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Cisco 7960G Administrator's Manual

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Cisco SIP IP Phone Administrator Guide
Release 6.0, 6.1, 7.0, 7.1, 7.2, 7.3, 7.4
March, 2005
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 7960G

  • Page 1 Cisco SIP IP Phone Administrator Guide Release 6.0, 6.1, 7.0, 7.1, 7.2, 7.3, 7.4 March, 2005 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 OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. CCSP, CCVP, the Cisco Square Bridge logo, Follow Me Browsing, and StackWise are trademarks of Cisco Systems, Inc.; Changing the Way We Work, Live, Play, and Learn, and iQuick Study are service marks of Cisco Systems, Inc.; and Access Registrar, Aironet, ASIST, BPX, Catalyst, CCDA, CCDP, CCIE, CCIP, CCNA, CCNP, Cisco, the...

  • Page 3: Table Of Contents

    SIP Capabilities SIP Components SIP Clients SIP Servers BTXML Support Cisco CallManager XML Support Network Capabilities Configuration Features Signaling Support 1-10 Dial-Plan and Messaging Support 1-10 Routing and Proxy Support 1-11 Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 4 Supported Protocols 1-13 Where to Go Next 1-14 Installing Cisco IP 7960G/7940G Phone Hardware on the Desktop or Wall C H A P T E R Placing the Phone on the Desktop Installing the Phone on the Wall Cabling the Phone Ports...
  • Page 5 Gateway to Cisco SIP IP Phone in a SIP Network Call Setup and Disconnect Call Setup and Hold Call to a Gateway Acting As an Emergency Proxy from a Cisco SIP IP Phone Cisco SIP IP Phone to Cisco SIP IP Phone Simple Call Hold...
  • Page 6 Three-Way Calling B-38 Call from a Cisco SIP IP Phone to a Gateway Acting As a Backup Proxy in a SIP Network B-42 Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Backup Proxy...
  • Page 7 Network engineers, system administrators, Cisco partners, and telecommunications engineers should use this guide to learn the tasks required to set up the Cisco IP phone in an SIP network. The described tasks are administration-level tasks and are not intended for the end users of the phones. Many of the tasks involve configuring network settings that could affect the ability of the phone to function in the network and require an understanding of IP networking and telephony concepts.

  • Page 8: Document Organization

    Objectives Objectives This guide provides necessary information to get the Cisco IP phone operational in a SIP network. It is not intended to provide information on how to implement a SIP or a VoIP network. For information on implementing SIP and VoIP networks, refer to the documents listed in the “Related Documentation”...

  • Page 9: Related Documentation

    Cisco IOS Voice Command Reference • http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123tcr/123tvr/ Cisco IOS IP Configuration Guide • http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/ip_vcg.htm Cisco IOS IP Command Reference, Volume 1 of 4: Addressing and Services • http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123tcr/123tip1r/ • Cisco IOS IP Command Reference, Volume 2 of 4: Routing Protocols http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123tcr/123tip2r/ •...

  • Page 10: Obtaining Documentation

    These sections explain how to obtain technical information from Cisco Systems. Cisco.com You can access the most current Cisco documentation on the World Wide Web at this URL: http://www.cisco.com/univercd/home/home.htm You can access the Cisco website at this URL: http://www.cisco.com...

  • Page 11: Documentation Feedback

    The Cisco TAC website is located at this URL: http://www.cisco.com/tac Accessing all the tools on the Cisco TAC website requires a Cisco.com user ID and password. If you have a valid service contract but do not have a login ID or password, register at this URL: http://tools.cisco.com/RPF/register/register.do...

  • Page 12: Obtaining Additional Publications And Information

    TAC Case Priority Definitions To ensure that all cases are reported in a standard format, Cisco has established case priority definitions. Priority 1 (P1)—Your network is “down” or there is a critical impact to your business operations. You and Cisco will commit all necessary resources around the clock to resolve the situation.
  • Page 13 Preface • Internet Protocol Journal is a quarterly journal published by Cisco Systems for engineering professionals involved in designing, developing, and operating public and private internets and intranets. You can access the Internet Protocol Journal at this URL: http://www.cisco.com/ipj •...
  • Page 14 Preface Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 15: New Information In This Release

    You can log your session by means of console or Telnet and save the log to a file. New tones and ringers are added for interoperation with the Cisco BTS 10200 Softswitch. • Selectable tones and ringing patterns already known to the phone are available to use as alternate •...

  • Page 16: Chapter 1 Product Overview

    Cisco IP 7960G/7940G Phone Overview Cisco IP 7960G/7940G phones are full-featured telephones that can be plugged directly into a SIP network and can be used very much like a standard PBX telephone. Phone terminals can attach to the existing data network infrastructure, using 10BASE-T and 100BASE-T interfaces on an Ethernet switch.
  • Page 17 LCD screen Displays information about the Cisco IP phone, such as the time, date, phone number, caller ID, line and call status, and the soft key tabs. The screen is 4.25 x 3 inches (10.79 x 7.62 cm) and has an adjustable contrast.
  • Page 18 Cisco IP 7960G/7940G Phone Overview Line or Opens a new line or speed-dials the number on the LCD screen. Phones in the speed-dial Cisco IP Phone 7960 series have six line or speed-dial buttons, and phones in the button 7940 series have two. Footstand Allows adjustment of the angle of the phone base.
  • Page 19 Chapter 1 Product Overview Cisco IP 7960G/7940G Phone Overview Figure 1-3 shows the connections on the back of the Cisco IP phone. Cisco IP 7960G/7940G phones have the same hardware configuration. Figure 1-3 Cisco IP Phone Cable Connections RS232 10/100 SW...

  • Page 20: Session Initiation Protocol Overview

    When a headset is used, an amplifier is not required. However, a coil cord is required to connect the headset to the headset port on the back of your Cisco IP 7960G/7940G phone. For information on ordering compatible headsets and coil cords for the Cisco IP 7960G/7940G phone, go to http://cisco.getheadsets.com...

  • Page 21: Sip Components

    Access Protocol (LDAP) servers, a database application, or an eXtensible Markup Language (XML) application. These provide back-end services such as directory, authentication, and billing. Figure 1-4 SIP Architecture SIP proxy and redirect servers SIP user agents (UAs) SIP gateway PSTN Legacy PBX Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 22: Sip Clients

    BTXML user interface control. Cisco CallManager XML Support The Cisco SIP IP phone supports Cisco CallManager XML cards that you configure to provide data such as stock quotes, calendars, and directory lookups. Phone users access this information using phone buttons such as the Services or Directories buttons or soft keys.

  • Page 23: Network Capabilities

    Chapter 1 Product Overview Network Capabilities For more information about using XML on your Cisco SIP IP phone, refer to the following: IP Telephony • http://www.hotdispatch.com/cisco-ip-telephony Cisco Call Manager Services Developer Kit • http://www.cisco.com/pcgi-bin/dev_support/access_level/product_support • Developing Cisco IP Phone Services by Darrick Deel, Mark Nelson, and Anne Smith,...

  • Page 24: Signaling Support

    Dial-Plan and Messaging Support The Cisco SIP IP phone supports a dial plan that enables automatic dialing and generation of a selectable secondary dial tone. Direct entries must be saved using the Keep soft key in order to be stored in the personal directory.

  • Page 25: Routing And Proxy Support

    The Domain Name Server (DNS) SRV query is used to locate servers for a given service. SIP on Cisco 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: Supported Languages And Character Set

    For example, an application that displayed the current weather in Sweden using Swedish characters can be displayed on the Cisco SIP IP phone. If you develop the same application for a Spanish locale, the application can be translated into Spanish.

  • Page 27: Supported Protocols

    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 defined in RFC 791. Real-Time Transport Protocol. Supports transport of real-time data (such as voice) over data networks.

  • Page 28: Where To Go Next

    Simple Network Time Protocol. Synchronizes computer clocks on an IP network. Current date and time are supported using SNTP including time zone and daylight saving time. The Cisco SIP IP phone uses SNTP for date and time support. Transmission Control Protocol. Provides a reliable byte-stream transfer service between endpoints on the Internet.

  • Page 29: Placing The Phone On The Desktop

    7.5-degree increments from flat to 60 degrees. Installing the Phone on the Wall Mount the Cisco IP 7960G/7940G phone on the wall by using the footstand as a mounting bracket or by using the optional locking bracket. To use the optional locking bracket, refer to the...

  • Page 30: Cabling The Phone Ports

    Step 5 Hang the phone on the wall. Cabling the Phone Ports The Cisco MGCP IP phone has several ports on the back of the phone, shown in Figure 1-3 on page 1-5. Plug the appropriate equipment into the appropriate port.

  • Page 31: Using The Phone With A Cisco Catalyst Switch

    IP addresses. For redundancy, you can use the AC adapter even if you are using inline power from the Cisco Catalyst switches. The Cisco IP 7960G/7940G phone can share the power load being used from the inline power and external power source.

  • Page 32: Connecting The Phone To Power

    WS-X4608-2PSU and WS-X4608—External –48VDC power shelf common equipment for the • Cisco Catalyst 4006 with two AC-to-DC power supply units (PSUs) and one empty bay for redundant option, and the 110-V 15-A AC-to-48VDC PSU redundant option for the power shelf.

  • Page 33: Where To Go Next

    Chapter 2 Installing Cisco IP 7960G/7940G Phone Hardware on the Desktop or Wall Where to Go Next Where to Go Next Chapter 3, “Initializing Cisco SIP IP Phones,” for information on installing firmware, • customizing configuration files, and connecting the phone to power sources and the network.
  • Page 34 Chapter 2 Installing Cisco IP 7960G/7940G Phone Hardware on the Desktop or Wall Where to Go Next Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 35: Prerequisites

    C H A P T E R Initializing Cisco SIP IP Phones This chapter describes the initial firmware installation tasks and configuration process for the Cisco IP 7960G/7940G phone in a Session Initiation Protocol (SIP) network. It provides information on the following: Prerequisites, page 3-1 •...

  • Page 36: Overview Of The Initialization Process

    Note Overview of the Initialization Process The initialization process for the Cisco SIP IP phone establishes network connectivity and makes the phone operational in your SIP network. After you connect your phone to the network and to a power supply, the phone begins initialization, during which the following occurs: The phone loads the firmware image.
  • Page 37 The phone learns its VLAN membership. If the phone is connected to a Cisco 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 send a DHCP request for its IP settings (if using DHCP).

  • Page 38: Information About Configuration Files

    Set only one variable per line. Indicate the end of a line with <lf> or <cr><lf>. – Put the variable and value on the same line, and do not break the line. – Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 39: How To Customize The Default Configuration File

    If you have an existing system from a release earlier than Release 7.x, upgrade your system firmware • as described in the “How to Upgrade Your Cisco SIP IP Phone Firmware Image” section on page 4-3 before proceeding. Procedure Obtain the default configuration file as follows: Step 1 Go to the Cisco.com SIP IP 7940/7960 phone software-download site at...

  • Page 40: How To Customize The Default Configuration File

    Save the file to the root directory of your TFTP server or to a subdirectory in which all phone-specific configuration files are stored. Configuration Example The following is an example of the SIPDefault configuration file that you downloaded from Cisco.com: # SIP Default Configuration File # Image Version...
  • Page 41 5060; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0; VAD setting 0-disable (Default), 1-enable ####### New Parameters added in Release 2.2 ###### # NAT/Firewall Traversal Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 42: How To Customize A Phone-Specific Configuration File

    Define the parameter in a phone-specific configuration file only if that phone needs to use a different dial plan than the one being used by the other phones in the same system. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 43 Procedure Obtain the phone-specific configuration file as follows: Step 1 Go to the Cisco.com SIP IP 7940/7960 phone software-download site at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960. Download the SIP<mac-addr>.cnf file to the root directory of your TFTP server or to a subdirectory in which all phone-specific configuration files are stored.

  • Page 44: How To Customize The Configuration From The Phone Menu

    – To delete any mistakes, press the << soft key. – To cancel all changes and exit a menu during configuration, press Cancel. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-10...

  • Page 45: Unlocking And Locking The Phone

    LCD changes to an unlocked state. If you are located elsewhere in the Cisco IP 7960G/7940G phone menus, the next time you access the Network Configuration or SIP Configuration menu, the unlocked icon displays, and you can modify the network and SIP configuration settings.
  • Page 46 “Prerequisites” section on page 3-1.) Using normal Cisco Discovery Protocol (CDP) processes, the phone locates the DHCP server upon connection to the network; the server in turn provides the TFTP server address. Provide it directly to the phone by means of the TFTPServer parameter as described in this –...
  • Page 47 Half-100—Port is configured to be a half-duplex, 100-MB • connection. Full-10—Port is configured to be a full-duplex, 10-MB • connection. • Half-10—Port is configured to be a half-duplex, 10-MB connection. Default is Auto. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-13...

  • Page 48: Setting And Restoring Phone-Specific Parameters

    TFTP Server IP address of the TFTP server. 1. If you have an administrative VLAN setting assigned on the Cisco Catalyst switch, that setting overrides any changes made on the phone. 2. DHCP must be disabled.
  • Page 49 Select Validate > Exit. If necessary, select Back to exit the menu. Select Save. The phone programs the new information into flash memory and resets. Step 6 Step 7 Relock the phone. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-15...

  • Page 50: Setting End-User Call Preferences

    3-8): – To enable end users to modify a preference, set to 0 or 1. – To prohibit end users from modifying a preference, set to 2 or 3. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-16...

  • Page 51: How To Set The Date And Time

    You can set date, time, and daylight savings time (DST) parameters. The current date and time is supported on the Cisco IP 7960G/7940G phone using Simple Network Time Protocol (SNTP) and is displayed on the LCD. DST and time-zone settings are also supported.

  • Page 52: How To Set The Date And Time

    Save the file to the root directory of your TFTP server. Step 5 To adjust the phone display to European Day-Month-Year format, add the following entry to the Note SIPDefault.cnf file: date_format:D/M/Y. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-18...
  • Page 53 1. If sntp_mode is set to anycast, the sntp_server address will be ignored and subsequent sntp requests will be sent to the first sntp server that responded (the first sntp request must be unconditionally sent to the broadcast address) . Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-19...
  • Page 54 GMT+02:00 Athens, Rome EET (Eastern European Time), USSR-zone1, MEST (Middle European Summer Time), FST (French Summer Time) GMT+03:00 Baghdad, Moscow BT (Baghdad Time), USSR-zone2 GMT+03:30 Tehran IT (Iran Time) Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-20...

  • Page 55: Time-Zone Configuration Examples

    The following is an example of an absolute DST configuration: 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 Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-21...

  • Page 56: How To Create Dial Plans

    How to Create Dial Plans Dial plans enable the Cisco SIP IP phone to support automatic dialing and generation of a secondary dial tone. If a single dial plan is used for a system of phones, the dial plan is best specified in the default configuration file.
  • Page 57 — — — 5678 Null output. Null output. — XYZ..XYZ1234 — X.Y.Z... X1Y2Z345 — 919%1%2%3 91912345678 12 appears twice. AB...X%1X.. AB123X12X45 You can reuse the string. X%1X%1X%1 X12X12X12 Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-23...
  • Page 58 If desired, specify at the end of each string to denote the type of plan (for example, <!-- comment --> <!-- Long Distance --> <!-- Corporate Dial Plan --> Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-24...

  • Page 59: Dial-Plan Configuration Examples

    For more information on Bellcore tones, refer to Bellcore GR-506-CORE. For more information Note on tones in BTS 10200 Softswitch features, refer to the Cisco BTS 10200 Softswitch website at http://www.cisco.com/en/US/partner/products/hw/vcallcon/ps531/index.html. Type the following to indicate the end of the dial-plan template: Step 6 </DIALTEMPLATE/>...

  • Page 60: How To Verify Initialization

    Chapter 4, “Managing Cisco SIP IP Phones,” for information on upgrading firmware and • performing other management tasks. Chapter 5, “Monitoring Cisco SIP IP Phones,” for information on debugging and on viewing • network statistics. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 3-26...

  • Page 61: How To Customize Cisco Ip 7960G/7940G Phone Rings

    Where to Go Next, page 4-7 • How to Customize Cisco IP 7960G/7940G Phone Rings The Cisco IP 7960G/7940G phone ships with two ring types: Chirp1 and Chirp2. However, you can create and add custom rings. Procedure Create a pulse-code-modulation (PCM) file for each desired ring type and store it in the root directory Step 1 of your TFTP server.

  • Page 62: How To Access Xml Cards

    Download a bitmap to be used as the phone logo (configured by means of the logo_url parameter). • Restrictions for XML Cards The phone supports Cisco CallManager XML up through version 3.0. It does not support the following XML objects in version 3.1 and later: CiscoIPPhoneIconMenu, CiscoIPPhoneExecute, CiscoIPPhoneError, and CiscoIPPhoneResponse.

  • Page 63: How To View Your Cisco Sip Ip Phone Firmware Image Version

    Chapter 4 Managing Cisco SIP IP Phones How to View Your Cisco SIP IP Phone Firmware Image Version How to View Your Cisco SIP IP Phone Firmware Image Version You can determine your firmware image version. Procedure Select Settings > Status.
  • Page 64 If your current release is earlier than Release 2.3, upgrade to Release 2.3 as follows: Step 3 From the Cisco.com website, do the following: Go to the Cisco.com SIP IP 7940/7960 phone software-download site at http://www.cisco.com/cgi-bin/tablebuild.pl/sip-ip-phone7960. Download the following to the root directory of your TFTP server: Release 2.3 ZIP archive, default configuration file, and release notes.

  • Page 65: Upgrading To Version 7.0 And Later Releases

    Chapter 4 Managing Cisco SIP IP Phones How to Upgrade Your Cisco SIP IP Phone Firmware Image Save the modified file to the TFTP server. Reinitialize each phone as described in Step above. If your current release is Release 5.2 or higher, upgrade to Release 6.0 or 6.1 as follows: Step 5 From the Cisco.com website listed in Step...

  • Page 66: How To Upgrade Your Cisco Sip Ip Phone Firmware Image And Reboot Remotely

    Chapter 4 Managing Cisco SIP IP Phones How to Upgrade Your Cisco SIP IP Phone Firmware Image and Reboot Remotely Step 2 Edit your SIPDefault.cnf and/or SIP<mac addr>.cnf file to have the correct image version. For example: image_version: P0S3-07-0-00 Reboot the phone.

  • Page 67: Where To Go Next

    During reboot, the phone sees the new image and upgrades to it with a synchronization value as specified in the syncinfo.xml file. The procedure is finished. Where to Go Next Chapter 5, “Monitoring Cisco SIP IP Phones,” for information on debugging and on viewing network statistics. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 68 Chapter 4 Managing Cisco SIP IP Phones Where to Go Next Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 69: Chapter 5 Monitoring Cisco Sip Ip Phones

    • How to Use the Command-Line Interface to Monitor Phones You can use Telnet or a console to connect to your Cisco IP 7960G/7940G phone, and you can and use the command-line interface (CLI) to debug or troubleshoot the phone.
  • Page 70 State Manager. • cc—Call control. • • cc-msg—Call-control messages. • error—General error debug output. sip-task—SIP task. • sip-state—SIP state machine. • sip-messages—SIP messaging. • sip-reg-state—SIP registration state machine. • Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 71 Range is 1 to 1480. Default is 100. • timeout—How long, in seconds, to wait before a request times out. Default is 2. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 72 CLI Commands (continued) Command Purpose Instructs the Cisco IP 7960G/7940G phone to register with SIP Phone> register {option value | line value } the proxy server. The keywords and argument are as follows: option value—Whether each line is registered. Valid •...
  • Page 73 Keywords are as follows: open—Enables the use of the test functionality. • close—Disables the use of the test functionality. • Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 74 A typical command is test key 23234. • onhook—Simulates a handset on-hook event. offhook—Simulates a handset off-hook event. • show—Shows test feedback. • hide—Hides test feedback. • Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 75: Output Examples

    W350 unprovisioned proxy_backup Telnet Session The following sample output shows the initial Telnet session using a UNIX server: UNIX% telnet 10.18.10.10 Trying 10.18.10.10... Connected to 10.18.10.10. Escape character is '^]'. Password :***** Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 76 1234 25640 26295@10.18.192.230 3333 user33 3434 user34 3636 user36 3737 user37 3838 user38 53@10.18.192.230 user53 user54 5550100 user1 5550101 5550102 Fid Mantel 9195550103 ciscotest@abccompany Fid Mantel croquet@abc.sip.com handball@abc.sip.com 9195550104 5550105 Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 77 Lst: SysBuf Length:040 Max:040 Head:0x0047d374 Tail:0x00479d74 Lst: GuiBufTmr Length:005 Max:005 Head:0x004950b4 Tail:0x00495094 Lst: DSPBuf Length:030 Max:030 Head:0x00478bd4 Tail:0x00478b74 Lst: DSP_Msg Length:000 Max:002 Head:0x00000000 Tail:0x004beb98 Lst: DSP_Msg (KA) Length:000 Max:001 Head:0x00000000 Tail:0x004bebbc Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 78 00000000, TxQMax 00000000, TxXCol 00000000, TxFram 00000398 Overflow Counters... UDP 00000000, ICMP 00000000, NonIP 00000000, TCP 00000000 CDP 00000000, Unknown 00000000, Arp 00000000 Use 'clear ethernet' to clear data Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-10...
  • Page 79 07. Pattern: 9,101....Rewrite: Timeout: 0000 UserMode: Phone RouteMode: Default 08. Pattern: 9,10....Rewrite: Timeout: 0000 UserMode: Phone RouteMode: Default 09. Pattern: 9,10* Rewrite: Timeout: 0006 UserMode: Phone RouteMode: Default Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-11...
  • Page 80 The following sample output shows the LSM parameters: Phone1> show lsm ------------------ LSM lcbs ------------------- call_id line state ----------------------------------------------- IDLE 0x004e1f00 IDLE 0x004e1f14 IDLE 0x004e1f28 IDLE 0x004e1f3c IDLE 0x004e1f50 IDLE 0x004e1f64 Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-12...
  • Page 81 0x004e2184 0x004e2284 0x004e2384 0x004e2484 FSMXFR Parameters The following sample output shows the FSMXFR parameters: Phone1> show fsmxfr ------------------------ FSMXFR xcbs ------------------------- xfr_id type method xfr_call_id cns_call_id -------------------------------------------------------------- 0x004e25c4 0x004e25f0 Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-13...
  • Page 82 REGISTERED 3595 2539 10.18.192.230:5060 REGISTERED 3595 2539 10.18.192.230:5060 REGISTERED 3595 2540 10.18.192.230:5060 REGISTERED 3595 2543 10.18.192.230:5060 REGISTERED 3595 2543 10.18.192.230:5060 1-BU NONE undefined:0 Note: APR is Authenticated, Provisioned, Registered Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-14...
  • Page 83 PortCnt: 0001 MacAddr: 0030:94c2:5d40 Type: 00000001 GTick:00001287 LastTry: 00000000 Mode: 00000001 Update: 00000000 [01] IPAddr: 10.18.199.1 PortCnt: 0001 MacAddr: 0000:0c07:ac08 Type: 00000001 GTick:00001287 LastTry: 00001287 Mode: 00000001 Update: 00000000 Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-15...
  • Page 84 : UNPROVISIONED voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : "1" xml_card_dir : "" xml_card_file : "CARD.XML" telnet_level : 2 services_url : "http://10.10.149.2/ciscodir/directory.xml" directory_url : "http://10.10.93.154/CiscoServices/Directory.asp" Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-16...
  • Page 85 : 5060 proxy5_port : 5060 proxy6_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 2000 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-17...

  • Page 86: How To Use The Phone Menus To Access Status Information

    This section contains the following: Viewing Status Messages, page 5-19 • Viewing Network Statistics, page 5-19 • Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-18...

  • Page 87: Viewing Status Messages

    Port 0 Half, 10—Indication that the network is in a linked state and has autonegotiated a half-duplex • 10-Mbps connection. Port 1 Full, 100—Indication that the network is in a linked state and has autonegotiated a full-duplex • 100-Mbps connection. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-19...
  • Page 88 Port 1 Half, 10—Indication that the network is in a linked state and has autonegotiated a half-duplex 10-Mbps connection. Select Exit. Step 3 Note To reset the values, power the phone off and on. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x 5-20...

  • Page 89: Appendix

    A P P E N D I X Compliance with RFC 3261 This appendix describes how the Cisco IP 7960G/7940G phone complies with the IETF definition of SIP as described in RFC 3261. It contains compliance information on the following: •...

  • Page 90: Sip Methods

    The phone supports both user and device registration. REFER — NOTIFY Used for REFER and remote reboot. SIP Responses The Cisco IP 7960G/7940G 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 91: 2Xx Response-Successful Responses

    If a 401 Unauthorized response is received, the phone accepts the response and sends a new request that contains the user’s authentication information in the format of the HTTP digest as modified by RFC 3261. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 92 414 Request—URL Too The user is notified if this response is received. Long 415 Unsupported Media The user is notified if this response is received. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 93: 5Xx Response-Server Failure Responses

    500 Internal Server Error The phone initiates a graceful call disconnect, and the user is notified. 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Gateway Timeout 505 Version Not Supported Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 94: 6Xx Response-Global Responses

    SIP Header Fields Header Field Supported? Accept Accept-Encoding Accept-Language Allow Also 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 Record-Route Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 95: Sip Session Description Protocol Usage

    SIP Session Description Protocol Usage SDP Headers Supported? v—Protocol version o—Owner or creator and session identifier s—Session name t—Time description c—Connection information m—Media name and transport address a—Media attribute lines Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 96: Transport Layer Protocols

    Supported? Digest Authentication Proxy Authentication SIP DNS Records Usage DNS Resource Record Type Supported? Type A Type SRV NAPTR SIP DTMF Digit Transport Transport Type Supported? RFC 2833 In-band tones Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 97: Appendix

    NOTIFY—Notifies the user of the status of a transfer using REFER. Also used for remote reboot • and message waiting indication (MWI). 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 98: Gateway To Cisco Sip Ip Phone In A Sip Network

    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 99 Disconnect—Gateway 1 to PBX A Gateway 1 sends a Disconnect message to PBX A. Release—PBX A to Gateway 1 PBX A sends a Release message to Gateway 1. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 100 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 101 The phone sends a SIP ACK to Gateway 1. The ACK confirms that the phone has received the 200 OK response. The call session is now temporarily inactive. No RTP packets are being sent. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 102 Call to a Gateway Acting As an Emergency Proxy from a Cisco SIP IP Phone Figure B-3 illustrates a successful call from a Cisco SIP IP phone to a gateway acting as an emergency proxy. Figure B-3 Successful Call from Cisco SIP IP Phone to SIP Gateway (Emergency Proxy)

  • Page 103: Cisco Sip Ip Phone To Cisco Sip Ip Phone

    The gateway sends a Release Complete message to the PBX and the call session the PBX terminates. Cisco SIP IP Phone to Cisco SIP IP Phone The following sections describe and illustrate successful calls from a Cisco SIP IP phone to another Cisco SIP IP phone: Simple Call Hold, page B-8 •...
  • Page 104 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 105 • The media capability User A is ready to receive is specified. 180 Ringing—Phone B to Phone A Phone B sends a SIP 180 Ringing response to Phone A. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 106: 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 107 14. INVITE (c=IN IP4 IP-User B) B is disconnected from C. 15. 200 OK 16. ACK A is taken off hold. The RTP channel between A and B is reestablished. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-11...
  • Page 108 INVITE—Phone B to Phone C Phone B sends a SIP INVITE request to Phone C. The request is an invitation to User C to participate in a call session. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-12...

  • Page 109: Call Waiting

    Figure B-6 illustrates a successful call between Cisco SIP IP phones in which two parties are in a call, and one of the participants receives a call from a third party and then returns to the original call. In this call flow scenario, the end users are User A, User B, and User C.
  • Page 110 B has disconnected from A, but the call with C (on hold) remains. 21. 200 OK 22. ACK C is taken off hold. The RTP channel between B and C is reestablished. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-14...
  • Page 111 ACK—Phone B to Phone A Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has received the 200 OK response from Phone A. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-15...
  • Page 112 Call_ID=2 SDP: c=IN IP4 10.10.10.0 200 OK—Phone C to Phone B Phone C sends a SIP 200 OK response to Phone B. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-16...

  • Page 113: Call Transfer Without Consultation

    This is called a blind or unattended transfer. 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. The call flow scenario is as follows: User A calls User B.
  • Page 114 11. BYE 12. 200 OK 13. INVITE (Referred-By: B) 14. 100 TRYING 15. 180 RINGING 16. 200 OK 17. ACK 18. NOTIFY (Event: Refer) 19. 200 OK 2-way voice path Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-18...
  • Page 115 Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has received the 200 OK response from Phone A. User B dials User C. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-19...

  • Page 116: Call Transfer Without Consultation Using Failover

    This is called a blind or unattended transfer. 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 using an IP network. The call flow scenario is as follows: User A calls User B.
  • Page 117 The media capability User A is ready to receive is specified. • 100 Trying—Phone B to Phone B sends a SIP 100 Trying response to Phone A. The response indicates that Phone A the INVITE request has been received. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-21...
  • Page 118 REFER message. 200 OK—Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B. The response notifies Phone B that the BYE message was received. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-22...

  • Page 119: Call Transfer With Consultation

    Call Transfer with Consultation 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 120 19. INVITE (Referred by: B, Replaces: B) 20. 200 OK 21. ACK 22. BYE 23. 200 OK 24. NOTIFY (Event: Refer) 25. 200 OK 26. BYE 27. 200 OK 2-way voice path Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-24...
  • Page 121 Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has received the 200 OK response from Phone A. User B dials User C. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-25...
  • Page 122 REFER message has been received. INVITE—Phone A to Phone C Phone A sends a SIP INVITE request to Phone C. The request contains the following information: Referred-By: B • Replaces: B • Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-26...

  • Page 123: Call Transfer With Consultation Using Failover

    Call Transfer with Consultation Using Failover Figure B-10 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 124 19. BYE(Also: C) 20. 200 OK 21. BYE 22. 200 OK 23. INVITE C (Requested-By: B) 24. 100 TRYING 25. 180 RINGING 26. 200 OK 27. ACK 2-way voice path Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-28...
  • Page 125 Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has received the 200 OK response from Phone A. User B dials User C. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-29...
  • Page 126 Phone B sends a BYE message to Phone A. The message includes the following information: Also: C • The message indicates that the 501 Not Implemented message was received in response to a REFER message. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-30...

  • Page 127: Network Call Forwarding (Unconditional

    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. The call flow scenario is as follows: User B requests that the network forward all calls to Phone C.
  • Page 128 The media capability User A is ready to receive is specified. • INVITE—SIP proxy server to The SIP proxy server sends the SIP INVITE request to the SIP redirect server. SIP redirect server Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-32...

  • Page 129: Network Call Forwarding (Busy

    Network Call Forwarding (Busy) Figure B-12 illustrates successful call forwarding between Cisco SIP IP phones in which User B has requested call forwarding from the network in the event the phone is busy. When User A calls User B, the SIP proxy server tries to place the call to Phone B, and, if the line is busy, the call is transferred to Phone C.
  • Page 130 4. ACK 5. INVITE B 6. 486 Busy Here 7. ACK 8. INVITE C 9. 180 Ringing 10. 200 OK 11. 200 OK 12. ACK 13. ACK 2-way RTP channel Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-34...
  • Page 131 The SIP proxy server forwards the SIP ACK to the Phone C. The ACK confirms Phone C that Phone A has received the 200 OK response from Phone C. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-35...

  • Page 132: Network Call Forwarding (No Answer

    Network Call Forwarding (No Answer) Figure B-13 illustrates successful call forwarding between Cisco SIP IP phones in which User B has requested call forwarding from the network in the event that there is no answer. When User A calls User B, the proxy server tries to place the call to Phone B, and, if there is no answer, the call is transferred to Phone C.
  • Page 133 Phone C sends a SIP 200 OK response to the SIP proxy server. server 200 OK—SIP proxy server to The SIP proxy server forwards the SIP 200 OK response to Phone A. Phone A Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-37...

  • Page 134: 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 IP 7960G/7940G phones, which are connected using an IP network.
  • Page 135 User A is taken off hold. The RTP channel 1 between User A and B is re-established. User B mixes the RTP channels 1 and 2 and establishes a conference brigde between Users A and C Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-39...
  • Page 136 Phone B sends a SIP ACK to Phone A. The ACK confirms that Phone B has received the 200 OK response from Phone A. The RTP channel between Phone A and Phone B is torn down. A is put on hold. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-40...
  • Page 137 Phone B acts as a bridge mixing the RTP channel between A and B with the channel between B and C, establishing a conference bridge between A and C. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-41...

  • Page 138: Call From A Cisco Sip Ip Phone To A Gateway Acting As A Backup Proxy In A Sip Network

    SIP Call Flows Call Flow Scenarios for Successful Calls Call from a Cisco SIP IP Phone to a Gateway Acting As a Backup Proxy in a SIP Network Figure B-15 illustrates a successful call from a Cisco SIP IP phone to a gateway acting as a backup proxy.
  • Page 139 The gateway sends a SIP 200 OK response to A. The response notifies A that phone (A) the gateway has received the BYE request. Release Complete—Gateway to the The gateway sends a Release Complete message to the PBX, and the call session terminates. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-43...

  • Page 140: Call From A Cisco Sip Ip Phone To A Cisco Sip Ip Phone Using A Sip Backup Proxy

    Cisco SIP IP phone to a Cisco SIP IP phone that uses a SIP backup proxy. Figure B-16 A Successful Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Backup Proxy...

  • Page 141: Call From A Cisco Sip Ip Phone To A Cisco Sip Ip Phone Using A Sip Emergency Proxy

    Cisco SIP IP phone (A) User A that the backup proxy has received the BYE request. Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Emergency Proxy Figure B-17 illustrates a successful call from a Cisco SIP IP phone to a Cisco SIP IP phone via an emergency proxy.
  • Page 142 Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-17 Successful Call from a Cisco IP Phone to a Cisco IP Phone Using a SIP Emergency Proxy SIP IP Phone SIP IP Phone Primary Proxy Emergency Proxy...

  • Page 143: Call Flow Scenarios For Failed Calls

    This section describes call flows for the following scenarios, which illustrate unsuccessful calls: Gateway to Cisco SIP IP Phone in a SIP Network, page B-47 • Cisco SIP IP Phone to Cisco SIP IP Phone in a SIP Network, page B-52 • Gateway to Cisco SIP IP Phone in a SIP Network...

  • Page 144: Called Number Is Busy

    Disconnect (Busy)—Gateway 1 to Gateway 1 sends a Disconnect message to PBX A. PBX A Release—PBX A to Gateway 1 PBX A sends a Release message to Gateway 1. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-48...

  • Page 145: Called Number Does Not Answer

    Step Action Description ACK—Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP ACK to the phone. The ACK confirms that A has received the 486 Busy Here response. The call session attempt terminates. Release Complete—Gateway 1 to...

  • Page 146: Client, Server, Or Global Error

    Client, Server, or Global Error Figure B-20 illustrates an unsuccessful call in which User A initiates a call to User B and receives a class 4xx, 5xx, or 6xx response. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-50...
  • Page 147 100 Trying—Cisco SIP IP phone to Gateway 1 The phone sends a SIP 100 Trying response to Gateway 1. The response indicates that the INVITE request has been received. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-51...

  • Page 148: Cisco Sip Ip Phone To Cisco Sip Ip Phone In A Sip Network

    Gateway 1 sends a Release Complete message to PBX A, and the call session attempt terminates. Cisco SIP IP Phone to Cisco SIP IP Phone in a SIP Network The following scenarios are Cisco SIP IP phone to Cisco SIP IP phone: Called Number Is Busy, page B-53 •...
  • Page 149 ACK—Phone A to Phone B Phone A sends a SIP ACK to the Phone B. The ACK confirms that Phone A has received the 486 Busy Here response from Phone B. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-53...

  • Page 150: Authentication Error

    User A initiates a call to User B but is prompted for authentication credentials by the proxy server. User A’s SIP IP phone then reinitiates the call with a SIP INVITE request that includes authentication credentials. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-54...
  • Page 151 Phone A sends a SIP ACK to the SIP proxy server acknowledging the 407 error server message. Resend INVITE—Cisco SIP IP Phone A resends a SIP INVITE to the SIP proxy server with authentication phone A to SIP proxy server credentials. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-55...
  • Page 152 Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x B-56...

  • Page 153: Appendix

    A P P E N D I X Technical Specifications of the Cisco IP 7960G/7940G Phone The following sections describe the technical specifications for the Cisco IP 7960G/7940G phone: Physical and Operating Environment Specifications, page C-1 • Cable Specifications, page C-2 •...

  • Page 154: Cable Specifications

    Use the PC-to-phone port to connect a network device, such as a computer, to the phone. For a diagram that identifies the different ports on the back of the Cisco IP 7960G/7940G phone, see the “Initializing Cisco SIP IP Phones” section on page 3-1.

  • Page 155: Appendix

    This parameter cannot be set in the configuration file. autocomplete (Phone-specific; optional) Configures automatic completion of numbers. Valid values are 0 (disable autocompletion) and 1 (enable autocompletion). Default is 1. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 156 AvgJit—Average jitter, which is an estimate of the statistical • variance of the RTP packet inter-arrival time, measured in timestamp unit and calculated according to RFC 1889. • a, b, c, d, e, f, g, h, and i—Integers. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...

  • Page 157: Appendix D Configurable Parameter For The Sip Ip Phone

    (Optional) Whether the conference bridge, when it hangs up, should attempt to join the two leaf nodes. Valid values are as follows: 0—Do not join the two leaf nodes. • 1—Join the two leaf nodes. • Default is 1. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 158 (Optional) Offset from the phone time when DST is in effect. When DST is over, the specified offset is no longer applied to the phone time. Valid values are hour/minute, –hour/minute, +hour/minute, hour, –hour, and +hour. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 159 Sunday or Sun, and 1 to 7, with 1 being Sunday and 7 being Saturday. When the name of the day is specified, the value is not case-sensitive. In the United States, the default is Sunday. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 160 5—6 dB above nominal • Default is 3. dtmf_inband (Optional) In-band signaling format. Valid values are 1 (generate DTMF digits in-band) and 0 (do not generate DTMF digits in-band). Default is 1. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 161 TFTP servers. Only dotted IP addresses are accepted. This value can be cleared by removing it from the configuration file or by changing its value to a null value “ ” or to “UNPROVISIONED.” Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x...
  • Page 162 (Optional) Number of the HTTP proxy port. Default is 80. image_version Firmware version that the phone should use. Enter the name of the image version as it is released by Cisco. Do not enter the filename extension (.bin). You cannot change the image version by changing the...
  • Page 163 If a value is not configured for the linex_password parameter when registration is enabled, the default password is used. Default is UNPROVISIONED. Valid values for x are (Cisco IP 7969G phone) 1 to 6 and (Cisco IP 7940G phone) 1 to 2. linex_shortname Labels a line key with a name other than the directory number.
  • Page 164 Half100—Port is configured to be a half-duplex, 100-MB • connection. Full10—Port is configured to be a full-duplex, 10-MB • connection. Half10—Port is configured to be a half-duplex, 10-MB • connection. Default is Auto. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-10...
  • Page 165 IP dotted-decimal notation. You must specify at least one address or the phones cannot Note register. proxy_backup_port (Optional) Port number of the backup proxy server. Default is 5060. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-11...
  • Page 166 The value of “x” can be from 1 to 6. If the value of the “x” is not specified in the proxyx_address parameter, the phone uses proxy1_address as the default. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-12...
  • Page 167 There are five possible numbers that can be configured on the Cisco IP 7960G but only one on the Cisco IP 7940G. The x variable is a string of up to 128 bytes. Note You cannot set this parameter in the configuration file.
  • Page 168 (Optional) Amount of time, in seconds, after which a SIP INVITE expires. This value is used in the Expire header field. Valid values are any positive number; however, we recommend 180. Default is 180. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-14...
  • Page 169 (Optional) UDP port used for SIP messages. All SIP REQUESTS use voip_control_port as the UDP source port when nat_enable = 1. Range is from 1025 to 65535. Default is 5060. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-15...
  • Page 170 DNS server list, and TFTP address). Valid values are Yes and No. Default is Yes. To manually configure your IP settings, you must set this parameter to No. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-16...
  • Page 171 IP address of the TFTP server. If you have an administrative VLAN Note setting assigned on the Catalyst switch, that setting overrides any changes made on the phone. DHCP must be disabled. Note Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-17...
  • Page 172 ID without the host name. Proxy Address (Phone-specific) IP address of the primary SIP proxy server that will be used by the phone. Enter this address in IP dotted-decimal notation. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-18...
  • Page 173 LCD instead. This parameter is used for display only. If a value is not specified for this parameter, the value in the Name variable is displayed. Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-19...
  • Page 174 Appendix D Configurable Parameters for the SIP IP Phone Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x D-20...

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