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Cisco 7960 Administrator's Manual
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Cisco 7960 Administrator's Manual

Cisco 7960 Administrator's Manual

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Cisco SIP IP Phone 7960
Administrator Guide
Version 2.0
Corporate Headquarters
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
USA
http://www.cisco.com
Tel:
408 526-4000
800 553-NETS (6387)
Fax: 408 526-4100
Customer Order Number: DOC-7810497=
Text Part Number: 78-10497-02

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Summary of Contents for Cisco 7960

  • Page 1 Cisco SIP IP Phone 7960 Administrator Guide Version 2.0 Corporate Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 http://www.cisco.com Tel: 408 526-4000 800 553-NETS (6387) Fax: 408 526-4100 Customer Order Number: DOC-7810497= Text Part Number: 78-10497-02...
  • Page 2 You can determine whether your equipment is causing interference by turning it off. If the interference stops, it was probably caused by the Cisco equipment or one of its peripheral devices. If the equipment causes interference to radio or television reception, try to correct the interference by using one or more of the following measures: •...
  • Page 3 Obtaining Technical Assistance Cisco Connection Online Technical Assistance Center Documentation Feedback xvii Product Overview C H A P T E R What is Session Initiation Protocol? Components of SIP SIP Clients SIP Servers Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 4: Table Of Contents

    1-13 Connecting to Power 1-14 Using a Headset 1-15 The Cisco SIP IP Phone with a Catalyst Switch 1-16 Getting Started with Your Cisco SIP IP Phone C H A P T E R Initialization Process Overview Installing the Cisco SIP IP Phone...
  • Page 5 Contents Using the Cisco SIP IP Phone Menu Interface 2-21 Reading the Cisco SIP IP Phone Icons 2-22 Customizing the Cisco SIP IP Phone Ring Types 2-24 Creating Dial Plans 2-24 Managing Cisco SIP IP Phones C H A P T E R...
  • Page 6 Transfer B-11 Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold B-16 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Hold with Consultation B-20 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting B-25 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer without...
  • Page 7 Contents Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Busy) B-44 Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (No Answer) B-48 Cisco SIP IP Phone-to Cisco SIP IP Phone 3-Way Calling B-52 Call Flow Scenarios for Failed Calls B-58 Gateway-to-Cisco SIP IP Phone—Called User is Busy...
  • Page 8 Contents G L O S S A R Y I N D E X Cisco SIP IP Phone 7960 Administrator Guide viii 78-10497-02...

  • Page 9: About This Guide

    Who Should Use This Guide Network engineers, system administrators, or telecommunication engineers should use this guide to learn the steps required to properly set up the Cisco SIP IP phone on the network. The tasks described are considered to be administration-level tasks and are not intended for end-users of the phones.
  • Page 10 Objectives The Cisco SIP IP Phone 7960 Administrator Guide provides necessary information to get the Cisco SIP IP phone operational in a Voice-over-IP (VoIP) network. It is not the intent of this administrator guide to provide information on how to implement a SIP VoIP network.

  • Page 11: Related Documentation

    About This Guide Related Documentation Related Documentation The following is a list of related Cisco SIP VoIP publications. For more information about implementing a SIP VoIP network refer to the following publications: • Session Initiation Protocol Gateway Call Flows Session Initiation for VoIP on Cisco Access Platforms •...
  • Page 12 (Voor vertalingen van de waarschuwingen die in deze publicatie verschijnen, kunt u het aanhangsel “Translated Safety Warnings” (Vertalingen van veiligheidsvoorschriften) raadplegen.) Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 13 La traduzione delle avvertenze riportate in questa pubblicazione si trova nell’appendice, “Translated Safety Warnings” (Traduzione delle avvertenze di sicurezza). Cisco SIP IP Phone 7960 Administrator Guide xiii 78-10497-02...
  • Page 14 (Se förklaringar av de varningar som förekommer i denna publikation i appendix "Translated Safety Warnings" [Översatta säkerhetsvarningar].) Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 15: Obtaining Documentation

    About This Guide Obtaining Documentation Obtaining Documentation World Wide Web You can access the most current Cisco documentation on the World Wide Web at http://www.cisco.com, http://www-china.cisco.com, or http://www-europe.cisco.com. Documentation CD-ROM Cisco documentation and additional literature are available in a CD-ROM package, which ships with your product.
  • Page 16 Obtaining Technical Assistance Cisco Connection Online Cisco continues to revolutionize how business is done on the Internet. Cisco Connection Online is the foundation of a suite of interactive, networked services that provides immediate, open access to Cisco information and resources at anytime, from anywhere in the world.

  • Page 17: Documentation Feedback

    Documentation Feedback If you are reading Cisco product documentation on the World Wide Web, you can submit technical comments electronically. Click Feedback in the toolbar and select Documentation. After you complete the form, click Submit to send it to Cisco.
  • Page 18 About This Guide Obtaining Technical Assistance Cisco SIP IP Phone 7960 Administrator Guide xviii 78-10497-02...

  • Page 19: Product Overview

    C H A P T E R Product Overview This chapter contains the following information about the Cisco SIP IP phone: What is Session Initiation Protocol?, page 1-1 • What is the Cisco SIP IP Phone 7960?, page 1-5 •...
  • Page 20 The term conference means an established session (or call) between Note two or more end points. In this document, the terms conference and call are used interchangeably. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 21: Components Of Sip

    Lightweght Directory Access Protocol (LDAP) servers, a database application, or an extensible markup language (XML) application. These application services provide back-end services such as directory, authentication, and billing services. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 22: Sip Clients

    SIP clients include: Phones—Can act as either a UAS or UAC. Softphones (PCs that have phone • capabilities installed) and Cisco SIP IP phones can initiate SIP requests and respond to requests. Gateways—Provide call control. Gateways provide many services, the most •...

  • Page 23: What Is The Cisco Sip Ip Phone 7960

    (PBX) telephone. The Cisco SIP IP phone is an IP telephony instrument that can be used in VoIP networks. The Cisco SIP IP phone model terminals can attach to the existing in place data network infrastructure, via 10BaseT/100BaseT interfaces on an Ethernet switch.
  • Page 24 • LCD screen—Desktop which displays information about your Cisco SIP IP phone, such as the time, date, your phone number, caller ID, line/call status and the soft key tabs. Line or speed dial buttons—Opens a new line or speed dials the number on •...

  • Page 25: Supported Features

    Chapter 1 Product Overview What is the Cisco SIP IP Phone 7960? • On-screen mode buttons—Retrieves information about current settings, recent calls, available services, and voice mail messages. Volume buttons—Adjusts the volume of the handset, headset, speaker, ringer • and adjusts the brightness contrast settings on the LCD screen.
  • Page 26 Chapter 1 Product Overview What is the Cisco SIP IP Phone 7960? • Ability to: Configure Ethernet port mode and speed – – Register with or unregister from a proxy server – Specify a TFTP boot directory Configure a label for phone identification display purposes –...
  • Page 27 SIP proxy server. Call hold—Allows the Cisco SIP IP phone user (user A) to place a call – (from user B) on hold. When user A places user B on hold, the 2-way RTP voice path between user A and user B is temporarily disconnected but the call session is still connected.

  • Page 28: Supported Protocols

    DHCP is used to dynamically allocate and assign IP addresses. DHCP allows you to move network devices from one subnet to another without administrative attention. If using DHCP, you can connect Cisco SIP IP phones to the network and become operational without having to manually assign an IP address and additional network parameters.
  • Page 29 IP is a network layer protocol that sends datagram packets between nodes on the Internet. IP also provides features for addressing, type-of-service (ToS) specification, fragmentation and reassembly, and security. The Cisco SIP IP phone supports IP as it is defined in RFC 791. Real-Time Transport Protocol (RTP) •...

  • Page 30: Prerequisites

    SIP can use UDP as the underlying transport protocol. If UDP is used, retransmissions are used to ensure reliability. The Cisco SIP IP phone supports UDP as it is defined in RFC 768 for SIP signaling. Prerequisites...

  • Page 31: Cisco Sip Ip Phone Connections

    (10/100 PC) Connecting to the Network The Cisco SIP IP phone has two RJ-45 ports that each support 10/100 Mbps half- or full-duplex Ethernet connections to external devices—network port (labeled 10/100 SW) and access port (labeled 10/100 PC). You can use either Category 3 or 5 cabling for 10 Mpbs connections, but use Category 5 for 100 Mbps connections.

  • Page 32: Connecting To Power

    • WS-X6348-RJ45V 10/100 switching module—Provides inline power to the Cisco SIP IP phone when connected to a Catalyst 3500, 4000, or 6000 family 10/100BaseTX switching module. This module sends power on pins 1 & 2 and 3 & 6.

  • Page 33: Using A Headset

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

  • Page 34: The Cisco Sip Ip Phone With A Catalyst Switch

    Catalyst switch, to obtain network connectivity. The Cisco SIP IP phone has an internal Ethernet switch, which enables it to switch traffic coming from the phone, access port, and the network port.

  • Page 35: Getting Started With Your Cisco Sip Ip Phone

    Getting Started with Your Cisco SIP IP Phone This chapter explains the Cisco SIP IP phone initialization and the process that you should follow to install and connect the Cisco SIP IP phone. This chapter provides the following major sections: •...
  • Page 36 (if using DHCP). An IP address is acquired. If the Cisco SIP IP phone is using DHCP to obtain the IP settings, the phone queries the DHCP server. If the phone is not using DHCP, then the phone will use IP settings that are stored in Flash memory.

  • Page 37: Installing The Cisco Sip Ip Phone

    Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Installing the Cisco SIP IP Phone This section contains information on how to install Cisco SIP IP phones in your IP network. Before getting started, read over the information in this section carefully.

  • Page 38: Downloading Files To Your Tftp Server

    Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Downloading Files to Your TFTP Server Before installing the Cisco SIP IP phones, copy the following files from CCO to the root directory of your TFTP server. File Description OS79XX.TXT...

  • Page 39: Configuring Sip Parameters

    For a complete list of the SIP parameters that you can configure, see the “Modifying the Phone’s SIP Settings” section on page 3-5. The SIP parameters are those parameters that a Cisco SIP IP phone needs to operate in a SIP VoIP environment. You can configure SIP parameters via a TFTP server or you can manually configure the parameters on a phone-by-phone basis after connecting the phones.

  • Page 40: Configuring Sip Parameters Via A Tftp Server

    Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Configuring SIP Parameters via a TFTP Server If you are configuring SIP parameters via a TFTP server, you must use configuration files. There are two configuration files that you can use to define the SIP parameters;...
  • Page 41 Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone • The default configuration file must be stored in the root directory of the TFTP server. The phone-specific configuration file can be stored in the root directory or in a subdirectory in which all phone-specific configuration files are located.
  • Page 42 Firmware version that the Cisco SIP IP phone should run. Enter the name of the image version (as it is released by Cisco). Do not enter the extension. You cannot change the image version by changing the file name because the version is also built into the file header.
  • Page 43 Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone The following is an example of a SIP default configuration file: ; sip default configuration file #Image Version image_version:P0S3 xxyy ; #Proxy server address proxy1_address: 192.168.1.1 ;...
  • Page 44 Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Procedure Using an ASCII editor, create a phone-specific configuration file for each phone Step 1 that you plan to install. In the phone-specific configuration file, define values for the following SIP parameters (where x is a number 1 through 6): linex_name—(Required) Number or e-mail address used when registering.

  • Page 45: Manually Configuring The Sip Parameters

    • Review the guidelines on using the Cisco SIP IP phone menus documented in the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-21. •...
  • Page 46 Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Step 5 Highlight and press the Select soft key to configure the following parameters: Name—(Required) Number or e-mail address used when registering. When •...

  • Page 47: Configuring Network Parameters

    Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Configuring Network Parameters Note This section describes how to configure the basic network parameters that are required for the phone to operate on the network.

  • Page 48: Configuring Network Parameters Via A Dhcp Server

    Configuring Network Parameters via a DHCP Server If you are using DHCP to configure the network parameters, configure the following DHCP options on your DHCP server before you connect your Cisco SIP IP phone: •...
  • Page 49 Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone • When configuring a domain name: Press the Number soft key if entering a numerical ID or press the Alpha – soft key to enter a name.

  • Page 50: Connecting The Phone

    Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone • Domain Name—Name of the DNS domain in which the phone resides. DNS Servers 1 through 5—IP address of the DNS server used by the phone •...
  • Page 51 Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Figure 2-1 Cisco SIP IP Phone Cable Connections Cisco IP Phone 7960 (rear view) Power outlet AC adapter port (DC48V) Headset port (optional power...

  • Page 52: Adjusting The Placement Of The Cisco Sip Phone

    Adjust the footstand to its desired height and release the knob. Mounting the Phone to the Wall You can mount the Cisco SIP IP phone on the wall using the footstand as a mounting bracket, or using the optional locking bracket. Use the following procedure to mount the phone on the wall using the standard footstand.
  • Page 53 Before You Begin Mounting the Cisco SIP IP phone on the wall requires some tools and • equipment that are not provided as standard equipment. Following are the tools and parts required for a typical Cisco SIP IP phone installation: – Screwdriver Screws to secure the Cisco SIP IP phone to the wall –...

  • Page 54: Verifying Startup

    After the phone has power connected to it, the phone begins its startup process by cycling through these steps: These buttons flash on and off in sequence: Headset – Mute – – Speaker The Cisco Systems, Inc. copyright displays on the LCD. Cisco SIP IP Phone 7960 Administrator Guide 2-20 78-10497-02...

  • Page 55: Using The Cisco Sip Ip Phone Menu Interface

    Chapter 2 Getting Started with Your Cisco SIP IP Phone Using the Cisco SIP IP Phone Menu Interface These messages display as phone starts up: Configuring VLAN—The phone is configuring the Ethernet connection. – – Configuring IP—The phone is contacting the DHCP server to obtain network parameters and the IP address of the TFTP server.

  • Page 56: Reading The Cisco Sip Ip Phone Icons

    Edit panel. Reading the Cisco SIP IP Phone Icons When using the Cisco SIP IP phone, a variety of icons can display on the phone’s LCD. Table 1 lists and describes each icon that you might see while using the Cisco SIP IP phone.
  • Page 57 The Cisco SIP IP phone configuration mode is locked. When the phone is locked, the phone’s network or SIP settings cannot be modified. The Cisco SIP IP phone configure mode is unlocked. When the phone is unlocked, the phone’s network or SIP settings can be modified.

  • Page 58: Customizing The Cisco Sip Ip Phone Ring Types

    Customizing the Cisco SIP IP Phone Ring Types Customizing the Cisco SIP IP Phone Ring Types The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By default, your ring type options will be those two choices. However, using the RINGLIST.DAT file, you can customize the ring types that are available to the...
  • Page 59 Chapter 2 Getting Started with Your Cisco SIP IP Phone Creating Dial Plans We recommend that you define the dial_template parameter in the Note default configuration file for maintenance and control purposes. Specify the dial_template parameter in a phone-specific configuration file only if that phone needs to use a different dial plan than is being used by the other phones in the same system.
  • Page 60 Chapter 2 Getting Started with Your Cisco SIP IP Phone Creating Dial Plans • User=”type” is the either IP or Phone. Enter User=phone or User=IP to have the tag automatically added to the dialed number. Rewrite=”altstrng” is the alternate string to be dialed instead of what the user •...

  • Page 61: Managing Cisco Sip Ip Phones

    • Entering Configuration Mode When you access the network configuration information on your Cisco SIP IP phone, you will notice that there is a padlock symbol located in the upper right corner of your LCD. By default, the network configuration information is locked.

  • Page 62: Unlocking Configuration Mode

    LCD will change to an unlocked state. If you are located elsewhere in the Cisco SIP IP phone menus, the next time you access the Network Configuration or the SIP Configuration panels, the lock icon will be displayed in an unlocked state.
  • Page 63 • Review the guidelines on using the Cisco SIP IP phone menus documented in the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-21. After making your changes, relock configuration mode as described in the •...
  • Page 64 Admin. VLAN Id—Unique identifier of the VLAN to which the phone is • attached. The value in this field is only used in non-Cisco switched networks. You can change the administrative VLAN used by the phone, however, if you have an administrative VLAN assigned on the Catalyst switch, that setting overrides any changes made on the phone.

  • Page 65: Modifying The Phone's Sip Settings

    “Locking Configuration Mode” section on page 3-2. Modifying the Phone’s SIP Settings You can modify the SIP parameters of a Cisco SIP IP phone. When modifying SIP parameters, remember the following: • Parameters defined in the default configuration file will override the values stored in Flash memory.
  • Page 66 Menu anonymous_call_block Anonymous Call Block autocomplete Auto-Complete Numbers callerid_blocking Caller ID Block dial_template dnd_control dst_auto_adjust dst_offset dst_start_day dst_start_day_of_week dst_start_month dst_start_time dst_start_week_of_mon dst_stop_day dst_stop_day_of_week dst_stop_month dst_stop_time dst_stop_week_of_mon dtmf_avt_payload dtmf_db_level dtmf_inband Do Not Disturb Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 67 Messages URI network_media_type Network Media Type phone_label Phone Label preferred_codec Preferred Codec proxy_register Register with Proxy proxy1_address Proxy Address proxy1_port Proxy Port sip_invite_retx sip_retx sntp_mode sntp_server sync tftp_cfg_dir TFTP Directory time_format_24hr time_zone timer_invite_expires Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 68: Modifying Sip Parameters Via A Tftp Server

    SIP parameters that are common to all of your phones. By maintaining these parameters in the default configuration file, you can perform global changes, such as upgrading the image version, without having to modify the phone-specific configuration file for each phone. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 69 Firmware version that the Cisco SIP IP phone should run. Enter the name of the image version (as it is release by Cisco). Do not enter the extension. You cannot change the image version by changing the file name because the version is also built into the file header.
  • Page 70 The default is 3. dtmf_outofband—(Optional) Whether to generate the out-of-band signaling • (for tone detection on the IP side of a gateway) and if so, when. The Cisco SIP IP phone supports out-of-bound signaling via the AVT tone method. Valid values are: none—Do not generate DTMF digits out-of-band.
  • Page 71 Mode in which the phone will listen for the – SNTP server. – sntp_server—(Optional) IP address of the SNTP server from which the phone will obtain time data. time_zone—(Optional) Time zone in which the phone is located. – Cisco SIP IP Phone 7960 Administrator Guide 3-11 78-10497-02...
  • Page 72 – off locally via the phone’s user interface. – 1—The Do Not Disturb feature is on by default, but can be turned on and off locally via the phone’s user interface. Cisco SIP IP Phone 7960 Administrator Guide 3-12 78-10497-02...
  • Page 73 1—The Anonymous Call Blocking features is enabled by default, but can – be turned on and off via the phone’s user interface. When enabled, anonymous calls will be rejected Cisco SIP IP Phone 7960 Administrator Guide 3-13 78-10497-02...
  • Page 74 1—The 24-hour format is displayed by default but can be changed to a – 12-hour format via the phone’s user interface. – 3—The 12-hour format is displayed and cannot be changed to a 24-hour format via the phone’s user interface. Cisco SIP IP Phone 7960 Administrator Guide 3-14 78-10497-02...

  • Page 75: Modifying The Phone-Specific Sip Configuration File

    Similarly, if you configure a line to use a number, that line can only be called using the number. Each line can have a different proxy configured. Cisco SIP IP Phone 7960 Administrator Guide 3-15 78-10497-02...
  • Page 76 If a value is not configured for the linex_password parameter for a line when registration is enabled, the value defined for line 1 is used. If a value is not defined for line 1, the default line1_password is UNPROVISIONED. Cisco SIP IP Phone 7960 Administrator Guide 3-16 78-10497-02...
  • Page 77 LCD. This field is • for end-user display only purposes. For example, a phone’s label can display “John Doe’s phone.” Approximately up to 11 characters can be used when specifying the phone label. Cisco SIP IP Phone 7960 Administrator Guide 3-17 78-10497-02...

  • Page 78: Modifying The Sip Parameters Manually

    Mode” section on page 3-2. By default, the SIP parameters are locked to ensure that end-users cannot modify settings that might affect their call capabilities. Review the guidelines on using the Cisco SIP IP phone menus documented in • the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-21.
  • Page 79 Authentication Password—(Required when registration is enabled) Password used by the phone for authentication if a registration is challenged by the proxy server during initialization. If a value is not configured for the Cisco SIP IP Phone 7960 Administrator Guide 3-19 78-10497-02...
  • Page 80 Out of Band DTMF—(Optional) Whether to detect and generate the out-of-band signaling (for tone detection on the IP side of a gateway) and if so, when. The Cisco SIP IP phone supports out-of-bound signaling via the AVT tone method. Valid values are: –...
  • Page 81 When done, press the Save soft key to save your changes and exit the Step 9 SIP Configuration menu. When you have completed your changes, ensure that you lock the Caution phone as described in the “Locking Configuration Mode” section on page 3-2. Cisco SIP IP Phone 7960 Administrator Guide 3-21 78-10497-02...

  • Page 82: Setting The Date, Time, And Daylight Savings Time

    Setting the Date, Time, and Daylight Savings Time Setting the Date, Time, and Daylight Savings Time The current date and time is supported on the Cisco SIP IP phone via SNTP and is displayed on the phone’s LCD. In addition to supporting the current date and time, daylight savings time (DST) and time zone settings are also supported.
  • Page 83 Cisco SIP IP Phone 7960 Administrator Guide 3-23 78-10497-02...
  • Page 84 SNTP from the local from other SNTP and the local servers. network broadcast servers. network broadcast address. address and ignores responses from other SNTP servers. Cisco SIP IP Phone 7960 Administrator Guide 3-24 78-10497-02...
  • Page 85 November, and December or 1 through 12 with January being 1 and December being 12. When specifying the name of a month, the value is case-sensitive and should be typed as cited in this description. Cisco SIP IP Phone 7960 Administrator Guide 3-25 78-10497-02...
  • Page 86 Valid values are 1 through 6 and 8 with 1 being the first week and each number thereafter being subsequent weeks and 8 specifying the last week in the month regardless of which week the last week is. Cisco SIP IP Phone 7960 Administrator Guide 3-26 78-10497-02...
  • Page 87 : 03/00 dst_offset : 01/00 dst_start_month : April dst_start_day : 1 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 1 dst_stop_time : 02/00 dst_stop_autoadjust : 1 (additional configuration text omitted) Cisco SIP IP Phone 7960 Administrator Guide 3-27 78-10497-02...

  • Page 88: Erasing The Locally-Defined Settings

    Select the Erase Config parameter by pressing the down arrow to scroll to and highlight the parameter or by pressing the number that represents the parameter (located to the left of the parameter name on the LCD). Cisco SIP IP Phone 7960 Administrator Guide 3-28 78-10497-02...

  • Page 89: Erasing The Locally-Defined Sip Settings

    3-2. Procedure Step 1 Press the settings key. The Settings menu is displayed. Step 2 Highlight SIP Configuration. Press the Select soft key. The SIP Configuration settings are displayed. Step 3 Cisco SIP IP Phone 7960 Administrator Guide 3-29 78-10497-02...

  • Page 90: Accessing Status Information

    Firmware Version—Displays information about the current firmware version • on the phone. In addition to the status messages available via the Setting Status menu, you can also obtain status messages for a current call. Cisco SIP IP Phone 7960 Administrator Guide 3-30 78-10497-02...

  • Page 91: Viewing Status Messages

    Xmit—Number of packets sent by the phone; not through the switch. REr—Number of packets received by the phone that contained errors. • • BCast—Number of broadcast packets received by the phone. Cisco SIP IP Phone 7960 Administrator Guide 3-31 78-10497-02...
  • Page 92 10Mbps connection. Step 6 To exit the Network Statistics panel, press the Exit soft key. Note To reset the values displayed on Network Statistics panel, power off and power on the phone. Cisco SIP IP Phone 7960 Administrator Guide 3-32 78-10497-02...

  • Page 93: Viewing The Firmware Version

    Step 6 Upgrading the Cisco SIP IP Phone Firmware There two methods that you can use to upgrade the firmware on your Cisco SIP IP phones. You can upgrade the firmware on one phone at a time via the phone-specific configuration or you can upgrade the firmware on a system of phones using the default configuration file.
  • Page 94 Chapter 3 Managing Cisco SIP IP Phones Upgrading the Cisco SIP IP Phone Firmware Procedure Copy the binary file P0S3xxyy.bin (where xx is the version number and yy is the Step 1 subversion number) from CCO to the root directory of the TFTP server.

  • Page 95: Performing An Image Upgrade And Remote Reboot

    Performing an Image Upgrade and Remote Reboot Performing an Image Upgrade and Remote Reboot With Version 2.0 of the Cisco SIP IP Phone 7960, you can perform an image upgrade and remote reboot using Notify messages and the synchinfo.xml file.
  • Page 96 The phone the performs a normal reboot process as described in “Initialization Process Overview” section on page 2-1, sees the new image, and upgrades to the new image with a sync value of 2. Cisco SIP IP Phone 7960 Administrator Guide 3-36 78-10497-02...

  • Page 97: Appendix

    A P P E N D I X SIP Compliance with RFC-2543 Information This section describes how the Cisco SIP IP phone complies with the IETF definition of SIP as described in RFC 2543. This section contains compliance information on the following: •...

  • Page 98: Sip Functions

    INVITE that contains an existing Call-ID. OPTIONS None. CANCEL REGISTER The Cisco SIP IP phone supports both user and device registration. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 99: Sip Responses

    Appendix A SIP Compliance with RFC-2543 Information SIP Responses SIP Responses Release 1.0 of the Cisco SIP IP phone supports the following SIP responses: 1xx Response—Information Responses, page A-4 • • 2xx Response—Successful Responses, page A-4 3xx Response—Redirection Responses, page A-5 •...

  • Page 100: 1Xx Response-Information Responses

    1xx Response—Information Responses 1xx Response Supported? Comments 100 Trying The Cisco SIP IP phone generates this response for an incoming INVITE. Upon receiving this response, the phone waits for a 180 Ringing, 183 Session progress, or 200 OK response. 180 Ringing...

  • Page 101: 3Xx Response-Redirection Responses

    300 Multiple Choices None 301 Moved Permanently 302 Moved The Cisco SIP IP phone does not Temporarily generate this response at this time. Upon receiving this response, the phone sends an INVITE containing the contact information received in the 302 Moved temporarily response.
  • Page 102 This response indicates that the SIP server has the request but will not provide service. 404 Not Found The Cisco SIP IP phone generates this response if it is unable to locate the callee. Upon receiving this response, the phone notifies the user.
  • Page 103 This response is only received by the phone in this release. The 410 Gone response indicates that a resource is no longer available at the server and no forwarding address is known. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 104 The user is notified if this response is received. If the phone does not understand the protocol extension specified in the Require field, the 420 Bad Extension response is generated. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 105 If a new contact is received, the phone might re-initiate the call. 486 Busy Here The Cisco SIP IP phone generates this response if the called party is off hook and the call cannot be presented as a call waiting call. Upon receiving this response, the phone notifies the user and generates a busy tone.

  • Page 106: 5Xx Response-Server Failure Responses

    5xx Response—Server Failure Responses 5xx Response Comments 500 Internal Server Error 501 Not Implemented For an incoming response, the Cisco SIP IP phone sends a new request if an additional 502 Bad Gateway contact address is present. If an additional 503 Service Unavailable contact address is not present, the gateway initiates a graceful call disconnect.
  • Page 107 SIP Compliance with RFC-2543 Information SIP Header Fields Header Field Supported? Authorization Call-ID Contact Content-Encoding Content-Length Content-Type Cseq Date Encryption Expires From Hide Max-Forwards Organization Priority Proxy-Authenticate Proxy-Authorization Proxy-Require ReBy Record-Route Require Response-Key Retry-After Route Cisco SIP IP Phone 7960 Administrator Guide A-11 78-10497-02...

  • Page 108: Sip Session Description Protocol (Sdp) Usage

    Server Subject Timestamp Unsupported User-Agent Warning WWW-Authenticate SIP Session Description Protocol (SDP) Usage SDP Headers Supported? v—Protocol version o—Owner/creator and session identifier a—Session name c—Connection information m—Media name and transport address Cisco SIP IP Phone 7960 Administrator Guide A-12 78-10497-02...

  • Page 109: Appendix

    OPTIONS—Queries the capabilities of servers. • REGISTER—Registers the address listed in the To header field with a SIP server. The following types of responses are used by SIP and generated by the Cisco SIP gateway: SIP 1xx—Informational Responses • SIP 2xx—Successful Responses •...

  • Page 110: Call Flow Scenarios For Successful Calls

    • Gateway to-Cisco SIP IP Phone—Successful Call Setup and Call Transfer, page B-11 • Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold, page B-16 • Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Hold with Consultation, page B-20 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting, page B-25 •...

  • Page 111: Gateway-To Cisco Sip Ip Phone-Successful Call Setup And Disconnect

    PBX A. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. User B is located at a Cisco SIP IP phone. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.
  • Page 112 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-1 Gateway-to-Cisco SIP IP Phone—Successful Setup and Disconnect SIP IP Phone User B IP Network PBX A User A 1. Setup 2. INVITE 3. Call Proceeding 4. 100 Trying 5.
  • Page 113 INVITE request to the address it receives as the dial peer which, in this scenario, is the Cisco SIP IP phone. In the INVITE request: The IP address of the Cisco SIP IP phone is inserted in • the Request-URI field.
  • Page 114 PBX A acknowledges Gateway 1’s Connect message. Gateway 1 ACK—Gateway 1 to Cisco SIP Gateway 1 sends a SIP ACK to the Cisco SIP IP phone. The IP phone ACK confirms that Gateway 1 has received the 200 OK response. The call session is now active.

  • Page 115: Gateway-To-Cisco Sip Ip Phone-Successful Call Setup And Call Hold

    PBX A. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. User B is located at a Cisco SIP IP phone. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.
  • Page 116 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-2 Gateway-to-Cisco SIP IP Phone Call—Successful Call Setup and Call Hold SIP IP Phone PBX A IP Network User B User A 1. Setup 2. INVITE 3. Call Proceeding 4.
  • Page 117 INVITE request to the address it receives as the dial peer which, in this scenario, is the Cisco SIP IP phone. In the INVITE request: The IP address of the Cisco SIP IP phone is inserted in • the Request-URI field.
  • Page 118 Connect message notifies PBX A that the connection has been made. ACK—Gateway 1 to Cisco SIP Gateway 1 sends a SIP ACK to the Cisco SIP IP phone. The IP phone ACK confirms that User A has received the 200 OK response.

  • Page 119: Gateway To-Cisco Sip Ip Phone-Successful Call Setup And Call Transfer

    User A, User B, and User C. User A is located at PBX A. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. User B is located at a Cisco SIP IP phone and is directly connected to the IP network. User C is located at PBX B. PBX B is connected to Gateway 2 (SIP Gateway) via a T1/E1.
  • Page 120 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-3 Gateway-to-Cisco SIP IP Phone Call—Successful Call Setup and Call Transfer SIP IP Phone User B PBX A PBX B IP Network User A User C 1. Setup 2.
  • Page 121 INVITE request to the address it receives as the dial peer which, in this scenario, is the Cisco SIP IP phone. In the INVITE request: The IP address of the Cisco SIP IP phone is inserted in • the Request-URI field.
  • Page 122 Connect message notifies PBX A that the connection has been made. ACK—Gateway 1 to Cisco SIP Gateway 1 sends a SIP ACK to the Cisco SIP IP phone. The IP phone ACK confirms that Gateway 1 has received the 200 OK response.
  • Page 123 PBX B ACK—Gateway 1 to Gateway 2 Gateway 1 sends a SIP ACK to Gateway 2. The ACK confirms that Gateway 1 has received the 200 OK message from Gateway 2. Cisco SIP IP Phone 7960 Administrator Guide B-15 78-10497-02...

  • Page 124: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Simple Call Hold

    Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold Figure B-4 illustrates a successful call between Cisco SIP IP phones in which one of the participants places the other on hold and then returns to the call. In this call flow scenario, the two end users are User A and User B.
  • Page 125 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-4 Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold SIP IP SIP IP IP Network Phone User A Phone User B 1. INVITE B 2. 180 RINGING 3.
  • Page 126 Action Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session.
  • Page 127 ACK is empty, Cisco SIP IP phone B uses the session description in the INVITE request. A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B. INVITE—Cisco SIP IP phone B...

  • Page 128: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Call Hold With Consultation

    (consultation), and then returns to the original call. In this call flow scenario, the end users are User A, User B, and User C. They are all using Cisco SIP IP phones, which are connected via an IP network.
  • Page 129 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-5 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Hold with Consultation SIP IP SIP IP SIP IP Phone Phone User B IP Network Phone User A User C 1.
  • Page 130 Action Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session.
  • Page 131 ACK is empty, Cisco SIP IP phone B uses the session description in the INVITE request. A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B. INVITE—Cisco SIP IP phone B...
  • Page 132 ACK is empty, Cisco SIP IP phone C uses the session description in the INVITE request. A two-way RTP channel is established between Cisco SIP IP phone B and Cisco SIP IP phone C. BYE—Cisco SIP IP phone B to The call continues and then User B hangs up.

  • Page 133: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Call Waiting

    A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A. A two-way RTP channel is reestablished between Cisco SIP IP phone A and Cisco SIP IP phone B. Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting...
  • Page 134 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-6 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting SIP IP SIP IP SIP IP Phone Phone User B Phone User A IP Network User C 1. INVITE B 2.
  • Page 135 Action Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session.
  • Page 136 ACK is empty, Cisco SIP IP phone B uses the session description in the INVITE request. A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B. INVITE—Cisco SIP IP phone C...
  • Page 137 A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A. The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down. 200 OK—Cisco SIP IP phone B...
  • Page 138 A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A. A two-way RTP channel is reestablished between Cisco SIP IP phone A and Cisco SIP IP phone B. BYE—Cisco SIP IP phone B to The call continues and then User B hangs up.

  • Page 139: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Call Transfer Without Consultation

    C. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A. A two-way RTP channel is reestablished between Cisco SIP IP phone B and Cisco SIP IP phone C. Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer without...
  • Page 140 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-7 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer without Consultation SIP IP SIP IP SIP IP Phone Phone User B IP Network Phone User A User C 1.
  • Page 141 Action Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session.
  • Page 142 B uses the session description in the INVITE request. A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B. User B then selects the option to transfer the call to User C.

  • Page 143: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Call Transfer With Consultation

    C. The ACK confirms that Cisco SIP IP phone A has received the 200 OK response from Cisco SIP IP phone C. A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone C. Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer with...
  • Page 144 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-8 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer with Consultation SIP IP SIP IP SIP IP Phone Phone User B Phone User A IP Network User C 1.
  • Page 145 Action Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session.
  • Page 146 B uses the session description in the INVITE request. A two-way RTP channel is established between Cisco SIP IP phone A and Cisco SIP IP phone B. User B then selects the option to transfer the call to User C.
  • Page 147 ACK is empty, Cisco SIP IP phone C uses the session description in the INVITE request. A two-way RTP channel is established between Cisco SIP IP phone B and Cisco SIP IP phone C. BYE—Cisco SIP IP phone B to The call continues and then User B hangs up.
  • Page 148 The call session between User A and User B is now terminated. The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down. INVITE—Cisco SIP IP phone A At the request of Cisco SIP IP phone B, Cisco SIP IP phone to Cisco SIP IP phone C A sends a SIP INVITE request to Cisco SIP IP phone C.

  • Page 149: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Network Call Forwarding (Unconditional

    User B has requested unconditional call forwarding from the network. When User A calls User B, the call is immediately transferred to Cisco SIP IP phone C. In this call flow scenario, the end users are User A, User B, and User C.
  • Page 150 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-9 Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Unconditional) IP Network SIP IP SIP IP SIP IP Proxy Redirect Phone Phone Phone Server Server User A...
  • Page 151 Action Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to the to SIP proxy server SIP proxy server. The INVITE request is an invitation to User B to participate in a call session.

  • Page 152: (Busy

    When User A calls User B, the SIP proxy server tries to place the call to Cisco SIP IP phone B and, if the line is busy, the call is transferred to Cisco SIP IP phone C. In this call flow scenario, the end users are User A, User B, and User C.
  • Page 153 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-10 Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Busy) IP Network SIP IP SIP IP SIP IP Proxy Redirect Phone Phone Phone Server Server User A...
  • Page 154 SIP proxy message to the SIP proxy server. The message indicates server that User B can be reached either at SIP phone B or Cisco SIP IP phone C. INVITE—SIP proxy server to SIP proxy server sends a SIP INVITE request to Cisco SIP Cisco SIP IP phone B IP phone B.
  • Page 155 Action Description 486 Busy Here—Cisco SIP IP Cisco SIP IP phone B sends a 486 Busy here message to the phone B to SIP proxy server SIP proxy server. The message indicates that Cisco SIP IP phone B is in use and the user is not willing or able to take additional calls.

  • Page 156: Answer

    When User A calls User B, the proxy server tries to place the call to Cisco SIP IP phone B and, if there is no answer, the call is transferred to Cisco SIP IP phone C. In this call flow scenario, the end users are User A, User B, and User C.
  • Page 157 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-11 Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (No Answer) IP Network SIP IP SIP IP SIP IP Proxy Redirect Phone Phone Phone Server Server...
  • Page 158 SIP proxy message to the SIP proxy server. The message indicates server that User B can be reached either at SIP phone B or Cisco SIP IP phone C. INVITE—SIP proxy server to SIP proxy server sends a SIP INVITE request to Cisco SIP Cisco SIP IP phone B IP phone B.
  • Page 159 B to cancel the invitation. phone B 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to the to SIP proxy server SIP proxy server. The response confirms receipt of the cancellation request.

  • Page 160: Cisco Sip Ip Phone-To Cisco Sip Ip Phone 3-Way Calling

    User B mixes two RTP channels and therefore establishes a conference bridge between User A and User C. In this call flow scenario, the end users are User A, User B, and User C. They are all using Cisco SIP IP phones, which are connected via an IP network.
  • Page 161 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-12 Cisco SIP IP Phone-to Cisco SIP IP Phone 3-Way Calling IP Network SIP IP SIP IP SIP IP Proxy Redirect Phone Phone Phone Server Server User A...
  • Page 162 Action Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session.
  • Page 163 A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A. The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down. User A is put on hold.
  • Page 164 Action Description INVITE—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP INVITE request to Cisco to Cisco SIP IP phone C SIP IP phone C. The INVITE request is an invitation to User B to participate in a call session.
  • Page 165 400 Bad Request response with a 304 Warning header field. ACK—Cisco SIP IP phone B to Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP Cisco SIP IP phone C phone C. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone C.

  • Page 166: Call Flow Scenarios For Failed Calls

    • • Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error, page B-63 • Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User is Busy, page B-66 Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User Does Not Answer, • page B-68 •...
  • Page 167 INVITE request to the address it receives as the dial peer which, in this scenario, is the Cisco SIP IP phone. In the INVITE request: The IP address of the Cisco SIP IP phone is inserted in • the Request-URI field.

  • Page 168: Gateway-To-Cisco Sip Ip Phone-Called User Does Not Answer

    PBX A sends a Release message to Gateway 1. ACK—Gateway 1 to Cisco SIP Gateway 1 sends a SIP ACK to the Cisco SIP IP phone. The IP phone ACK confirms that User A has received the 486 Busy Here response.
  • Page 169 Cisco SIP IP phone. In the INVITE request: • The IP address of the Cisco SIP IP phone is inserted in the Request-URI field. • PBX A is identified as the call session initiator in the From field.
  • Page 170 PBX A and the call session attempt is terminated. 200 OK—Cisco SIP IP phone to The Cisco SIP IP phone sends a SIP 200 OK response to Gateway 1 Gateway 1. The 200 OK response confirms that User A has received the 486 Busy Here response.

  • Page 171: Gateway-To-Cisco Sip Ip Phone-Client, Server, Or Global Error

    Figure B-15 illustrates an unsuccessful call in which User A initiates a call to User B and receives a class 4xx, 5xx, or 6xx response. Figure B-15 Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error SIP IP Phone IP Network...
  • Page 172 Cisco SIP IP phone. In the INVITE request: • The IP address of the Cisco SIP IP phone is inserted in the Request-URI field. • PBX A is identified as the call session initiator in the From field.
  • Page 173 Action Description 4xx/5xx/6xx Failure—Cisco SIP The Cisco SIP IP phone sends a class 4xx, 5xx, or class 6xx IP phone to Gateway 1 failure response to Gateway 1. Depending on which class the failure response is, the call actions differ.

  • Page 174: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone-Called User Is Busy

    Figure B-16 illustrates an unsuccessful call in which User A initiates a call to User B but User B is on the phone and is unable or unwilling to take another call. Figure B-16 Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User is Busy SIP IP...
  • Page 175 ACK—Cisco SIP IP phone A to Cisco SIP IP phone A sends a SIP ACK to the Cisco SIP IP Cisco SIP IP phone B phone B. The ACK confirms that Cisco SIP IP phone A has received the 486 Busy here response from Cisco SIP IP phone B.
  • Page 176 Not Answer Figure B-17 illustrates an unsuccessful call in which User A initiates a call to User B but User B does not answer. Figure B-17 Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User Does Not Answer SIP IP SIP IP...
  • Page 177 Action Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session.

  • Page 178: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone-Authentication Error

    User B but is prompted for authentication credentials by the proxy server. User A’s SIP IP phone then reinitiates the call with an SIP INVITE request that includes it’s authentication credentials. Figure B-18 Cisco SIP IP Phone-to-Cisco SIP IP Phone—Authentication Error Proxy SIP IP...
  • Page 179 Cisco SIP IP phone A. phone A ACK—Cisco SIP IP phone A to Cisco SIP IP phone A sends a SIP ACK to the SIP proxy SIP proxy server server acknowledging the 407 error message. Resend INVITE—Cisco SIP IP...
  • Page 180 Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Cisco SIP IP Phone 7960 Administrator Guide B-72 78-10497-02...

  • Page 181: Appendix

    A P P E N D I X Technical Specifications This appendix provides physical and operating environment and cable technical specifications for the Cisco SIP IP phones. This appendix also provides the connection specifications of your Cisco SIP IP phone. Physical and Operating Environment Specifications The following table lists the physical and operating specifications of the Cisco SIP IP phone.
  • Page 182 Appendix C Technical Specifications Physical and Operating Environment Specifications Table C-1 Cisco SIP IP Phone Operational and Physical Specifications (continued) Specification Value or Range Power 48 VDC, supplied locally at the desktop using an optional AC-to-DC poser supply Regulatory CE Marking...

  • Page 183: Cable Specifications

    Switchcraft 760 or equivalent. Connections Specifications The Cisco SIP IP phone has two RJ-45 ports that each support 10/100 Mbps half- or full-duplex connections to external devices—the network port and access port. You can use either Category 3 or 5 cabling for 10 Mpbs connections, but use Category 5 for 100 Mbps connections.
  • Page 184 Appendix C Technical Specifications Connections Specifications Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 185: Appendix

    A P P E N D I X Translated Safety Warnings This appendix repeats in multiple languages the warnings that appear in the “Getting Started with Your Cisco SIP IP Phone” chapter of this guide. Installation Warning Warning Read the installation instructions before you connect the system to its power source.

  • Page 186: Product Disposal Warning

    ¡Advertencia! El desecho final de este producto debe realizarse según todas las leyes y regulaciones nacionales. Varning! Slutlig kassering av denna produkt bör skötas i enlighet med landets alla lagar och föreskrifter. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 187: Lightning Activity Warning

    ¡Advertencia! No operar el sistema ni conectar o desconectar cables durante el transcurso de descargas eléctricas en la atmósfera. Varning! Vid åska skall du aldrig utföra arbete på systemet eller ansluta eller koppla loss kablar. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 188: Selv Circuit Warning (Other Versions Available

    Zur Vermeidung von Elektroschock die Warnung Sicherheits-Kleinspannungs-Stromkreise (SELV-Kreise) nicht an Fernsprechnetzspannungs-Stromkreise (TNV-Kreise) anschließen. LAN-Ports enthalten SELV-Kreise, und WAN-Ports enthalten TNV-Kreise. Einige LAN- und WAN-Ports verwenden auch RJ-45-Steckverbinder. Vorsicht beim Anschließen von Kabeln. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 189 För att undvika elektriska stötar, koppla inte säkerhetskretsar med extra låg spänning (SELV-kretsar) till kretsar med telefonnätspänning (TNV-kretsar). LAN-portar innehåller SELV-kretsar och WAN-portar innehåller TNV-kretsar. Vissa LAN- och WAN-portar är försedda med RJ-45-kontakter. Iaktta försiktighet vid anslutning av kablar. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

  • Page 190: Circuit Breaker (15A) Warning

    Dette produktet er avhengig av bygningens installasjoner av Advarsel kortslutningsbeskyttelse (overstrøm). Kontroller at det brukes en sikring eller strømbryter som ikke er større enn 120 VAC, 15 A (USA) (240 VAC, 10 A internasjonalt) på faselederne (alle strømførende ledere). Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 191 Denna produkt är beroende av i byggnaden installerat kortslutningsskydd (överströmsskydd). Kontrollera att säkring eller överspänningsskydd används på fasledarna (samtliga strömförande ledare) ¥ för internationellt bruk max. 240 V växelström, 10 A (iþUSA max. 120 V växelström, 15 A). Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 192 Appendix D Translated Safety Warnings Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 193 G L O S S A R Y Authentication, Authorization, and Accounting. AAA is a suite of network security services that provides the primary framework through which access control can be set up on your Cisco router or access server.
  • Page 194 DNS is also used to assist in the locating remote gatekeepers and to reverse-map raw IP addresses to host names of administrative domains. DNIS Dialed number identification service (the called number). Digital signal processor. DTMF Dual tone multi-frequency. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 195 (H.245, H.225.0, and Q.931) to describe its actual protocol. H.323 RAS Registration, admission, and status. The RAS signaling function performs registration, admissions, bandwidth changes, status and disengage procedures between the VoIP gateway and the gatekeeper. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 196 This can be necessary in networks which do not support multicast. node A H.323 entity that uses RAS to communicate with the gatekeeper, for example, an endpoint such as a terminal, proxy, or gateway. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 197 MAY offer location services. Registration, admission, and status protocol. This is the protocol that is used between endpoints and the gatekeeper to perform management functions. Robbed bit signaling. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 198 User Agent Server (or user agent): A user agent server is a server application that contacts the user when a SIP request is received, then returns a response on behalf of the user. The response accepts, rejects or redirects the request. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 199 Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with POTs-like functionality, reliability, and voice quality. VoIP is a blanket term, which generally refers to Cisco’s standards based (for example H.323) approach to IP voice traffic.
  • Page 200 Glossary Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 201 Accept-Language header field A-10 overview accessing buttons firmware version 3-33 information network statistics 3-31 line status messages 3-31 volume access port 1-14 address proxy server 3-20 TFTP server adjusting, phone placement 2-18 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 202 SIP parameters compliance information manually 2-11 configuration, erasing via TFTP configuration files connections 1-13, 2-16 default Contact header field A-11 creating Content-Encoding header field A-11 example 3-15 Content-Length header field A-11 modifying guidelines Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 203 2-14 DHCP IP subnet mask 2-14 registration 3-11, 3-21 TFTP server 2-14 Encryption header field A-11 releasing address endpoint, SIP server parameter erasing dialing pad configuration directory services parameters 3-28 settings 3-28 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 204 Hide header field A-11 OS79XX.txt host name parameter RINGLIST.DAT SIPDefault.cnf firmware image ICMP, description 1-11 updating 3-33 image version version, viewing 3-33 information button Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 205 1-11 manually configuring SIP parameters 3-18 Max-Forwards header field A-11 messages, status 3-31 keys message URI parameter 3-20 on-screen mode methods scroll soft CANCEL INVITE OPTIONS LCD screen REGISTER line buttons Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 206 DHCP server domain name erase configuration guidelines 2-13 parameters host name common 2-7, 3-8 IP address configuring MAC address network 2-13 operational VLAN ID subnet mask erasing 3-28 TFTP server network 2-13 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 207 3-19 Authentication Password 3-19 password dtmf_db_level configuring 3-10 3-16 dtmf_outofbound 3-10 line 3-16 image_version phone linex_authname 3-16 adjusting placement 2-18 linex_name connecting 3-16 2-16 linex_password 3-16 linex_shortname 3-16 Message URI 3-20 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 208 Cisco Catalyst switches 2-18 1-14 overview external 1-14 prerequisites prerequisites 1-12 1-12 secondary directory number Priority header field A-11 supported features product supported protocols 1-10 overview DHCP product disposal warning 1-10 1-10 protocols 1-10 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 209 1-10 2543 1-1, 1-5 1-12 Real-Time Transport Protocol (RTP) 1-11 1-11 ReBy header field A-11 1-11 Record-Route header field A-11 RINGLIST.DAT redirect server Route header field A-11 registrar server RTP, description 1-11 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 210 3-16 Name 3-19 Simple Network Time Protocol (SNTP) 1-11 Out of Band DTMF 3-20 phone-specific configuration file architecture Preferred Codec 3-20 call flows Proxy Address 3-20 successful Proxy Port 3-20 unsuccessful B-58 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 211 SIPDefault.cnf 2-4, 2-7 SIP parameters technical specifications configuring manually 3-18 cablet configuring via TFTP server operating environment SNTP, description 1-11 physical soft keys TFTP, description 1-12 specifications TFTP server parameter cable connections Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...
  • Page 212 Trivial File Transfer Protocol (TFTP) 1-12 wall mounting phone 2-18 Warning header field A-12 WWW-Authenticate header field A-12 UDP, description 1-12 unlocking, configuration mode Unsupported header field A-12 updating firmware 3-33 URL dialing 1-10 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02...

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