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Cisco Unified CallManager 5.1 SIP Trunk Integration Guide for Cisco Unity 4.2

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Table Of Contents

Cisco Unified CallManager 5.1 SIP Trunk Integration Guide for Cisco Unity 4.2

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 Cisco Unified CallManager Phone System

Creating a New Integration with the Cisco Unified CallManager Phone System

Disabling Transcoding into the G.729a Audio Format

Testing the Integration


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, Obtaining Support, and Security Guidelines


Cisco Unified CallManager 5.1 SIP Trunk Integration Guide for Cisco Unity 4.2


Revised May 14, 2007

This document provides instructions for integrating the Cisco Unified CallManager phone system with Cisco Unity through a SIP trunk.

Cisco Unity supports a SIP trunk integration when the Cisco Unified CallManager phone system has only SIP phones (best practice) or has both SCCP and SIP phones.


Note Cisco Unity failover is not available for the Cisco Unified CallManager SIP trunk integration.

If you are configuring MWI relay across trunks in a distributed phone system, you must refer to the Cisco Unified CallManager documentation for requirements and instructions. Configuring MWI relay across trunks does not involve Cisco Unity settings.


Integration Tasks

Before doing the following tasks to integrate Cisco Unity with the Cisco Unified CallManager 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 Cisco Unified CallManager 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 Cisco Unified CallManager. See the "Programming the Cisco Unified CallManager Phone System" section.

4. Create the integration. See the "Creating a New Integration with the Cisco Unified CallManager Phone System" section.

5. If want to disable transcoding into the G.729a audio format, remove the G.729a codec from the Cisco Unity server. See the "Disabling Transcoding into the G.729a Audio Format" section.

6. Test the integration. See the "Testing the Integration" section.

Requirements

The Cisco Unified CallManager integration supports configurations of the following components:

Phone System

A Cisco IP telephony applications server consisting of Cisco Unified CallManager 5.1(x), running on a Cisco Media Convergence Server (MCS) or customer-provided server meeting approved Cisco configuration standards.

For the Cisco Unified CallManager extensions, one of the following configurations:

(Best practice) Only SIP phones that support DTMF relay as described in RFC-2833.

Both SCCP and SIP phones.

Note that older SCCP phone models may require a Media Termination Point (MTP) to function correctly.

A LAN connection in each location where you will plug a SIP phone into the network.

For multiple Cisco Unified CallManager clusters, the capability for users to dial an extension on another Cisco Unified CallManager cluster without having to dial a trunk access code or prefix.

Cisco Unity Server

The applicable version of Cisco Unity. For details on compatible versions of Cisco Unity and Cisco Unified CallManager, refer to the SIP Trunk Compatibility Matrix: Cisco Unity and Cisco Unified CallManager at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_device_support_table09186a0080624ba2.html.

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 appropriate number of voice messaging ports.

Integration Description

The Cisco Unified CallManager integration uses the LAN to connect Cisco Unity and the phone system. The gateway provides connections to the PSTN.

Call Information

The phone system 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 Cisco Unified CallManager integration with Cisco Unity provides the following 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)

The functionality of this integration may be affected by the issues described below.

Use of Cisco Unified Survivable Remote Site Telephony (SRST) Router

When a Cisco Unified Survivable Remote Site Telephony (SRST) router is part of the network and the Cisco Unified SRST router takes over call processing functions from Cisco Unified CallManager (for example, because the WAN link is down), phones at a branch office can continue to function. In this situation, however, the integration features have the following limitations:

Call forward to busy greeting—When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call is forwarded from a branch office to Cisco Unity, the busy greeting cannot play.

Call forward to internal greeting—When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call is forwarded from a branch office to Cisco Unity, the internal greeting cannot play. Because the PSTN provides the calling number of the FXO line, the caller is not identified as a subscriber.

Call transfers—Because an access code is needed to reach the PSTN, call transfers from Cisco Unity to a branch office will fail.

Identified subscriber messaging—When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a subscriber at a branch office leaves a message or forwards a call, the subscriber is not identified. The caller appears as an unidentified caller.

Message waiting indication—MWIs are not updated on branch office phones, so MWIs will not correctly reflect when new messages arrive or when all messages have been listened to. We recommend resynchronizing MWIs after the WAN link is reestablished.

Message notification—Because an access code is needed to reach the PSTN, message notifications from Cisco Unity to a branch office will fail.

Routing rules—When the Cisco Unified SRST router uses FXO/FXS connections to the PSTN and a call arrives from a branch office to Cisco Unity (either a direct or forwarded call), routing rules will fail.

When the Cisco Unified SRST router uses PRI/BRI connections, the caller ID for calls from a branch office to Cisco Unity may be the full number (exchange plus extension) provided by the PSTN and therefore may not match the extension of the Cisco Unity subscriber. If this is the case, you can let Cisco Unity recognize the caller ID by using alternate extensions (for instructions, see the "Appendix: Using Alternate Extensions and MWIs" section) or by using extension remapping (for instructions, refer to the "Remapping Extension Numbers" section of the "System Settings" chapter in the applicable Cisco Unity System Administration Guide, available at http://www.cisco.com/en/US/products/sw/voicesw/ps2237/prod_maintenance_guides_list.html.

Redirected Dialed Number Information Service (RDNIS) needs to be supported when using SRST.

For information on setting up Cisco Unified SRST routers, refer to the "Integrating Voice Mail with Cisco Unified SRST" section of the Cisco Unified SRST System Administrator Guide at http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122z/122zj15/index.htm.

Impact of Non-Delivery of RDNIS on Voice Mail Calls Routed via AAR

RDNIS needs to be supported when using Automated Alternate Routing (AAR).

AAR can route calls over the PSTN when the WAN is oversubscribed. However, when calls are rerouted over the PSTN, RDNIS can be affected. Incorrect RDNIS information can affect voice mail calls that are rerouted over the PSTN by AAR when Cisco Unity is remote from its messaging clients. If the RDNIS information is not correct, the call will not reach the voice mail box of the dialed user but will instead receive the automated attendant prompt, and the caller might be asked to reenter the extension number of the party they wish to reach. This behavior is primarily an issue when the telephone carrier is unable to ensure RDNIS across the network. There are numerous reasons why the carrier might not be able to ensure that RDNIS is properly sent. Check with your carrier to determine whether it provides guaranteed RDNIS delivery end-to-end for your circuits. The alternative to using AAR for oversubscribed WANs is simply to let callers hear reorder tone in an oversubscribed condition.

Impact of Multiple Cisco Unified CallManager Clusters on MWIs (Cisco Unity 4.2 Only)

When configured for multiple SIP clusters, Cisco Unity 4.2 sends MWIs to only one cluster. As a result, all Cisco Unified CallManager clusters that integrate through a SIP trunk must be part of a single phone system integration in UTIM.

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. 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, to make AMIS deliveries, and to make telephone record and playback (TRAP) connections.

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.


Caution All voice messaging ports connecting to the Cisco Unified CallManager 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.

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 ports that will be set to dial out only.

The number of calls made for message notification.

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 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.

The Number of Voice Messaging Ports That Will Only Dial Out, and Not Answer Calls

Ports that will only dial out and will not answer calls can do one or more of the following:

Notify subscribers by phone, pager, or e-mail of messages that have arrived.

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.

Typically, these voice messaging ports are the least busy ports.


Caution In programming the phone system, 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 Message Notification, do not send calls to it.

Preparing for Programming the Phone System

Record your decisions about the voice messaging ports to guide you in programming the phone system.

Programming the Cisco Unified CallManager Phone System

Do the following procedures in the order given.


Note There must be a calling search space that is used by all subscriber phones (directory numbers). Otherwise, the integration will not function correctly. For instructions on setting up a calling search space and assigning subscriber phones to it, refer to the Cisco Unified CallManager Help.


To Create the SIP Trunk Security Profile


Step 1 In Cisco Unified CallManager Administration, on the System menu, click Security Profile > SIP Trunk Security Profile.

Step 2 On the Find and List SIP Trunk Security Profiles page, click Add New.

Step 3 On the SIP Trunk Security Profile Configuration page, under SIP Trunk Security Profile Information, enter the following settings.

Table 2 Settings for the SIP Trunk Security Profile Configuration Page 

Field
Setting

Name

Enter Cisco Unity SIP Trunk Security Profile or another name.

Description

Enter SIP trunk security profile for Cisco Unity or another description.

Device Security Mode

Accept the default of Non-secure.

Accept Out-of-Dialog REFER

Check this check box.

Accept Unsolicited Notification

Check this check box.

Accept Header Replacement

Check this check box.


Step 4 Click Save.


To Create the SIP Profile


Step 1 On the Device menu, click Device Settings > SIP Profile.

Step 2 On the Find and List SIP Profiles page, click Add New.

Step 3 On the SIP Profile Configuration page, enter the following settings.

Table 3 Settings for the SIP Profile Configuration Page 

Field
Setting

Name

Enter Cisco Unity SIP Profile or another name.

Description

Enter SIP profile for Cisco Unity or another description.


Step 4 Click Save.


To Create the SIP Trunk


Step 1 On the Device menu, click Trunk.

Step 2 On the Find and List Trunks page, click Add New.

Step 3 On the Trunk Configuration page, in the Trunk Type field, click SIP Trunk.

Step 4 In the Device Protocol field, click SIP and click Next.

Step 5 Under Device Information, enter the following settings.

Table 4 Settings for Device Information on the Trunk Configuration Page 

Field
Setting

Device Name

Enter Cisco_Unity_SIP_Trunk or another name.

Description

Enter SIP trunk for Cisco Unity or another description.


Step 6 If subscriber phones are contained in a calling search space, under Inbound Calls, enter the following settings. Otherwise, continue to Step 7.

Table 5 Settings for Inbound Calls on the Trunk Configuration Page 

Field
Setting

Calling Search Space

Click the name of the calling search space that contains the subscriber phones.

Redirecting Diversion Header Delivery - Inbound

Check this check box.


Step 7 Under Outbound Calls, check the Redirecting Diversion Header Delivery - Outbound check box.

Step 8 Under SIP Information, enter the following settings.

Table 6 Settings for SIP Information on the Trunk Configuration Page 

Field
Setting

Destination Address

Enter the IP address of the Cisco Unity SIP port to which Cisco Unified CallManager will connect.

Destination Port

We recommend that you accept the default of 5060.

SIP Trunk Security Profile

Click the name of the SIP trunk security profile that you created in the "To Create the SIP Trunk Security Profile" procedure. For example, click "Cisco Unity SIP Trunk Security Profile."

Rerouting Calling Search Space

Click the name of the calling search space that is used by subscriber phones.

Out-of-Dialog Refer Calling Search Space

Click the name of the calling search space that is used by subscriber phones.

SIP Profile

Click the name of the SIP profile that you created in the "To Create the SIP Profile" procedure. For example, click "Cisco Unity SIP Profile."


Step 9 Adjust any other settings that are needed for your site.

Step 10 Click Save.


To Create a Route Pattern


Step 1 On the Call Routing menu, click Route/Hunt > Route Pattern.

Step 2 On the File and List Route Patterns page, click Add New.

Step 3 On the Route Pattern Configuration page, enter the following settings.

Table 7 Settings for the Route Pattern Configuration Page 

Field
Setting

Route Pattern

Enter the voice mail pilot number for Cisco Unity.

Gateway/Route List

Click the name of the SIP trunk that you created in the "To Create the SIP Trunk" procedure. For example, click "Cisco_Unity_SIP_Trunk."


Step 4 Click Save.


To Create the Voice Mail Pilot


Step 1 On the Voice Mail menu, click Voice Mail Pilot.

Step 2 On the Find and List Voice Mail Pilots page, click Add New.

Step 3 On the Voice Mail Pilot Configuration page, enter the following voice mail pilot number settings.

Table 8 Settings for the Voice Mail Pilot Configuration Page 

Field
Setting

Voice Mail Pilot Number

Enter the voice mail pilot number that users will dial to listen to their voice messages. This number must match the route pattern that you entered in the "To Create a Route Pattern" procedure.

Calling Search Space

Click the calling search space that includes partitions containing the user phones and the partition that you set up for the voice mail pilot number.

Description

Enter Cisco Unity Pilot or another description.

Make This the Default Voice Mail Pilot for the System

Check this check box. When this check box is checked, this voice mail pilot number replaces the current default pilot number.


Step 4 Click Save.


To Create the Voice Mail Profile


Step 1 On the Voice Mail menu, click Voice Mail Profile.

Step 2 On the Find and List Voice Mail Profiles page, click Add New.

Step 3 On the Voice Mail Profile Configuration page, enter the following voice mail profile settings.

Table 9 Settings for the Voice Mail Profile Configuration Page 

Field
Setting

Voice Mail Profile Name

Enter Cisco Unity Profile or another name to identify the voice mail profile.

Description

Enter Profile for Cisco Unity or another description.

Voice Mail Pilot

Click the voice mail pilot number that you defined in the "To Create the Voice Mail Pilot" procedure.

Voice Mail Box Mask

When multitenant services are not enabled on Cisco Unified CallManager, leave this field blank.

When multitenant services are enabled, each tenant uses its own voice mail profile and must create a mask to identify the extensions (directory numbers) in each partition that is shared with other tenants. For example, one tenant can use a mask 972813XXXX, while another tenant can use the mask 214333XXXX. It is also necessary to set up translation patterns for MWIs.

Make This the Default Voice Mail Profile for the System

Check this check box to make this voice mail profile the default.

When this check box is checked, this voice mail profile replaces the current default voice mail profile.


Step 4 Click Save.


Do the following two procedures only if you want to set up SIP Digest authentication.

If you do not want to set up SIP digest authentication, continue to the "Creating a New Integration with the Cisco Unified CallManager Phone System" procedure.

(Optional) To Set Up SIP Digest Authentication


Step 1 On the System menu, click Security Profile > SIP Trunk Security Profile.

Step 2 On the Find and List SIP Trunk Security Profiles page, click the SIP trunk security profile that you created in the "To Create the SIP Trunk Security Profile" procedure.

Step 3 On the SIP Trunk Security Profile Configuration page, check the Enable Digest Authentication check box.

Step 4 Click Save.


(Optional) To Create the Application User


Step 1 On the User Management menu, click Application User.

Step 2 On the Find and List Application Users page, click Add New.

Step 3 On the Application User Configuration page, enter the following settings.

Table 10 Settings for the Application User Configuration Page 

Field
Setting

User ID

Enter the application user identification name. Cisco Unified CallManager does not permit modifying the user ID after it is created. You may use the following special characters: =, +, <, >, #, ;, \, , "", and blank spaces.

Password

Enter the same password that you use for the digest credentials.

Confirm Password

Enter the password again.

Digest Credentials

Enter the name of the digest credentials.

Presence Group

Used with the Presence feature, the application user (for example, IPMASysUser) serves as the watcher because it requests status about the presence entity.

If you want the application user to receive the status of the presence entity, make sure that the Application User Presence group is allowed to view the status of the Presence group that is applied to the directory number, as indicated in the Presence Group Configuration window.

Accept Presence Subscription

Leave this check box unchecked.

Accept Out-of-Dialog REFER

Check this check box.

Accept Unsolicited Notification

Check this check box.

Accept Header Replacement

Leave this check box unchecked.

Available Devices

This list box displays the devices that are available for association with this application user.

To associate a device with this application user, select the device and click the Down arrow below this list box.

If the device that you want to associate with this application user does not appear in this pane, click one of these buttons to search for other devices:

Find More Phones—Click this button to find more phones to associate with this application user. The Find and List Phones window appears to enable a phone search.

Find More Route Points—Click this button to find more phones to associate with this application user. The Find and List CTI Route Points window displays to enable a CTI route point search.

Associated CAPF Profiles

In the Associated CAPF Profile pane, the Instance ID for the Application User CAPF Profile displays if you configured an Application User CAPF Profile for the user. To edit the profile, click the Instance ID; then, click Edit Profile. The Application User CAPF Profile Configuration window appears.

Groups

This list box appears after an application user has been added. The list box displays the groups to which the application user belongs.

Roles

This list box appears after an application user has been added. The list box displays the roles that are assigned to the application user.


Step 4 Click Save.


Creating a New Integration with the Cisco Unified CallManager Phone System

After ensuring that the Cisco Unified CallManager 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 11 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 12 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 13 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; in most circumstances, we recommend using the default setting; however, for multiple phone system integrations, do not use 5061 as the setting for this integration>


Caution Do not use 5061 as the Cisco Unity SIP port for this integration. Otherwise, MWIs and dialouts may not function correctly.

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 14 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 15 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 16 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.


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 cluster.

Step 4 In the right pane, click the Ports tab.

Step 5 Enter the settings shown in Table 1 for the voice messaging ports.

For the voice messaging ports assigned to a given Cisco Unified CallManager cluster, to get the best performance use the first voice messaging ports for incoming calls and the last ports to dial out. This helps minimize 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. Set the ports assigned to each Cisco Unified CallManager cluster in this manner.

Table 17 Settings for the Voice Messaging Ports 

Field
Considerations

Enabled

Check this check box.

Answer Calls

Check this check box.


Caution All voice messaging ports connecting to the Cisco Unified CallManager 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.

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 Repeat Step 3 and Step 6 for the remaining clusters, if any.

Step 8 If Cisco Unity integrates with only one cluster of Cisco Unified CallManager, exit UTIM, skip the remaining procedure in this section, and continue to the "Testing the Integration" section.

If Cisco Unity integrates with multiple clusters of Cisco Unified CallManager, continue to the next procedure.


To Create an Integration with a Second Cluster of Cisco Unified CallManager

If Cisco Unity integrates with only one cluster of Cisco Unified CallManager, skip this procedure.


Step 1 In the left pane of the UTIM window, click the Cisco Unified CallManager integration.

Step 2 On the Cluster menu, click New. The Add Server Dialog box appears.

Step 3 Enter the following settings.

Table 18 Settings for the Add Server Dialog Box 

Field
Setting

IP Address or Host Name

<the IP address (or DNS name) of a subscriber Cisco Unified CallManager server>

Port

<the IP port of the Cisco Unified CallManager server that you are connecting to Cisco Unity; we recommend using the default setting>


Step 4 Click OK.

Step 5 When prompted to enter the remaining settings for the cluster, click OK.

Step 6 Click the Servers tab, and, in the Display Name field, enter Cisco Unified CallManager Cluster 02 or another name that you will use to identify this Cisco Unified CallManager cluster.

Step 7 If there are no additional Cisco Unified CallManager servers in this cluster, continue to Step 11.

If there are additional Cisco Unified CallManager servers in this cluster, click Add. The Servers dialog box appears.

Step 8 Enter the following settings.

Table 19 Settings for the Add Server Dialog Box 

Field
Setting

IP Address or Host Name

<the IP address (or DNS name) of a Cisco Unified CallManager server>

Port

<the IP port of the Cisco Unified CallManager server that you are connecting to Cisco Unity; we recommend using the default setting>


Step 9 Click OK.

Step 10 Repeat Step 7 through Step 9 for all remaining Cisco Unified CallManager servers in the cluster.

Step 11 Click the SIP Info tab.

Table 20 Settings for the SIP Info Page 

Field
Setting

Contact Line Name

Enter the voice messaging line name that subscribers will use to contact Cisco Unity and that Cisco Unity uses to register with the Cisco Unified CallManager server.

Cisco Unity SIP Port

Verify the IP port on Cisco Unity that callers and Cisco Unified CallManager use to connect to voice mail. In most circumstances, we recommend using the default setting. However, for multiple phone system integrations, do not use 5061 as the setting for this integration.


Caution Do not use 5061 as the Cisco Unity SIP port for this integration. Otherwise, MWIs and dialouts may not function correctly.

Preferred Codec

Verify the codec that Cisco Unity will first attempt to use on outgoing calls.

Preferred Transport Protocol

Click UDP.


Step 12 Enter other settings on the tab as needed.

Step 13 Click the Ports tab, and click Add Port.

Step 14 In the Add Port dialog box, enter the number of voice messaging ports on Cisco Unity that you want to connect to the Cisco Unified CallManager cluster, and click OK.

This number cannot be more than the number of ports set up on the Cisco Unified CallManager cluster. This number cannot bring the total number of port installed on the Cisco Unity server to more than the number of ports enabled by the Cisco Unity license.

Step 15 Enter the settings shown in Table 21 for the voice messaging ports.

Table 21 Settings for the Voice Messaging Ports 

Field
Considerations

Enabled

Check this check box.

Answer Calls

Check this check box.


Caution All voice messaging ports connecting to the Cisco Unified CallManager 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.

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 16 Click the RTP tab, and confirm that the Automatically Assign option is selected.

Step 17 In the UTIM window, click Save.

Step 18 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 19 Exit UTIM.


Disabling Transcoding into the G.729a Audio Format

If you want to disable transcoding into the G.729a audio format, do the following procedure. Otherwise, continue to the "Testing the Integration" section.


Caution Disabling transcoding into the G.729a audio format will block the audio stream for phones that use this audio format when connected to Cisco Unity. For the phones that use the G.729a audio format to receive the audio stream from Cisco Unity, you must set up a Cisco Unified CallManager transcoder to transcode the audio stream into the G.729a audio format.

When Cisco Unity has multiple integrations, disabling transcoding into the G.729a audio format will block G.729 audio streams to the Cisco Unity server for other integrations that use the G.729a audio format (for example, Cisco Unified CallManager SCCP integrations or integrations through PIMG units).

To Disable Transcoding into the G.729a Audio Format


Step 1 On the Windows Start menu, click Settings > Control Panel > Sounds and Multimedia.

Step 2 In the Sounds and Multimedia dialog box, click the Hardware tab.

Step 3 Under Devices, click Audio Codecs and click Properties.

Step 4 In the Audio Codecs Properties dialog box, click the Properties tab.

Step 5 Under Audio Compression Codecs, click Sipro Labs G.729A and click Remove.

Step 6 When prompted to confirm removing the codec, click Yes.

Step 7 If prompted to restart the system, click Restart Later.

Step 8 In the Audio Codecs Properties dialog box, click OK.

Step 9 In the Sounds and Multimedia dialog box, click OK.

Step 10 Browse to Windows\System32.

Step 11 Rename the file Sl_g729a.acm to be Sl_g729a.old.

Step 12 On the Windows Start menu, click Programs > Cisco Unity > Manage Integrations.

Step 13 In the left pane of the UTIM window, expand the Cisco Unified CallManager SIP trunk integration and click the first cluster.

Step 14 In the right pane, click the SIP Info tab.

Step 15 In the Preferred Codec field, confirm that the setting is G.711 (mu-law). If the field has this setting, continue to Step 16.

If this field has a different setting, do the following substeps.

a. Click G.711 (mu-law).

b. Click Save.

c. When prompted to restart the Cisco Unity services, click No.

Step 16 Repeat Step 13 through Step 15 on all remaining clusters in the Cisco Unified CallManager SIP trunk integration.

Step 17 Restart the Cisco Unity server.


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. 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 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, refer to the "Subscriber Template Call Transfer Settings" section in the Cisco Unity Administrator Help.

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


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.



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 Cisco Unified CallManager 5.1 SIP Trunk Integration Guide for Cisco Unity 4.2 uses the following conventions.

Table 22 Cisco Unified CallManager 5.1 SIP Trunk Integration Guide for Cisco Unity 4.2 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 Cisco Unified CallManager 5.1 SIP Trunk Integration Guide for Cisco Unity 4.2 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, Obtaining Support, and Security Guidelines

For information on obtaining documentation, obtaining support, providing documentation feedback, security guidelines, and also recommended aliases and general Cisco documents, 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