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B-43

Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8.0 (SIP)

OL-7890-01

Appendix B SIP Call Flows

Call Flow Scenarios for Successful Calls

Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Backup Proxy

Figure B-16

illustrates a successful call from a Cisco SIP IP phone to a Cisco SIP IP phone that uses a SIP

backup proxy.

INVITE—Cisco SIP IP phone to
primary proxy (fifth try)

The phone retries a fifth time to connect to the proxy by sending out the
INVITE message.

INVITE—Cisco SIP IP phone to
primary proxy (sixth try)

The phone retries a sixth time to connect to the proxy by sending out the
INVITE message.

INVITE—Cisco SIP IP phone to
primary proxy (seventh try)

The phone retries a seventh time to connect to the proxy. If the connection is
not successful after this trial, the “Network Delay, Trying Backup” message
displays on the phone.

INVITE—Cisco SIP IP phone to the
gateway (backup proxy)

The phone tries to connect to the gateway (backup proxy) by sending out the
INVITE message.

Setup—Gateway to the PBX

Call setup is initiated between the gateway and the PBX. The setup includes
the standard transactions that take place as A attempts to call B.

10.

Call Proceeding—PBX to the gateway The PBX sends a Call Proceeding message to the gateway to acknowledge

the Call Setup request.

11.

100 Trying—Gateway to Cisco SIP IP
phone (A)

The gateway sends a SIP 100 Trying response to A. The response indicates
that the INVITE request has been received.

12.

Alerting—PBX to the gateway

The PBX sends an Alert message to the gateway. The message indicates that
the PBX has received a 100 Trying Ringing response from the gateway.

13.

180 Ringing—Gateway to Cisco SIP
IP phone (A)

The gateway sends a SIP 180 Ringing response to A. The response indicates
that the gateway is being alerted.

14.

Connect—PBX to the gateway

The PBX sends a Connect message to the gateway. The message notifies the
gateway that the connection has been made.

15.

200 OK—Gateway to Cisco SIP IP
phone (A)

The gateway sends a SIP 200 OK response to A. The response notifies A that
the connection has been made.

16.

ACK—Cisco SIP IP phone (A) to the
gateway

A sends a SIP ACK to the gateway. The ACK confirms that A has received
the 200 OK response. The call session is now active.

17.

Connect ACK—Gateway to the PBX

The gateway acknowledges PBX Connect message.

18.

BYE—Cisco SIP IP phone
(A) to the gateway

A terminates the call session and sends a SIP BYE request to the gateway.
The request indicates that A wants to release the call.

19.

Disconnect—Gateway to the PBX

The gateway sends a Disconnect message to the PBX.

20.

Release—PBX to the gateway

The PBX sends a Release message to the gateway.

21.

200 OK—Gateway to Cisco SIP IP
phone (A)

The gateway sends a SIP 200 OK response to A. The response notifies A that
the gateway has received the BYE request.

22.

Release Complete—Gateway to the
PBX

The gateway sends a Release Complete message to the PBX, and the call
session terminates.

Step

Action

Description

Summary of Contents for 7940G

Page 1: ... 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 Unified IP Phone 7960G and 7940G Administration Guide for Release 8 0 SIP February 2007 Text Part Number OL 7890 01 ...

Page 2: ... INDIRECT SPECIAL CONSEQUENTIAL OR INCIDENTAL DAMAGES INCLUDING WITHOUT LIMITATION LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES CCVP the Cisco Logo and the Cisco Square Bridge logo are trademarks of Cisco Systems Inc Changing the Way We Work Live Play and Learn is a ...

Page 3: ...ing Security Problems in Cisco Products xii Obtaining Technical Assistance xiii Cisco Technical Support Documentation Website xiii Submitting a Service Request xiii Definitions of Service Request Severity xiv Obtaining Additional Publications and Information xiv C H A P T E R 1 Product Overview 1 1 New Information for This Release 1 1 Cisco Unified IP Phone 7960G and 7940G Overview 1 1 Session Ini...

Page 4: ...es 3 1 Overview of the Initialization Process 3 3 About Configuration Files 3 3 How to Customize the Default Configuration File 3 4 How to Customize a Phone Specific Configuration File 3 8 Configuring the SIP Parameters Manually 3 10 Unlocking the Phone 3 10 Setting and Restoring Network Parameters 3 11 Configuring Network Parameters Manually 3 14 Setting and Restoring Phone Specific Parameters 3 ...

Page 5: ...Functions A 1 SIP Methods A 2 SIP Responses A 2 1xx Response Information Responses A 3 2xx Response Successful Responses A 3 3xx Response Redirection Responses A 3 4xx Response Request Failure Responses A 4 5xx Response Server Failure Responses A 6 6xx Response Global Responses A 6 SIP Header Fields A 6 SIP Session Description Protocol Usage A 8 Transport Layer Protocols A 8 SIP Security Authentic...

Page 6: ...e to a Cisco SIP IP Phone Using a SIP Backup Proxy B 43 Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Emergency Proxy B 45 Call Flow Scenarios for Failed Calls B 47 Gateway to Cisco SIP IP Phone in a SIP Network B 47 Called Number Is Busy B 47 Called Number Does Not Answer B 49 Client Server or Global Error B 50 Cisco SIP IP Phone to Cisco SIP IP Phone in a SIP Network B 52 Ca...

Page 7: ...rmation page xiv Overview Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8 0 SIP provides information about how to set up cable and configure a Cisco Unified IP Phone 7960G and 7940G in a SIP network It also provides information on how to configure the network and SIP parameters and change the settings and options of the Cisco IP phone The appendixes include reference info...

Page 8: ...Chapter 5 Monitoring Cisco SIP IP Phones Describes how to debug and troubleshoot Appendix A Compliance with RFC 3261 Provides reference information about Cisco SIP IP phone compliance to RFC 3261 Appendix B SIP Call Flows Provides reference information about Cisco SIP IP phone call flows Appendix C Technical Specifications of the Cisco Unified IP Phone 7960G and 7940G Provides physical and operati...

Page 9: ..._ipphon english ipp7960 index htm Session Initiation Protocol Session Initiation Protocol Gateway Call Flows http www cisco com univercd cc td doc product software ios123 rel_docs sip flo Implementing a VoIP Network Cisco IOS Voice Configuration Library http www cisco com univercd cc td doc product software ios123 123cgcr vcl htm Cisco IOS Voice Command Reference http www cisco com univercd cc td ...

Page 10: ...stance and other technical resources These sections explain how to obtain technical information from Cisco Systems Cisco com You can access the most current Cisco documentation at this URL http www cisco com techsupport You can access the Cisco website at this URL http www cisco com You can access international Cisco websites at this URL http www cisco com public countries_languages shtml Product ...

Page 11: ...Cisco direct customers can order documentation from the Ordering tool http www cisco com en US partner ordering Instructions for ordering documentation using the Ordering tool are at this URL http www cisco com univercd cc td doc es_inpck pdi htm Nonregistered Cisco com users can order documentation through a local account representative by calling Cisco Systems Corporate Headquarters California U...

Page 12: ... to delivering secure products We test our products internally before we release them and we strive to correct all vulnerabilities quickly If you think that you might have identified a vulnerability in a Cisco product contact PSIRT Emergencies security alert cisco com An emergency is either a condition in which a system is under active attack or a condition for which a severe and urgent security v...

Page 13: ...k under Documentation Tools Choose Cisco Product Identification Tool from the Alphabetical Index drop down list or click the Cisco Product Identification Tool link under Alerts RMAs The CPI tool offers three search options by product ID or model name by tree view or for certain products by copying and pasting show command output Search results show an illustration of your product with the serial n...

Page 14: ...tions is available from various online and printed sources Cisco Marketplace provides a variety of Cisco books reference guides documentation and logo merchandise Visit Cisco Marketplace the company store at this URL http www cisco com go marketplace Cisco Press publishes a wide range of general networking training and certification titles Both new and experienced users will benefit from these pub...

Page 15: ...cisco com ipj Networking products offered by Cisco Systems as well as customer support services can be obtained at this URL http www cisco com en US products index html Networking Professionals Connection is an interactive website for networking professionals to share questions suggestions and information about networking products and technologies with Cisco experts and other networking profession...

Page 16: ...xvi Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8 0 SIP OL 7890 01 Preface Obtaining Additional Publications and Information ...

Page 17: ...s and Character Set page 1 12 Supported Protocols page 1 13 Where to Go Next page 1 14 New Information for This Release The following changes have been made and the following features have been added to release 8 0 New configurable parameters have been added See Appendix D SIP IP Phone Parameters Caveats can be found on the product release notes page at this URL http www cisco com en US partner pr...

Page 18: ... proprietary telephone set and key system or PBX The Cisco Unified IP Phone 7960G and 7940G also supports an adjustable ring tone a hearing aid compatible handset and a headset The Cisco Unified IP Phone 7960G and 7940G complies with RFC 3261 as described in Appendix A Compliance with RFC 3261 See Figure 1 1 and Figure 1 2 to identify the buttons and hardware on your Cisco IP phone Figure 1 1 Cisc...

Page 19: ...s adjustment of the angle of the phone base 6 Directories button Provides access to call histories and directories 7 i or button Provides online help for selected keys or features and network statistics about the active call Pressing the button and then the up or down scroll key displays a descriptor of the key For example pressing the i or button and then the up or down scroll key displays a scre...

Page 20: ...o IP phone Cisco Unified IP Phone 7960G 7940G models have the same hardware configuration 15 Navigation button Allows scrolling through text and selection of features displayed on the LCD screen 16 Dial pad Works exactly like the dial pad on a traditional telephone 17 Softkeys Activates any functions displayed on the corresponding LCD screen tabs Softkeys point to feature options displayed along t...

Page 21: ... inline power and external power source If either the inline power or the external power goes down the phone can switch entirely to the other power source 2 Power supply with AC plug 3 Power cable with wall socket plug for connecting to power 4 Network port 10 and 100 SW RJ 45 to connect the phone to the network supporting 10 or 100 Mbps half or full duplex Ethernet connections to external devises...

Page 22: ...completed SIP establishes a session between the endpoints SIP also supports midcall changes such as the addition of another endpoint to the conference or the changing of a media characteristic or codec Handle the transfer and termination of calls SIP supports the transfer of calls from one endpoint to another During a call transfer SIP simply establishes a session between the transferee and a new ...

Page 23: ... the user when a SIP request is received 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 ...

Page 24: ...st 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 Location services are often included with redirect ser...

Page 25: ...one to see if it is operational and assess how long the response time from the phone is Traceroute support Allows you to see the path that the signal traverses in the route to its desired destination Configuration Features With the Cisco Unified IP Phone 7960G and 7940G you can do the following Configure an Ethernet port mode and speed Register with or unregister from a proxy server or backup prox...

Page 26: ... the light remains on If you listen to the new message and save or delete it the light goes off The message waiting indicator is controlled by the voice mail server The indication is saved over a phone upgrade or reboot Dial plan support for the asterisk and pound characters In previous versions of the dial plan the and characters were not matched if specified as dialed digits The character was a ...

Page 27: ...that fail to communicate with the primary proxy Once the backup proxy is used it is active for the duration of the call The location of the backup SIP proxy can be defined as an IP address in the default configuration file Emergency SIP proxy An optional emergency SIP proxy can be configured with the route attribute of the template tag in the dial plan template file When an emergency SIP proxy is ...

Page 28: ...following languages French 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 It does not support the following languages Zulu zu and other Bantu languages that use Latin Extended B letters Arabic in North Africa ...

Page 29: ...ames of endpoints into IP addresses Dynamic DNS and TFTP You can configure additional DNS and TFTP servers Upon bootup the phone first goes to the default TFTP server to download the configuration files If a new dynamic TFTP server is specified in the files the phone requests a new set of files from the specified server If new DNS addresses are specified in the files the phone uses those addresses...

Page 30: ...k Current date and time are supported using SNTP including time zone and daylight saving time The Cisco SIP IP phone uses SNTP for date and time support TCP Transmission Control Protocol Provides a reliable byte stream transfer service between endpoints on the Internet The Cisco SIP IP phone supports TCP for Telnet sessions only TFTP Trivial File Transfer Protocol Allows files to be transferred fr...

Page 31: ...P phone includes an adjustable footstand When placing the phone on a desktop surface you can adjust the tilt height to several different angles in 7 5 degree increments from flat to 60 degrees Installing the Phone on the Wall Mount the Cisco Unified IP Phone 7960G and 7940G on the wall by using the footstand as a mounting bracket or by using the optional locking bracket To use the optional locking...

Page 32: ...e phone shown in Figure 1 3 on page 1 5 Plug the appropriate equipment into the appropriate port Caution Consider use of an uninterruptible power supply UPS Without one when you use either a local transformer or inline power on the LAN the phone is inoperable during a power outage This affects your ability to make emergency calls 911 in USA and Canada 999 in the UK and 112 in mainland Europe Note ...

Page 33: ...devices PC connected to the same port Data traffic that is present on the VLAN that supports 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 connected to a phone The switch port configured for connecting a phone would have separate VLANs configured for carrying the following traffic Voice traffic ...

Page 34: ...power the phone automatically begins startup and initialization Use this section to determine how you will power your phones Do not connect the phones to power though until you address the prerequisites and follow the configuration instructions in Chapter 3 Initializing Cisco Unified IP Phones You can connect the Cisco IP phone to the following power sources External power source Optional AC adapt...

Page 35: ...se 8 0 SIP OL 7890 01 Chapter 2 Installing Cisco Unified IP Phone 7960G and 7940G Hardware on the Desktop or Wall Connecting the Phone to Power If you are using a Cisco Catalyst switch to provide power see the Using the Phone with a Cisco Catalyst Switch section on page 2 3 ...

Page 36: ... to determine where to go next in this guide See Chapter 3 Initializing Cisco Unified IP Phones for information on installing firmware customizing configuration files and connecting the phone to power sources and the network See Chapter 4 Managing Cisco SIP IP Phones for information on upgrading firmware and performing other management tasks See Chapter 5 Monitoring Cisco SIP IP Phones for informa...

Page 37: ... Date and Time page 3 18 How to Create Dial Plans page 3 23 How to Verify Initialization page 3 27 Where to Go Next page 3 28 Prerequisites Ensure that your network meets the following requirements A working IP network is established and configured for SIP For information on configuring IP refer to the Cisco IOS IP Configuration Guide http www cisco com univercd cc td doc product software ios123 1...

Page 38: ...e phone Ring list file RINGLIST DAT A file listing audio files that are the custom ring type options for the phones These audio files must also be in the root directory of the TFTP server For more information see the How to Customize Cisco Unified IP Phone 7960G and 7940G Rings section on page 4 1 Synchronization file syncinfo xml Controls the image version and associated synchronization value to ...

Page 39: ...CP server as described in Prerequisites If you are not using DHCP to configure network parameters manually configure the required network parameters as described in Configuring Network Parameters Manually Step 4 Connect the phone to the network as described in Appendix C Connection Specifications and to a power supply as described in Connecting the Phone to Power About Configuration Files Configur...

Page 40: ...llowing format variable name value optional comments Configuration file variable entries must adhere to the following rules Associate only one value with one variable Separate variable names and values with colons Set only one variable per line Indicate the end of a line with lf or cr lf Put the variable and value on the same line and do not break the line You can include white space before or aft...

Page 41: ... image version as it is released by Cisco Do not enter the extension Note You cannot change the image version by changing the filename because the version is also built into the file header Trying to change the image version by changing the filename causes the firmware to fail when it compares the version in the header against the filename proxy1_address IP address of the SIP proxy server that is ...

Page 42: ...eters added in Release 2 0 Dialplan template xml format file relative to the TFTP root directory dial_template dialplan TFTP Phone Specific Configuration File Directory tftp_cfg_dir Example sip_phone Time Server There are multiple values and configurations refer to Admin Guide for Specifics sntp_server SNTP Server IP Address sntp_mode anycast default unicast multicast or directedbroadcast time_zon...

Page 43: ...s 5060 Configurable VAD option enable_vad 0 VAD setting 0 disable Default 1 enable New Parameters added in Release 2 2 NAT Firewall Traversal nat_enable 0 0 Disabled default 1 Enabled nat_address WAN IP address of NAT box dotted IP or DNS A record only voip_control_port 5060 UDP port used for SIP messages default 5060 start_media_port 16384 Start RTP range for media default 16384 end_media_port 32...

Page 44: ...nly by using the e mail address Similarly if you configure a line to use a number that line can be called only by using the number Each line can have a different proxy configured Define the dial_template parameter in a phone specific configuration file only if that phone needs to use a different dial plan than the default For a complete alphabetical list of configurable parameters see Appendix D S...

Page 45: ... number without any dashes For example enter 555 1212 as 5551212 When entering an e mail address enter the e mail ID without the host name e Save the file to the root directory of your TFTP server or to a subdirectory that contains all the phone specific configuration files Name the file SIP mac addr cnf Type the MAC address in uppercase and the extension cnf in lowercase for example SIP00503EFFD8...

Page 46: ...light the parameter or press the number that represents the parameter located to the left of the parameter on the LCD During configuration use for dots periods or press the softkey when available on the LCD During configuration To enter a number press the Number softkey To enter a name press the Alpha softkey To enter a new value use the buttons on the dial pad If entering letters use the numbers ...

Page 47: ...e Settings menu lockable menus automatically relock Step 4 To manually relock the phone select Settings Lock Config Setting and Restoring Network Parameters To modify network parameters using the phone menus follow these steps Procedure Step 1 Unlock the Network Configuration menu See the Unlocking the Phone section on page 3 10 Step 2 Select Settings Network Configuration The Network Configuratio...

Page 48: ...ther IP address from the DHCP server When you move the phone to a new network segment first release the DHCP address DHCP Enabled Whether the phone uses DHCP to configure network settings IP address subnet mask domain name default router list DNS server list and TFTP address Valid values are Yes and No Default is Yes To manually configure your IP settings including the TFTP server s IP address set...

Page 49: ...e about the connection use the default value Using a value of PC and connecting port 2 to a switch could result in spanning tree loops and network confusion Subnet Mask2 IP subnet mask used by the phone A subnet mask partitions the IP address into a network and a host identifier TFTP Server2 IP address of the TFTP server 1 If you have an administrative VLAN setting assigned on the Cisco Catalyst s...

Page 50: ...on on page 3 10 By default the network parameters are locked to ensure that end users cannot modify settings that might affect their network connectivity 3 Review the guidelines on using the Cisco SIP IP phone menus 4 If configuring a domain name a Press the Number softkey to enter a numerical ID or press the Alpha softkey to enter a name b If entering letters use the numbers on the dial pad assoc...

Page 51: ...pecific configuration file exists the phone uses locally configured parameters until the next reboot If a phone specific configuration file does not exist you must configure the phone locally with parameters specific to that phone To configure the preferred codec and out of band DTMF parameters press Change until the option appears then press Save If your system has been set up to have the phones ...

Page 52: ...To and From fields proxy1_port Port number of the SIP proxy server that is used by line 1 If the proxy server with which the phone communicates has authentication enabled set the following parameters as well line1_authname Name used by the phone for authentication if a registration is challenged by the proxy server during initialization Default is UNPROVISIONED line1_password Password used by the ...

Page 53: ...ation is enabled the default logical password is used The default logical password is SIPmac address where mac address is the MAC address of the phone Display Name Identification as it should appear for caller identification For example instead of jdoe company com appearing on phones that have caller ID you can specify John Doe in this parameter to have John Doe appear on the callee end instead If...

Page 54: ...hone Specific Configuration File section on page 3 8 Step 3 Press the Save softkey The phone programs the new information into flash memory and resets How to Set the Date and Time You can set date time and daylight savings time DST parameters The current date and time is supported on the Cisco Unified IP Phone 7960G and 7940G using Simple Network Time Protocol SNTP and is displayed on the LCD DST ...

Page 55: ...on DST parameters as needed dst_offset dst_auto_adjust dst_start_month dst_stop_month dst_start_time dst_stop_time Step 4 Do one of the following Modify the following absolute DST parameters as needed dst_start_day dst_stop_day Modify the following relative DST parameters as needed dst_start_day_of_week dst_start_week_of_month dst_stop_day_of_week dst_stop_week_of_month Step 5 Save the file to the...

Page 56: ...irst SNTP response is received the phone switches to multicast mode Receives No known server with which to communicate Multicast data using 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 responded to the network broadcast request SNTP data from the SNTP NTPmulticast address and the local netwo...

Page 57: ...DT 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 Eastern Standard Time CDT Central Daylight Time NYC AST GMT 04 00 La Paz AST Atlantic Standard Time EDT Eastern Daylight Time NST GMT 03 30 Newfoundland NST Newfoundland Standard Time BST GMT 03 00 Buenos Aires BST Brazil Standard Time ADT Atlantic Dayli...

Page 58: ...o USSR zone5 ZP6 GMT Plus 6 Hours SUM GMT 06 30 North Sumatra NST North Sumatra Time WAST GMT 07 00 Bangkok Hanoi SST South Sumatra Time USSR zone6 WAST West Australian Standard Time HST GMT 08 00 Beijing Hong Kong CCT China Coast Time HST Hong Kong Standard Time USSR zone7 WADT West Australian Daylight Time JST GMT 09 00 Tokyo Seoul JST Japan Standard Time Tokyo KST Korean Standard Time SSR zone8...

Page 59: ...in its phone specific configuration file Special Characters You can specify the pound sign and asterisk as dialed digits if needed The is processed as a dial now event by default You can override this by specifying in the dial plan template in which case the phone does not dial immediately when the is pressed but does continue to match the dial plan template that specifies the The is not matched b...

Page 60: ...uration file Note To simplify maintenance and control define this parameter in the default configuration file Define it in a phone specific configuration file only if that phone needs to use a different dial plan than the one being used by the other phones in the same system Procedure Step 1 Using an ASCII text editor such as vi open a new file Step 2 Type the following to indicate the start of th...

Page 61: ...ore nonwildcard matches than an incomplete rule Comments are allowed with the following syntax comment Rules allow for substitution of up to five replacement strings as well as picking off of replaced digits one at a time For example a match string of ab cd ef and an input string of ab12cd34ef5678 result in the following Rewrite Output Notes s ab12cd34ef5678 0 ab12cd34ef5678 1 12 2 34 3 5678 4 Nul...

Page 62: ... DIALTEMPLATE TEMPLATE MATCH 123 45 6 TIMEOUT 0 User Phone Match TEMPLATE MATCH 34 TIMEOUT 0 User Phone Match TEMPLATE MATCH TIMEOUT 15 User Phone DIALTEMPLATE In the example above the 123 45 6 string is matched if the user dials 123 45 6 Pressing the pound sign does not cause the phone to dial immediately because is explicitly specified However dialing 1 or 123 4 causes the phone to dial immediat...

Page 63: ... MATCH 9 TIMEOUT 0 Tone Zip Play Zip Tone TEMPLATE MATCH 8 TIMEOUT 0 Tone Hold Play Hold Tone TEMPLATE MATCH 8 123 TIMEOUT 0 Tone Hold Tone Zip Play Hold Tone after 8 Play Zip Tone after 123 DIALTEMPLATE How to Verify Initialization The initialization process establishes network connectivity and makes the phone operational in your IP network Procedure Step 1 After the phone has power connected to ...

Page 64: ...splays the following Primary directory number Softkeys If the phone successfully cycles through these steps it has started up properly Where to Go Next See Chapter 4 Managing Cisco SIP IP Phones for information on upgrading firmware and performing other management tasks See Chapter 5 Monitoring Cisco SIP IP Phones for information on debugging and on viewing network statistics ...

Page 65: ...o Unified IP Phone 7960G and 7940G ships with two ring types Chirp1 and Chirp2 However you can create and add custom rings Procedure Step 1 Create a pulse code modulation PCM file for each desired ring type and store it in the root tftp directory of your TFTP server PCM files must contain no header information and must comply with the following format guidelines 8000 Hz sampling rate 8 bits per sa...

Page 66: ...s button and selecting the External Directory option The phone removes white space when Cisco CallManager Cisco IP Phone Services are displayed Multiple spaces are consolidated into a single space The phone does not allow setting of x and y coordinates for the CiscoIPPhoneImage object The image always appears at location 0 0 Centering of the image is not supported if x and y are set to 1 The phone...

Page 67: ...ow to Upgrade Your Cisco SIP IP Phone Firmware Image For instructions about how to upgrade the firmware image on a Cisco SIP IP Phone refer to Cisco IP Phone 7960 and 7940 Firmware Upgrade Matrix which is available at this URL http www cisco com univercd cc td doc product voice c_ipphon english ipp7960 addprot mgcp frmwrup htm How to Upgrade Your Cisco SIP IP Phone Firmware Image and Reboot Remote...

Page 68: ...check sync Event header Date Mon 10 Jul 2000 16 28 53 0700 Call ID 1349882 ipaddress CSeq 1300 NOTIFY Contact sip webadmin ipaddress Content Length 0 During a remote reboot the phone does the following 1 If it is idle the phone waits 20 seconds and contacts the TFTP server for the syncinfo xml and dialplan xml files Otherwise it waits until it is idle for 20 seconds and then contacts the TFTP serv...

Page 69: ...7940G Administration Guide for Release 8 0 SIP OL 7890 01 Chapter 4 Managing Cisco SIP IP Phones Where to Go Next Where to Go Next See Chapter 5 Monitoring Cisco SIP IP Phones for information on debugging and on viewing network statistics ...

Page 70: ...4 6 Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8 0 SIP OL 7890 01 Chapter 4 Managing Cisco SIP IP Phones Where to Go Next ...

Page 71: ...ones You can use Telnet or a console to connect to your Cisco Unified IP Phone 7960G and 7940G and you can and use the command line interface CLI to debug or troubleshoot the phone Table 5 1 shows the available CLI commands and their syntax Note You need the phone IP address to use the CLI in a Telnet session To get the IP address select Settings Network Configuration IP Address The default Telnet...

Page 72: ...sages sip reg state sip trx dns config sntp sntp packet http arp broadcast xml events xml deck xml vars xml post Shows detailed debug output for the following depending on the keywords used arp ARP cache console stall Console stall driver output mode cpr error Cisco Portable Runtime error conditions cdp Cisco Discovery Protocol dsp keepalive Messaging between the DSP and the main phone control str...

Page 73: ...d decks xml vars XML content variables xml post 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 Note To turn the debugging off use the undebug command works just as does the no debug command SIP Phone dns p c s ip address b ip address hostname Manipulates the DNS Keywords and arguments ar...

Page 74: ...e if you enter 0 the phone registers to the backup proxy SIP Phone reset Resets the phone line This command can be used only if the telnet_level parameter is set to allow privileged commands to be executed SIP Phone show arp cdp debug ethernet ip strpool memorymap malloc table stacks status abort_vector flash dspstate rtp tcp lsm fsm fsmdef fsmcnf fsmxfr fim gsm register reset log network config p...

Page 75: ... fim Current status of the Feature Interaction Manager control blocks interface control blocks and state control blocks gsm Global State Manager status that includes these parameters vcm lsm fim fsm and gsm register Current registration status of SIP lines reset log Debugging information about the internal state of the phone at the time that it was last restarted network Network information such a...

Page 76: ...the test functionality close Disables the use of the test functionality test command keywords continued key Simulates key presses The arguments k1 through k12 are as follows k1 voldn Volume down k2 volup Volume up k3 headset Headset k4 spkr Speaker k5 mute Mute k6 info Info k7 msgs Messages k8 serv Services k9 dir Directories k10 set Settings k11 navup Navigate up k12 navdn Navigate down Note You ...

Page 77: ...om the SIP IP phone to the specified IP address The arguments are as follows ip address Dotted IP address or alphanumeric address host name of the host to which you are sending the traceroute ttl Optional Time to live value or the number of routers hops through which the datagram can pass Default is 30 SIP Phone tty echo on off mon time value kill session msg prompt Controls the Telnet system Argu...

Page 78: ...ode Phone1 show strpool node_id refcount string 1 1 sip 48 10 18 192 230 2 1 sip 48 10 18 192 230 3 1 sip 47 10 18 192 230 4 1 sip 47 10 18 192 230 5 1 sip 46 10 18 192 230 6 1 sip 46 10 18 192 230 7 1 sip duval 10 18 192 230 8 1 sip duval 10 18 192 230 9 1 sip 44 10 18 192 230 10 1 sip 44 10 18 192 230 11 1 sip 43 10 18 192 230 12 1 sip 43 10 18 192 230 13 1 1234 14 1 25640 15 1 26295 10 18 192 2...

Page 79: ...974 stklo 00491973 Size 4096 Unused 1012 Task NET 19 stkhi 00491974 stklo 00492173 Size 2048 Unused 536 Task CFG 18 stkhi 00492174 stklo 00492973 Size 2048 Unused 912 Task TTY 17 stkhi 00492974 stklo 00493973 Size 4096 Unused 3036 Task AUD 16 stkhi 00493974 stklo 00494173 Size 2048 Unused 1724 Task PTMR 28 stkhi 00494174 stklo 00494973 Size 2048 Unused 1932 Task TMR 27 stkhi 004bb60c stklo 004bbe0...

Page 80: ...Chksum OK chksum 00001c7a applen 00016d90 cmpchksum 00000000 cmplen 00000000 DSP Status The following sample output shows the status of the DSPs Phone1 show dspstate DSP State READY DSP Audio mode None DSP IsStreaming flag False Keep Alive Pending False Ringer state Off number 2 volume dB 17 Progress tone state Off Number of DSP resets since boot 0 Times DSP was not able to get a buffer 0 Volumes ...

Page 81: ...1 AttFail 00000000 EstRsts 00000000 CurrEstab 00000001 InSegs 00000530 OutSegs 00000330 RetransSegs 00000000 OutPeer 00000011 InErrs 00000000 OutRsts 00000001 PktBufErrs 00000000 Telnet Stats Conn 1 Throttles 00000000 Conn 2 Throttles 00000000 Dial Plan Configuration The following sample output shows the dial plan Phone1 show dialplan Dialplan is 01 Pattern 0 Rewrite Timeout 0001 UserMode Phone Ro...

Page 82: ...9 02 3838 6 326 L1 T45 2003 7 8 11 08 53 53 10 10 10 0 7 65 L1 T45 2003 6 26 14 42 49 54 8 53 L1 T45 2003 3 31 17 04 17 5550100 9 6 L1 T45 2002 12 20 13 42 50 5550110 Kazoo 9 Phone 10 13 L1 T45 2002 8 29 16 38 14 9195550111 11 6 L1 T45 2002 3 1 12 37 29 9195550111 abc com Fid Mantel 12 12 L1 T45 2002 1 7 17 42 10 9195550111 13 6 L1 T45 2003 7 9 17 07 54 5550111 14 5 L1 T45 2002 3 8 17 19 59 ciscot...

Page 83: ...708 UNDEFINED IDLE 0x00000000 0x00000000 9 0 0x004e2724 UNDEFINED IDLE 0x00000000 0x00000000 10 0 0x004e2740 UNDEFINED IDLE 0x00000000 0x00000000 11 0 0x004e275c UNDEFINED IDLE 0x00000000 0x00000000 12 0 0x004e2778 UNDEFINED IDLE 0x00000000 0x00000000 13 0 0x004e2794 UNDEFINED IDLE 0x00000000 0x00000000 14 0 0x004e27b0 UNDEFINED IDLE 0x00000000 0x00000000 15 0 0x004e27cc UNDEFINED IDLE 0x00000000 ...

Page 84: ...2990 0x00000000 0x004e29a4 0x00000000 0x004e284c 15 0 DEF 0x004e29a4 0x00000000 0x00000000 0x00000000 0x004e285c 16 0 HEAD 0x004e29b8 0x004e2a08 0x004e29cc 0x00000000 0x004e282c 17 0 CNF 0x004e29cc 0x00000000 0x004e29e0 0x00000000 0x004e283c 18 0 XFR 0x004e29e0 0x00000000 0x004e29f4 0x00000000 0x004e284c 19 0 DEF 0x004e29f4 0x00000000 0x00000000 0x00000000 0x004e285c 20 0 HEAD 0x004e2a08 0x0000000...

Page 85: ...Status Flags 12300000 Running Network Configuration The following sample output shows the running configuration Phone1 show network running Network RUNNING Configuration Platform Cisco IP Phone 7960 Elasped Time 00 18 11 dhcp_server 10 18 192 230 my_ip_addr 10 18 199 14 subnet_mask 255 255 255 0 defaultgw 10 18 199 1 dyn_dns_addr_1 0 0 0 0 dyn_dns_addr_2 0 0 0 0 dns_addr 10 18 192 48 tftp_addr 10 ...

Page 86: ...tatus Flags 12300000 image_version P0S3 05 8 10 FirmLoadID PC13K030 DSPLoadID PS03AT36 network_media_type Half10 network_port2_type Hub Switch tos_media 5 phone_label user4X tftp_cfg_dir phone_password phone_prompt Phone1 language english sntp_mode DirectedBroadcast sntp_server 10 10 10 150 time_zone EST dst_offset 1 dst_start_month April dst_start_day 0 dst_start_day_of_week Sun dst_start_week_of...

Page 87: ... UNPROVISIONED line5_authname UNPROVISIONED line6_authname UNPROVISIONED line1_password line2_password line3_password line4_password line5_password line6_password line1_shortname UNPROVISIONED line2_shortname UNPROVISIONED line3_shortname UNPROVISIONED line4_shortname UNPROVISIONED line5_shortname UNPROVISIONED line6_shortname UNPROVISIONED line1_displayname user43 line2_displayname user44 line3_d...

Page 88: ...peed_label2 speed_line3 speed_label3 speed_line4 speed_label4 speed_line5 speed_label5 speed_line6 speed_label6 IP Statistics The following sample output shows the IP statistics Phone1 show ip IP Statistics Received 00002623 RxDrops 00000006 RxFrags 00000000 RxFragDrops 00000000 RxReassembled 00000000 Transmitted 00000869 TxDrops 00000000 TxFragments 00000000 Use clear ip to clear data How to Use ...

Page 89: ...by the phone Phone State Message TCP messages that indicate the state of the phone The following are possible messages Phone Initialized TCP connection has not gone down since the phone was powered on Phone Closed TCP TCP connection was closed by the phone TCP Timeout TCP connection was closed because of a retry timeout Error Code Error messages that indicate unusual reasons for which the TCP conn...

Page 90: ...ication that the network is in a linked state and has autonegotiated a half duplex 100 Mbps connection Port 1 Full 10 Indication that the network is in a linked state and has autonegotiated a full duplex 10 Mbps connection Port 1 Half 10 Indication that the network is in a linked state and has autonegotiated a half duplex 10 Mbps connection Step 3 Select Exit Note To reset the values power the pho...

Page 91: ...t 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 6 SIP Session Description Protocol Usage page A 8 Transport Layer Protocols page A 8 SIP Security Authentication page A 8 SIP DNS Records Usage page A 9 SIP DTMF Digit Transport page A 9 SIP Functions Function Supported User agent client UAC Yes User agent ...

Page 92: ...Responses page A 6 6xx Response Global Responses page A 6 Note A SIP response that cannot be constructed properly because of a syntactically incorrect received request will be ignored A syntactically incorrect response to a sent request will be ignored Method Supported Comments INVITE Yes The phone supports midcall changes such as putting a call on hold as signaled by a new INVITE that contains an...

Page 93: ...one processes these responses in the same way that it processes the 100 Trying response 182 Queued 183 Session Progress The phone does not generate this message Upon receiving this response the phone provides early media cut through and then waits for a 200 OK response 2xx Response Supported Comments 200 OK Yes 202 Accepted Yes 3xx Response Supported Comments 300 Multiple Choices Yes 301 Moved Per...

Page 94: ...ves a 405 Method Not Allowed response it notifies the user of the response 406 Not Acceptable Yes The phone does not generate a 406 Not Acceptable response For an incoming response the phone initiates a graceful call disconnect during which the caller hears a busy or fast busy tone before clearing the call request 407 Proxy Authentication Required Yes If a 407 Proxy Authorization Required response...

Page 95: ...quire field the 420 Bad Extension response is generated 480 Temporarily Unavailable Yes The response is received only by the phone The user is notified if this response is received 481 Call Leg Transaction Does Not Exist Yes The user is notified if this response is received 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous Yes If a new contact is received the phone might rei...

Page 96: ... is notified 501 Not Implemented Yes 502 Bad Gateway Yes 503 Service Unavailable Yes 504 Gateway Timeout Yes 505 Version Not Supported Yes 6xx Response Supported Comments 600 Busy Everywhere Yes The phone initiates a graceful call disconnect and the user is notified 603 Decline Yes 604 Does Not Exist Anywhere Yes 606 Not Acceptable Yes Header Field Supported Accept Yes Accept Encoding Yes Accept L...

Page 97: ...Forwards Yes Mime Version Yes Organization No Priority No Proxy Authenticate Yes Proxy Authorization Yes Proxy Require Yes Record Route Yes Referred By Yes Refer To Yes Remote Party ID Yes Note The Screen parameter is ignored on received Remote Party ID and is always set to Yes in sent Remote Party ID Replaces Yes Requested By Yes Require Yes Response Key No Retry After Yes Route Yes RTP RxStat Ye...

Page 98: ...upported 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 information Yes m Media name and transport address Yes a Media attribute lines Yes Protocol Supported Unicast UDP Yes Multicast UDP No TCP No Basic Authentication Supported Digest Authe...

Page 99: ...de for Release 8 0 SIP OL 7890 01 Appendix A Compliance with RFC 3261 SIP DNS Records Usage SIP DNS Records Usage SIP DTMF Digit Transport DNS Resource Record Type Supported Type A Yes Type SRV Yes NAPTR No Transport Type Supported RFC 2833 Yes In band tones Yes ...

Page 100: ...A 10 Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8 0 SIP OL 7890 01 Appendix A Compliance with RFC 3261 SIP DTMF Digit Transport ...

Page 101: ...P 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 reboot and message waiting indication MWI The following types of responses are used by SIP and generated by the Cisco SIP gateway SIP 1xx Informational Responses SIP 2xx Successful Responses SIP 3xx Redir...

Page 102: ...scribe and illustrate successful calls in a gateway to a Cisco SIP IP phone Call Setup and Disconnect page B 2 Call Setup and Hold page B 4 Call to a Gateway Acting As an Emergency Proxy from a Cisco SIP IP Phone page B 7 Call Setup and Disconnect Figure B 1 illustrates a successful 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 ...

Page 103: ...scenario is the IP phone In the INVITE request The IP address of the 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 that User A is ready to receiv...

Page 104: ...esponse from the 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 phone sends a SIP 200 OK response to Gateway 1 The response notifies Gateway 1 that the connection has been made 8 Connect Gateway 1 to PBX A Gateway 1 sends a Connect message to the PBX A The message notifies PBX A that the connection has been made 9 Conne...

Page 105: ... B 2 Successful Call Setup and 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 a sendonly 14 INVITE a sendrecv 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 137201 ...

Page 106: ... 7960G and 7940G to Gateway 1 The phone sends a SIP 100 Trying response to Gateway 1 The response indicates that the INVITE request has been received 5 180 Ringing Cisco SIP IP hone to Gateway 1 The phone sends a SIP 180 Ringing response to Gateway 1 The response indicates that the user is being alerted 6 Alerting Gateway 1 to the PBX A Gateway 1 sends an Alert message to User A The message indica...

Page 107: ...phone to Gateway 1 User B takes User A off hold Phone B sends a SIP INVITE request to Phone A with the same call ID as the previous INVITE and a new SDP attribute parameter sendrecv which is used to reestablish the call 15 200 OK Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP 200 OK response to the phone The response notifies the phone that the INVITE was successfully processed 16 ACK Cisco...

Page 108: ...PBX to the gateway The PBX sends an Alert message to the gateway The message indicates that the PBX has received a 100 Trying Ringing response from the gateway 6 180 Ringing Gateway to Cisco SIP IP phone User A The gateway sends a SIP 180 Ringing response to User A The response indicates that the gateway is being alerted 7 Connect PBX to the gateway The PBX sends a Connect message to the gateway T...

Page 109: ...to a Cisco SIP IP Phone Using a SIP Backup Proxy page B 43 Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone Using a SIP Emergency Proxy page B 45 Simple Call Hold Figure B 4 illustrates a successful call between Cisco SIP IP phones in which one of the participants places the other on hold and then returns to the call In this call flow scenario the two end users are User A and User B User A a...

Page 110: ... in the INVITE request to User B appears as INVITE sip 555 0199 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a username 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 c...

Page 111: ...Phone A sends a SIP ACK to Phone B The ACK confirms that Phone A has received the 200 OK response from Phone B The ACK might contain a message body with the final session description to be used by Phone B If the message body of the ACK is empty Phone B uses the session description in the INVITE request A two way RTP channel is established between Phone A and Phone B 5 INVITE Phone B to Phone A Pho...

Page 112: ...tation IP IP IP 2 180 Ringing 3 200 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 a sendonly 6 200 OK 7 ACK 14 INVITE a sendrecv 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 Ph...

Page 113: ... Phone A The response notifies Phone A that the connection has been made If Phone B supports the media capability advertised in the INVITE message sent by Phone A it advertises the intersection of its own and Phone A s media capability in the 200 OK response If Phone B does not support the media capability advertised by Phone A it sends back a 400 Bad Request response with a 304 Warning header fie...

Page 114: ...lity advertised by Phone A it sends back a 400 Bad Request response with a 304 Warning header field 11 ACK Phone B to Phone C Phone B sends a SIP ACK to Phone C The ACK confirms that Phone B has received the 200 OK response from Phone C The ACK might contain a message body with the final session description to be used by Phone C If the message body of the ACK is empty Phone C uses the session desc...

Page 115: ...with C on 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 a sendonly 8 200 OK 9 ACK 15 INVITE a sendrecv 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 a sendrec...

Page 116: ...o Phone A The response notifies Phone A that the connection has been made If Phone B supports the media capability advertised in the INVITE message sent by Phone A it advertises the intersection of its own and Phone A media capability in the 200 OK response If Phone B does not support the media capability advertised by Phone A it sends back a 400 Bad Request response with a 304 Warning header fiel...

Page 117: ...User A off hold Phone B sends a SIP INVITE request to Phone A with the same call ID as the previous INVITE and a new SDP attribute parameter sendrecv which is used to reestablish the call 16 200 OK Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B 17 ACK Phone B to Phone A Phone B sends a SIP ACK to Phone A The ACK confirms that Phone B has received the 200 OK response from Phone A...

Page 118: ...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 Figure B 7 Call Transfer Without Consultation IP Network SIP IP Phone User B IP IP IP 2 100 TRYING 3 180 RINGING 4 200 OK 12 BYE 13 200 OK 14 INVITE Referred By B 15 100 TRYING 16 180 RINGING 17 200 OK 18 ACK ...

Page 119: ...equest has been received 3 180 Ringing Phone B to Phone A Phone B sends a SIP 180 Ringing response to Phone A 4 200 OK Phone B to Phone A Phone B sends a SIP 200 OK response to Phone A The response notifies Phone A that the connection has been made If Phone B supports the media capability advertised in the INVITE message sent by Phone A it advertises the intersection of its own and Phone A s media...

Page 120: ...s started 12 BYE Phone B to Phone A Phone B sends a BYE message to Phone A The message indicates that Phone B will disconnect from the call 13 200 OK Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B The response notifies Phone B that the BYE message was received 14 INVITE Phone A to Phone C Because of the REFER message from Phone B Phone A sends a SIP INVITE request to Phone C The...

Page 121: ...Failover 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 a sendonly 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 137205 Use...

Page 122: ...uest has been received 3 180 Ringing Phone B to Phone A Phone B sends a SIP 180 Ringing response to Phone A 4 200 OK Phone B to Phone A Phone B sends a SIP 200 OK response to Phone A The response notifies Phone A that the connection has been made If Phone B supports the media capability advertised in the INVITE message sent by Phone A it advertises the intersection of its own and Phone A s media c...

Page 123: ...cates that the REFER message is not supported and that Phone B should failover to Bye Also 11 BYE Phone B to Phone A Phone B sends a BYE message to Phone A The message includes the following information Also C The message indicates that the 501 Not Implemented message was received in response to a REFER message 12 200 OK Phone A to Phone B Phone A sends a SIP 200 OK response to Phone B The respons...

Page 124: ...OK 11 180 RINGING 12 200 OK 13 ACK 2 way voice path 14 INVITE a sendrecv 15 200 OK 16 ACK 17 REFER Refer To C Replaces B Referred By B 18 202 ACCEPTED 19 NOTIFY Event Refer Subscription State Active 20 INVITE Referred by B Replaces B 21 200 OK 22 ACK 23 BYE 24 200 OK 25 NOTIFY Event Refer Subscription State Terminated 26 200 OK 27 BYE 28 200 OK 2 way voice path 5 ACK 2 way voice path User B presse...

Page 125: ...request has been received 3 180 Ringing Phone B to Phone A Phone B sends a SIP 180 Ringing response to Phone A 4 200 OK Phone B to Phone A Phone B sends a SIP 200 OK response to Phone A The response notifies Phone A that the connection has been made If Phone B supports the media capability advertised in the INVITE message sent by Phone A it advertises the intersection of its own and Phone A s medi...

Page 126: ...ne C The ACK might contain a message body with the final session description to be used by Phone C If the message body of the ACK is empty Phone C uses the session description in the INVITE request A two way RTP channel is established between Phone B and Phone C User B then selects the option to transfer the call to User C 14 INVITE Phone B to Phone C Phone B sends a midcall INVITE to Phone C with...

Page 127: ... The request contains the following information Referred By B Replaces B 21 200 OK Phone C to Phone A Phone C sends a SIP 200 OK response to Phone A The response notifies Phone A that the INVITE request has been received 22 ACK Phone A to Phone C Phone A sends a SIP ACK to Phone C The ACK confirms that Phone A has received the 200 OK response from Phone C 23 BYE Phone C to Phone B Phone C sends a ...

Page 128: ...G 4 200 OK 11 180 RINGING 12 200 OK 13 ACK 2 way voice path 14 INVITE a sendonly 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 a sendonly 7 200 OK 8 ACK 9 INVITE C 10 100 TRY...

Page 129: ...0 Ringing Phone B to Phone A Phone B sends a SIP 180 Ringing response to Phone A 4 200 OK Phone B to Phone A Phone B sends a SIP 200 OK response to Phone A The response notifies Phone A that the connection has been made If Phone B supports the media capability advertised in the INVITE message sent by Phone A it advertises the intersection of its own and Phone A s media capability in the 200 OK res...

Page 130: ...tablished between Phone B and Phone C User B then selects the option to transfer the call to User C 14 INVITE Phone B to Phone C Phone B sends a midcall INVITE to Phone C with new Session Description Protocol SDP attribute parameter SDP a sendonly The a SDP field of the SIP INVITE contains sendonly This value places the call on hold 15 200 OK Phone C to Phone B Phone C sends a SIP 200 OK response ...

Page 131: ...ends a SIP BYE request to Phone C 22 200 OK Phone C to Phone B Phone C sends a SIP 200 OK response to Phone B The response notifies Phone B that the BYE request has been received 23 INVITE Phone A to Phone C Phone A sends a SIP INVITE request to Phone C The request contains the following information Requested By B The message indicates that the INVITE was requested by Phone B 24 100 Trying Phone C...

Page 132: ...address For example the Request URI field in the INVITE request to User B appears as INVITE sip 555 0199 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a username 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...

Page 133: ...edirect server sends a SIP 302 Moved temporarily message to the SIP proxy server The message indicates that User B is not available at Phone B and includes instructions to locate User B at Phone C 4 ACK SIP proxy server to redirect server The SIP proxy server sends the SIP ACK to the SIP redirect server 5 INVITE SIP proxy server to Phone C The SIP proxy server sends a SIP INVITE request to Phone C...

Page 134: ...os for Successful Calls Figure B 12 Network Call Forwarding Busy IP IP 11 200 OK 5 INVITE B 9 180 Ringing 10 200 OK 13 ACK 8 INVITE C 6 486 Busy Here 7 ACK 1 INVITE B 12 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 41472 2 way RTP channel ...

Page 135: ...s a SIP 300 Multiple choices message to the SIP proxy server The message indicates that User B can be reached either at Phone B or Phone C 4 ACK SIP proxy server to redirect server The SIP proxy server sends the SIP ACK to the SIP redirect server 5 INVITE SIP proxy server to Phone B The SIP proxy server sends a SIP INVITE request to Phone B The request is an invitation to User B to participate in ...

Page 136: ...o 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 the phone Phone B is not answered within a set amount of time the network should forward incoming calls to Phone C 2 User A calls User B 3 User B does not answered 4 The network transfers the call to Phone C Figure ...

Page 137: ...00 Multiple Choices SIP redirect server to SIP proxy server The 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 Phone B or Phone C 4 ACK SIP proxy server to redirect server The SIP proxy server sends the SIP ACK to the SIP redirect server 5 INVITE SIP proxy server to Phone B The SIP proxy server sends a...

Page 138: ...ey are all using Cisco Unified IP Phone 7960G and 7940G models which are connected using 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 14 ACK Phone A to SIP proxy server Phone A sends a SIP ACK to the SIP proxy server The ACK confirms that...

Page 139: ...endonly 1 INVITE B Call ID 1 3 200 OK 2 180 Ringing 6 200 OK 12 INVITE A Call ID 1 a sendrecv 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 137208 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...

Page 140: ...a SIP 180 Ringing response to Phone A 3 200 OK Phone B to Phone A Phone B sends a SIP 200 OK response to Phone A The response notifies Phone A that the connection has been made If Phone B supports the media capability advertised in the INVITE message sent by Phone A it advertises the intersection of its own and Phone A s media capability in the 200 OK response If Phone B does not support the media...

Page 141: ... specified 9 180 Ringing Phone C to Phone B Phone C sends a SIP 180 Ringing response to Phone B 10 200 OK Phone C to Phone B Phone C sends a SIP 200 OK response to Phone B The response notifies Phone B that the connection has been made If Phone C supports the media capability advertised in the INVITE message sent by Phone B it advertises the intersection of its own and Phone B s media capability i...

Page 142: ... the proxy by sending out the INVITE message 3 INVITE Cisco SIP IP phone to primary proxy third try The 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 The 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 r...

Page 143: ... SIP 100 Trying response to A The response indicates that the INVITE request has been received 12 Alerting PBX to the gateway The PBX sends an Alert message to the gateway The message indicates that the PBX has received a 100 Trying Ringing response from the gateway 13 180 Ringing Gateway to Cisco SIP IP phone A The gateway sends a SIP 180 Ringing response to A The response indicates that the gate...

Page 144: ...SIP IP phone A to primary proxy third try The phone retries a third time to connect to the primary proxy by sending out the INVITE message 4 INVITE Cisco SIP IP phone A to primary proxy fourth try The phone retries a fourth time to connect to the primary proxy by sending out the INVITE message 5 INVITE Cisco SIP IP phone A to primary proxy fifth try The phone retries a fifth time to connect to the...

Page 145: ...een received 12 180 Ringing Cisco SIP IP phone B to backup proxy Phone B sends a SIP 180 Ringing response to the backup proxy The response indicates that B is being alerted 13 180 Ringing Backup proxy to Cisco SIP IP phone A The backup proxy sends a SIP 180 Ringing response to A The response indicates that the backup proxy is being alerted 14 200 OK Cisco SIP IP phone B to backup proxy Phone B sen...

Page 146: ...he emergency proxy The response indicates that the INVITE request has been received 5 180 Ringing Cisco SIP IP phone B to emergency proxy User B sends a SIP 180 Ringing response to the emergency proxy The response indicates that User B is being alerted 6 180 Ringing Emergency proxy to Cisco SIP IP phone A The emergency proxy sends a SIP 180 Ringing response to User A The response indicates that th...

Page 147: ...r unwilling to take another call 9 ACK Cisco SIP IP phone A to emergency proxy A acknowledges the emergency proxy s Connect message 10 ACK Emergency proxy to Cisco SIP IP phone B The emergency proxy acknowledges B s Connect message 11 BYE Cisco SIP IP phone A to emergency proxy A terminates the call session and sends a SIP BYE request to the emergency proxy The request indicates that A wants to re...

Page 148: ...d in the Call ID field The transaction number within a single call leg is identified in the CSeq field The media capability 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 Setup request 4 100 Trying Cisco SIP IP phone to G...

Page 149: ...t Answer 8 ACK Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP ACK to the phone The ACK confirms that A has received the 486 Busy Here response The call session attempt terminates 9 Release Complete Gateway 1 to PBX A Gateway 1 sends a Release Complete message to PBX A and the call session attempt terminates Step Action Description IP SIP IP Phone User B 3 Call Proceeding 11 Release Complete...

Page 150: ... 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 Setup request 4 100 Trying Cisco SIP IP phone to Gateway 1 The phone sends a SIP 100 Trying response to Gateway 1 The response indicates that the INVITE request has been received 5 180 Ringing Cisco SIP IP phone to Gateway 1 The p...

Page 151: ... request The IP address of the 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 A is ready to receive is specified The port on which the gateway is ...

Page 152: ...If the 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 phone sends a class 6xx failure response a global error the search for B terminates because the 6xx response indicates that a server has definite information about B but not for the particular instance indicated in the Reque...

Page 153: ...her a domain name or a numeric network address For example the Request URI field in the INVITE request to B appears as INVITE sip 555 0199 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a username Phone A is identified as the call session initiator in the From field A unique numeric identifier is assigned to the call an...

Page 154: ...etwork address For example the Request URI field in the INVITE request to B appears as INVITE sip 555 0199 companyb com user phone The user phone parameter distinquishes that the Request URI address is a telephone number rather than a username 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 fie...

Page 155: ... a call session In the INVITE request The phone number of B is inserted in the Request URI field in the form of a SIP URL The SIP URL identifies the address of User B and takes a form similar to an e mail address user host where 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 INV...

Page 156: ...s Call Flow Scenarios for Failed Calls 3 ACK Phone A to SIP proxy server Phone A sends a SIP ACK to the SIP proxy server acknowledging the 407 error message 4 Resend INVITE Cisco SIP IP phone A to SIP proxy server Phone A resends a SIP INVITE to the SIP proxy server with authentication credentials Step Action Description ...

Page 157: ...sco Unified IP Phone 7960G and 7940G Table C 1 Cisco Unified IP Phone 7960G and 7940G 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 to 240 VAC 5...

Page 158: ...utory information on Cisco IP phone models in the 7900 series refer to Regulatory Compliance and Safety Information for the Cisco IP Phone 7900 Series at http www cisco com univercd cc td doc product voice c_ipphon english ipp7960 iphrcsi3 htm Connection Specifications The Cisco Unified IP Phone 7960G and 7940G has two RJ 45 ports that each support 10 100 Mbps half or full duplex connections to ex...

Page 159: ...nonymous calls are received 1 Enabled by default but can be turned on and off using the user interface When enabled anonymous calls are rejected 2 Disabled permanently and cannot be turned on and off locally using the user interface Specify this parameter in the phone specific configuration file 3 Enabled permanently and cannot be turned on and off locally using the user interface Specify this par...

Page 160: ... parameter for a specific phone configure it in that phone s configuration file Default is 1 call_hold_ringback Phone specific optional If you have a call on hold and are talking on another call when you hang up the call this parameter causes the phone to ring letting you know that you still have another party on hold Valid values are as follows 0 Off by default but can be turned on and off locall...

Page 161: ...ackets received not including late RTP packets AvgJit Average jitter which is an estimate of the statistical variance of the RTP packet inter arrival time measured in timestamp unit and calculated according to RFC 1889 a b c d e f g h and i Integers call_waiting Phone specific optional Configures call waiting Valid values are as follows 0 Disabled by default but can be turned on and off using the ...

Page 162: ...ration file 3 Enabled permanently and cannot be turned on and off locally using the user interface Specify this parameter in the phone specific configuration file Default is 0 cfwd_url Optional Configures the call forwarding feature The maximum allowable characters for the string is 128 The character can be a telephone number or a URL Note This parameter cannot be set in the configuration file cnf...

Page 163: ... On permanently and cannot be turned on and off locally using the user interface This setting sets the phone to be a call out phone only Default is 0 Domain Name Name of the DNS domain in which the phone resides dscpForAudio Differentiated Services Code Point DSCP specifies the class of service for each audio packet Valid value is any positive integer Default is 184 dst_auto_adjust Optional Whethe...

Page 164: ...f month in which DST begins Valid values are 1 to 6 and 8 1 is the first week each subsequent number is a subsequent week 8 is the last week in the month regardless of which week the last week is In the United States the default is 1 dst_stop_day Optional Day of the month on which DST ends Valid values are as follows 1 to 31 for the days of the month 0 ignore this field and use the value in the ds...

Page 165: ...6 to 127 If the value specified is null or invalid default is 101 dtmf_db_level Optional In band DTMF digit tone level Valid values are as follows 1 6 dB below nominal 2 3 dB below nominal 3 nominal 4 3 dB above nominal 5 6 dB above nominal Default is 3 dtmf_inband Obsolete dtmf_outofband Optional Configures the out of band signaling for tone detection on the IP side of a gateway Note The Cisco SI...

Page 166: ... new dynamic TFTP server After initially querying the default TFTP server the phone rerequests the default and phone specific configuration files from the new TFTP server The dynamic TFTP server address is not stored in flash memory The number of dyn_tftp_addr values supported by the phone is limited to prevent the phone configuration being downloaded repeatedly from multiple TFTP servers Only dot...

Page 167: ...e default name is used Default is UNPROVISIONED The x argument can be 1 or 2 linex_contact Phone specific optional The URL that appears in the contact field for SIP messages on a particular line x is a line number on the phone Default value is null linex_displayname Phone specific optional Identification as it should appear for caller identification purposes For example instead of jdoe company com...

Page 168: ...il number that is dialed when the messages button is pressed Value is typically a phone number but can be a URI mwi_status Optional Displays the message waiting status Note You cannot set this parameter in the configuration file nat_address Optional WAN IP address of the Network Address Translation NAT or firewall server Value is either a dotted IP address or a DNS name nat_enable Optional Enables...

Page 169: ...tiated 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 Default is Auto network_port2_type Optional Configures the device type that is connected to port 2 of the phone Valid values are Hub Swit...

Page 170: ...the source IP address phone_label Phone specific optional Text to display on the top right status line of the LCD This field is for end user display only and has no effect on caller identification or messaging For example a phone label can display User A s phone Limited to 11 characters phone_password Phone specific optional Password to be used for console or Telnet access Limited to 31 characters...

Page 171: ... parameter is provisioned with an FQDN the phone sends REGISTER and INVITE messages by using the FQDN in the Req URI To and From fields If the value of x is not specified in the proxyx_address parameter the phone uses proxy1_address as the default value proxyx_port Port number of the SIP proxy server that will be used by phone lines other than line 1 The x variable represents a phone line Valid va...

Page 172: ...er in the Max Forwards header of the SIP requests that it generates Default is 70 sip_retx Optional Maximum number of times that a SIP message other than an INVITE request will be retransmitted Valid value is any positive integer Default is 10 sntp_mode Optional Mode in which the phone listens for the SNTP server Valid values are unicast multicast anycast or directedbroadcast Default is anycast sn...

Page 173: ...al Whether a 12 or 24 hour time format is displayed by default on the user interface Valid values are as follows 0 12 hour format is displayed by default but can be changed to a 24 hour format using the user interface 1 24 hour format is displayed by default but can be changed to a 12 hour format using the user interface 2 12 hour format is displayed and cannot be changed to a 24 hour format using...

Page 174: ...eger greater than timer_t1 Default is 4000 tos_media Obsolete transfer_onhook_enabled If enabled a user can transfer a call to a second user by hanging up the phone once the second user picks up the line If disabled a user must before hanging up press the Transfer button again to complete the transfer Valid values are 0 Disabled 1 Enabled Default is 0 user_info Phone specific optional Configures t...

Page 175: ...te TFTP server Default is No Default Router 1 to 5 IP address 1 of the default gateway used by the phone and 2 to 5 of the gateways that the phone attempts to use as an alternate gateway if the default gateway is unavailable Note Default Router 1 always takes precedence If Router 1 is unavailable the phone moves down the list of available routers in order based on the configuration in Routers 2 th...

Page 176: ...ts the default and phone specific configuration files from the new TFTP server The dynamic TFTP server address is not stored in flash memory Erase Configuration Whether to erase all of the locally defined network settings on the phone and reset the values to the defaults Valid values are Yes and No Yes reenables DHCP HTTP Proxy Address IP address of the HTTP proxy server You can use either a dotte...

Page 177: ... 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 SIPmac address where mac address is the MAC address of the phone Note Required when registration is enabled and the registrar challenges registration Display Name Phone specific Identification as...

Page 178: ...e parameter is the e mail address username company com you can specify the username to have just the username 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 Table D 3 SIP Parameters continued Parameter Description Table D 4 Manual SIP Parameter Configuration Parameter Description Line 1 se...

Page 179: ...server during initialization Valid values are 0 disable registration during initialization and 1 enable registration during initialization Default is 0 Note You can also use this parameter in a phone specific configuration file After a phone has initialized and registered with a proxy server you can remove the registration by changing this value to 0 in the phone specific configuration file To rei...

Page 180: ...e port when nat_enable 1 Range is from 1025 to 65535 Default is 5060 Start Media Port Optional Start RTP range for media Range is from 16384 to 32766 Default is 16384 End Media Port Optional Configures the Real Time Transport Protocol RTP end range for media Valid values are 16384 to 32766 Default is 32766 Backup Proxy Optional IP address of the backup proxy server or gateway Enter this address in...

Page 181: ...and outbound proxy modes can be independently enabled or disabled The received tag is added to the Via header of all responses if there is no received tag in the uppermost Via header and if the source IP address is different from the IP address in the uppermost Via header Keep the following rules in mind If a received tag is in the uppermost Via header the response is sent back to the IP address c...

Page 182: ...ss is invalid or UNPROVISIONED the Contact header appears as follows Contact sip lineN_name phone_ip_address voip_co ntrol_port and the Via header appears as follows Via SIP 2 0 UDP phone_ip_address voip_control_port If NAT is enabled the Session Description Protocol SDP message uses the nat_address and an RTP port between the start_media_port and the end_media_port range in the C and M fields All...

Page 183: ... b Oct c LatePkt d LostPkt e Av gJit f RTP TxStat Dur g Pkt h Oct i where the following apply Dur Total number of seconds since the beginning of reception or transmission Pkt Total number of RTP packets received or transmitted Oct Total number of RTP payload octets received or transmitted not including RTP header LatePkt Total number of late RTP packets received LostPkt Total number of lost RTP pa...

Page 184: ...D 26 Cisco Unified IP Phone 7960G and 7940G Administration Guide for Release 8 0 SIP OL 7890 01 Appendix D SIP IP Phone Parameters ...

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