IP address or URL of the TFTP
server from which a phone obtains
a profile. Required if the DHCP
server does not provide the TFTP
server address.
If you set the Alternate TFTP
parameter to Yes, you must enter a
non-zero value for the TFTP
Server parameter.
Default Router
IP address of the network gateway
used by the phone. You can use this
parameter to assign this address if
DHCP does not provide it.
If you assign a network gateway
address with this parameter, you
must also assign an IP address and
a subnet mask for the phone. See
the IP Address and the Subnet
Mask parameters in this table.
DNS Server 1,
DNS Server 2
IP addresses of the Primary
Domain Name System (DNS)
server (DNS Server 1) and
secondary DNS server (DNS
Server 2) used by the phone.
To Change
1.
Unlock configuration parameters.
2.
Set the TFTP Enabled parameter to Yes.
3.
Set the Alternate TFTP parameter to Yes.
4.
Scroll to the TFTP Server parameter,
press the Edit softkey, and then enter a
new TFTP server IP address or URL.
5.
Press the Validate softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Set the DHCP Enabled parameter to No.
3.
Scroll to the Default Router parameter,
press the Edit softkey, and then enter a
new network gateway IP address.
4.
Press the Validate softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Set the Alternate DNS parameter to Yes.
3.
Scroll to the DNS Server 1 parameter or
the DNS Server 2 parameter, press the
Edit softkey, and then enter a new DNS
server IP address.
4.
Press the Validate softkey, and then press
the Save softkey.
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Network Configuration Menu
Table 5-1
Network Configuration Menu Parameters (continued)
Parameter
Description
NTP Server 1,
NTP Server 2
IP addresses of the Primary
Network Time Protocol (NTP)
server (NTP Server 1) and
secondary NTP server (NTP Server
2) used by the phone.
Time Zone
To Change
Offset to apply to the Greenwich
Mean Time (GMT) returned by an
NTP server to determine local time
(to use for Call ID display, for
example).
Local time is generated as follows:
•
Local time =
GMT + Time Zone,
if Time Zone <= 12.
•
Local time =
GMT + Time Zone – 25,
if Time Zone > 12.
1.
Unlock configuration parameters.
2.
Set the Alternate NTP parameter to Yes.
3.
Scroll to the NTP Server 1 parameter or
the NTP Server 2 parameter, press the
Edit softkey, and then enter a new NTP
server IP address.
4.
Press the Validate softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Time Zone parameter, press
the Edit softkey, and then enter a new
time zone offset value.
3.
Press the Validate softkey, and then press
the Save softkey.
For example, Time Zone = 17 for
Pacific Standard Time.
Note
You must update this
parameter manually when
daylight savings time goes
into effect and ends.
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Network Configuration Menu
Table 5-1
Network Configuration Menu Parameters (continued)
Parameter
Description
To Change
Operational
VLAN Id
Auxiliary Virtual Local Area
Network (VLAN) configured on a
Cisco Catalyst switch in which the
phone is a member.
The phone obtains its Operational VLAN Id
via Cisco Discovery Protocol (CDP) from the
switch to which the phone is attached. To
assign a VLAN ID manually, use the Admin
VLAN Id parameter.
If the phone has not received an
auxiliary VLAN, this parameter
indicates the Administrative
VLAN.
If neither the auxiliary VLAN nor
the Administrative VLAN are
configured, this parameter is blank.
Admin. VLAN Id Auxiliary VLAN in which the
phone is a member.
1.
Unlock configuration parameters.
2.
Scroll to the Admin. VLAN Id parameter,
press the Edit softkey, and then enter a
new Admin VLAN value.
3.
Press the Validate softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the VLAN Enabled parameter
and press the No softkey to disable
VLAN, or press the Yes softkey to enable
VLAN.
3.
Press the Save softkey.
Overrides the value specified by
the Operation VLAN Id parameter.
VLAN Enabled
Indicates whether VLAN is
enabled for the phone.
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Network Configuration Menu
Table 5-1
Network Configuration Menu Parameters (continued)
Parameter
Description
To Change
CDP Enabled
Indicates whether Cisco Discovery
Protocol (CDP) is enabled for the
phone. CDP allows the phone to be
discovered by the CiscoWorks
network management system.
For more information about
CiscoWorks2000, refer to the
documentation available at this
URL:
1.
Unlock configuration parameters.
2.
Scroll to the CDP Enabled parameter and
press the No softkey to disable CDP, or
press the Yes softkey to enable CDP.
3.
Press the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the DHCP Enabled parameter
and press the No softkey to disable
DHCP, or press the Yes softkey to enable
DHCP.
3.
Press the Save softkey.
http://www.cisco.com/univercd/cc
/td/doc/product/rtrmgmt/cw2000/i
ndex.htm
DHCP Enabled
Specifies whether to use the DHCP
server to obtain information such
as its IP address, its subnet mask,
and the IP addresses of the network
gateway and the DNS, NTP, and
TFTP servers.
If this parameter is 0, you must
manually specify the IP address of
the phone, and the subnet mask and
the network gateway (default
router) used by the phone. See the
“Minimum Configuration Settings
for the Cisco IP Phone Models
7905G and 7912G” section on
page 3-9 for more information.
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Network Configuration Menu
Table 5-1
Network Configuration Menu Parameters (continued)
Parameter
Description
TFTP Enabled
Enables or disables downloading
of a profile from a TFTP server. If
you set this parameter to Yes and
subsequently make changes using
the web interface, you must set this
parameter to No before saving
those changes; otherwise, the
changes will be overwritten by the
profile obtained from a TFTP
server.
To Change
1.
Unlock configuration parameters.
2.
Scroll to the TFTP Enabled parameter
and press the No softkey to disable TFTP,
or press the Yes softkey to enable TFTP.
3.
Press the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Alternate TFTP parameter
and press the Yes softkey if the phone
should use an alternative TFTP server.
Press the No softkey otherwise.
3.
Press the Save softkey.
If you set this parameter to Yes and
do not specify a TFTP server, the
phone will use the TFTP IP address
or URL obtained from the DHCP
server to contact the TFTP server.
If you set this parameter to Yes, set
the Alternate TFTP parameter to
Yes, and specify a value for the
TFTP Server parameter, the phone
will use the server that you specify.
Alternate TFTP
Indicates whether the phone
obtains the IP address or URL of
the TFTP server from the DHCP
server (No) or uses the IP address
or URL specified by the TFTP
Server parameter (Yes).
If you set the Alternate TFTP
server to Yes, you must specify a
value for the TFTP Server
parameter.
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Network Configuration Menu
Table 5-1
Network Configuration Menu Parameters (continued)
Parameter
Description
To Change
Alternate DNS
Indicates whether the phone
obtains the IP address of the DNS
server from the DHCP server (No)
or uses the IP addresses specified
by the DNS Server 1 and,
optionally, the DNS Server 2
parameters (Yes).
1.
Unlock configuration parameters.
2.
Scroll to the Alternate DNS parameter
and press the Yes softkey if the phone
should use an alternative DNS. Press the
No softkey otherwise.
3.
Press the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Alternate NTP parameter
and press the Yes softkey if the phone
should use an alternative NTP server.
Press the No softkey otherwise.
3.
Press the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Alternate Domain parameter
and press the Yes softkey if the phone
should use an alternative domain name.
Press the No softkey otherwise.
3.
Press the Save softkey.
If you set the Alternate DNS server
to Yes, you must specify a value
for the DNS Server 1 parameter.
You may also specify a value for
the DNS Server 2 parameter.
Alternate NTP
Indicates whether the phone
obtains the IP address of the NTP
server from the DHCP server (No)
or uses the IP addresses specified
by the NTP Server 1 and,
optionally, the NTP Server 2
parameters (Yes).
If you set the Alternate NTP server
to Yes, you must specify a value
for the NTP Server 1 parameter.
You may also specify a value for
the NTP Server 2 parameter.
Alternate
Domain
Indicates whether the phone is
using an alternative domain name.
If you set this parameter to Yes,
you must specify a value for the
Domain Name parameter.
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SIP Configuration Menu
Table 5-1
Network Configuration Menu Parameters (continued)
Parameter
Description
Profile Encrypt
Key
Specifies an encryption key used to
decrypt the profile that the phone
downloads from the TFTP server.
For more information about
encryption keys, see “Using
Encryption” section on page 6-15.
Erase
Configuration
Changes these settings to their
default values: Network
Configuration menu settings
(including password), SIP
Configuration menu settings, Call
Preferences menu settings, volume
settings, and contrast settings.
To Change
1.
Unlock configuration parameters.
2.
Scroll to the Profile Encryption Key
parameter, press the Edit softkey, and
then enter a new encryption key.
3.
Press the Validate softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Erase Configuration
parameter, press the Yes softkey, and then
press the Save softkey.
SIP Configuration Menu
The SIP Configuration menu lets you configure a variety of SIP signaling
parameters. Table 5-2 describes these parameters and explains how to change
them.
Note
The Edit, Yes, or No softkeys for changing SIP configuration parameters appear
only if parameters are unlocked as described in the “Unlocking and Locking
Configuration Parameters” section on page 5-4. For information about the keys
you can use to edit parameters, see the “Editing Parameter Values” section on
page 5-5.
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SIP Configuration Menu
Table 5-2
SIP Configuration Menu Parameters
Parameter
Description
SIP Proxy
IP address or URL of the SIP proxy
server used by the phone.
User ID
Password
Use Login ID
Login ID
To Change
User ID for registration and
authentication of the phone.
Password for registration and
authentication of the phone.
Indicates whether to use the value
specified with the Login ID
parameter instead of the value
specified with the User ID
parameter for registration and
authentication of the phone.
Alternate ID used for
authentication of the phone.
1.
Unlock configuration parameters.
2.
Scroll to the SIP Proxy parameter, press
the Edit softkey, and then enter the IP
address or URL of a SIP proxy server.
3.
Press the Accept softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the User ID parameter, press the
Edit softkey, and then enter a user ID.
3.
Press the Accept softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Password parameter, press
the Edit softkey, and then enter a
Password.
3.
Press the Accept softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Use Login ID parameter, and
then press the Yes softkey to use the value
specified with the Login ID parameter for
registration and authentication.
3.
Press the Save softkey.
1.
Unlock configuration parameters.
2.
Set the Use Login ID parameter to Yes.
3.
Scroll to the Login ID parameter, press
the Edit softkey, and then enter a login
ID.
4.
Press the Accept softkey, and then press
the Save softkey.
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SIP Configuration Menu
Table 5-2
SIP Configuration Menu Parameters (continued)
Parameter
Description
Local SIP Port
Port where the phone listens for
incoming requests and sends
outgoing requests.
Local RTP Port
Backup Proxy
Timeout
Port from which the phone
transmits and receives media
streams.
Timeout, in seconds, for switching
from backup proxy server to
primary proxy server.
The value can range from 0 to
4294967295.
Outbound Proxy
IP address or URL of the SIP
outbound proxy server used by the
phone.
To Change
1.
Unlock configuration parameters.
2.
Scroll to the Local SIP Port parameter,
press the Edit softkey, and then enter a
SIP port from 5060 to 65535.
3.
Press the Accept softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Local RTP Port parameter,
press the Edit softkey, and then enter a
SIP port, which must be an even number
from 0 to 65535.
3.
Press the Accept softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Backup Proxy Timeout
parameter, press the Edit softkey, and
then enter a timeout value.
3.
Press the Accept softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Outbound Proxy parameter,
press the Edit softkey, and then enter the
IP address or URL of the outbound proxy
server.
3.
Press the Accept softkey, and then press
the Save softkey.
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Model Information Menu
Table 5-2
SIP Configuration Menu Parameters (continued)
Parameter
Description
To Change
Register Expires
SIP registration interval, in
seconds, between each registration
renewal to the SIP proxy server.
The value can range from 1 to
86400.
Register with
Proxy
NAT WAN IP
Address
Indicates whether to enable SIP
registration. This parameter is
ignored if a SIP proxy server is not
specified.
External Network Address
Translation (NAT) IP address
where other SIP user agents can
communicate with the phone.
1.
Unlock configuration parameters.
2.
Scroll to the Register Expires parameter,
press the Edit softkey, and then enter the
SIP registration interval.
3.
Press the Accept softkey, and then press
the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the Register with Proxy and
press the Yes softkey to enable SIP
registration, or press the No softkey to
disable SIP registration.
3.
Press the Save softkey.
1.
Unlock configuration parameters.
2.
Scroll to the NAT WAN IP Address
parameter, press the Edit softkey, and
then enter the NAT IP address.
3.
Press the Accept softkey, and then press
the Save softkey.
Model Information Menu
The Model Information menu provides information that can assist with
troubleshooting the phone.
The Model Information menu shows the following information:
•
Model Number—The model number of the phone.
•
MAC Address—The MAC address of the phone.
•
App Load ID—Identifier of the firmware running on the phone.
•
Boot Load ID—Identifier of the factory-installed load running on the phone.
•
Serial Number—Serial number of the phone.
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Status Menu
Status Menu
The Status menu contains the following options, which provide information about
the phone and its operation:
•
Status Messages—Reserved for future use.
•
Network Statistics—Displays Ethernet traffic statistics.
•
Firmware Versions—Displays this information about the software and
firmware running on the phone:
– Application Load ID—Identifier of the firmware.
– Boot Load ID—Identifier of the factory-installed load.
– Version—Software version number.
To access the Status menu, choose Status from the Settings menu.
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6
Configuring the Cisco IP Phone
Models 7905G and 7912G Using a TFTP
Server
This chapter explains how to configure the Cisco IP Phone using a Trivial File
Transfer Protocol (TFTP) server. This configuration method is useful when you
have many Cisco IP Phones because you can use a TFTP server for remote, batch
configuration of the phones. The TFTP server can also host one unique
configuration file for each phone.
To obtain configuration information from a TFTP server, a phone downloads a
configuration file, called a profile. A profile contains various parameters and
values for phone and network settings.
You can also configure many parameters for a Cisco IP Phone by using the
configuration menus on a phone and by using the phone’s web page. For more
information, see Chapter 5, “Configuring the Cisco IP Phone Models 7905G and
7912G Using the Phone Menus,” and Chapter 7, “Configuring the
Cisco IP Phone Models 7905G and 7912G Using Web Pages.”
For an explanation of the parameters that you can use in a profile, see Chapter 8,
“Profile and Web Page Parameters and Defaults.”
For a list of the configurable parameters on the phone menus and their equivalent
profile and web page parameters, see Appendix B, “Phone Menu, Web Page, and
Profile Equivalent Parameters.”
For an overview of configuring the Cisco IP Phone, see Chapter 3, “Configuring
the Cisco IP Phone Models 7905G and 7912G for SIP.”
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Using Profiles with TFTP Configuration
Configuring the Cisco IP Phone Models 7905G and 7912G Using a TFTP Server
If you are not using a Dynamic Host Configuration Protocol (DHCP) server, you
must manually define IP addresses, subnet masks, and gateways for the phones
before you can access profiles. For more information, see the “Minimum
Configuration Settings for the Cisco IP Phone Models 7905G and 7912G” section
on page 3-9.
Note
Parameter values that you set in a profile will overwrite values that you set from
a phone’s Network Configuration menu, its SIP Configuration menu, or its web
page.
This chapter includes the following topics:
•
Using Profiles with TFTP Configuration, page 6-2
•
Setting Up the TFTP Server with Cisco IP Phone Software, page 6-4
•
Configuring the Cisco IP Phone to Obtain its Configuration File from the
TFTP Server, page 6-5
•
Creating or Updating a Profile, page 6-9
•
Using Encryption, page 6-15
•
Refreshing or Resetting the Cisco IP Phone, page 6-17
Using Profiles with TFTP Configuration
After the Cisco IP Phone has been configured with the minimum settings as
explained in the “Minimum Configuration Settings for the
Cisco IP Phone Models 7905G and 7912G” section on page 3-9, you can
configure many additional parameters using a file called a profile. A profile is
stored on the TFTP server and contains a list of tag, length, and value triplets
(TLVs) that represent some of the configuration parameters for the Cisco IP
Phone. You can edit a text version of a profile file and then convert it to a binary
file for the phone to download.
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Using Profiles with TFTP Configuration
TFTP configuration works as follows:
1.
The Cisco IP Phone model 7905G will try to download the profile
ldxxxxxxxxxxxx from the TFTP server. The Cisco IP Phone model 7912G will
try to download the profile gkxxxxxxxxxxxx from the TFTP server. In these
file names, each xx is the two-digit lowercase hexadecimal representation of
each integer in the phone’s MAC address.
2.
If the ldxxxxxxxxxxxx or the gkxxxxxxxxxxxx profile does not exist, the
Cisco IP Phone 7905G will try to download profile lddefault.cfg, and the
Cisco IP Phone 7912G will try to download profile gkdefault.cfg
Use the ldxxxxxxxxxxxx or the gkxxxxxxxxxxxx profile to configure a particular
Cisco IP Phone. Use the lddefault.cfg or the gkdefault.cfg profile to
simultaneously configure multiple phones of the same model with the same
information.
Parameter values in a profile will overwrite any values configured through the
web or through the phone configuration menus when the phone powers up or
refreshes.
Note
If you name a profile something other than ldxxxxxxxxxxxx or gkxxxxxxxxxxxx,
you must configure DHCP with the name of the profile or else the phone will not
recognize it.
Related Topics
•
Setting Up the TFTP Server with Cisco IP Phone Software, page 6-4
•
Configuring the Cisco IP Phone to Obtain its Configuration File from the
TFTP Server, page 6-5
•
Creating or Updating a Profile, page 6-9
•
Using Encryption, page 6-15
•
Refreshing or Resetting the Cisco IP Phone, page 6-17
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Setting Up the TFTP Server with Cisco IP Phone Software
Setting Up the TFTP Server with Cisco IP Phone
Software
Before you can use a TFTP server to configure a Cisco IP Phone, the firmware
image that you are using for the phone must exist on the TFTP server. To obtain
the correct firmware image and install it on the TFTP server, perform the
following steps.
Note
You must be a registered Cisco.com user to download the firmware image.
Procedure
Step 1
Go to the following URL:
http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml
Step 2
Download the zip file that contains the software for the applicable release and
firmware image you are using. The contents of each file are described next to the
file name.
Note
Step 3
A firmware image can contain a signature that allows a phone to verify
the authenticity of the image. A firmware image with such a signature has
the extension .sbin. A firmware image without such a signature has the
extension .zup. For the Cisco IP Phone 7905G, you should obtain both the
.sbin and the .zup versions of a firmware image. After a phone upgrades
to a .zup firmware image, it will request .sbin firmware images for future
firmware upgrades. If a phone upgrades to a .zup firmware image released
after March 26, 2003, the phone cannot be upgraded or downgraded to any
other .zup firmware image. In addition, a phone cannot be downgraded
from a .sbin firmware image to a .zup firmware image. (The
Cisco IP Phone 7912G is distributed with a .sbin firmware image.)
Extract the firmware files onto the TFTP server.
This server should be the same one that will contain the binary profile file that you
create as described in this chapter.
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Configuring the Cisco IP Phone to Obtain its Configuration File from the TFTP Server
Related Topics
•
Using Profiles with TFTP Configuration, page 6-2
•
Configuring the Cisco IP Phone to Obtain its Configuration File from the
TFTP Server, page 6-5
•
Creating or Updating a Profile, page 6-9
•
Using Encryption, page 6-15
•
Refreshing or Resetting the Cisco IP Phone, page 6-17
Configuring the Cisco IP Phone to Obtain its
Configuration File from the TFTP Server
This section describes the following ways in which the Cisco IP Phone can
contact the TFTP server to obtain its profile:
•
Using a DHCP Server, page 6-5
– The phone contacts the DHCP server, which provides the IP address of
the TFTP server
– The phone uses the DHCP server but the DHCP server does not know
about the TFTP server
•
Without Using a DHCP Server, page 6-8
Using a DHCP Server
Configuration settings for a DHCP server vary depending on whether the DHCP
server is under the control of the Cisco IP Phone system administrator or the
service provider. The simplest configuration is when the DHCP server is under the
control of the Cisco IP Phone administrator, in which case the DHCP server
provides the IP address of the TFTP server. Depending on who controls the DHCP
server, follow the applicable configuration procedure:
•
If the DHCP Server is Under Control of the Cisco IP Phone Administrator,
page 6-6
•
If the DHCP Server is not Under the Control of the Cisco IP Phone
Administrator, page 6-7
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This section also includes this topic:
•
Other DHCP Options You Can Set, page 6-7
The procedures in the following sections instruct you to set various parameter
values using a phone’s configuration menus or web page. For information about
phone menu configuration and parameters, see Chapter 5, “Configuring the Cisco
IP Phone Models 7905G and 7912G Using the Phone Menus.” For information
about web page configuration, see Chapter 7, “Configuring the
Cisco IP Phone Models 7905G and 7912G Using Web Pages.” For information
about web page parameters, see the “Network Parameters” section on page 8-2.
You can configure parameters through configuration menus on a phone without
first obtaining network connectivity for the phone. If you want to configure a
parameter using the phone’s web page, you first must obtain network connectivity
by statically configuring IP with the settings as explained in the “Minimum
Configuration Settings for the Cisco IP Phone Models 7905G and 7912G” section
on page 3-9.
If the DHCP Server is Under Control of the Cisco IP Phone Administrator
If you as the Cisco IP Phone administrator control the DHCP server, follow these
steps to make appropriate DHCP server configuration settings:
Procedure
Step 1
On the DHCP server, set one of the following options:
•
DHCP option 150 (TFTP server IP address)
•
Standard DHCP option 66 (TFTP server name)
If you use DHCP option 150, the Cisco IP Phone will ignore DHCP option 66.
However, if you use DHCP option 66, you must turn off DHCP option 150 or set
its value to 0.
Note
You can turn off the DHCP option 150 request by using OpFlags
parameter. See the “Network Parameters” section on page 8-2.
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Step 2
Make sure to use these default values for the following configuration parameters:
•
TftpURL=0
•
UseTftp=1
•
DHCP=1
After you complete these steps, the Cisco IP Phone will contact the DHCP server
for the IP address of the TFTP server that contains the Cisco IP Phone profile.
If the DHCP Server is not Under the Control of the Cisco IP Phone Administrator
If you as the Cisco IP Phone adminstrator do not control the DHCP server, you
must manually configure the URL of the TFTP server.
Procedure
Step 1
Set the TFTP Server parameter (on the phone’s Network Configuration menu) or
the TftpURL parameter (on the phone’s web page) to the IP address or the URL
of the TFTP server.
Step 2
Make sure that the DHCP Enabled parameter on the phone's Network
Configuration menu is set to Yes (the default value) or that the Dhcp parameter on
the phone's web page is set to 1 (the default value).
Step 3
Make sure that the TFTP Enabled parameter on the phone's Network
Configuration menu is set to Yes (the default value) or that the UseTftp parameter
on the phone's web page is set to 1 (the default value).
After you complete these steps, the Cisco IP Phone will contact the
manually-configured TFTP server that contains the phone’s profile.
Other DHCP Options You Can Set
The DHCP server can use option 60 (DHCP_VENDOR_CLASS_ID) to identify
Cisco IP Phone models 7905G and 7912G devices on the network. If specific
phones with specific MAC addresses need to download a TFTP file, the file name
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Configuring the Cisco IP Phone to Obtain its Configuration File from the TFTP Server
can be set in the bootfile name field of the DHCP header. This field is limited to
31 characters, even though the DHCP specification allows for a longer bootfile
name.
You can configure the following phone configuration parameters using DHCP:
•
Client IP address
•
DHCP option 1: Client subnet mask
•
DHCP option 3: Routers on the client’s subnet
•
DHCP option 6: Domain Name System (DNS) server (up to two)
•
DHCP option 42: Network Time Protocol (NTP) server (up to two)
•
DHCP option 66: TFTP server
The values for DNS server, NTP server, and TFTP server can be overwritten by
values in the corresponding parameters in the phone profile (DNS1IP, DNS2IP,
NTPIP, AltNTPIP, and TftpURL). If these parameters are not set to 0, their values
are used in place of those corresponding values that are supplied by the DHCP
options.
Without Using a DHCP Server
If you are not using a DHCP server in your environment but are still using a TFTP
server to obtain the Cisco IP Phone profile, follow these steps so that the
Cisco IP Phone can contact the TFTP server without using DHCP:
Procedure
Step 1
Set the DHCP Enabled parameter on the phone’s Network Configuration menu to
No or set the Dhcp parameter on the phone’s web page to 0.
Step 2
Set the TFTP Enabled parameter on the phone’s Network Configuration menu to
Yes or set the UseTftp parameter on the phone’s web page to 1.
Step 3
Set the TFTP Server parameter on the phone’s Network Configuration menu to the
IP address or the URL of the TFTP server or set the TftpURL parameter on the
phone’s web page to the IP address or the URL of the TFTP server.
Step 4
If you have done already done so, statically configure the following parameters
using the phone’s Network Configuration menu. These parameters are required
for the phone to obtain network connectivity:
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•
IP Address
•
Network gateway (default router)
•
Subnet Mask
You may also provide other parameters that are normally supplied by DHCP by
configuring their values. These parameters are:
•
DNS Server 1
•
DNS Server 2
•
NTP Server 1
•
NTP Server 2
•
Domain Name
After you complete these steps, the Cisco IP Phone will be able to contact the
TFTP server (without using DHCP) that will contain the configuration file for the
phone.
Related Topics
•
Using Profiles with TFTP Configuration, page 6-2
•
Setting Up the TFTP Server with Cisco IP Phone Software, page 6-4
•
Creating or Updating a Profile, page 6-9
•
Using Encryption, page 6-15
•
Refreshing or Resetting the Cisco IP Phone, page 6-17
Creating or Updating a Profile
The profile from which a Cisco IP Phone obtains information is a binary file.
However, when you create or edit a profile, you work with a text version of the
file, then create a binary version from the text file. Before you can create a binary
profile, you will need the cfgfmt.exe tool, which converts a text profile file to
binary format.
The cfgfmt.exe tool requires a parameter tag file, which specifies the type and size
of each parameter. By default, the cfgfmt.exe tool looks in the folder from which
you run the tool for a parameter tag file named ptag.dat. Cisco recommends that
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you use the parameter tag file called sip_ptag.dat, which Cisco provides. You can
rename sip_ptag.dat to ptag.dat before you run the cfgfmt.exe tool, or you can use
the cfgfmt.exe tool -t option to specify the file name sip_ptag.dat. (See the
“Creating a Phone-Specific Profile” section on page 6-11 or the “Creating a
Default Profile” section on page 6-13 for more information.)
The cfgfmt.exe tool and the sip_ptag.dat file are bundled with the Cisco IP Phone
software.
This section includes these topics:
•
Profile Conventions, page 6-10
•
Creating a Phone-Specific Profile, page 6-11
•
Creating a Default Profile, page 6-13
Related Topics
•
Using Profiles with TFTP Configuration, page 6-2
•
Setting Up the TFTP Server with Cisco IP Phone Software, page 6-4
•
Configuring the Cisco IP Phone to Obtain its Configuration File from the
TFTP Server, page 6-5
•
Using Encryption, page 6-15
•
Refreshing or Resetting the Cisco IP Phone, page 6-17
Profile Conventions
A text profile file must conform to the following conventions:
•
The characters #txt must appear at the beginning of the first line of the file so
that the formatting tool, cfgfmt.exe, will treat the file as a text file.
•
A pound sign (#) must appear at the beginning of a line that is a comment.
•
Parameter/value pairs must be in the form parameter:value, where parameter
is the exact name of the parameter and value is a valid value.
•
Each parameter/value pair must appear on its own line.
•
All parameter/value pairs are optional, but the Cisco IP Phone might not
function properly if some values are missing.
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•
A parameter value must be one of the following types:
– Alphanumeric string
– Numeric digit string
– Array of short integers
– IP address (for example, 192.168.2.170)
– Extended IP address (IP address with port; for example,
192.168.2.170.9001)
– Boolean (1 or 0)
– Bitmap (unsigned hexadecimal integer, for specifying bits in 32-bit
integer)
– Integer (32-bit integer)
Related Topics
•
Creating a Phone-Specific Profile, page 6-11
•
Creating a Default Profile, page 6-13
•
Using Encryption, page 6-15
•
Refreshing or Resetting the Cisco IP Phone, page 6-17
Creating a Phone-Specific Profile
The following steps explain how to create a phone-specific profile and how to
convert it to binary format.
When you create a phone-specific profile, you can use the sipexample.txt file as
a template. The sipexample.txt file is bundled with the Cisco IP Phone software
contains default values for parameters.
Tip
If you will create several unique profiles for various Cisco IP Phones, copy the
sipexample.txt file (or another existing profile) to a file with a meaningful name,
such as common.txt. Edit the new file, configure parameters and values that are
common to all of the phones, and save this file. Then, when you create a unique
profile for a phone, you can use the Include parameter to include common
parameters from the common file you just created. This way, you just need to
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configure parameters in the unique profile that are specific to a phone. For more
information about the Include parameter, see the “Profile Parameters” section on
page 8-26.
Procedure
Step 1
Use a text editor to open the sipexample.txt file or to open another existing text
profile.
Step 2
Edit or enter parameter:value pairs in the profile as needed.
For an explanation of parameters that you can include in a profile, see Chapter 8,
“Profile and Web Page Parameters and Defaults.”
If you use the Include parameter, Cisco recommends that it be the first
parameter in a text profile file. Any parameters in a text profile that come
before an include statement will be overwritten with values from the same
parameters, if they exist, in a common profile.
Note
Step 3
Save the file and name it as follows, where each xx is the two-digit lowercase
hexadecimal representation of each integer in the phone’s MAC address. A file
name must contain exactly 14 characters.
•
File name for the Cisco IP Phone 7905G: ldxxxxxxxxxxxx.txt
•
File name for the Cisco IP Phone 7912G: gkxxxxxxxxxxxx.txt
You can determine the MAC address of a phone from the MAC Address parameter
in the Network Configuration menu or the Device Information area in the phone’s
web page.
Step 4
Open a command window and type the following command to generate the binary
profile file from the text file:
cfgfmt [-eRC4Password] [-tpTagFile] input-text-file output-binary-file
where:
•
RC4Password is the optional RC4 key, up to eight alphanumeric characters,
to encrypt the binary profile that is generated by the cfgfmt tool. For more
information, see the “Using Encryption” section on page 6-15.
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•
pTagFile is an optional path and file name (or file name only) of the parameter
tag file. Use this option if the parameter tag file is not in the folder from which
you are running the cfgfmt.exe tool, and if the parameter tag file is not named
ptag.dat.
•
input-text-file is the name of the text version of the profile file. Make sure to
include any extension, such as .txt.
•
output-binary-file is the name of the binary file that the Cisco IP Phone uses
as the TFTP configuration profile file. Specify the file name as follows, as
follows, where each xx is the two-digit lowercase hexadecimal representation
of each integer in the phone’s MAC address. The file name must contain
exactly 14 characters.
– File name for the Cisco IP Phone 7905G: ldxxxxxxxxxxxx
– File name for the Cisco IP Phone 7912G: gkxxxxxxxxxxxx
Step 5
Put the binary file in the root directory on the TFTP server
When the Cisco IP Phone powers up or refreshes, it will retrieve its unique
configuration file from the TFTP server.
If you want to refresh the phone immediately, see the “Refreshing or Resetting the
Cisco IP Phone” section on page 6-17.) If you do not perform a refresh procedure,
the phone will update its configuration the next time it contacts the TFTP server.
Related Topics
•
Creating a Default Profile, page 6-13
•
Using Encryption, page 6-15
•
Refreshing or Resetting the Cisco IP Phone, page 6-17
Creating a Default Profile
You can create a default profile, called lddefault.cfg for the Cisco IP Phone 7905G, or
gkdefault.cfg for the Cisco IP Phone 7912G, that is common to all Cisco IP Phones
of the same model. This profile is applied to a phone only if a phone-specific profile
(ldxxxxxxxxxxxx or gkxxxxxxxxxxxx) does not exist for the phone on the TFTP server
when the phone powers up.
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The following steps explain how to create a default profile, how to convert it to
binary format, and how to store it on the TFTP server so that the Cisco IP Phone
will download it during the boot-up process:
Procedure
Step 1
Use a text editor to open the sipexample.txt file or to open another existing text
profile.
Step 2
Edit or enter parameter:value pairs in the profile as needed.
For an explanation of parameters that you can include in a profile, see Chapter 8,
“Profile and Web Page Parameters and Defaults.”
Step 3
Step 4
Save the file and name it as follows:
•
File name for the Cisco IP Phone 7905G: lddefault.txt
•
File name for the Cisco IP Phone 7912G: gkdefault.txt
Open a command window and type the following command to generate the binary
profile file from the text file:
cfgfmt [-eRC4Password] [-tpTagFile] input-text-file {lddefault.cfg |
gkdefault.cfg}
where:
•
RC4Password is the optional RC4 key, up to eight alphanumeric characters,
to encrypt the binary profile that is generated by the cfgfmt tool. For more
information, see the “Using Encryption” section on page 6-15.
•
pTagFile is an optional path and file name (or file name only) of the parameter
tag file. Use this option if the parameter tag file is not in the folder from which
you are running the cfgfmt.exe tool, and if the parameter tag file is not named
ptag.dat.
•
input-text-file is the name of the text version of the profile file. Make sure to
include any extension, such as .txt.
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Step 5
Put the binary file in the root directory on the TFTP server
When the Cisco IP Phone powers up or refreshes, the phone will download this
file as its profile file unless it first finds a phone-specific profile.
If you want to refresh the phone immediately, see the “Refreshing or Resetting the
Cisco IP Phone” section on page 6-17.) If you do not perform a refresh procedure,
the phone will update its configuration the next time it contacts the TFTP server.
Related Topics
•
Creating a Phone-Specific Profile, page 6-11
•
Using Encryption, page 6-15
•
Refreshing or Resetting the Cisco IP Phone, page 6-17
Using Encryption
The EncryptKey parameter encrypts binary files that are transferred over TFTP.
You can change this key for each Cisco IP Phone so that only one specific phone
can decode the information.
By default, the phone-specific profile is not encrypted. If encryption is required,
you must manually configure the phone with the encryption key. Use either the
Profile Encrypt Key parameter on the phone’s Network Configuration menu or the
EncryptKey key parameter in the Network Parameters area on a phone’s web
page.
Note
Because the factory-fresh phone cannot accept encrypted configuration files, the
first unencrypted file, if intercepted, can easily be read. (You would still have to
know the data structure format in order to decode the binary information from the
unencrypted file.) Therefore, the new encryption key in the unencrypted file can
be compromised.
When the EncryptKey parameter is set to a nonzero value, the Cisco IP Phone
assumes that the binary configuration file on the TFTP server is to be encrypted
with this key by means of the RC4 cipher algorithm. The phone will use this key
to decrypt the configuration file.
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The EncryptKey parameter or the profile Encrypt Key parameter and the
encryption key used in the cfgfmt tool command syntax must match.
The cfgfmt.exe syntax affects how the EncryptKey parameter is used, as shown
in the following examples. In these examples, input_text is the ldxxxxxxxxxxxx.txt
or the gkxxxxxxxxxxxx.txt file that you will convert to a binary profile,
output_binary is the binary ldxxxxxxxxxxxx or gkxxxxxxxxxxxx file, and Secret is
the encryption key.
Syntax examples
•
cfgfmt input-text-file output-binary-file
If input-text-file sets the EncryptKey parameter to 0, then output-binary-file
is not encrypted. If input-text-file sets EncryptKey to a non-zero value, then
output-binary-file is encrypted with that value.
•
cfgfmt -eSecret input-text-file output-binary-file
If the EncryptKey parameter has the value of 0 or is not included in
input-text-file, the Secret is used to encrypt the output-binary-file. If
input-text-file sets the EncryptKey parameter to a nonzero value and the -e
option is used, then output-binary-file is encrypted with the EncryptKey
parameter set in input-text-file and Secret is ignored.
•
cfgfmt -E input-text-file output-binary-file
The -E (uppercase) option means that any value specified for the EncryptKey
parameter in input-text-file is ignored. However, because Secret is not
specified in this example, output-binary-file is not encrypted. Nevertheless,
the EncryptKey parameter and its value, if specified in input-file-text, will be
included in output-binary-file for possible encryption at a later time.
•
cfgfmt -E -eSecret input-text-file
output-binary-file
The -E (uppercase) option means that any value specified for the EncryptKey
parameter in input-text-file is ignored and the output-binary-file is encrypted
with the Secret key. However, the EncryptKey parameter and its value, if
specified in input-text-file, will be included in output-binary-file.
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Refreshing or Resetting the Cisco IP Phone
Refreshing or Resetting the Cisco IP Phone
When you make configuration changes to a Cisco IP Phone profile, you can
refresh or reset the Cisco IP Phone for these configuration changes to
immediately take effect. If you do not refresh or reset the phone, the configuration
changes will take effect the next time the phone contacts the TFTP server, which
occurs based on the configured value of the CfgInterval parameter.
A refresh procedure will update a phone with the current profile. A reset
procedure will update a phone with the current profile and will power-down and
power-up the phone. A reset is not necessary if your only goal is to update a
phone’s profile.
Note
You use bits 8 and 0 of the OpFlags parameter prevent end users from refreshing
or resetting a phone. See the “Network Parameters” section on page 8-2 for more
information.
This section includes the following topics:
•
Refreshing the Cisco IP Phone, page 6-17
•
Resetting the Cisco IP Phone, page 6-17
Refreshing the Cisco IP Phone
To refresh the Cisco IP Phone, enter the following command from your web
browser:
http://<ipaddress>/refresh
where ipaddress is the IP address of the Cisco IP Phone that you are refreshing.
Resetting the Cisco IP Phone
To reset the Cisco IP Phone, enter the following command from your web
browser:
http://<ipaddress>/reset
where ipaddress is the IP address of the Cisco IP Phone that you are resetting.
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7
Configuring the
Cisco IP Phone Models 7905G and
7912G Using Web Pages
Each phone has a web page from which you can view and update many device and
network parameter values. You can also use the web page to obtain a variety of
statistical and operational information about the phone. This chapter describes
how use a phone’s web page to configure parameters and to obtain information.
You can also configure many parameters for a Cisco IP Phone using the
configuration menus on the phone and using profiles. For more information, see
Chapter 5, “Configuring the Cisco IP Phone Models 7905G and 7912G Using the
Phone Menus,” and Chapter 6, “Configuring the Cisco IP Phone Models 7905G
and 7912G Using a TFTP Server.”
For an explanation of the parameters that you can configure from a phone’s web
page, see Chapter 8, “Profile and Web Page Parameters and Defaults.”
For a list of the configurable parameters on the phone menus and their equivalent
web page and profile parameters, see Appendix B, “Phone Menu, Web Page, and
Profile Equivalent Parameters.”
If you are not using a Dynamic Host Configuration Protocol (DHCP) server, you
must manually configure the IP address of the phone, the subnet mask used by the
phone, and the network gateway (default router) IP address for the phone before
you can access the web page for the phone. For more information, see the
“Minimum Configuration Settings for the Cisco IP Phone Models 7905G and
7912G” section on page 3-9.
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Accessing a Phone’s Web Page
Note
If you configure a phone using a TFTP server, parameter values in a profile will
overwrite values that you set from the phone’s web page for the same parameters.
This chapter includes the following topics:
•
Accessing a Phone’s Web Page, page 7-2
•
Viewing Statistical and Operational Information, page 7-4
Accessing a Phone’s Web Page
You can display and configure device and network information for a
Cisco IP Phone through the phone’s web page. You can access the web page using
any graphically capable web browser.
Note
Cisco recommends that you take care to prevent unauthorized access to a phone’s
web page and that you increase security by setting a password using the
UIPassword parameter in the phone’s profile. For more information about this
parameter, see the “Profile Parameters” section on page 8-26. For security
reasons, Cisco also recommends that you do not use the web interface over the
public network. Disable the web interface by setting the OpFlags parameter bit 7
(mask 0x80) to 1 before the phone leaves the service-provider site. After the
parameter bit is set, limited configuration can be done through the configuration
menus on a phone and full configuration can be done through the TFTP server.
To access the web page for a Cisco IP Phone, perform the following steps.
If you want to edit parameters in the Network Parameters, SIP Parameters, Call
Preferences, Tone Parameters, or Audio Parameters areas, you must first set up a
password with the UIPassword parameter in the profile. If you do not set up a
password, you will be able to view information in these areas, but you will not be
able to change parameters, and Apply will not be available.
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Accessing a Phone’s Web Page
Procedure
Step 1
Open a web browser and enter the URL of the web page for the phone as follows:
http://IP_address/dev
where IP_address is the IP address of the Cisco IP Phone.
You can determine the IP address on a phone by pressing the Menu button,
choosing Settings > Network Configuration, and then scrolling to the IP
Address option.
For example, the configuration page for a Cisco IP Phone with the IP address
192.168.3.225 is:
http://192.168.3.225/dev
Step 2
From the phone’s web page, choose the hyperlink for the information that you
want to display or configure:
•
Device Information—Display only. For more information, see the “Device
Information” section on page 7-4.
•
Network Configuration—Display only. For more information, see the
“Network Configuration” section on page 7-5.
•
Network Statistics—Display only. For more information, see the “Network
Statistics” section on page 7-6.
•
Device Logs—Reserved for future use.
•
Network Parameters—Configurable parameters. For more information, see
the “Network Parameters” section on page 8-2.
•
SIP Parameters—Configurable parameters. For more information, see the
“SIP Parameters” section on page 8-9.
•
Call Preferences—Configurable parameters. For more information, see the
“Call Preference Parameters” section on page 8-13.
•
Tone Parameters—Configurable parameters. For more information, see the
“Tone Parameters” section on page 8-21.
•
Audio Parameters—Configurable parameters. For more information, see
the “Audio Parameters” section on page 8-24.
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Step 3
Step 4
Configuring the Cisco IP Phone Models 7905G and 7912G Using Web Pages
If you selected Network Parameters, SIP Parameters, Call Preferences, Tone
Parameters, or Audio Parameters, follow these steps:
a.
If you have set up a password with the UIPassword parameter in the profile,
enter the password in the UIPassword field, and then click Apply. (If you
want to change the password, enter the new password in the two
ChangeUIPassword fields before clicking Apply.)
b.
Make changes to parameters as needed and click Apply to save your changes.
Close your web browser.
Viewing Statistical and Operational Information
The following sections describe the information that you can obtain from a
phone’s web page. You can use this information to remotely monitor the
operation of a phone and to assist with troubleshooting.
•
Device Information, page 7-4
•
Network Configuration, page 7-5
•
Network Statistics, page 7-6
Device Information
Table 7-1 explains the fields displayed in the Device Information area of the
Cisco IP Phone web page.
Note
To access the Device Information area directly from a web browser, enter
ip_address/DeviceInformation, where ip_address is the IP address of your
Cisco IP Phone.
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Table 7-1
Field
Explanation
MAC Address
Unique Media Access Control (MAC) address of
the phone
Software Version
Version of the software running on the phone
Hardware Revision
Version of the phone hardware
Serial Number
Serial number of the phone
Product ID
Product identifier of the phone
H/W features
Reserved for future use
BTXML cards version
Version of the BTXML cards in the phone
Network Configuration
Table 7-2 explains the fields displayed in the Network Configuration area of the
Cisco IP Phone web page.
Note
To access the Network Configuration area from a web browser, enter
ip_address/NetworkConfiguration, where ip_address is the IP address of your
Cisco IP Phone.
Table 7-2
Field
Explanation
DHCP Server
IP address of the Dynamic Host Configuration
Protocol (DHCP) server from which the phone
obtains its IP address
BOOTP Server
Indicates whether the phone obtains its
configuration from a Bootstrap Protocol (BootP)
server rather than a DHCP server
MAC Address
Unique Media Access Control (MAC) address of
the phone
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Table 7-2
Field
Explanation
Host Name
Unique host name of the phone
Domain Name
Name of the Domain Name System (DNS) domain
in which the phone resides
IP Address
Internet Protocol (IP) address of the phone
Default Router
Network gateway used by the phone
Subnet Mask
Subnet mask used by the phone
TFTP Server 1
Trivial File Transfer Protocol (TFTP) server used
by the phone
NTP Server 1
Primary Network Time Protocol (NTP) server
used by the phone
NTP Server 2
Secondary NTP server used by the phone
DNS Server 1
Primary Domain Name System (DNS) server used
by the phone
DNS Server 2
Secondary DNS server (DNS Server 2) used by the
phone
Alt NTP Server 1
Primary alternate NTP server used by the phone
Alt NTP Server 2
Secondary alternate NTP server used by the phone
Network Statistics
Table 7-3 describes the fields displayed on the Network Statistics area of the
Cisco IP Phone web page.
Note
To access the Network Statistics area from a web browser, enter
ip_address/EthernetInformation, where ip_address is the IP address of your
Cisco IP Phone.
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Table 7-3
Field
Explanation
Elapsed time
Time that has elapsed since the phone was last
reset
Receive Packets
Number of packets that the phone has received
during the elapsed time
Transmit Packets
Number of packets that the phone has transmitted
during the elapsed time
Broadcast
Number of packets that the network has broadcast
during the elapsed time
Multicast
Number of packets that the network has multicast
during the elapsed time
Receive Errors
Number of receive errors that the phone has
experienced during the elapsed time
Transmit Errors
Number of transmit errors that the phone has
experienced during the elapsed time
Receive Overflow
Number of packet overflows that the phone has
experienced during the elapsed time
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8
Profile and Web Page Parameters and
Defaults
This chapter describes the parameters that you can use to configure the
Cisco IP Phone models 7905G and 7912G through a profile or through a phone’s
web page. These parameters are organized as follows:
•
Network parameters—Controls various network-related activities of the
phone.
•
SIP parameters—Specifies SIP parameters of the phone.
•
Call preferences—Controls various user features on the phone, and specifies
which parameters appear and which options can be configured on a phone’s
Call Preference menu.
•
Tone Parameters—Controls how the phone handles the various tones that it
plays, and related options.
•
Audio parameters—Controls how the phone handles various audio-related
activities.
•
Profile parameters—Parameters that are available only from the phone’s
profile.
You can also configure many parameters for a Cisco IP Phone using the
configuration menus on the phone or using the phone’s web page. For more
information, see Chapter 5, “Configuring the Cisco IP Phone Models 7905G and
7912G Using the Phone Menus,” and Chapter 7, “Configuring the
Cisco IP Phone Models 7905G and 7912G Using Web Pages.”
For an explanation of profiles and how to use them, see Chapter 6, “Configuring
the Cisco IP Phone Models 7905G and 7912G Using a TFTP Server.”
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Network Parameters
For instructions about accessing a phone’s web page, see Chapter 7, “Configuring
the Cisco IP Phone Models 7905G and 7912G Using Web Pages.”
For a list of the configurable parameters on the phone menus and their equivalent
profile and web page parameters, see Appendix B, “Phone Menu, Web Page, and
Profile Equivalent Parameters.”
This chapter includes the following sections:
•
Network Parameters, page 8-2
•
SIP Parameters, page 8-9
•
Call Preference Parameters, page 8-13
•
Tone Parameters, page 8-21
•
Audio Parameters, page 8-24
•
Profile Parameters, page 8-26
•
DialPlan Parameter Detail, page 8-38
•
Parameter Details, page 8-27
Network Parameters
Table 8-1describes the network parameters that you can configure through a
phone’s profile or through the Network Parameters area on the phone’s web page.
Tip
To access the Network Parameters area directly from a web browser, enter
ip_address/NetCfg, where ip_address is the IP address of your Cisco IP Phone.
This area appears by default when you enter ip_address/dev.
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Network Parameters
Table 8-1
Network Parameters
Parameter
Description
UseTftp
Enables or disables downloading of a profile from a TFTP server. If you set this
parameter to 1 and subsequently make changes using the web interface, you
must set this parameter to 0 before saving those changes; otherwise, the
changes will be overwritten by the profile obtained from a TFTP server.
•
0=Do not use a TFTP server.
•
1=Use a TFTP server:
– If TftpURL is set to 0, use the TFTP IP address or URL obtained from
the DHCP server to contact the TFTP server.
– If TftpURL is set to a value other than 0, use the specified IP address
or URL to contact the TFTP server.
Note
Do not specify a port. The Cisco IP Phone always contacts the TFTP
server at port 69.
Value type: Boolean.
Default value: 1
TftpURL
IP address or URL of the TFTP server from which a phone obtains a profile.
Required if the DHCP server does not provide the TFTP server address.
Optionally, you can include the path prefix to the profile to download. For
example, if the TFTP server IP address is 192.168.2.170 or www.cisco.com,
and the path to the file is /ip7905, you can specify the URL as
192.168.2.170/ip7905 or www.cisco.com/ip7905.
Do not specify a port. The Cisco IP Phone always contacts the TFTP server at
port 69.
If DHCP is used, a nonzero values of TftpURL overwrites any DHCP-supplied
addresses and values of 0 tell the phone to use DHCP-supplied addresses.
This parameter is ignored if UseTftp is 0.
Value type: IP address, up to 31 characters.
Default value: 0.0.0.0
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Network Parameters
Table 8-1
Network Parameters (continued)
Parameter
Description
CfgInterval
Number of seconds between automatic profile refreshes from the TFTP server.
At the earliest idle time following each interval expiration, the phone retrieves
its profile using the tftp get command. Intervals should vary among phones on
the network to prevent simultaneous contact of the TFTP server by many
phones.This value can range from 60 to 4294967295.
This parameter is ignored if UseTftp is 0.
Value type: Integer.
Default value: 3600
EncryptKey
Key for decrypting the configuration profile that is downloaded from the TFTP
server.
•
0=The configuration profile is not encrypted.
•
string=The profile is encrypted with this key.
This parameter is ignored if UseTftp is 0.
Value type: Alphanumeric string, up to 8 characters.
Default value: 0
Dhcp
Specifies whether to use the DHCP server to obtain information such as IP
addresses for the phone, the network gateway, and the DNS, NTP, and TFTP
servers.
If this parameter is 0, you must manually specify the IP address of the phone,
and the IP addresses of the subnet mask and the default router used by the
phone. See the “Minimum Configuration Settings for the
Cisco IP Phone Models 7905G and 7912G” section on page 3-9 for more
information.
•
0=Do not use the DHCP server. Use hard-coded values.
•
1=Use the DHCP server.
Value type: Boolean.
Default value: 1
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Network Parameters
Table 8-1
Network Parameters (continued)
Parameter
Description
StaticIP
Internet Protocol (IP) address of the phone. You can use this parameter to
assign the IP address if DHCP does not provide it.
If you assign an IP address with this parameter, you must also assign a default
router and a subnet mask. See the StaticRoute and the StaticNetMask
parameters in this table.
This parameter is ignored if the Dchp parameter is set to 1.
Value type: IP address.
Default value: 0.0.0.0
StaticRoute
Static IP address of the network gateway. You can use this parameter to assign
the IP address if DHCP does not provide it.
This parameter is ignored if the Dchp parameter is set to 1.
Value type: IP address.
Default value: 0.0.0.0
StaticNetMask
Subnet mask used by the phone. You can use this parameter to assign the
subnet mask if DHCP does not provide it.
If you assign a Subnet Mask address with this parameter, you must also assign
an IP address and a default router for the phone. See the StaticIP and the
StaticRoute parameters in this table.
This parameter is ignored if the Dchp parameter is set to 1.
Value type: IP address.
Default value: 0.0.0.0
Domain
Domain name of the phone.
If set to 0, the phone will use the domain name supplied by the DHCP server.
Value type: Alphanumeric string, up to 31 characters.
Default value: 0
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Network Parameters
Table 8-1
Network Parameters (continued)
Parameter
Description
DNS1IP, DNS2IP
IP addresses of the primary and secondary domain name servers if DHCP is not
used to provide the DNS addresses. If DHCP is used, nonzero values of
DNS1IP and DNS2IP overwrite any DHCP-supplied addresses and values of 0
tell the phone to use DHCP-supplied addresses. Do not specify a port; the
phone uses the default DNS port only.
Value type: IP address.
Default value: 0.0.0.0
NTPIP
IP address of the primary NTP server. DHCP may also supply a NTP server but
NTPIP, if specified, overwrites that value. Do not specify a port; the phone uses
the default NTP port only.
If DHCP is used, a nonzero values of NTPIP overwrites any DHCP-supplied
addresses and values of 0 tell the phone to use DHCP-supplied addresses.
Value type: IP address.
Default value: 0 (use DHCP)
AltNTPIP
IP address of a secondary NTP server if redundancy is desired. If only one NTP
server exists, set this to 0 or to the address of the primary server. Do not specify
a port; the phone uses the default NTP port only.
Value type: IP address.
Default value: 0 (use DHCP)
TimeZone
Offset to apply to the Greenwich Mean Time (GMT) returned by an NTP server
to determine local time (to use for Call ID display, for example).
Local time is generated as follows:
•
Local time = GMT + TimeZone, if TimeZone <= 12.
•
Local time = GMT + TimeZone – 25, if TimeZone > 12.
For example, TimeZone = 17 for Pacific Standard Time.
Note
You must update this parameter manually when daylight savings time
goes into effect and ends.
Value type: Integer.
Default value: 17
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Network Parameters
Table 8-1
Network Parameters (continued)
Parameter
Description
OpFlags
Enables or disables various operational features.
•
Bit 0: If 0, the TFTP phone-specific profile file name supplied by the
DHCP server overwrites the default file name for each phone. If 1, the
phone uses the default profile file phone-specific name (recommended).
•
Bit 1: If 0, enables probing of the static network router when the phone
powers up. If 1, probing is disabled (recommended).
•
Bit 2: Reserved.
•
Bit 3: If 0, request DHCP option 150 in the DHCP discovery message. If
1, do not request this option. (Some DHCP servers do not respond if option
150 is requested.)
•
Bit 4: If 0, do not use VLAN ID set with the VLANSetting parameter. If
1, use VLAN ID (recommended).
•
Bit 5: If 0, use VLAN IP encapsulation. If 1, do not use VLAN IP
encapsulation.
•
Bit 6: If 0, perform CDP discovery. If 1, do not perform CDP discovery.
•
Bit 7: If 0, allow configuration of the phone through the web. If 1, do not
allow configuration through the web.
•
Bit 8: If 0, allow refreshing the phone through the web. If 1, do not allow
refreshing the phone through the web.
•
Bit 9: If 0, allow resetting the phone through the web. If 1, do not allow
resetting the phone through the web.
•
Bits 10-31: Reserved.
Value type: Bitmap.
Default value: 0x00000002
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Network Parameters
Table 8-1
Network Parameters (continued)
Parameter
Description
VLANSetting
Controls the use of VLAN in the phone.
•
Bits 0-2: Specifies 802.1Q priority for Signaling IP packets.
•
Bits 3-5: Specifies 802.1Q priority for Audio Voice IP packets.
•
Bits 6-17: Reserved.
•
Bits 18-29: User-specified 802.1Q VLAND ID.
•
Bits 30-31: Reserved.
Value type: Bitmap.
Default value: 0x0000002B
TOS
ToS (Type of Service) bits. Specifies the precedence and delay of Audio and
Signaling IP packets. Higher values increase routing priority for data packets,
resulting in less latency.
•
Bits 0-7: ToS value for audio data packets. Range: 0-255; Default: 184
•
Bits 8-15: ToS value for signaling data packets. Range: 0-255; Default:
104
•
Bits 16-31: Reserved
Value type: Bitmap.
Default value:0x000068b8
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SIP Parameters
Table 8-1
Network Parameters (continued)
Parameter
Description
NPrintf
IP address and port of a host PC to which phone debug messages are sent, in
the format IP_address.port (for example: 192.168.2.159.9000).
The host PC must be running the prserv.exe application at the
corresponding port. This application is available at Cisco.com.
Note
To turn off debugging, set to 0 or to 0.0.0.0.0.
Value type: Extended IP address.
Default value: 0.0.0.0.0
For more information about using Nprintf and prserv.exe to collect information
for debugging, see the “Debugging” section on page 9-5.
TraceFlags
Enables specific trace features when Nprintf is set to a valid host address and
port.
•
0=Disable debug messages.
•
1=Enable debug messages.
Value type: Bitmap.
Default value: 0x00000000
SIP Parameters
Table 8-2 describes the SIP parameters that you can configure through a phone’s
profile or through the SIP Parameters area on the phone’s web page
Tip
To access the SIP Parameters area directly from a web browser, enter
IP_address/SIPConfiguration, where IP_address is the IP address of your
Cisco IP Phone.
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SIP Parameters
Table 8-2
SIP Parameters
Parameter
Description
UID, PWD
User ID and password for registration and authentication. Authentication can
be performed on the (UID, PWD) pair or on the (LoginID, PWD) pair
depending on the UseLoginID flag. However, the phone identifies itself to the
outside world with the UID only and can be reached only with the
corresponding UID but not the LoginID. If UID is set to “.” or to “0” (zero),
the phone is disabled (does not register with the proxy server).
Value type: Alphanumeric string.
Default value: UID=123, PWD=0
Proxy
SIP proxy server.
Value type: Alphanumeric string.
Default value: 0
AltProxyTimeOut
Timeout, in seconds, for switching from backup proxy server to primary proxy
server.
The value can range from 0 to 4294967295.
Value type: Integer.
Default value: 0
UseLoginID
Specifies which value to use for authentication:
•
0=Use UID for authentication.
•
1=Use LoginID for authentication.
Value type: Boolean.
Default value: 0
LoginID
Alternate user name used for authentication.
Value type: Alphanumeric string.
Default value: 0
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SIP Parameters
Table 8-2
SIP Parameters (continued)
Parameter
Description
SIPRegInterval
SIP registration interval, in seconds, between each registration renewal to the
SIP proxy server. If set to 0, the phone uses the default value.
The value can range from 0 to 86400.
Value type: Integer.
Default value: 3600
MAXRedirect
Maximum number of redirections the phone attempts to reach a callee.
The value can range from 0 to 10.
Value type: Integer.
Default value: 5
SIPRegOn
Whether to enable SIP registration. This parameter is ignored if a SIP proxy
server is not specified. If set to 1, the phone periodically registers to the SIP
proxy server at an interval given by the SIPRegInterval parameter.
Valid values are 0 or 1.
Value type: Boolean.
Default value: 0
NATIP
Network Address Translation (NAT) WAN IP address where other SIP user
agents can communicate with the phone.
Value type: IP address.
Default value: 0.0.0.0
SIPPort
Port where the phone listens for incoming requests and sends outgoing
requests.
Valid values are 5060 to 65535.
Value type: Integer.
Default value: 5060
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SIP Parameters
Table 8-2
SIP Parameters (continued)
Parameter
Description
MediaPort
Port from which the phone transmits and receives media streams. This value
must be an even number; each connection uses the next available
even-numbered port for RTP. 0=Use the default value.
This value can range from 0 to 65535.
Value type: Even integer.
Default value: 16384
OutBoundProxy
IP address or URL of the outbound proxy server, with or without a port
parameter; for example, 209.165.201.30, 209.165.201.30.5060,
209.165.201.30:5061, sip.cisco.com, sip.xyz.cisco.com:5061.
•
For IP address, the port, if included, can be preceded by a period (.) or a
colon (:).
•
For URL, the port must be preceded by a colon (:).
•
If no port is specified, the default port 5060 is used.
Value type: Alphanumeric string, up to 31 characters.
Default value: 0
NatServer
Specifies a server to which a dummy, single-byte UDP packed is sent to
maintain a Network Address Translation (NAT) during a session.
This parameter can contain up to 47 characters in fully qualified domain name
(FQDN) or IP format with an optional port parameter (separated from the
address by a colon); for example, xyz.cisco.com;1234. If no port is specified,
the phone uses port 5060.
Value type: IP address, up to 47 characters.
Default value: 0.0.0.0 (port 5060 will be used)
NatTimer
Retransmission interval (in seconds) for sending a dummy packet to the server
specified with the NatServer parameter, specified in bits 0-11 the parameter.
The upper 20 bits are reserved and should be set to 0.
Value type: Bitmap.
Default value: 0x00000000 (no dummy packets will be sent to NatServer)
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Table 8-2
SIP Parameters (continued)
Parameter
Description
DialPlan
Dial plan rules. No syntax check is performed by the implementation. The
administrator must ensure that the dial plan is syntactically valid.
See the “DialPlan Parameter Detail” section on page 8-38 for syntax and usage
details.
Value type: Alphanumeric string, up to 199 characters.
Default value: *St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.-
IPDialPlan
Allows for detection of IP-like destination addresses in the dial plan:
•
1=If two periods (.) are detected, assume that this is an IP address.
•
2=If three periods (.) are detected, assume that this is an IP address.
Value type: Integer.
Default value: 1
Call Preference Parameters
Table 8-4 describes the call preference parameters that you can configure through
a phone’s profile or through the Call Preferences area on the phone’s web page.
These parameters specify various user features on the phone, and specifies which
parameters appear and which options can be configured on a phone’s Call
Preference menu.
Tip
To access the Call Preferences area directly from a web browser, enter
IP_address/CallPrefConfiguration, where IP_address is the IP address of your
Cisco IP Phone.
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Call Preference Parameters
Table 8-3
Call Preference Parameters
Parameter
Description
Allow Call Waiting
(on web page)
Whether the call waiting feature is enabled on a phone.
CallWaiting
(in profile)
•
0=Disable the call waiting feature.
•
1=Enable the call waiting feature.
Value type: Boolean.
Default value: 1
Allow Call Transfer
(on web page)
AttendedTransfer
(in profile)
Whether the call transfer feature is enabled on a phone. If this feature is
enabled, the Trnsfer softkey will appear when appropriate, allowing a user to
transfer a call. If this feature is disabled, this softkey will never be available.
•
0=Disable the call transfer feature.
•
1=Enable the call transfer feature.
Value type: Boolean.
Default value: 1
Allow Blind Transfer Whether the blind call transfer feature is enabled on a phone. If this feature is
enabled, the BlndXfr softkey will appear when appropriate, allowing a blind
(on web page)
transfer of a call. If this feature is disabled, this softkey will never be available.
BlindTransfer
(in profile)
• 0=Disable the blind call transfer feature.
•
1=Enable the blind call transfer feature.
Value type: Boolean.
Default value: 1
Allow Conference
(on web page)
Conference
(in profile)
Whether the conference feature is enabled on a phone. If this feature is
enabled, the Confrn softkey will appear when appropriate, allowing a
three-way conference to be established. If this feature is disabled, this softkey
will never be available.
•
0=Disable the conference feature.
•
1=Enable the conference feature.
Value type: Boolean.
Default value: 1
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Call Preference Parameters
Table 8-3
Call Preference Parameters (continued)
Parameter
Description
Block Caller ID
(on web page)
Whether a phone’s caller ID information, which includes the value specified
by the Display Name parameter and the phone number, is sent with an outgoing
call. If this parameter is enabled, outgoing calls will use “Anonymous” as the
phone’s caller ID.
BlockCallerId
(in profile)
•
0=Do not block outgoing caller ID information.
•
1=Block outgoing caller ID information.
Value type: Boolean.
Default value: 0
Block Anonymous
Calls
(on web page)
BlockAnonymous
(in profile)
Whether a phone rejects (blocks) an incoming call with “Anonymous” caller
ID will be rejected.
•
0=Do not block anonymous incoming calls.
•
1=Block anonymous incoming calls.
Value type: Boolean.
Default value: 0
Do Not Disturb
(on web page)
DoNotDisturb
(in profile)
Whether the Do Not Disturb feature is enabled on a phone. If this feature is
enabled, calls to the phone will receive a busy signal.
•
0=Disable the do not disturb feature.
•
1=Enable the do not disturb feature.
Value type: Boolean.
Default value: 0
Voice Mail Number
(on web page)
VoiceMailNumber
(in profile)
Specifies a telephone number where voice messages are stored and retrieved.
If a number (other than 0) is specified:
•
The Message softkey will appear on the phone when appropriate.
•
Incoming calls forward to this number if the phone is busy or if the calls
are not answered within the number of seconds specified with the Forward
to VMail Delay(s) parameter.
Value type: Alphanumeric string, up to 31 characters.
Default value: 0 (disables this feature)
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Call Preference Parameters
Table 8-3
Call Preference Parameters (continued)
Parameter
Description
Call Forward Number Specifies a telephone number to which all calls to the phone are forwarded. If
(on web page)
a number (other than 0) is specified, the Call Forward All feature is activated.
CallForwardNumber
(in profile)
Value type: Alphanumeric string, up to 31 characters.
Display Name
(on web page)
Specifies a name to be used as part of the phone’s caller ID and that will be
displayed on the phone’s LCD screen. A 0 (zero) or a blank value disables this
feature.
DisplayName
(in profile)
Default value: 0 (disables call forwarding)
Note
If the Block Caller ID parameter is set to Yes, outgoing calls will use
“Anonymous” as the phone’s caller ID regardless of the Display Name
setting.
Value type: Alphanumeric string, up to 31 characters.
Default value: blank (no character string)
Short Name
(on web page)
ShortName
(in profile)
Specifies a name to be displayed on the phone’s LCD screen in place of the
Display Name value. Short Name will not affect the Display Name value that
is used as part of the phone’s caller ID.
Value type: Alphanumeric string, up to 31 characters.
Default value: 0 (causes the Display Name value to be displayed)
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Table 8-3
Call Preference Parameters (continued)
Parameter
Description
Time Format
(on web page)
Specifies the format for the time that appears on the phone’s LCD screen. This
format is specified by one or more of the following characters:
TimeFormat
(in profile)
h—designates the hour in 12-hour format.
H—designates the hour in 24-hour format.
i or I—designates minutes.
a or A—for 12-hour format, include “p” with times from noon until one minute
before midnight.
: (colon)—displays a colon that blinks every second.
For example, when the current time is 1:30 P.M.,
•
h:ia causes the time to appear as 1:30 p.
•
H:I causes the time to appear as 13:30.
Value type: Alphanumeric string, up to 15 characters.
Default value: h:ia
Date Format
(on web page)
Specifies the format for the date that appears on the phone’s LCD screen. This
format is specified by one or more of the following characters:
DateFormat
(in profile)
m—designates the month as a number 1 through 12.
M—designates the month as a three-letter abbreviation, Jan through Dec.
d or D—designates the day of the month.
y—designates a two-digit year (such as 03 for 2003).
Y—designates a four-digit year.
Other characters—appear as entered.
For example, when the current date is March 10, 2003:
•
m-d-y causes the date to appear as 3-10-03.
•
M d, Y causes the date to appear as Mar 20, 2003.
•
Y/m/d causes the date to appear as 2003/3/10.
Value type: Alphanumeric string, up to 15 characters.
Default value: m-d-y
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Call Preference Parameters
Table 8-3
Call Preference Parameters (continued)
Parameter
Description
Forward to VMail
Delay
(on web page)
Specifies number of seconds after which an incoming call ringing at the phone
will be forwarded to the telephone number specified by the Voice Mail Number
parameter.
ForwardToVMDelay
(in profile)
Valid values are from 0 to 2147483647.
Value type: Integer.
Default value: 20 (seconds)
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Call Preference Parameters
Table 8-3
Call Preference Parameters (continued)
Parameter
Description
GUI Show Mask
(on web page)
Specifies which parameters, if any, appear on the Call Preferences menu on the
phone.
CallPrefGuiShow
(in profile)
Set a bit to 0 if the corresponding parameter should not appear. Set the bit to 1
if the parameter should appear.
•
Bit 0: Do Not Disturb.
•
Bit 1: Allow Call Waiting.
•
Bit 2: Block Caller ID.
•
Bit 3: Call Forward Number.
•
Bits 4–5: Reserved.
•
Bit 6: Display Name.
•
Bit 7: Time Format.
•
Bit 8: Date Format.
•
Bit 9: Voice Mail Number.
•
Bit 10: Allow Call Transfer.
•
Bit 11: Allow Attended Transfer.
•
Bit 12: Allow Conference.
•
Bit 13: Short Name.
•
Bits 14–23: Reserved.
•
Bit 24: Block Anonymous Call.
•
Bit 25: Reserved.
•
Bit 26: Forward to Voice Mail Delay.
•
Bit 27 and bits 29–31: Reserved.
In addition, bit 28, if set to 1, causes a registration status icon to appear on the
phone’s LCD screen. This icon indicates whether the phone is registered to the
SIP proxy server.
Value type: Bitmap.
Default value: 0xFFFFFFFF
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Table 8-3
Call Preference Parameters (continued)
Parameter
Description
GUI Set Mask
(on web page)
Specifies which parameters on the Call Preferences menu on the phone can be
changed by an end user.
CallPrefGuiSet
(in profile)
Set a bit to 0 if an end user should not be able to change the corresponding
parameter. Set the bit to 1 if an end user should be able to change the parameter.
•
Bit 0: Do Not Disturb.
•
Bit 1: Allow Call Waiting.
•
Bit 2: Block Caller ID.
•
Bit 3: Call Forward Number.
•
Bits 4–5: Reserved.
•
Bit 6: Display Name.
•
Bit 7: Time Format.
•
Bit 8: Date Format.
•
Bit 9: Voice Mail Number.
•
Bit 10: Allow Call Transfer.
•
Bit 11: Allow Blind Transfer.
•
Bit 12: Allow Conference.
•
Bit 13: Short Name.
•
Bit 24: Block Anonymous Call.
•
Bit 25: Reserved.
•
Bit 26: Forward to Voice Mail Delay.
Value type: Bitmap.
Default value: 0xFFFFFFFFparameters:network<$startrange>
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Tone Parameters
Tone Parameters
Table 8-4 describes the tone parameters that you can configure through a phone’s
profile or through the Tone Parameters area on the phone’s web page. These
parameters specify various tone characteristics for the Cisco IP Phone.
Tip
To access the Tone Parameters area directly from a web browser, enter
IP_address/ToneConfiguration, where IP_address is the IP address of your
Cisco IP Phone.
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Tone Parameters
Table 8-4
Tone Parameters
Parameter
Description
SigTimer
Timeout values for signal events.
•
Bits 0–7: CWT Period, which the number of 0.1-second intervals to wait
between each burst of call-waiting tone. The range is 0 to 255. The default
value is 100 (0x64, 10 seconds). 0=Use default.
•
Bits 8–13: Reserved.
•
Bits 14–19: Ring Timeout, which is the number of 10-second intervals to
wait between when the phone starts ringing and the phone rejects the
incoming call. Range: 0 to 63. 0=Never time out. Default value: 6 (0x6,
60 seconds).
•
Bits 20–25: NoAns Timeout, which is the number of seconds to wait
between when the phone starts ringing and the phone initiates call
forwarding on no answer. Range: 0 to 63. Default value: 20 (0x14,
20 seconds).
•
Bits 28–29: First Key Repeat Interval, which specifies how long a user
must hold the Volume or the Navigation button up or down before the
desired change begins to repeat. Valid values: 0 (1 second), 1 (disable
repeat), 2 (2 seconds), and 3 (3 seconds). Default value: 0 (1 second).
•
Bits 30–31: Subsequent Key Repeat Interval, which specifies the intervals
at which a change continues to repeat (after the First Key Repeat Interval)
when a user continues to hold the Volume or the Navigation button up or
down. Valid values: 0 (0.25 second), 1 (0.5 second), 2 (0.75 second), and
3 (1 second). Default value: 0 (0.25 second).
Value type: Bitmap.
Default value: 0x01418064
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Table 8-4
Tone Parameters (continued)
Parameter
Description
RingOnOffTime
Specifies the ringer cadence pattern, expressed as three comma-separated
integers a,b,c, where:
•
a—Number of seconds to wait before turning the ring on.
•
b—Number of seconds to wait before turning the ring off.
•
c—Ring frequency.
Value type: Three comma-separated integers.
Default value: 2,4,25
DialTone
Tone that plays when the phone is ready to accept the first digit of a telephone
number or an IP address.
See the “BusyTone, CallWaitTone, DialTone, DialTone2, and RingBackTone
Parameter Details” section on page 8-28 for syntax and usage details.
Value type: Array of 11 short integers.
Default value: 2,31538,814,30831,2032,0,0,0,0,0,0
DialTone2
Secondary dial tone. For example, this tone might play after you dial a number
to obtain an outside line.
See the “BusyTone, CallWaitTone, DialTone, DialTone2, and RingBackTone
Parameter Details” section on page 8-28 for syntax and usage details.
Value type: Array of 11 short integers.
Default value: 2,30743,1384,29864,1252,0,0,0,0,0,0
BusyTone
Tone that plays when the called party’s line is busy.
See the “BusyTone, CallWaitTone, DialTone, DialTone2, and RingBackTone
Parameter Details” section on page 8-28 for syntax and usage details.
Value type: Array of 11 short integers.
Default value: 2,30467,1104,28959,1404,1,4000,4000,0,0,0
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Audio Parameters
Table 8-4
Tone Parameters (continued)
Parameter
Description
ReorderTone
Tone that plays when the called number does not exist or when the external
network circuit is busy.
See the “ReorderTone Parameter Details” section on page 8-30 for syntax and
usage details.
Value type: Array of 17 short integers.
Default value: 0,2,30467,1104,28959,1404,0,0,1,2000,2000,0,0,0,0,0,0
RingBackTone
Tone that plays when the called party’s line is ringing.
See the “BusyTone, CallWaitTone, DialTone, DialTone2, and RingBackTone
Parameter Details” section on page 8-28 for syntax and usage details.
Value type: Array of 11 short integers.
Default value: 2,30831,2032,30467,1104,1,16000,32000,0,0,0
CallWaitTone
Tone that plays to indicate that you have a call waiting.
See the “BusyTone, CallWaitTone, DialTone, DialTone2, and RingBackTone
Parameter Details” section on page 8-28 for syntax and usage details.
Value type: Array of 11 short integers.
Default value: 1,30831,2412,0,0,1,2400,2400,0,0,4800
Audio Parameters
Table 8-5 describes the audio parameters that you can configure through a phone’s
profile or through the Audio Parameters area on the phone’s web page. These
parameters that specify encoding and other audio characteristics for the
Cisco IP Phone.
Tip
To access the Audio Parameters area directly from the a browser, enter
IP_address/AudioConfiguration, where IP_address is the IP address of your
Cisco IP Phone.
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Table 8-5
Audio Parameters
Parameter
Description
RxCodec
Preferred audio decoder (receiving codec):
•
1: G.711A-law
•
2: G.711u-law
•
3: G.729a
Value type: Integer.
Default value: 2
TxCodec
Preferred audio encoder (transmitting codec):
•
1=G.711A-law
•
2=G.711u-law
•
3=G.729a
Value type: Integer.
Default value: 2
AudioMode
Audio operating mode.
•
Bit 0: G.711 silence suppression:
– 0=Disable.
– 1=Enable.
•
Bits 1–3: Reserved.
•
Bits 4–5: DTMF transmission method:
– 0=Always inband.
– 1=Negotiated via SDP.
– 2=Always out-of-band.
•
Bits 6–31: Reserved.
Value type: Bitmap.
Default value: 0x00000011
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Table 8-5
Audio Parameters (continued)
Parameter
Description
ConnectMode
Connection mode for the selected call-signaling protocol.
See the “ConnectMode Detail” section on page 8-36 for syntax and usage
details.
Value type: Bitmap.
Default value: 0x00000000
NumTxFrames
Number of frames per outbound audio RTP packet. For G.711 and G.729
codecs, a frame is 10 ms. Cisco recommends that you use the default value.
Valid values are 1, 2, 3, 4, 5, or 6.
Value type: Integer.
Default value: 2
Profile Parameters
Table 8-6 describes the parameters that you can configure through a phone’s
profile only.
For more information about profiles, see Chapter 6, “Configuring the Cisco IP
Phone Models 7905G and 7912G Using a TFTP Server.”
Table 8-6
Profile Parameters
Parameter
Description
UIPassword
Password to access the phone’s configuration web page, and to unlock the
phone’s Network Configuration and SIP Configuration menus for editing.
If set to 0, configuration parameters cannot be modified from a phone’s web
page. In addition, if set to 0, the phone will not prompt for a password when
you unlock configuration parameters on the phone.
Value type: Alphanumeric string up to nine characters.
upgradecode
Provides information for upgrading the firmware image on the phone.
See the “Upgradecode Parameter Detail” section on page 8-41 for syntax and
usage details.
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Table 8-6
Profile Parameters (continued)
Parameter
Description
upgradelogo
Allows you to specify the graphic that appears on the phone’s LCD screen.
See the “Upgradelogo Parameter Detail” section on page 8-43 for syntax and
usage details.
Include
Includes the parameters and values from another profile when you create the
binary profile.
Specify the file to include as an alphanumeric string. For example, if you are
creating a phone-specific profile and want to include some values from a
common profile, called common.txt, put this parameter and value in the text
phone-specific profile:
include:common.txt
When you run the cfgfmt.exe to create the binary profile from the text profile,
the parameters and values from common.txt will be included in the binary
profile.
Cisco recommends that the Include parameter, if used, be the first parameter in
a text profile file. Any parameters in a text profile that come before an include
statement will be overwritten with values from the same parameters, if they
exist, in a common profile.
Parameter Details
This section includes details about using the following parameters:
•
BusyTone, CallWaitTone, DialTone, DialTone2, and RingBackTone
Parameter Details, page 8-28
•
ReorderTone Parameter Details, page 8-30
•
ConnectMode Detail, page 8-36
•
DialPlan Parameter Detail, page 8-38
•
Upgradecode Parameter Detail, page 8-41
•
Upgradelogo Parameter Detail, page 8-43
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BusyTone, CallWaitTone, DialTone, DialTone2, and
RingBackTone Parameter Details
The following parameters specify tones that can consist of a single frequency or
of two frequencies that play simultaneously, and of a cadence with up to two
on-off pairs:
•
BusyTone—Tone that plays when the called party’s line is busy
•
CallWaitTone—Tone that plays to indicate that you have a call waiting
•
DialTone—Tone that plays when the phone is ready to accept the first digit of
a telephone number or an IP address
•
DialTone2—Secondary dial tone (for example, the dial tone that plays when
you dial a number to obtain an outside line)
•
RingBackTone—Tone that plays to indicate that the called party’s line is
ringing
For the default values of these parameters, which are suitable for the United Sates,
see Table 8-4.
The syntax of these parameters is:
parameter:NumOfFreqs,TFreq_1,TAmplitude_1,TFreq_2,TAmplitude_2,
NumOfOnOffPairs,OnTime_1,OffTime_1,OnTime_2,OffTime_2,TotalToneTime
where:
•
parameter is BusyTone, CallWaitTone, DialTone, DialTone2, or
RingBackTone.
•
NumOfFreqs is the number of frequencies in the tone (1 or 2).
•
TFreq_1 and TFreq_2 are the transformed frequencies of the first and second
frequencies, respectively. Calculate each value as follows, where Frequency
is the desired frequency in Hz:
32767 * cos(2 * pi * Frequency/8000)
The range of each value is –32768 to 32767.
Set a value to 0 if the frequency does not exist.
For negative values, use the 16-bit 2’s complement value. For example, enter
–1 as 65535 or as 0xffff.
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•
TAmplitude_1 and TAmplitude_2 are the transformed amplitudes (volumes)
of the first and second frequencies, respectively. Calculate each value as
follows:
A * 32767 * sin(2 * pi * Frequency/8000)
where:
– A is the amplitude factor, which controls the volume. This value is
calculated as follows, where k is the desired volume in dBm and n is the
number of frequencies (1 or 2):
A = 0.5 * 10((k + 9 – (n – 1) * 3) / 20)
For example, for a one-frequency tone with a volume of –19 dBm,
A = 0.5 * 10((–19 + 9)/20) = 0.16
– Frequency is the desired frequency in Hz
•
NumOfOnOffPairs is the number of on-off pairs in the cadence of the tone (0,
1, or 2). For a steady tone, use 0.
If this value is 0, the OnTime_1, OnTime_2, OffTime_1, and OffTime_2 values
must also be 0.
•
OnTime_1 and OnTime_2 are the lengths of time that the first and second
on-off pairs of a cadence play a sound, respectively. (See Figure 8-1 for a
graphical representation.)
Specify each value as a number of samples with the sampling rate of 8 kHz.
The range of each value is 0 to 0xffff.
For example, for a length of 0.3 seconds, set a value to 2400.
•
OffTime_1 and OffTime_2 are the lengths of silence after the sound of the first
and second on-off pairs of a cadence, respectively. (See Figure 8-1 for a
graphical representation.)
Specify each value as a number of samples with the sampling rate of 8 kHz.
The range of each value is 0 to 0xffff.
For example, for a length of 0.3 seconds, set a value to 2400.
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Cadence with Two On-Off Pairs
OnTime_1
OnTime_2
Sound
Silence
•
OffTime_1
OffTime_2
99267
Figure 8-1
TotalToneTime is the total length of time that the tone plays. If this value is 0,
the tone will play until another call event stops the tone.
For the BusyTone, DialTone, DialTone 2, and RingBackTone parameters,
specify this value in 10 ms units (100 ms = 1 second). For the CallWaitTone
parameter, specify this value as a number of samples with the sampling rate
of 8 kHz.
ReorderTone Parameter Details
The ReorderTone parameter specifies the tone that plays when the called number
is not available or the external circuit is busy. This tone can consist of either
•
Up to three frequencies played simultaneously and a cadence of up to three
on-off pairs. The first on-off pair can repeat multiple times before the second
on-off pair plays.
For example, a 400 Hz frequency plays four times for 0.75 second followed
by 0.1 second of silence after each play and then plays one time for 0.75
second followed by 0.4 second of silence. This pattern can be set to repeat
until another call event stops it.
•
Up to three frequencies played sequentially with a cadence of up to three
on-off pairs
For example, the frequencies 900 Hz, 1400 Hz, and 1800 Hz play sequentially
for 0.33 seconds each with no silence after the first and second frequencies
but 1 second of silence after the third frequency.
For the default value of this parameter, which is suitable for the United States, see
Table 8-4.
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The syntax of the ReorderTone parameters is:
ReorderTone:Sequential,NumOfFreqs,TFreq_1,TAmplitude_1,TFreq_2,
TAmplitude_2,TFreq_3,TAmplitude_3,NumOfOnOffPairs,OnTime_1,OffTime_1,
OnTime_2,OffTime_2,OnTime_3,OffTime_3,NumOfRepeats,TotalToneTime
where:
•
Sequential specifies whether multiple frequencies in a tone play
simultaneously (0) or sequentially (1). Set to 0 for a tone with one frequency.
•
NumOfFreqs is the number of frequencies in the tone (1, 2, or 3). The
frequencies can play simultaneously or sequentially, depending on the
Sequential setting.
•
TFreq_1, TFreq_2, and TFreq_3 are the transformed frequencies of the first,
second, and third frequencies, respectively. Calculate each value as follows,
where Frequency is the desired frequency in Hz:
32767 * cos(2 * pi * Frequency/8000)
The range of each value is –32768 to 32767.
Set a value to 0 if the frequency does not exist.
For negative values, use the 16-bit 2’s complement value. For example, enter
–1 as 65535 or as 0xffff.
•
TAmplitude_1, TAmplitude_2, and TAmplitude_3 are the transformed
amplitudes (volumes) of the first, second, and third frequencies, respectively.
Calculate each value as follows:
A * 32767 * sin(2 * pi * Frequency/8000)
where:
– A is the amplitude factor, which controls the volume. Calculate this value
as follows, where k is the desired volume in dBm and n is the number of
frequencies (1, 2, or 3):
A = 0.5 * 10((k + 9 – (n – 1) * 3) / 20)
For example, for a one-frequency tone with a volume of –19 dBm,
A = 0.5 * 10((–19 + 9)/20) = 0.16
– Frequency is the desired frequency in Hz
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•
NumOfOnOffPairs is the number of on-off pairs in the cadences of the tone
(0, 1, 2, or 3). For a steady tone, use 0.
If this value is 0, the OnTime_1, OnTime_2, OnTime_ 3, OffTime_1,
OffTime_2, and OffTime_3 values must also be 0.
•
OnTime_1, OnTime_2, and OnTime_3 are the lengths of time that the first,
second, and third on-off pairs of a cadence play a sound, respectively. (See
Figure 8-2 for a graphical representation.)
Specify each value as a number of samples with the sampling rate of 8 kHz.
The range of each value is 0 to 0xffff.
For example, for a length of 0.3 seconds, set a value to 2400.
•
OffTime_1, OffTime_2, and OffTime_3 are the lengths of silence after the
sound of the first, second, and third on-off pairs of a cadence, respectively.
Specify each value as a number of samples with the sampling rate of 8 kHz.
The range of each value is 0 to 0xffff.
For example, for a length of 0.3 seconds, set a value to 2400. (See Figure 8-2
for a graphical representation.)
Cadence with Three On-Off Pairs
OnTime_1
Silence
•
OnTime_3
OnTime_2
Sound
OffTime_1
OffTime_2
OffTime_3
99267
Figure 8-2
NumOfRepeats is the number of times that the first on-off pair of the cadence
(specified by OnTime_1, OffTime_1) repeats before the second on-off pair
(specified by OnTime_2, OffTime_2) plays.
For example, if NumOfRepeats is 2, the first on-off pair will play three times
(it will play once and then repeat two times), then the second on-off pair will
play.
•
TotalToneTime is the total length of time that the tone plays. If this value is 0,
the tone will play until another call event stops the tone.
This value is in 10 ms units (100 ms = 1 second).
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ReorderTone Parameter Example1
Assume that you want a reorder tone in which:
•
The frequencies 900 Hz, 1400 Hz, and 1800 Hz play sequentially.
•
Each frequency plays once for 0.33 seconds.
•
There is no silence after the first and the second frequencies.
•
There is 1 second of silence after the third frequency (before the first
frequency starts again)
•
The volume of each frequency is –19 dBm.
•
The tone plays until another call event stops the tone.
For this reorder tone, make the following setting. See Table 8-7 for a detailed
explanation.
ReorderTone:1,3,24917,3405,14876,4671,5126,5178,3,2640,0,2640,0,
2640,8000,0,0
Table 8-7
Reorder Tone Parameter Example 1 Explanation
Component
Setting
Explanation
Sequential
1
Frequencies play sequentially
NumOfFreqs
3
Three frequencies in the tone
TFreq_1
24917
First frequency is 900 Hz
TAmplitude_1
3405
First frequency volume is –19 dBm
TFreq_2
14876
Second frequency is 1400 Hz
TAmplitude_2
4671
Second frequency volume is –19 dBm
TFreq_3
5126
Third frequency is 1800 Hz
TAmplitude_3
5178
Third frequency volume is –19 dBm
NumOfOnOffPairs
3
Three on-off pairs in the cadence of the tone
OnTime_1
2640
Sound in first on-off pair plays for 0.33
seconds
OffTime_1
0
No silence after the first sound (the second
sound plays immediately)
OnTime_2
2640
Sound in second on-off pair plays for 0.33
seconds
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Table 8-7
Reorder Tone Parameter Example 1 Explanation (continued)
Component
Setting
Explanation
OffTime_2
0
No silence after the second sound (the third
sound plays immediately)
OnTime_3
2640
Sound in third on-off pair plays for 0.33
seconds
OffTime_3
8000
1 second of silence after the sound in the third
on-off pair (before the pattern repeats,
beginning with the first on-off pair)
NumOfRepeats
0
First on-off pair of the cadence plays once
(does not repeat), then the second on-off pair
plays
TotalToneTime
0
Tone plays continuously (set of three on-off
pairs of the cadence repeat continuously) until
another call event stops the tone
ReorderTone Parameter Example 2
Assume that you want a reorder tone in which:
•
The only frequency is 400 Hz.
•
The frequency plays six times, each time for 0.1 second followed by 0.9
second of silence.
•
The frequency then plays once for 0.3 second followed by 0.7 second of
silence.
•
The volume of the frequency is –19 dBm.
•
The tone plays until another call event stops the tone.
For this reorder tone, make the following setting. See Table 8-8 for a detailed
explanation.
ReorderTone:0,1,31164,1620,0,0,0,0,2,800,7200,2400,5600,0,0,5,0
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Table 8-8
Reorder Tone Parameter Example 2 Explanation
Component
Setting
Explanation
Sequential
0
Required setting for a tone with one frequency
NumOfFreqs
1
One frequency in the tone
TFreq_1
31164
First frequency is 400 Hz
TAmplitude_1
1620
First frequency volume is –19 dBm
TFreq_2
0
No second frequency
TAmplitude_2
0
No second frequency
TFreq_3
0
No third frequency
TAmplitude_3
0
No third frequency
NumOfOnOffPairs
2
Two on-off pairs in the cadence of the tone
OnTime_1
800
Sound in first on-off pair plays for 0.1 second
OffTime_1
7200
Sound in first on-off pair is followed by 0.9
second of silence
OnTime_2
2400
Sound in second on-off pair plays for 0.3
seconds
OffTime_2
5600
Sound in second on-off pair is followed by 0.7
second of silence
OnTime_3
0
No third on-off pair in the cadence
OffTime_3
0
No third on-off pair in the cadence
NumOfRepeats
5
First on-off pair of the cadence plays six times
(plays once and then repeats five times), then
the second on-off pair plays
TotalToneTime
0
Tone plays continuously (set of two on-off
pairs of the cadence repeat continuously) until
another call event stops the tone
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ConnectMode Detail
The ConnectMode parameter specifies the connection mode for the selected
call-signaling protocol, as described in Table 8-9.
Table 8-9
ConnectMode Parameter
Bit
Description
0–15
Reserved. Must be set to 0.
16
Registration removal prior to re-registration.
•
0=Disable (default).
•
1=When the phone powers up, “Contact: *” is used to remove all registrations. On
subsequent registration cycles, “Contact: current_SIP_URL;expires=0” is used.
17–18
Reserved. Must be set to 0.
19
IP ringback and early media.
20
21
•
0=Do not send a ringback tone to the caller (default).
•
1=Send a ring back tone to the caller.
Include “action=proxy” in REGISTER request. Do not enable if bit 21 is enabled.
•
0=Disable (default).
•
1=Enable.
Include “action=redirect” in REGISTER request.
•
0=Disable (default).
•
1=Enable.
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Table 8-9
ConnectMode Parameter (continued)
Bit
Description
22
Process a received = parameter in the VIA header to extract the external IP addresses
used by the Network Address Translation (NAT) router.
•
0=Disable (default).
•
1=Enable.
When a Cisco IP Phone is operating behind a NAT, the NATIP parameter must be set to
the external IP address of the NAT router. This setting allows the correct IP address to
be placed in the Contact and SDP headers.
You may leave the NATIP address set to the default value of 0 (or 0.0.0.0) and let the
phone automatically scan the VIA header for a received = parameter. The parameter, if
present, would indicate that the phone is operating behind a firewall.
The phone proceeds as follows:
1.
If the received = parameter is an INVITE response, the current INVITE is canceled
and a new INVITE is sent with the new IP address extracted from the
received =NAT_IP_Address parameter in the Contact header. This step is performed
only if registration is currently in an idle state.
2.
If the received = parameter is in a REGISTER response as a result of a REGISTER
command, the phone will cancel all previous registrations and re-register with the
new IP address extracted from the received = NAT_IP_Address parameter in the
Contact header.
Note
23
For the phone to automatically detect its presence behind a NAT, the SIP proxy
server or remote user agent server must include the received = parameter in the
VIA header in the responses to the phone if the proxy detects that the source
address and port do not match those in the VIA header.
Reserved. Must be set to 0.
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Table 8-9
ConnectMode Parameter (continued)
Bit
Description
24
Include RTP statistics in BYE request and response.
•
0=Disable (default).
•
1=Enable.
If this bit is enabled, the phone will insert the headers RxStat and TxStat as follows:
RxStat: Dur=a,Pkt=b,Oct=c,LatePkt=d,LostPkt=e,AvgJit=f
TxStat: Dur=g,Pkt=h,Oct=i
where:
25–31
•
Dur is the total number of seconds since the beginning of reception or transmission
•
Pkt is the total number of RTP packets received or transmitted
•
Oct is the total number of RTP payload octets received or transmitted (not including
RTP header)
•
LatePkt is the total number of late RTP packets received
•
LostPkt is the total number of lost RTP packets received (not including late RTP
packets)
•
AvgJit is the average jitter, which is an estimate of the statistical variance of the RTP
packet inter-arrival time, measured in timestamp unit and calculated according to
RFC 1889.
•
a, b, c, d, e, f, g, h, and i are integers
Reserved. Must be set to 0.
DialPlan Parameter Detail
The DialPlan parameter specifies dial plan rules.
Note
No syntax check is performed by the implementation of this parameter. It is the
responsibility of the provisioner to make sure that the dial plan is syntactically
valid.
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The DialPlan field provides programmable strings of dial plan that allow you to
specify:
•
Special rule—Itimeout to control default interdigit timeout. Specifying this
rule also has the side effect of preventing nonmatching dial strings from being
sent out.
•
Optional send character to use. For example, the pound sign (#) or the asterisk
(*).
•
Number of digits before auto-send.
•
Send after timeout at any specified number of digits (timeout can be changed
as digits are entered).
The DialPlan field uses the following conventions:
•
Special characters translate as shown in Table 8-10.
Table 8-10 DialPlan Parameter Special Characters
Character
Action
Period (.)
Match any digit.
Comma (,)
Play secondary dial tone as specified by DialTone2.
Hyphen (-)
More digits can be entered. If this character is needed, it
must appear at the end of an individual rule. For example,
1408t5- is legal, but 1408t5-3... is illegal.
Pound sign (#)
The terminating key to send is #, and termination can be
applied only after matching hits >#. So >* means that the
terminating character is the asterisk (*); that is, the
terminating key must follow the greater-than sign (>).
Characters “tn”
Set the timeout to n seconds. Note: n is 0–9 and a–z, which
ranges from 0 to 26.
Characters “rn”
Repeat the last pattern n times. Note: The characters “>#”
and “tn” are modifiers, not patterns. n is 0–9 and a–z, which
ranges from 0 to 26. Use the repeat modifier to specify
more rules in less space.
Modifier “S”
Cause rule-matching to cease (that is, if a rule matches and
the modifier “S” is seen, all other rules after that matching
rule are not used for matching).
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•
Rules are applied in the listed order. The first rule that completely matches
triggers the send.
•
Multiple rules are separated by a pipe (|).
Dial Plan Rules Example 1
The following two sets of dial plan rules are translated as described in Table 8-11:
“.t7>#......t4-|911|1t7>#..........t1-|0t4>#.t7-”
“.t7>#r6t4-|911|1t7>#.r9t1-|0t4>#.t7-”
Table 8-11
Dial Plan Rules Example 1 Explanation
String
Effect
.t7>#......t4-
At least one digit needs to be entered. After that, send
after 7 seconds. The terminating # character can also be
applied after the first digit is entered. After 7 digits are
entered, the timeout changes to 4 seconds. The
* character means that more digits can be entered, as
long as the string is not terminated by timeout or #.
911
Send immediately.
1t7>#..........t1-
At least one digit needs to be entered. After that, send
after 7 seconds. The terminating character # can also be
applied after the first digit is entered. After 10 digits are
entered, the timeout changes to 1 second. The
* character means that more digits can be entered, as
long as the string is not terminated by timeout or #.
0t4>#.t7-”
After a 0, if no other digit is entered, send after
4 seconds. If another digit is entered, send after
7 seconds. Again, # is the terminating digit.
Dial Plan Rules Example 2
The following set of dial plan rules is translated as described in Table 8-12:
"911|1>#.r9t3.t5-|0t411t9-"
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Table 8-12 Dial Plan Rules Example 2 Explanation
String
Effect
911
Send immediately.
1>#.r9t3.t5-
If 14088713344 is entered, send after 3 seconds. If another
digit is entered (for example, 140887133445), send after
5 seconds.
0t411t9-
If 0 is entered, send after 4 seconds. If 011 is entered, send
after 9 seconds.
Upgradecode Parameter Detail
The upgradecode parameter provides the phone with instructions it uses to
download the latest firmware image from the TFTP server. (For information about
editing a profile, see Chapter 6, “Configuring the Cisco IP Phone Models 7905G
and 7912G Using a TFTP Server.”)
The syntax of the upgradecode parameter is:
upgradecode:3,type,0x0400,0x0100,tftp_server_ip,69,image_id,
image_file_name
where:
•
type is 0x501 for the Cisco IP Phone 7905G or 0x601 for the
Cisco IP Phone 7912G.
•
tftp_server_ip is the IP address of the TFTP server on which the firmware
upgrade image is stored. Enter 0 to use the TFTP server specified by the
TFTPServer1 parameter or by DHCP.
•
image_id is a unique 32-bit integer value. This value must be different for
each upgrade. For this value, use 0x followed by the build date on the
firmware image file name. (For example, if the image file name is
CP79050101SIP030323A.ZUP, then the build date is 030323A, and image_id
would be 0x030323A.)
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•
image_file_name is the file name of the firmware image, in the following
format. (This format includes spaces between elements for clarity. Do not
include spaces between elements when you enter and image file name in the
upgradecode parameter.)
CP model VV vv RR SIP yymmdd a-f ext
where:
– model is 7905 for the Cisco IP Phone 7905G or 7912 for the
Cisco IP Phone 7912G
– VV is the major version number (always two digits)
– vv is the minor version number (always two digits)
– RR is build revision number (always two digits)
– SIP is the protocol and should not be changed
– yymmdd is a two-digit year, two-digit month, and two-digit day
– a-f is the build letter (yymmdd and a-f together form the build date of the
image
– ext is the extension .zup or .sbin
When you enter the upgradecode parameter, you must enter the following values
exactly as shown in the syntax:
•
3 (preceding type)
•
0x0400, 0x0100 (preceding tftp_server_ip)
•
69 (default TFTP server port number, preceding image_id)
Note
A firmware image can contain a signature that allows a phone to
verify the authenticity of the image. A firmware image with such a
signature has the extension .sbin. A firmware image without such a
signature has the extension .zup. A Cisco IP Phone 7905G that cannot
perform image authentication will request a .zup image, which will
provide it with the ability to verify the authenticity of .sbin images in
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future upgrades. A phone that can perform image authentication will
only request a .sbin file. If the profile of a phone that can perform
image authentication specifies a .zup firmware image instead of a
.sbin firmware image, the phone will automatically modify the
request to obtain a .sbin firmware image. (A Cisco IP Phone 7912G
will always request a .sbin image.)
The default value for the upgradecode parameter is:
3,0x501,0x0400,0x0100,0.0.0.0,69,0,none
Use this default value as a starting point when you enter a value. (This default
value includes 0x501 for a Cisco IP Phone 7905G. If you are using a
Cisco IP Phone 7912G, make sure to change 0x501 to 0x601.)
Process
When you store a new firmware image on the TFTP server and you update
image_id in the upgradecode parameter, the Cisco IP Phone upgrades its firmware
with the new image. The phone contacts the TFTP server that is specified by
tftp_server_ip in the upgradecode parameter.
Example
upgradecode:3,0x501,0x0400,0x0100,192.168.2.170,69,0x030609a,CP7905010
1SIP030609A.zup
This parameter setting instructs a Cisco IP Phone 7905G to upgrade its firmware
to CP79050101SIP030609A.zup by downloading this file from the TFTP server
IP address of 192.168.2.170. This download occurs after the phone downloads its
configuration file that contains the upgradecode parameter and only if the phone’s
internally-cached image_id is different than the value 0x030609a.
Upgradelogo Parameter Detail
When the Cisco IP Phone is not in use for a call, a figure appears on the LCD
screen above the soft key labels and below the time and date. Using TFTP
configuration, you can change this figure to one that you provide. For example,
you might change the figure to your company logo.
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Before you can change the figure on the phone, you must use a graphics program
to create an image with the following attributes:
•
Width of 88 pixels.
•
Height of 27 pixels.
•
Monochrome (black-and-white).
•
Saved as bitmap file with the extension .bmp. For example, mylogo.bmp.
In addition, you will need the bmp2logo.exe tool, which converts your bitmap
image file to a .logo file that can be downloaded to the phone. This tool is bundled
with the Cisco IP Phone software.
To change the figure that appears on a Cisco IP Phone, perform the following
steps. As part of this procedure, you will need to update the phone’s profile. For
information about editing a profile, see Chapter 6, “Configuring the Cisco IP
Phone Models 7905G and 7912G Using a TFTP Server.”
Procedure
Step 1
Open a command window and type the following command:
bmp2logo imageID image.bmp image.logo
where:
•
imageID is a unique identifier for the new figure. This identifier must be a
number from 0 through 4294967295 and must be different than the identifier
of the figure that is currently on the phone.
•
image is the base file name of the image that you previously created and saved
with the graphics program.
Note
The imageID of the image that comes with the phone is 1.
For example, if the image identifier is 10 and the base name of your image file is
mylogo, type this command:
bmp2logo 10 mylogo.bmp mylogo.logo
Step 2
Copy the image.logo file to the root directory on the TFTP server.
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Step 3
Add the following line to the Cisco IP Phone profile:
upgradelogo:imageID,TFTPServerIP,image.logo
where:
•
imageID is the same unique identifier that you specified in Step 1.
•
TFTPServerIP is the IP address of the TFTP server on which the image.logo
file is stored. If the image.logo file is stored on the same TFTP server as the
Cisco IP Phone configuration file, replace TFTPServerIP with the numeral 0.
•
image is the base file name of the image file.
For example, if the image identifier is 10, the converted file is stored on the same
TFTP server as the Cisco IP Phone configuration file, and the base name of the
converted image file is mylogo, add the following line to the configuration file:
upgradelogo:10,0,mylogo.logo
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C H A P T E R
9
Troubleshooting the
Cisco IP Phone Models 7905G and
7912G
This chapter describes some basic testing and troubleshooting procedures for the
Cisco IP Phone models 7905G and 7912G. It contains the following sections:
•
General Troubleshooting Tips, page 9-1
•
Resolving Problems, page 9-2
•
Contacting TAC, page 9-5
•
Debugging, page 9-5
General Troubleshooting Tips
The suggestions in this section are general troubleshooting tips.
•
If using a Dynamic Host Configuration Protocol (DHCP) server, make sure
that it is operating correctly. The DHCP server should show an incoming
request from the MAC address listed on the product label.
•
If you do not see the status screen after you connect the Ethernet cable, make
sure the power cord and the Ethernet connection are secure.
•
If you do not hear a dial tone, make sure that the telephone-handset line cord
is plugged into the Cisco IP Phone port. Also make sure that the appropriate
Ethernet cable (crossover or straight-through) is used, and that all cable
connections are secure.
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•
A fast-busy tone indicates that the number you called is not valid or that
external circuits are busy. Verify the number or try your call again later.
•
If you place a call to another IP telephone, hear ringback, and the called party
answers but you cannot hear the speaker’s voice, verify that the
Cisco IP Phone and the other IP telephone support at least one common audio
codec (G.711 A-law, G.711 u-law, or G.729a).
•
If you use a Cisco PIX firewall, make sure that it is version 5 or later.
Related Topics
•
Resolving Problems, page 9-2
•
Contacting TAC, page 9-5
•
Debugging, page 9-5
Resolving Problems
This section describes errors or unexpected results that you may encounter with
the Cisco IP Phone models 7905G and 7912G. It includes possible causes and
recommended actions, where appropriate.
Symptom Parameters with values set through a phone’s web interface or LCD
screen revert to their original settings.
Possible Cause You are using TFTP for configuring the phone (the UseTFTP
parameter is set to 1). Parameter values in a profile will overwrite values that
you set from a phone’s Network Configuration menu, its SIP Configuration
menu, or its web page.
Recommended Action If you are using TFTP to configure phones, do not use
the web page or the phone menus to modify SIP or network parameter values.
Use the phone’s Network Configuration menu only to initially configure
phone.
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Symptom Time is not displayed or is displayed incorrectly on the phone.
Possible Cause
– A Network Time Protocol (NTP) server is not specified for the phone or
the NTP server is returning an incorrect time.
– The TimeZone parameter (in the profile or a phone’s web page) or the
Time Zone parameter (in the Network Configuration menu) is not set
correctly.
Recommended Action
– Specify at least one NTP server for the phone and check the NTP server.
– Verify the setting of the TimeZone or the Time Zone parameter. (You
must update the TimeZone or the Time Zone parameter value manually
when daylight savings time goes into effect and ends.)
Symptom The phone plays a fast busy tone.
Possible Cause User name or password is incorrect.
Recommended Action Verify that the user name and password are correct.
Symptom Ready screen does not appear on the phone.
Possible Cause
– The Cisco IP Phone is trying to obtain the DHCP address or the firmware
image is being upgraded.
– The Ethernet cable is unplugged.
Recommended Action
– Wait for a few moments.
– Plug in the Ethernet cable.
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Symptom There is no dial tone on the phone.
Possible Cause No user ID was entered, or the UID parameter is set to 0 or a
period (.).
Recommended Action Verify the UID parameter value and enter the correct
user ID.
Symptom The phone plays an incorrect busy tone, call waiting tone, dial tone,
reorder tone, or ringback tone.
Possible Cause There is an incorrect setting for the BusyTone, CallWaitTone,
DialTone, DialTone2, RingBackTone, or ReorderTone parameter.
Recommended Action Verify that these parameters are set correctly for your
country. For more information, see the “BusyTone, CallWaitTone, DialTone,
DialTone2, and RingBackTone Parameter Details” section on page 8-28 or
the “ReorderTone Parameter Details” section on page 8-30.
Symptom Cisco IP Phone 7905G resets repeatedly after upgrading its firmware
image.
Possible Cause The phone cannot perform image authentication, but it has
upgraded to a .SBIN firmware image file instead of to a .ZUP image file. (A
.SBIN firmware image file requires that the phone be able to perform image
authentication.)
Recommended Action Upgrade the phone to a .ZUP image file. For more
information, see the “Upgradecode Parameter Detail” section on page 8-41.
Symptom The Cisco logo does not appear properly on the web page for a phone.
Possible Cause You are using Netscape version 4.78 or earlier.
Recommended Action When accessing a phones web page, use Netscape
release 6.0 or later, or use Internet Explorer.
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Contacting TAC
Related Topics
•
General Troubleshooting Tips, page 9-1
•
Contacting TAC, page 9-5
•
Debugging, page 9-5
Contacting TAC
If you need to contact the Cisco Technical Assistance Center (TAC) as explained
in the “Obtaining Technical Assistance” section on page xiv, be prepared to
provide the following information:
•
Product type.
•
Software version number, hardware version number, and serial number. To
identify these numbers from a phone, choose Menu > Settings > Status >
Firmware Versions. You can also identify the MAC address from the Device
Information area of the Cisco IP Phone web page.
•
MAC address. To identify this information from a phone, choose Menu >
Settings > Network Configuration and scroll to the MAC Address
parameter. You can also identify the MAC address from the Device
Information area of the Cisco IP Phone web page.
Debugging
You can collect information relating to the operations of a Cisco IP Phone and
store this information in a log file on a server that you specify.This information
can be useful for debugging and troubleshooting.
To collect information for debugging, you will need the prserv.exe tool. This tool
is bundled with the Cisco IP Phone software.
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Debugging
To collect debugging information in a log file, perform the following steps. For
information about editing the Cisco IP Phone profile, see Chapter 6, “Configuring
the Cisco IP Phone Models 7905G and 7912G Using a TFTP Server.”
Step 1
Add the following line to the Cisco IP Phone profile:
Nprintf:IPaddress.listen_port
where:
•
IPaddress is the IP address of the server to which information will be sent.
•
listen_port is the port to which information is sent on the server specified.
For example, if you want to send information to port 9001 on a server with an IP
address of 192.168.2.170, set the Nprintf parameter to 192.168.2.170.9001.
Note
Step 2
If you set this parameter to 0 or to 0.0.0.0.0, no information will be sent
from the phone.
On the server that will receive information, open a command window and type the
following command:
prserv [listen_port]
where listen_port is the same port that you specified with the Nprintf parameter.
If you do not specify a port, 9001 is used by default.
While prserv is running, information from the Cisco IP Phone is sent to a file
named listen_port.log, where listen_port is the port you specified with the Nprintf
parameter and the prserv command. This file is stored on the server you specified
with the Nprintf parameter in the folder from which you ran the prserv command.
If the file exists, new information is appended to it.
You can open a listen_port.log file using a text editor.
Step 3
To exit prserv.exe and stop collecting information, press Ctrl-C in the command
window.
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Note
If you want to collect information from more than one Cisco IP Phone, Cisco
recommends that you store information in a separate listen_port.log file for each
phone. If you store all information in a single file, it can be difficult to identify
which messages belong to which phones. To generate separate listen_port.log
files, configure each phone using its own profile or web page. For each phone, use
the Nprintf parameter in the profile or web page to specify a unique port number
on the server to which information will be sent. Then, run prserv.exe multiple
times, once for each phone from which you want to collect information. Run each
instance of prserv.exe in its own command window. For each instance, make sure
to specify a port number that corresponds to a port number in a Nprintf parameter.
Related Topics
•
General Troubleshooting Tips, page 9-1
•
Resolving Problems, page 9-2
•
Contacting TAC, page 9-5
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A P P E N D I X
A
Additional End User Features on the
Cisco IP Phone Models 7905G and
7912G
This appendix describes Cisco IP Phone end user features that are not included in
the current version of the Cisco IP Phone end user documentation. This
information is provided here so that you can make it available to end users.
This appendix includes these topics:
•
Call Preferences Menu, page A-1
•
Softkeys, page A-6
•
Other Features, page A-7
Call Preferences Menu
End users access the Call Preferences menu from the Settings menu on a
Cisco IP Phone. What parameters appear on this menu and whether end users can
change the value of a parameter depends on the Call Preference settings that you
make for the phone in the profile or from the configuration web pages. For more
information about configuring access to the Call Preferences menu and
parameters, see the “Call Preference Parameters” section on page 8-13.
For information about editing parameter values, see the “Editing Parameter
Values” section on page 5-5.
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Call Preferences Menu
To display the Call Preferences menu, follow these steps:
Procedure
Step 1
Press the Menu button to access the Services menu.
Step 2
In the Services menu, use the Navigation button to select Settings, and then press
the Select softkey.
Step 3
In the Settings menu, use the Navigation button to select Call Preferences, and
then press the Select softkey.
To exit the Call Preferences menu, press the Back softkey until the Exit softkey
appears, and then press the Exit softkey.
Table A-1 describes the parameters that can appear in the Call Preferences menu
and explains how to change them.
Table A-1
Call Preference Menu Parameters
Parameter
Description
DisplayName
Specifies an alpha-numeric string that is
used as part of the phone’s caller ID and that
appears on the phone’s LCD screen.
1.
Scroll to the DisplayName
parameter, press the Edit softkey,
and then enter a name.
If the Block Caller ID parameter is
set to Yes, outgoing calls will use
“Anonymous” as the phone’s caller
ID regardless of the Display Name
value.
2.
Press the Accept softkey, and then
press the Save softkey.
Specifies a name to be displayed on the
phone’s LCD screen in place of the Display
Name value. If the Short Name parameter is
blank or set to 0, the Display Name value
will be displayed. Short Name will not
affect the Display Name value that is used
as part of the phone’s caller ID.
1.
Scroll to the ShortName parameter,
press the Edit softkey, and then
enter a name.
2.
Press the Accept softkey, and then
press the Save softkey.
Note
ShortName
To Change
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Table A-1
Call Preference Menu Parameters (continued)
Parameter
Description
To Change
Do Not Disturb Specifies whether calls to the phone will
receive a busy signal.
1.
Scroll to the Do Not Disturb
parameter and press the Yes
softkey to enable this parameter, or
press the No softkey to disable it.
2.
Press the Save softkey.
1.
Scroll to the Allow Call Waiting
parameter and press the Yes
softkey to enable this parameter, or
press the No softkey to disable it.
2.
Press the Save softkey.
Specifies whether caller ID is blocked on
outgoing calls. If this parameter is set to
Yes, outgoing calls will use “Anonymous”
as the phone’s caller ID. If it is set to No, the
phone’s caller ID will include the Display
Name value and phone number.
1.
Scroll to the Block Caller ID
parameter and press the Yes
softkey to enable this parameter, or
press the No softkey to disable it.
2.
Press the Save softkey.
Specifies whether anonymous calls are
rejected. If this parameter is set to Yes,
incoming calls with “Anonymous” caller
IDs will be rejected. If it is set to No, these
calls will ring at the phone.
1.
Scroll to the Block Anonymous
Call parameter and press the Yes
softkey to enable this parameter, or
press the No softkey to disable it.
2.
Press the Save softkey.
Specifies whether attended call transfers
can be made from the phone. If this
parameter is set to Yes, the Trnsfer softkey
will appear when appropriate, allowing
attended transfers of a calls. If this
parameter is set to No, this softkey will
never be available.
1.
Scroll to the Allow Call Transfer
parameter and press the Yes
softkey to enable this parameter, or
press the No softkey to disable it.
2.
Press the Save softkey
When this parameter is set to Yes, the DND
softkey becomes available on the phone.
See the “Softkeys” section on page A-6 for
more information.
Allow Call
Waiting
Block Caller
ID
Block
Anonymous
Call
Allow Call
Transfer
Specifies whether call waiting is enabled on
the phone.
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Call Preferences Menu
Table A-1
Call Preference Menu Parameters (continued)
Parameter
Description
Allow Blind
Transfer
Specifies whether blind call transfers can be
made from the phone. If this parameter is
set to Yes, the BlndXfr softkey will appear
when appropriate, allowing blind transfers
of calls. If this parameter is set to No, this
softkey will never be available.
To Change
1.
Scroll to the Allow Blind Transfer
parameter and press the Yes
softkey to enable this parameter, or
press the No softkey to disable it.
2.
Press the Save softkey
Specifies whether conference calls can be
established from the phone. If this
parameter is set to Yes, the Confrn softkey
will appear when appropriate, allowing
conference calls to be established. If this
parameter is set to No, this softkey will
never be available.
1.
Scroll to the Allow Conference
parameter and press the Yes
softkey to enable this parameter, or
press the No softkey to disable it.
2.
Press the Save softkey
Specifies the format for the time that
appears on the phone’s LCD screen. This
format is specified by one or more of the
following characters:
1.
Scroll to the Time Format
parameter, press the Edit softkey,
and then enter a time format.
2.
Press the Accept softkey, and then
press the Save softkey.
See the “Other Features” section on
page A-7 for more information about blind
transfer.
Allow
Conference
Time Format
h—designates the hour in 12-hour format
H—designates the hour in 24-hour format
i or I—designates minutes
a or A—for 12-hour format, include “p”
with times from noon until one minute
before midnight
: (colon)—displays a colon that blinks
every second
For example, when the current time is 1:30
P.M.,
•
h:ia causes the time to appear as 1:30p.
•
H:I causes the time to appear as 13:30.
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Table A-1
Call Preference Menu Parameters (continued)
Parameter
Description
To Change
Date Format
Specifies the format for the date that
appears on the phone’s LCD screen. This
format is specified by one or more of the
following characters:
1.
Scroll to the Date Format
parameter, press the Edit softkey,
and then enter a date format.
2.
Press the Accept softkey, and then
press the Save softkey.
1.
Scroll to the Voice Mail Number
parameter, press the Edit softkey,
and then enter a phone number.
2.
Press the Accept softkey, and then
press the Save softkey.
m—designates the month as a number 1
through 12
M—designates the month as a three-letter
abbreviation, Jan through Dec
d or D—designates the day of the month
y—designates a two-digit year (such as 03
for 2003)
Y—designates a four-digit year
Other characters—appear as entered
For example, when the current date is
March 10, 2003:
Voice Mail
Number
•
m-d-y causes the date to appear as
3-10-03.
•
M d, Y causes the date to appear as Mar
20, 2003.
•
Y/m/d causes the date to appear as
2003/3/10.
Specifies a phone number where voice
messages are stored and retrieved. If a
number is specified:
•
The Message softkey will appear when
appropriate.
•
Incoming calls forward to this number
if the phone is busy or if the calls are
not answered within the number of
seconds specified with the Forward to
VMail Delay(s) parameter.
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Appendix A
Additional End User Features on the Cisco IP Phone Models 7905G and 7912G
Softkeys
Table A-1
Call Preference Menu Parameters (continued)
Parameter
Description
Call Forward
Number
Specifies a phone number to which all calls
to the phone are forwarded. If a number is
entered, the Call Forward All feature is
activated.
1.
Scroll to the Call Forward Number
parameter, press the Edit softkey,
and then enter a phone number.
2.
Press the Accept softkey, and then
press the Save softkey.
Specifies the number of seconds after which
an incoming ringing call will be forwarded
to the number specified by the Voice Mail
Number parameter.
1.
Scroll to the Forward to VMail
Delay(s) parameter, press the Edit
softkey, and then enter a value.
2.
Press the Accept softkey, and then
press the Save softkey.
Forward to
VMail
Delay(s)
To Change
Softkeys
Table A-2 describes the softkeys that can appear on the
Cisco IP Phone models 7905G and 7912G and that are not described in the User
Guide for the phone.
Table A-2
Sofkteys
Softkey
Description
Back
Displays the previous menu.
BlndXfr
Allows the blind transfer of a call when Allow Blind
Transfer is enabled in the Call Preferences menu. In a
blind transfer, you transfer a caller to another number
without talking to the party who is to receive the call or
verifying that the new party is there.
To perform a blind transfer, follow these steps:
1.
During a call, press the More soft key, then press
the BlndXfr soft key. The call goes on hold.
2.
Dial the number to which to transfer the call. The
phone hangs up automatically.
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Additional End User Features on the Cisco IP Phone Models 7905G and 7912G
Other Features
Table A-2
Sofkteys (continued)
Softkey
Description
Clear
In the Call Preferences menu, clears the contents of the
currently-displayed parameter.
Default
In the Call Preferences menu, changes the value of the
currently-displayed parameter to its default value.
DND
Disables the Do Not Disturb feature and changes the
setting of the Do Not Disturb parameter in the Call
Preferences menu to No.
Drop
Allows the initiator of a conference call to disconnect
the most recently connected active party in the call.
Factory
Changes volume and contrast settings to their default
values. You access this softkey by pressing the Menu
button, selecting Settings, and then pressing the More
softkey.
Missed
Displays the Missed Calls directory.
Restore
Restores volume and contrast settings to the values
currently stored in non-volatile memory on the phone.
You access this softkey by pressing the Menu button,
selecting Settings, and then pressing the More softkey.
Other Features
Table A-3 describes features on the Cisco IP Phone models 7905G and 7912G
that are not described elsewhere in this appendix or in the User Guide for the
phone.
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Appendix A
Additional End User Features on the Cisco IP Phone Models 7905G and 7912G
Other Features
Table A-3
Other Features
Feature
Description
Ringer sound
The ringer sound cannot be changed on the phone.
Stutter tone for
message waiting
In addition to lighting the red light on the handset, the
Cisco IP Phone plays a stutter tone to indicate one or
more new voice mail messages. A stutter tone is a series
of short dial tones that a user hears instead of a regular
dial tone when placing a call.
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A P P E N D I X
B
Phone Menu, Web Page, and Profile
Equivalent Parameters
Table B-1 shows the configurable parameters on the phone menus and their
equivalent parameters in the phone configuration web pages and profile.
Table B-1
Phone Menu, Web Page and Profile Parameters
Phone Menu Parameter
Phone Web Page and Profile Parameter
Admin. VLAN Id
VLANSetting
Alternate DNS
DNS2IP
Alternate NTP
AltNTPIP
Alternate TFTP
TftpURL
Backup Proxy Timeout
AltProxyTimeOut
CDP Enabled
OpFlags
Default Router
StaticRoute
DHCP Enabled
Dhcp
DNS Server 1
DNS1IP
DNS Server 2
DNS2IP
Domain (if the Alternate Domain
option is set to yes)
Domain
IP Address
StaticIP
Local RTP Port
MediaPort
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Appendix B
Table B-1
Phone Menu, Web Page, and Profile Equivalent Parameters
Phone Menu, Web Page and Profile Parameters (continued)
Phone Menu Parameter
Phone Web Page and Profile Parameter
Local SIP Port
SIPPort
Login ID
LoginID
NAT WAN IP Address
NATIP
NTP Server 1
NTPIP
NTP Server 2
AltNTPIP
Outbound Proxy
OutboundProxy
Password
PWD
Profile Encrypt Key
EncryptKey
Register Expires
SipRegInterval
Register with Proxy
SipRegOn
SIP Proxy
Proxy
Subnet Mask
StaticNetMask
TFTP Enabled
UseTftp
TFTP Server
TftpURL
Time Zone
TimeZone
Use Login ID
UseLoginID
User ID
UID
VLAN Enabled
OpFlags
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A P P E N D I X
C
SIP Call Flows
This appendix describes some basic call flows for the
Cisco IP Phone models 7905G and 7912G. It includes the following sections:
•
Supported SIP Request Methods, page C-1
•
Call Flow Scenarios for Successful Calls, page C-2
Supported SIP Request Methods
The Cisco IP Phone supports the following SIP request methods:
•
INVITE—Indicates a user or service is being invited to participate in a call
session.
•
ACK—Confirms that the client has received a final response to an INVITE
request.
•
BYE—Terminates a call and can be sent by either the caller or the callee.
•
CANCEL—Cancels any pending searches but does not terminate a call that
has already been accepted.
•
REGISTER—Registers the address listed in the To header field with a SIP
proxy.
•
NOTIFY—Notifies the user of the status of a transfer using REFER. Also
used for remote reset and for message waiting.
•
OPTIONS—Provides requestors with SIP methods and codecs supported by
the phone.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
The following types of responses are used by SIP and generated by the Cisco SIP
gateway:
•
SIP 1xx—Informational responses
•
SIP 2xx—Successful responses
•
SIP 3xx—Redirection responses
•
SIP 4xx—Client Failure responses
•
SIP 5xx—Server Failure responses
•
SIP 6xx—Global Failure responses
Call Flow Scenarios for Successful Calls
This section describes call flows for the following scenarios:
•
Cisco IP Phone-to-SIP Server—Registration without Authentication,
page C-2
•
Cisco IP Phone-to-SIP Server—Registration with Authentication, page C-4
•
Cisco IP Phone-to-Cisco IP Phone—Basic SIP to SIP Call without
Authentication, page C-7
•
Cisco IP Phone-to-Cisco IP Phone—Basic SIP to SIP Call with
Authentication, page C-15
Each of the call flows includes a call diagram, action descriptions table, and a
sample log file.
Cisco IP Phone-to-SIP Server—Registration without
Authentication
Figure C-1 illustrates the Cisco IP Phone registering with the SIP server where
authentication is not required for registration. Table C-1 describes these actions
in detail. Table C-2 shows the log of these actions.
The call flow is as follows:
1.
Cisco IP Phone requests registration.
2.
Registration is completed.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Figure C-1
Cisco IP Phone-to-SIP Server—Registration without
Authentication
Cisco IP Phone
IP network
SIP server
IP
1. REGISTER
2. 100 Trying
Table C-1
94052
3. 200 OK
Action Descriptions
Step
Action
Description
Step 1
REGISTER—Cisco IP Phone to SIP server Cisco IP Phone sends a REGISTER message to
the SIP server to register the address in the To
header field.
Step 2
100 Trying—SIP Server to Cisco IP Phone SIP server returns a 100 Trying message,
indicating that the REGISTER request has been
received.
Step 3
200 OK—SIP server to Cisco IP Phone
SIP server returns a final 200 OK response,
confirming that registration is complete.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-2
Log Listing
1.
REGISTER sip:cisco.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=1692201497
To: <sip:9313@cisco.com;user=phone>
Call-ID: 2923175725@192.168.2.163
CSeq: 1 REGISTER
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>;expires=3600
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
2.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 2923175725@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=1692201497
To: <sip:9313@cisco.com;user=phone>
CSeq: 1 REGISTER
Content-Length: 0
3.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 2923175725@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=1692201497
To: <sip:9313@cisco.com;user=phone>
CSeq: 1 REGISTER
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>;expires="Fri, 02
May 2003 00:52:58 GMT"
Expires: 3600
Content-Length: 0
Cisco IP Phone-to-SIP Server—Registration with Authentication
Figure C-2 illustrates the Cisco IP Phone registering with the SIP server where
authentication is required for registration. Table C-3 describes these actions in
detail. Table C-4 shows the log of these actions.
The call flow is as follows:
1.
Cisco IP Phone requests registration.
2.
SIP server requests authentication credential.
3.
Authentication is received and registration is completed.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Figure C-2
Cisco IP Phone-to-SIP Server—Registration with Authentication
IP network
Cisco IP Phone
SIP server
IP
1. REGISTER
2. 100 Trying
3. 401 Unauthorized
4. REGISTER
5. 100 Trying
Table C-3
94053
6. 200 OK
Action Descriptions
Step
Action
Description
Step 1
REGISTER—Cisco IP Phone to SIP server Cisco IP Phone sends a REGISTER message to
the SIP server to register the address in the To
header field.
Step 2
100 Trying—SIP server to Cisco IP Phone
SIP server returns a 100 trying message,
indicating that the REGISTER request has been
received.
Step 3
401 Unauthorized— SIP server to
Cisco IP Phone
SIP server returns a request for authentication.
Step 4
REGISTER—Cisco IP Phone to SIP server Cisco IP Phone attempts to register using its
authentication credential.
Step 5
100 Trying—SIP server to Cisco IP Phone
SIP server returns a 100 trying message,
indicating that the new REGISTER request has
been received.
Step 6
200 OK—SIP server to Cisco IP Phone
SIP server returns a final 200 OK response,
confirming that the authentication credential has
been verified and registration is complete.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-4
Log Listing
1.
REGISTER sip:cisco.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=118440597
To: <sip:9313@cisco.com;user=phone>
Call-ID: 948614169@192.168.2.163
CSeq: 1 REGISTER
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>;expires=3600
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
2.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 948614169@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=118440597
To: <sip:9313@cisco.com;user=phone>
CSeq: 1 REGISTER
Content-Length: 0
3.
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 948614169@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=118440597
To: <sip:9313@cisco.com;user=phone>
CSeq: 1 REGISTER
WWW-Authenticate: DIGEST realm="CISCO", nonce="3eb1ad3b", algorithm=MD5
Content-Length: 0
4.
REGISTER sip:cisco.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=118440597
To: <sip:9313@cisco.com;user=phone>
Call-ID: 948614169@192.168.2.163
CSeq: 2 REGISTER
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>;expires=3600
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Authorization: Digest
username="9313",realm="CISCO",nonce="3eb1ad3b",uri="sip:cisco.com",response="e52
5065aac09e798058122020e3d1921"
Content-Length: 0
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-4
Log Listing (continued)
5.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 948614169@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=118440597
To: <sip:9313@cisco.com;user=phone>
CSeq: 2 REGISTER
Content-Length: 0
6.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 948614169@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=118440597
To: <sip:9313@cisco.com;user=phone>
CSeq: 2 REGISTER
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>;expires="Fri, 02
May 2003 00:26:51 GMT"
Expires: 3600
Content-Length: 0
Cisco IP Phone-to-Cisco IP Phone—Basic SIP to SIP Call
without Authentication
Figure C-3 illustrates a call from one Cisco IP Phone to another where
authentication by the SIP server is not required. Table C-5 describes these actions
in detail. Table C-6 shows the log of these actions.
The call flow is as follows:
1.
Call is established between Cisco IP Phone A and Cisco Cisco IP Phone B.
2.
Call is terminated.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Figure C-3
Cisco IP Phone-to-Cisco IP Phone—Basic SIP to SIP Call without Authentication
Cisco IP Phone A
IP network
SIP server
IP network
IP
Cisco IP Phone B
IP
1. INVITE
2. 100 Trying
3. INVITE
4. 100 Trying
5. 180 Ringing
6. 180 Ringing
7. 200 OK
8. 200 OK
9. ACK
10. ACK
2-way voice path
11. BYE
13. BYE
14. 200 OK
94054
15. 200 OK
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-5
Action Descriptions
Step
Action
Description
Step 1
INVITE—Cisco IP Phone A to SIP server
Cisco IP Phone A sends a call session
INVITE request to the SIP server to pass on to
Cisco IP Phone B.
Step 2
100 Trying—SIP server to Cisco IP Phone A SIP server returns a 100 trying message,
indicating that the INVITE request has been
received.
Step 3
INVITE—SIP server to Cisco IP Phone B
Step 4
100 Trying—Cisco IP Phone B to SIP server Cisco IP Phone B returns a 100 trying
message indicating that the INVITE request
has been received.
Step 5
180 Ringing—Cisco IP Phone B to SIP
server
Cisco IP Phone B sends a 180 ringing
response to the SIP server to pass on to
Cisco IP Phone A.
Step 6
180 Ringing—SIP server to Cisco IP Phone
A
SIP server sends the 180 ringing response to
Cisco IP Phone A.
Step 7
200 OK—Cisco IP Phone B to SIP server
Cisco IP Phone B sends a 200 OK message to
the SIP server indicating that a connection has
been established.
Step 8
200 OK—SIP server to Cisco IP Phone A
SIP server passes the 200 OK message to
Cisco IP Phone A.
Step 9
ACK—Cisco IP Phone A to SIP server
Cisco IP Phone A sends acknowledgement of
the 200 OK response to the SIP server to pass
on to Cisco IP Phone B.
Step 10
ACK—SIP server to Cisco IP Phone B
SIP server passes ACK response to
Cisco IP Phone B.
SIP server sends the call session INVITE
request to Cisco IP Phone B.
A two-way voice path is established between Cisco IP Phone A and Cisco IP Phone B.
Step 11
BYE—Cisco IP Phone A to SIP server
Cisco IP Phone A terminates the call session
and sends a BYE request to the SIP server
indicating that Cisco IP Phone A wants to
terminate the call.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-5
Action Descriptions (continued)
Step
Action
Description
Step 12
BYE—SIP server to Cisco IP Phone B
SIP server passes the BYE request to
Cisco IP Phone B.
Step 13
200 OK—Cisco IP Phone B to SIP server
Cisco IP Phone B sends a 200 OK message to
the SIP server indicating that Cisco IP Phone
B has received the BYE request.
Step 14
200 OK—SIP server to Cisco IP Phone A
SIP server passes the BYE request to
Cisco IP Phone A.
Table C-6
1.
Log Listing
INVITE sip:9314@cisco.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>
Call-ID: 4074632257@192.168.2.163
CSeq: 1 INVITE
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Expires: 300
Content-Length: 259
Content-Type: application/sdp
v=0
o=9313 58115 58115 IN IP4 192.168.2.163
s=Cisco 7905 SIP Call
c=IN IP4 192.168.2.163
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-6
Log Listing (continued)
2.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 4074632257@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>
CSeq: 1 INVITE
Content-Length: 0
3.
INVITE sip:9314@192.168.3.232:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=5b495e54-d5a8a93e-aa5f9258-be025820-1
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>
Call-ID: 4074632257@192.168.2.163
CSeq: 1 INVITE
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Expires: 300
Content-Length: 259
Content-Type: application/sdp
v=0
o=9313 58115 58115 IN IP4 192.168.2.163
s=Cisco 7905 SIP Call
c=IN IP4 192.168.2.163
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
4.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=5b495e54-d5a8a93e-aa5f9258-be025820-1
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 1 INVITE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-6
Log Listing (continued)
5.
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=5b495e54-d5a8a93e-aa5f9258-be025820-1
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 1 INVITE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
6.
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 1 INVITE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
7.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=5b495e54-d5a8a93e-aa5f9258-be025820-1
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 1 INVITE
Contact: <sip:9314@192.168.3.232:5060;user=phone;transport=udp>
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 205
Content-Type: application/sdp
v=0
o=9314 58362 58362 IN IP4 192.168.3.232
s=Cisco 7905 SIP Call
c=IN IP4 192.168.3.232
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-6
8.
Log Listing (continued)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 1 INVITE
Contact: <sip:9314@192.168.3.232:5060;user=phone;transport=udp>
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 205
Content-Type: application/sdp
v=0
o=9314 58362 58362 IN IP4 192.168.3.232
s=Cisco 7905 SIP Call
c=IN IP4 192.168.3.232
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
9.
ACK sip:9314@192.168.3.232;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 1 ACK
User-Agent: Cisco-CP7905/1.01-030429A
Content-Length: 0
10.
ACK sip:9314@192.168.3.232;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 1 ACK
User-Agent: Cisco-CP7905/1.01-030429A
Content-Length: 0
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-6
Log Listing (continued)
11.
BYE sip:9314@192.168.3.232;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 2 BYE
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
12.
BYE sip:9314@192.168.3.232;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 2 BYE
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
13.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 2 BYE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
14.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=2274744465
To: <sip:9314@cisco.com;user=phone>;tag=252438364
Call-ID: 4074632257@192.168.2.163
CSeq: 2 BYE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Cisco IP Phone-to-Cisco IP Phone—Basic SIP to SIP Call with
Authentication
Figure C-4 illustrates a call from one Cisco IP Phone to another where
authentication by the SIP server is required. Table C-7 describes these actions in
detail. Table C-8 shows the log of these actions.
The call flow is as follows:
1.
Authentication is requested for call initiated by Cisco IP Phone A.
2.
Call is established between Cisco IP Phone A and Cisco IP Phone B.
3.
Call is terminated.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Figure C-4
Cisco IP Phone-to-Cisco IP Phone—Basic SIP to SIP Call with Authentication
Cisco IP Phone A
IP network
SIP server
IP network
IP
Cisco IP Phone B
IP
1. INVITE
2. 100 Trying
3. 407 Proxy Authentication Required
4. ACK
5. INVITE
6. 100 Trying
7. INVITE
8.100 Trying
9. 180 Ringing
10. 180 Ringing
11. 200 OK
12. 200 OK
13. ACK
14. ACK
2-way voice path
15. BYE
17. BYE
18. 200 OK
94055
19. 200 OK
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-7
Action Descriptions
Step
Action
Description
Step 1
INVITE—Cisco IP Phone A to SIP server
Cisco IP Phone A sends a call session
INVITE request to the SIP server to pass on to
Cisco IP Phone B.
Step 2
100 Trying—SIP server to Cisco IP Phone A SIP server returns a 100 Trying message,
indicating that the INVITE request has been
received.
Step 3
407 Proxy Authentication Required— SIP
server to Cisco IP Phone A
SIP server returns a request for authentication
to Cisco IP Phone A.
Step 4
ACK—Cisco IP Phone A to SIP server
Cisco IP Phone A acknowledges the request
for authentication.
Step 5
INVITE—Cisco IP Phone A to SIP server
Cisco IP Phone A sends a call session
INVITE request along with authentication
credential to the SIP server to pass on to
Cisco IP Phone B.
Step 6
100 Trying—SIP server to Cisco IP Phone A SIP server returns a 100 Trying message,
indicating that the INVITE request has been
received.
Step 7
INVITE—SIP server to Cisco IP Phone B
Step 8
100 Trying—Cisco IP Phone B to SIP server Cisco IP Phone B returns a 100 trying
message indicating that the INVITE request
has been received.
Step 9
180 Ringing—Cisco IP Phone B to SIP server Cisco IP Phone B sends a 180 ringing
response to the SIP server to pass on to
Cisco IP Phone A.
Step 10
180 Ringing—SIP server to Cisco IP Phone A SIP server sends the 180 ringing response to
Cisco IP Phone A.
Step 11
200 OK—Cisco IP Phone B to SIP server
Cisco IP Phone B sends a 200 OK message to
the SIP server indicating that a connection has
been established.
Step 12
200 OK—SIP server to Cisco IP Phone A
SIP server passes the 200 OK message to
Cisco IP Phone A.
SIP server sends the call session INVITE
request to Cisco IP Phone B.
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-7
Action Descriptions (continued)
Step
Action
Description
Step 13
ACK—Cisco IP Phone A to SIP server
Cisco IP Phone A sends acknowledgment of
the 200 OK response to the SIP server to pass
on to Cisco IP Phone B.
Step 14
ACK—SIP server to Cisco IP Phone B
SIP server passes ACK response to
Cisco IP Phone B.
A two-way voice path is established between Cisco IP Phone A and Cisco IP Phone B.
Step 15
BYE—Cisco IP Phone A to SIP server
Cisco IP Phone A terminates the call session
and sends a BYE request to the SIP server
indicating that Cisco IP Phone A wants to
terminate the call.
Step 16
BYE—SIP server to Cisco IP Phone B
SIP server passes the BYE request to
Cisco IP Phone B.
Step 17
200 OK—Cisco IP Phone B to SIP server
Cisco IP Phone 186 B sends a 200 OK
message to the SIP server indicating that
Cisco IP Phone 186 B has received the BYE
request.
Step 18
200 OK—SIP server to Cisco IP Phone A
SIP server passes the BYE request to
Cisco IP Phone A.
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-8
1.
Log Listing
INVITE sip:9314@cisco.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>
Call-ID: 2765277948@192.168.2.163
CSeq: 1 INVITE
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Expires: 300
Content-Length: 257
Content-Type: application/sdp
v=0
o=9313 8737 8737 IN IP4 192.168.2.163
s=Cisco 7905 SIP Call
c=IN IP4 192.168.2.163
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
2.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 2765277948@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>
CSeq: 1 INVITE
Content-Length: 0
3.
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 2765277948@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=4d241cc8-1145efd4
CSeq: 1 INVITE
Proxy-Authenticate: DIGEST realm="CISCO", nonce="3eb1be0a", algorithm=MD5
Content-Length: 0
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-8
Log Listing (continued)
4.
ACK sip:9314@cisco.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=4d241cc8-1145efd4
Call-ID: 2765277948@192.168.2.163
CSeq: 1 ACK
User-Agent: Cisco-CP7905/1.01-030429A
Content-Length: 0
5.
INVITE sip:9314@cisco.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>
Call-ID: 2765277948@192.168.2.163
CSeq: 2 INVITE
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Proxy-Authorization: Digest
username="9313",realm="CISCO",nonce="3eb1be0a",uri="sip:9314@cisco.com",response
="40190e28f6f098437e4a7edcd73c4ce6"
Expires: 300
Content-Length: 257
Content-Type: application/sdp
v=0
o=9313 8749 8749 IN IP4 192.168.2.163
s=Cisco 7905 SIP Call
c=IN IP4 192.168.2.163
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
6.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
Call-ID: 2765277948@192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>
CSeq: 2 INVITE
Content-Length: 0
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-8
7.
Log Listing (continued)
INVITE sip:9314@192.168.3.232:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=fa2b71c3-292cb31a-a4158bcf-2cb346be-1
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>
Call-ID: 2765277948@192.168.2.163
CSeq: 2 INVITE
Contact: <sip:9313@192.168.2.163:5060;user=phone;transport=udp>
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Expires: 300
Content-Length: 257
Content-Type: application/sdp
v=0
o=9313 8749 8749 IN IP4 192.168.2.163
s=Cisco 7905 SIP Call
c=IN IP4 192.168.2.163
t=0 0
m=audio 16384 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
8.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=fa2b71c3-292cb31a-a4158bcf-2cb346be-1
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 2 INVITE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
9.
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=fa2b71c3-292cb31a-a4158bcf-2cb346be-1
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 2 INVITE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-8
Log Listing (continued)
10.
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 2 INVITE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
11.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=fa2b71c3-292cb31a-a4158bcf-2cb346be-1
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 2 INVITE
Contact: <sip:9314@192.168.3.232:5060;user=phone;transport=udp>
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 203
Content-Type: application/sdp
v=0
o=9314 8644 8644 IN IP4 192.168.3.232
s=Cisco 7905 SIP Call
c=IN IP4 192.168.3.232
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-8
12.
Log Listing (continued)
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.163:5060;received=192.168.2.163
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 2 INVITE
Contact: <sip:9314@192.168.3.232:5060;user=phone;transport=udp>
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 203
Content-Type: application/sdp
v=0
o=9314 8644 8644 IN IP4 192.168.3.232
s=Cisco 7905 SIP Call
c=IN IP4 192.168.3.232
t=0 0
m=audio 16384 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13.
ACK sip:9314@192.168.3.232;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 2 ACK
User-Agent: Cisco-CP7905/1.01-030429A
Proxy-Authorization: Digest
username="9313",realm="CISCO",nonce="3eb1be0a",uri="sip:9314@cisco.com",response
="40190e28f6f098437e4a7edcd73c4ce6"
Content-Length: 0
14.
ACK sip:9314@192.168.3.232;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 2 ACK
User-Agent: Cisco-CP7905/1.01-030429A
Proxy-Authorization: Digest
username="9313",realm="CISCO",nonce="3eb1be0a",uri="sip:9314@cisco.com",response
[0]DPKT 1st: 800 560, pt 13
="40190e28f6f098437e4a7edcd73c4ce6"
Content-Length: 0
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Appendix C
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table C-8
Log Listing (continued)
15.
BYE sip:9314@192.168.3.232;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 3 BYE
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Proxy-Authorization: Digest
username="9313",realm="CISCO",nonce="3eb1be0a",uri="sip:9314@cisco.com",response
="8b95de72274483131516ebb2b4086d71"
Content-Length: 0
16.
BYE sip:9314@192.168.3.232;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 3 BYE
User-Agent: Cisco-CP7905/1.01-030429A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Proxy-Authorization: Digest
username="9313",realm="CISCO",nonce="3eb1be0a",uri="sip:9314@cisco.com",response
="8b95de72274483131516ebb2b4086d71"
Content-Length: 0
17.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 3 BYE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
18.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.163:5060
From: <sip:9313@cisco.com;user=phone>;tag=3907367229
To: <sip:9314@cisco.com;user=phone>;tag=338213918
Call-ID: 2765277948@192.168.2.163
CSeq: 3 BYE
Server: Cisco-CP7905/1.01-030422A
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 0
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
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A P P E N D I X
D
Specifications for the
Cisco IP Phone Models 7905G and
7912G
This appendix provides specifications for the Cisco IP Phone models 7905G and
7912G. It contains the following sections:
•
Physical Specifications, page D-2
•
Electrical Specifications, page D-2
•
Environmental Specifications, page D-3
•
Physical Interfaces, page D-3
•
Software Specifications, page D-3
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Appendix D
Specifications for the Cisco IP Phone Models 7905G and 7912G
Physical Specifications
Physical Specifications
Table D-1
Cisco IP Phone models 7905G and 7912G Physical Specifications
Description
Specification
Dimensions
Length: 8.5 in. (21.7 cm)
Width: 7.0 in. (17.8 cm)
Depth: 3.8 in. (9.5 cm) without snap-on base or 5.9 in.
(14.9 cm) with base
Weight
2.0 lb (0.9 kg)
Electrical Specifications
Table D-2
Cisco IP Phone models 7905G and 7912G Electrical Specifications
Description
Specification
Power
2.5 to 3.5W (idle to peak)
DC input voltage
+48 VDC at 1.5A maximum
Power adaptor
Universal AC/DC
~3.3 x 2.0 x 1.3 in. (~8.5 x 5.0 x 3.2 cm)
~4.8 oz (135 g) for the AC-input external power adaptor
~4 ft (1.2 m) DC cord
6 ft (1.8 m) cord
UL/CUL, CE approved
Class II transformer
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Appendix D
Specifications for the Cisco IP Phone Models 7905G and 7912G
Environmental Specifications
Environmental Specifications
Table D-3
Cisco IP Phone models 7905G and 7912G Environmental
Specifications
Description
Specification
Operating temperature
41 to 104°F (5 to 40°C)
Storage temperature
–4 to 140°F (–20 to 65°C)
Relative humidity
10 to 90% noncondensing, operating, and
nonoperating/storage
Physical Interfaces
Table D-4
Cisco IP Phone models 7905G and 7912G Physical Interfaces
Description
Specification
Ethernet
RJ-45 8-wire connector, IEEE 802.3
10BASE-T standard
Power
48-VDC power connector
Software Specifications
Table D-5
Cisco IP Phone models 7905G and 7912G Software Specifications
Description
Specification
Voice-over-IP (VoIP) protocols
SIP (RFC 2543)
Voice coder-decoders (codecs)
G.729, G.729A, G.729AB
G.711 A-law
G.711 u-law
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Appendix D
Specifications for the Cisco IP Phone Models 7905G and 7912G
Software Specifications
Table D-5
Cisco IP Phone models 7905G and 7912G Software Specifications
Description
Specification
Provisioning and configuration
DHCP (RFC 2131)
Web configuration via built-in web server
Basic boot provisioning (RFC 1350 TFTP
Profiling)
Dial plan provisioning
Cisco Discovery Protocol (CDP)
Security
RC4 encryption for TFTP configuration
profiles
Call progress tones
Configurable for up to two sets of
frequencies and multiple on/off cadences
Voice features
Voice activity detection (VAD)
Comfort noise generation (CNG)
Dynamic jitter buffer (adaptive)
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G L O S S A RY
Numerics
10BASE-T
110-Mbps baseband Ethernet specification using two pairs of twisted-pair
cabling (Categories 3, 4, or 5): one pair for transmitting data and the other for
receiving data. 10BASE-T, which is part of the IEEE 802.3 specification, has a
distance limit of approximately 328 feet (100 meters) per segment.
A
A-law
ITU-T companding standard used in the conversion between analog and digital
signals in PCM systems. A-law is used primarily in European telephone
networks and is similar to the North American u-law standard. See also
companding and u-law.
B
BTXML
Basic Telephony Extensible Markup Language. See XML.
C
category x cable
One of five grades of UTP cabling described in the EIA/TIA-586 standard.
Category 3 cabling is used in 10BASE-T networks and can transmit data at
speeds up to 10 Mbps. Category 5 cabling can transmit data at speeds up to
100 Mbps.
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GL-1
Glossary
CDP
Cisco Discovery Protocol. Used primarily to obtain protocol addresses of
neighboring devices and discover the platform of those devices. CDP can also
be used to show information about the interfaces your router uses. CDP is mediaand protocol-independent, and runs on all Cisco-manufactured equipment
including routers, bridges, access servers, and switches.
CED tone
CallED station identification. A 3-second, 2100-Hz tone generated by a fax
machine while answering a call, which is used in the handshaking used to set the
call; the response from a called fax machine to a CNG tone.
CELP
Code excited linear prediction compression. Compression algorithm used in low
bit-rate voice encoding. Used in ITU-T Recommendations G.728, G.729, and
G.723.1.
CLIP
Calling Line Identification Presentation. Shows your identity to callers with
Caller ID.
CLIR
Calling Line Identification Restriction. Hides your identity from callers with
Caller ID.
CNG
Comfort noise generation or calling tone. Distinctive, repeating 1100-Hz tone (on
for 0.5 seconds, off for 3 seconds) generated by a fax machine when placing a call.
codec
Coder-decoder. In Voice over IP, Voice over Frame Relay, and Voice over
ATM, a DSP software algorithm used to compress/decompress speech or audio
signals.
companding
Contraction derived from the opposite processes of compression and expansion.
Part of the PCM process whereby analog signal values are rounded logically to
discrete scale-step values on a nonlinear scale. The decimal step number then is
coded in its binary equivalent before transmission. The process is reversed at the
receiving terminal using the same nonlinear scale. Compare with compression
and expansion. See also A-law and u-law.
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Glossary
compression
The running of a data set through an algorithm that reduces the space required
to store or the bandwidth required to transmit the data set. Compare with
companding and expansion.
CoS
Class of service. An indication of how an upper-layer protocol requires a
lower-layer protocol to treat its messages. In SNA subarea routing, CoS
definitions are used by subarea nodes to determine the optimal route to establish
a given session. A CoS definition comprises a virtual route number and a
transmission priority field. Also called ToS.
D
DHCP
Dynamic Host Configuration Protocol. Provides a mechanism for allocating IP
addresses dynamically so that addresses can be reused when hosts no longer
need them.
dial peer
An addressable call endpoint. In Voice over IP (VoIP), there are two types of
dial peers: POTS and VoIP.
DNS
Domain Name System. System used on the Internet for translating names of
network nodes into IP addresses.
DSL
Digital subscriber line. Public network technology that delivers high bandwidth
over conventional copper wiring at limited distances. There are four types of
DSL: ADSL, HDSL, SDSL, and VDSL. All are provisioned via modem pairs,
with one modem located at a central office and the other at the customer site.
Because most DSL technologies do not use the whole bandwidth of the twisted
pair, there is room remaining for a voice channel.
DSP
Digital signal processor. Specialized hardware and software algorithms that
perform complex processing of digitized data that was originally analog data.
Typically segments a voice signal into frames and stores the frames in voice
packets.
DTMF
Dual tone multifrequency. A type of signaling that combines two distinct
frequencies to generate a tone for each digit or character dialed, which is used by
customers to signal the network. Sometimes referred to as “touchtone,” because a
customer generally touches keypad keys to generate the tones.
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GL-3
Glossary
E
E.164
The international public telecommunications numbering plan. A standard set by
the ITU-T that addresses telephone numbers.
endpoint
A source or sink of voice data, which may be physical (such as a trunk interface
or a line interface in a media gateway) or logical (such as an announcement
stored on a server).
expansion
The process of running a compressed data set through an algorithm that restores
the data set to its original size. Compare with companding and compression.
F
firewall
Router or access server, or several routers or access servers, designated as a
buffer between any connected public networks and a private network. A firewall
router uses access lists and other methods to ensure the security of the private
network.
FQDN
Fully qualified domain name. FQDN is the full name of a system, including the
domain name and not just the host name. For example, aldebaran is a host name,
and aldebaran.interop.com is an FQDN.
FSK
Frequency shift key.
FXO
Foreign exchange office. An FXO interface connects to the public switched
telephone network (PSTN) central office and is the interface offered on a
standard telephone. The Cisco FXO interface is an RJ-11 connector that allows
an analog connection at the PSTN central office or to a station interface on a
PBX.
FXS
Foreign exchange station. An FXS interface connects directly to a standard
telephone and supplies ring, voltage, and dial tone. The Cisco FXS interface is
an RJ-11 connector that allows connections to basic telephone service
equipment, keysets, and PBXs.
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Glossary
G
G.711
The 64-kbps PCM voice coding technique. In G.711, encoded voice is already
in the correct format for digital voice delivery in the PSTN or through PBXs.
Described in the ITU-T standard in its G-series recommendations.
G.723.1
Compression technique that can be used for compressing speech or audio signal
components at a very low bit rate as part of the H.324 family of standards. This
codec has two bit rates associated with it: 5.3 and 6.3 kbps. The higher bit rate
is based on ML-MLQ technology and provides a somewhat higher quality of
sound. The lower bit rate is based on CELP and provides system designers with
additional flexibility. Described in the ITU-T standard in its G-series
recommendations.
G.729A
CELP compression where voice is coded into 8-kbps streams. There are two
variations of this standard (G.729 and G.729 Annex A) that differ mainly in
computational complexity; both provide speech quality similar to 32-kbps
ADPCM. Described in the ITU-T standard in its G-series recommendations.
GMT
Greenwich Mean Time.
I
ICMP
Internet Control Message Protocol. Network-layer Internet protocol that reports
errors and provides other information relevant to IP packet processing.
Documented in RFC 792.
IETF
Internet Engineering Task Force. Task force consisting of over 80 working
groups responsible for developing Internet standards.
IP
Internet Protocol. Network-layer protocol in the TCP/IP stack offering a
connectionless internetwork service. IP provides features for addressing,
type-of-service specification, fragmentation and reassembly, and security.
Defined in RFC 791.
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Glossary
ITU
International Telecommunication Union. An organization established by the
United Nations to set international telecommunications standards and to allocate
frequencies for specific uses.
IVR
Interactive voice response. Describes systems that provide information in the
form of recorded messages over telephone lines in response to user input in the
form of spoken words or, more commonly, DTMF signaling. Examples include
banks that allow you to check your balance from any telephone and automated
stock quote systems.
L
LDAP
Lightweight Directory Access Protocol. Protocol that provides access for
management and browser applications that provide read/write interactive access
to the X.500 Directory.
LEC
Local exchange carrier. A local telephone company or communications common
carrier that provides ordinary local voice-grade telecommunications service
under regulation within a specified service area.
location server
A SIP redirect or proxy server uses a location server to get information about a
caller's location. Location services are offered by location servers.
M
MCU
Multipoint control unit. A bridging or switching device that supports multipoint
videoconferencing.
MGCP
Media Gateway Control Protocol.
MWI
Message waiting indication.
mu-law
Commonly written u-law. North American companding standard used in
conversion between analog and digital signals in PCM systems. Similar to the
European A-law. See also A-law and companding.
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Glossary
N
NAT
Network Address Translation. Mechanism for reducing the need for globally
unique IP addresses. NAT allows an organization with addresses that are not
globally unique to connect to the Internet by translating those addresses into
address space with global routing. Also known as Network Address Translator.
NTP
Network Time Protocol. Protocol built on top of TCP that ensures accurate local
time-keeping with reference to radio and atomic clocks located on the Internet.
This protocol is capable of synchronizing distributed clocks within milliseconds
over long time periods.
P
POTS
Plain Old Telephone Service. Basic telephone service that supplies standard
single-line telephones, telephone lines, and access to the PSTN.
proxy server
An intermediary program that acts as both a server and a client for the purpose
of making requests on behalf of other clients. Requests are serviced internally or
by passing them on, possibly after translation, to other servers. A proxy
interprets and, if necessary, rewrites a request message before forwarding it.
PSTN
Public Switched Telephone Network. PSTN refers to the local telephone
company.
Q
QoS
Quality of service. The ability of a network, whether the network is a complex
network, small corporate network, Internet service provider (ISP), or enterprise
network, to provide better service to selected network traffic over various
technologies, including Frame Relay, ATM, Ethernet and 802.1 networks, and
SONET, as well as IP-routed networks that may use any or all of these
underlying technologies.
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Glossary
R
RAS
Registration, Admission, and Status Protocol. Protocol that is used between
endpoints and a gatekeeper to perform management functions. RAS signalling
function performs registration, admissions, bandwidth changes, status, and
disengage procedures between the VoIP gateway and the gatekeeper.
redirect server
A server that accepts a SIP request, maps the address into zero or more new
addresses, and returns these addresses to the client. It does not initiate its own
SIP request nor accept calls.
registrar server
A server that accepts REGISTER requests. A registrar is typically colocated
with a proxy or redirect server and may offer location services.
RFC 2833
IETF standard that describes RTP payload for DTMF digits, telephony tones,
and telephony signals.
router
Network layer device that uses one or more metrics to determine the optimal
path along which network traffic should be forwarded. Routers forward packets
from one network to another based on network layer information. Occasionally
called a gateway (although this definition of gateway is becoming increasingly
outdated). Compare with gateway.
RSVP
Resource Reservation Protocol. Protocol that supports the reservation of
resources across an IP network. Applications running on IP end systems can use
RSVP to indicate to other nodes the nature (bandwidth, jitter, maximum burst,
and so on) of the packet streams that they want to receive. RSVP depends on
IPv6. Also known as Resource Reservation Setup Protocol.
RTP
Real-Time Transport Protocol. One of the IPv6 protocols. RTP is designed to
provide end-to-end network transport functions for applications that transmit
real-time data, such as audio, video, or simulation data, over multicast or unicast
network services. RTP provides services such as payload type identification,
sequence numbering, time stamping, and delivery monitoring to real-time
applications.
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Glossary
S
SCCP
Skinny Call Control Protocol. A VoIP protocol utilized by a CallManager
server.
SDP
Session Definition Protocol. An IETF protocol for the definition of multimedia
services. SDP messages can be part of SIP, MGCP, and SCCP and messages.
SIP
Session Initiation Protocol. Protocol developed by the IETF MMUSIC Working
Group as an alternative to H.323. SIP features are compliant with IETF
RFC 2543, published in March 1999. SIP equips platforms to signal the setup of
voice and multimedia calls over IP networks.
SIP endpoint
A terminal or gateway that acts as a source or sink of SIP voice data. An
endpoint can call or be called, and it generates or terminates the information
stream.
SLIC
Subscriber Line Interface Circuit. An integrated circuit providing central
office-like telephone interface functionality.
SNA
Systems Network Architecture. Large, complex, feature-rich network
architecture developed in the 1970s by IBM. Similar in some respects to the OSI
reference model but with a number of differences. SNA essentially is composed
of seven layers.
SOHO
Small office, home office. Networking solutions and access technologies for
offices that are not directly connected to large corporate networks.
T
TCP
Transmission Control Protocol. Connection-oriented transport layer protocol
that provides reliable full-duplex data transmission. TCP is part of the TCP/IP
protocol stack.
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Glossary
TFTP
Trivial File Transfer Protocol. Simplified version of FTP that allows files to be
transferred from one computer to another over a network, usually without the
use of client authentication (for example, username and password).
ToS
Type of service. See CoS.
U
UAC
User agent client. A client application that initiates the SIP request.
UAS
User agent server (or user agent). A server application that contacts the user
when a SIP request is received, and then returns a response on behalf of the user.
The response accepts, rejects, or redirects the request.
UDP
User Datagram Protocol. Connectionless transport layer protocol in the TCP/IP
protocol stack. UDP is a simple protocol that exchanges datagrams without
acknowledgments or guaranteed delivery, requiring that error processing and
retransmission be handled by other protocols. UDP is defined in RFC 768.
u-law
See mu-law.
\
V
VAD
Voice activity detection. When enabled on a voice port or a dial peer, silence is
not transmitted over the network, only audible speech. When VAD is enabled,
the sound quality is slightly degraded but the connection monopolizes much less
bandwidth.
VoIP
Voice over IP. The capability to carry normal telephony-style voice over an
IP-based Internet with POTS-like functionality, reliability, and voice quality.
VoIP enables a router to carry voice traffic (for example, telephone calls and
faxes) over an IP network. In VoIP, the DSP segments the voice signal into
frames, which then are coupled in groups of two and stored in voice packets.
VoIP is a blanket term, which generally refers to Cisco’s standard-based
approach to IP voice traffic.
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Glossary
X
XML
eXtensible Markup Language. XML is an open standard for defining data
elements on web pages and in business documents. See also BTXML.
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Glossary
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I N D EX
Numerics
assistance, technical 9-5
attended call transfer A-3
10/100 PC port 2-4
AttendedTransfer parameter 8-14
10/100 SW port 2-4
AudioMode parameter 4-5, 8-25
10 BASET port 2-4
audio parameters
802.1Q VLAN ID 3-5
AudioMode 4-5, 8-25
ConnectMode 4-8, 4-10, 8-26
MediaPort 4-3, 4-7
A
NumTxFrames 8-26
AC adapter 2-5
RxCodec 8-25
ACK C-1
TOS 4-5
Admin. VLAN Id parameter 5-10
TxCodec 8-25
Alert-Info header 4-6
Audio Parameters screen 8-24
Allow Blind Transfer parameter 8-14
authentication 4-4
Allow Call Transfer parameter 8-14
Allow Call Waiting parameter 8-14
Allow Conference parameter 8-14
B
Alpha softkey 5-5
Back softkey A-6
Alternate DNS parameter 5-13
Backup Proxy Timeout parameter 5-16
Alternate Domain parameter 5-13
basic services, list of supported 1-5
Alternate NTP parameter 5-13
blind call transfer A-4, A-6
Alternate TFTP parameter 5-12
BlindTransfer parameter 8-14
AltNTPIP parameter 4-8, 8-6
BlndXfr softkey 8-14, A-6
AltProxyTimeOut parameter 8-10
Block Anonymous Calls parameter 8-15
anonymous calls, blocking 8-15, A-3
BlockAnonymous parameter 8-15
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Index
Block Caller ID parameter 8-15
Block Caller ID 8-15
BlockCallerId parameter 8-15
BlockCallerId 8-15
bmp2logo.exe 8-44
Call Forward Number 8-16
boot load firmware image 2-9
CallForwardNumber 8-16
boot load process 3-2
CallPrefGuiSet 4-5, 8-20
busy tone 8-23, 8-28, 9-3
CallPrefGuiShow 4-5, 8-19
BusyTone parameter 8-23, 8-28
CallWaiting 8-14
BYE C-1
Conference 8-14
Date Format 8-17
DateFormat 8-17, 8-18
C
Display Name 8-16
cable connections
DisplayName 8-16
Cisco IP Phone 7905G 2-10
Do Not Disturb 8-15
Cisco IP Phone 7912G 2-11
DoNotDisturb 8-15
Forward to VMail Delay 8-18
caller ID
blocking A-2, A-3
GUI Set Mask 4-5, 8-20
display name 8-16, A-2
GUI Show Mask 4-5, 8-19
call forwarding 8-16, A-6
Short Name 8-16
Call Forward Number parameter 8-16
ShortName 8-16
CallForwardNumber parameter 8-16
Time Format 8-17
call preference parameters
TimeFormat 8-17
Allow Blind Transfer 8-14
Voice Mail Number 8-15
Allow Call Transfer 8-14
VoiceMailNumber 8-15
Allow Call Waiting 8-14
Allow Conference 8-14
AttendedTransfer 8-14
BlindTransfer 8-14
BlockAnonymous 8-15
Block Anonymous Calls 8-15
Call Preferences menu
allowing users to change parameters on 4-5,
8-20
controlling parameters displayed on 4-5, 8-19
customizing 4-5, 8-19, 8-20
displaying A-2
overview 5-2, A-1
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Index
parameters A-2 to A-6
Call Preferences menu parameters
see Cisco Discovery Protocol
CDP Enabled parameter 5-11
Allow Blind Transfer A-4
cfgfmt.exe tool 6-9, 6-16
Allow Call Transfer A-3
CfgInterval parameter 8-4
Allow Call Waiting A-3
Cisco Discovery Protocol (CDP) 3-3, 3-4, 5-11,
8-7
Allow Conference A-4
Block Anonymous Call A-3
Block Caller ID A-3
Call Forward Number A-6
Date Format A-5
DisplayName A-2
Do Not Disturb A-3
Forward to VMail Delay(s) A-6
ShortName A-2
Time Format A-4
Voice Mail Number A-5
Call Preferences screen 8-13
CallPrefGuiSet parameter 4-5, 8-20
CallPrefGuiShow parameter 4-5, 8-19
call transfer
attended 8-14, A-3
blind 8-14, A-4, A-6
call waiting 8-14, A-3
CallWaiting parameter 8-14
call waiting tone 8-24, 8-28
CallWaitTone parameter 8-24, 8-28
CANCEL C-1
Cancel softkey 5-5
CDP
Cisco IP Phone 7905G
electrical specifications D-2
environmental specifications D-3
graphic on LCD screen 8-27, 8-43
mounting to wall 2-12
physical interface specifications D-3
physical specifications D-2
port 2-4
resets repeatedly 9-4
software specifications D-3
Cisco IP Phone 7912G
electrical specifications D-2
environmental specifications D-3
graphic on LCD screen 8-27, 8-43
mounting to wall 2-12
physical interface specifications D-3
physical specifications D-2
ports 2-4
software specifications D-3
Clear softkey A-7
codec
number of text frames for 8-26
receiving 8-25
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Index
specification D-3
TFTP server 3-1
supported 1-5
through phone menus 3-2
transmitting 8-25
through TFTP 6-2
comfort noise 4-5
conference call 8-14, A-4
Conference parameter 8-14
configuration
alphabetical listing of features with related
parameters 4-12
authentication 4-4
basic 2-13
tone 8-28
configuration file
see profile
configuration menu
displaying 5-3
entering parameter values 5-5
locking parameters 5-4
unlocking parameters 5-4
cfgfmt.exe tool 6-16
dial plan 4-5
effecting changes 3-8
Confrn softkey 8-14, A-4
connecting, phone to network 2-4
ConnectMode parameter 4-8, 4-10, 8-26, 8-36
features and related parameters 4-12
from a Cisco IP Phone 5-4
individual phone 6-3
D
in TFTP server environment, basic
steps 3-6 to 3-8
date format, on LCD screen 8-17, A-5
menus on phone 5-2
minimum settings for network
connectivity 3-9
Date Format parameter 8-17
DateFormat parameter 8-17, 8-18
daylight savings time 5-9, 8-6, 9-3
multiple phones 6-3
debugging 4-5, 8-9, 9-5
overview of methods 3-1
default profile 6-13
refresh interval 4-4, 8-4
default router IP address 5-8, 8-5
required parameters 4-2
Default Router parameter 3-10, 5-8
required SIP parameters 4-2 to 4-3
Default softkey A-7
services and related parameters 4-12
default values, setting parameters to 5-14
silence suppression 4-5
Device Information screen 7-4
SIP proxy server redundancy 4-10
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DHCP
DisplayName parameter 8-16
disabling 5-11, 8-4
distinctive ringing 4-6
enabling 5-11, 8-4
DND softkey A-3, A-7
option 1 6-8
DNS
option 150 6-6, 8-7
disabling alternate 5-13
option 3 6-8
enabling alternate 5-13
option 42 6-8
DNS1IP parameter 8-6
option 6 6-8
DNS2IP parameter 8-6
option 60 6-7
DNS server
IP address of primary 5-8, 8-6
option 66 6-6, 6-8
DHCP Enabled parameter 5-11
IP address of secondary 5-8, 8-6
Dhcp parameter 8-4
DNS Server 1 parameter 5-8
DHCP server
DNS Server 2 parameter 5-8
configuration overview 3-2
DNS SRV lookup 4-6
configuration using 6-5, 6-8
documentation, related xiii
configuration without using 3-3, 3-10, 6-8
domain name
IP address 5-6
disabling alternate 5-13
DHCP Server parameter 5-6
enabling alternate 5-13
diagnostics 4-5
of phone 8-5
dial plan
Domain Name parameter 5-7
configuration 4-5
Domain parameter 8-5
rules 8-13, 8-38
Do Not Disturb 8-15, A-3, A-7
DialPlan parameter 4-5, 8-13, 8-38
Do Not Disturb parameter 8-15
dial tone 8-23, 8-28, 9-4
DoNotDisturb parameter 8-15
DialTone2 parameter 8-23, 8-28, 8-39
downloading, Cisco IP Phone software from
CCO 6-4
DialTone parameter 8-23, 8-28
direct IP-to-IP calls 4-10
display name, for caller ID 8-16, A-2
Drop softkey A-7
DTMF transmission method 8-25
Display Name parameter 8-16
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
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IN-5
Index
E
G
Edit softkey 5-6, 5-14
gkdefault.cfg profile 3-7, 6-3, 6-13
electrical specifications D-2
graphic, on LCD screen 8-27, 8-43
enabling SIP registration 4-2
GUI Set Mask parameter 4-5, 8-20
encryption, of profile 6-15
GUI Show Mask parameter 4-5, 8-19
encryption key 5-14, 8-4
EncryptKey parameter 6-15, 6-16, 8-4
environmental specifications D-3
Erase Configuration parameter 5-14
H
handset 1-3
hardware overview 1-1
Hold button 1-3
F
Host Name parameter 5-6
Factory softkey A-7
fast-busy tone 9-2
features, software 1-4
I
features and related parameters 4-12
Include parameter 3-7, 8-27
firewall 9-2
inline power 2-4, 2-5
firmware image
installation
.sbin extension 6-4, 8-42
configuration 2-13
.zup extension 6-4, 8-42
prerequisites 2-1
boot load 2-9
procedure 2-6
installing on TFTP server 6-4
safety 2-2
signature for authentication 6-4, 8-42
verifying 2-15
upgrading 3-6, 8-26, 8-41
Firmware Versions screen 5-18
Forward to VMail Delay parameter 8-18
INVITE C-1
IP address
default router 5-8, 8-5
DHCP server 5-6
network gateway 5-8
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
IN-6
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Index
phone 5-7, 8-5
Status 5-2, 5-18
specifying manually 3-10
menu, displaying on phone 5-3
IP Address parameter 3-10, 5-7
Menu button 1-3
IPDialPlan parameter 8-13
message waiting A-8
Missed softkey A-7
Model Information menu 5-2, 5-17
L
lddefault.cfg profile 3-4, 3-7, 6-3, 6-13
N
light strip, on handset 1-3
Local RTP Port parameter 4-3, 5-16
name
Local SIP Port parameter 4-3, 5-16
displayed on LCD screen 8-16, A-2
locking configuration parameters 5-4
for caller ID 8-16, A-2
log file 9-6
NATIP parameter 4-3, 4-7, 8-11
Login ID parameter 5-15
NatServer parameter 4-8, 8-12
LoginID parameter 8-10
NatTimer parameter 4-8, 8-12
logo, on LCD screen 8-27, 8-43
NAT WAN IP Address parameter 4-3, 5-17
Navigation button 1-3
network
M
connecting phone to 2-4
requirements for installation 2-1
MAC Address parameter 5-6
MAXRedirect parameter 4-2, 8-11
Media Access Control (MAC) address 5-6
media port 4-7
MediaPort parameter 4-3, 4-7, 8-12
menu
Call Preferences 5-2, A-1
Network Address Translation (NAT),
overview 4-7
Network Address Translation (NAT) WAN, IP
address 5-17, 8-11
Network Address Translation/Port Address
Translation (NAT/PAT) 4-8
Network Configuration menu
Model Information 5-2, 5-17
overview 5-2
Network Configuration 5-2, 5-6
parameters 5-6 to 5-14
SIP Configuration 5-2, 5-14
resetting parameters to default values 5-14
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
OL-4277-01
IN-7
Index
Network Configuration menu parameters
AltNTPIP 4-8, 8-6
Admin. VLAN Id 5-10
CfgInterval 8-4
Alternate DNS 5-13
Dhcp 8-4
Alternate Domain 5-13
DNS1IP 8-6
Alternate NTP 5-13
DNS2IP 8-6
Alternate TFTP 5-12
Domain 8-5
CDP Enabled 5-11
EncryptKey 8-4
Default Router 3-10, 5-8
NPrintf 4-5, 8-9
DHCP Enabled 5-11
NTPIP 4-8, 8-6
DHCP Server 5-6
OpFlags 3-4, 6-17, 8-7
DNS Server 1 5-8
StaticIP 3-10, 8-5
DNS Server 2 5-8
StaticNetMask 3-10, 8-5
Domain Name 5-7
StaticRoute 3-10, 4-7, 8-5
Erase Configuration 5-14
TftpURL 8-3
Host Name 5-6
TimeZone 4-8, 8-6
IP Address 3-10, 5-7
TOS 8-8
MAC Address 5-6
TraceFlags 4-5, 8-9
NTP Server 1 5-9
UseTftp 8-3
NTP Server 2 5-9
VLANSetting 3-5, 8-8
Operational VLAN Id 5-10
Network Parameters screen 8-2
Profile Encrypt Key 5-14
Network Statistics screen 5-18
Subnet Mask 3-10, 5-7
next hop server redundancy 4-9
TFTP Enabled 5-12
No softkey 5-6, 5-14
TFTP Server 5-8
NOTIFY C-1
Time Zone 5-9
NPrintf parameter 4-5, 8-9, 9-6
VLAN Enabled 5-10
NTPIP parameter 4-8, 8-6
network connectivity, minimum settings 3-9
NTP server
network gateway IP address 5-8
address of primary 5-9
network parameters
address of secondary 5-9
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
IN-8
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Index
disabling alternate 5-13
Network Configuration menu 5-6 to 5-14
enabling alternate 5-13
settings revert 9-2
IP address of primary 8-6
SIP 8-9 to 8-13
IP address of secondary 8-6
SIP Configuration menu 5-15 to 5-17
NTP Server 1 parameter 5-9
tone 8-21 to 8-24
NTP Server 2 parameter 5-9
value types 6-11
Number softkey 5-5
parameter tag file 6-9, 6-13, 6-14
NumTxFrames parameter 8-26
password 5-15, 8-10, 8-26
Password parameter 5-15
phone menu configuration 3-2
O
physical interface specifications D-3
Operational VLAN Id parameter 5-10
physical specifications D-2
OpFlags parameter 3-4, 6-17, 8-7
placing call without using SIP proxy 4-10
OPTIONS C-1
port
OutBoundProxy parameter 4-3, 8-12
10/100 PC 2-4
Outbound Proxy parameter 4-3, 4-9, 5-16
10/100 SW 2-4
outbound proxy server, address 4-3
10 BASET 2-4
cable type for 2-4
for incoming SIP requests 4-3
P
media 4-7
padlock icon 5-4
power patch panel, providing power using 2-5
parameters
power source
audio 8-24 to 8-26
description 2-5
call preference 8-13 to 8-20
design 2-5
Call Preference menu A-2 to A-6
external 2-5
format in profile 6-10
inline power 2-5
in profile only 8-26 to 8-27
power patch panel 2-5
menu, web page, and profile equivalents B-1
redundancy 2-6
network 8-2 to 8-9
probing, of static network router 8-7
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
OL-4277-01
IN-9
Index
list of supported 1-4
profile
about 2-7, 6-2
SIP overview 1-7
binary 6-9
specification D-3
common 6-13
Proxy parameter 4-3, 4-9, 8-10
conventions 6-10
proxy redundancy 4-9
creating 6-9
proxy server 1-10
default 6-13
prserv.exe 9-5, 9-6
default values 6-11
ptag.dat 6-9
defined 6-1
PWD parameter 8-10
disabling downloading from TFTP
server 5-12, 8-3
enabling downloading from TFTP
server 5-12, 8-3
encrypting 6-15
example file 6-11
for individual phone 6-3
for multiple phones 6-3
gkdefault.cfg 3-7, 6-3
including parameters from another
profile 8-27
IP address of TFTP server containing 8-3
lddefault.cfg 3-4, 3-7, 6-3
overwriting default name 8-7
phone-specific 6-11
SEPDefault.cnf 3-4
ready screen, on phone 2-9, 9-3
redirection, maximum number of times to
try 4-2
redirect server 1-10
redundancy, for power 2-6
refreshing
Cisco IP Phone 6-17
disabling 8-7
enabling 8-7
refresh interval, configuration 3-8, 4-4, 8-4
REGISTER C-1
Register Expires parameter 4-2, 5-17
Register with Proxy parameter 4-2, 5-17
text 6-10
updating 6-9
XMLDefault.cnf.xml 3-4
Profile Encrypt Key parameter 5-14
protocols
R
registrar server 1-10
registration
with authentication C-4
without authentication C-2
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
IN-10
OL-4277-01
Index
registration renewal, configuring seconds
between 4-2, 5-17, 8-11
Device Information 7-4
registration status icon 8-19
SIP Parameters 8-9
related documentation xiii
secondary dial tone 8-28
reorder tone 8-24, 8-30
SEPDefault.cnf profile 3-4
ReorderTone parameter 8-24, 8-30
services
required parameters, description of 4-2
resetting
SIP, list of supported 1-6
disabling 8-7
supplemental, list of supported 1-5
enabling 8-7
Short Name parameter 8-16
Restore softkey A-7
ShortName parameter 8-16
ring back tone 8-24, 8-28
signal events, timeout values 8-22
RingBackTone parameter 8-24, 8-28
SigTimer parameter 8-22
ringer cadence pattern 8-23
silence suppression 4-5, 8-25
ringer sound A-8
SIP
ringing, distinctive 4-6
call flow scenarios C-2
RingOnOffTime parameter 8-23
clients 1-9
router IP address, specifying manually 3-10
RxCodec parameter 8-25
basic 1-5
related parameters 4-12
Cisco IP Phone 6-17
RTP media port 4-3
Network Parameters 8-2
components 1-8
functions 1-7
overview 1-7
request methods C-1
responses C-2
S
servers 1-10
sample configuration text file 6-11
sip_ptag.dat 6-10
Save softkey 5-5
SIP Configuration menu
screens
overview 5-2
Audio Parameters 8-24
parameters 5-14 to 5-17
Call Preferences 8-13
resetting parameters to default values 5-14
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
OL-4277-01
IN-11
Index
SIP Configuration menu parameters
SIPRegInterval 4-2, 8-11
Backup Proxy Timeout 5-16
SIPRegOn 8-11
Local RTP Port 4-3, 5-16
UID 8-10
Local SIP Port 4-3, 5-16
UseLoginID 4-4, 8-10
Login ID 5-15
SIP Parameters screen 8-9
NAT WAN IP Address 4-3, 5-17
SIPPort parameter 4-3, 4-7, 8-11
Outbound Proxy 4-3, 4-9, 5-16
SIP proxy not used for call 4-10
Password 5-15
SIP Proxy parameter 4-3, 5-15
Register Expires 4-2, 5-17
SIP proxy server
Register with Proxy 4-2, 5-17
IP address 4-3, 5-15, 8-10
SIP Proxy 4-3, 5-15
redundancy 4-10
Use Login ID 5-15
SIPRegInterval parameter 4-2, 8-11
User ID 5-15
SIP registration
sipexample.txt sample configuration file 6-11
disabling 5-17, 8-11
SIP outbound proxy server, IP address 5-16,
enabling 4-2, 5-17, 8-11
8-12
SIPRegOn parameter 4-2, 8-11
SIP parameters
AltProxyTimeOut 8-10
DialPlan 8-13
IPDialPlan 8-13
LoginID 8-10
MAXRedirect 4-2, 8-11
MediaPort 8-12
NATIP 4-7, 8-11
NatServer 4-8, 8-12
NatTimer 4-8, 8-12
OutBoundProxy 4-3, 8-12
Proxy 4-3, 4-9, 8-10
PWD 8-10
SIP services
advanced 4-4
basic 4-2
list of supported 1-6
SIP to SIP call
with authentication C-15
without authentication C-7
softkey
Back A-6
back 5-3
BlndXfr A-6
Cancel 5-5
Clear A-7
SIPPort 4-3, 4-7, 8-11
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
IN-12
OL-4277-01
Index
Default A-7
stutter tone A-8
DND A-7
subnet mask
Drop A-7
specifying manually 3-10
Edit 5-6, 5-14
used by phone 5-7, 8-5
Exit 5-3
Subnet Mask parameter 3-10, 5-7
Factory A-7
subscription features 4-5
Missed A-7
supplemental services, list of supported 1-5
No 5-6, 5-14
Restore A-7
Save 5-5
T
Validate 5-5
TAC, contacting 9-5
Yes 5-6, 5-14
tag, length, and value triplets 6-2
software
downloading from CCO 6-4
technical assistance 9-5
TFTP
overview 1-4
configuration, about 6-2
protocols, list of supported 1-4
disabling 5-12
software specifications D-3
disabling alternate server 5-12
specifications
enabling 5-12
electrical D-2
enabling alternate server 5-12
environmental D-3
server configuration
physical D-2
overview 3-1
physical interface D-3
TFTP Enabled parameter 5-12
software D-3
TFTP server
startup process, of phone 2-9
configuration overview 3-1
StaticIP parameter 3-10, 8-5
disabling downloading of profile from 5-12,
StaticNetMask parameter 3-10, 8-5
StaticRoute parameter 3-10, 4-7, 8-5
Status menu 5-2, 5-18
Status Messages screen 5-18
8-3
enabling downloading of profile from 5-12,
8-3
IP address 5-8, 6-6, 8-3
name 6-6
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
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IN-13
Index
URL 5-8
RingOnOffTime 8-23
TFTP Server parameter 5-8
SigTimer 8-22
TftpURL parameter 8-3
ToS (Type of Service) 8-8
time, display 9-3
TOS parameter 4-5, 8-8
time format, on LCD screen 8-17, A-4
TraceFlags parameter 4-5, 8-9
Time Format parameter 8-17
Trnsfer softkey 8-14
TimeFormat parameter 8-17
troubleshooting
time zone, of time on phone 5-9, 8-6
contacting TAC 9-5
Time Zone parameter 5-9
general tips 9-1
TimeZone parameter 4-8, 8-6
resolving problems 9-2
tone
TxCodec parameter 8-25
busy tone 8-23, 8-28
cadences in 8-28
call waiting tone 8-24, 8-28
U
configuration 8-28
UID parameter 8-10
dial tone 8-23, 8-28, 9-4
UIPassword parameter 8-26
fast busy tone 9-2, 9-3
unlocking configuration parameters 5-4
frequencies in 8-28
upgradecode parameter 3-7, 8-26, 8-41
reorder 8-24, 8-30
upgradelogo parameter 8-27, 8-45
ring back tone 8-24, 8-28
Use Login ID parameter 5-15
secondary dial tone 8-23, 8-28
UseLoginID parameter 4-4, 8-10
tone parameters
user-agent client (UAC) 1-8
BusyTone 8-23
user-agent server (UAS) 1-8
CallWaitTone 8-24
user ID 5-15, 8-10
DialPlan 4-5
User ID parameter 5-15
DialTone 8-23
UseTftp parameter 8-3
DialTone2 8-23
ReorderTone 8-24
RingBackTone 8-24
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
IN-14
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Index
phone number for 8-15, A-5
V
Volume button 1-3
Validate softkey 5-5
VLAN
W
auxiliary 5-10
disabling 5-10
wall mounting, Cisco IP Phone 2-12
enabling 5-10
WAN, address of attached router/NAT 4-3
IP encapsulation, disabling 8-7
warnings
IP encapsulation, enabling 8-7
related parameters 3-4
connecting SELV circuits to TNV
circuits 2-8
tagging 3-3
connecting to power source 2-3
use of 8-8
disposal 2-2
installation by service personnel only 2-3
VLAN CoS bit value
specifying for TCP packets 3-5
lightning 2-3
specifying for UDP packets 3-5
plug-socket combination 2-3
VLAN Enabled parameter 5-10
short-circuit (overcurrent) protection 2-3
VLAN ID
TN power systems 2-3
web page
enabling user-specified voice 3-4
example 3-5
accessing for a phone 7-2
specifying a preconfigured 3-4
Audio Parameters 8-24
Call Preferences 8-13
VLAN IP encapsulation
configuration options 7-3
disabling 3-4
VLAN IP encapsulation, disabling 3-4
configuring 7-2
VLANSetting parameter 3-5, 8-8
Device Information 7-4
Voice Mail Number parameter 8-15
disabling access to 4-5
VoiceMailNumber parameter 8-15
for Cisco IP Phone 7905G 7-1, 7-2
voice messages 8-15, A-5
for Cisco IP Phone 7912G 7-1, 7-2
voice message system
Network Parameters 8-2
delay before call is forwarded to 8-18, A-6
password 7-4
SIP Parameters 8-9
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
OL-4277-01
IN-15
Index
X
XMLDefault.cnf.xml profile 3-4
Y
Yes softkey 5-6, 5-14
Cisco IP Phone Models 7905G and 7912G Administrator Guide (SIP)
IN-16
OL-4277-01
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