Table of Contents
Task List to Create the Integration
Integrations with Multiple Phone Systems
Planning How the Voice Messaging Ports Are Used
The Number Voice of Messaging Ports to Install
The Number of Voice Messaging Ports That Answers Calls
The Number of Voice Messaging Ports That Dials Out
When Both Servers Are Functioning Normally
When Only One Server Is Functioning
Creating a New Integration with Cisco Unified Communications Manager Express
Adding New User Templates for Multiple Integrations
Appendix: Documentation and Technical Assistance
Cisco Unity Connection Documentation
Obtaining Documentation and Submitting a Service Request
Cisco Product Security Overview
Cisco Unified Communications Manager Express SIP Trunk Integration Guide for Cisco Unity Connection Release 10.x
This document provides instructions for setting up a Cisco Unified Communications Manager Express SIP trunk integration with Cisco Unity Connection.
This document does not apply to the configuration in which Cisco Unity Connection is installed as Cisco Business Edition—on the same server with Cisco Unified Communications Manager.
Note Cisco Unified Communications Manager (CM) Express does not support Keypad Markup Language (KPML) for sending DTMF keystrokes in a SIP message (out-of-band).
Integration Tasks
Before doing the following tasks to integrate Cisco Unity Connection with Cisco Unified Communications Manager Express through a SIP trunk, confirm that the Cisco Unity Connection server is ready for the integration by completing the applicable tasks in the Installation Guide for Cisco Unity Connection.
The following task list describes the process for creating the integration.
Task List to Create the Integration
Use the following task list to set up a Cisco Unified CM Express SIP trunk integration.
1. Review the system and equipment requirements to confirm that all phone system and Cisco Unity Connection server requirements have been met. See the “Requirements” section.
2. Plan how the voice messaging ports will be used by Cisco Unity Connection. See the “Planning How the Voice Messaging Ports Are Used” section.
3. Program Cisco Unified Communications Manager Express. See the “Programming the Cisco Unified Communications Manager Express Phone System for Integrating with Cisco Unity Connection” section.
4. Create the integration. See the “Creating a New Integration with Cisco Unified Communications Manager Express” section.
5. Test the integration. See the “Testing the Integration” section.
6. If this integration is a second or subsequent integration, add the applicable new user templates for the new phone system. See the “Adding New User Templates for Multiple Integrations” section.
Requirements
The Cisco Unified CM Express SIP trunk integration supports configurations of the following components:
- A compatible version of Cisco Unified CM Express.
- For details on compatible versions of Cisco Unified CM Express, see the SIP Trunk Compatibility Matrix: Cisco Unity Connection, Cisco Unified Communications Manager, and Cisco Unified Communications Manager Express at http://www.cisco.com/en/US/products/ps6509/products_device_support_tables_list.html.
- A compatible Cisco IOS software version. See the Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_device_support_tables_list.html.
- Cisco Unified CM Express feature license.
- Cisco IP phone feature licenses, and Cisco licenses for other H.323-compliant devices or software (such as Cisco VirtualPhone and Microsoft NetMeeting clients) that will be connected to the network, as well as one license for each Cisco Unity Connection port.
- For the Cisco Unified CM Express extensions, SIP phones that support DTMF relay as described in RFC-2833. For a list of supported Cisco IP phone models, see the applicable compatibility information document at http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_device_support_tables_list.html.
- For the Cisco Unified CM Express extensions, one of the following configurations:
- Only SIP phones.
- Both SCCP phones and SIP phones.
- Note that older SCCP phone models may require a Media Termination Point (MTP) to function correctly.
- A LAN Unity Connection in each location where you will plug the applicable phone into the network.
- The applicable version of Cisco Unity Connection. For details on compatible versions of Cisco Unity Connection, see the SIP Trunk Compatibility Matrix: Cisco Unity Connection, Cisco Unified Communications Manager, and Cisco Unified Communications Manager Express at http://www.cisco.com/en/US/products/ps6509/products_device_support_tables_list.html.
- Cisco Unity Connection installed and ready for the integration, as described in the Installation Guide for Cisco Unity Connection at http://www.cisco.com/en/US/products/ps6509/prod_installation_guides_list.html.
- A license that enables the applicable number of voice messaging ports.
Cisco Unity Connection supports centralized voice messaging through the phone system, which supports various inter-phone system networking protocols including proprietary protocols such as Avaya DCS, Nortel MCDN, or Siemens CorNet, and standards-based protocols such as QSIG or DPNSS. Note that centralized voice messaging is a function of the phone system and its inter-phone system networking, not voicemail. Unity Connection will support centralized voice messaging as long as the phone system and its inter-phone system networking are properly configured. For details, see the “Centralized Voice Messaging” section in the “Integrating Cisco Unity Connection with the Phone System” chapter of the Design Guide for Cisco Unity Connection Release 10.x at www.cisco.com/en/US/docs/voice_ip_comm/connection/10x/design/guide/10xcucdgx.html.
Integration Description
The Cisco Unified Communications Manager (CM) Express SIP trunk integration uses a LAN to connect Cisco Unity Connection and the phone system. The Cisco Unified Communications Manager Express router also provides connections to the PSTN. Figure 1 shows the connections.
Figure 1 Connections Between the Phone System and Cisco Unity Connection
For a list of supported versions of Cisco Unified CM Express that are qualified to integrate with Cisco Unity Connection through a SIP trunk, see the SIP Trunk Compatibility Matrix: Cisco Unity Connection, Cisco Unified Communications Manager, and Cisco Unified Communications Manager Express at http://www.cisco.com/en/US/products/ps6509/products_device_support_tables_list.html.
This document does not apply to the configuration in which Cisco Unity Connection is installed as Cisco Business Edition—on the same server with Cisco Unified CM.
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 Connection uses this information to answer the call appropriately. For example, a call forwarded to Cisco Unity Connection is answered with the personal greeting of the user. If the phone system routes the call to Cisco Unity Connection without this information, Cisco Unity Connection answers with the opening greeting.
Integration Functionality
The Cisco Unified CM Express SIP trunk integration with Cisco Unity Connection provides the following features:
- Call forward to personal greeting
- Call forward to busy greeting
- Caller ID
- Easy message access (a user can retrieve messages without entering an ID; Cisco Unity Connection identifies a user based on the extension from which the call originated; a password may be required)
- Identified user messaging (Cisco Unity Connection automatically identifies a user who leaves a message during a forwarded internal call, based on the extension from which the call originated)
- Message waiting indication (MWI)
Integrations with Multiple Phone Systems
When Cisco Unity Connection is installed as Cisco Business Edition—on the same server with Cisco Unified Communications Manager—Cisco Unity Connection cannot be integrated with multiple phone systems at one time.
When Cisco Unity Connection is not installed as Cisco Business Edition, Cisco Unity Connection can be integrated with two or more phone systems at one time. For information on and instructions for integrating Cisco Unity Connection with multiple phone systems, see the Multiple Phone System Integration Guide for Cisco Unity Connection Release 10.x at http://www.cisco.com/en/US/products/ps6509/products_installation_and_configuration_guides_list.html.
Planning How the Voice Messaging Ports Are Used
Before programming the phone system, you need to plan how the voice messaging ports will be used by Cisco Unity Connection. 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):
For a Cisco Unity Connection cluster, each Cisco Unity Connection server must have enough ports to handle all voice messaging traffic in case the other server stops functioning.
- 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 set message waiting indicators (MWIs), and to make telephone record and playback (TRAP) connections.
The following table describes the voice messaging port settings in Cisco Unity Connection that can be set on Telephony Integrations > Port of Cisco Unity Connection Administration.
The Number Voice of Messaging Ports to Install
The number of voice messaging ports to install depends on numerous factors, including:
- The number of calls Cisco Unity Connection will answer when call traffic is at its peak.
- The expected length of each message that callers will record and that users will listen to.
- The number of users.
- The number of calls made for message notification.
- The number of MWIs that will be activated when call traffic is at its peak.
- The number of TRAP connections needed when call traffic is at its peak. (TRAP connections are used by Cisco Unity Connection 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.
- Whether a Cisco Unity Connection cluster is configured. For considerations, see the “Considerations for a Cluster” section.
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 Answers Calls
The calls that the voice messaging ports answer can be incoming calls from unidentified callers or from users. 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.
If your system is configured for a Cisco Unity Connection cluster, see the “Considerations for a Cluster” section.
The Number of Voice Messaging Ports That Dials Out
Ports that will only dial out can do one or more of the following:
- Notify users by phone, pager, or email of messages that have arrived.
- Turn MWIs on and off for user extensions.
- Make a TRAP connection so that users can use the phone as a recording and playback device in Cisco Unity Connection web applications.
If your system is configured for a Cisco Unity Connection cluster, see the “Considerations for a Cluster” section.
Considerations for a Cluster
If your system is configured for a Cisco Unity Connection cluster, consider how the voice messaging ports will be used in different scenarios.
When Both Servers Are Functioning Normally
- A hunt group is configured to send incoming calls first to the subscriber server, then to the publisher server if no answering ports are available on the subscriber server.
- Both Cisco Unity Connection servers are active and handle voice messaging traffic for the system.
- In Cisco Unity Connection Administration, the voice messaging ports are configured so that an equal number of voice messaging ports are assigned to each Cisco Unity Connection server. This guide directs you to assign the voice messaging ports to their specific server at the applicable time.
- The number of voice messaging ports that are assigned to one Cisco Unity Connection server must be sufficient to handle all of the voice messaging traffic for the system (answering calls and dialing out) when the other Cisco Unity Connection server stops functioning.
If both Cisco Unity Connection servers must be functioning to handle the voice messaging traffic, the system will not have sufficient capacity when one of the servers stops functioning.
If all the voice messaging ports are assigned to one Cisco Unity Connection server, the other Cisco Unity Connection server will not be able to answer calls or to dial out.
When Only One Server Is Functioning
- The hunt group on the phone system sends all calls to the functioning Cisco Unity Connection server.
- The functioning Cisco Unity Connection server receives all voice messaging traffic for the system.
- The number of voice messaging ports that are assigned to the functioning Cisco Unity Connection server must be sufficient to handle all of the voice messaging traffic for the system (answering calls and dialing out).
- The functioning Cisco Unity Connection server must have voice messaging ports that will answer calls and that can dial out (for example, to set MWIs).
If the functioning Cisco Unity Connection server does not have voice messaging ports for answering calls, the system will not be able to answer incoming calls. Similarly, if the functioning Cisco Unity Connection server does not have voice messaging ports for dialing out, the system will not be able to dial out (for example, to set MWIs).
Programming the Cisco Unified Communications Manager Express Phone System for Integrating with Cisco Unity Connection
For details on programming the Cisco Unified Communications Manager Express router for the integration with Cisco Unity Connection, see the “Integrating Voice Mail” chapter of the Cisco Unified Communications Manager Express System Administrator Guide at http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_installation_and_configuration_guides_list.html.
After you have configured the Cisco Unified CM Express router for the integration, do the applicable following procedures:
- For Cisco Unified CM Express 4.1 or later, if calls can be received from a Cisco Unified Communications Manager SIP trunk, do the To Configure the Cisco Unified Communications Manager Express Router When It is Connected to a Cisco Unified Communications Manager SIP Trunk (Cisco Unified Communications Manager Express 4.1 or Later).
- For a Cisco Unity Connection cluster, do the To Configure the Cisco Unified Communications Manager Express Router for a Cisco Unity Connection Cluster.
To Configure the Cisco Unified Communications Manager Express Router When It is Connected to a Cisco Unified Communications Manager SIP Trunk (Cisco Unified Communications Manager Express 4.1 or Later)
Step 1 On the Cisco Unified CM Express router, go into the global configuration mode by entering the following command:
Step 2 To enter the voice service configuration mode, enter the following command:
Step 3 To disable the 302 “Moved Temporarily” SIP message, enter the following command:
no supplementary-service sip moved-temporarily
Step 4 To exit the global configuration mode, enter the following command:
To Configure the Cisco Unified Communications Manager Express Router for a Cisco Unity Connection Cluster
Step 1 On the Cisco Unified CM Express router, go into the global configuration mode by entering the following command:
Step 2 To enter dial-peer configuration mode for port group for the publisher server, enter the following command:
Step 3 To set the description for the dial-peer, enter the following command:
description <name of publisher server>
Step 4 To set the Cisco Unity Connection pilot number for the dial-peer, enter the following command:
destination-pattern <pilot number>
Step 5 To configure the dial-peer to use Session Initiation Protocol (SIP) for calls, enter the following command:
Step 6 To specify the IP address (or DNS name) of the publisher server, enter the following command:
session target {ipv4:<IP address>|dns:<host name>}
Step 7 To enable DTMF relay, enter the following command:
Step 8 To set the codec for calls, enter the following command:
Step 9 To equalize the number of calls sent to each Cisco Unity Connection server in the Unity Connection cluster, enter the following command:
max-conn <number of ports handled by publisher server>
Step 10 To set the dial-peer preference for port group so that the calls will be routed first to the subscriber server, then to the publisher server if no ports are available on the subscriber server, enter the following command:
Step 11 To enable huntstop, enter the following command:
Step 12 To exit the global configuration mode, enter the following command:
Step 13 To set a dial-peer that will prevent Cisco Unified CM Express from hunting beyond the dial-peers for the Cisco Unity Connection port groups, entering the following command:
Step 14 To enter dial-peer configuration mode, enter the following command:
Step 15 To set the description for the dial-peer, enter the following command:
description <name of subscriber server>
Step 16 To set the Cisco Unity Connection pilot number for the dial-peer, enter the following command:
destination-pattern <pilot number>
Step 17 To configure the dial-peer to use Session Initiation Protocol (SIP) for calls, enter the following command:
Step 18 To specify the IP address (or DNS name) of the subscriber server, enter the following command:
session target {ipv4:<IP address>|dns:<host name>}
Step 19 To enable DTMF relay, enter the following command:
Step 20 To set the codec for calls, enter the following command:
Step 21 To equalize the number of calls sent to each Cisco Unity Connection server in the Unity Connection cluster, enter the following command:
max-conn <number of ports handled by subscriber server>
Step 22 To set the dial-peer preference so that the calls will be routed first to the subscriber server, then to the publisher server if no ports are available on the subscriber server, enter the following command:
Step 23 To disable huntstop so that the calls will use the next available voice messaging port on the subscriber server, then use voice messaging ports on the publisher server if no ports are available on the subscriber server, enter the following command:
Step 24 To exit the global configuration mode, enter the following command:
The following is an example of the configuration without a Cisco Unity Connection cluster:
The following is an example of the configuration with a Cisco Unity Connection cluster configured:
Creating a New Integration with Cisco Unified Communications Manager Express
After ensuring that Cisco Unified Communications Manager Express and Cisco Unity Connection are ready for the integration, do the following procedure to set up the integration and to enter the port settings.
Step 1 Sign in to Cisco Unity Connection Administration.
Step 2 In Cisco Unity Connection Administration, expand Telephony Integrations, then select Phone System.
Step 3 On the Search Phone Systems page, under Display Name, select the name of the default phone system.
Step 4 On the Phone System Basics page, in the Phone System Name field, enter the descriptive name that you want for the phone system.
Step 5 If you want to use this phone system as the default for TRaP connections so that administrators and users without voicemail boxes can record and playback through the phone in Cisco Unity Connection web applications, check the Default TRAP Switch check box. If you want to use another phone system as the default for TRaP connections, uncheck this check box.
Step 7 On the Phone System Basics page, in the Related Links drop-down box, select Add Port Group and select Go.
Step 8 On the New Port Group page, enter the applicable settings and select Save.
Select the name of the phone system that you entered in Step 4.
Select Port Group Template and select SIP in the drop-down box.
Enter a descriptive name for the port group. You can accept the default name or enter the name that you want.
Check this check box if you want Cisco Unity Connection to authenticate with the Cisco Unified CM Express router.
Enter the name that Cisco Unity Connection will use to authenticate with the Cisco Unified CM Express router.
Enter the password that Cisco Unity Connection will use to authenticate with the Cisco Unified CM Express router.
Enter the voice messaging line name (or pilot number) that users will use to contact Cisco Unity Connection and that Cisco Unity Connection will use to register with the Cisco Unified CM Express router.
Select the SIP security profile that Cisco Unity Connection will use.
Note Cisco Unified CM Express does not support Cisco Unified CM authentication and encryption.
Select the SIP transport protocol that Cisco Unity Connection will use.
Enter the IP address (or host name) of the Cisco Unified CM Express router that you are integrating with Cisco Unity Connection.
Do not enter a value in this field. IPv6 is not supported for Cisco Unified CM Express integrations.
Enter the IP address (or host name) of the Cisco Unified CM Express router that you are integrating with Cisco Unity Connection.
Enter the IP port of the Cisco Unified CM Express router that you are integrating with Cisco Unity Connection. We recommend that you use the default setting.
Step 9 On the Port Group Basics page, in the Related Links drop-down box, select Add Ports and select Go.
Step 10 On the New Port page, enter the following settings and select Save.
Enter the number of voice messaging ports that you want to create in this port group.
Note For a Cisco Unity Connection cluster, you must enter the total number of voice messaging ports that will be used by all Cisco Unity Connection servers. Each port will later be assigned to a specific Cisco Unity Connection server.
Select the name of the phone system that you entered in Step 4.
Select the name of the port group that you added in Step 8.
Step 11 On the Search Ports page, select the display name of the first voice messaging port that you created for this phone system integration.
Note By default, the display names for the voice messaging ports are composed of the port group display name followed by incrementing numbers.
Step 12 On the Port Basics page, set the voice messaging port settings as applicable. The fields in the following table are the ones that you can change.
Step 15 Repeat Step 12 through Step 14 for all remaining voice messaging ports for the phone system.
Step 16 If another phone system integration exists, in Cisco Unity Connection Administration, expand Telephony Integrations, then select Trunk. Otherwise, skip to Step 20.
Step 17 On the Search Phone System Trunks page, on the Phone System Trunk menu, select New Phone System Trunk.
Step 18 On the New Phone System Trunk page, enter the following settings for the phone system trunk and select Save.
Step 19 Repeat Step 17 and Step 18 for all remaining phone system trunks that you want to create.
Step 20 In the Related Links drop-down list, select Check Telephony Configuration and select Go to confirm the phone system integration settings.
If the test is not successful, the Task Execution Results displays one or more messages with troubleshooting steps. After correcting the problems, test the Unity Connection again.
Step 21 In the Task Execution Results window, select Close.
Testing the Integration
To test whether Cisco Unity Connection and the phone system are integrated correctly, do the following procedures in the order listed.
If any of the steps indicate a failure, see the following documentation as applicable:
- The installation guide for the phone system.
- Troubleshooting Guide for Cisco Unity Connection Release 10.x at http://www.cisco.com/en/US/docs/voice_ip_comm/connection/10x/troubleshooting/guide/10xcuctsgx.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 Connection is connected to.
Step 2 Set Phone 1 to forward calls to the Cisco Unity Connection pilot number when calls are not answered.
Caution The phone system must forward calls to the Cisco Unity Connection pilot number in no fewer than four rings. Otherwise, the test may fail.
Step 3 In Cisco Unity Connection Administration, expand Users, then select Users.
Step 4 On the Search Users page, select the display name of a user to use for testing. The extension for this user must be the extension for Phone 1.
Step 5 On the Edit User Basics page, uncheck the Set for Self-enrollment at Next Login check box.
Step 6 In the Voice Name field, record a recorded name for the test user.
Step 8 On the Edit menu, select Message Waiting Indicators.
Step 9 On the Message Waiting Indicators page, select the message waiting indicator. If no message waiting indication is in the table, select Add New.
Step 10 On the Edit Message Waiting Indicator page, enter the following settings.
Step 12 On the Edit menu, select Transfer Rules.
Step 13 On the Transfer Rules page, select the active transfer rule.
Step 14 On the Edit Transfer Rule page, under Transfer Action, select Extension and enter the extension of Phone 1.
Step 15 In the Transfer Type field, select Release to Switch.
Step 17 Minimize the Cisco Unity Connection Administration window.
Do not close the Cisco Unity Connection Administration window because you will use it again in a later procedure.
Step 18 Sign in to the Real-Time Monitoring Tool (RTMT).
Step 19 On the Unity Connection menu, select Port Monitor. The Port Monitor tool appears in the right pane.
Step 20 In the right pane, select Start Polling. The Port Monitor will display which port is handling the calls that you will make.
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 Connection.
Step 2 In the Port 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 Connection 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 Connection and that you hear the greeting for the test user. Hearing the greeting means that the phone system forwarded the unanswered call and the call-forward information to Cisco Unity Connection, which correctly interpreted the information.
Step 7 On the Port Monitor, note which port handles this call.
Step 8 Leave a message for the test user and hang up Phone 2.
Step 9 In the Port 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 Connection are successfully integrated for turning on MWIs.
Step 1 From Phone 1, enter the internal pilot number for Cisco Unity Connection.
Step 2 When asked for your password, enter the password for the test user. Hearing the request for your password means that the phone system sent the necessary call information to Cisco Unity Connection, which correctly interpreted the information.
Step 3 Confirm that you hear the recorded name for the test user (if you did not record a name for the test user, you will hear the extension number for Phone 1). Hearing the recorded name means that Cisco Unity Connection correctly identified the user by the extension.
Step 5 After listening to the message, 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 Connection are successfully integrated for turning off MWIs.
Step 8 On the Port 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 Connection
Step 1 In Cisco Unity Connection Administration, on the Edit Transfer Rule page for the test user, in the Transfer Type field, select Supervise Transfer.
Step 2 In the Rings to Wait For field, enter 3.
Step 4 Minimize the Cisco Unity Connection Administration window.
Do not close the Cisco Unity Connection Administration window because you will use it again in a later procedure.
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 Connection.
Step 2 On the Port 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).
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 Connection is supervising the transfer.
Step 6 Confirm that, after three rings, you hear the greeting for the test user. Hearing the greeting means that Cisco Unity Connection successfully recalled the supervised-transfer call.
Step 7 During the greeting, hang up Phone 2.
Step 8 On the Port 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.
Adding New User Templates for Multiple Integrations
When you create the first phone system integration, this first phone system is automatically selected in the default user template. The users that you add after creating this phone system integration will be assigned to this phone system by default.
However, for each additional phone system integration that you create, you must add the applicable new user templates that will assign users to the new phone system. You must add the new templates before you add new users who will be assigned to the new phone system.
For details on adding new user templates, or on selecting a user template when adding a new user, see System Administration Guide for Cisco Unity Connection Release 10.x. The guide is available at
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/10x/administration/guide/10xcucsagx.html.
Documentation Conventions
The Cisco Unified Communications Manager Express SIP Trunk Integration Guide for Cisco Unity Connection Release 10.x uses the following conventions.
The Cisco Unified Communications Manager Express SIP Trunk Integration Guide for Cisco Unity Connection Release 10.x also uses the following conventions:
Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the document.
Cisco Unity Connection Documentation
For descriptions and URLs of Cisco Unity Connection documentation on Cisco.com, see the Documentation Guide for Cisco Unity Connection. The document is shipped with Cisco Unity Connection and is available at http://www.cisco.com/en/US/products/ps6509/products_documentation_roadmaps_list.html.
Obtaining Documentation and Submitting a Service Request
For information on obtaining documentation, submitting a service request, and gathering additional information, see the monthly What’s New in Cisco Product Documentation, which also lists all new and revised Cisco technical documentation, at:
http://www.cisco.com/en/US/docs/general/whatsnew/whatsnew.html
Subscribe to the What’s New in Cisco Product Documentation as a Really Simple Syndication (RSS) feed and set content to be delivered directly to your desktop using a reader application. The RSS feeds are a free service and Cisco currently supports RSS Version 2.0.
Cisco Product Security Overview
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