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B-57

Cisco SIP IP Phone 7960 Administrator Guide

78-10497-02

Appendix B SIP Call Flows

Call Flow Scenarios for Successful Calls

10

200 OK—Cisco SIP IP phone C
to Cisco SIP IP phone B

Cisco SIP IP phone C sends a SIP 200 OK response to
Cisco SIP IP phone B. The 200 OK response notifies Cisco
SIP IP phone B that the connection has been made.

If Cisco SIP IP phone C supports the media capability
advertised in the INVITE message sent by Cisco SIP IP
phone B, it advertises the intersection of its own and Cisco
SIP IP phone B’s media capability in the 200 OK response.
If Cisco SIP IP phone C does not support the media
capability advertised by Cisco SIP IP phone B, it sends
back a 400 Bad Request response with a 304 Warning
header 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 the session description in the INVITE request.

A two-way RTP channel is established between SIP IP phone B and SIP IP phone C.

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 B, the
IP address of phone B is inserted into the c= SDP field.

13

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.

14

ACK—Cisco SIP IP phone 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.

SIP IP phone B acts as a bridge mixing the RTP channel between User A and User B with the channel
between User B and User C; establishing a conference bridge between User A and User C.

Step

Action

Description

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Summary of Contents for Cisco 7960

Page 1: ... 1706 USA http www cisco com Cisco Systems Inc Corporate Headquarters Tel 800 553 NETS 6387 408 526 4000 Fax 408 526 4100 Cisco SIP IP Phone 7960 Administrator Guide Version 2 0 Customer Order Number DOC 7810497 Text Part Number 78 10497 02 ...

Page 2: ...om 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 breakers or fuses Modifications to this product not authorized by Cisco Systems Inc could void the FCC approval and negate your authority to operate the product The Cis...

Page 3: ...tation xi Document Conventions xi Obtaining Documentation xv World Wide Web xv Documentation CD ROM xv Ordering Documentation xv Obtaining Technical Assistance xv Cisco Connection Online xvi Technical Assistance Center xvi Documentation Feedback xvii C H A P T E R 1 Product Overview 1 1 What is Session Initiation Protocol 1 1 Components of SIP 1 3 SIP Clients 1 4 SIP Servers 1 5 ...

Page 4: ...etting Started with Your Cisco SIP IP Phone 2 1 Initialization Process Overview 2 1 Installing the Cisco SIP IP Phone 2 3 Installation Task Summary 2 3 Downloading Files to Your TFTP Server 2 4 Configuring SIP Parameters 2 5 Configuring SIP Parameters via a TFTP Server 2 6 Manually Configuring the SIP Parameters 2 11 Configuring Network Parameters 2 13 Configuring Network Parameters via a DHCP Ser...

Page 5: ...Phone s SIP Settings 3 5 Modifying SIP Parameters via a TFTP Server 3 8 Modifying the Default SIP Configuration File 3 8 Modifying the Phone Specific SIP Configuration File 3 15 Modifying the SIP Parameters Manually 3 18 Setting the Date Time and Daylight Savings Time 3 22 Erasing the Locally Defined Settings 3 28 Erasing the Locally Defined Network Settings 3 28 Erasing the Locally Defined SIP Se...

Page 6: ...Call Flows B 1 Call Flow Scenarios for Successful Calls B 2 Gateway to Cisco SIP IP Phone Successful Call Setup and Disconnect B 3 Gateway to Cisco SIP IP Phone Successful Call Setup and Call Hold B 7 Gateway to Cisco SIP IP Phone Successful Call Setup and Call Transfer B 11 Cisco SIP IP Phone to Cisco SIP IP Phone Simple Call Hold B 16 Cisco SIP IP Phone to Cisco SIP IP Phone Call Hold with Consu...

Page 7: ... Gateway to Cisco SIP IP Phone Client Server or Global Error B 63 Cisco SIP IP Phone to Cisco SIP IP Phone Called User is Busy B 66 Cisco SIP IP Phone to Cisco SIP IP Phone Called User Does Not Answer B 68 Cisco SIP IP Phone to Cisco SIP IP Phone Authentication Error B 70 A P P E N D I X C Technical Specifications C 1 Physical and Operating Environment Specifications C 1 Cable Specifications C 3 C...

Page 8: ...Contents viii Cisco SIP IP Phone 7960 Administrator Guide 78 10497 02 G L O S S A R Y I N D E X ...

Page 9: ...co 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 engineers system administrators or telecommunication engineers should use this guide to learn the steps required to properly set up the Cisco SIP IP phone on the network The tasks described are considered to be administration...

Page 10: ...one 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 how to modify the Cisco SIP IP phone s network and SIP settings how to access network and call status information and how to upgrade the firmware Appendix A SIP Compliance with RFC 2543 Information provides reference infor...

Page 11: ... for Voice over IP introduced in Cisco IOS Release 12 0 3 T Cisco IOS IP and IP Routing Configuration Guide Cisco IOS Release 12 1 Multiservice Applications Configuration Guide Voice over IP for the Cisco 2600 and Cisco 3600 Series Routers Voice over IP for the Cisco AS5300 Documents Document Conventions This document uses the following conventions Commands and keywords are in boldface font Argume...

Page 12: ...ork on any equipment be aware of the hazards involved with electrical circuitry and be familiar with standard practices for preventing accidents To see translations of the warnings that appear in this publication refer to the appendix Translated Safety Warnings Waarschuwing Dit waarschuwingssymbool betekent gevaar U verkeert in een situatie die lichamelijk letsel kan veroorzaken Voordat u aan enig...

Page 13: ...rant dans cette publication veuillez consulter l annexe intitulée Translated Safety Warnings Traduction des avis de sécurité Warnung Dieses Warnsymbol bedeutet Gefahr Sie befinden sich in einer Situation die zu einer Körperverletzung führen könnte Bevor Sie mit der Arbeit an irgendeinem Gerät beginnen seien Sie sich der mit elektrischen Stromkreisen verbundenen Gefahren und der Standardpraktiken z...

Page 14: ...áticas comuns que possam prevenir possíveis acidentes Para ver as traduções dos avisos que constam desta publicação consulte o apêndice Translated Safety Warnings Traduções dos Avisos de Segurança Advertencia Este símbolo de aviso significa peligro Existe riesgo para su integridad física Antes de manipular cualquier equipo considerar los riesgos que entraña la corriente eléctrica y familiarizarse ...

Page 15: ...ion Ordering Documentation Registered CCO users can order the Documentation CD ROM and other Cisco Product documentation through our online Subscription Services at http www cisco com cgi bin subcat kaojump cgi Nonregistered CCO users can order documentation through a local account representative by calling Cisco s corporate headquarters California USA at 408 526 4000 or in North America call 800 ...

Page 16: ...services download and test software packages and order Cisco learning materials and merchandise Valuable online skill assessment training and certification programs are also available Customers and partners can self register on CCO to obtain additional personalized information and services Registered users may order products check on the status of an order and view benefits specific to their relat...

Page 17: ...eading Cisco product documentation on the World Wide Web you can submit technical comments electronically Click Feedback in the toolbar and select Documentation After you complete the form click Submit to send it to Cisco You can e mail your comments to bug doc cisco com To submit your comments by mail for your convenience many documents contain a response card behind the front cover Otherwise you...

Page 18: ...About This Guide Obtaining Technical Assistance xviii Cisco SIP IP Phone 7960 Administrator Guide 78 10497 02 ...

Page 19: ...tion Protocol Session Initiation Protocol SIP is the Internet Engineering Task Force s IETF s standard for multimedia conferencing over IP SIP is an ASCII based application layer control protocol defined in RFC 2543 that can be used to establish maintain and terminate calls between two or more end points Like other VoIP protocols SIP is designed to address the functions of signaling and session ma...

Page 20: ...dicating why the target end point was unavailable Establish a session between the originating and target end point If the call can be completed SIP establishes a session between the end points SIP also supports mid call changes such as the addition of another end point to the conference or the changing of a media characteristic or codec Handle the transfer and termination of calls SIP supports the...

Page 21: ...P end point 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 servers Figure 1 1 illustrates the architecture of a SIP network...

Page 22: ...quests Gateways Provide call control Gateways provide many services the most common being a translation function between SIP conferencing endpoints and other terminal types This function includes translation between transmission formats and between communications procedures In addition the gateway also translates between audio and video codecs and performs call setup and clearing on both the LAN s...

Page 23: ...mation about the next hop or hops that a message should take and then the client contacts the next hop server or UAS directly Registrar server Processes requests from UACs for registration of their current location Registrar servers are often co located with a redirect or proxy server What is the Cisco SIP IP Phone 7960 Cisco SIP IP phones 7960s hereafter referred to as Cisco SIP IP phones are ful...

Page 24: ...he soft key tabs Line or speed dial buttons Opens a new line or speed dials the number on the LCD screen Footstand adjustment Adjusts the angle of the phone base Soft keys Activates the feature described by the text message directly above on the LCD screen Information i button Provides online help for selected keys or features and network statistics about the active call This feature will be avail...

Page 25: ...al pad buttons to dial a phone number Dial pad buttons work exactly like those on your existing telephone Handset Lift the handset and press the dial pad numbers to place a call review voice mail messages answer a call and so on Supported Features In addition to the physical features illustrated in Figure 1 2 the Cisco SIP IP phone also provides the following An adjustable ring tone A hearing aid ...

Page 26: ...automatic dialing and automatic generation of a secondary dial tone Current date and time support via Simple Network Time Protocol SNTP and time zone and daylight savings time support Call redirection information support via the CC Diversion header Third party call control via delayed media negotiation A delayed media negotiation is one where the Session Description Protocol SDP information is not...

Page 27: ...from user B on hold When user A places user B on hold the 2 way RTP voice path between user A and user B is temporarily disconnected but the call session is still connected When user A takes user B off hold the 2 way RTP voice path is reestablished Call transfer Allows the Cisco SIP IP phone user user A to transfer a call from one user user B to another user user C User A places user B on hold and...

Page 28: ...ony features and URL dialing refer to the Getting Started Cisco IP Phone 7960 and Quick Reference Cisco IP Phone 7960 documents that shipped with the phone Supported Protocols The Cisco SIP IP phone supports the following standard protocols Domain Name System DNS DNS is used in the Internet for translating names of network nodes into addresses SIP uses DNS to resolve the host names of end points t...

Page 29: ... features for addressing type of service ToS specification fragmentation and reassembly and security The Cisco SIP IP phone supports IP as it is defined in RFC 791 Real Time Transport Protocol RTP RTP transports real time data such as voice data over data networks RTP also the ability to obtain Quality of Service QoS information The Cisco SIP IP phone supports RTP as a media channel Session Descri...

Page 30: ...ling Prerequisites For the Cisco SIP IP phone to successfully operate as a SIP endpoint in your network your network must meet the following requirements A working IP network is established For more information about configuring IP refer to Cisco IOS IP and IP Routing Configuration Guide VoIP is configured on your Cisco routers For more information about configuring VoIP refer to the Cisco IOS Rel...

Page 31: ...onnections Connecting to the Network The Cisco SIP IP phone has two RJ 45 ports that each support 10 100 Mbps half or full duplex Ethernet connections to external devices network port labeled 10 100 SW and access port labeled 10 100 PC You can use either Category 3 or 5 cabling for 10 Mpbs connections but use Category 5 for 100 Mbps connections On both the network port and access port use full dup...

Page 32: ...ng to a standard wall receptacle WS X6348 RJ45V 10 100 switching module Provides inline power to the Cisco SIP IP phone when connected to a Catalyst 3500 4000 or 6000 family 10 100BaseTX switching module This module sends power on pins 1 2 and 3 6 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 50...

Page 33: ...alyst switch Next connect the un powered Cisco SIP IP phone to the network After the phone powers up connect the external power supply to the phone Using a Headset The Cisco SIP IP phone supports a four or six wire headset jack Specifically the Cisco SIP IP phone supports the following Plantronics headset models Tristar Monaural Encore Monaural H91 Encore Binaural H101 The Volume and Mute controls...

Page 34: ... configured on an IP subnet basis and additional IP addresses might not be available to assign the phone to a port so that it belongs to the same subnet as other devices PC connected to the same port Data traffic present on the VLAN supporting phones might reduce the quality of VoIP traffic You can resolve these issues by isolating the voice traffic onto a separate VLAN on each of the ports connec...

Page 35: ...talling the Cisco SIP IP Phone page 2 3 Verifying Startup page 2 20 Using the Cisco SIP IP Phone Menu Interface page 2 21 Reading the Cisco SIP IP Phone Icons page 2 22 Customizing the Cisco SIP IP Phone Ring Types page 2 24 Creating Dial Plans page 2 24 Initialization Process Overview The initialization process of the Cisco SIP IP phone is responsible for establishing network connectivity and for...

Page 36: ...will use IP settings that are stored in Flash memory 4 The TFTP server is contacted On the TFTP server is the latest Cisco SIP IP phone firmware image and the dual boot file OS79XX TXT that enables the phone to automatically determine and initialize for the VoIP environment in which it is being installed If the phone is using the TFTP server to obtain its SIP parameters there should also be a conf...

Page 37: ...erver create and store the configuration files as described in the Configuring SIP Parameters via a TFTP Server section on page 2 6 3 If you are using DCHP to configure the phones network settings configure the required network parameters on your DHCP server as described in the Configuring Network Parameters via a DHCP Server section on page 2 14 4 Connect the phone to the network and to a power s...

Page 38: ...all phones For more information on using the SIPDefault cnf file see the Creating the Default SIP Configuration File section on page 2 7 SIPConfigGeneric cnf Required File which 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 ri...

Page 39: ...configuration file from the TFTP server If the default configuration file has been configured and stored in the root directory of the TFTP server the phone reads the parameters defined in the file and stores those parameters that differ in Flash memory The phone then requests its phone specific configuration file If the phone specific configuration file has been configured and placed on the TFTP s...

Page 40: ...o apply to all phones in the default configuration file SIPDefault cnf Phone specific parameters should only be defined via a phone specific configuration file or manually configured Phone specific parameters should not be defined in the default configuration file Configuration File Guidelines When modifying the default configuration file and creating the phone specific configuration files adhere ...

Page 41: ... you can include optional comments Use the semicolon and pound delimiters to distinguish the comments Blank lines are allowed Comment lines are allowed Variable names are not case sensitive Only one variable can be set per line Distinguish 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 an...

Page 42: ...r open the SIPDefault cnf file and define values for the following SIP global parameters image_version Required Firmware version that the Cisco SIP IP phone should run Enter the name of the image version as it is released by Cisco Do not enter the extension You cannot change the image version by changing the file name because the version is also built into the file header Trying to change the imag...

Page 43: ...e define the parameters that are specific to a phone such as the lines configured on a phone and the users defined for those lines Before You Begin Review the guidelines and restrictions documented in the Configuration File Guidelines section on page 2 6 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 onl...

Page 44: ...authname parameter when registration is enabled the default name is used The default name is UNPROVISIONED linex_password Required when registration is enabled and the proxy requires authentication Password used by the phone for authentication if a registration is challenged by the proxy server during initialization If a value is not configured for the linex_password parameter when registration is...

Page 45: ...ings that might affect their call capabilities Review the guidelines on using the Cisco SIP IP phone menus documented in the Using the Cisco SIP IP Phone Menu Interface section on page 2 21 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 configur...

Page 46: ...hentication Password Required when registration is enabled Password used by the phone for authentication if a registration is challenged by the proxy server during initialization If a value is not configured for the Authentication Password parameter when registration is enabled the default logical password is used The default logical password is SIPmacaddress where macaddress is the MAC address of...

Page 47: ...equired 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 not required TFTP server IP address When configuring th...

Page 48: ...n 1 IP subnet mask dhcp option 3 Default IP gateway dhcp option 15 Domain name dhcp option 6 DNS server IP address dhcp option 66 TFTP server IP address Manually Configuring the Network Parameters If you are not using DCHP to configure your network parameters you must manually configure them Before You Begin Connect your phone as described in the Connecting the Phone section on page 2 16 Unlock co...

Page 49: ... complete list of the SIP parameters that you can configure see the Modifying the Phone s Network Settings section on page 3 2 Procedure Step 1 Press the settings key The Settings menu is displayed Step 2 Highlight Network Configuration Step 3 Press the Select soft key The Network Configuration menu is displayed Step 4 Highlight DHCP Enabled Step 5 Press the No soft key DHCP is now disabled Step 6...

Page 50: ...e will attempt to use DNS Servers 2 through 5 if DNS Server 1 is unavailable Step 7 When done press the Save soft key The phone programs the new information into Flash memory and resets Caution When you have completed your changes ensure that you lock the phone as described in the Locking Configuration Mode section on page 3 2 Connecting the Phone You must connect the phone to the network and to a...

Page 51: ...le from the switch or hub to the network port on the phone See Connecting to the Network section on page 1 13 for more information on the network port Step 2 Connect the handset and headset to their respective ports See Using a Headset section on page 1 15 for more information on the headset port Handset port Headset port Cisco IP Phone 7960 rear view 38006 Power outlet RJ 11 port optional power c...

Page 52: ...ou can adjust the tilt height to several different angles in 7 5 degree increments from flat to 60 degrees Alternatively you can mount the phone to the wall using the footstand or using the optional locking accessory Adjusting Phone Placement on the Desktop Adjust the footstand to the height that provides optimum view of the display and use of the buttons and keys To adjust the phone placement on ...

Page 53: ...al overview of these procedures Procedure Step 1 Push in the footstand adjustment knob Step 2 Adjust the footstand so it is flat against the back of the phone Step 3 Modify the handset rest so that the handset remains on the ear piece rest when the phone is vertically placed a Remove the handset from the ear piece rest b Locate the tab handset wall hook at the base of the ear piece rest c Slide th...

Page 54: ...s power connected to it the phone begins its startup process by cycling through these steps 1 These buttons flash on and off in sequence Headset Mute Speaker 2 The Cisco Systems Inc copyright displays on the LCD Cisco IP Phone 7960 rear view Footstand adjustment button raises and lowers adjustment plate Adjustment plate raises and lowers phone vertically Adjustment plate installation screws holes ...

Page 55: ... automatically reboot to run the new image 4 The main LCD screen appears displaying Primary directory number Soft keys If the phone successfully passes through these stages it has started up properly Using the Cisco SIP IP Phone Menu Interface As you configure your phone s settings via the menu interface follow these guidelines Select a parameter by pressing the down arrow to scroll to and highlig...

Page 56: ...the dial pad to enter a new value Press the soft key to delete any mistakes After editing a parameter press the Validate soft key to save the value that you have entered and exit the Edit panel Reading the Cisco SIP IP Phone Icons When using the Cisco SIP IP phone a variety of icons can display on the phone s LCD Table 1 lists and describes each icon that you might see while using the Cisco SIP IP...

Page 57: ...ng the call The character x displayed to the right of the icon indicates that registration has failed The line is configured for URL dialing and 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 more soft key and then the Nu...

Page 58: ...dding specify the name as you want it to display on the Ring Type menu press Tab and then specify the filename of the ring type For example the format of a pointer in your RINGLIST DAT file should appear similar to the following Ring Type 1 ringer1 pcm Step 3 After defining pointers for each of the ring types you are adding save your modifications and close the RINGLIST DAT file Creating Dial Plan...

Page 59: ... must use a dial plan that differs from the one being used by other phones in the same system DIALTEMPLATE indicates the start of a template and DIALTEMPLATE indicates the end of a template Rules are matched from start to finish with the longest matching rule taken as the one to use Matches against a period are not counted for the length to be the longest Step 1 Using an ASCII editor open a new fi...

Page 60: ...he 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 Example 2 1 Example of a PBX North American Dial Plan DIALTEMPLATE TEMPLATE MAT...

Page 61: ...ght Savings Time page 3 22 Erasing the Locally Defined Settings page 3 28 Accessing Status Information page 3 30 Upgrading the Cisco SIP IP Phone Firmware page 3 33 Entering Configuration Mode When you access the network configuration information on your Cisco SIP IP phone you will notice that there is a padlock symbol located in the upper right corner of your LCD By default the network configurat...

Page 62: ...ate The unlocked symbol indicates that you can modify the network and SIP configuration settings Locking Configuration Mode To lock the Cisco SIP IP phone when you are done modifying the settings press If the Network Configuration or SIP Configuration panel is displayed the lock icon in the upper right corner of your LCD will change to a locked state If you are located elsewhere in the Cisco SIP I...

Page 63: ...work Configuration Step 3 Press the Select soft key The Network Configuration menu is displayed The following network parameters are available on the Network Configuration menu DHCP Server IP address of the DHCP server from which the phone received its IP address and additional network settings You cannot change the information in this field MAC Address Factory assigned unique 48 bit hexadecimal M...

Page 64: ...e phone is attached The value in this field is only used in non Cisco switched networks You can change the administrative VLAN used by the phone however if you have an administrative VLAN assigned on the Catalyst switch that setting overrides any changes made on the phone Network Media Type Ethernet port negotiation mode Possible values are Auto Port is auto negotiated Full 100 Port is configured ...

Page 65: ... Yes will re enable DHCP For more information on erasing the local configuration see the Erasing the Locally Defined Settings section on page 3 28 Step 4 When done press the Save soft key The phone programs the new information into Flash memory and resets Caution When you have completed your changes ensure that you lock the phone as described in the Locking Configuration Mode section on page 3 2 M...

Page 66: ...can cannot be defined via that menu Table 3 1 SIP Parameters Summary Configuration File SIP Configuration Menu Network Configuration Menu Services Menu anonymous_call_block NA NA Anonymous Call Block autocomplete NA NA Auto Complete Numbers callerid_blocking NA NA Caller ID Block dial_template NA NA NA dnd_control NA NA NA dst_auto_adjust NA NA NA dst_offset NA NA NA dst_start_day NA NA NA dst_sta...

Page 67: ...me line1 to line6 Shortname NA NA messages_uri Messages URI NA NA network_media_type NA Network Media Type NA phone_label Phone Label NA NA preferred_codec Preferred Codec NA NA proxy_register Register with Proxy NA NA proxy1_address Proxy Address NA NA proxy1_port Proxy Port NA NA sip_invite_retx NA NA NA sip_retx NA NA NA sntp_mode NA NA NA sntp_server NA NA NA sync NA NA NA tftp_cfg_dir TFTP Di...

Page 68: ...recommend that you use the default configuration file to define values for SIP parameters that are common to all phones Doing so will make controlling and maintaining your network an easier task You can then define only those parameters that are specific to a phone in the phone specific configuration file Phone specific parameters should only be defined in a phone specific configuration file or ma...

Page 69: ...ot change the image version by changing the file name because the version is also built into the file header Trying to change the image version by changing the file name will cause the firmware to fail when it compares the version in the header against the file name proxy1_address Required IP address of the primary SIP proxy server that will be used by the phones Enter this address in IP dotted de...

Page 70: ...avt If requested by the remote side generate DTMF digits out of band and disable in band DTMF signaling otherwise do not generate DTMF digits out of band avt_always Always generate DTMF digits out of band This option disables in band DTMF signaling The default is avt dtmf_avt_payload Optional Payload type for AVT packets Possible range is 96 to 127 If the value specified exceeds 127 the phone will...

Page 71: ... enable registration and authentication is required you must specify values for the linex_authname and linex_password parameters where x is a number 1 through 6 in the phone specific configuration file For information on configuring the phone specific configuration file see the Modifying the Phone Specific SIP Configuration File section on page 3 15 timer_register_expires Optional The amount of ti...

Page 72: ...ay of the week on which DST ends dst_stop_week_of_month Optional Week of month in which DST ends dst_stop_time Optional Time of day on which DST ends dst_auto_adjust Optional Whether or not DST is automatically adjusted on the phones dnd_control Optional Whether the Do Not Disturb feature is enabled or disabled by default on the phone or whether the feature is permanently enabled When the feature ...

Page 73: ...e caller identification is included in the Request URI header field 1 The Caller ID Blocking feature is enabled by default but can be turned on and off via the phone s user interface When enabled Anonymous is included in place of the user identification in the Request URI header field 2 The Caller ID Blocking feature is disabled permanently and cannot be turned on and off locally via the phone s u...

Page 74: ...ull100 Port is configured to be a full duplex 100MB connection Half100 Port is configured to be a half duplex 100MB connection Full10 Port is configured to be a full duplex 10MB connection Half10 Port is configured to be a half duplex 10MB connection The default is Auto autocomplete Optional Whether to have numbers automatically completed when dialing Valid values are 0 disable auto completion or ...

Page 75: ... address proxy1_address 192 168 1 1 Modifying the Phone Specific SIP Configuration File In the phone specific SIP configuration file maintain those parameters that are specific to a phone such as the lines configured on a phone and the users defined for those lines Before You Begin Review the guidelines and restrictions documented in the Configuration File Guidelines section on page 2 6 Line param...

Page 76: ...he 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 only purposes If a value is not specified for 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 authenticati...

Page 77: ...ally 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 interface If specifying this value specify this parameter in the phone specific configuration file 3 The Do Not Disturb feature is on permane...

Page 78: ...e 1 authentication name password line1_password password Modifying the SIP Parameters Manually If you did not configure the SIP parameters via a TFTP server you can configure them manually 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 user...

Page 79: ...55 1212 as 5551212 When entering an e mail address enter the e mail ID without the host name Short Name Optional Name or number associated with the linex_name as you want it to display on the phone s LCD if the linex_name value exceeds the display area For example if the linex_name value is the phone number 111 222 333 4444 you can specify 34444 for this parameter to have 3444 display on the LCD i...

Page 80: ...ep 6 Press the Back soft key exit the Line 1 Configuration menu Step 7 To configure additional lines on the phone highlight the next Line x Settings press the Select soft key and repeat Step 5 and Step 6 and then continue with Step 8 Step 8 In addition to the line settings you can highlight and press Select to configure the following parameters on the SIP Configuration menu Message URI Number to c...

Page 81: ... must specify values for the Authentication Name and Authentication Password parameters Register Expires Optional The amount of time in seconds after which a REGISTRATION request will expire This value is used the Expire header field The valid value is any positive number however we recommend 3600 seconds The default is 3600 TFTP Directory Required if phone specific configuration files are located...

Page 82: ...r example starts the first Sunday in April and ends on the last day of October configuration We recommend that date and time related parameters be defined in the SIPDefault cnf file Before You Begin When configuring the date time time zone and DST settings remember the following Review the guidelines and restrictions documented in the Configuration File Guidelines section on page 2 6 Determine whe...

Page 83: ... to the local network address After the first SNTP response is received the phone switches to unicast mode with the server being set as the one who first responded SNTP packet to the local network 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 loc...

Page 84: ...ectedbroadcast Sends SNTP request to the SNTP server Nothing When in multicast mode SNTP requests are not sent SNTP request to the SNTP server SNTP packet to the SNTP server After the first SNTP response is received the phone switches to multicast mode Receives SNTP response from the SNTP server and ignores responses from other SNTP servers SNTP data via the SNTP NTP multicast address from the loc...

Page 85: ...T or EST Step 2 To configure common DST settings specify values for the following parameters dst_offset Offset from the phone s time when DST is in effect When DST is over the specified offset is no longer applied to the phone s time Valid values are the same as for the time_zone parameter dst_auto_adjust Whether or not DST is automatically adjusted on the phones Valid values are 0 disable automat...

Page 86: ...of the month on which DST ends Valid values are 1 through 31 for the days of the month or 0 when specifying relative DST to specify that this field be ignored and that the value in the dst_stop_day_of_week parameter be used instead Step 4 To configure relative DST specify values for the following parameters dst_start_day_of_week Day of the week on which DST begins Valid values are Sunday or Sun Mo...

Page 87: ...t_stop_week_of_month Week of month in which DST ends 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 Step 5 Save the file with the same file name SIPDefault cnf to the root directory of your TFTP server The following is an example of the configurati...

Page 88: ...00 dst_stop_autoadjust 1 additional configuration text omitted Erasing the Locally Defined Settings You can erase the locally defined network settings and the SIP settings that have been 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 ...

Page 89: ...ngs the values are reset to the defaults Note If your system has been set up to have the phones retrieve their SIP parameters via a TFTP server you will need to edit the configuration file in which a parameter is defined to delete the parameter When deleting a parameter leave the variable in the file but change its value to a null value or UNPROVISIONED If both the variable and its value are remov...

Page 90: ...s the new information into Flash memory and resets Accessing Status Information There are several types of status information that you can access via the settings key The information that you can obtain via the settings key can aid in system management To access status information select settings and then select Status from the Settings menu From the Status which the following three options are av...

Page 91: ... Messages panel press the Exit soft key Viewing Network Statistics To view statistical information about the phone and network performance complete the following steps Step 1 Press the settings key The Settings menu is displayed Step 2 Highlight Status Step 3 Press the Select soft key The Setting Status menu is displayed Step 4 Highlight Network Statistics Step 5 Press the Select soft key The Netw...

Page 92: ...is in a linked state and has auto negotiated a half duplex 100Mbps connection Port 0 Full 10 Indicates that the network is in a linked state and has auto negotiated a full duplex 10Mbps connection Port 0 Half 10 Indicates that the network is in a linked state and has auto negotiated a half duplex 10Mbps connection Port 1 Full 100 Indicates that the network is in a linked state and has auto negotia...

Page 93: ...e phone This image name does not change Step 6 To exit the Firmware Versions panel press the Exit soft key Upgrading the Cisco SIP IP Phone Firmware There two methods that you can use to upgrade the firmware on your Cisco SIP IP phones You can upgrade the firmware on one phone at a time via the phone specific configuration or you can upgrade the firmware on a system of phones using the default con...

Page 94: ...TP server and requests its configuration files The phone compares the image defined in the file to the image that it has stored in Flash memory If the phone determines that the image defined in the file differs from the image in Flash memory it downloads 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 p...

Page 95: ... ASCII editor open the SIPDefault cnf file located in the root directory of your TFTP server and change the image_version parameter to the name of the latest image 2 Using an ASCII editor open the syncinfo xml file located in the root directory of your TFTP server and specify values for the image version and sync parameter as follows IMAGE VERSION image_version SYNC sync_number Where image_version...

Page 96: ...ct the TFTP server for the syncinfo xml file If the phone is not in an idle state the phone will wait until it is in an idle state for 20 seconds and then contact the TFTP server for the syncinfo xml file 2 The phone reads the syncinfo xml file and performs the following as appropriate a Determines whether the current image is specified If so the phone proceeds to c If not the phone proceeds to b ...

Page 97: ...is section describes how the Cisco SIP IP phone complies with the IETF definition of SIP as described in RFC 2543 This section contains compliance information on the following SIP Functions page A 2 SIP Methods page A 2 SIP Responses page A 3 SIP Header Fields page A 10 SIP Session Description Protocol SDP Usage page A 12 ...

Page 98: ...Supported User Agent Client UAC Yes User Agent Server UAS Yes Proxy Server Third party only Redirect Server Third party only 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 No BYE Yes CANCEL Yes REGISTER Yes The Cisco SIP IP phone supports both use...

Page 99: ...Release 1 0 of the Cisco SIP IP phone supports the following SIP responses 1xx Response Information Responses page A 4 2xx Response Successful Responses page A 4 3xx Response Redirection Responses page A 5 4xx Response Request Failure Responses page A 5 5xx Response Server Failure Responses page A 10 6xx Response Global Responses page A 10 ...

Page 100: ...ne waits for a 180 Ringing 183 Session progress or 200 OK response 180 Ringing Yes None 181 Call is being forwarded See comments The Cisco SIP IP phone does not generate these responses however the phone does receive them The phone processes these responses the same way that it processes the 100 Trying response 182 Queued 183 Session Progress The SIP IP phone does not generate this message Upon re...

Page 101: ... time Upon receiving this response the phone sends an INVITE containing the contact information received in the 302 Moved temporarily response 305 Use Proxy Yes The phone does not generate these responses The gateway contacts the new address in the Contact header field 380 Alternate Service Yes 4xx Response Supported Comments 400 Bad Request Yes The phone generates a 400 Bad Request response for a...

Page 102: ...ne receives a 403 Forbidden response it notifies the 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 only received in this rel...

Page 103: ... 408 Request Timeout See comments The SIP phone does not generate a 408 Request Timeout response For an incoming response the gateway initiates a graceful call disconnect during which the caller hears a busy or fast busy tone before clearing the call request 409 Conflict See comments This response is only received the phone in this release The 409 Conflict response indicates that the INVITE reques...

Page 104: ...a retry after header field is contained in this response then the user can attempt the call once again in the retry time provided 414 Request URL Too Long See comments This response is only received by the phone in this release The user is notified if this response is received 415 Unsupported Media See comments This response is only received by the phone in this release The user is notified if thi...

Page 105: ...is response is received 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous See comments This response is only received by the phone in this release If a new contact is received the phone might re initiate 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 receiv...

Page 106: ...tact address is present If an additional contact address is not present the gateway initiates a graceful call disconnect 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Gateway Timeout 505 Version Not Supported 6xx Response Comments 600 Busy Everywhere The Cisco SIP IP phone does not generate these 6xx responses For an incoming response the gateway initiates a graceful call disconn...

Page 107: ...ll 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 Forwards Yes Organization No Priority No Proxy Authenticate Yes Proxy Authorization Yes Proxy Require Yes ReBy Yes Record Route Yes Require Yes Response Key No Retry After Yes Route Yes Header Field Supported ...

Page 108: ...2 SIP Session Description Protocol SDP Usage Server No 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 creator and session identifier Yes a Session name Yes c Connection information Yes m Media name and transport address Yes Header Field Supported ...

Page 109: ...either the caller or the callee CANCEL Cancels any pending searches but does not terminate a call that has already been accepted OPTIONS Queries the capabilities of servers REGISTER Registers the address listed in the To header field with a SIP server The following types of responses are used by SIP and generated by the Cisco SIP gateway SIP 1xx Informational Responses SIP 2xx Successful Responses...

Page 110: ...isco SIP IP Phone to Cisco SIP IP Phone Simple Call Hold page B 16 Cisco SIP IP Phone to Cisco SIP IP Phone Call Hold with Consultation page B 20 Cisco SIP IP Phone to Cisco SIP IP Phone Call Waiting page B 25 Cisco SIP IP Phone to Cisco SIP IP Phone Call Transfer without Consultation page B 31 Cisco SIP IP Phone to Cisco SIP IP Phone Call Transfer with Consultation page B 35 Cisco SIP IP Phone to...

Page 111: ...es a successful gateway to Cisco SIP IP phone call setup and disconnect In this scenario the two end users are 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 The call flow is as follows 1 User A calls User B 2 User B answers the call 3 ...

Page 112: ...e B 1 Gateway to Cisco SIP IP Phone Successful Setup and Disconnect IP 3 Call Proceeding 6 Alerting 8 Connect 12 Disconnect 1 Setup PBX A SIP IP Phone User B User A GW1 IP Network 4 100 Trying 11 BYE 5 180 Ringing 7 200 OK 2 INVITE 2 way RTP channel 2 way voice path 10 ACK 14 200 OK 9 Connect ACK 13 Release 15 Release Complete 41724 ...

Page 113: ...d 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 receive is specified The port on which the Gateway is prepared to receive the RTP data is specif...

Page 114: ...X A acknowledges Gateway 1 s Connect message 10 ACK Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP ACK to the Cisco SIP IP phone The ACK confirms that Gateway 1 has received the 200 OK response The call session is now active 11 BYE Cisco SIP IP phone to Gateway 1 User B terminates the call session at his Cisco SIP IP phone and the phone sends a SIP BYE request to Gateway 1 The BYE request i...

Page 115: ...ay to Cisco SIP IP phone call setup and call hold In this scenario the two end users are 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 The call flow is as follows 1 User A calls User B 2 User B answers the call 3 User B puts User A on ...

Page 116: ...essful Call Setup and Call Hold IP SIP IP Phone User B 3 Call Proceeding 6 Alerting 8 Connect 10 Connect ACK 1 Setup PBX A User A GW1 IP Network 4 100 Trying 13 ACK 16 ACK 11 INVITE c IN IP4 0 0 0 0 14 INVITE c IN IP4 IP User B 5 180 Ringing 7 200 OK 2 INVITE 2 way RTP channel No RTP packets being sent 2 way VP 2 way voice path 2 way voice path 12 200 OK 9 ACK 15 200 OK 41728 ...

Page 117: ...d 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 receive is specified The port on which the Gateway is prepared to receive the RTP data is specif...

Page 118: ...Connect message 11 INVITE Cisco SIP IP phone to Gateway 1 User B puts User A on hold The Cisco SIP IP phone sends a SIP INVITE request to Gateway 1 12 200 OK Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP 200 OK response to the Cisco SIP IP phone The 200 OK response notifies the Cisco SIP IP phone that the INVITE was successfully processed 13 ACK Cisco SIP IP phone to Gateway 1 The Cisco SI...

Page 119: ...ree end users User A User B and User C User A is located at PBX A PBX A is connected to Gateway 1 SIP Gateway via a T1 E1 User B is located at a Cisco SIP IP phone and is directly connected to the IP network User C is located at PBX B PBX B is connected to Gateway 2 SIP Gateway via a T1 E1 Gateway 1 Gateway 2 and the Cisco SIP IP phone are connected to one another over an IP network The call flow ...

Page 120: ...ing 8 Connect 9 Connect ACK 14 Setup 25 Connect ACK 16 Call Proceeding 17 Alerting 20 Connect 1 Setup PBX A PBX B User A User C GW1 GW2 IP Network 4 100 Trying 13 INVITE Requested By B 11 BYE Also C 15 100 Trying 18 180 Ringing 21 200 OK 5 180 Ringing 7 200 OK 2 INVITE 2 way RTP channel 2 way voice path 19 Alerting 22 Connect 23 Connect ACK 2 way voice path 2 way voice path 2 way RTP channel 12 20...

Page 121: ...ed 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 receive is specified The port on which the Gateway is prepared to receive the RTP data is speci...

Page 122: ...BX A acknowledges Gateway 1 s Connect message 11 BYE Cisco SIP IP phone to Gateway 1 User B transfers User A s call to User C and then hangs up The Cisco SIP IP phone sends a SIP BYE request to Gateway 1 The SIP BYE request includes the Also header In this scenario the Also header indicates that User C needs to be brought into the call while User B hangs up This header distinguishes the call trans...

Page 123: ...2 18 180 Ringing Gateway 2 to Gateway 1 Gateway 2 sends a SIP 180 Ringing response to Gateway 1 The 180 Ringing response indicates that Gateway 2 has located and is trying to alert User C 19 Connect PBX B to Gateway 2 User C answers the phone PBX B sends a Connect message to Gateway 2 The Connect message notifies Gateway 2 that the connection has been made 20 200 OK Gateway 2 to Gateway 1 Gateway ...

Page 124: ...n Cisco SIP IP phones in which one of the participants places the other on hold and then returns to the call In this call flow scenario the two end users are User A and User B User A and User B are both 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 places User A on hold 4 User B takes Us...

Page 125: ...co 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 126: ...user is the telephone 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 user name Cisco SIP IP phone A is identified as the call session initiator in the Fr...

Page 127: ...response from Cisco SIP IP phone B The ACK might contain a message body 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 A and Cisco SIP IP phone B 5 INVITE Cisco SIP IP phone B to Cisco SIP IP phone A C...

Page 128: ...om User C 6 User B takes User A off hold 7 The original call continues 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 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 ...

Page 129: ...0 OK 2 way 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 Ph...

Page 130: ...user is the telephone 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 user name Cisco SIP IP phone A is identified as the call session initiator in the Fr...

Page 131: ...K response from Cisco SIP IP phone B The ACK might contain a message body 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 A and Cisco SIP IP phone B 5 INVITE Cisco SIP IP phone B to Cisco SIP IP phone A...

Page 132: ... sends back a 400 Bad Request response with a 304 Warning header 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 ...

Page 133: ...notified of the remaining call with User C 8 User B answers the notification and continues the call with User C 14 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...

Page 134: ...hold remains 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 ...

Page 135: ...user is the telephone 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 user name Cisco SIP IP phone A is identified as the call session initiator in the Fr...

Page 136: ...IP IP phone B The ACK might contain a message body 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 A and Cisco SIP IP phone B 5 INVITE Cisco SIP IP phone C to Cisco SIP IP phone B Cisco SIP IP phone C s...

Page 137: ...sage body 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 p...

Page 138: ...P IP phone B 18 BYE Cisco SIP IP phone B to Cisco SIP IP phone A The call continues 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 Ci...

Page 139: ... called a blind 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 transfers the call to User C 16 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 140: ...Cisco SIP IP Phone to Cisco SIP IP Phone Call Transfer without Consultation IP IP IP 2 180 Ringing 3 200 OK 2 way RTP channel A and B are disconnected 4 ACK 5 BYE Also C 6 200 OK 1 INVITE B 8 180 Ringing 9 200 OK 2 way RTP channel 10 ACK 7 INVITE C Requested by B IP Network SIP IP Phone User A SIP IP Phone User B SIP IP Phone User C 41468 ...

Page 141: ...user is the telephone 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 user name Cisco SIP IP phone A is identified as the call session initiator in the Fr...

Page 142: ... might contain a message body 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 A and Cisco SIP IP phone B User B then selects the option to transfer the call to User C 5 BYE Cisco SIP IP phone B to Cisco...

Page 143: ...all to User C 7 INVITE Cisco SIP IP phone A to Cisco SIP IP phone C Requested by Cisco SIP IP phone B At the request of Cisco SIP IP phone B Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone C The INVITE request is an invitation to User C to participate in a call session 8 180 Ringing Cisco SIP IP phone C to Cisco SIP IP phone A Cisco SIP IP phone C sends a SIP 180 Ringing resp...

Page 144: ...K 2 way RTP channel A is put on hold The RTP channel between A and B is torn down A and B are disconnected 4 ACK 2 way RTP channel B and C are disconnected 2 way RTP channel 5 INVITE c IN IP4 0 0 0 0 6 200 OK 7 ACK 14 BYE Also C 15 200 OK 16 INVITE C Requested by B 18 200 OK 19 ACK 17 180 Ringing 1 INVITE B 13 200 OK 11 ACK 12 BYE 8 INVITE C 9 180 Ringing 10 200 OK IP Network SIP IP Phone User A S...

Page 145: ...user is the telephone 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 user name Cisco SIP IP phone A is identified as the call session initiator in the Fr...

Page 146: ...ion 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 A and Cisco SIP IP phone B User B then selects the option to transfer the call to User C 5 INVITE Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a m...

Page 147: ...isco 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 the session description in the INVITE reques...

Page 148: ...nel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down 16 INVITE Cisco SIP IP phone A to Cisco SIP IP phone C Requested by Cisco SIP IP phone B At the request of Cisco SIP IP phone B Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone C The INVITE request is an invitation to User C to participate in a call session 17 180 Ringing Cisco SIP IP phone C to Cisco SIP I...

Page 149: ...h User B has requested unconditional call forwarding from the network When User A calls User B the call is immediately transferred to Cisco SIP IP phone C In this call flow scenario the end users are User A User B and User C 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...

Page 150: ...02 Figure B 9 Cisco SIP IP Phone to Cisco SIP IP Phone Network Call Forwarding Unconditional IP IP 7 200 OK 4 INVITE C 6 200 OK 9 ACK 5 180 Ringing 1 INVITE B 8 ACK 3 302 Moved Temporarily 2 INVITE B IP Network SIP IP Phone User A SIP IP Phone User B SIP IP Phone User C Proxy Server Redirect Server IP 41471 2 way RTP channel ...

Page 151: ... rather than a user name 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 number within a single call leg is identified in the CSeq field The media capability User A is ready to receive is specified 2 INVITE SIP proxy server to SIP redirect server SIP proxy ser...

Page 152: ...r 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 6 200 OK Cisco SIP IP phone C to SIP proxy server Cisco SIP IP phone C sends a SIP 200 OK response to the SIP proxy server 7 200 OK SIP proxy server to Cisco SIP IP pho...

Page 153: ...0 Cisco SIP IP Phone to Cisco SIP IP Phone Network Call Forwarding Busy IP IP 10 200 OK 4 INVITE B 8 180 Ringing 9 200 OK 12 ACK 7 INVITE C 5 486 Busy Here 6 ACK 1 INVITE B 11 ACK 3 300 Multiple Choices 2 INVITE B IP Network SIP IP Phone User A SIP IP Phone User B SIP IP Phone User C Proxy Server Redirect Server IP 41472 2 way RTP channel ...

Page 154: ...e user phone parameter distinquishes that the Request URI address is a telephone number rather than a user name 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 number within a single call leg is identified in the CSeq field The media capability User A is ready...

Page 155: ...ne C The INVITE request is an invitation to User C to participate in a call session 8 180 Ringing Cisco SIP IP phone C to SIP proxy server Cisco SIP IP phone C sends a SIP 180 Ringing response to the SIP proxy server 9 200 OK Cisco SIP IP phone C to SIP proxy server Cisco SIP IP phone C sends a SIP 200 OK response to the SIP proxy server 10 200 OK SIP proxy server to Cisco SIP IP phone A SIP proxy...

Page 156: ...the proxy server tries to place the call to Cisco SIP IP phone B and if there is no answer the call is transferred to Cisco SIP IP phone C In this call flow scenario the end users are User A User B and User C 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 not answered wi...

Page 157: ... Phone to Cisco SIP IP Phone Network Call Forwarding No Answer IP IP 12 200 OK 6 180 Ringing 4 INVITE B 8 200 OK 11 200 OK 10 180 Ringing 14 ACK 9 INVITE C 7 CANCEL 5 180 Ringing 1 INVITE B 13 ACK 3 300 Multiple Choices 2 INVITE B IP Network SIP IP Phone User A SIP IP Phone User B SIP IP Phone User C Proxy Server Redirect Server IP 73 2 way RTP channel ...

Page 158: ...s a telephone number rather than a user name 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 number within a single call leg is identified in the CSeq field The media capability User A is ready to receive is specified 2 INVITE SIP proxy server to SIP redirect ...

Page 159: ...o Cisco SIP IP phone C The INVITE request is an invitation to User C to participate in a call session 10 180 Ringing Cisco SIP IP phone C to SIP proxy server Cisco SIP IP phone C sends a SIP 180 Ringing response to the SIP proxy server 11 200 OK Cisco SIP IP phone C to SIP proxy server Cisco SIP IP phone C sends a SIP 200 OK response to the SIP proxy server 12 200 OK SIP proxy server to Cisco SIP ...

Page 160: ...ones in which User B mixes two RTP channels and therefore establishes a conference bridge between User A and User C In this call flow scenario the end users are User A User B and User C They are all using Cisco SIP IP phones which are connected via an IP network The call flow scenario is as follows 5 User A calls User B 6 User B answers the call 7 User B puts User A on hold 8 User B calls User C 9...

Page 161: ...0 0 0 0 1 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 cha...

Page 162: ...user is the telephone 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 user name Cisco SIP IP phone A is identified as the call session initiator in the Fr...

Page 163: ...om Cisco SIP IP phone B The ACK might contain a message body 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 A and Cisco SIP IP phone B 5 INVITE Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP...

Page 164: ...mber and host is either a domain name or a numeric network address For example the Request URI field in the INVITE request to User C 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 user name Cisco SIP IP phone B is identified as the call session initiator in the From field A unique numeri...

Page 165: ...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 the session description in the INVITE request A two way RTP channel is established between SIP IP phone B and SIP IP phone C 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...

Page 166: ...SIP IP Phone to Cisco SIP IP Phone Called User is Busy page B 66 Cisco SIP IP Phone to Cisco SIP IP Phone Called User Does Not Answer page B 68 Cisco SIP IP Phone to Cisco SIP IP Phone Authentication Error page B 70 Gateway to Cisco SIP IP Phone Called User is Busy Figure B 13 illustrates an unsuccessful call in which User A initiates a call to User B but User B is on the phone and is unable or un...

Page 167: ... 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 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 se...

Page 168: ...o PBX A 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 a...

Page 169: ...dentified 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 receive is specified The port on which the Gateway is prepared to receive the RTP data is specified 3 Call Proceeding Gateway 1 to PB...

Page 170: ...header field values 8 Disconnect Gateway 1 to PBX A Gateway 1 sends a Disconnect message to PBX A 9 Release Complete Gateway 1 to PBX A Gateway 1 sends a Release Complete message to PBX A and the call session attempt is terminated 10 200 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 confirms that User A has received the 486 B...

Page 171: ...or Figure B 15 illustrates an unsuccessful call in which User A initiates a call to User B and receives a class 4xx 5xx or 6xx response Figure B 15 Gateway to Cisco SIP IP Phone Client Server or Global Error IP SIP IP Phone User B 3 Call Proceeding 6 Disconnect 1 Setup PBX A User A GW1 IP Network 4 100 Trying 5 4xx 5xx 6xx Failure 8 ACK 2 INVITE 9 Release Complete 7 Release 41727 ...

Page 172: ...co 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 receive...

Page 173: ...sible 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 because the 6xx response indicates that a server has definite information about User B but not for the particular instance indicated in the Request URI field Therefore all further searches for this user will fail 6 Disconnect Gateway 1 to PBX A Gateway 1 sends a...

Page 174: ...P Phone Called User is Busy Figure B 16 illustrates an unsuccessful call in which User A initiates a call to User B but User B is on the phone and is unable or unwilling to take another call Figure B 16 Cisco SIP IP Phone to Cisco SIP IP Phone Called User is Busy IP IP 2 486 Busy Here 3 ACK 1 INVITE B IP Network SIP IP Phone User A SIP IP Phone User B 41475 ...

Page 175: ...sip 555 0002 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a user name 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 number within a single call leg is identified in the CSeq fie...

Page 176: ...Cisco SIP IP Phone Called User Does Not Answer Figure B 17 illustrates an unsuccessful call in which User A initiates a call to User B but User B does not answer Figure B 17 Cisco SIP IP Phone to Cisco SIP IP Phone Called User Does Not Answer IP IP 2 180 Ringing 4 200 OK 3 CANCEL 1 INVITE B IP Network SIP IP Phone User A SIP IP Phone User B 41476 ...

Page 177: ... sip 555 0002 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a user name 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 number within a single call leg is identified in the CSeq fi...

Page 178: ...ich 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 an SIP INVITE request that includes it s authentication credentials Figure B 18 Cisco SIP IP Phone to Cisco SIP IP Phone Authentication Error Proxy Server IP IP 2 407 Authentication Error 4 Resend INVITE B 3 ACK 1 INVITE B IP Network SIP IP P...

Page 179: ...rs 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 user name 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 number within a single call leg is identified in...

Page 180: ...Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls B 72 Cisco SIP IP Phone 7960 Administrator Guide 78 10497 02 ...

Page 181: ... phone Physical and Operating Environment Specifications The following table lists the physical and operating specifications of the Cisco SIP IP phone Table C 1 Cisco SIP IP Phone Operational and Physical Specifications Specification 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 2...

Page 182: ... A FCC CFR47 Part 68 FCC Part 68 EN 55022 Class A UL 1459 CSA C22 2 No 225 M90 EN 60950 1992 IEC 950 AS NZS 3260 TS001 Safety UL 1950 EN 60950 CSA C22 2 No 950 IEC 950 AS NZS 3260 TS001 Note See also Appendix D Translated Safety Warnings EMC FCC CFR Part 15 Class B ICES 003 Class B EN55022 Class B CISPR22 Class B AS NZS 3548 Class B VCCI Class B Certification CD Marking Table C 1 Cisco SIP IP Phon...

Page 183: ...evices the network port and access port You can use either Category 3 or 5 cabling for 10 Mpbs connections but use Category 5 for 100 Mbps connections 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...

Page 184: ...Appendix C Technical Specifications Connections Specifications C 4 Cisco SIP IP Phone 7960 Administrator Guide 78 10497 02 ...

Page 185: ...anwijzingen voordat u het systeem met de voeding verbindt Varoitus Lue asennusohjeet ennen järjestelmän yhdistämistä virtalähteeseen Attention Avant de brancher le système sur la source d alimentation consulter les directives d installation Warnung Lesen Sie die Installationsanweisungen bevor Sie das System an die Stromquelle anschließen Avvertenza Consultare le istruzioni di installazione prima d...

Page 186: ...sia lakeja ja säännöksiä noudattaen Attention La mise au rebut définitive de ce produit doit être effectuée conformément à toutes les lois et réglementations en vigueur Warnung Dieses Produkt muß den geltenden Gesetzen und Vorschriften entsprechend entsorgt werden Avvertenza L eliminazione finale di questo prodotto deve essere eseguita osservando le normative italiane vigenti in materia Advarsel E...

Page 187: ...rancher ou débrancher les câbles pendant un orage du foudre Warnung Arbeiten Sie nicht am 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 t...

Page 188: ...lä kytke pienjännitteisiä SELV suojapiirejä puhelinverkkojännitettä TNV käyttäviin virtapiireihin LAN portit sisältävät SELV piirejä ja WAN portit puhelinverkkojännitettä käyttäviä piirejä Osa sekä LAN että WAN porteista käyttää RJ 45 liittimiä Ole varovainen kytkiessäsi kaapeleita Attention Pour éviter une électrocution ne raccordez pas les circuits de sécurité basse tension Safety Extra Low Volt...

Page 189: ...uitos 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 Tenha o devido cuidado ao conectar os cabos Advertencia Para evitar la sacudida eléctrica no conectar circuitos de seguridad de voltaje muy bajo safety extra low voltage SELV con circuitos de voltaje de red telefónica telephone network volta...

Page 190: ...tä Attention Pour ce qui est de la protection contre les courts circuits surtension ce produit dépend de l installation électrique du local Vérifier qu un fusible ou qu un disjoncteur de 120 V alt 15 A U S maximum 240 V alt 10 A international est utilisé sur les conducteurs de phase conducteurs de charge Warnung Dieses Produkt ist darauf angewiesen daß im Gebäude ein Kurzschluß bzw Überstromschutz...

Page 191: ...rotección contra cortocircuitos o sobrecorrientes deló propio edificio Asegurarse de que se utiliza un fusible o interruptor automático de no más de 240 voltios en corriente alterna VAC 10 amperios del estándar internacional 120 VAC 15 amperios del estándar USA en los hilos de fase todos aquéllos portadores de corriente Varning Denna produkt är beroende av i byggnaden installerat kortslutningsskyd...

Page 192: ...Appendix D Translated Safety Warnings D 8 Cisco SIP IP Phone 7960 Administrator Guide 78 10497 02 ...

Page 193: ...A is a suite of network security services that provides the primary framework through which access control can be set up on your Cisco router or access server ANI Automatic number identification C CAS Channel associated signaling CCAPI Call control applications programming interface CLI Command line interface CO Central office ...

Page 194: ...ching module D dial peer An addressable call endpoint In Voice over IP V0IP there are two types of dial peers POTS and VoIP DNS Domain name system used to address translation to convert H 323 IDs URLs or e mail IDs to IP addresses DNS is also used to assist in the locating remote gatekeepers and to reverse map raw IP addresses to host names of administrative domains DNIS Dialed number identificati...

Page 195: ... other protocols by converting protocols A gateway is the point where a circuit switched call is encoded and repackaged into IP packets H H 323 An 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 22...

Page 196: ...quencies that provide 15 two frequency combinations for indication digits 0 9 and KP ST signals multicast A process of transmitting PDUs from one source to many destinations The actual mechanism that is IP multicast multi unicast and so forth for this process might be different for LAN technologies multipoint unicast A process of transferring PDUs Protocol Data Units where an endpoint sends more t...

Page 197: ...roxy interprets and if necessary rewrites a request message before forwarding it PSTN Public switched telephone network PSTN refers to the local telephone company R Redirect Server A redirect server is a server that accepts a SIP request maps the address into zero or more new addresses and returns these addresses to the client It does not initiate its own SIP request nor accept calls Registrar A r...

Page 198: ...ision multiplexing Technique in which information from multiple channels can be allocated bandwidth on a single wire based on preassigned time slots 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 u...

Page 199: ...oIP Voice over IP The ability to carry normal telephone style voice over an IP based Internet with POTs like functionality reliability and voice quality VoIP is a blanket term which generally refers to Cisco s standards based for example H 323 approach to IP voice traffic ...

Page 200: ...Glossary 8 Cisco SIP IP Phone 7960 Administrator Guide 78 10497 02 ...

Page 201: ...age header field A 10 accessing firmware version 3 33 network statistics 3 31 status messages 3 31 access port 1 14 address proxy server 3 20 TFTP server 3 4 adjusting phone placement 2 18 administrative VLAN ID parameter 3 4 Allow header field A 10 Also header field A 10 alternate TFTP server enabling 3 5 authentication name configuring 3 16 services 1 3 Authorization header field A 11 B billing ...

Page 202: ...nformation A 1 configuration erasing 3 5 configuration files default creating 2 7 example 3 15 modifying 3 8 guidelines 2 6 phone specific 2 6 2 10 creating 2 9 example 3 18 modifying 3 15 naming convention 2 6 SIPDefault cnf 2 7 storing 2 7 configuration mode entering into 3 1 locking 3 2 unlocking 3 2 configuring lines authentication name 3 16 name 3 16 password 3 16 short name 3 16 network para...

Page 203: ...e 2 14 IP address 2 14 IP subnet mask 2 14 TFTP server 2 14 releasing address 3 4 server parameter 3 3 dialing pad 1 7 directory services 1 3 DNS description 1 10 server parameters 3 4 documentation conventions xi related xi domain name parameter 3 3 Domain Name System DNS 1 10 do not disturb 1 9 downloading required files 2 4 DTMF DB level 3 10 inband 3 10 outofbound 3 10 3 20 Dynamic Host Contro...

Page 204: ... phone specific 3 18 files audio 2 4 dual boot 2 4 firmware image 2 4 OS79XX txt 2 4 RINGLIST DAT 2 4 SIPDefault cnf 2 4 firmware image 2 4 updating 3 33 version viewing 3 33 footstand adjustment 1 6 From header field A 11 functions proxy server A 2 redirect server A 2 UAC A 2 UAS A 2 G gateways 1 4 guidelines 2 13 H handset 1 7 header fields A 10 headset supported types 1 15 using 1 15 headset an...

Page 205: ...r 3 3 description 1 11 K keys on screen mode 1 7 scroll 1 7 soft 1 6 L LCD screen 1 6 line buttons 1 6 lines configuring authentication name 3 16 name 3 16 password 3 16 short name 3 16 linex_authname parameter 3 16 linex_name parameter 3 16 linex_password parameter 3 16 linex_shortname parameter 3 16 locking configuration mode 3 2 M MAC address parameter 3 3 manually configuring SIP parameters 3 ...

Page 206: ... DHCP server 3 3 domain name 3 3 erase configuration 3 5 guidelines 2 13 host name 3 3 IP address 3 3 MAC address 3 3 operational VLAN ID 3 4 subnet mask 3 3 TFTP server 3 4 port 1 14 statistics 3 31 network connections access port 1 14 O on screen mode keys 1 7 operating environment specifications C 1 operational VLAN ID parameter 3 4 Organization header field A 11 OS79XX txt 2 4 Out of Band DTMF...

Page 207: ...hentication Password 3 19 dtmf_db_level 3 10 dtmf_outofbound 3 10 image_version 3 9 linex_authname 3 16 linex_name 3 16 linex_password 3 16 linex_shortname 3 16 Message URI 3 20 Name 3 19 Out of Band DTMF 3 20 Prefered Codec 3 20 preferred_codec 3 9 proxy_register 3 11 proxy1_address 3 9 proxy1_port 3 9 Proxy Address 3 20 Proxy Port 3 20 Register Expires 3 21 Register with Proxy 3 21 required 2 8 ...

Page 208: ... 1 5 mounting to wall 2 18 overview 1 5 prerequisites 1 12 secondary directory number 1 9 supported features 1 7 supported protocols 1 10 DHCP 1 10 DNS 1 10 ICMP 1 11 IP 1 11 RTP 1 11 SDP 1 11 SNTP 1 11 TFTP 1 12 UDP 1 12 telephony features telephony 1 9 URL dialing 1 10 verifying startup 2 20 phone specific configuration file creating 2 10 example 2 9 3 18 modifying 3 15 physical specifications C...

Page 209: ... ReBy header field A 11 Record Route header field A 11 redirect server 1 5 registrar server 1 5 registration enabling 3 11 timer 3 11 3 21 related documentation xi release DHCP address 3 4 request methods B 1 Require header field A 11 resetting network statistics 3 32 Response Key header field A 11 responses A 3 global 6xx A 10 information 1xx A 4 redirection 3xx A 5 request failure 4xx A 5 server...

Page 210: ...1 11 settings erasing 3 28 short name configuring 3 16 Simple Network Time Protocol SNTP 1 11 SIP architecture 1 4 call flows B 1 successful B 2 unsuccessful B 58 clients 1 3 1 4 gateways 1 4 phones 1 4 compliance information A 1 components 1 3 UAC 1 3 user agent server 1 3 default configuration file example 2 9 dtmf_inband 3 10 end point 1 3 funtions A 2 gateways 1 4 header fields A 10 IP phone o...

Page 211: ...s configuring manually 3 18 configuring via TFTP server 2 6 SNTP description 1 11 soft keys 1 6 specifications C 1 cable C 3 connections C 3 operating environment C 1 physical C 1 specifying 3 9 codec 3 9 3 20 DTMF level 3 10 DTMF signaling 3 10 image version 3 9 proxy port 3 9 proxy server 3 9 retransmission timers 3 10 TOS media 3 9 specifying out of bound 3 20 startup verifying 2 20 statistics ...

Page 212: ...tning activity warning D 3 product disposal warning D 2 SELV circuit warning D 4 Trivial File Transfer Protocol TFTP 1 12 U UAC 1 3 UDP description 1 12 unlocking configuration mode 3 2 Unsupported header field A 12 updating firmware 3 33 URL dialing 1 10 user agent server 1 3 User Agent header field A 12 User Datagram Protocol UDP 1 12 V verifying startup 2 20 Via header field A 12 viewing firmwa...

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