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Cisco SIP IP Phone Administrator's Manual
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Cisco SIP IP Phone Administrator's Manual

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Cisco SIP IP Phone Administrator Guide
Version 4.0
August 2002
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

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Summary of Contents for Cisco SIP IP Phone

  • Page 1 Cisco SIP IP Phone Administrator Guide Version 4.0 August 2002 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...
  • 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: Table Of Contents

    Product Overview C H A P T E R What Is Session Initiation Protocol? Components of SIP SIP Clients SIP Servers What Is the Cisco SIP IP Phone? BTXML Support Cisco CallManager XML Support Supported Features Physical Features Network Features...
  • Page 4 1-13 Connecting to Power 1-13 Using a Headset 1-14 The Cisco SIP IP Phone with a Catalyst Switch 1-14 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 3-43 Viewing Network Statistics 3-43 Viewing the Firmware Version 3-44 Upgrading the Cisco SIP IP Phone Firmware 3-44 Upgrading from Release 2.2 or Later Releases to Release 4.0 3-45 Upgrading from Release 2.1 or Earlier Releases to Release 4.0 3-45 Dual Booting from SCCP or MGCP to Release 4.0...
  • Page 6 Call from a Cisco SIP IP Phone to a Gateway Acting as a Backup Proxy B-54 Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone via a Backup Proxy B-56 Call from a Cisco SIP IP Phone to a Gateway Acting as an Emergency Proxy...
  • Page 7 Contents SELV Circuit Warning Circuit Breaker (15A) Warning L O S S A R Y N D E X Cisco SIP IP Phone Administrator Guide...
  • Page 8 Contents Cisco SIP IP Phone Administrator Guide...
  • Page 9 Cisco Session Initiation Protocol (SIP) IP phone 7940 or 7960 (hereafter referred to as a Cisco SIP IP phone). It also provides information on how to configure the network and SIP settings and change the settings and options of the Cisco SIP IP phone. The administrator guide also includes reference information such as Cisco SIP IP phone call flows and compliance information.

  • Page 10: Appendix A Sip Compliance With Rfc 3261 Information

    Objectives Objectives The Cisco SIP IP Phone 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: Document Conventions

    Preface Document Conventions The following is a list of Cisco VoIP publications that provide information about implementing a VoIP network: • Cisco IOS Voice, Video, and Fax Configuration Guide, Release 12.2 Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2 •...
  • Page 12 Innan du utför arbete på någon utrustning måste du vara medveten om farorna med elkretsar och känna till vanligt förfarande för att förebygga skador. (Se förklaringar av de varningar som förekommer i denna publikation i appendix "Translated Safety Warnings" [Översatta säkerhetsvarningar].) Cisco SIP IP Phone Administrator Guide...

  • Page 13: Obtaining Documentation

    The following sections explain how to obtain documentation from Cisco Systems. World Wide Web You can access the most current Cisco documentation on the World Wide Web at the following URL: http://www.cisco.com Translated documentation is available at the following URL: http://www.cisco.com/public/countries_languages.shtml...

  • Page 14: Obtaining Technical Assistance

    Technical Assistance Center The Cisco TAC is available to all customers who need technical assistance with a Cisco product, technology, or solution. Two types of support are available through the Cisco TAC: the Cisco TAC Web Site and the Cisco TAC Escalation Center.
  • Page 15 Cisco TAC Web Site. The Cisco TAC Web Site requires a Cisco.com login ID and password. If you have a valid service contract but do not have a login ID or password, go to the following URL to register: http://www.cisco.com/register/...
  • Page 16 Preface Obtaining Technical Assistance Cisco SIP IP Phone Administrator Guide...

  • Page 17: What Is Session Initiation Protocol?

    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?, page 1-3 • •...

  • Page 18: Components Of Sip

    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. Figure 1-1 SIP Architecture SIP Proxy and Redirect Servers SIP User Agents (UA) SIP Gateway PSTN Legacy PBX Cisco SIP IP Phone Administrator Guide...

  • Page 19: Sip Clients

    Cisco SIP IP phones are full-featured telephones that can be plugged directly into an IP network and can be used very much like a standard private branch exchange (PBX) telephone. The Cisco SIP IP phone is an IP telephony instrument that can be used in VoIP networks.
  • Page 20 Scroll Function toggles LCD screen—Desktop, which displays information about your Cisco SIP IP phone, such as the • time, date, your phone number, caller ID, line and call status and the soft key tabs. Line or speed-dial buttons—Opens a new line or speed dials the number on the LCD screen.

  • Page 21: Btxml Support

    Chapter 3, “Managing Cisco SIP IP Phones” for information about configuring these parameters. The Cisco SIP IP phone supports Cisco CallManger XML up to version 3.0. It does not support the XML objects added in Cisco CallManager XML version 3.1: •...

  • Page 22: Supported Features

    “Managing Cisco SIP IP Phones” section on page 3-1 for more information on configuration parameters. Ping support—Allows the user to use ping to see if a Cisco SIP IP phone is operational and how long • the response time is from the phone.

  • Page 23: Configuration Features

    Chapter 1 Product Overview What Is the Cisco SIP IP Phone? Configuration Features The Cisco SIP IP phone provides the ability to: Configure Ethernet port mode and speed • • Register with or unregister from a proxy server • Specify a TFTP boot directory •...

  • Page 24: Call Options

    (via a third-party tool that enables this feature to be configured). When a call is placed to the user’s phone, it is redirected to the appropriate forward destination by the SIP proxy server. Call hold—Allows the Cisco SIP IP phone user (user A) to place a call (from user B) on hold. When •...
  • Page 25 The Domain Name Server RR (DNS SRV) is used to locate servers for a given service. SIP on Cisco’s SIP IP phones uses a DNS SRV query to determine the IP address of the SIP proxy or redirect server. The query string generated is in compliance with RFC 2782, and prepends the protocol label with an underscore _, as in “_protocol._transport.”...

  • Page 26: Character Support

    Caller ID information. When a SIP message is received with ISO 8859-1 Latin1 characters in the caller ID strings, those caller ID strings are displayed on the Cisco SIP IP phone's LCD screen with the correct ISO 8859-1 Latin1 characters.

  • Page 27: Supported Protocols

    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. Hypertext Transfer Protocol (HTTP)—The phone contains limited support for HTTP 1.1. The •...

  • Page 28: Prerequisites

    Chapter 1 Product Overview Prerequisites Prerequisites For the Cisco SIP IP phone to successfully operate as a SIP endpoint in your network, your network must meet the following requirements: A working IP network is established. • For more information about configuring IP, refer to...

  • Page 29: Connecting To The Network

    For redundancy, you can use the Cisco AC adapter even if you are using inline power from the 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 30: Using A Headset

    Voice traffic to and from the Cisco SIP IP phone (auxiliary VLAN) • Data traffic to and from the PC connected to the switch through the access port of the Cisco SIP IP • phone (native VLAN) Isolating the phones on a separate, auxiliary VLAN increases the quality of the voice traffic and allows a large number of phones to be added to an existing network where there are not enough IP addresses.
  • Page 31 Chapter 1 Product Overview The Cisco SIP IP Phone with a Catalyst Switch http://www.cisco.com/univercd/home/home.htm Cisco SIP IP Phone Administrator Guide 1-15...
  • Page 32 Chapter 1 Product Overview The Cisco SIP IP Phone with a Catalyst Switch Cisco SIP IP Phone Administrator Guide 1-16...

  • Page 33: Initialization Process Overview

    The VLAN is configured. If the Cisco SIP IP phone is connected to a Catalyst switch, the switch notifies the phone of the voice VLAN defined on the switch. The phone needs to know its VLAN membership before it can proceed with the Dynamic Host Configuration Protocol (DHCP) request for its IP settings (if using DHCP).

  • Page 34: Installing The Cisco Sip Ip Phone

    Installing the Cisco SIP IP Phone 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 uses IP settings that are stored in Flash memory.

  • Page 35: 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 Cisco.com to the root directory of your TFTP server. File...

  • Page 36: Configuring Sip Parameters Via A Tftp Server

    Installing the Cisco SIP IP Phone 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 37 Before You Begin Ensure that you have downloaded the SIPDefault.cnf file from Cisco.com to the root directory of • your TFTP server. Review the guidelines and restrictions documented in the “Configuration File Guidelines”...
  • Page 38 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 39: Manually Configuring The Sip Parameters

    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 2-15.
  • Page 40 Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Procedure Step 1 Press the settings key. The Settings menu is displayed. Step 2 Highlight SIP Configuration. The SIP Configuration menu is displayed. Step 3 Highlight Line 1 Settings.

  • Page 41: Configuring Network Parameters

    Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Table 2-1 Manual SIP Configuration Parameters (continued) Parameter Required or Optional Description Proxy Address Required for the first IP address of the primary SIP proxy server that will be line configured on the used by the phone.

  • Page 42: Configuring Network Parameters Via A Dhcp Server

    3-2. By default, the network parameters are locked to ensure that end users cannot modify settings that might affect their network connectivity. • Review the guidelines on using the Cisco SIP IP phone menus documented in the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-15.

  • Page 43: Connecting The Phone

    Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Step 3 Press the Select soft key. The Network Configuration menu is displayed. Highlight DHCP Enabled. Step 4 Press the No soft key. DHCP is now disabled.

  • Page 44: Adjusting The Placement Of The Cisco Sip Phone

    Adjusting the Placement of the Cisco SIP Phone The Cisco SIP IP phone includes an adjustable footstand. When placing the phone on a desktop surface, you can adjust the tilt height to several different angles in 7.5 degree increments from flat to 60 degrees.
  • Page 45 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 46: Verifying Startup

    Chapter 2 Getting Started with Your Cisco SIP IP Phone Verifying Startup Figure 2-2 Adjusting the Footstand Cisco IP Phone (rear view) Footstand adjustment Adjustment plate Adjustment plate button raises and installation raises and lowers lowers adjustment screws holes (2)

  • Page 47: Using The Cisco Sip Ip Phone Menu Interface

    • exit the Edit panel. Reading the Cisco SIP IP Phone Icons When using the Cisco SIP IP phone, a variety of icons can display on the phone’s LCD. Table 2-2 lists and describes each icon that you might see while using the Cisco SIP IP phone.
  • Page 48 Number soft key. The character x displayed to the right of the icon indicates that registration has failed. The Cisco SIP IP phone configuration mode is locked. When the phone is locked, the phone’s network or SIP settings cannot be modified.

  • Page 49: 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 Cisco SIP IP phone users.
  • Page 50 Chapter 2 Getting Started with Your Cisco SIP IP Phone Creating Dial Plans • <DIALTEMPLATE> indicates the start of a template and </DIALTEMPLATE> indicates the end of a template • Rules are matched from start to finish with the longest matching rule taken as the one to use.
  • Page 51 Chapter 2 Getting Started with Your Cisco SIP IP Phone Creating Dial Plans 919 919 (No need to use the input.) .... 12345678 (Note that nothing goes in for the extra dots.) Route= “route” is default, emergency, or FQDN. FQDN is treated the same as default proxy. Route •...
  • Page 52 Chapter 2 Getting Started with Your Cisco SIP IP Phone Creating Dial Plans Cisco SIP IP Phone Administrator Guide 2-20...

  • Page 53: Changing Your Configuration

    Parameters via a TFTP Server” section on page 3-8. Use Telnet or a console to connect to your Cisco SIP IP phone and use the command-line interface • (CLI). You will need to know your phone’s IP address. Press Settings, select Network Configuration, and scroll down to IP Address to find this address.

  • Page 54: Modifying The Phone's Network Settings

    If the Network Configuration or SIP Configuration panel is displayed, the lock icon in the upper-right corner of your LCD changes 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 55 3-2. By default, the network parameters are locked to ensure that end users cannot modify settings that might affect their network connectivity. • Review the guidelines on using the Cisco SIP IP phone menus documented in the “Using the Cisco SIP IP Phone Menu Interface” section on page 2-15.
  • Page 56 IP Address Yes, but DHCP must IP address of the phone that either was assigned by DHCP or was be disabled. locally configured. MAC Address Factory-assigned unique 48-bit hexadecimal MAC address of the phone. Cisco SIP IP Phone Administrator Guide...

  • Page 57: Modifying The Phone's Sip Settings

    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 override the values stored in Flash memory.
  • Page 58 Date Format dial_template dnd_control Do Not Disturb dst_auto_adjust dst_offset dst_start_day dst_start_day_of_week dst_start_month dst_start_time dst_start_week_of_month dst_stop_day dst_stop_day_of_week dst_stop_month dst_stop_time dst_stop_week_of_month dtmf_avt_payload dtmf_db_level dtmf_inband dtmf_outofband Out of Band DTMF enable_vad Enable VAD end_media_port End Media Port Cisco SIP IP Phone Administrator Guide...
  • Page 59 Backup Proxy Port proxy_emergency Emergency Proxy proxy_emergency_port Emergency Proxy Port proxy_register Register with Proxy proxyN_address (N=1 to 6) Proxy Address proxyN_port (N=1 to 6) Proxy Port remote_party_id sip_invite_retx sip_retx sntp_mode sntp_server start_media_port Start Media Port sync Cisco SIP IP Phone Administrator Guide...

  • Page 60: Modifying Sip Parameters Via A Tftp Server

    While it is not required, Cisco recommends that you use the default configuration file to define values for SIP parameters that are common to all phones. Doing so will make controlling and maintaining your network easier.
  • Page 61 The default value is 0. autocomplete Optional Whether to have numbers automatically completed when dialing. Valid values are 0 (disable auto completion) or 1 (enable auto completion). The default is 1. Cisco SIP IP Phone Administrator Guide...
  • Page 62 3—The call waiting feature is enabled permanently • and cannot be turned on and off locally via the phone’s user interface. If specifying this value, specify this parameter in the phone-specific configuration file. The default value is 1. Cisco SIP IP Phone Administrator Guide 3-10...
  • Page 63 The default value is 1, or join two leaf nodes. date_format Optional The format to use for dates. Valid values are: M/D/Y—Month/day/year • D/M/Y—Day/ month/year • Y/M/D—Year/month/day • Y/D/M—Year/day/month • Y-M-D—Year-month-day • • YY-M-D—4-digit year-month-day The default is M/D/Y. Cisco SIP IP Phone Administrator Guide 3-11...
  • Page 64 This setting sets the phone to be a “call out” phone only. If specifying this value, specify this parameter in the phone-specific configuration file. The default value is 0. Cisco SIP IP Phone Administrator Guide 3-12...
  • Page 65 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 66 The IP address of the HTTP proxy server. You can use either a dotted IP address or a DNS name (a record only). http_proxy_port Optional The port number of the HTTP proxy port. The default is Cisco SIP IP Phone Administrator Guide 3-14...
  • Page 67 Description image_version Required 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 68 2 to a switch results in spanning tree loops and network confusion. outbound_proxy Optional The IP address of the outbound proxy server. You can use either a dotted IP address or a DNS name. Cisco SIP IP Phone Administrator Guide 3-16...
  • Page 69 Port number of the backup proxy server. Default is 5060. proxy_emergency Optional IP address of the emergency proxy server or gateway. Enter this address in IP dotted-decimal notation. proxy_emergency_port Optional Port number of the emergency proxy server. Default is 5060. Cisco SIP IP Phone Administrator Guide 3-17...
  • Page 70 IP address, the proxyN_address parameters should be configured as dotted IP addresses. proxyN_port Optional Port number of the SIP proxy server that will be used by phone lines other than line 1. Cisco SIP IP Phone Administrator Guide 3-18...
  • Page 71 16,384 to 32,766. sync Optional Value against which to compare the value in the syncinfo.xml file before performing a remote reboot. Valid value is a character string up to 32 characters long. Cisco SIP IP Phone Administrator Guide 3-19...
  • Page 72 The amount of time, in seconds, after which a SIP INVITE expires. This value is used in the Expire header field. The valid value is any positive number; however, Cisco recommends 180 seconds. The default is 180. timer_register_expires Optional The amount of time, in seconds, after which a REGISTRATION request expires.
  • Page 73 ; sip default configuration file # Image Version image_version: “P0S3-xx-y-zz” # Proxy Server proxy1_address: "proxy.company.com" proxy2_address: "" proxy3_address: "" proxy4_address: "" proxy5_address: "" proxy6_address: "" # Proxy Server Port (default - 5060) proxy1_port:"5060" proxy2_port:"" Cisco SIP IP Phone Administrator Guide 3-21...
  • Page 74 # TFTP Phone Specific Configuration File Directory tftp_cfg_dir: "./" # Time Server sntp_mode: "directedbroadcast" sntp_server: "172.16.10.150" #sntp_server: "sntp.company.com" time_zone: "EST" dst_offset: "1" dst_start_month: "April" dst_start_day: "" dst_start_day_of_week: "Sun" dst_start_week_of_month: "1" dst_start_time: "02" dst_stop_month: "Oct" dst_stop_day: "" dst_stop_day_of_week: "Sunday" Cisco SIP IP Phone Administrator Guide 3-22...

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

    Using an ASCII editor, create a phone-specific configuration file for each phone that you plan to install. Step 1 In the phone-specific configuration file, define values for SIP parameters shown in Table 3-4. Cisco SIP IP Phone Administrator Guide 3-23...
  • Page 76 ID, you can specify John Doe in this parameter to have John Doe appear on the callee end instead. If a value is not specified for this parameter, nothing is used. Cisco SIP IP Phone Administrator Guide 3-24...
  • Page 77 ; phone-specific configuration file sample line1_displayname: "jdoe43" line1_name: "43" line2_displayname: "jdoe44" line2_name: "44" line3_displayname: "pgatour" line3_name: "duval" line4_displayname: "jdoe46" line4_name: "46" line5_displayname: "jdoe47" line5_name: "47" line6_displayname: "jdoe48" line6_name: "48" phone_label: "jdoe4X" phone_prompt: "John-43" Cisco SIP IP Phone Administrator Guide 3-25...

  • Page 78: Modifying The Sip Parameters Directly On Your Phone

    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 2-15.
  • Page 79 6, and then continue with Step In addition to the line settings, you can highlight and press Select to configure the parameters on the SIP Step 8 Configuration menu shown in Table 3-6: Cisco SIP IP Phone Administrator Guide 3-27...
  • 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 SIP Configuration menu. Caution When you have completed your changes, ensure that you lock the phone as described in the “Locking Configuration Mode” section on page 3-2. Cisco SIP IP Phone Administrator Guide 3-29...

  • Page 82: Using The Command-Line Interface

    Managing Cisco SIP IP Phones Using the Command-Line Interface Using the Command-Line Interface You can use Telnet or a console to connect to your Cisco SIP IP phone to debug or troubleshoot the phone. Table 3-7 shows the available CLI commands:...
  • Page 83 • Do not use the debug all command, because it can Note cause the phone to become inoperable. This command is for use only by Cisco TAC personnel. Manipulates the DNS system. The following arguments are SIP Phone> dns used: -p: Prints out the DNS cache table.
  • Page 84 2. Instructs the Cisco SIP IP phone to register with the proxy SIP Phone> register {option | line} server. Option values are 0 and 1; 0 is unregister and 1 is register.
  • Page 85 Table 3-7 CLI Commands (continued) Command Purpose Shows information about the SIP IP phone. The following SIP Phone> show {arp | debug | strpool | memorymap | dump | malloctable | stacks | status | abort_vector keywords are used: | flash | dspstate | rtp | tcp | lsm | fsm | fsmdef arp: Displays contents of the ARP cache.
  • Page 86 This command can be used only if the telnet_level parameter is set to allow privileged commands to be executed. • dialplan: Shows the phone’s dial plan. • timers: Shows the current status of the platform timers Cisco SIP IP Phone Administrator Guide 3-34...
  • Page 87 Telnet session specified by the session argument. The msg keyword allows you to send a message to another terminal logged into the phone; for example, you can send a message telling everyone else that is logged in to log off. Cisco SIP IP Phone Administrator Guide 3-35...

  • Page 88: Setting The Date, Time, And Daylight Saving Time

    Setting the Date, Time, and Daylight Saving 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 Saving Time (DST) and time zone settings are also supported.
  • Page 89 Table 3-8 Table 3-9 for an explanation on how these values work, depending on the sntp_server parameter value. • sntp_server—(Required) IP address of the SNTP server from which the phone will obtain time data. Cisco SIP IP Phone Administrator Guide 3-37...
  • Page 90 BST (British Summer Time), MEWT( Middle European Winter Time), SWT (Swedish Winter Time), FWT (French Winter Time) GMT+02:00 Athens, Rome EET (Eastean European Time), USSR-zone1, MEST (Middle European Summer Time), FST (French Summer Time) Cisco SIP IP Phone Administrator Guide 3-38...
  • Page 91 In the United States, the default value is October. dst_start_time—Time of day on which DST begins. Valid values are hour/minute (02/00) or hour • (02:00). In the United States, the default value is 02:00. Cisco SIP IP Phone Administrator Guide 3-39...
  • Page 92 (additional configuration text omitted) time_zone : PST 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 Administrator Guide 3-40...

  • Page 93: Erasing The Locally Defined Settings

    Press the Select soft key. The Network Configuration settings are displayed. Step 3 Highlight Erase Configuration. Step 4 Press the Yes soft key. Step 5 Press the Save soft key. The phone programs the new information into Flash memory and resets. Step 6 Cisco SIP IP Phone Administrator Guide 3-41...

  • Page 94: Erasing The Locally Defined Sip Settings

    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 Administrator Guide 3-42...

  • Page 95: Viewing Status Messages

    Port 0 Full, 10—Indicates that the network is in a linked state and has autonegotiated a full-duplex • 10-Mbps connection. Port 0 Half, 10—Indicates that the network is in a linked state and has autonegotiated a half-duplex • 10-Mbps connection. Cisco SIP IP Phone Administrator Guide 3-43...

  • Page 96: Viewing The Firmware Version

    Upgrading the Cisco SIP IP Phone Firmware You can use one of two methods to upgrade the firmware on your Cisco SIP IP phones. You can upgrade the firmware on one phone at a time using the phone-specific configuration, or you can upgrade the firmware on a system of phones using the default configuration file.

  • Page 97: Upgrading From Release 2.2 Or Later Releases To Release 4.0

    Managing Cisco SIP IP Phones Upgrading the Cisco SIP IP Phone Firmware • Ensure that the latest version of the Cisco SIP IP phone firmware has been copied from Cisco.com to the root directory of your TFTP server. See the upgrade scenarios in Table 3-11 to determine how to upgrade.

  • Page 98: Performing An Image Upgrade And Remote Reboot

    Performing an Image Upgrade and Remote Reboot With Version 2.0 and newer of the Cisco SIP IP phone, you can perform an image upgrade and remote reboot using NOTIFY messages and the syncinfo.xml file. The dialplan.xml file can also be pushed down to the phones using a NOTIFY with a check-sync Event header.
  • Page 99 The phone the performs a normal reboot process as described in the “Initialization Process Overview” section on page 2-1, sees the new image, and upgrades to the new image with a synchronization value of what is specified in the syncinfo.xml file. Cisco SIP IP Phone Administrator Guide 3-47...
  • Page 100 Chapter 3 Managing Cisco SIP IP Phones Performing an Image Upgrade and Remote Reboot Cisco SIP IP Phone Administrator Guide 3-48...

  • Page 101: Appendix

    A P P E N D I X SIP Compliance with RFC 3261 Information This section describes how the Cisco SIP IP phone complies with the IETF definition of SIP as described in RFC 3261. This section contains compliance information on the following: SIP Functions, page A-1 •...

  • Page 102: Sip Methods

    REFER None. NOTIFY Used for REFER and remote reboot. SIP Responses Release 4.0 of the Cisco SIP IP phone supports the following SIP responses: 1xx Response—Information Responses, page A-2 • 2xx Response—Successful Responses, page A-3 • 3xx Response—Redirection Responses, page A-3 •...

  • Page 103: Xx Response—Redirection Responses

    182 Queued processes the 100 Trying response. 183 Session The SIP IP phone does not generate this message. Upon Progress receiving this response, the phone provides early media cut-through and then waits for a 200 OK response.
  • Page 104 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 105 This response is received only by the phone in this release. 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 Administrator Guide...
  • Page 106 5xx Response—Server Failure Responses 5xx Response Comments 500 Internal Server Error The Cisco SIP IP phone does not generate these 5xx responses. For an incoming response, the SIP IP phone initiates a graceful call 501 Not Implemented disconnect. 502 Bad Gateway...

  • Page 107: Sip Header Fields

    6xx Response—Global Responses 6xx Response Comments 600 Busy Everywhere The Cisco SIP IP phone does not generate these 6xx responses. For an incoming response, the SIP IP phone initiates a graceful call 603 Decline disconnect. 604 Does Not Exist Anywhere...

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

    Require Response-Key Retry-After Route Server Subject Timestamp Unsupported User-Agent Warning WWW-Authenticate SIP Session Description Protocol (SDP) Usage SDP Headers Supported? v—Protocol version o—Owner or creator and session identifier s—Session name t—Time description c—Connection information Cisco SIP IP Phone Administrator Guide...

  • Page 109: Transport Layer Protocols

    Multicast UDP SIP Security Authentication Basic Authentication Digest Authentication Proxy Authentication SIP DNS Records Usage DNS Resource Record Type Supported? Type A Type SRV SIP DTMF Digit Transport Transport Type Supported? RFC 2833 In-band tones Cisco SIP IP Phone Administrator Guide...
  • Page 110 Appendix A SIP Compliance with RFC 3261 Information SIP DTMF Digit Transport Cisco SIP IP Phone Administrator Guide A-10...

  • Page 111: Appendix

    REFER—Indicates that the user (recipient) should contact a third party for use in transferring parties. NOTIFY—Notifies the user of the status of a transfer using REFER. Also used for remote reset. • The following types of responses are used by SIP and generated by the Cisco SIP gateway: SIP 1xx—Informational Responses • •...

  • Page 112: Appendix B Sip Call Flow

    User A and User B. 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. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.
  • Page 113 SIP 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 User A and User B. 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. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.
  • Page 115 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 116: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Simple Call Hold

    In this call flow scenario, the two end users are User A and User B. User A and User B are both using Cisco SIP IP phones, which are connected via an IP network.
  • Page 117 Call Flow Scenarios for Successful Calls User B places User A on hold. User B takes User A off hold. The call continues. Figure B-3 Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold SIP IP SIP IP Phone User A IP Network Phone User B 1.
  • Page 118 B to Cisco SIP IP phone A 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the connection has been made.

  • Page 119: 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 120 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-4 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 121 B to Cisco SIP IP phone A 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
  • Page 122 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. INVITE—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP INVITE request to Cisco SIP IP phone C.

  • Page 123: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Call Waiting

    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 124 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-5 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting SIP IP SIP IP SIP IP Phone Phone User B IP Network Phone User A User C 1. INVITE B 2.
  • Page 125 B to Cisco SIP IP phone A 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
  • Page 126 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 Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone C.

  • Page 127: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Call Transfer Without Consultation

    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 128 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-6 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer without Consultation SIP IP SIP IP SIP IP Phone User B Phone User A Phone User C IP Network 1.
  • Page 129 B to Cisco SIP IP phone A 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
  • Page 130 Cisco SIP IP phone B will be disconnecting from the call. 200 OK—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B. The to Cisco SIP IP phone B 200 OK response notifies Cisco SIP IP phone B that the BYE message was received.
  • Page 131 NOTIFY message notifies Cisco SIP IP phone C of the REFER event. 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the NOTIFY message was received.

  • Page 132: Bye/Also

    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 Using Bye/Also SIP IP SIP IP SIP IP Phone User B Phone User A Phone User C IP Network 1.
  • Page 133 B to Cisco SIP IP phone A 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
  • Page 134 REFER message. 200 OK—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B. The to Cisco SIP IP phone B 200 OK response notifies Cisco SIP IP phone B that the BYE message was received.

  • Page 135: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Call Transfer With Consultation

    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 Consultation...
  • Page 136 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 User B Phone User A Phone User C IP Network 1.
  • Page 137 B to Cisco SIP IP phone A 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
  • Page 138 C to Cisco SIP IP phone B 200 OK—Cisco SIP IP phone C Cisco SIP IP phone C sends a SIP 200 OK response to Cisco SIP IP phone B. The to Cisco SIP IP phone B 200 OK response notifies Cisco SIP IP phone B that the connection has been made.
  • Page 139 Replaces: B 200 OK—Cisco SIP IP phone C Cisco SIP IP phone C sends a SIP 200 OK message to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the INVITE request has been received.
  • Page 140 Failover to Bye/Also Figure B-9 illustrates a successful call between Cisco SIP IP phones in which two parties are in a call, one of the participants contacts a third party, and then that participant transfers the call to the third party.

  • Page 141: Bye/Also

    Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-9 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer with Consultation Using Failover to Bye/Also SIP IP SIP IP SIP IP Phone User B Phone User A...
  • Page 142 B to Cisco SIP IP phone A 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
  • Page 143 C to Cisco SIP IP phone B 200 OK—Cisco SIP IP phone C Cisco SIP IP phone C sends a SIP 200 OK response to Cisco SIP IP phone B. The to Cisco SIP IP phone B 200 OK response notifies Cisco SIP IP phone B that the connection has been made.
  • Page 144 REFER message. 200 OK—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP 200 OK message to Cisco SIP IP phone B. The to Cisco SIP IP phone B 200 OK response notifies Cisco SIP IP phone B that the BYE request has been received.

  • Page 145: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Network Call Forwarding (Unconditional

    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. They are all using Cisco SIP IP phones, which are connected via an IP network.
  • Page 146 Description INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to the SIP proxy server. The to SIP proxy server INVITE request is an invitation to User B to participate in a call session.

  • Page 147: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Network Call Forwarding (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.
  • Page 148 INVITE request is an invitation to User B to participate in a call session. 486 Busy Here—Cisco SIP IP Cisco SIP IP phone B sends a 486 Busy here message to the SIP proxy server. The phone B to SIP proxy server message indicates that Cisco SIP IP phone B is in use and the user is not willing or able to take additional calls.

  • Page 149: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone Network Call Forwarding (No 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.
  • Page 150 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-12 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 151 B 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to the SIP proxy server. The to SIP proxy server response confirms receipt of the cancellation request. INVITE—SIP proxy server to SIP proxy server sends a SIP INVITE request to Cisco SIP IP phone C.

  • Page 152: Cisco Sip Ip Phone-To Cisco Sip Ip Phone Three-Way Calling

    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 153 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-13 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 154 B to Cisco SIP IP phone A 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
  • Page 155 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. INVITE—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP INVITE request to Cisco SIP IP phone C.

  • Page 156: Call Flow Scenarios For Failed Calls

    Cisco SIP IP phone A. SIP IP phone B acts as a bridge mixing the RTP channel between User A and User B with the channel between User B and User C; establishing a conference bridge between User A and User C.
  • Page 157 Cisco SIP IP phone. 486 Busy Here—Cisco SIP IP The Cisco SIP IP phone sends a SIP 486 Busy Here response to Gateway 1. The phone to Gateway 1 486 Busy Here response is a client error response that indicates that User B was successfully contacted but that User B was not willing or was unable to take the call.

  • Page 158: 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 ACK confirms that IP phone User A has received the 486 Busy Here response. The call session attempt is now being terminated.
  • Page 159 200 OK—Cisco SIP IP phone to The Cisco SIP IP phone sends a SIP 200 OK response to Gateway 1. The 200 OK Gateway 1 response confirms that User A has received the 486 Busy Here response. The call session attempt is now being terminated.

  • Page 160: Gateway-To-Cisco Sip Ip Phone-Client, Server, Or Global Error

    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 161: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone-Called User Is Busy

    Cisco SIP IP phone. 4xx/5xx/6xx Failure— The Cisco SIP IP phone sends a class 4xx, 5xx, or class 6xx failure response to Cisco SIP IP phone to Gateway 1 Gateway 1. Depending on which class the failure response is, the call actions differ.

  • Page 162: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone-Called User Does Not Answer

    • 486 Busy Here—Cisco SIP IP Cisco SIP IP phone B sends a 486 Busy here message to the Cisco SIP IP phone A. phone B to Cisco SIP IP phone A 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 163: Cisco Sip Ip Phone-To-Cisco Sip Ip Phone-Authentication Error

    A to Cisco SIP IP phone B 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A. The to Cisco SIP IP phone A response confirms receipt of the cancellation request.

  • Page 164: Call From A Cisco Sip Ip Phone To A Gateway Acting As A Backup Proxy

    Call from a Cisco SIP IP Phone to a Gateway Acting as a Backup Proxy Figure B-20 illustrates a successful call from a Cisco SIP IP phone to a gateway acting as a backup proxy. Cisco SIP IP Phone Administrator Guide...
  • Page 165 (second try) the INVITE message. INVITE—Cisco SIP IP phone to Cisco SIP IP phone retries a third time to connect to the proxy by sending out the primary proxy (third try) INVITE message. INVITE—Cisco SIP IP phone to...

  • Page 166: Call From A Cisco Sip Ip Phone To A Cisco Sip Ip Phone Via A Backup Proxy

    Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone via a Backup Proxy Figure B-21 illustrates a successful call from a Cisco SIP IP phone to a Cisco SIP IP phone via a backup proxy. Cisco SIP IP Phone Administrator Guide...

  • Page 167: Call From A Cisco Sip Ip Phone To A Cisco Sip Ip Phone Via A Backup Proxy

    Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Figure B-21 A Successful Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone via a Backup Proxy SIP IP Phone SIP IP Phone Primary Proxy...
  • Page 168 (User A) to primary proxy sending out the INVITE message. (fourth try) INVITE—Cisco SIP IP phone Cisco SIP IP phone retries a fifth time to connect to the primary proxy by sending (User A) to primary proxy (fifth out the INVITE message. try) INVITE—Cisco SIP IP phone...

  • Page 169: Call From A Cisco Sip Ip Phone To A Gateway Acting As An Emergency Proxy

    Call from a Cisco SIP IP Phone to a Gateway Acting as an Emergency Proxy Figure B-22 illustrates a successful call from a Cisco SIP IP phone to a gateway acting as an emergency proxy. Figure B-22 A Successful Call from Cisco SIP IP Phone to Gateway (Emergency Proxy)

  • Page 170: Call From A Cisco Sip Ip Phone To A Cisco Sip Ip Phone Via Emergency Proxy

    Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone via Emergency Proxy Figure B-23 illustrates a successful call from a Cisco SIP IP phone to a Cisco SIP IP phone via emergency proxy. User B is the extension of the dial template with the “Route” attribute as “emergency” in the dialplan.xml file.

  • Page 171: Call From A Cisco Sip Ip Phone To A Cisco Sip Ip Phone Via Emergency Proxy

    Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Figure B-23 A Successful Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone via Emergency Proxy SIP IP Phone SIP IP Phone Primary Proxy Emergency Proxy...
  • Page 172 Description 200 OK—Emergency proxy to Emergency proxy sends a SIP 200 OK response to User A. The 200 OK response Cisco SIP IP phone (User A) notifies User A that the connection has been made. ACK—Cisco SIP IP phone User A acknowledges the emergency proxy’s Connect message.

  • Page 173: Appendix

    Regulatory Safety Compliance, page C-2 Connections Specifications, page C-3 • Physical and Operating Environment Specifications The following table lists the physical and operating specifications of the Cisco SIP IP phone. Table C-1 Cisco SIP IP Phone Operational and Physical Specifications Specification...

  • Page 174: Cable Specifications

    .1 inches (2.5 mm). The center pin is positive (+) voltage. The miniature power plug required to mate with the power jack on the phone is a Switchcraft 760 or equivalent. Regulatory Safety Compliance The Cisco IP Phone models 7960, 7940, and 7910 meet the following regulatory safety and compliance approvals: CE Marking •...

  • Page 175: Connections Specifications

    LAN-to-phone jack. Use the PC-to-phone port to connect a network device, such as a computer, to the phone. For a diagram identifying the different ports on the back of the Cisco SIP IP phone, see the “Connecting the Phone” section on page 2-11.
  • Page 176 Appendix C Technical Specifications Connections Specifications Cisco SIP IP Phone Administrator Guide...

  • Page 177: Appendix

    A P P E N D I X Translated Safety Warnings This appendix contains in multiple languages the warnings that should be used with 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 178: Lightning Activity Warning

    LAN (Lokaal netwerk) poorten bevatten SELV circuits en WAN (Regionaal netwerk) poorten bevatten TNV circuits. Sommige LAN en WAN poorten gebruiken allebei RJ-45 connectors. Ga voorzichtig te werk wanneer u kabels verbindt. Cisco SIP IP Phone Administrator Guide...
  • Page 179 (overstroom)beveiliging. Controleer of er een zekering of stroomverbreker van niet meer dan 120 Volt wisselstroom, 15 A voor de V.S. (240 Volt wisselstroom, 10 A internationaal) gebruikt wordt op de fasegeleiders (alle geleiders die stroom voeren). Cisco SIP IP Phone Administrator Guide...
  • Page 180 Denna produkt är beroende av i byggnaden installerat kortslutningsskydd (överströmsskydd). Varning! 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 Administrator Guide...
  • Page 181 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 182 (PIN), and so on. Local exchange carrier. A SIP redirect or proxy server uses a a location service to get information about a caller’s locations. location server Location services are offered by location servers. Cisco SIP IP Phone Administrator Guide GL-2...
  • Page 183 A registrar is a server that accepts REGISTER requests. A registrar is typically colocated with a proxy registrar or redirect server and 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 Administrator Guide GL-3...
  • Page 184 Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with VoIP POTs-like functionality, reliability, and voice quality. VoIP is a blanket term, that generally refers to the Cisco standards-based (for example H.323) approach to IP voice traffic. Cisco SIP IP Phone Administrator Guide GL-4...
  • Page 185 VLAN ID parameter Call-ID header field Allow header field call preferences menu Also header field call transfer alternate TFTP server, enabling call waiting disabled authentication call waiting enabled name, configuring 3-24 character support 1-10 services Cisco SIP IP Phone Administrator Guide IN-1...
  • Page 186 3-24 dialing pad password 3-24 directory services short name 3-24 network parameters description 1-11 manually 2-10 server parameters via DHCP 2-10 documentation SIP parameters conventions manually related viii via TFTP domain name parameter Cisco SIP IP Phone Administrator Guide IN-2...
  • Page 187 1-14 call waiting disabled using 1-14 call waiting enabled headset and speaker toggle do not disturb Hide header field secondary directory number host name parameter Cisco SIP IP Phone Administrator Guide IN-3...
  • Page 188 3-24 naming convention, phone-specific configuration file language support 1-10 3-15 LCD screen network line buttons connections 1-13 lines, configuring parameters authentication name 3-24 administrative VLAN ID name 3-24 Cisco SIP IP Phone Administrator Guide IN-4...
  • Page 189 Organization header field subnet mask address OS79XX.txt TFTP server Out of Band DTMF parameter 3-28 required overview services_url Cisco SIP IP phone initialization process Authentication Name 3-27 product Authentication Password 3-27 dtmf_avt_payload 3-13 Cisco SIP IP Phone Administrator Guide IN-5...
  • Page 190 TFTP 1-11 phone 1-11 adjusting placement URL dialing 2-12 connecting verifying startup 2-11 2-14 connections 1-12 phone-specific configuration file access port 1-13 creating network example 1-13 2-6, 3-25 network port modifying 1-13 3-23 Cisco SIP IP Phone Administrator Guide IN-6...
  • Page 191 3-27 1-11 port 3-27 registration, enabling 3-18, 3-28 specifying 3-18 safety warnings, translated circuit breaker (15A) warning installation warning Real-Time Transport Protocol (RTP) lightning activity warning 1-11 Record-Route header field product disposal warning Cisco SIP IP Phone Administrator Guide IN-7...

  • Page 192: Selv Circuit Warning D

    3-26 funtions configuring on your phone 3-26 gateways configuring via TFTP server header fields SNTP, description 1-11 IP phone, overview soft keys methods specifications overview cable parameters connections Authentication Name operating environment 3-27 Cisco SIP IP Phone Administrator Guide IN-8...
  • Page 193 3-20 timer_t2 3-21 verifying startup 2-14 timers, retransmission 3-21 Via header field Timestamp header field viewing firmware version 3-44 time zone abbreviations 3-38 VLAN time zone setting administrative setting operational time zone 3-36 Cisco SIP IP Phone Administrator Guide IN-9...
  • Page 194 Index volume buttons wall mounting phone 2-13 Warning header field WWW-Authenticate header field Cisco SIP IP Phone Administrator Guide IN-10...

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