Table Of Contents
QSIG-Enabled Phone System with Cisco ISR Voice Gateway Integration Guide for Cisco Unity 5.0
Integration Tasks
Task List to Create the 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
Programming the QSIG-Enabled Phone System
Configuring the Cisco ISR Voice Gateway
Creating an Integration with the Phone System
Testing the Integration
Appendix: Documentation and Technical Assistance
Documentation Conventions
Obtaining Documentation and Submitting a Service Request
QSIG-Enabled Phone System with Cisco ISR Voice Gateway Integration Guide for Cisco Unity 5.0
Revised April 7, 2009
This document provides instructions for integrating a QSIG-enabled phone system with Cisco Unity through a Cisco ISR voice gateway.
Note 
Cisco Unity failover and AMIS networking are not available for the integration with a QSIG-enabled phone system.
Integration Tasks
Before doing the following tasks to integrate Cisco Unity with a QSIG-enabled phone system through a Cisco ISR voice gateway, 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 list describes the process for creating the integration.
Task List to Create the Integration
Use the following task list to set up a new integration with a QSIG-enabled phone system through a Cisco ISR voice gateway. 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 QSIG-enabled phone system and extensions. See the "Programming the QSIG-Enabled Phone System" section.
4. Configure the Cisco ISR voice gateway. See the Configuring the Cisco ISR Voice Gateway.
5. Create the integration. See the Creating an Integration with the Phone System.
6. Test the integration. See the Testing the Integration.
Note Cisco Unity failover is not available for the integration with a QSIG-enabled phone system.
Requirements
Revised April 7, 2009
The QSIG-enabled integration supports configurations of the following components:
Phone System
•A QSIG-enabled phone system.
•The phone system is ready for the integration.
Cisco ISR Voice Gateway
•Cisco IOS version 12.4(11)T or later.
•The QSIG-enabled phone system connected to the Cisco ISR voice gateway.
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.
Centralized Voice Messaging
Cisco Unity supports centralized voice messaging by supporting various inter-phone system networking protocols including, for example, proprietary protocols such as Avaya DCS, Nortel MCDN, or Siemens CorNet, and standards-based protocols such as QSIG or DPNSS. For details, see the "Centralized Voice Messaging" section in the "Integrating Cisco Unity with the Phone System" chapter of the Cisco Unity Design Guide Release 5.x at http://www.cisco.com/en/US/docs/voice_ip_comm/unity/5x/design/guide/5xcudgx.html.
Integration Description
This integration uses a Cisco ISR voice gateway and a LAN or WAN to connect Cisco Unity and a QSIG-enabled phone system. The Cisco ISR voice gateway converts the QSIG communications to SIP. Figure 1 shows the required connections.
Figure 1 Connections Between the Phone System and Cisco Unity
Call Information
The QSIG-enabled phone system integration sends the following information with forwarded calls:
•The extension of the called party
•The extension of the calling party (for internal calls) or the phone number of the calling party (if it is an external call and the system uses caller ID)
•The reason for the forward (the extension is busy, does not answer, or is set to forward all calls)
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 QSIG-enabled phone system 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)
Note Some phone systems may not support all integration features (such as MWIs). For details, see the application note at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_installation_and_configuration_guides_list.html.
Integrations with Multiple Phone Systems
Cisco Unity can be integrated with multiple 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 for Cisco Unity 5.0 at http://cisco.com/en/US/products/sw/voicesw/ps2237/products_installation_and_configuration_guides_list.html.
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. The following considerations will affect the programming for the phone system (for example, setting up the hunt group or call forwarding for the voice messaging ports):
•The number of voice messaging ports installed.
•The number of voice messaging ports that will answer calls.
•The number of voice messaging ports that will only dial out, for example, to send message notification, and to make telephone record and playback (TRAP) connections.
Note The Cisco ISR voice gateway will perform transfers by hairpinning two independent calls across two b-channels on the QSIG trunk. Hairpinned calls will use more QSIG channels in comparison to the number of Cisco Unity voice messaging ports that are available to answer calls.
Release (blind) transfers that are forwarded back to Cisco Unity will use three b-channels for the remainder of the call. However, supervised transfers pull back the consulting call when the target is unavailable so that only one b-channel is used for the remainder of the call.
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
|
Enabled
|
Check this check box.
|
Answer Calls
|
Check this check box to designate the port for answering calls.
Caution All voice messaging ports connecting to the Cisco ISR voice gateway 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.
|
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. Assign TRAP Connection to the least busy ports.
|
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 ports that will be set to dial out only.
•The number of calls made for message notification.
•The number of TRAP connections needed when call traffic is at its peak. (TRAP connections are used by Cisco Unity web applications 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. Typically, the voice messaging ports that answer calls are the busiest.
You can set voice messaging ports to both answer calls and to dial out (for example, to send message notifications). However, when the voice messaging ports perform more than one function and are very active (for example, answering many calls), the other functions may be delayed until the voice messaging port is free (for example, message notifications cannot be sent until there are fewer calls to answer). For best performance, dedicate certain voice messaging ports for only answering incoming calls, and dedicate other ports for only dialing out. Separating these port functions eliminates the possibility of a collision, in which an incoming call arrives on a port at the same time that Cisco Unity takes the port off-hook to dial out.
Note The Cisco ISR voice gateway will perform transfers by hairpinning two independent calls across two b-channels on the QSIG trunk. Hairpinned calls will use more QSIG channels in comparison to the number of Cisco Unity voice messaging ports that are available to answer calls. If your system uses Cisco Unity auto-attendant transfers, you must provision a larger number of QSIG b-channels than the number of Cisco Unity voice messaging ports that will answer calls.
The Number of Voice Messaging Ports That Will Dial Out
Ports that will only dial out can do one or more of the following:
•Notify subscribers by phone, pager, or e-mail of messages that have arrived.
•Make a TRAP connection so that subscribers can use the phone as a recording and playback device in Cisco Unity web applications.
Preparing for Programming the Phone System
Record your decisions about the voice messaging ports to guide you in programming the phone system.
Programming the QSIG-Enabled Phone System
For information on provisioning the phone system for QSIG interoperability, see the phone system documentation, and see the application note that is available at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_installation_and_configuration_guides_list.html.
Note that you must program each extension to forward calls to the pilot number assigned to the voice messaging ports, based on one of the call transfer types shown in Table 2.
Table 2 Call Transfer Types
Transfer Types
|
Usage
|
Release transfer
(blind transfer)
|
Program the phone to forward calls to the pilot number when:
•The extension is busy
•The call is not answered
|
Supervised transfer
|
Program the phone to forward calls to the pilot number only when the call is not answered (on the phone system, the number of rings before forwarding must be more than the number of rings to supervise the call). Confirm that call forwarding is disabled when the extension is busy.
|
Configuring the Cisco ISR Voice Gateway
For information on configuring the Cisco ISR voice gateway, see the application note that is available at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_installation_and_configuration_guides_list.html.
Note If the Cisco Unity SIP Port setting will not be 5060 (for example, it is set to 5061), you must change the SIP port that the Cisco ISR voice gateway uses. You can use a command similar to the following:
dial-peer voice 1 voip
session protocol sipv2
session target ipv4:10.00.00.00:5061
Creating an Integration with the Phone System
After ensuring that the QSIG-enabled 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 the Cisco Unity Telephony Integration Manager (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 SIP (including CUCM/CCM) and click Next.
Step 5 On the Name This SIP Integration and Clusters page, enter the following settings, then click Next.
Table 3 Settings for the Name This SIP Integration and Cluster Page
Field
|
Setting
|
Integration Name
|
Enter the name you will use to identify this QSIG-enabled phone system integration. Accept the default name or enter another name.
|
Cluster Name
|
Enter the name you will use to identify the Cisco ISR voice gateway. Accept the default name or enter another name.
|
Step 6 On the Enter Primary and Secondary SIP Server page, enter the following settings, then click Next.
Table 4 Settings for the Enter Primary and Secondary SIP Server Page
Field
|
Setting
|
Primary:
IP Address/Name
|
Enter the first virtual IP address (or name alias) of the primary Cisco ISR voice gateway that you are connecting to Cisco Unity.
|
Primary:
Port
|
Enter the IP port of the primary Cisco ISR voice gateway that you are connecting to Cisco Unity.
|
Secondary:
IP Address/Name
|
(Optional) Enter the second virtual IP address (or name alias) of the secondary Cisco ISR voice gateway that you are connecting to Cisco Unity.
|
Secondary:
Port
|
(Optional) Enter the IP port of the secondary Cisco ISR voice gateway that you are connecting to Cisco Unity.
|
You can click Ping Servers to confirm that the IP address is correct.
Step 7 On the Set Number of Cisco Unity Voice Messaging Ports page, enter the number of voice messaging ports on Cisco Unity that you want to connect to the Cisco ISR voice gateway, then click Next.
This number must not be more than the number of ports set up on the Cisco ISR voice gateway.
Step 8 On the Configure Cisco Unity SIP Settings page, enter the following settings, then click Next.
Table 5 Settings for the Configure Cisco Unity SIP Settings Page
Field
|
Setting
|
Contact Line Name
|
Enter the voice messaging pilot number that matches the Cisco ISR voice gateway dial plan configuration.
|
Cisco Unity SIP Port
|
Enter the IP port on Cisco Unity that callers and the Cisco ISR voice gateway use to connect to voice mail. We recommend using the default setting.
Note If the Cisco Unity SIP Port setting will not be 5060 (for example, it is set to 5061), you must change the SIP port that the Cisco ISR voice gateway uses.
|
Preferred Codec
|
Click the codec that Cisco Unity will first attempt to use on outgoing calls.
|
Preferred Transport Protocol
|
Click UDP.
|
Step 9 On the Enter SIP Server Authentication page, enter the following settings, then click Next.
Table 6 Settings for the Enter SIP Server Authentication Page
Field
|
Setting
|
Authenticate with the SIP Server
|
Uncheck the check box.
|
Name
|
Leave this field blank.
|
Password
|
Leave this field blank.
|
Step 10 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 11 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 12.
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 7 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.
|
Note You can reassign subscribers later. For details, refer to UTIM Help.
Step 12 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 13.
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 8 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.
|
Note You can reassign call handlers later. For details, refer to UTIM Help.
Step 13 On the Completing page, verify the settings you entered, then click Finish.
Step 14 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.
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 created.
Step 3 In the left pane, click the name of the QSIG-enabled phone system integration.
Step 4 In the right pane, click the Ports tab.
Step 5 Enter the settings shown in Table 9 for the voice messaging ports.
Table 9 Settings for the Voice Messaging Ports
Field
|
Considerations
|
Enabled
|
Check this check box.
|
Answer Calls
|
Check this check box to designate the port for answering calls. These calls can be incoming calls from unidentified callers or from subscribers.
Caution All voice messaging ports connecting to the Cisco ISR voice gateway 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.
|
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.
Testing the Integration
To test whether Cisco Unity and the phone system are integrated correctly, do the following procedures in the order listed.
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 Microsoft Exchange, do the following:
a. In the Cisco Unity Administrator, go to the Subscribers > Subscribers > Profile page.
b. Click the Add icon.
c. In the New Subscriber field, click Exchange.
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. In the New Subscriber field, 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 In the Transfer Rule Applies To field, click Standard.
Step 9 Under Transfer Incoming Calls to a Phone, click Yes, Ring Extension For, and confirm that the extension number is for Phone 1.
Step 10 Under Transfer Type, click Release to Switch.
Step 11 Click the Save icon.
Step 12 In the navigation bar, click Messages to go to the Subscribers > Subscribers > Messages page for the test subscriber.
Step 13 Under Message Waiting Indicators (MWIs), check Use MWI for Message Notification.
Step 14 In the Extension field, enter x.
Step 15 Click the Save icon.
Step 16 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 message, 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.
To Set Up Supervised Transfer on Cisco Unity
Step 1 In the Cisco Unity Administrator, go to the Subscribers > Subscribers > Call Transfer 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.
For more information on transfer settings, see the "Subscriber Template Call Transfer Settings" section in Cisco Unity Administrator Help.
Step 2 Under Transfer Incoming Calls to a Phone, click Yes, Ring Extension For, and confirm that the extension number is for Phone 1.
Step 3 Under Transfer Type, click Supervise Transfer.
Step 4 Set the Rings to Wait For field to 3.
Step 5 Click the Save icon.
To Test Supervised 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 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
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.
Step 4 When prompted to confirm deleting the subscriber, click OK.
Appendix: Documentation and Technical Assistance
Documentation Conventions
The QSIG-Enabled Phone System with Cisco ISR Voice Gateway Integration Guide for Cisco Unity 5.0 uses the following conventions.
Table 10 QSIG-Enabled Phone System with Cisco ISR Voice Gateway Integration Guide for Cisco Unity 5.0 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 QSIG-Enabled Phone System with Cisco ISR Voice Gateway Integration Guide for Cisco Unity 5.0 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 Documentation Guide for Cisco Unity. The document is shipped with Cisco Unity and is available at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_documentation_roadmaps_list.html.
Obtaining Documentation and Submitting a Service Request
For information on obtaining documentation, submitting a service request, and gathering additional information, see the monthly What's New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at:
http://www.cisco.com/en/US/docs/general/whatsnew/whatsnew.html
Subscribe to the What's New in Cisco Product Documentation as a Really Simple Syndication (RSS) feed and set content to be delivered directly to your desktop using a reader application. The RSS feeds are a free service and Cisco currently supports RSS version 2.0.
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