Table Of Contents
Session Initiation Protocol (SIP) Integration Guide for Cisco Unity 4.0
Integration Tasks
Task List to Create the Integration
Task List to Make Changes to an Integration
Task List to Delete an Existing Integration
Requirements
Integration Description
Call Information
Integration Functionality
Integrations with Multiple Phone Systems
Planning How the Voice Messaging Ports Will Be Used by Cisco Unity
Preparing for Programming the Phone System
Programming the SIP Phone System
Configure the SIP Gateway Servicing Cisco Unity for the SIP Integration
Creating a New Integration with the SIP Phone System
Testing the Integration
Integrating a Secondary Server for Cisco Unity Failover
Requirements
Integration Description
Setting Up the Secondary Server for Failover
Changing the Settings for an Existing Integration
Deleting an Existing Integration
Appendix: Compatibility of Phone System Components
Appendix: Using Alternate Extensions and MWIs
Alternate Extensions
Setting Up Alternate Extensions
Alternate MWIs
Setting Up Alternate MWIs
Appendix: Documentation and Technical Assistance
Conventions
Obtaining Documentation
Cisco.com
Product Documentation DVD
Ordering Documentation
Documentation Feedback
Cisco Product Security Overview
Reporting Security Problems in Cisco Products
Obtaining Technical Assistance
Cisco Technical Support & Documentation Website
Submitting a Service Request
Definitions of Service Request Severity
Obtaining Additional Publications and Information
Session Initiation Protocol (SIP) Integration Guide for Cisco Unity 4.0
Revised April 4, 2006
This document provides instructions for integrating the phone system with Cisco Unity.
Integration Tasks
Before doing the following tasks to integrate Cisco Unity with the SIP phone system, confirm that the Cisco Unity server is ready for the integration by completing the applicable tasks in the applicable Cisco Unity installation guide.
The following task lists describe the process for creating, changing, and deleting integrations.
Task List to Create the Integration
Use the following task list to set up a new integration with the SIP phone system. If you are installing a new Cisco Unity server by using the applicable Cisco Unity installation guide, you may have already completed some of the following tasks.
1. 
Review the system and equipment requirements to confirm that all phone system and Cisco Unity server requirements have been met. See the "Requirements" section.
2. Plan how the voice messaging ports will be used by Cisco Unity. See the "Planning How the Voice Messaging Ports Will Be Used by Cisco Unity" section.
3. Program the SIP proxy server and other call processing components. See the "Programming the SIP Phone System" section.
4. Set up the SIP gateway that services Cisco Unity. See the "Configure the SIP Gateway Servicing Cisco Unity for the SIP Integration" section.
5. Create the integration. See the "Creating a New Integration with the SIP Phone System" section.
6. Test the integration. See the "Testing the Integration" section.
7. If you have a secondary server for Cisco Unity failover, integrate the secondary server. See the "Integrating a Secondary Server for Cisco Unity Failover" section.
Task List to Make Changes to an Integration
Use the following task list to make changes to an integration after it has been created.
1. Start the Cisco Unity Telephony Integration Manager (UTIM). See the "Changing the Settings for an Existing Integration" section.
2. Make the changes you want to the existing integration. See the "Changing the Settings for an Existing Integration" section.
Task List to Delete an Existing Integration
Use the following task list to remove an existing integration.
1. Start the Cisco Unity Telephony Integration Manager (UTIM). See the "Deleting an Existing Integration" section.
2. Delete the existing integration. See the "Deleting an Existing Integration" section.
Requirements
The SIP integration supports configurations of the following components:
Phone System
•SIP proxy server (Cisco SIP Proxy Server).
•SIP-enabled phones (for example, SIP-enabled Cisco IP Phone 7960 or Pingtel xpressa).
The SIP phones must use the REFER method for call transfers.
•SIP-enabled gateways (for example, Cisco AS5300 Access Server, Cisco 2600 series router, or Cisco 3600 series router) for access to the PSTN.
For details on compatibility of the phone system components with the integration, see the "Appendix: Compatibility of Phone System Components" section.
Cisco Unity Server
•Cisco Unity installed and ready for the integration, as described in the applicable Cisco Unity installation guide at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_installation_guides_list.html.
•A license that enables the applicable number of voice messaging ports.
Network Configuration
•Cisco Unity server, SIP proxy server, SIP-enabled phones, and SIP-enabled gateways installed on the same subnet (ensures adequate bandwidth and avoids latency issues affecting integration behavior).
Integration Description
The SIP integration uses the SIP proxy server to set up communications between the voice messaging ports on the Cisco Unity server and the applicable end point (for example, a SIP-enabled phone). The communications occur through:
•An IP network (LAN, WAN, or Internet) to all SIP-enabled devices connected to it.
•A SIP-enabled gateway to the PSTN and all phones connected to it.
Figure 1 shows the connections.
Figure 1 Connections Between the SIP Phone System and Cisco Unity
Call Information
The proxy server sends the following information in the SIP message with the calls forwarded:
•In the Diversion header, the extension of the called party
•In the Diversion header, the reason for the forward (the extension is busy, does not answer, or is set to forward all calls)
•In the From header, the extension of the calling party (for internal calls) or the SIP URL of the calling party (if it is an external call and the system uses caller ID)
Cisco Unity uses this information to answer the call appropriately. For example, a call forwarded to Cisco Unity is answered with the personal greeting of the subscriber. If the phone system routes the call to Cisco Unity without this information, Cisco Unity answers with the opening greeting.
Integration Functionality
The SIP integration with Cisco Unity provides the following integration features:
•Call forward to personal greeting
•Call forward to busy greeting
•Caller ID
•Easy message access (a subscriber can retrieve messages without entering an ID because Cisco Unity identifies the subscriber based on the extension from which the call originated; a password may be required)
•Identified subscriber messaging (Cisco Unity identifies the subscriber who leaves a message during a forwarded internal call, based on the extension from which the call originated)
•Message waiting indication (MWI)
Integrations with Multiple Phone Systems
Depending on the version, Cisco Unity can be integrated with two or more phone systems:
•Cisco Unity 4.0 and 4.1 can be integrated with a maximum of two phone systems at one time. For information on and instructions for integrating Cisco Unity with two phone systems, refer to the Dual Phone System Integration Guide at http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/integuid/multi/itmultin.htm.
•Cisco Unity 4.2 and later can be integrated with two or more phone systems at one time. For information on the maximum supported combinations and instructions for integrating Cisco Unity with multiple phone systems, refer to the Multiple Phone System Integration Guide at http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/integuid/multi/multcu42.htm.
Planning How the Voice Messaging Ports Will Be Used by Cisco Unity
Before programming the phone system, you need to plan how the voice messaging ports will be used by Cisco Unity.
Unlike other integrations, the hunt group mechanism for SIP integrations is implemented on the Cisco Unity server. Within an integration cluster, each incoming call hunts for an available voice messaging port among all the ports in a round-robin (or circular) fashion. If a voice messaging port in the cluster is set not to answer calls or is not enabled, a call reaching that port may receive a busy signal.
The following table describes the voice messaging port settings in Cisco Unity that can be set in UTIM, and that are displayed as read-only text on the System > Ports page of the Cisco Unity Administrator.
Table 1 Settings for the Voice Messaging Ports
Field
|
Considerations
|
Extension
(not shown for Cisco Unity 4.0(4) and later)
|
Cisco Unity automatically enters the extension. Accept the default unless a different extension is needed.
|
Enabled
|
Check this check box.
|
Answer Calls
|
Check this check box.
Caution All voice messaging ports connecting to the SIP proxy server must have the Answer Calls box checked. Otherwise, calls to Cisco Unity may not be answered.
|
Message Notification
|
Check this check box to designate the port for notifying subscribers of messages.
|
Dialout MWI
(not shown for Cisco Unity 4.0(4) and later)
|
Check this check box to designate the port for turning MWIs on and off.
|
AMIS Delivery
(available with the AMIS licensed feature only)
|
Check this check box to designate the port for making outbound AMIS calls to deliver voice messages from Cisco Unity subscribers to users on another voice messaging system. Cisco Unity supports the Audio Messaging Interchange Specification (AMIS) protocol, which provides an analog mechanism for transferring voice messages between different voice messaging systems.
This setting affects outbound AMIS calls only. All ports are used for incoming AMIS calls.
Because the transmission of outgoing AMIS messages may tie up voice ports for long periods of time, you may want to adjust the schedule on the Network > AMIS > Schedule page so that outgoing AMIS calls are placed during closed hours or at times when Cisco Unity is not processing many calls.
|
TRAP Connection
|
Check this check box so that subscribers can use the phone as a recording and playback device in Cisco Unity web applications and e-mail clients.
|
The Number of Voice Messaging Ports to Install
The number of voice messaging ports to install depends on numerous factors, including:
•The number of calls Cisco Unity will answer when call traffic is at its peak.
•The expected length of each message that callers will record and that subscribers will listen to.
•The number of subscribers.
•The number of calls made for message notification.
•The number of MWIs that will be activated when call traffic is at its peak.
•The number of AMIS delivery calls.
•The number of TRAP connections needed when call traffic is at its peak. (TRAP connections are used by Cisco Unity web applications and e-mail clients to play back and record over the phone.)
•The number of calls that will use the automated attendant and call handlers when call traffic is at its peak.
It is best to install only the number of voice messaging ports that are needed so that system resources are not allocated to unused ports.
The Number of Voice Messaging Ports That Will Answer Calls
The calls that the voice messaging ports answer can be incoming calls from unidentified callers or from subscribers. Assign all of the voice messaging ports to answer calls.
You can set voice messaging ports to both answer calls and to dial out (for example, to set MWIs).
The Number of Voice Messaging Ports That Will Dial Out
Ports that will dial out can do one or more of the following:
•Notify subscribers by phone, pager, or e-mail of messages that have arrived.
•Turn MWIs on and off for subscriber extensions.
•Make outbound AMIS calls to deliver voice messages from Cisco Unity subscribers to users on another voice messaging system. (This action is available only with the AMIS licensed feature.)
•Make a TRAP connection so that subscribers can use the phone as a recording and playback device in Cisco Unity web applications and e-mail clients.
Preparing for Programming the Phone System
Record your decisions about the voice messaging ports to guide you in programming the phone system.
Programming the SIP Phone System
If you use programming options other than those supplied in the following procedure, the performance of the integration may be affected.
Do the following procedure.
To Program the SIP Phone System
Step 1 Install and set up the SIP proxy server as described in the server documentation.
Step 2 Program each phone to forward calls to <the contact line name>@<SIP proxy server>, the voice messaging line name that subscribers will use to contact Cisco Unity.
Step 3 If Cisco Unity will authenticate with the SIP proxy server, enter a subscriber record for the contact line name that Cisco Unity will use.
Note You can use alternate extensions to create multiple line appearances, enable easy message access from cell phones, and simplify addressing messages to subscribers at different locations in Cisco Unity. Enabling alternate MWIs lets Cisco Unity turn MWIs on at more than one extension. This feature can also be used to enable alphanumeric extension numbers in Cisco Unity. For details, see the "Appendix: Using Alternate Extensions and MWIs" section.
Configure the SIP Gateway Servicing Cisco Unity for the SIP Integration
To configure the SIP gateway for the SIP integration with Cisco Unity, do the following three procedures.
To Configure Application Session on the Sip Gateway
Step 1 On the VoIP dial-peer servicing Cisco Unity, use the following command:
Step 2 Create a destination pattern that matches the voice messaging port numbers. For example, if the system has voice messaging ports 1001 through 1016, enter the dial-peer destination pattern 10xx.
Step 3 Repeat Step 1 and Step 2 for all remaining VoIP dial-peers servicing Cisco Unity.
To Disable the SIP Media Inactivity Timer
Step 1 On the gateway, go into the gateway configuration mode by entering the following command:
Step 2 Disable the RTCP timer by entering the following command:
Router(config-gateway)# no timer receive-rtcp
Step 3 Exit the gateway configuration mode by entering the following command:
Router(config-gateway)# exit
To Enable DTMF Relay for SIP Calls by Using Named Telephony Events
Step 1 On the gateway, go into dial-peer configuration mode and define the VoIP dial peer by entering the following command:
Router(config)# dial-peer voice <dial peer number> voip
Step 2 Configure the SIP protocol on the gateway by entering the following command:
Router(config-dial-peer)# session protocol sipv2
Step 3 Enable DTMF relay using NTE RTP packets by entering the following command:
Router(config-dial-peer)# dtmf-relay rtp-nte
Step 4 Configure the type of payload in the NTE packet by entering the following command:
Router(config-dial-peer)# rtp payload-type nte <NTE packet payload type>
Creating a New Integration with the SIP Phone System
After ensuring that the SIP phone system and the Cisco Unity server are ready for the integration, do the following procedures to set up the integration and to enter the port settings.
To Create an Integration
Step 1 If UTIM is not already open, on the Windows Start menu of the Cisco Unity server, click Programs > Cisco Unity > Manage Integrations. UTIM appears.
Step 2 In the left pane of the UTIM window, click Cisco Unity Server.
Step 3 On the Integration menu of the UTIM window, click New. The Telephony Integration Setup Wizard appears.
Step 4 On the Welcome page, click the applicable phone system type, depending on your version of Cisco Unity:
•Cisco Unity 4.2 or later—SIP (including Cisco Unified CallManager)
•Cisco Unity 4.0 or 4.1—SIP
Step 5 Click Next.
Step 6 On the Name This SIP Integration and Cluster page, enter the following settings, then click Next.
Table 2 Settings for the Name This SIP Integration and Cluster Page
Field
|
Setting
|
Integration Name
|
<the name you will use to identify this SIP integration; accept the default name or enter another name>
|
Cluster Name
|
<the name you will use to identify this SIP server cluster; accept the default name or enter another name>
|
Step 7 On the Enter Primary and Secondary SIP Server page, enter the following settings, then click Next.
Table 3 Settings for the Enter Primary and Secondary SIP Server Page
Field
|
Setting
|
Primary:
IP Address/Name
|
<the IP address of the primary SIP server that you are connecting to Cisco Unity>
|
Primary:
Port
|
<the IP port of the primary SIP server that you are connecting to Cisco Unity>
|
Secondary:
IP Address/Name
|
<optional; the IP address of the secondary SIP server that you are connecting to Cisco Unity>
|
Secondary:
Port
|
<optional; the IP port of the secondary SIP server that you are connecting to Cisco Unity>
|
You can click Ping Servers to confirm that the IP address is correct.
Step 8 On the Set Number of Voice Messaging Ports page, enter the number of voice messaging ports on Cisco Unity that you want to connect to the SIP server, then click Next.
This number must not be more than the number of ports set up on the SIP server.
Step 9 On the Configure Cisco Unity SIP Settings page, enter the following settings, then click Next.
Table 4 Settings for the Configure Cisco Unity SIP Settings Page
Field
|
Setting
|
Contact Line Name
|
<the voice messaging line name that subscribers will use to contact Cisco Unity and that Cisco Unity uses to register with the SIP server>
|
Cisco Unity SIP Port
|
<the IP port on Cisco Unity that callers and the SIP server use to connect to voice mail; we recommend using the default setting>
|
Preferred Codec
|
<the codec Cisco Unity will first attempt to use on outgoing calls>
|
Preferred Transport Protocol
|
Click UDP.
|
Step 10 On the Enter SIP Server Authentication page, enter the following settings, then click Next.
Table 5 Settings for the Enter SIP Server Authentication Page
Field
|
Setting
|
Authenticate with the SIP Server
|
<your indication whether Cisco Unity will authenticate with the SIP server>
|
Name
|
<the name the SIP server will use for authentication>
|
Password
|
<the password the SIP server will use for authentication>
|
Step 11 If other integrations already exist, the Enter Trunk Access Code page appears. Enter the extra digits that Cisco Unity must use to transfer calls through the gateway to extensions on the other phone systems with which it is integrated. Then click Next.
Step 12 (Cisco Unity 4.2 and later only) On the Reassign Subscribers page, any subscribers whose phone system integration has been deleted and who are not currently assigned to a phone system integration will appear in the list.
If no subscribers appear in the list, click Next and continue to Step 13.
Otherwise, select the subscribers that you want to assign to this phone system integration and click Next. You can use the following selection controls for selecting subscribers.
Table 6 Selection Controls for the Reassign Subscribers Page
Selection Control
|
Effect
|
Check All
|
Checks the check boxes for all subscribers in the list.
|
Uncheck All
|
Unchecks the check boxes for all subscribers in the list.
|
Toggle Selected
|
For the subscribers highlighted in the list, toggles between checking and unchecking the check boxes.
If some highlighted subscriber check boxes are checked and others are unchecked, clicking this button will check all the check boxes. Clicking again will uncheck all the check boxes.
|
Step 13 (Cisco Unity 4.2 and later only) On the Reassign Call Handlers page, any call handlers whose phone system integration has been deleted and that are not currently assigned to a phone system integration will appear in the list.
If no call handlers appear in the list, click Next and continue to Step 14.
Otherwise, select the call handlers that you want to assign to this phone system integration and click Next. You can use the following selection controls for selecting call handlers.
Table 7 Selection Controls for the Reassign Call Handlers Page
Selection Control
|
Effect
|
Check All
|
Checks the check boxes for all call handlers in the list.
|
Uncheck All
|
Unchecks the check boxes for all call handlers in the list.
|
Toggle Selected
|
For the call handlers highlighted in the list, toggles between checking and unchecking the check boxes.
If some highlighted call handler check boxes are checked and others are unchecked, clicking this button will check all the check boxes. Clicking again will uncheck all the check boxes.
|
Step 14 On the Completing page, verify the settings you entered, then click Finish.
Step 15 At the prompt to restart the Cisco Unity services, click Yes. The Cisco Unity services restart.
Alternatively, you can restart the Cisco Unity services in UTIM on the Tools menu by clicking Restart Cisco Unity.
Unlike other integrations, the hunt group mechanism for SIP integrations is implemented on the Cisco Unity server. Within an integration cluster, each incoming call hunts for an available voice messaging port among all the ports in a round-robin fashion. If a voice messaging port in the cluster is set not to answer calls or is not enabled, a call reaching that port may receive a busy signal.
To Enter the Voice Messaging Port Settings for the Integration
Step 1 After the Cisco Unity services restart, on the View menu, click Refresh.
Step 2 In the left pane of the UTIM window, expand the phone system integration that you are creating.
Step 3 In the left pane, click the name of the phone system.
Step 4 In the right pane, click the Ports tab.
Step 5 Enter the settings shown in Table 8 for the voice messaging ports.
Table 8 Settings for the Voice Messaging Ports
Field
|
Considerations
|
Extension
(not shown for Cisco Unity 4.0(4) and later)
|
Cisco Unity automatically enters the extension. Accept the default unless a different extension is needed.
|
Enabled
|
Check this check box.
|
Answer Calls
|
Check this check box.
Caution All voice messaging ports connecting to the SIP proxy server must have the Answer Calls box checked. Otherwise, calls to Cisco Unity may not be answered.
|
Message Notification
|
Check this check box to designate the port for notifying subscribers of messages.
|
Dialout MWI
(not shown for Cisco Unity 4.0(4) and later)
|
Check this check box to designate the port for turning MWIs on and off.
|
AMIS Delivery
(available with the AMIS licensed feature only)
|
Check this check box to designate the port for making outbound AMIS calls to deliver voice messages from Cisco Unity subscribers to users on another voice messaging system. Cisco Unity supports the Audio Messaging Interchange Specification (AMIS) protocol, which provides an analog mechanism for transferring voice messages between different voice messaging systems.
This setting affects outbound AMIS calls only. All ports are used for incoming AMIS calls.
Because the transmission of outgoing AMIS messages may tie up voice ports for long periods of time, you may want to adjust the schedule on the Network > AMIS > Schedule page so that outgoing AMIS calls are placed during closed hours or at times when Cisco Unity is not processing many calls.
|
TRAP Connection
|
Check this check box so that subscribers can use the phone as a recording and playback device in Cisco Unity web applications and e-mail clients.
|
Step 6 Click Save.
Step 7 Exit UTIM.
When integrated with a SIP phone system, Cisco Unity can use a non-20 ms packet size. See the following table to determine whether you must enable a non-20 ms packet size for your system.
Cisco Unity 4.0(4) or earlier
|
Non-20 ms packet sizes are not supported. Skip to the "Testing the Integration" section.
|
Cisco Unity 4.0(5) or later that uses only the 20 ms packet size
|
Non-20 ms packet sizes are not needed. Skip to the "Testing the Integration" section.
|
Cisco Unity 4.0(5) or later that uses a non-20 ms packet size
|
If your system meets one of the following conditions, you must enable a non-20 ms packet size on Cisco Unity:
•The SIP endpoints do not support ptime, and the SIP phone system uses a non-20 ms packet size.
•You want Cisco Unity to initiate calls with a non-20ms packet size.
Continue to the procedure "To Enable a Non-20 ms Packet Size on Cisco Unity" procedure.
Note Cisco Unity does not support different packet-size intervals for sending and receiving.
|
To Enable a Non-20 ms Packet Size on Cisco Unity
Step 1 On the Windows Start menu, click Run.
Step 2 Enter regedit and click OK. The Registry Editor window appears.
Caution Changing the wrong registry key or entering an incorrect value can cause the server to malfunction. Before you edit the registry, confirm that you know how to restore it if a problem occurs. (Refer to the "Restoring" topics in Registry Editor Help.) Note that for Cisco Unity failover, registry changes on one Cisco Unity server must be made manually on the other Cisco Unity server, because registry changes are not replicated. If you have any questions about changing registry key settings, contact Cisco TAC.
Step 3 If you do not have a current backup of the registry, click Registry > Export Registry File, and save the registry settings to a file.
Step 4 Select the key HKEY_LOCAL_MACHINE\Software\ActiveVoice\MIU\1.0\Initialization\Integrations\Integration<n> where <n> is the number of the SIP integration.
Step 5 On the Edit menu, click New > DWORD Value.
Step 6 In the right pane, for the name of the new value, enter G711 Packetization and press Enter.
Step 7 On the Edit menu, click New > DWORD Value.
Step 8 In the right pane, for the name of the new value, enter G729 Packetization and press Enter.
Step 9 Double-click the G711 Packetization value.
Step 10 In the Edit DWORD Value dialog box, Under Base, click Decimal.
Step 11 In the Value Data field, enter one of the following values that you want to use for the packet size (in ms) for the G.711 codec:
•10
•20
•30
Caution If you enter a setting other than one that appears in the list above, Cisco Unity will use the 20 ms default packet size.
Step 12 Click OK.
Step 13 Double-click the G729 Packetization value.
Step 14 In the Edit DWORD Value dialog box, Under Base, click Decimal.
Step 15 In the Value Data field, enter one of the following values that you want to use for packet size (in ms) for the G.729 codec:
•10
•20
•30
•40
•50
•60
Caution If you enter a setting other than one that appears in the list above, Cisco Unity will use the 20 ms default packet size.
Step 16 Click OK.
Step 17 Close the Registry Editor window.
Step 18 Restart the Cisco Unity server for these changes to take effect.
Step 19 If Cisco Unity is configured for failover, repeat this procedure on the secondary server.
When a non-20 ms packet size is enabled and depending on the situation, Cisco Unity will use the following packet sizes:
•When the initial SDP offer does not contain a ptime attribute, Cisco Unity will use the enabled packet size.
•When the initial SDP offer contains a ptime attribute, Cisco Unity will use the requested packet size.
•When Cisco Unity initiates the initial SDP offer, Cisco Unity will use the enabled packet size.
Testing the Integration
To test whether Cisco Unity and the phone system are integrated correctly, do the following procedures in the order listed. If you are testing Cisco Unity versions 4.0(3) and earlier, however, skip the procedures that use supervised transfers (as indicated) because Cisco Unity versions 4.0(3) and earlier do not support supervised transfers.
If any of the steps indicate a failure, refer to the following documentation as applicable:
•The installation guide for the phone system.
•Cisco Unity Troubleshooting Guide, available at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_troubleshooting_guides_list.html.
•The setup information earlier in this guide.
To Set Up the Test Configuration
Step 1 Set up two test extensions (Phone 1 and Phone 2) on the same phone system that Cisco Unity is connected to.
Step 2 Set Phone 1 to forward calls to the Cisco Unity pilot number when calls are not answered.
Caution The phone system must forward calls to the Cisco Unity pilot number in no fewer than four rings. Otherwise, the test may fail.
Step 3 In the Cisco Unity Administrator, create a test subscriber to use for testing by doing the applicable substeps below.
If your message store is Exchange, do the following:
a. In the Cisco Unity Administrator, go to the Subscribers > Subscribers > Profile page.
b. Click the Add icon.
c. Select New Exchange Subscriber.
d. On the Add Subscriber page, enter the applicable information.
e. Click Add.
If your message store is IBM Lotus Domino, do the following:
a. In the Cisco Unity Administrator, go to the Subscribers > Subscribers > Profile page.
b. Click the Add icon.
c. Click Notes.
d. In the Address Book list, confirm that the address book listed is the one that contains the user data that you want to import.
If the address book that you want to use is not listed, go to the System > Configuration > Subscriber Address Books page and add a different address book.
e. In the Find Domino Person By list, indicate whether to search by short name, first name, or last name.
f. Enter the applicable short name or name. You also can enter * to display a list of all users, or enter one or more characters followed by * to narrow your search.
g. Click Find.
h. On the list of matches, click the name of the user to import.
i. On the Add Subscriber page, enter the applicable information.
j. Click Add.
Step 4 In the Extension field, enter the extension of Phone 1.
Step 5 In the Active Schedule field, click All Hours - All Days.
Step 6 Click the Save icon.
Step 7 In the navigation bar, click Call Transfer to go to the Subscribers > Subscribers > Call Transfer page for the test subscriber.
For more information on transfer settings, refer to the "Subscriber Template Call Transfer Settings" section in the Cisco Unity Administrator Help.
Step 8 Under Transfer Incoming Calls, click Yes, Ring Subscriber's Extension, and confirm that the extension number is for Phone 1.
Step 9 Under Transfer Type, click Release to Switch.
Step 10 Click the Save icon.
Step 11 In the navigation bar, click Messages to go to the Subscribers > Subscribers > Messages page for the test subscriber.
Step 12 Under Message Waiting Indicators (MWIs), check Use MWI for Message Notification.
Step 13 In the Extension field, enter x.
Step 14 Click the Save icon.
Step 15 Open the Status Monitor by doing one of the following:
•In Internet Explorer, go to http://<Cisco Unity server name>/web/sm.
•Double-click the desktop shortcut to the Status Monitor.
•In the status bar next to the clock, right-click the Cisco Unity tray icon and click Status Monitor.
To Test an External Call with Release Transfer
Step 1 From Phone 2, enter the access code necessary to get an outside line, then enter the number outside callers use to dial directly to Cisco Unity.
Step 2 On the Status Monitor, note which port handles this call.
Step 3 When you hear the opening greeting, enter the extension for Phone 1. Hearing the opening greeting means that the port is configured correctly.
Step 4 Confirm that Phone 1 rings and that you hear a ringback tone on Phone 2. Hearing a ringback tone means that Cisco Unity correctly released the call and transferred it to Phone 1.
Step 5 Leaving Phone 1 unanswered, confirm that the state of the port handling the call changes to "Idle." This state means that release transfer is successful.
Step 6 Confirm that, after the number of rings that the phone system is set to wait, the call is forwarded to Cisco Unity and that you hear the greeting for the test subscriber. Hearing the greeting means that the phone system forwarded the unanswered call and the call-forward information to Cisco Unity, which correctly interpreted the information.
Step 7 On the Status Monitor, note which port handles this call.
Step 8 Leave a message for the test subscriber and hang up Phone 2.
Step 9 On the Status Monitor, confirm that the state of the port handling the call changes to "Idle." This state means that the port was successfully released when the call ended.
Step 10 Confirm that the MWI on Phone 1 is activated. The activated MWI means that the phone system and Cisco Unity are successfully integrated for turning on MWIs.
To Test Listening to Messages
Step 1 From Phone 1, enter the internal pilot number for Cisco Unity.
Step 2 When asked for your password, enter the default password. Hearing the request for your password means that the phone system sent the necessary call information to Cisco Unity, which correctly interpreted the information.
Step 3 Confirm that you hear the recorded voice name for the test subscriber (if you did not record a voice name for the test subscriber, you will hear the extension number for Phone 1). Hearing the voice name means that Cisco Unity correctly identified the subscriber by the extension.
Step 4 When asked whether you want to listen to your messages, press 1.
Step 5 After listening to the message, press 3 to delete the message.
Step 6 Confirm that the MWI on Phone 1 is deactivated. The deactivated MWI means that the phone system and Cisco Unity are successfully integrated for turning off MWIs.
Step 7 Hang up Phone 1.
Step 8 On the Status Monitor, confirm that the state of the port handling the call changes to "Idle." This state means that the port was successfully released when the call ended.
If you are testing Cisco Unity versions 4.0(3) and earlier, skip to the "To Delete the Test Subscriber (All Cisco Unity Versions)" procedure because Cisco Unity versions 4.0(3) and earlier do not support supervised transfers.
If you are testing Cisco Unity versions 4.0(4) and later, continue with the following procedure.
To Set Up Supervised Transfer on Cisco Unity (Cisco Unity Version 4.0(4) and Later Only)
Step 1 In the Cisco Unity Administrator, go to Subscribers > Subscribers > Call Transfer.
If the name of the test subscriber is not displayed, click the Find icon (the magnifying glass) in the title bar, then click Find, and select the name of the test subscriber in the list that appears.
For more information on transfer settings, refer to the "Subscriber Template Call Transfer Settings" section in the Help for the Cisco Unity Administrator.
Step 2 Under Transfer Type, click Supervise Transfer.
Step 3 Set the Rings to Wait For field to 3.
Step 4 Click the Save icon.
To Test Supervised Transfer (Cisco Unity Version 4.0(4) and Later Only)
Step 1 From Phone 2, enter the access code necessary to get an outside line, then enter the number outside callers use to dial directly to Cisco Unity.
Step 2 On the Status Monitor, note which port handles this call.
Step 3 When you hear the opening greeting, enter the extension for Phone 1. Hearing the opening greeting means that the port is configured correctly.
Step 4 Confirm that Phone 1 rings and that you do not hear a ringback tone on Phone 2. Instead, you should hear the indication your phone system uses to mean that the call is on hold (for example, music or beeps).
Step 5 Leaving Phone 1 unanswered, confirm that the state of the port handling the call remains "Busy." This state and hearing an indication that you are on hold mean that Cisco Unity is supervising the transfer.
Step 6 Confirm that, after three rings, you hear the greeting for the test subscriber. Hearing the greeting means that Cisco Unity successfully recalled the supervised-transfer call.
Step 7 During the greeting, hang up Phone 2.
Step 8 On the Status Monitor, confirm that the state of the port handling the call changes to "Idle." This state means that the port was successfully released when the call ended.
To Delete the Test Subscriber (All Cisco Unity Versions)
Step 1 In the Cisco Unity Administrator, go to the Subscribers > Subscribers > Profile page.
If the name of the test subscriber is not displayed, click the Find icon (the magnifying glass) in the title bar, then click Find, and select the name of the test subscriber in the list that appears.
Step 2 In the title bar, click the Delete Subscriber icon (the X).
Step 3 Click Delete.
Integrating a Secondary Server for Cisco Unity Failover
The Cisco Unity failover feature enables a secondary server to provide voice messaging services when the primary server becomes inactive. For information on installing a secondary server for failover, refer to the applicable Cisco Unity installation guide, available at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_installation_guides_list.html.
For information on failover, refer to the Cisco Unity Failover Guide. The Domino version of the guide is available at http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/fail/fail401/dom/index.htm. The Exchange version of the guide is available at http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/fail/fail401/ex/index.htm.
Requirements
The following components are required to integrate a secondary server:
•One secondary server for each primary server installed and ready for the integration, as described in the applicable Cisco Unity installation guide and earlier in this integration guide.
•A license that enables failover.
Integration Description
The phone system communicates with both the primary and secondary servers through the LAN. Figure 2 shows the required connections.
Figure 2 Connections Between the SIP Proxy Server and the Cisco Unity Servers
The primary and secondary servers act in the following manner:
•When the primary server is operating normally, the secondary server is inactive.
•When the primary server becomes inactive, the secondary server becomes active.
•When the primary server becomes active again, the secondary server becomes inactive.
Setting Up the Secondary Server for Failover
Do the following procedure to integrate the secondary server.
To Set Up the Secondary Server for Failover
Step 1 Install a secondary server with the same configuration as the primary server. For installation instructions, refer to the applicable Cisco Unity installation guide.
Step 2 On the Windows Start menu of the secondary server, click Programs > Cisco Unity > Manage Integrations. The UTIM window appears.
Step 3 On the Integration menu of the UTIM window, click New. The Telephony Integration Setup Wizard appears.
Step 4 Enter the settings to match the integration settings on the primary server.
The Contact Line Name must be the same for both the primary and secondary servers.
Note We recommend not reassigning any unassigned subscribers and call handlers to the new integration, if you are asked by the wizard. Failover replication will automatically assign the correct integration.
Step 5 At the prompt to restart the Cisco Unity services, click Yes.
Note When restarting the Cisco Unity services, use the UTIM prompt instead of the Cisco Unity icon in the Windows taskbar. The taskbar icon does not restart all of the Cisco Unity services.
Step 6 After Cisco Unity restarts, on the Windows Start menu of the Cisco Unity server, click Programs > Cisco Unity > Manage Integrations. UTIM appears.
Step 7 In the left pane of the UTIM window, click the phone system integration that you created in Step 3.
Step 8 For Cisco Unity 4.0 and 4.1, continue to Step 9.
For Cisco Unity 4.2 and later, do the following substeps.
a. In the right pane, click Properties.
b. On the Integration tab, compare the setting of the Integration ID field for the secondary server to the setting of the Integration ID field for the primary server.
c. If the integration IDs of the phone system on the primary and secondary server are the same, continue to Step 9.
If the integration IDs of the phone system on the primary and secondary servers are different, on the secondary server, click Modify Integration ID.
d. When cautioned that subscribers associated with the current Integration ID setting will not be automatically associated with the new Integration ID setting, click OK.
e. In the Modify Integration ID dialog box, in the Enter New Integration ID field, enter the Integration ID setting for the phone system on the primary server and click OK.
f. Click Save.
g. At the prompt to restart the Cisco Unity services, click Yes.
h. In the left pane, click the phone system integration that you created in Step 3.
Step 9 In the right pane of the UTIM window, click the Ports tab.
Step 10 Enter the port settings to match the port settings on the primary server.
Caution In programming the SIP proxy server, do not send calls to voice messaging ports in Cisco Unity that cannot answer calls (voice messaging ports that are not set to Answer Calls). For example, if a voice messaging port is set only to Dialout MWI, do not send calls to it.
Step 11 Click Save.
Step 12 Exit UTIM.
Changing the Settings for an Existing Integration
After the integration is set up, if you want to change any of its settings (for example, to add or remove voice messaging ports for an integration), do the following procedure.
To Change the Settings for an Integration
Step 1 On the Cisco Unity server, on the Windows Start menu, click Programs > Cisco Unity > Manage Integrations. The UTIM window appears.
Step 2 In the left pane, double-click Unity Server. The existing integrations appear.
Step 3 Click the integration you want to modify.
Step 4 In the right pane, click the name of the cluster, phone system, or PIMG unit (depending on your integration) for the integration.
Step 5 In the right pane, click the applicable tab for the integration.
Step 6 Enter new settings in the fields that you want to change.
Caution If you are adding or removing voice messaging ports, make sure you change the settings for the individual ports so that there are an appropriate number of ports set to answer calls and an appropriate number of ports set to dial out.
Step 7 In the UTIM window, click Save.
Step 8 If prompted, restart the Cisco Unity services.
Deleting an Existing Integration
If you want to delete an existing integration (for example, you have replaced the phone system with which Cisco Unity originally integrated), do the following procedure.
To Delete an Existing Integration
Step 1 On the Cisco Unity server, on the Windows Start menu, click Programs > Cisco Unity > Manage Integrations. The UTIM window appears.
Step 2 In the left pane, double-click Unity Server. The existing integrations appear.
Step 3 Click the integration that you want to delete.
Step 4 On the Integration menu, click Delete.
Step 5 Follow the on-screen instructions to assign the subscribers of the deleted phone system integration to another phone system integration.
Step 6 At the prompt to restart the Cisco Unity services, click Yes. The Cisco Unity services restart.
Alternatively, you can restart the Cisco Unity services in UTIM on the Tools menu by clicking Restart Cisco Unity.
Step 7 If the integration you deleted used voice cards, remove the voice cards from the Cisco Unity server.
Appendix: Compatibility of Phone System Components
Testing has shown compatibility of the following phone system components with Cisco Unity in a SIP integration.
Table 9 Cisco SIP Proxy Server Compatibility with the Integration
Version
|
Comments
|
1.3
|
If Cisco Unity authenticates with the SIP proxy server, the authentication name entered in Cisco Unity must be the same as the contact line name in the SIP proxy server.
|
2.0
|
If Cisco Unity authenticates with the SIP proxy server, the authentication name entered in Cisco Unity must be the same as the contact line name in the SIP proxy server.
|
Table 10 Cisco 7960 IP Phone Compatibility with the Integration
Version
|
Comments
|
7960 P0S3-03-1-00
|
|
7960 P0S3-03-2-00
|
|
7960 P0S3-04-0-00
|
When the phone initiates a call, release transfer of the call is not available to Cisco Unity.
|
7960 P0S3-04-1-00
|
When the phone initiates a call, release transfer of the call is not available to Cisco Unity.
|
7960 P0S3-04-2-00
|
|
Table 11 Pingtel xpressa Compatibility with the Integration
Version
|
Comments
|
1.2.6
|
To get the call forwarding to busy greeting integration feature, forwarding must be programmed on the SIP proxy server rather than configured on the Pingtel xpressa phones.
|
2.0.1
2.0.2
|
Not compatible. Silence is inserted into the audio stream every few seconds.
To get the call forwarding to busy greeting integration feature, forwarding must be programmed on the SIP proxy server rather than configured on the Pingtel xpressa phones.
|
Table 12 Gateway IOS Compatibility with the Integration
Version
|
Comments
|
12.2(2)XB4
|
|
12.2(2)XB6
|
|
Other compatibility issues are:
•The Pingtel xpressa cannot connect to a backup SIP proxy server.
•To enable call forwarding when Cisco Unity is configured for failover, set the forwarding destinations in MySQL to be <contact line name>@proxy instead of <contact line name>@Unity.
•Caveat: CSCdx74350.
•Caveat: CSCdx66707.
•Caveat: CSCdx71968.
Appendix: Using Alternate Extensions and MWIs
Alternate Extensions
In addition to the "primary" extension that you specify for subscribers, you can assign subscribers up to nine alternate extensions. (The primary extension is the one that you assign to each subscriber when you create his or her subscriber account; it is listed on the Subscribers > Subscribers > Profile page.)
Reasons to Use Alternate Extensions
There are several reasons that you may want to specify alternate extensions for subscribers. For example, if you have more than one Cisco Unity server that accesses a single, corporate-wide directory, you may want to use alternate extensions to simplify addressing messages to subscribers at the different locations. With alternate extensions, the number that a subscriber uses when addressing a message to someone at another location can be the same number that the subscriber dials when calling. You may also want to use alternate extensions to:
•Handle multiple line appearances on subscriber phones.
•Offer easy message access on direct calls from a cell phone, home phone, or phone at an alternate work site (assuming that the phone number is passed along to Cisco Unity from these other phone systems). In addition, when such phones are used as alternate extensions, and are set to forward to Cisco Unity, callers can listen to the subscriber greeting, and leave messages for the subscriber just as they would when dialing the primary extension for the subscriber.
Tip To reduce the number of requests from subscribers who want alternate extensions set up for multiple cell phones, home phones, and other phones, give subscribers class of service (COS) rights to specify their own set of alternate extensions. (See the Subscribers > Class of Service > Profile page.) With proper COS rights, a subscriber can specify up to five alternate extensions in the Cisco Unity Assistant—in addition to the nine that you can specify on the Subscribers > Alternate Extensions page in the Cisco Unity Administrator.
•Enable URL-based extensions in Cisco Unity for an integration with a SIP phone system.
How Alternate Extensions Work
Before you set up alternate extensions, review the following list for information on how alternate extensions work:
•Alternate extensions cannot exceed 30 characters in length. By default, each administrator-defined alternate extension must be at least 3 characters in length, while subscriber-defined alternate extensions must be at least 10 characters.
You can use the Advanced Settings tool in Tools Depot to specify a minimum extension length for the extensions entered in the Cisco Unity Administrator and the Cisco Unity Assistant. Refer to the Advanced Settings Tool Help for details on using the settings. Respectively, the settings are Administration—Set the Minimum Length for Locations, and Administration—Set the Minimum Length for Subscriber-Defined Alternate Extensions.
•You can control whether subscribers can use the Cisco Unity Assistant to view the alternate extensions that you specify in the Cisco Unity Administrator. To do so, see the Subscribers > Class of Service > Profile page. The Subscriber-Defined Alternate Extension table displays the alternate extensions that the subscriber adds.
•Neither the Cisco Unity Administrator nor the Cisco Unity Assistant will accept an extension that is already assigned to another subscriber (either as a primary or alternate extension), or to a public distribution list, call handler, directory handler, or interview handler. Cisco Unity verifies that each alternate extension is unique—up to the dialing domain level, if applicable—before allowing either an administrator or a subscriber to create it.
•All alternate extensions use the same transfer settings as the primary extension.
•In many cases, Cisco Unity can activate a message waiting indicator (MWI) for an alternate extension. However, depending on the phones and phone systems involved, some additional phone system programming may be required to set this up.
Setting Up Alternate Extensions
Do the applicable procedure to add, modify, or delete alternate extensions:
•To Add Administrator-Defined Alternate Extensions
•To Modify or Delete Alternate Extension(s)
To Add Administrator-Defined Alternate Extensions
Step 1 In the Cisco Unity Administrator, go to any Subscribers > Alternate Extensions page.
Step 2 In the Administrator-Defined Alternate Extensions table, enter an extension in any row. When entering characters in the Alternate Extensions table, consider the following:
•You can enter an extension up to 30 characters in length. (SIP integrations can use up to 30 alphanumeric characters.)
•Each extension must be unique—up to the dialing domain level, if applicable.
•Enter digits 0 through 9. Do not use spaces, dashes, or parentheses.
•For SIP integrations, you can also enter a valid alias for a SIP URL. For example, if the URL is SIP:aabade@cisco.com, enter aabade. Do not use spaces.
•Rows are numbered as a convenience. You can enter alternate extensions in any order, and you can have blank rows.
Step 3 Repeat Step 2 as necessary.
Step 4 Click the Save icon. Alternate extensions are enabled for all rows in the table.
To Modify or Delete Alternate Extension(s)
Step 1 In the Cisco Unity Administrator, go to any Subscribers > Alternate Extensions page.
Step 2 Do any of the following:
•To modify an extension, change the extension in the Alternate Extensions table.
•To delete extensions, check the check boxes next to the alternate extensions that you want to delete.
•To remove all alternate extensions listed in the table, click Select All.
Step 3 Click the Save icon.
Step 4 Repeat Step 2 and Step 3 as necessary.
Note You can run the Cisco Unity Bulk Import wizard when you want to add alternate extensions for multiple subscribers at once. When you do, the Cisco Unity Bulk Import wizard appends the new alternate extensions to the existing table of alternate extensions, beginning with the first blank row.
Alternate MWIs
You can set up Cisco Unity to activate alternate MWIs when you want a new message for a subscriber to activate the MWIs at up to 10 extensions. For example, a message left at extension 1001 can activate the MWIs on extensions 1001 and 1002.
Cisco Unity uses MWIs to alert the subscriber to new voice messages. MWIs are not used to indicate new e-mail, fax, or return receipt messages.
Setting Up Alternate MWIs
Cisco Unity can activate alternate MWIs. Note that depending on the phones and phone systems, some additional phone system programming may be necessary. Refer to the installation guide for the phone system.
To enable alternate MWIs for extensions, do the following procedure for each subscriber who needs alternate MWIs.
To Set Up Alternate MWIs for Extensions
Step 1 In the Cisco Unity Administrator, go to the applicable Subscribers > Subscribers > Messages page.
Step 2 Confirm that the Use MWI for Message Notification check box is checked.
Step 3 Click the Add button located beneath the MWI Extensions table to add a row to the table. By default, the first row in the table contains an "X" to indicate the primary extension assigned to a subscriber. If you want one more extension and do not need to activate the MWI on the primary extension, you can also modify the first row.
Step 4 Enter the applicable extension in the Extension field of the table. MWIs are automatically enabled for all rows in the table. When entering characters in the MWI Extensions table, consider the following:
•Enter digits 0 through 9. Do not use spaces, dashes, or parentheses.
•Enter , (comma) to insert a one-second pause.
•Enter # and * to correspond to the # and * keys on the phone.
Step 5 Click the Save icon.
Step 6 Repeat Step 3 through Step 5 as necessary.
Note You can run the Cisco Unity Bulk Import wizard when you want to set up alternate MWIs for multiple subscribers at once.
To change or delete alternate MWIs for extensions, do the following procedure.
To Modify or Delete Alternate MWIs
Step 1 In the Cisco Unity Administrator, go to the applicable Subscribers > Subscribers > Messages page.
Step 2 Do either of the following:
•To modify an extension, change the extension in the MWI Extensions table.
•To delete extensions, check the check boxes next to the rows that you want to delete in the MWI Extensions table, and then click the Delete button.
Step 3 Click the Save icon.
Step 4 Repeat Step 2 and Step 3 as necessary.
Appendix: Documentation and Technical Assistance
Conventions
The Session Initiation Protocol (SIP) Integration Guide for Cisco Unity 4.0 uses the following conventions.
Table 13 Session Initiation Protocol (SIP) Integration Guide for Cisco Unity 4.0 Conventions
Convention
|
Description
|
boldfaced text
|
Boldfaced text is used for:
•Key and button names. (Example: Click OK.)
•Information that you enter. (Example: Enter Administrator in the User Name box.)
|
< >
(angle brackets)
|
Angle brackets are used around parameters for which you supply a value. (Example: In the Command Prompt window, enter ping <IP address>.)
|
-
(hyphen)
|
Hyphens separate keys that must be pressed simultaneously. (Example: Press Ctrl-Alt-Delete.)
|
>
(right angle
bracket)
|
A right angle bracket is used to separate selections that you make:
•On menus. (Example: On the Windows Start menu, click Settings > Control Panel > Phone and Modem Options.)
•In the navigation bar of the Cisco Unity Administrator. (Example: Go to the System > Configuration > Settings page.)
|
[x]
(square brackets)
|
Square brackets enclose an optional element (keyword or argument). (Example: [reg-e164])
|
[x | y]
(vertical line)
|
Square brackets enclosing keywords or arguments separated by a vertical line indicate an optional choice. (Example: [transport tcp | transport udp])
|
{x | y}
(braces)
|
Braces enclosing keywords or arguments separated by a vertical line indicate a required choice. (Example: {tcp | udp})
|
The Session Initiation Protocol (SIP) Integration Guide for Cisco Unity 4.0 also uses the following conventions:
Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the document.
Caution Means
reader be careful. In this situation, you might do something that could result in equipment damage or loss of data.
For descriptions and URLs of Cisco Unity documentation on Cisco.com, see the About Cisco Unity Documentation. The document is shipped with Cisco Unity and is available at http://www.cisco.com/univercd/cc/td/doc/product/voice/c_unity/about/aboutdoc.htm.
Obtaining Documentation
Cisco documentation and additional literature are available on Cisco.com. Cisco also provides several ways to obtain technical assistance 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 Documentation DVD
The Product Documentation DVD is a comprehensive library of technical product documentation on a portable medium. The DVD enables you to access multiple versions of installation, configuration, and command guides for Cisco hardware and software products. With the DVD, you have access to the same HTML documentation that is found on the Cisco website without being connected to the Internet. Certain products also have .PDF versions of the documentation available.
The Product Documentation DVD is available as a single unit or as a subscription. Registered Cisco.com users (Cisco direct customers) can order a Product Documentation DVD (product number DOC-DOCDVD= or DOC-DOCDVD=SUB) from Cisco Marketplace at this URL:
http://www.cisco.com/go/marketplace/
Ordering Documentation
Registered Cisco.com users may order Cisco documentation at the Product Documentation Store in the Cisco Marketplace at this URL:
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Nonregistered Cisco.com users can order technical documentation from 8:00 a.m. to 5:00 p.m. (0800 to 1700) PDT by calling 1 866 463-3487 in the United States and Canada, or elsewhere by calling 011 408 519-5055. You can also order documentation by e-mail at tech-doc-store-mkpl@external.cisco.com or by fax at 1 408 519-5001 in the United States and Canada, or elsewhere at 011 408 519-5001.
Documentation Feedback
You can rate and provide feedback about Cisco technical documents by completing the online feedback form that appears with the technical documents on Cisco.com.
You can submit comments about Cisco documentation by using the response card (if present) behind the front cover of your document or by writing to the following address:
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We appreciate your comments.
Cisco Product Security Overview
Cisco provides a free online Security Vulnerability Policy portal at this URL:
http://www.cisco.com/en/US/products/products_security_vulnerability_policy.html
From this site, you will find information about how to:
•Report security vulnerabilities in Cisco products.
•Obtain assistance with security incidents that involve Cisco products.
•Register to receive security information from Cisco.
A current list of security advisories, security notices, and security responses for Cisco products is available at this URL:
http://www.cisco.com/go/psirt
To see security advisories, security notices, and security responses as they are updated in real time, you can subscribe to the Product Security Incident Response Team Really Simple Syndication (PSIRT RSS) feed. Information about how to subscribe to the PSIRT RSS feed is found at this URL:
http://www.cisco.com/en/US/products/products_psirt_rss_feed.html
Reporting Security Problems in Cisco Products
Cisco is committed 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 have identified a vulnerability in a Cisco product, contact PSIRT:
•For Emergencies only — 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 vulnerability should be reported. All other conditions are considered nonemergencies.
•For Nonemergencies — psirt@cisco.com
In an emergency, you can also reach PSIRT by telephone:
•1 877 228-7302
•1 408 525-6532
Tip We encourage you to use Pretty Good Privacy (PGP) or a compatible product (for example, GnuPG) to encrypt any sensitive information that you send to Cisco. PSIRT can work with information that has been encrypted with PGP versions 2.x through 9.x.
Never use a revoked or an expired encryption key. The correct public key to use in your correspondence with PSIRT is the one linked in the Contact Summary section of the Security Vulnerability Policy page at this URL:
http://www.cisco.com/en/US/products/products_security_vulnerability_policy.html
The link on this page has the current PGP key ID in use.
If you do not have or use PGP, contact PSIRT at the aforementioned e-mail addresses or phone numbers before sending any sensitive material to find other means of encrypting the data.
Obtaining Technical Assistance
Cisco Technical Support provides 24-hour-a-day award-winning technical assistance. The Cisco Technical Support & Documentation website on Cisco.com features extensive online support resources. In addition, if you have a valid Cisco service contract, Cisco Technical Assistance Center (TAC) engineers provide telephone support. If you do not have a valid Cisco service contract, contact your reseller.
Cisco Technical Support & Documentation Website
The Cisco Technical Support & Documentation website provides online documents and tools for troubleshooting and resolving technical issues with Cisco products and technologies. The website is available 24 hours a day, at this URL:
http://www.cisco.com/techsupport
Access to all tools on the Cisco Technical Support & Documentation website requires a Cisco.com user ID and password. If you have a valid service contract but do not have a user ID or password, you can register at this URL:
http://tools.cisco.com/RPF/register/register.do
Note Use the Cisco Product Identification (CPI) tool to locate your product serial number before submitting a web or phone request for service. You can access the CPI tool from the Cisco Technical Support & Documentation website by clicking the Tools & Resources link 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 number label location highlighted. Locate the serial number label on your product and record the information before placing a service call.
Submitting a Service Request
Using the online TAC Service Request Tool is the fastest way to open S3 and S4 service requests. (S3 and S4 service requests are those in which your network is minimally impaired or for which you require product information.) After you describe your situation, the TAC Service Request Tool provides recommended solutions. If your issue is not resolved using the recommended resources, your service request is assigned to a Cisco engineer. The TAC Service Request Tool is located at this URL:
http://www.cisco.com/techsupport/servicerequest
For S1 or S2 service requests, or if you do not have Internet access, contact the Cisco TAC by telephone. (S1 or S2 service requests are those in which your production network is down or severely degraded.) Cisco engineers are assigned immediately to S1 and S2 service requests to help keep your business operations running smoothly.
To open a service request by telephone, use one of the following numbers:
Asia-Pacific: +61 2 8446 7411 (Australia: 1 800 805 227)
EMEA: +32 2 704 55 55
USA: 1 800 553-2447
For a complete list of Cisco TAC contacts, go to this URL:
http://www.cisco.com/techsupport/contacts
Definitions of Service Request Severity
To ensure that all service requests are reported in a standard format, Cisco has established severity definitions.
Severity 1 (S1)—An existing network is down, or there is a critical impact to your business operations. You and Cisco will commit all necessary resources around the clock to resolve the situation.
Severity 2 (S2)—Operation of an existing network is severely degraded, or significant aspects of your business operations are negatively affected by inadequate performance of Cisco products. You and Cisco will commit full-time resources during normal business hours to resolve the situation.
Severity 3 (S3)—Operational performance of the network is impaired, while most business operations remain functional. You and Cisco will commit resources during normal business hours to restore service to satisfactory levels.
Severity 4 (S4)—You require information or assistance with Cisco product capabilities, installation, or configuration. There is little or no effect on your business operations.
Obtaining Additional Publications and Information
Information about Cisco products, technologies, and network solutions is available from various online and printed sources.
•The Cisco Product Quick Reference Guide is a handy, compact reference tool that includes brief product overviews, key features, sample part numbers, and abbreviated technical specifications for many Cisco products that are sold through channel partners. It is updated twice a year and includes the latest Cisco offerings. To order and find out more about the Cisco Product Quick Reference Guide, go to this URL:
http://www.cisco.com/go/guide
•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 publications. For current Cisco Press titles and other information, go to Cisco Press at this URL:
http://www.ciscopress.com
•Packet magazine is the Cisco Systems technical user magazine for maximizing Internet and networking investments. Each quarter, Packet delivers coverage of the latest industry trends, technology breakthroughs, and Cisco products and solutions, as well as network deployment and troubleshooting tips, configuration examples, customer case studies, certification and training information, and links to scores of in-depth online resources. You can access Packet magazine at this URL:
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•iQ Magazine is the quarterly publication from Cisco Systems designed to help growing companies learn how they can use technology to increase revenue, streamline their business, and expand services. The publication identifies the challenges facing these companies and the technologies to help solve them, using real-world case studies and business strategies to help readers make sound technology investment decisions. You can access iQ Magazine at this URL:
http://www.cisco.com/go/iqmagazine
or view the digital edition at this URL:
http://ciscoiq.texterity.com/ciscoiq/sample/
•Internet Protocol Journal is a quarterly journal published by Cisco Systems for engineering professionals involved in designing, developing, and operating public and private internets and intranets. You can access the Internet Protocol Journal at this URL:
http://www.cisco.com/ipj
•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 professionals. Join a discussion at this URL:
http://www.cisco.com/discuss/networking
•World-class networking training is available from Cisco. You can view current offerings at this URL:
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Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.
© 2006 Cisco Systems, Inc. All rights reserved.
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