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3-26

Cisco SIP IP Phone Administrator Guide

Chapter 3 Managing Cisco SIP IP Phones

Modifying the Phone’s SIP Settings

proxy1_address: 1.2.3.4

proxy2_address: 1.2.3.4

proxy3_address: 1.2.3.4

proxy4_address: 1.2.3.4

proxy5_address: 1.2.3.4

proxy6_address: 1.2.3.4

proxy1_port: 5060

proxy2_port: 5060

proxy3_port: 5060

proxy4_port: 5060

proxy5_port: 5060

proxy6_port: 5060

callerid_blocking: 0

dtmf_outofband: avt

network_media_type: auto

tos_media: 5

dtmf_avt_payload: 101

time_zone: EST

call_waiting: 1

cnf_join_enable : 1

semi_attended_transfer : 1

Modifying the SIP Parameters Directly on Your Phone

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

Before You Begin

•

Unlock configuration mode as described in the

“Unlocking Configuration Mode” section on

page 3-2

. By default, the SIP parameters are locked to ensure that end users cannot modify settings

that might affect their call capabilities.

•

Review the guidelines on using the Cisco SIP IP phone menus documented in the

“Using the

Cisco SIP IP Phone Menu Interface” section on page 2-15

.

•

Line parameters (those identified as linex) define a line on the phone. If you configure a line to use
an e-mail address, that line can be called only by using an e-mail address. Similarly, if you configure
a line to use a number, that line can be called only by using the number.

•

When configuring the Preferred Codec and Out of Band DTMF parameters, press the Change soft
key until the option you desire is displayed and then press the Save soft key.

•

After making your changes, relock configuration mode as described in the

“Locking Configuration

Mode” section on page 3-2

.

Step 1

Press the settings key. The Settings menu appears.

Step 2

Highlight SIP Configuration. The SIP Configuration menu appears.

Step 3

Highlight Line 1 Settings.

Step 4

Press the Select soft key. The Line 1 Configuration menu appears.

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Summary of Contents for SIP IP Phone

Page 1: ...eadquarters 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 Cisco SIP IP Phone Administrator Guide Version 4 0 August 2002 ...

Page 2: ...the television or radio antenna until the interference stops Move the equipment to one side or the other of the television or radio Move the equipment farther away from the television or radio Plug the equipment into an outlet that is on a different circuit from the television or radio That is make certain the equipment and the television or radio are on circuits controlled by different circuit br...

Page 3: ...al Assistance xii Cisco com xii Technical Assistance Center xii Cisco TAC Web Site xiii Cisco TAC Escalation Center xiii C H A P T E R 1 Product Overview 1 1 What Is Session Initiation Protocol 1 1 Components of SIP 1 2 SIP Clients 1 3 SIP Servers 1 3 What Is the Cisco SIP IP Phone 1 3 BTXML Support 1 5 Cisco CallManager XML Support 1 5 Supported Features 1 6 Physical Features 1 6 Network Features...

Page 4: ...guring SIP Parameters via a TFTP Server 2 4 Manually Configuring the SIP Parameters 2 7 Configuring Network Parameters 2 9 Configuring Network Parameters via a DHCP Server 2 10 Manually Configuring the Network Parameters 2 10 Connecting the Phone 2 11 Adjusting the Placement of the Cisco SIP Phone 2 12 Verifying Startup 2 14 Using the Cisco SIP IP Phone Menu Interface 2 15 Reading the Cisco SIP IP...

Page 5: ...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 3 46 Performing an Image Upgrade and Remote Reboot 3 46 A P P E N D I X A SIP Compliance with RFC 3261 Information A 1 SIP Functions A 1 SIP Methods A 2 SIP Responses A 2 1xx ...

Page 6: ... SIP IP Phone to Cisco SIP IP Phone Network Call Forwarding No Answer B 39 Cisco SIP IP Phone to Cisco SIP IP Phone Three Way Calling B 42 Call Flow Scenarios for Failed Calls B 46 Gateway to Cisco SIP IP Phone Called User Is Busy B 46 Gateway to Cisco SIP IP Phone Called User Does Not Answer B 48 Gateway to Cisco SIP IP Phone Client Server or Global Error B 50 Cisco SIP IP Phone to Cisco SIP IP P...

Page 7: ...Contents v Cisco SIP IP Phone Administrator Guide SELV Circuit Warning D 2 Circuit Breaker 15A Warning D 3 G L O S S A R Y I N D E X ...

Page 8: ...Contents vi Cisco SIP IP Phone Administrator Guide ...

Page 9: ...on 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 Who Should Use This Guide Network engine...

Page 10: ...or the Cisco IP Phone 7960 7940 and 7910 Series Installing the Wall Mount Kit for the Cisco IP Phone Table 1 Document Organization Section Title Description Chapter 1 Product Overview Describes SIP and the Cisco SIP IP phone Chapter 2 Getting Started with Your Cisco SIP IP Phone Describes how to install connect and configure the Cisco SIP IP phone Chapter 3 Managing Cisco SIP IP Phones Describes h...

Page 11: ... the system displays are in screen font Information you must enter is in boldface screen font Note Means reader take note Notes contain helpful suggestions or references to material not covered in the publication Caution Means reader be careful In this situation you might do something that could result in equipment damage or loss of data Warning This warning symbol means danger You are in a situat...

Page 12: ...standard per la prevenzione di incidenti La traduzione delle avvertenze riportate in questa pubblicazione si trova nell appendice Translated Safety Warnings Traduzione delle avvertenze di sicurezza Advarsel Dette varselsymbolet betyr fare Du befinner deg i en situasjon som kan føre til personskade Før du utfører arbeid på utstyr må du være oppmerksom på de faremomentene som elektriske kretser inne...

Page 13: ...subscription Ordering Documentation Cisco documentation is available in the following ways Registered Cisco Direct Customers can order Cisco product documentation from the Networking Products MarketPlace http www cisco com cgi bin order order_root pl Registered Cisco com users can order the Documentation CD ROM through the online Subscription Store http www cisco com go subscription Nonregistered ...

Page 14: ... that provides a broad range of features and services to help you to Streamline business processes and improve productivity Resolve technical issues with online support Download and test software packages Order Cisco learning materials and merchandise Register for online skill assessment training and certification programs You can self register on Cisco com to obtain customized information and ser...

Page 15: ...er http www cisco com register If you cannot resolve your technical issues by using the Cisco TAC Web Site and you are a Cisco com registered user you can open a case online by using the TAC Case Open tool at the following URL http www cisco com tac caseopen If you have Internet access it is recommended that you open P3 and P4 cases through the Cisco TAC Web Site Cisco TAC Escalation Center The Ci...

Page 16: ...xiv Cisco SIP IP Phone Administrator Guide Preface Obtaining Technical Assistance ...

Page 17: ...ion name mapping and call redirection Determine the media capabilities of the target endpoint Via Session Description Protocol SDP SIP determines the lowest level of common services between the endpoints Conferences are established using only the media capabilities that can be supported by all endpoints Determine the availability of the target endpoint If a call cannot be completed because the tar...

Page 18: ... and that returns a response on behalf of the user Typically a SIP endpoint is capable of functioning as both a UAC and a UAS but functions only as one or the other per transaction Whether the endpoint functions as a UAC or a UAS depends on the UA that initiated the request From an architecture standpoint the physical components of a SIP network can also be grouped into two categories clients and ...

Page 19: ... requests strips out the address in the request checks its address tables for any other addresses that may be mapped to the one in the request and then returns the results of the address mapping to the client Basically redirect servers provide the client with information about the next hop or hops that a message should take and then the client contacts the next hop server or UAS directly Registrar...

Page 20: ...creen instructing you how to scroll up and down on the LCD On screen mode buttons Retrieves information about current settings recent calls available services and voice mail messages Volume buttons Adjust the volume of the handset headset speaker and ringer and adjust the brightness contrast settings on the LCD screen Function toggles Includes these options Headset and speaker Toggles these functi...

Page 21: ...pport the XML objects added in Cisco CallManager XML version 3 1 CiscoIPPhoneIconMenu CiscoIPPhoneExecute CiscoIPPhoneError CiscoIPPhoneResponse SoftKeyItem The following exceptions apply to the Cisco SIP IP phone External directories cannot be appended to the main list of directories under the directory button If external directories are provisioned for the Cisco SIP IP phone then they can be acc...

Page 22: ...ne Hearing aid compatible handset Headset compatibility Integrated two port Ethernet switch that allows the telephone and a computer to share a single Ethernet jack Direct connection to a 10BASE T or 100BASE T Ethernet RJ 45 network half or full duplex connections are supported Large 4 25 x 3 in or 10 79 cm x 7 62 cm display with adjustable contrast Network Features IP address assignment Dynamic H...

Page 23: ... is not completely advertised in the initial call setup Support for endpoints specified as fully qualified domain names FQDNs in the SDP Remote reset and dial plan update support via the Event header in NOTIFY messages Note See the Supported Protocols section on page 1 11 for additional supported protocols Dialing and Messaging Features Dial plan support that enables automatic dialing and automati...

Page 24: ...ree way conference call see the Making Conference Calls section in Chapter 3 of the Cisco IP Phone Models 7960 and 7940 User Guide Call Options Call forward network Allows the Cisco SIP IP phone user to request forwarding service from the network 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 d...

Page 25: ... DNS SRV 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 The addition of the underscore reduces the risk of the same...

Page 26: ...h fr Spanish es Catalan ca Basque eu Portuguese pt Italian it Albanian sq Rhaeto Romanic rm Dutch nl German de Danish da Swedish sv Norwegian no Finnish fi Faroese fo Icelandic is Irish ga Scottish gd English en Afrikaans af and Swahili sw The following languages are not supported Zulu zu and other Bantu languages using Latin Extended B letters Arabic in North Africa and Guarani gn missing GEIUY w...

Page 27: ...ragmentation and reassembly and security The Cisco SIP IP phone supports IP as it is defined in RFC 791 Real Time Transport Protocol RTP Transports real time data such as voice data over data networks RTP also has the ability to obtain quality of service QoS information The Cisco SIP IP phone supports RTP as a media channel Session Description Protocol SDP An ASCII based protocol that describes mu...

Page 28: ...more information about configuring SIP VoIP refer to the Configuring SIP for VoIP chapter VoIP gateways are configured for SIP A TFTP server is active and contains the latest Cisco SIP IP phone firmware image in its root directory A proxy server is active and configured to receive and forward SIP messages Cisco SIP IP Phone Connections The Cisco SIP IP phone has connections for connecting to the d...

Page 29: ...le WS X6348 RJ45V 10 100 switching module Provides inline power to the Cisco SIP IP phone when connected to a Catalyst 3500 4000 or 6000 family 10 100BASE TX switching module This module sends power on pins 1 and 2 and 3 and 6 WS PWR PANEL Power patch panel provides power to the Cisco SIP IP phone which allows the Cisco SIP IP phone to be connected to existing Catalyst 4000 5000 and 6000 family 10...

Page 30: ...in network connectivity The Cisco SIP IP phone has an internal Ethernet switch which enables it to switch traffic coming from the phone access port and the network port If a computer is connected to the access port packets traveling to and from the computer and to and from the phone share the same physical link to the switch and the same port on the switch This configuration has these implications...

Page 31: ...1 15 Cisco SIP IP Phone Administrator Guide Chapter 1 Product Overview The Cisco SIP IP Phone with a Catalyst Switch http www cisco com univercd home home htm ...

Page 32: ...1 16 Cisco SIP IP Phone Administrator Guide Chapter 1 Product Overview The Cisco SIP IP Phone with a Catalyst Switch ...

Page 33: ...ng network connectivity and for making the phone operational in your IP network Once you connect your phone to the network and to an electrical supply the phone begins its initialization process During the initialization process the following events take place 1 The stored image is loaded The Cisco SIP IP phone has nonvolatile Flash memory in which it stores the firmware images user defined prefer...

Page 34: ...performing a firmware upgrade the phone downloads the firmware image from the TFTP server programs the image into Flash memory and reboots Installing the Cisco SIP IP Phone This section contains information on how to install Cisco SIP IP phones in your IP network Before getting started read over the information in this section carefully Installation Task Summary To successfully install the Cisco S...

Page 35: ...e see the Creating the Default SIP Configuration File section on page 2 5 SIPConfigGeneric cnf Required File that can be used as a template to configure SIP parameters specific to a phone When customized for a phone this file must be renamed to the MAC address of the phone RINGLIST DAT Optional Lists audio files that are the custom ring type options for the phones The audio files listed in the RIN...

Page 36: ...SIP parameters via a TFTP server you must use configuration files There are two configuration files that you can use to define the SIP parameters the default configuration file optional and the phone specific configuration file required If you choose to use a default configuration file you must store the file in the root directory of your TFTP server Phone specific configuration files can be store...

Page 37: ...ish the end of a line using lf or cr lf The variable and value must be on the same line and cannot break the line Except for parameters used to defined the lines and users on a phone all other SIP parameters can be defined in either the default configuration file or the phone specific configuration file However for network control and maintenance purposes Cisco recommends that you define the param...

Page 38: ...lowing is an example of a SIP default configuration file sip default configuration file Image Version image_version P0S3 xx y zz Proxy server address proxy1_address 192 168 1 1 Subdirectory config file location tftp_cfg_dir tftpboot configs sipphone Creating the Phone Specific SIP Configuration File In the phone specific SIP configuration file define the parameters that are specific to a phone suc...

Page 39: ...MAC address of the phone The MAC address must be in uppercase and the extension cnf must be in lowercase for example SIP00503EFFD842 cnf The following is an example of a configuration file phone specific configuration file sample Line 1 phone number line1_name 5551212 Line 1 name for authentication with proxy server line1_authname 5551212 Line 1 authentication name password line1_password password...

Page 40: ...the value for the linex_name parameter is the email address username company com you can specify the username to have just the user name appear on the LCD instead This parameter is used for display purposes only If a value is not specified for this parameter the value in the Name variable is displayed Authentication Name Required when registration is enabled Name used by the phone for authenticati...

Page 41: ...e to operate in an IP network You can configure the required network parameters via DHCP or manually configure them after you have connected the phone to a power supply The following parameters must be defined for your phone to establish network connectivity Phone s IP address Subnet mask Default gateway for the subnet use 0 0 0 0 if not required Domain name DNS server IP address use 0 0 0 0 if no...

Page 42: ... Connecting the Phone section on page 2 11 Unlock configuration mode as described in the Unlocking Configuration Mode section on page 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 se...

Page 43: ...ters 2 through 5 are the IP addresses of the gateways that the phone attempts to use as an alternate gateway if the primary gateway is not available Domain Name Name of the DNS domain in which the phone resides DNS servers 1 through 5 IP address of the DNS server used by the phone to resolve names to IP addresses The phone attempts to use DNS servers 2 through 5 if DNS server 1 is unavailable Step...

Page 44: ...k device such as a desktop computer to the access port on the phone optional See the Connecting to the Network section on page 1 13 for more information on the access port Step 4 Connect the power plug to the Cisco AC adapter port optional See the Connecting to Power section on page 1 13 for more information Adjusting the Placement of the Cisco SIP Phone The Cisco SIP IP phone includes an adjustab...

Page 45: ...t for the Cisco IP Phone document Before You Begin Mounting the Cisco SIP IP phone on the wall requires some tools and equipment that are not provided as standard equipment Following are the tools and parts required for a typical Cisco SIP IP phone installation Screwdriver Screws to secure the Cisco SIP IP phone to the wall Procedure Step 1 Push in the footstand adjustment knob Step 2 Adjust the f...

Page 46: ...CP server to obtain network parameters and the IP address of the TFTP server Requesting Configuration The phone is contacting the TFTP server to request its configuration files and compare firmware images Upgrading Software The Upgrade Software message displays only if the phone has determined that an image upgrade is required After upgrading the image the phone automatically reboots to run the ne...

Page 47: ...bers on the dial pad associated with a particular letter For example the 2 key has the letters A B and C For a lowercase a press the 2 key once To scroll through the available letters and numbers press the key repeatedly Press the soft key to delete any mistakes When configuring an network IP address or ID parameter Use the buttons on the dial pad to enter a new value Press the soft key to delete ...

Page 48: ...igured for URL dialing and is ready for you to place the call When a line is configured for URL dialing you can enter both numbers and letters when placing the call You can change to E 164 number dialing at any time while dialing on a line by pressing the Number soft key The character x displayed to the right of the icon indicates that registration has failed The Cisco SIP IP phone configuration m...

Page 49: ...phone to support automatic dialing and automatic generation of a secondary dial tone If a single dial plan is to be used for a system of phones the dial plan is best specified in the default configuration file However you can create multiple dial plans and specify which phones are to use which dial plan by defining the dial_template parameter in the phone specific configuration file If one phone i...

Page 50: ... a timeout occurs and the number is dialed as entered by the user To have the number dial immediately specify 0 User type is the either IP or Phone Enter User phone or User IP to have the tag automatically added to the dialed number This entry is not case sensitive Rewrite xxx is the alternate string to be dialed instead of what the user enters The rewrite rules are matched from start to finish wi...

Page 51: ... in the phone specific configuration file If the dial plan applies to a system of phones add the path to the dial plan via the dial_template parameter in the default configuration file For more information on defining the dial_template parameter see the Modifying the Phone s SIP Settings section on page 3 5 The following is an example of a North American dial plan DIALTEMPLATE TEMPLATE MATCH 0 Tim...

Page 52: ...2 20 Cisco SIP IP Phone Administrator Guide Chapter 2 Getting Started with Your Cisco SIP IP Phone Creating Dial Plans ...

Page 53: ...first follow the instructions in the Entering Configuration Mode section on page 3 2 Edit the default and phone specific configuration files on the TFTP server See the Modifying SIP 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 ...

Page 54: ...ess Note You have activated the configuration mode for your phone There is no indication that an action has taken place 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 Configu...

Page 55: ...ernate TFTP server This field enables an administrator to specify the remote TFTP server rather than the local one Possible values for this parameter are Yes and No The default is No When Yes is specified the IP address in the TFTP Address parameter must be changed to the address of the alternate TFTP server Default Routers 1 through 5 Yes but DHCP must be disabled IP address of the default gatewa...

Page 56: ...server After initially querying the default TFTP server the phone will re request the default and MAC specific configuration files from the new TFTP server The dynamic TFTP server is not stored in Flash memory Erase Configuration Yes Whether to erase all of the locally defined network settings on the phone and reset the values to the defaults Selecting Yes reenables DHCP For more information on er...

Page 57: ...lex 100 MB connection 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 Network Port 2 Device Type Yes The device type that is connected to port 2 of the phone Valid values are Hub Switch default PC Note If the value is PC port 2 can be connected only to a...

Page 58: ...nu Table 3 2 SIP Parameters Summary Configuration File SIP Configuration Menu Network Configuration Menu Call Preferences Time Date anonymous_call_block NA NA Anonymous Call Block NA autocomplete NA NA Auto Complete Numbers NA callerid_blocking NA NA Caller ID Blocking NA call_waiting NA NA Call Waiting NA cnf_join_enable NA NA NA NA date_format NA NA NA Date Format dial_template NA NA NA NA dnd_c...

Page 59: ... NA NA outbound_proxy Outbound Proxy NA NA NA outbound_proxy_port Outbound Proxy Port NA NA NA phone_label Phone Label NA NA NA phone_password NA NA NA NA phone_prompt NA NA NA NA preferred_codec Preferred Codec NA NA NA proxy_backup Backup Proxy NA NA NA proxy_backup_port Backup Proxy Port NA NA NA proxy_emergency Emergency Proxy NA NA NA proxy_emergency_port Emergency Proxy Port NA NA NA proxy_r...

Page 60: ...ou can then define only those parameters that are specific to a phone in the phone specific configuration file Phone specific parameters should be defined only in a phone specific configuration file or should be manually configured Phone specific parameters should not be defined in the default configuration file Modifying the Default SIP Configuration File In the default configuration file SIPDefa...

Page 61: ...ll Blocking feature is disabled by default but can be turned on and off via the phone s user interface When disabled anonymous calls are received 1 The Anonymous Call Blocking feature is enabled by default but can be turned on and off via the phone s user interface When enabled anonymous calls are rejected 2 The Anonymous Call Blocking feature is disabled permanently and cannot be turned on and of...

Page 62: ... can be turned on and off via the phone s user interface When enabled call waiting calls are accepted 2 The call waiting feature is disabled 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 3 The call waiting feature is enabled permanently and cannot be turned on and off local...

Page 63: ...ng feature is disabled 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 3 The Caller ID Blocking feature is enabled permanently and cannot be turned on and off locally via the phone s user interface If specifying this value specify this parameter in the phone specific configur...

Page 64: ...e calls in the Missed Calls directory Valid values are 0 The Do Not Disturb feature is off by default but can be turned on and off locally via the phone s user interface 1 The Do Not Disturb feature is on by default but can be turned on and off locally via the phone s user interface 2 The Do Not Disturb feature is off permanently and cannot be turned on and off locally via the phone s user interfa...

Page 65: ... nominal 3 nominal 4 3 db above nominal 5 6 db above nominal The default is 3 dtmf_inband Optional Whether to detect and generate in band signaling format Valid values are 1 generate DTMF digits in band and 0 do not generate DTMF digits in band The default is 1 dtmf_outofband Optional Whether to generate the out of band signaling for tone detection on the IP side of a gateway and if so when The Ci...

Page 66: ...the IP address of a new dynamic TFTP server After initially querying the default TFTP server the phone will re request the default and MAC specific configuration files from the new TFTP server The dynamic TFTP server is not stored in Flash memory The number of dyn_tftp_addr values supported by the phone is limited to prevent the phone from bouncing between two TFTP servers Only dotted IP addresses...

Page 67: ... PhoneImage objects are not supported for this parameter Using anything other than a Windows bitmap bmp file can cause unpredictable results messages_uri Optional Number to call to check voice mail This number is called when the Messages key is pressed nat_address Optional The WAN IP address of the Network Address Translation NAT or firewall server You can use either a dotted IP address or a DNS n...

Page 68: ...d Full100 Port is configured to be a full duplex 100 MB connection 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 The default is Auto network_port2_type Optional The device type that is connected to port 2 of the phone Valid values are Hub Switch default P...

Page 69: ...eived tag If there is no received tag and the IP address in the uppermost Via header is different than the source IP address the response is sent back to the source IP Otherwise the response is sent back to the IP address in the uppermost Via header phone_password Optional Password to be used for console or Telnet access The default password is cisco phone_prompt Optional Prompt to be displayed wh...

Page 70: ...hat will be used by the phones Enter this address in IP dotted decimal notation proxy1_port Optional Port number of the primary SIP proxy server This is the port on which the SIP client listens for messages The default is 5060 Note For additional phone lines proxyN_address and proxyN_port parameters can be used to assign different proxy addresses to different phone lines N in the parameters repres...

Page 71: ..._url Optional URL of the services BTXML files This URL is accessed when the Services button is pressed For example use services_url http 10 10 10 10 CiscoServices Services asp sip_invite_retx Optional Maximum number of times an INVITE request will be retransmitted The valid value is any positive integer The default is 6 sip_retx Optional Maximum number of times a SIP message other than an INVITE r...

Page 72: ...rmat is displayed and cannot be changed to a 24 hour format via the phone s user interface 3 The 24 hour format is displayed and cannot be changed to a 12 hour format via the phone s user interface The default value is 1 time_zone Optional See the Setting the Date Time and Daylight Saving Time section on page 3 36 section for more information timer_invite_expires Optional The amount of time in sec...

Page 73: ...sitive integer greater than timer_t1 The default is 4000 tos_media Optional Type of service ToS level for the media stream being used Valid values are 0 IP_ROUTINE 1 IP_PRIORITY 2 IP_IMMEDIATE 3 IP_FLASH 4 IP_OVERIDE 5 IP_CRITIC The default is 5 user_info Optional Configures the user parameter in the REGISTER message Valid values are none No value is inserted phone The value user phone is inserted...

Page 74: ... dtmf_inband 1 Out of band DTMF Settings none disable avt avt enable default avt_always always avt dtmf_outofband avt DTMF dB Level Settings 1 6dB down 2 3db down 3 nominal default 4 3db up 5 6dB up dtmf_db_level 3 SIP Timers timer_t1 500 Default 500 msec timer_t2 4000 Default 4 sec sip_retx 10 Default 11 sip_invite_retx 6 Default 7 timer_invite_expires 180 Default 180 sec Setting for Message spee...

Page 75: ...4hr default time_format_24hr 1 Enable or Disbale VAD 0 disabled default 1 enabled enable_vad 0 telnet_level 0 phone_password cisco services_url http www company com phone services asp directory_url http www company com phone companydirectory asp logo_url http www company com phone logo bmp Modifying the Phone Specific SIP Configuration File In the phone specific SIP configuration file maintain tho...

Page 76: ...r this parameter the value in the linex_name variable is displayed linex_authname Required for line 1 when registration is enabled and the proxy server requires authentication Name used by the phone for authentication if a registration is challenged by the proxy server during initialization If a value is not configured for the linex_authname parameter for a line when registration is enabled the va...

Page 77: ...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 3 The Do Not Disturb feature is on permanently and cannot be turned on and off locally via the phone s user interface This setting sets the phone to be a call out phone only If specifying this value specify this parameter in the phone specific ...

Page 78: ...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 Line parameters those identified as linex define a line on the phone If you configure a line to use an e mail address that line can be called...

Page 79: ... linex_name parameter is the e mail address username company com you can specify the username to have just the user name appear on the LCD instead This parameter is used for display only If a value is not specified for this parameter the value in the Name variable is displayed Authentication Name Required when registration is enabled Name used by the phone for authentication if a registration is c...

Page 80: ...itialization Select the Yes soft key to enable registration during initialization The default is No After a phone has initialized and registered with a proxy server changing the value of this parameter to No unregisters the phone from the proxy server To reinitiate a registration change the value of this parameter back to Yes Note If you enable registration and authentication is required you must ...

Page 81: ...s address in IP dotted decimal notation Backup Proxy Port Optional Port number of the backup proxy server Default is 5060 Emergency Proxy Optional IP address of the emergency proxy server or gateway Enter this address in IP dotted decimal notation Emergency Proxy Port Optional Port number of the emergency proxy Default is 5060 Outbound Proxy Optional The IP address of the outbound proxy server You...

Page 82: ...hen used with the following keywords arp Shows debug output for the ARP cache console stall Shows debug output for the console stall driver output mode strlib Shows debug output for the string library malloc Shows debug output for memory allocation malloc table Enables the population of the memory allocation table The table can be viewed with the show malloc table command sk platform Shows debug o...

Page 83: ...ests and responses xml events Shows XML events that are posted to the XML application chain xml deck Shows XML requests for XML cards and decks xml vars Shows XML content variables xml post Shows XML post strings Note Do not use the debug all command because it can cause the phone to become inoperable This command is for use only by Cisco TAC personnel SIP Phone dns Manipulates the DNS system The ...

Page 84: ... size of the packet you can send any size packet up to 1480 bytes and the default packet size is 100 The timeout value is measured in seconds and identifies how long to wait before the request times out the default is 2 SIP Phone register option line Instructs the Cisco SIP IP phone to register with the proxy server Option values are 0 and 1 0 is unregister and 1 is register These values are set f...

Page 85: ...stacks Shows tasks and buffer lists status Shows the current phone status including errors abort_vector Shows the address of the last recorded abort vector flash Shows flash memory information dspstate Shows the DSP status including whether the DSP is ready the audio mode if keepalive pending is turned on and the ringer state rtp Shows packet statistics for the RTP streams tcp Shows the status of ...

Page 86: ...ne MAC address domain name and phone name config Shows the current Flash configuration including network information phone label and password SNTP server address DST information time and date format and input and output port numbers personaldir Displays the current contents of the personal directory This command can be used only if the telnet_level parameter is set to allow privileged commands to ...

Page 87: ...dir Directories set Settings navup Navigate up navdn Navigate down The keys 0 through 9 and may be entered in continuous strings to better express typical dialing strings A typical command would be test ky 23234 test onhook Simulates a handset onhook event test offhook Simulates a handset offhook event test show Shows test feedback test hide Hides test feedback SIP Phone tty echo on off mon timeou...

Page 88: ...er the following Review the guidelines and restrictions documented in the Configuration File Guidelines section on page 2 4 Determine whether you want to configure absolute DST or relative DST The SNTP parameters specify how the phone will obtain the current time from an SNTP server Review the guidelines in Table 3 8 and Table 3 9 before configuring the SNTP parameters Table 3 8 lists the actions ...

Page 89: ...r being set as the one who first responded SNTP packet to the local network broadcast address After the first SNTP response is received the phone switches to multicast mode Receives Nothing No known server with which to communicate SNTP data via the SNTP NTP multicast address from the local network broadcast address from any server on the network Unicast SNTP data from the SNTP server that first r...

Page 90: ...on YST Yukon Standard Time PST GMT 08 00 Los Angeles PST Pacific Standard Time MST GMT 07 00 Phoenix MST Mountain Standard Time PDT Pacific Daylight Time CST GMT 06 00 Dallas Mexico City CST Central Standard Time MDT Mountain Daylight Time Chicago EST GMT 05 00 New York EST Eastan Standard Time CDT Central Daylight Time NYC AST GMT 04 00 La Paz AST Atlantic Standard Time EDT Eastan Daylight Time N...

Page 91: ...ot case sensitive 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 BT GMT 03 00 Baghdad Moscow BT Baghdad Time USSR zone2 IT GMT 03 30 Tehran IT Iran Time ZP4 GMT 04 00 Abu Dhabi USSR zone3 ZP4 GMT Plus 4 Hours AFG GMT 04 30 Kabul Afghanistan ZP5 GMT 05 0...

Page 92: ...case sensitive In the United States the default value is Sunday dst_start_week_of_month Week of month in which DST begins Valid values are 1 through 6 and 8 with 1 being the first week and each number thereafter being subsequent weeks and 8 specifying the last week in the month regardless of which week the last week is In the United States the default value is 1 dst_stop_day_of_week Day of the wee...

Page 93: ...een configured in the phone Erasing the Locally Defined Network Settings When you erase the locally defined settings the values are reset to the defaults Before You Begin Unlock configuration mode as described in the Unlocking Configuration Mode section on page 3 2 If DHCP has been disabled on a phone clearing the phone s settings reenables it Select the Erase Config parameter by pressing the down...

Page 94: ... Settings menu appears Step 2 Highlight SIP Configuration Step 3 Press the Select soft key The SIP Configuration settings are displayed Step 4 Highlight the parameter for which you want to erase the setting Step 5 Press the Edit soft key Step 6 Press the soft key to delete the current value Step 7 Press the Validate soft key to save your change and exit the Edit panel Step 8 If modifying a line pa...

Page 95: ...r of packets received by the phone not through the switch Xmit Number of packets sent by the phone not through the switch REr Number of packets received by the phone that contained errors BCast Number of broadcast packets received by the phone Phone State Message TCP messages indicating the state of the phone Possible messages are Phone Initialized TCP connection has not gone down since the phone ...

Page 96: ... Step 2 Highlight Status Step 3 Press the Select soft key The Setting Status menu appears Step 4 Highlight Firmware Versions Step 5 Press the Select soft key The Firmware Versions panel appears The following information is displayed on this panel Application Load ID Current software image on the phone Boot Load ID Bootstrap loader image version that is manufactured on the phone This image name doe...

Page 97: ... the image defined in the configuration file which is stored in the root directory on the TFTP server Once the new image has been downloaded the phone programs that image into Flash memory and then reboots Note If you do not define the image_version parameter in the default configuration file only phones that have an updated phone specific configuration file with the new image version and that hav...

Page 98: ...P server Step 2 If you are dual booting from a Cisco IP phone running the SCCP or MGCP protocol open the OS79XX TXT file with a text editor and change the file to include P0S30202 Step 3 Copy the new Release 4 0 binary image P0S3 xx y zz bin where xx is the release major version y is the release minor version and zz is the maintenance number from Cisco com to the root directory of the TFTP server ...

Page 99: ...dim ipaddress To sip lineX_name ipaddress Event check sync Date Mon 10 Jul 2000 16 28 53 0700 Call ID 1349882 ipaddress CSeq 1300 NOTIFY Contact sip webadmin ipaddress Content Length 0 After the remote reboot process is initiated on the phone via the NOTIFY message the following actions take place 1 If the phone is currently in an idle state the phone waits 20 seconds and then contacts the TFTP se...

Page 100: ...3 48 Cisco SIP IP Phone Administrator Guide Chapter 3 Managing Cisco SIP IP Phones Performing an Image Upgrade and Remote Reboot ...

Page 101: ...contains compliance information on the following SIP Functions page A 1 SIP Methods page A 2 SIP Responses page A 2 SIP Header Fields page A 7 SIP Session Description Protocol SDP Usage page A 8 Transport Layer Protocols page A 9 SIP Security page A 9 SIP DTMF Digit Transport page A 9 SIP Functions Function Supported User agent client UAC Yes User agent server UAS Yes Proxy server Third party only...

Page 102: ...esponse Global Responses page A 7 1xx Response Information Responses Method Supported Comments INVITE Yes The Cisco SIP IP phone supports mid call changes such as putting a call on hold as signaled by a new INVITE that contains an existing call ID ACK Yes None OPTIONS Response only BYE Yes CANCEL Yes REGISTER Yes The Cisco SIP IP phone supports both user and device registration REFER Yes None NOTI...

Page 103: ... Upon receiving this response the phone provides early media cut through and then waits for a 200 OK response 1xx Response Supported Comments 2xx Response Supported Comments 200 OK Yes None 202 Accepted Yes None 3xx Response Supported Comments 300 Multiple Choices Yes None 301 Moved Permanently Yes 302 Moved Temporarily Yes The Cisco SIP IP phone does not generate this response at this time Upon r...

Page 104: ...e user of the response This response indicates that the SIP server has the request but will not provide service 404 Not Found Yes 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 405 Method Not Allowed See comments This response is received only in this release If the phone receives a 405 Method Not Allowed...

Page 105: ... response is received by the phone only in this release This response indicates that the user refuses to accept the request without a defined content length If received the phone resends the INVITE request if it can add a valid Content Length header field 413 Request Entity Too Large See comments This response is received only by the phone in this release If a retry after header field is contained...

Page 106: ...omments This response is received only by the phone in this release If a new contact is received the phone might reinitiate the call 486 Busy Here Yes The Cisco SIP IP phone generates this response if the called party is off hook and the call cannot be presented as a call waiting call Upon receiving this response the phone notifies the user and generates a busy tone 487 Request Canceled Yes This r...

Page 107: ...ses For an incoming response the SIP IP phone initiates a graceful call disconnect 603 Decline 604 Does Not Exist Anywhere 606 Not Acceptable Header Field Supported Accept Yes Accept Encoding Yes Accept Language Yes Allow Yes Also Yes Authorization Yes Call ID Yes Contact Yes Content Encoding Yes Content Length Yes Content Type Yes Cseq Yes Date Yes Encryption No Expires Yes From Yes Hide No Max F...

Page 108: ...es Referred By Yes Referred To Yes Remote Party ID Yes Replaces Yes Requested By Yes Require Yes Response Key No Retry After Yes Route Yes Server Yes Subject No Timestamp Yes To Yes Unsupported Yes User Agent Yes Via Yes Warning Yes WWW Authenticate Yes SDP Headers Supported v Protocol version Yes o Owner or creator and session identifier Yes s Session name Yes t Time description Yes c Connection ...

Page 109: ... Records Usage SIP DTMF Digit Transport m Media name and transport address Yes a Media attribute lines Yes Protocol Supported Unicast UDP Yes Multicast UDP No TCP No Basic Authentication No Digest Authentication Yes Proxy Authentication No PGP No DNS Resource Record Type Supported Type A Yes Type SRV Yes Transport Type Supported RFC 2833 Yes In band tones Yes SDP Headers Supported ...

Page 110: ...A 10 Cisco SIP IP Phone Administrator Guide Appendix A SIP Compliance with RFC 3261 Information SIP DTMF Digit Transport ...

Page 111: ...listed in the To header field with a SIP server 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 SIP 2xx Successful Responses SIP 3xx Re...

Page 112: ...page B 30 Cisco SIP IP Phone to Cisco SIP IP Phone Network Call Forwarding Unconditional page B 35 Cisco SIP IP Phone to Cisco SIP IP Phone Network Call Forwarding Busy page B 37 Cisco SIP IP Phone to Cisco SIP IP Phone Network Call Forwarding No Answer page B 39 Cisco SIP IP Phone to Cisco SIP IP Phone Three Way Calling page B 42 Gateway to Cisco SIP IP Phone Successful Call Setup and Disconnect ...

Page 113: ... 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 A unique numeric identifier is assigned to the call and is inserted in the Call ID field The transaction number within a single call leg is identified in the CSeq field The media capability User A is ready to rec...

Page 114: ...nse from the Cisco SIP IP phone User A hears the ringback tone that indicates that User B is being alerted 7 200 OK Cisco SIP IP phone to Gateway 1 The Cisco SIP IP phone sends a SIP 200 OK response to Gateway 1 The 200 OK response notifies Gateway 1 that the connection has been made 8 Connect Gateway 1 to PBX A Gateway 1 sends a Connect message to PBX A The Connect message notifies PBX A that the...

Page 115: ...ich 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 PBX A is identified as the call session initiator in the From field A unique numeric identifier is assigned to the call and is inserted in the Call ID field The transaction number within a single call leg is identified in the CSeq field The media capabi...

Page 116: ...ponse to Gateway 1 The 200 OK response notifies Gateway 1 that the connection has been made 8 Connect Gateway 1 to PBX A Gateway 1 sends a Connect message to PBX A The Connect message notifies PBX A that the connection has been made 9 ACK Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP ACK to the Cisco SIP IP phone The ACK confirms that User A has received the 200 OK response The call sessio...

Page 117: ...ure B 3 Cisco SIP IP Phone to Cisco SIP IP Phone Simple Call Hold IP IP 2 180 RINGING 3 200 OK 2 way RTP channel 4 ACK A is taken off hold The RTP channel between A and B is reestablished A is on hold The RTP channel between A and B is torn down 5 INVITE c IN IP4 0 0 0 0 6 200 OK 7 ACK 8 INVITE c IN IP4 IP User B 9 200 OK 10 ACK 1 INVITE B IP Network SIP IP Phone User A SIP IP Phone User B 41465 ...

Page 118: ...se to Cisco SIP IP phone A 3 200 OK Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A The 200 OK response notifies Cisco SIP IP phone A that the connection has been made If Cisco SIP IP phone B supports the media capability advertised in the INVITE message sent by Cisco SIP IP phone A it advertises the intersection of its own and ...

Page 119: ...e B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down 8 INVITE Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a mid call INVITE to Cisco SIP IP phone A wit...

Page 120: ...ay RTP channel A is put on hold The RTP channel between A and B is torn down 4 ACK 2 way RTP channel B is disconnected from C A is taken off hold The RTP channel between A and B is reestablished 5 INVITE c IN IP4 0 0 0 0 6 200 OK 7 ACK 14 INVITE c IN IP4 IP User B 15 200 OK 16 ACK 1 INVITE B 9 180 Ringing 10 200 OK 8 INVITE C 13 200 OK 12 BYE 11 ACK IP Network SIP IP Phone User A SIP IP Phone User...

Page 121: ...80 Ringing response to Cisco SIP IP phone A 3 200 OK Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A The 200 OK response notifies Cisco SIP IP phone A that the connection has been made If Cisco SIP IP phone B supports the media capability advertised in the INVITE message sent by Cisco SIP IP phone A it advertises the intersectio...

Page 122: ...eader field 11 ACK Cisco SIP IP phone B to Cisco SIP IP phone C Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone C The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone C The ACK might contain a message body with the final session description to be used by Cisco SIP IP phone C If the message body of the ACK is empty Cisco SIP IP phone C uses ...

Page 123: ...via an IP network The call flow scenario is as follows 1 User A calls User B 2 User B answers the call 3 User C calls User B 4 User B accepts the call from User C 5 User B switches back to User A 6 User B hangs up ending the call with User A 7 User B is notified of the remaining call with User C 8 User B answers the notification and continues the call with User C 16 ACK Cisco SIP IP phone B to Cis...

Page 124: ...ins 4 ACK C is taken off hold The RTP channel between B and C is reestablished 2 way RTP channel C is on hold The RTP channel between B and C is torn down A is taken off hold The RTP channel between A and B is reestablished 7 INVITE c IN IP4 0 0 0 0 8 200 OK 9 ACK 15 INVITE c IN IP4 IP User B 16 200 OK 17 ACK 18 BYE 19 200 OK 1 INVITE B 11 ACK 13 200 OK 10 200 OK 21 200 OK 20 INVITE c IN IP4 IP Us...

Page 125: ...esponse to Cisco SIP IP phone A 3 200 OK Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A The 200 OK response notifies Cisco SIP IP phone A that the connection has been made If Cisco SIP IP phone B supports the media capability advertised in the INVITE message sent by Cisco SIP IP phone A it advertises the intersection of its own...

Page 126: ...dy with the final session description to be used by Cisco SIP IP phone B If the message body of the ACK is empty Cisco SIP IP phone B uses the session description in the INVITE request A two way RTP channel is established between Cisco SIP IP phone B and Cisco SIP IP phone C 12 INVITE Cisco SIP IP phone B to Cisco SIP IP phone C Cisco SIP IP phone B sends a mid call INVITE to Cisco SIP IP phone C ...

Page 127: ...es and then User B hangs up Cisco SIP IP phone B sends a SIP BYE request to Cisco SIP IP phone A The BYE request indicates that User B wants to release the call 19 200 OK Cisco SIP IP phone A to Cisco SIP IP phone B Cisco SIP IP phone A sends a SIP 200 OK message to Cisco SIP IP phone B The 200 OK response notifies Cisco SIP IP phone B that the BYE request has been received The call session betwee...

Page 128: ...SIP IP Phone User B IP IP 2 100 TRYING 3 180 RINGING 4 200 OK 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 5 ACK 2 way voice path User B presses blind transfer 6 INVITE c 0 0 0 0 7 200 OK 8 ACK 9 REFER Refer To C Referred By B 10 202 ACCEPTED 1 INVITE SIP IP Phone User A SIP IP Phone User C IP 62503 User B d...

Page 129: ... a SIP 100 Trying response to Cisco SIP IP phone A The 100 Trying response indicates that the INVITE request has been received by Cisco SIP IP phone B 3 180 Ringing Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A 4 200 OK Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco ...

Page 130: ... B sends a BYE message to Cisco SIP IP phone A This message indicates that Cisco SIP IP phone B will be disconnecting from the call 12 200 OK Cisco SIP IP phone A to Cisco SIP IP phone B Cisco SIP IP phone A sends a SIP 200 OK response to Cisco SIP IP phone B The 200 OK response notifies Cisco SIP IP phone B that the BYE message was received 13 INVITE Cisco SIP IP phone A to Cisco SIP IP phone C B...

Page 131: ...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 1 User A calls User B 2 User B answers the call 3 User B transfers the call to User C 18 NOTIFY Cisco SIP IP phone A to Cisco SIP IP phone B Cisco SIP IP phone A sends a NOTIFY message to Cisco SIP IP phone B The NOTIFY message notifies Cisco SIP IP phone C of the REFER ev...

Page 132: ...Also IP Network SIP IP Phone User B IP IP 2 100 TRYING 3 180 RINGING 4 200 OK 11 BYE Also C 12 200 OK 13 INVITE Requested By B 14 100 TRYING 15 180 RINGING 16 200 OK 17 ACK 2 way voice path 5 ACK 2 way voice path User B presses blind transfer 6 INVITE c 0 0 0 0 7 200 OK 8 ACK 9 REFER Refer To C Referred By B 10 501 NOT IMPLEMENTED 1 INVITE SIP IP Phone User A SIP IP Phone User C IP 62504 User B di...

Page 133: ... a SIP 100 Trying response to Cisco SIP IP phone A The 100 Trying response indicates that the INVITE request has been received by Cisco SIP IP phone B 3 180 Ringing Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A 4 200 OK Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco ...

Page 134: ...nd that Cisco SIP IP phone B should failover to Bye Also 11 BYE Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a BYE message to Cisco SIP IP phone A The BYE message includes the following information Also C This message indicates that the 501 Not Implemented message was received in response to a REFER message 12 200 OK Cisco SIP IP phone A to Cisco SIP IP phone B Cisco SIP...

Page 135: ...r 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 1 User A calls User B 2 User B answers the call 3 User B calls User C and User C consents to take the call 4 User B transfers the call to User C 5 User B disconnects with User C 6 User C and User A connect to each other 17 ACK Cisco SIP IP phone A to Cisco SIP IP ...

Page 136: ...11 180 RINGING 12 200 OK 13 ACK 2 way voice path 14 INVITE c 0 0 0 0 15 200 OK 16 ACK 17 REFER Refer To C Replaces B Referred By B 18 202 ACCEPTED 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 5 ACK 2 way voice path User B presses transfer 6 INVITE c 0 0 0 0 7 200 OK 8 ACK 9 INVITE C 10 100 TRYING 1 INVITE SIP...

Page 137: ...n received by Cisco SIP IP phone B 3 180 Ringing Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A 4 200 OK Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A The 200 OK response notifies Cisco SIP IP phone A that the connection has been made If Cisco SIP IP p...

Page 138: ...SIP IP phone A it advertises the intersection of its own and Cisco SIP IP phone A s media capability in the 200 OK response If Cisco SIP IP phone B does not support the media capability advertised by Cisco SIP IP phone A it sends back a 400 Bad Request response with a 304 Warning header field 13 ACK Cisco SIP IP phone B to Cisco SIP IP phone C Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP p...

Page 139: ...sends a SIP ACK to Cisco SIP IP phone C The ACK confirms that Cisco SIP IP phone A has received the 200 OK response from Cisco SIP IP phone C 22 BYE Cisco SIP IP phone C to Cisco SIP IP phone B Cisco SIP IP phone C sends a SIP BYE request to Cisco SIP IP phone B 23 200 OK Cisco SIP IP phone B to Cisco SIP IP phone C Cisco SIP IP phone B sends a SIP 200 OK message to Cisco SIP IP phone C The 200 OK...

Page 140: ...s contacts a third party and then that participant transfers the call to the third party This is called an attended 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 1 User A calls User B 2 User B answers the call 3 User B calls User C and User C consents t...

Page 141: ...NG 4 200 OK 11 180 RINGING 12 200 OK 13 ACK 2 way voice path 14 INVITE c 0 0 0 0 15 200 OK 16 ACK 17 REFER Refer To C Replaces B Referred By B 18 501 NOT IMPLEMENTED 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 5 ACK 2 way voice path User B presses transfer 6 INVITE c 0 0 0 0 7 200 OK 8 ACK 9 INVITE C 10 100 TRYI...

Page 142: ...n received by Cisco SIP IP phone B 3 180 Ringing Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 180 Ringing response to Cisco SIP IP phone A 4 200 OK Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A The 200 OK response notifies Cisco SIP IP phone A that the connection has been made If Cisco SIP IP p...

Page 143: ... SIP IP phone A it advertises the intersection of its own and Cisco SIP IP phone A s media capability in the 200 OK response If Cisco SIP IP phone B does not support the media capability advertised by Cisco SIP IP phone A it sends back a 400 Bad Request response with a 304 Warning header field 13 ACK Cisco SIP IP phone B to Cisco SIP IP phone C Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP ...

Page 144: ... phone B sends a SIP BYE request to Cisco SIP IP phone C 22 200 OK Cisco SIP IP phone C to Cisco SIP IP phone B Cisco SIP IP phone C sends a SIP 200 OK message to Cisco SIP IP phone B The 200 OK response notifies Cisco SIP IP phone B that the BYE request has been received 23 INVITE Cisco SIP IP phone A to Cisco SIP IP phone C Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone C ...

Page 145: ...cenario 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 1 User B requests that the network forward all calls to Cisco SIP IP phone C 2 User A calls User B 3 The network transfers the call to Cisco SIP IP phone C Figure B 10 Cisco SIP IP Phone to Cisco SIP IP Phone Network Call Forwarding Un...

Page 146: ...TE SIP proxy server to SIP redirect server SIP proxy server sends the SIP INVITE request to the SIP redirect server 3 302 Moved Temporarily SIP redirect server to SIP proxy server SIP redirect server sends a SIP 302 Moved temporarily message to the SIP proxy server The message indicates that User B is not available at SIP phone B and includes instructions to locate User B at Cisco SIP IP phone C 4...

Page 147: ...he 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 1 User B requests that if their phone Cisco SIP IP phone B is busy the network should forward incoming calls to Cisco SIP IP phone C 2 User A calls User B 3 User B s phone is busy 4 The network transfers the call to Cisco SIP IP phone C Figure ...

Page 148: ...irect server 3 300 Multiple Choices SIP redirect server to SIP proxy server SIP redirect server sends a SIP 300 Multiple choices message to the SIP proxy server The message indicates that User B can be reached either at SIP phone B or Cisco SIP IP phone C 4 INVITE SIP proxy server to Cisco SIP IP phone B SIP proxy server sends a SIP INVITE request to Cisco SIP IP phone B The INVITE request is an i...

Page 149: ...sco SIP IP phones which are connected via an IP network The call flow scenario is as follows 1 User B requests that if the phone Cisco SIP IP phone B is not answered within a set amount of time the network should forward incoming calls to Cisco SIP IP phone C 2 User A calls User B 3 User B s phone is not answered 4 The network transfers the call to Cisco SIP IP phone C 11 ACK Cisco SIP IP phone A ...

Page 150: ...o SIP IP Phone Network Call Forwarding No Answer IP IP 13 200 OK 7 180 Ringing 5 INVITE B 9 200 OK 12 200 OK 11 180 Ringing 15 ACK 10 INVITE C 8 CANCEL 6 180 Ringing 1 INVITE B 14 ACK 3 300 Multiple Choices 2 INVITE B 4 ACK IP Network SIP IP Phone User A SIP IP Phone User B SIP IP Phone User C Proxy Server Redirect Server IP 41473 2 way RTP channel ...

Page 151: ...t server sends a SIP 300 Multiple choices message to the SIP proxy server The message indicates that User B can be reached either at SIP phone B or Cisco SIP IP phone C 4 INVITE SIP proxy server to Cisco SIP IP phone B SIP proxy server sends a SIP INVITE request to Cisco SIP IP phone B The INVITE request is an invitation to User B to participate in a call session 5 180 Ringing Cisco SIP IP phone B...

Page 152: ...h are connected via an IP network The call flow scenario is as follows 1 User A calls User B 2 User B answers the call 3 User B puts User A on hold 4 User B calls User C 5 User C answers the call 6 User B takes User A off hold 13 ACK Cisco SIP IP phone A to SIP proxy server Cisco SIP IP phone A sends a SIP ACK to the SIP proxy server The ACK confirms that Cisco SIP IP phone A has received the 200 ...

Page 153: ... INVITE B Call ID 1 3 200 OK 2 180 Ringing 6 200 OK 12 INVITE A Call ID 1 c IN IP4 IP User B 13 200 OK 7 ACK IP Network SIP IP Phone User A SIP IP Phone User B SIP IP Phone User C Proxy Server Redirect Server 50212 User A is taken off hold The RTP channel 1 between User A and B is re established User A is on hold The RTP channel 1 between User A and B is torn down User B mixes the RTP channels 1 a...

Page 154: ... phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP 200 OK response to Cisco SIP IP phone A The 200 OK response notifies Cisco SIP IP phone A that the connection has been made If Cisco SIP IP phone B supports the media capability advertised in the INVITE message sent by Cisco SIP IP phone A it advertises the intersection of its own and Cisco SIP IP phone A s media capability in the 2...

Page 155: ...ier is assigned to the call and is inserted in the Call ID field The transaction number within a single call leg is identified in the CSeq field The media capability User B is ready to receive is specified 9 180 Ringing Cisco SIP IP phone C to Cisco SIP IP phone B Cisco SIP IP phone C sends a SIP 180 Ringing response to Cisco SIP IP phone B 10 200 OK Cisco SIP IP phone C to Cisco SIP IP phone B Ci...

Page 156: ... is on the phone and is unable or unwilling to take another call 12 INVITE Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a mid call INVITE to Cisco SIP IP phone A with the same call ID as the previous INVITE and new SDP session parameters IP address which are used to reestablish the call Call_ID 1 SDP c IN IP4 181 23 250 2 To reestablish the call between phone A and phone...

Page 157: ...er is assigned to the call and is inserted in the Call ID field The transaction number within a single call leg is identified in the CSeq field The media capability User A is ready to receive is specified The port on which the gateway is prepared to receive the RTP data is specified 3 Call Proceeding Gateway 1 to PBX A Gateway 1 sends a Call Proceeding message to PBX A to acknowledge the Call Setu...

Page 158: ...7 Release PBX A to Gateway 1 PBX A sends a Release message to Gateway 1 8 ACK Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP ACK to the Cisco SIP IP phone The ACK confirms that User A has received the 486 Busy Here response The call session attempt is now being terminated 9 Release Complete Gateway 1 to PBX A Gateway 1 sends a Release Complete message to PBX A and the call session attempt i...

Page 159: ...X A to acknowledge the Call Setup request 4 100 Trying Cisco SIP IP phone to Gateway 1 The Cisco SIP IP phone sends a SIP 100 Trying response to Gateway 1 The 100 Trying response indicates that the INVITE request has been received by the Cisco SIP IP phone 5 180 Ringing Cisco SIP IP phone to Gateway 1 The Cisco SIP IP phone sends a SIP 180 Ringing response to Gateway 1 The 180 Ringing response ind...

Page 160: ...e dial peer includes the IP address and the port number of the SIP enabled entity to contact Gateway 1 sends a 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 PBX A is identified as the call session initiator in the From field A unique ...

Page 161: ...e a definite failure response that is a client error the request will not be retried without modification If the Cisco SIP IP phone sends a class 5xx failure response an indefinite failure that is a server error the request is not terminated but rather other possible locations are tried If the Cisco SIP IP phone sends a class 6xx failure response a global error the search for User B is terminated ...

Page 162: ...the Request URI field in the INVITE request to User B appears as INVITE sip 555 0002 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a username Cisco SIP IP phone A is identified as the call session initiator in the From field A unique numeric identifier is assigned to the call and is inserted in the Call ID field The tr...

Page 163: ...elephone number and host is either a domain name or a numeric network address For example the Request URI field in the INVITE request to User B appears as INVITE sip 555 0002 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a username Cisco SIP IP phone A is identified as the call session initiator in the From field A uni...

Page 164: ... URI field in the INVITE request to User B appears as INVITE sip 555 0002 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a username Cisco SIP IP phone A is identified as the call session initiator in the From field A unique numeric identifier is assigned to the call and is inserted in the Call ID field The transaction n...

Page 165: ...he INVITE message 3 INVITE Cisco SIP IP phone to primary proxy third try Cisco SIP IP phone retries a third time to connect to the proxy by sending out the INVITE message 4 INVITE Cisco SIP IP phone to primary proxy fourth try Cisco SIP IP phone retries a fourth time to connect to the proxy by sending out the INVITE message IP SIP IP Phone User A Primary 1 INVITE 2 INVITE retry 3 INVITE retry 4 IN...

Page 166: ...e indicates that the INVITE request has been received by the gateway 12 Alerting PBX to gateway PBX sends an Alert message to gateway The Alert message indicates that PBX has received a 100 Trying Ringing response from the gateway 13 180 Ringing Gateway to Cisco SIP IP phone User A The gateway sends a SIP 180 Ringing response to User A The 180 Ringing response indicates that the gateway is being a...

Page 167: ...cond try Cisco SIP IP phone retries a second time to connect to the primary proxy by sending out the INVITE message 3 INVITE Cisco SIP IP phone User A to primary proxy third try Cisco SIP IP phone retries a third time to connect to the primary proxy by sending out the INVITE message IP IP SIP IP Phone User A 1 INVITE 2 INVITE retry 3 INVITE retry 4 INVITE retry 5 INVITE retry 6 INVITE retry 7 INVI...

Page 168: ... User B 12 180 Ringing Cisco SIP IP phone User B to backup proxy User B sends a SIP 180 Ringing response to the backup proxy The 180 Ringing response indicates that User B is being alerted 13 180 Ringing Backup proxy to Cisco SIP IP phone User A The backup proxy sends a SIP 180 Ringing response to User A The 180 Ringing response indicates that the backup proxy is being alerted 14 200 OK Cisco SIP ...

Page 169: ...y and PBX The call setup includes the standard transactions that take place as User A attempts to call User B 3 Call Proceeding PBX to gateway PBX sends a Call Proceeding message to gateway to acknowledge the Call Setup request 4 100 Trying Gateway to Cisco SIP IP phone User A Gateway sends a SIP 100 Trying response to User A The 100 Trying response indicates that the INVITE request has been recei...

Page 170: ... OK response to the User A The 200 OK response notifies User A that the connection has been made 9 ACK Cisco SIP IP phone User A to gateway User A sends a SIP ACK to the gateway The ACK confirms that User A has received the 200 OK response The call session is now active 10 Connect ACK Gateway to PBX Gateway acknowledges PBX s Connect message 11 BYE Cisco SIP IP phone User A to gateway User A termi...

Page 171: ...ergency proxy User B sends a SIP 100 Trying response to the emergency proxy The 100 Trying response indicates that the INVITE request has been received by User B 5 180 Ringing Cisco SIP IP phone User B to emergency proxy User B sends a SIP 180 Ringing response to the emergency proxy The 180 Ringing response indicates that User B is being alerted 6 180 Ringing Emergency proxy to Cisco SIP IP phone ...

Page 172: ...s the call session and sends a SIP BYE request to the emergency proxy The BYE request indicates that User A wants to release the call 12 BYE Emergency proxy to Cisco SIP IP phone User B Emergency proxy terminates the call session and sends a SIP BYE request to User B The BYE request indicates that the emergency proxy wants to release the call 13 200 OK Cisco SIP IP phone User B to emergency proxy ...

Page 173: ...o SIP IP Phone Operational and Physical Specifications Specification Value or Range Operating temperature 32 to 104 F 0 to 40 C Operating relative humidity 10 to 95 noncondensing Storage temperature 14 to 140 F 10 to 60 C Height 8 in 20 32 cm Width 10 5 in 26 67 cm Depth 6 in 15 24 cm Weight 3 5 lb 1 6 kg Power 100 240 VAC 50 60 Hz 0 5 A when using the AC adapter 48 Vdc 0 2 A when using the in lin...

Page 174: ...wer jack Switchcraft 712A is 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 Safety UL1950 CSA C22 2 No 950 EN 60950 IEC 60950 AS NZS 3260 T...

Page 175: ...ither Category 3 or 5 cabling for 10 Mpbs connections but use Category 5 for 100 Mbps connections On both the LAN to phone port left RJ 45 port facing the back of the phone and PC to phone port right port use full duplex to avoid collisions Use the LAN to phone port to connect the phone to the network a LAN to phone jack Use the PC to phone port to connect a network device such as a computer to th...

Page 176: ...C 4 Cisco SIP IP Phone Administrator Guide Appendix C Technical Specifications Connections Specifications ...

Page 177: ...ma di collegare il sistema all alimentatore Advarsel Les installasjonsinstruksjonene før systemet kobles til strømkilden Aviso Leia as instruções de instalação antes de ligar o sistema à sua fonte de energia Advertencia Ver las instrucciones de instalación antes de conectar el sistema a la red de alimentación Varning Läs installationsanvisningarna innan du kopplar systemet till dess strömförsörjni...

Page 178: ... System und schließen Sie keine Kabel an bzw trennen Sie keine ab wenn es gewittert Avvertenza Non lavorare sul sistema o collegare oppure scollegare i cavi durante un temporale con fulmini Advarsel Utfør aldri arbeid på systemet eller koble kabler til eller fra systemet når det tordner eller lyner Aviso Não trabalhe no sistema ou ligue e desligue cabos durante períodos de mau tempo trovoada Adver...

Page 179: ...ganger inneholder TNV kretser Det finnes både LAN utganger og WAN utganger som bruker RJ 45 kontakter Vær forsiktig når du kobler kabler Aviso Para evitar choques eléctricos não conecte os circuitos de segurança de baixa tensão SELV aos circuitos de tensão de rede telefónica TNV As portas LAN contêm circuitos SELV e as portas WAN contêm circuitos TNV Algumas portas LAN e WAN usam conectores RJ 45 ...

Page 180: ...icare che un fusibile o interruttore automatico non superiore a 120 VCA 15 A U S 240 VCA 10 A internazionale sia stato usato nei fili di fase tutti i conduttori portatori di corrente Advarsel Dette produktet er avhengig av bygningens installasjoner av kortslutningsbeskyttelse overstrøm Kontroller at det brukes en sikring eller strømbryter som ikke er større enn 120 VAC 15 A USA 240 VAC 10 A intern...

Page 181: ...face CO Central office CPE Customer premises equipment Terminating equipment such as terminals telephones and modems supplied by the telephone company installed at the customer sites and connected to the telephone company network CSM Call switching module D dial peer An addressable call endpoint In Voice over IP VoIP there are two types of dial peers POTS and VoIP DNS Domain Name System Used to ad...

Page 182: ...n International Telecommunication Union ITU T standard that describes packet based video audio and data conferencing H 323 is an umbrella standard that describes the architecture of the conferencing system and refers to a set of other standards H 245 H 225 0 and Q 931 to describe its actual protocol H 323 RAS Registration admission and status The RAS signaling function performs registration admiss...

Page 183: ... telephone service supplying standard single line telephones telephone lines and access to the PSTN proxy server An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients Requests are serviced internally or by passing them on possibly after translation to other servers A proxy interprets and if necessary rewrites a request message...

Page 184: ...ots Bandwidth is allocated to each channel regardless of whether the station has data to transmit U user agent See UAS UAC User agent client A user agent client is a client application that initiates the SIP request UAS User agent server or user agent A user agent server is a server application that contacts the user when a SIP request is received then returns a response on behalf of the user The ...

Page 185: ...g phone placement 2 12 administrative VLAN ID parameter 3 3 Allow header field A 7 Also header field A 7 alternate TFTP server enabling 3 3 authentication name configuring 3 24 services 1 2 Authorization header field A 7 B basic telephony extensible markup language BTXML 1 5 billing services 1 2 book objectives viii organization viii BTXML 1 5 buttons information 1 4 line 1 4 volume 1 4 C cables c...

Page 186: ...rk parameters 2 9 manually 2 10 via DHCP 2 10 SIP parameters manually 2 7 via TFTP 2 4 connections 1 12 2 11 Contact header field A 7 Content Encoding header field A 7 Content Length header field A 7 Content Type header field A 7 conventions document ix Cseq header field A 7 D Date header field A 7 daylight savings time 3 36 default configuration file 2 4 2 5 example 2 6 3 21 guidelines 2 4 modify...

Page 187: ...3 25 Expires header field A 7 F features call forward 1 8 call hold 1 8 call transfer 1 8 call waiting disabled 1 8 call waiting enabled 1 8 do not disturb 1 8 secondary directory number 1 8 URL dialing 1 8 file default 3 21 phone specific 3 25 files audio 2 3 dual boot 2 3 firmware image 2 3 OS79XX txt 2 3 RINGLIST DAT 2 3 SIPDefault cnf 2 3 firmware image 2 3 updating 3 44 version viewing 3 44 f...

Page 188: ...uage support 1 10 LCD screen 1 4 line buttons 1 4 lines configuring authentication name 3 24 name 3 24 password 3 24 short name 3 24 linex_authname parameter 3 24 linex_name parameter 3 24 linex_password parameter 3 24 linex_shortname parameter 3 24 locking configuration mode 3 2 M MAC address parameter 3 4 Max Forwards header field A 7 menu call preferences 3 6 messages status 3 43 messages URI p...

Page 189: ...eter 3 5 Organization header field A 7 OS79XX txt 2 3 Out of Band DTMF parameter 3 28 overview Cisco SIP IP phone 1 3 initialization process 2 1 product 1 1 SIP 1 1 P parameters common 2 5 3 8 configuring network 2 9 SIP 2 3 directory_url 1 5 erasing 3 41 logo_url 1 5 nat_enable 3 15 network 2 9 administrative VLAN ID 3 3 alternate TFTP 3 3 default routers 3 3 DHCP address release 3 3 DHCP enable ...

Page 190: ...3 21 password configuring 3 24 line 3 24 phone 3 8 adjusting placement 2 12 connecting 2 11 connections 1 12 access port 1 13 network 1 13 network port 1 13 features dialing pad 1 4 footstand adjustment 1 4 handset 1 4 headset 1 14 headset and speaker toggle 1 4 information button 1 4 LCD screen 1 4 line buttons 1 4 mute toggle 1 4 on screen mode keys 1 4 physical 1 3 scroll key 1 4 soft keys 1 4 ...

Page 191: ...cifying 3 18 R Real Time Transport Protocol RTP 1 11 Record Route header field A 8 redirect server 1 3 registrar server 1 3 registration enabling 3 18 timer 3 20 3 28 related documentation viii release DHCP address 3 3 request methods B 1 Require header field A 8 resetting network statistics 3 44 Response Key header field A 8 responses A 2 global 6xx A 7 information 1xx A 2 redirection 3xx A 3 req...

Page 192: ...gateways 1 3 header fields A 7 IP phone overview 1 3 methods A 2 overview 1 1 parameters Authentication Name 3 27 Authentication Password 3 27 configuring on your phone 3 26 Message URI 3 28 Name 3 27 Out of Band DTMF 3 28 phone specific configuration file 2 4 Preferred Codec 3 28 Proxy Address 3 27 Proxy Port 3 27 Register Expires 3 28 Register with proxy 3 28 Short Name 3 27 request methods B 1 ...

Page 193: ...sion 3 20 timer_t2 3 21 timers retransmission 3 21 Timestamp header field A 8 time zone abbreviations 3 38 time zone setting setting time zone 3 36 toggle headset and speaker 1 4 mute 1 4 To header field A 8 TOS media specifying 3 21 traceroute command 3 36 translated safety warnings D 1 circuit breaker 15A warning D 3 installation warning D 1 lightning activity warning D 2 product disposal warnin...

Page 194: ...Index IN 10 Cisco SIP IP Phone Administrator Guide volume buttons 1 4 W wall mounting phone 2 13 Warning header field A 8 WWW Authenticate header field A 8 X XML 1 5 ...

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