Table Of Contents
Cisco Unified CallManager 5.0 SIP Trunk Integration Guide for Cisco Unity 4.2
Integration Tasks
Task List to Create the Integration
Task List to Make Changes to an Integration
Task List to Change the Number of Voice Messaging Ports
Task List to Delete an Existing Integration
Requirements
Integration Description
Call Information
Integration Functionality
Integrations with Multiple Phone Systems
Planning How the Voice Messaging Ports Will Be Used by Cisco Unity
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
Changing the Settings for an Existing Integration
Changing the Number of Voice Messaging Ports
Deleting an Existing 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 5.0 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.
Task List to Make Changes to an Integration
Use the following task list to make changes to an integration after it has been created.
1. Start the Cisco Unity Telephony Integration Manager (UTIM). See the "Changing the Settings for an Existing Integration" section.
2. Make the changes you want to the existing integration. See the "Changing the Settings for an Existing Integration" section.
Task List to Change the Number of Voice Messaging Ports
Use the following task list to change the number of voice messaging ports for an integration after it has been created.
1. Change the number of voice messaging ports in Cisco Unified CallManager Administration. See the "Changing the Number of Voice Messaging Ports" section.
2. Start the Cisco Unity Telephony Integration Manager (UTIM). See the "Changing the Number of Voice Messaging Ports" section.
3. Make the changes you want to the number of voice messaging ports in the Cisco Unity Administrator. See the "Changing the Number of Voice Messaging Ports" section.
Task List to Delete an Existing Integration
Use the following task list to remove an existing integration.
1. Start the Cisco Unity Telephony Integration Manager (UTIM). See the "Deleting an Existing Integration" section.
2. Delete the existing integration. See the "Deleting an Existing Integration" section.
Requirements
The Cisco Unified CallManager integration supports configurations of the following components:
Phone System
•A Cisco IP telephony applications server consisting of Cisco Unified CallManager 5.0(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 Deliver - 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.
Changing the Settings for an Existing Integration
After the integration is set up, if you want to change any of its settings (for example, to change the MWI settings), do the following procedure.
If you want to change the number of voice messaging ports, see the Changing the Number of Voice Messaging Ports.
To Change the Settings for an Integration
Step 1 On the Cisco Unity server, on the Windows Start menu, click Programs > Cisco Unity > Manage Integrations. The UTIM window appears.
Step 2 In the left pane, double-click Unity Server. The existing integrations appear.
Step 3 Click the integration you want to modify.
Step 4 In the right pane, click the cluster for the integration.
Step 5 In the right pane, click the applicable tab for the integration.
Step 6 Enter new settings in the fields that you want to change.
Caution If you are adding or removing voice messaging ports, make sure you change the settings for the individual ports so that there are an appropriate number of ports set to answer calls and an appropriate number of ports set to dial out.
Step 7 In the UTIM window, click Save.
Step 8 If prompted, restart the Cisco Unity services.
Changing the Number of Voice Messaging Ports
To change the number of voice messaging ports after you have finished installing and setting up Cisco Unified CallManager, do the following procedure.
To Update Cisco Unity for Additional Voice Messaging Ports
Step 1 If the Cisco Unity license does not enable the additional voice messaging ports you added, see your sales representative to request the applicable license.
Step 2 When you have the license, on the Cisco Unity server, click Programs > Cisco Unity > Licensing.
Step 3 On the Action menu, click Install License Files.
Step 4 Follow the on-screen instructions.
Note If you increase the number of voice messaging ports from 32 or fewer to more than 32, you must also install SQL Server 2000 as described in the applicable Cisco Unity installation guide.
Step 5 On the Cisco Unity server, on the Windows Start menu, click Programs > Cisco Unity > Manage Integrations. The UTIM window appears.
Step 6 In the left pane, double-click Unity Server. The existing integrations appear.
Step 7 Click the integration you want to modify.
Step 8 In the right pane, click the cluster for the integration.
Step 9 In the right pane, click the applicable tab for the integration.
Step 10 Enter new settings in the fields that you want to change.
Caution If you are adding or removing voice messaging ports, make sure you change the settings for the individual ports so that there are an appropriate number of ports set to answer calls and an appropriate number of ports set to dial out.
Step 11 In the UTIM window, click Save.
Step 12 If prompted, restart the Cisco Unity services.
Deleting an Existing Integration
If you want to delete an existing integration (for example, you have replaced the phone system with which Cisco Unity originally integrated), do the following procedure.
To Delete an Existing Integration
Step 1 On the Cisco Unity server, on the Windows Start menu, click Programs > Cisco Unity > Manage Integrations. The UTIM window appears.
Step 2 In the left pane, double-click Unity Server. The existing integrations appear.
Step 3 Click the integration that you want to delete.
Step 4 On the Integration menu, click Delete.
Step 5 Follow the on-screen instructions to assign the subscribers of the deleted phone system integration to another phone system integration.
Step 6 At the prompt to restart the Cisco Unity services, click Yes. The Cisco Unity services restart.
Alternatively, you can restart the Cisco Unity services in UTIM on the Tools menu by clicking Restart Cisco Unity.
Step 7 If the integration you deleted used voice cards, remove the voice cards from the Cisco Unity server.
Appendix: 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 5.0 SIP Trunk Integration Guide for Cisco Unity 4.2 uses the following conventions.
Table 22 Cisco Unified CallManager 5.0 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 5.0 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
Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.
© 2007 Cisco Systems, Inc. All rights reserved.
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