Installing and Configuring Guide for Cisco HCS for CC 11.0(1)

SPAN-Based Monitoring

Install SPAN based Silent Monitoring
SPAN-Based Silent Monitoring Configuration

Install SPAN based Silent Monitoring
SPAN-Based Silent Monitoring Configuration

Install SPAN based Silent Monitoring

Procedure

Step 1	Mount the Cisco CCE CTI ISO image.
Step 2	Run setup.exe file to install SPAN based Silent Monitoring.
Step 3	On CTIOS Silent Monitoring Service page, Click Yes to stop CTIOS Silent Monitor process
Step 4	In Software License Agreement page Click Yes and click Continue.
Step 5	Enter the MR patch browse location and click Next. If you do not know the MR patch browser location, leave the filed blank and click Next.
Step 6	In Choose Destination Location page, browse to the directory where you want to install and click Next.
Step 7	Enter the following information in the Cisco CTIOS Silent Monitor - Install Shield Wizard: Host Name\IP Address: Host Name of the silent monitor server. Port: Enter the port number 42228 on which the Silent Monitor Service listens for incoming connections. Check Silent Monitor Server: Select this to allow the Silent Monitor Service to monitor multiple Mobile Agents simultaneously. Enter peer(s) information: Select this if this Silent Monitor Service is part of a cluster of Silent Monitor Services.
Step 8	Click Next.
Step 9	On CTIOS Silent Monitor page, do not check Enable Security. Click OK.
Step 10	Click Finish to complete the installation.

SPAN-Based Silent Monitoring Configuration

Configurations for SPAN from Gateway
Configurations for SPAN from Call Manager

Configurations for SPAN from Gateway
Configurations for SPAN from Call Manager

Configurations for SPAN from Gateway

This section describes the additional configuration required for Mobile Agent deployment:

For Mobile Agents, the voice path crosses the Public Switched Telephone Network (PSTN) and two gateways.
One gateway control calls from customer phones. The other gateway controls calls from agents, known as agent gateway.
In a Mobile Agent deployment, the Silent Monitor service uses a SPAN port to receive the voice traffic that passes through the agent gateway. This requires the computer running the Silent Monitor service to have two NIC cards; one to handle communications with clients and another to receive all traffic spanned from the switch.

For example, if the agent gateway is connected to port 1 and the NIC (on the Silent Monitor Server that receives SPAN traffic) is connected on port 10, use the following commands to configure the SPAN session:

monitor session 1 source interface fastEthernet0/1

monitor session 1 destination interface fastEthernet0/10
To deploy Silent Monitoring for the Mobile Agent, there must be two gateways; one gateway for agent traffic and another for caller traffic.
If you use one gateway for both agent and caller traffic, the voice traffic does not leave or cross the agent gateway and therefore cannot be silently monitored.
For example, agent-to-agent and consultation calls between Mobile Agents share the same gateway and cannot be silently monitored. Most Mobile Agent deployments only allow silent monitoring for calls between agents and customers.
Install Silent Monitor service on the supervisors desktop, but you need not configure Silent Monitor service for the Mobile Agents. You must configure the agent to use one or more Silent Monitor Servers in the CTI OS Server setup program.
Agents who are both mobile and regular agents require at least two profiles.

The profiles for regular agents do not contain any Silent Monitor service information.

The profiles for Mobile Agents, contains information used to connect to a Silent Monitor Server.

Silent Monitor Service Clusters

Silent Monitor Service Clusters

If more than one agent gateway is present in the call center and an agent can use either gateway to log in, cluster the Silent Monitor services to support Silent Monitor as follows.

Deploy a separate silent monitor server for each gateway.
Configure a SPAN port for each silent monitor server as described in the previous section.
Run the Silent Monitor server installer to install and configure two Silent Monitor servers as peers.
Configure the following to set up a connection profile to instruct the agent desktops to connect to one of the peers:
1. Check the Enter peers information check box.
2. Enter the IP address of the other silent monitor service in the Hostname/IP address.

Configurations for SPAN from Call Manager

Span from Call Manager is recommended for small agent contact center only as in this deployment model CUCM software resources are being used .

Before You Begin

To Span from CUCM ensure that SM server should be on the same blade as CUCM. Ensure that CUCM uses its own mtp resources ,when the agent is logged into a phone across a gateway.

This requires the computer running the Silent Monitor service to have two NIC cards; one to handle communications with clients and another to receive all traffic spanned from the nexus.

Procedure

Use the following commands to configure the LOCAL SPAN session in nexus :

monitor session 1
description LOCAL-SPAN
source interface Vethernet76 both

where : Vethernet76 is the interface of CUCM(used for spanning) on the switch.

Unified CCE AW-HDS-DDS

To install Unified CCE AW-HDS-DDS, see Create Golden Template for Unified CCE AW-HDS-DDS and to configure see Configure Unified CCE AW-HDS-DDS.

Cisco RSM

Create Golden Template for Cisco Remote Silent Monitoring
Configure Cisco RSM

Create Golden Template for Cisco Remote Silent Monitoring
Configure Cisco RSM

Create Golden Template for Cisco Remote Silent Monitoring

Follow this sequence of tasks to create the golden template for the Cisco RSM server.

After each task, return to this page to mark the task "done" and continue the sequence.

Sequence	Tasks	Notes
1	Download HCS-CC_11.0(1)_CCE-RSM_vmv9_v1.0.ova.	See Open Virtualization Format Files.
2	Create the virtual machine for the Cisco RSM server.	Follow the procedure Create Virtual Machines.
3	Install Microsoft Windows Server	Follow the procedure Install Microsoft Windows Server 2012 R2 Standard Edition.
5	Install antivirus software.	Follow the procedure Install Antivirus Software.
6	Install the JTAPI Client.	Follow the procedure Install the JTAPI Client.
7	Configure SNMP Traps for Cisco RSM	Follow the procedure Configuring SNMP Traps for Cisco RSM
8	Install the Cisco RSM server.	Follow the procedure Install the Cisco RSM Server.
9	Convert the virtual machine to a template.	Follow the procedure Convert the Virtual Machine to a Golden Template.

After you create all golden templates, you can run the automation process (Automated Cloning and OS Customization). After you run the automation process, you can configure the Cisco RSM server on the destination system. See Configure Cisco RSM.

Install the JTAPI Client
Install the Cisco RSM Server
Configuring SNMP Traps for Cisco RSM

Install the JTAPI Client

Complete the following procedure to install JTAPI on the Cisco RSM server.

Procedure

Step 1

Start the Unified Communications Manager Administration application in a browser.

Step 2

Login using the administrator credentials.

Step 3

Navigate to Application > Plugins and then click Find.

Step 4

Download the Cisco JTAPI 32-bit Client for Windows.

Step 5

Install the downloaded file and accept all the default settings.

Note

In the Cisco TFTP IP Address text-box enter the CUCM Subscriber IP Address.
For Small Contact Center agent deployment model this is optional as RSM needs to be connected to multiple sub customer CUCM clusters

Install the Cisco RSM Server

Complete the following procedure to install the Cisco RSM Server.

Procedure

Step 1	Mount the Cisco RSM ISO image to the virtual machine. For more information, see Mount and Unmount ISO Files.
Step 2	Run the setup.exe file to install the RSM server. The RSM installer program starts and it displays the Cisco Remote Silent Monitoring(RSM) InstallShield window.
Step 3	Click Next. It displays the Licence Agreement page.
Step 4	In the Licence Agreement page, accept the License. Click Next.
Step 5	In the service Login Information page, provide the administrator credentials of RSM Virtual machine. Click Next.
Step 6	In the Launch Configuration Settings page, click Exit from the setup. Click Yes on the pop-up window.
Step 7	Click Finish.

What to Do Next

Configure RSM

Configuring SNMP Traps for Cisco RSM

Simple Network Management Protocol (SNMP) traps may be raised from Cisco RSM by configuring Windows to send selected events to an SNMP monitor. This is achieved using a Windows utility called evntwin.exe. This utility converts events written to the Windows Event log into SNMP traps that are raised and forwarded by the Windows SNMP service to an SNMP management tool.

Complete the following procedures to configure SNMP traps for use with Cisco RSM:

Configure SNMP Agent in MIB

Configure SNMP Agent in MIB

The following information is to connect the RSM SNMP Agent and to root the MIB Object.

RSM SNMP Agent Connection: < RSM Server IP >:33161
RSM SNMP Agent Root OID: .1.3.6.1.4.1.9.9.2776 - ciscoRSMMIB

Configure Cisco RSM

Configure Cisco RSM for 500 and 1000 Agent Deployment
Configure Cisco RSM for 4000 Agent Deployment
Configure Cisco RSM for Small Contact Center Deployment
Configure Cisco RSM for 12000 Agent Deployment
Configure Cisco RSM for A-Law Codec

The following figure shows the configuration topology for Remote Silent Monitoring.

Figure 1. Cisco Remote Silent Monitoring Configuration Topology

Configure Cisco RSM for 500 and 1000 Agent Deployment
Configure Cisco RSM for 4000 Agent Deployment
Configure Cisco RSM for 12000 Agent Deployment
Configure Cisco RSM for Small Contact Center Deployment
Configure Cisco RSM for A-Law Codec

Configure Cisco RSM for 500 and 1000 Agent Deployment

Configure the Cisco RSM (Remote Silent Monitoring) Server for 500 and 1000 agent deployment in the distributed mode, in the following order:

Required Software	Tasks
Configure RSM	Set RSM Configuration Settings for 500 and 1000 Agent Deployment
	Configure JTAPI Client Preferences
	Edit Registry Settings
Configure Gateway	Set Up the VXML Gateway
Configure Unified CVP	Upload RSM Prompts
	Integrate the CVP Call Flow
	Call Flow Deployment
Configure Unified CCE	Set the Agent Target Rule
	Create the Supervisor Login Account
	Create Routing Script for RSM
Configure Unified Call Manager	Configure Simulated Phone
	Set Up the Login Pool Simphone
	Create RSM Application User

Configure RSM
Configure Gateway
Configure Unified CVP
Configure Unified CCE
Configure Unified Communication Manager

Configure RSM

Set RSM Configuration Settings for 500 and 1000 Agent Deployment

Procedure

Step 1

Complete the Mail Server configuration settings:

Choose Start > CiscoRSM > RSM Configuration Manager.
Check Send Email Alert checkbox.
Enter the Host Name/IP address of the mail server in Mail Server Host Name/IP text box.
Enter the email port number in Port text box.
Enter the sender email ID in Sender Email Address text box.
Enter the receiver email ID in Receiver Email Address text box.
Click Next.

Step 2

Complete the Miscellaneous configuration settings:

Enter 1800 in Problem Call Minimum Duration text box.
Enter 4 in Problem Call Min Holds text box.
Enter 3600 in Max Stale Call Duration text box.
Set blank value for CTI OS Trace Mask.
Select INFO from the Log Level drop down list for VL Engine.
Enter 8080 in HTTP Listen Port text box for VL Engine.
Enter 480 in the Audio Buffer Len To VRU text box for PhoneSim.
Note The default value of Audio Buffer Len to VRU is 160, for CVP environment the value is set to 480.
Select INFO from the Log Level drop down list for PhoneSim.
Enter 29001 in HTTP Listen Port text box for PhoneSim.
Enter 29554 in RTSP Listen Port text box for PhoneSim.
Select the RTSP u-law for Audio Encoding from the drop down list for Phonesim.
Select No from the Do HTTP Chunked Transfers drop down list for PhoneSim.
Enter the IP Address of RSM server in the Host Data IP text box.
Click Next.

Step 3

Define Cluster configuration settings:

These settings are used to configure each Unified Communications Manager cluster with the agents to be monitored by RSM.

Click Add Cluster
Enter a cluster name in ClusterN_Name text box.
Note
Name should be alphanumeric.
Enter 5 in No. of Login Pool Simphones text box.
Enter 60 in No. of Monitoring Phones text box. (this is to monitor 60 concurrent).
Enter 5000 in the Peripheral ID text box.
Enter the rsmuser in JTAPI Username text box.
Enter the rsmuser password in JTAPI Password text box.
Enter the first MAC address to use for auto-generation of MAC range for simphone device names in Start MAC Range text box.

Enter the first extension number to use for auto-generation of line extension range for simphone DNs in Start Line Num Range text box.

Note

Line extension ranges must not overlap between clusters. Correlates to ClusterN_PhoneSim_StartMACRange value.
The Start Line Num Range should be between 4 to 15 digits.

Select TCP from the SIP Transport drop down list.
Click Next.

Step 4

Define Unified Communications Manager configuration settings:

Enter the host name / IP address of CUCM1 server(Subscriber1) in Host Name/IP text box.
Enter CUCM1 port as 5060 in Port text box.
Enter the host name / IP address of CUCM2 server(Subscriber 2) in Host Name/IP text box.
Enter CUCM2 port as 5060 in Port text box.
Click Next.

Step 5

In UCCE Integration page select UCCE integrate with CTIOS OR UCCE integrate with CTI

If UCCE Integration with CTIOS is selected, perform the following instructions:
1. Enter the host name / IP address of CTIOS 1A in CTIOS 1A Host Name/IP.
2. Enter 42028 in CTIOS 1A Port text box.
3. Enter the host name / IP address of CTIOS 1B in CTIOS 1B Host Name/IP.
4. Enter 42028 in CTIOS 1B Port text box.
5. Click Next.

If UCCE Integration with CTI is selected, perform the following instructions:
1. Enter the host name / IP address of CTI 1A in CTI 1A Host Name/IP.
2. Enter 42027 in CTI 1A Port text box.
3. Enter the host name / IP address of CTI 1B in CTI 1B Host Name/IP.
4. Enter 43027 in CTI 1B Port text box.
5. Click Next.

Step 6

Click Next and Check Start PhoneSim Service and Start VLEngine Service check boxes.

Step 7

Click Finish.

Configure JTAPI Client Preferences

Procedure

Step 1	Choose Start > All Programs > Cisco JTAPI and click Cisco Unified Communications Manager JTAPI Preferences.
Step 2	Click Language tab.
Step 3	Select English from the Select Language drop-down list.
Step 4	Enter the TFTP Server IP Address.
Step 5	Click OK.

Edit Registry Settings

RSM requires numeric supervisor accounts, so that users can log in through the telephone. However, Unified CCE supervisor agent accounts are also Active Directory user accounts and an Active Directory security policy can prevent numeric-only accounts. To resolve this issue, modify the "VLEngine_PassPrefix" parameter.

Procedure

Step 1

Access the Registry Editor, Start > Run > regedit.

Step 2

Navigate to HKEY_Local_Machine > Software > Wow6432Node > Cisco Systems, Inc. > Remote Silent Monitoring.

Step 3

SetVLEngine_PassPrefix with a string that prepends the password before it submits for CTI OS Validation.

For Example: If "VLEngine_PassPrefix" String is set to RSM1RSM and you want a supervisor to log in with PIN 1234, then set supervisor's password to RSM1RSM1234.

Note

The valid values are any string of letters, numbers, and valid password symbols (no whitespace and control characters).

Configure Gateway

Set Up the VXML Gateway

RSM is supported on any VXML gateway models and versions of Cisco IOS supporting CVP. The Ingress/VXML gateway can be shared between RSM and other features.

To set up the VXML gateway for RSM, make sure that the IVR prompt memory is at least 8 Mb, by issuing the ivr prompt memory 8000 command.

Note

If the gateway is shared with RSM along with other features, the gateway performance reduces by 20%.

Configure Unified CVP

Upload RSM Prompts

Procedure

Step 1	Navigate to your media server directory, at C:\inetpub\wwwroot\en-us\, and create a new directory labeled VL.
Step 2	Navigate to your RSM server and copy prompts.zip from C:\ CiscoRSM\callflows and unzip the contents into the VL directory of the media server.
Step 3	Right-click the VL directory, then click Properties.
Step 4	Click the Security tab, click Advanced and click Change Permission.
Step 5	Select Include inheritable permission from object's parent and Replace all child object permission with inheritable permissions from this object check-boxes.
Step 6	Click OK and click Yes on the windows security pop up window.
Step 7	Open your web browser and navigate to the VL directory of your media server, that is http://<SERVER IP>/en-us/VL. Ensure that the prompt files are listed and accessible.

Integrate the CVP Call Flow

Procedure

Step 1

Navigate to the C:\CiscoRSM\callflows\vxml-cvp folder on the RSM server.

Step 2

Copy all the contents from the folder to a directory that can be accessed by the desktop machine hosting Cisco Unified Call Studio software (for example, C:\RSM-Callflow).

Step 3

Launch the Call Studio. Navigate to File> Import> Call Studio>Existing Call Studio Project in the menu bar to import the RSM project into the workspace and click Next.

Step 4

Browse the vxml-cvp folder and click Finish.

Step 5

Navigate to the DoLogin page in the Callflow Editor Navigator pane for RSM Project.

Step 6

Select the SetBaseSessionVars element, and then click Data under Element Configuration.

Step 7

Modify the VoiceXML Variable settings for RSM Project as follows:

Note

Ensure that you click Update after modifying the variable settings below; else, the fields will be set with default values.

VL_VLENGINE_HOSTNAME— Hostname or IP address of the server running on VLEngine service.
VL_VLENGINE_PORT— Port number used by the VLEngine service. Port value is 8080.
VL_PHONESIM_HOSTNAME— Hostname or IP address of the server running on PhoneSim service. Value is same as VL_VLENGINE_HOSTNAME.
VL_PHONESIM_RTSP_PORT— RTSP port number used by PhoneSim service. This is usually 29554.
VL_PHONESIM_HTTP_PORT— HTTP port number used by PhoneSim service. This is usually 29001.
MAX_NUM_LOGIN_ATTEMPTS— Maximum number of failed login attempts allowed before RSM disconnects user.

CVP_MEDIASVR_AUDIO_PATH— Points to the URL path where RSM prompts are uploaded (for example, /en-us/VL).

Note

This path, and the Path component specified in the RSM CVP project's Audio Settings - Default Audio Path URL text field, should be identical.

CVP_MEDIASVR_HOSTNAME— Hostname or IP address of the CVP media server with RSM prompts, as found in the /en-us/VL directory.
MAIN_MENU_TIMEOUT— Time, in seconds. This is usually 12 seconds.
CVP_VXMLSVR_HOSTNAME—Hostname or IP address of the server running on VXML server.
CVP_VXMLSVR_PORT—Port number used by the VXML server.
CVP_MEDIASVR_PORT—Port number of the media server domain. This is usually port 80.
MONITOR_NEWEST_REPOLL_PERIOD—Time, in seconds, before monitoring any new agent conversations. This value is normally set to 4 seconds.
MONITOR_NEWEST_PROMPT_TO_END_EVERYN—Number of pollings before the progress prompt is stated to the caller (that is, “System is still busy. Press any key to return to main menu or continue to hold.”). This value is normally set to 3.
SUPERVISOR_LOGIN_TIMEOUT—Time out for Supervisor Login. This value is normally set to 1500.

Step 8

Click Save to save the RSM Project.

Step 9

Right-click RSM Project in the navigator pane then click Properties.

Step 10

Under Call Studio, click Audio Settings.

Step 11

Navigate to the Default Audio Path URI text field and enter the VL directory on your media server, for example, http://<cvp_media_server_IP_address>/en-us/VL . Click OK.

Step 12

Repeat the above steps to create RSM project for each CVP Server in your deployment.

Note

For small contact center deployment, repeat the steps 1 through 11 for creating unique RSM projects for each sub customer in each CVP server.

Call Flow Deployment

Once the call flow script is installed on the CVP server, it must be deployed for use by CVP VXML Server.

Perform the procedure to deploy the call flow script.

Note

Deploy the VXML script in all the CVP boxes with the appropriate CVP_VXMLSVR_HOSTNAME

Procedure

Step 1	Open Cisco Unified Call Studio.
Step 2	Right-click RSM Project in the navigator pane, then click Deploy.
Step 3	Select Archive File radio button.
Step 4	Browse to the loction where you want to save the VXML Application file.
Step 5	Click Finish. The call flow script will be saved at the specified location.
Step 6	Open the CVP OAMP portal.
Step 7	Navigate to Bulk Administration > File Transfer > VXML application.
Step 8	Choose the desired CVP VXML servers from Available to Selected and browse the VXML application file that you saved in step 4.
Step 9	Click Transfer and click File Transfer Status to check the status.
Step 10	Go to the SCC-CVP-SVR-A server and navigate to the C:\Cisco\CVP\VXMLServer\applications\RSM\admin Directory in CVP call server, then double-click the deployApp.bat file. The batch file is executed in a separate DOS window.
Step 11	Enter Y for yes when prompted to deploy the application. The call flow script is now accessible from the CVP VXML Server.
Step 12	Configure the appropriate micro applications on your VXML gateway (VXML Gateway dial peers, Unified Communication Manager route patterns, and so on) so they can access the script.

Configure Unified CCE

Set the Agent Target Rule

Note

Ignore this procedure, if Unified CCE Day 1 configuration includes the Extension range.

Procedure

Step 1

From the Administration Workstation (AW), click Start > All Programs > Configuration Manager.

Step 2

Expand Tools, then expand List Tools. Double-click Agent Targeting Rule.

Step 3

Click Retrieve to return a list of all existing agent targets in the environment.

Step 4

Highlight the existing agent targeting rule. Under extension ranges click Add. A blank extension range section appears.

Step 5

Add the range of extension, that is the DN and then click OK.

Step 6

Click Save.

Note

In 4000 deployment there are two CUCM PGs configured according to CUCM application user. ATRs are configured based on CUCM PG. Therefore, two ATRs are pre-configured as a part of load base configuration and extensions should be added to ATRs based on the Application user.

Create the Supervisor Login Account

You must create a new account for each supervisor who will be using RSM, according to your current CTI OS supervisor/agent accounts. If CTI OS authentication is used, separate supervisor agent accounts must be created in the Unified CCE environment, to allow dialed-in supervisors to log in to the system.

To create a supervisor login account, follow this procedure Create an Agent

Note

Check the Supervisor check box and ensure that the supervisor password should meet AD Password Policy.
Ensure that the agent password is numeric of any length.
Ensure that the Supervisor is added to Team List.

Create Routing Script for RSM

The following Routing Scripts are used for Cisco RSM.

Figure 2. Routing Scripts used for Cisco RSM

Configure Unified Communication Manager

Configure Simulated Phone

Before You Begin

You must determine the number of simulated phones ( also called as simphones) to assign to each Unified Communication Manager cluster. Each cluster must have a number of simphones greater than or equal to the maximum number of agents that will be simultaneously monitored through RSM for the cluster. This section provides the following information:

Configure the simphone device dependencies
Create the simphone devices

Perform the following steps to add new cluster in RSM:

Create Simphone Device Dependencies

Perform the below steps to create Simphone device dependencies:

Create Simphone Device

Perform the below steps to create Simphone devices:

Add Phones
Disable Built-in Bridge see, Enable or Disable the Built-in-Bridge

Set Up the Login Pool Simphone

The first five simphone devices that are created for each cluster are automatically assigned to the VLEngine login pool. The login pool performs a test login to CTI OS when a caller is authenticated by RSM, to support the VLEngine authentication mechanism. Because CTI OS logins are performed on these simphone devices, they must be associated with the pguser account on each Unified Communications Manager cluster. They must also have Cisco Unified Intelligent Contact Management Enterprise device targets created.

What to Do Next

Create RSM Application User
Associate first five phones to application user. see, Associate Phone to Application User

.

Configure Cisco RSM for 4000 Agent Deployment

Configure the Cisco RSM (Remote Silent Monitoring) Server for 4000 agent deployment in the distributed mode, in the following order:

Required Software	Tasks
Configure RSM	Set RSM Configuration Settings for 4000 and 12000 Agent Deployment
	Configure JTAPI Client Preferences
	Edit Registry Settings
Configure Gateway	Set Up the VXML Gateway
Configure Unified CVP	Upload RSM Prompts
	Integrate the CVP Call Flow
	Call Flow Deployment
Configure Unified CCE	Set the Agent Target Rule
	Create the Supervisor Login Account
	Create Routing Script for RSM
Configure Unified Call Manager	Configure Simulated Phone
	Set Up the Login Pool Simphone
	Create RSM Application User

Set RSM Configuration Settings for 4000 and 12000 Agent Deployment

Set RSM Configuration Settings for 4000 and 12000 Agent Deployment

Procedure

Step 1

Complete the Mail Server configuration settings:

Choose Start > CiscoRSM > RSM Configuration Manager.
Check Send Email Alert check box.
Enter the Host Name/IP address of the mail server in Mail Server Host Name/IP text box.
Enter the email port number in Port text box.
Enter the sender email ID in Sender Email Address text box.
Enter the receiver email ID in Receiver Email Address text box.
Click Next.

Step 2

Complete the Miscellaneous configuration settings:

Enter 1800 in Problem Call Minimum Duration text box.
Enter 4 in Problem Call Min Holds text box.
Enter 3600 in Max Stale Call Duration text box.
Set blank value for CTI OS Trace Mask.
Select INFO from the Log Level drop down list for VL Engine.
Enter 8080 in HTTP Listen Port text box for VL Engine.
Enter 480 in the Audio Buffer Len To VRU text box for PhoneSim.
Note The default value of Audio Buffer Len to VRU is 160, for CVP environment the value is set to 480.
Select INFO from the Log Level drop down list for PhoneSim.
Enter 29001 in HTTP Listen Port text box for PhoneSim.
Enter 29554 in RTSP Listen Port text box for PhoneSim.
Select the RTSP u-law for Audio Encoding from the drop down list for Phonesim.
Select No from the Do HTTP Chunked Transfers drop down list for PhoneSim.
Enter the IP Address of RSM server in the Host Data IP text box.
Click Next.

Step 3

Define first Cluster configuration settings:

These settings are used to configure the Unified Communications Manager cluster with the agents to be monitored by RSM.

Click Add Cluster
Enter a cluster name in ClusterN_Name text box.
Note
Name should be alphanumeric.
Enter 5 in No. of Login Pool Simphones text box.
Enter 60 in No. of Monitoring Phones text box. (this is to monitor 60 concurrent).
Enter 5000 in the Peripheral ID text box.
Enter the rsmuser1 in JTAPI Username text box.
Enter the rsmuser1 password in JTAPI Password text box.
Enter the first MAC address to use for auto-generation of MAC range for simphone device names in Start MAC Range text box.

Enter the first extension number to use for auto-generation of line extension range for simphone DNs in Start Line Num Range text box.

Note

Line extension ranges must not overlap between clusters. Correlates to ClusterN_PhoneSim_StartMACRange value.
The Start Line Num Range should be between 4 to 15 digits.

Select TCP from the SIP Transport drop down list.
Click Next.

Step 4

Define Unified Communications Manager configuration settings for first cluster:

Enter the host name / IP address of CUCM1 server(Subscriber1) in Host Name/IP text box.
Enter CUCM1 port as 5060 in Port text box.
Enter the host name / IP address of CUCM2 server(Subscriber 2) in Host Name/IP text box.
Enter CUCM2 port as 5060 in Port text box.
Click Next.

Step 5

In UCCE Integration page select UCCE integrate with CTI

Enter the host name / IP address of CTI 1A in CTI 1A Host Name/IP.
Enter 42027 in CTI 1A Port text box.
Enter the host name / IP address of CTI 1B in CTI 1B Host Name/IP
Enter 43027 in CTI 1B Port text box.
Click Next.

Step 6

Define second Cluster configuration settings:

These settings are used to configure the Unified Communications Manager cluster with the agents to be monitored by RSM.

Click Add Cluster
Enter a cluster name in ClusterN_Name text box.
Note
Name should be alphanumeric.
Enter 5 in No. of Login Pool Simphones text box.
Enter 60 in No. of Monitoring Phones text box. (this is to monitor 60 concurrent).
Enter 5001 in the Peripheral ID text box.
Enter the rsmuser2 in JTAPI Username text box.
Enter the rsmuser2 password in JTAPI Password text box.
Enter the first MAC address to use for auto-generation of MAC range for simphone device names in Start MAC Range text box.

Enter the first extension number to use for auto-generation of line extension range for simphone DNs in Start Line Num Range text box.

Note

Line extension ranges must not overlap between clusters. Correlates to ClusterN_PhoneSim_StartMACRange value.
The Start Line Num Range should be between 4 to 6 digits.

Select TCP from the SIP Transport drop down list.
Click Next.

Step 7

In UCCE Integration page select UCCE integrate with CTI

Enter the host name / IP address of CG 2A in CTI 1A Host Name/IP.
Enter 42027 in CTI 1A Port text box.
Enter the host name / IP address of CG 2B in CTI 1B Host Name/IP.
Enter 43027 in CTI 1B Port text box.
Click Next.

Step 8

Check Start PhoneSim Service and Start VLEngine Service check boxes.

Step 9

Click Finish.

Note

For 12000 agent deployment model, repeat the steps from step 3 to add new clusters.

Configure Cisco RSM for 12000 Agent Deployment

Configure the Cisco RSM (Remote Silent Monitoring) Server for 12000 agent deployment in the distributed mode, in the following order:

Required Software	Tasks
Configure RSM	Set RSM Configuration Settings for 4000 and 12000 Agent Deployment
	Configure JTAPI Client Preferences
	Edit Registry Settings
Configure Gateway	Set Up the VXML Gateway
Configure Unified CVP	Upload RSM Prompts
	Integrate the CVP Call Flow
	Call Flow Deployment
Configure Unified CCE	Set the Agent Target Rule
	Create the Supervisor Login Account
	Create Routing Script for RSM
Configure Unified Call Manager	Configure Simulated Phone
	Set Up the Login Pool Simphone
	Create RSM Application User

Configure Cisco RSM for Small Contact Center Deployment

Configure the Cisco RSM (Remote Silent Monitoring) Server for Small Contact Center deployment in the distributed mode, in the following order.

Note

Each Sub customer will have Individual RSM configured.

Required Software	Tasks
Configure RSM	Set RSM Configuration Settings for Small Contact Center Deployment
	Configure JTAPI Client Preferences
	Edit Registry Settings
Configure Gateway	Set Up the VXML Gateway
Configure Unified CVP	Upload RSM Prompts
	Integrate the CVP Call Flow
	Call Flow Deployment
Configure Unified CCE	Set the Agent Target Rule
	Create the Supervisor Login Account
	Create Routing Script for RSM
Configure Unified Call Manager	Configure Simulated Phone
	Set Up the Login Pool Simphone
	Create RSM Application User

Set RSM Configuration Settings for Small Contact Center Deployment

Set RSM Configuration Settings for Small Contact Center Deployment

Procedure

Step 1

Complete the Mail Server configuration settings:

Choose Start > CiscoRSM > RSM Configuration Manager.
Check Send Email Alert check box.
Enter the Host Name/IP address of the mail server in Mail Server Host Name/IP text box.
Enter the email port number in Port text box.
Enter the sender email ID in Sender Email Address text box.
Enter the receiver email ID in Receiver Email Address text box.
Click Next.

Step 2

Complete the Miscellaneous configuration settings:

Enter 1800 in Problem Call Minimum Duration text box.
Enter 4 in Problem Call Min Holds text box.
Enter 3600 in Max Stale Call Duration text box.
Set blank value for CTI OS TraceMask.
Select INFO from the Log Level drop down list for VL Engine.
Enter 8080 in HTTP Listen Port text box for VL Engine.
Enter 480 in the Audio Buffer Len To VRU text box for PhoneSim.
Note The default value of Audio Buffer Length to VRU is 160, for CVP environment the value is set to 480.
Select INFO from the Log Level drop down list for PhoneSim.
Enter 29001 in HTTP Listen Port text box for PhoneSim.
Enter 29554 in RTSP Listen Port text box for PhoneSim.
Select the RTSP u-law for Audio Encoding VRU from the drop down list for Phonesim.
Select No from the Do HTTP Chunked Transfers drop down list for PhoneSim.
Enter the IP Address of RSM server in the Host Data IP text box.
Click Next.

Step 3

Define Cluster configuration settings:

These settings are used to configure the Unified Communications Manager cluster with the agents to be monitored by RSM.

Click Add Cluster
Enter a cluster name in ClusterN_Name text box.
Note
Name should be alphanumeric.

N represents the cluster number.
Enter 5 in No. of Login Pool Simphones text box.
Enter 10 in No. of Monitoring Phones text box. (this is to monitor 10 concurrent).
Enter the Peripheral ID of Agent PG in thePeripheral ID text box.
Enter the rsmuser in JTAPI Username text box.
Enter the rsmuser password in JTAPI Password text box.
Enter the first MAC address to use for auto-generation of MAC range for simphone device names in Start MAC Range text box.

Enter the first extension number to use for auto-generation of line extension range for simphone DNs in Start Line Num Range text box.

Note

Line extension ranges must not overlap between clusters. Correlates to ClusterN_PhoneSim_StartMACRange value.
The Start Line Num Range should be between 4 to 15 digits.

Select TCP from the SIP Transport drop down list.
Click Next.

Step 4

Define Unified Communications Manager configuration settings for the cluster:

Enter the host name / IP address of CUCM1 server(Publisher) in Host Name/IP text box.
Enter CUCM1 port as 5060 in Port text box.
Enter the host name / IP address of CUCM2 server(Subscriber 1) in Host Name/IP text box.
Enter CUCM2 port as 5060 in Port text box.
Click Next.

Step 5

Select UCCE Integration with CTI in UCCE Integration window and enter the following:

Enter the host name / IP address of Agent PG 1A in CTI 1A Host Name/IP .
Enter 42027 in CTI 1A Port text box.
Enter the host name / IP address of Agent PG 1B in CTIOS 1B Host Name/IP.
Enter 43027 in CTI 1B Port text box.
Click Next.

Step 6

Check Start PhoneSim Service and Start VLEngine Service check boxes.

Step 7

Click Finish.

Configure RSM
Configure Gateway
Configure Unified CVP
Configure Unified Communications Manager

Configure RSM

For more information about configuring Cisco RSM, see Configure RSM.

Note

Ensure that you select rtsp-alaw for Miscellaneous configuration settings (Step 2-k) in RSM configuration.

Configure Gateway

For more information, see Configure Gateway.

Configure Unified CVP

For more information, see Configure Unified CVP.

Configure Unified Communications Manager

Configure Service Parameters
Configure Regions

Configure Service Parameters

Configure Service Parameters

For more information, see Configure Unified Communication Manager.

Cisco MediaSense

Create Golden Template for Cisco MediaSense
Automated Cloning and OS Customization
Configure Cisco MediaSense

Create Golden Template for Cisco MediaSense
Configure Cisco MediaSense

Create Golden Template for Cisco MediaSense

Follow the below sequence of tasks to create the golden template for Cisco MediaSense. After each task, return to this page to mark the task "done" and continue the sequence:

Sequence	Done?	Tasks	Notes
1		Download cms_11.0_vmv8_v1.0.ova.	See Open Virtualization Format Files.
2		Create the virtual machine from the OVA.	Follow the procedure that is documented in, Create Virtual Machines.
3		Install Cisco MediaSense.	Follow the procedure for installing VOS applications for golden templates. See Install Unified Communications Voice OS based Applications.
4		Convert the virtual machine to a Golden Template.	Follow the procedure Convert the Virtual Machine to a Golden Template.

After creating all the golden templates, you can run the automation process Automated Cloning and OS Customization. After you run the automation process, you can configure Cisco MediaSense on the destination system. See Configure Cisco MediaSense.

Configure Cisco MediaSense

Cisco MediaSense Primary
Cisco MediaSense Secondary
Configure MediaSense Forking

Cisco MediaSense Primary
Cisco MediaSense Secondary
Configure MediaSense Forking

Cisco MediaSense Primary

Configure Cisco MediaSense Primary
Complete Setup for Primary Server
Configure Incoming Call

Configure Cisco MediaSense Primary
Complete Setup for Primary Server
Configure Incoming Call

Configure Cisco MediaSense Primary

Before You Begin

If there is a value in the optional DNS_IP_NIC1 cell of the automation spreadsheet, configure the DNS server by adding the machine in forward and reverse lookup. For more information, see Configure DNS Server.

Procedure

Step 1

Ensure that Connect at Power On is checked for the network adapters and the floppy drive and click OK.

Step 2

Power on the Primary. This begins the installation based on the information in the .flp file. The installation begins automatically and runs with no interaction from you. After an hour or more, a message appears indicating a successful installation.

Step 3

Click the Console tab for the VM. Log in to the publisher machine, using the credentials for the Administration User. The machine opens to the CLI interface.

Step 4

Edit settings and uncheck Connect at Power on for the floppy drive.

Note

During the customization of the publisher/primary, the username and the password are modified as follows. The customer should change the password.

Default Password for OS Administrator: c1sco@123
Application UserName: Administrator
Default Password for Application User: c1sco@123
Sftp password: c1sco@123
IPSec password: c1sco@123

After rebooting, the VM installation is complete with all the parameters provided in the spreadsheet for the VM.

Complete Setup for Primary Server

Follow this procedure to complete the setup for the primary server in any MediaSense deployment:

Procedure

Step 1	After you complete the installation procedure, the system automatically restarts. Sign in to MediaSense Administration for the primary server. (https://<server>:8443/oraadmin) Welcome screen of the MediaSense First Server Setup wizard is displayed.
Step 2	When you are ready to proceed, click Next. The Service Activation screen is displayed.
Step 3	The system internally verifies the IP address of this server and automatically begins enabling the MediaSense feature services in this server. Wait until all the features services show as enabled in the Service Activation window. After all the services are successfully enabled, click Next. After you click Next, the AXL Service Provider screen appears.
Step 4	Enter the AXL service provider (IP address) and the AXL administrator username and password in the respective fields for the Unified CM that should communicate with MediaSense and click Next, the Call Control Service Provider screen appears. The AXL authentication allows you to enter the Unified CM cluster and retrieve the list of Unified CM servers within that cluster. The AXL administrator username may not be same as the Unified CM Administrator username for that cluster. Make sure to add the username for the AXL Administrator to the Standard Unified CM Administrators group and “Standard AXL API Access” roles in Unified CM.
Step 5	Select and move the Unified CM IP address for Call Control Service from Available Call Control Service Providers window to Selected Call Control Service Providers window and click Next.
Step 6	The MediaSense Setup Summary window appears with successfully configured services. Click Done to complete the initial setup for the primary server. When you finish the post-installation process for any MediaSense server, you must access the Unified CM server for your deployment and you will need to configure the SIP trunk, route pattern, route group, route list, recording profile and end user. You have now completed the initial setup of the primary server for MediaSense. Before you install MediaSense on a secondary server or an expansion server, you must configure details for these servers on the primary server. You configure details for these servers using the MediaSense Administration user interface.
Step 7	Login to MediaSense Administration > API User Configuration
Step 8	Select the available Unified CM User and add it to MediaSense API Users list. Using this user you can login to Search and Play.

What to Do Next

Configure Incoming Call.

Configure Incoming Call

Procedure

Step 1	Login to Cisco MediaSense Administration page.
Step 2	Goto Incoming Call Configuration page.
Step 3	Click Add.
Step 4	Enter recording profile number created in CUCM in Address field.
Step 5	Choose Record from Action drop-down list.
Step 6	Click Save.

Cisco MediaSense Secondary

Before You Begin

If there is a value in the optional DNS_IP_NIC1 cell of the automation spreadsheet, configure the DNS server by adding the machine in forward and reverse lookup. See Configure DNS Server.

Procedure

Step 1	Ensure that Connect at Power on is checked for the network adapters and the floppy drive and click OK.
Step 2	Power on the Secondary. This begins the installation based on the information in the .flp file. The installation begins automatically and runs with no interaction from you. After an hour or more, a message appears indicating a successful installation.
Step 3	Click the Console tab for the VM. Log in to the secondary machine, using the credentials for the Administration User. The machine opens to the CLI interface.
Step 4	Right-click the VM and choose Edit settings and uncheck Connect at Power on for the floppy drive. During the customization of the secondary node, the username and the password are modified as follows. The customer should change the password. Default Password for OS Administrator: cisco@123 Application UserName: Administrator Default Password for Application User: cisco@123 Sftp password: c1sco@123 IPSec password: c1sco@123 After rebooting, the VM installation is complete with all the parameters provided in the spreadsheet for theVM.

Add Secondary Node

Procedure

Step 1

Login to the web portal of MediaSense

Step 2

From the System menu on the left, select MediaSense Server Configuration.

Step 3

In the MediaSense Server Configuration screen, click Add MediaSense Server. The Add MediaSense Server screen in the primary node opens.

Step 4

If your installation uses DNS suffixes, enter the hostname of the server that you want to add.

Step 5

If your installation does not use DNS suffixes, enter the IP address of the server that you want to add.

Step 6

Enter the description of the server that you want to add (Optional).

Step 7

Enter the MAC address of the server that you want to add (Optional).

Step 8

Click Save.

Step 9

Click Back to MediaSense Server List.

MediaSense displays a confirmation message. You see the configuration details of the server that you added in the MediaSense Server List.

Note

You can not assign the server type in this web page. You can assign the server type only during the post-installation procedure. Between the time that a new server is added to the MediaSense Server list and until the time that its post-installation is successfully completed, the type for the new server remains unknown.

Configure Cisco MediaSense Secondary

Before You Begin

If there is a value in the optional DNS_IP_NIC1 cell of the automation spreadsheet, configure the DNS server by adding the machine in forward and reverse lookup. See Configure DNS Server.

Procedure

Step 1

In the client computer where the automation tool was run, navigate to C:\GoldenTemplateTool_10\PlatformConfigRepository\MediaSense.

Step 2

Copy the file named MEDIASENSE_SECONDARY_platformConfig.xml.

Step 3

Paste it to another location and rename it to platformConfig.xml.

Step 4

Launch WinImage and select File > New > 1.44 MB and click OK.

Step 5

Drag and drop platformConfig.xmlinto WinImage.

Click Yes at the message asking if you want to inject the file
Select File > Save as and save the file as a Virtual Floppy Image with the filename platformConfig.flp.

Tip

If drag and drop does not work, select Image > Inject.Then browse to the file.

Step 6

Open vSphere infrastructure client and connect to the vCenter. Go to the customer ESXi host where the VMs are deployed.

Step 7

Navigate to the Configuration tab and in the storage section right click on the Datastore and choose Browse Datastore.

Step 8

Create the folder CMS_SEC and upload platformConfig.flp to it.

Step 9

Edit the Virtual Machine settings for the Unified Communications Manager Subscriber VM.

Step 10

On the Hardware tab, click the floppy drive, choose the radio button Use The Existing Floppy Image in Datastore and mount the platformConfig.flp from the CMS_SEC folder on the data store.

Step 11

Ensure that Connect at Power On is checked for the network adapters and the floppy drive. Click OK and then power on the VM. This begins the installation and customizes the installation based on the information in the .flp file.

Step 12

If there is a value in the optional DNS_IP_NIC1 cell of the automation spreadsheet, configure the DNS server by adding the machine in forward and reverse lookup.

Note

During the customization of the subscriber node, the username and the password are modified as follows. The customer should change the password.

Step 13

After you complete the installation uncheck Connect at Power on for the floppy drive.

Note

During the customization of the publisher/primary, the username and the password are modified as follows. The customer should change the password.

After rebooting, the VM installation is complete with all the parameters provided in the spreadsheet for the VM.

Complete Setup for Secondary Server

Follow this procedure to complete the setup for the secondary server in any MediaSense deployment:

Procedure

Step 1

After you complete the installation procedure of the previous section, the system restarts automatically and you must sign in to MediaSense Administration for secondary servers. When you sign in, the Welcome screen of the MediaSense Secondary Server Setup wizard appears.

Step 2

When you are ready to proceed, click Next.

You determine the type of server in this Welcome screen.

Select the server type Secondary and click Next. The Service Activation screen is displayed.

Step 3

After the services are enabled, click Finish to complete the initial setup for a subsequent server.

The MediaSense Setup Summary window displays the result of the initial setup and MediaSense restarts.

You have now completed the initial setup of a subsequent server. This subsequent server is ready to record.

Repeat this setup procedure for each expansion server in the cluster.

Configure MediaSense Forking

Provisioning Cisco Unified CM for Cisco MediaSense BIB Forking
Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking
Provisioning TDM Gateway for Media Forking

Provisioning Cisco Unified CM for Cisco MediaSense BIB Forking
Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking
Provisioning TDM Gateway for Media Forking

Provisioning Cisco Unified CM for Cisco MediaSense BIB Forking

Sequence	Task	Done?
1	Set Up SIP Options
2	Add SIP Trunks
3	Add Route Pattern
4	Set up Recording Profile
5	Configure Device
6	Disable iLBC, iSAC and g.722 for Recording Device
7	Configure End User

Configure Device
Configure End User

Configure Device

Procedure

Step 1

Login to Cisco Unified Communication Domain Manager as provider.

Step 2

Ensure that hierarchy is set to appropriate customer level.

Step 3

Navigate to Subscriber Management > Phones.

Step 4

Choose the phone from the list that you want to configure.

Step 5

Choose ON from Built-in Bridge drop-down list to enable Built-in Bridge.

Step 6

In Lines tab, choose Automatic Call Recording Enabled from Recording Flag drop-down list.

Step 7

Enter Recording Profile Name.

Note

Enter the exact name of recording profile created in CUCM.

Step 8

Click Save.

Configure End User

Procedure

Step 1	Login to Cisco Unified Communication Domain Manager as provider, reseller or customer admin.
Step 2	Ensure that hierarchy is set to appropriate customer/site.
Step 3	Navigate to Subscriber Management > Subscribers.
Step 4	Click Add.
Step 5	Enter unique Userid and Last Name, in User tab.
Step 6	Enter Password and Repeat Password.
Step 7	Click Save.

Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking

Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking for HCS Deployment Models
Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking for SCC Deployment Models

Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking for HCS Deployment Models
Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking for SCC Deployment Models

Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking for HCS Deployment Models

Sequence	Task	Done?
1	Setup Global Level
2	Dial-Peer Level Setup
3	Set Up CUBE Dial-Peers for MediaSense Deployments

Setup Global Level
Dial-Peer Level Setup
Set Up CUBE Dial-Peers for MediaSense Deployments

Setup Global Level

Procedure

Step 1

Connect to your CUBE gateway using SSH or Telnet.

Step 2

Enter the global configuration mode.

cube# configure terminal
Enter configuration commands, one per line. End with CNTL/Z.
cube(config)#

Step 3

Enter VoIP voice-service configuration mode.

cube(config)# voice service voip
cube(config-voi-serv)#

Step 4

Calls may be rejected with a 403 Forbidden response if Toll Fraud security is not configured correctly. The solution is to add the IP address as a trusted endpoint, or else disable the IP address trusted list authentication altogether using the following configuration entry:

cube(config-voi-serv)# no ip address trusted authenticate

Step 5

Enable CUBE and CUBE Redundancy.

cube(config-voi-serv)# mode border-element
cube(config-voi-serv)# allow-connections sip to sip
cube(config-voi-serv)# sip
cube(config-voi-serv)# asymmetric payload full
cube(config-voi-serv)# video screening

In the example above, the final 3 lines are only required if video calls are to be passed through CUBE.

Step 6

At this point, you will need to save the CUBE configuration and reboot CUBE.

Caution

Be sure to reboot CUBE during off-peak hours.

Save your CUBE configuration.
```
cube# copy run start
```
Reboot CUBE.
```
cube# reload
```

Step 7

After you reboot CUBE, configure the media class to determine which calls should be recorded.

cube(config-voi-serv)# media class 3
cube(config-voi-serv)# recorder parameter
cube(config-voi-serv)# media-recording 3000

Step 8

Exit the VoIP voice-service configuration mode.

cube(config-voi-serv)# exit

Step 9

Create one voice codec class to include five codecs (including one for video). These codecs will be used by the inbound and outbound dial-peers to specify the voice class.

cube(config)# voice class codec 3
cube(config)# codec preference 1 mp4a-latm
cube(config)# codec preference 2 g711ulaw
cube(config)# codec preference 3 g722-64
cube(config)# codec preference 4 g729br8
cube(config)# video codec h264

In the example above, the first codec preference and video codec definition are only required if AAC-LD/LATM media is part of the customer's call flow.

Step 10

To simplify debugging, you must synchronize the local time in CUBE with the local time in Cisco MediaSense servers. For example, if you specify the NTP server as 10.10.10.5, then use the following command in CUBE:

cube(config)# ntp update-calendar
cube(config)# sntp server 10.10.10.5

Dial-Peer Level Setup

Note

This information describes a sample configuration. CUBE may be deployed in multiple ways.

Each Cisco MediaSense deployment for CUBE contains three dial-peers:

Inbound dial-peer: In this example, the unique name is 1000
Outbound dial-peer: In this example, the unique name is 2000
Forking dial-peer: In this example, the unique name is 3000

Before you begin this procedure, obtain the details for these three dial-peers from your CUBE administrator.

Note

The order in which you configure these three dial-peers is not important.

Set Up CUBE Dial-Peers for MediaSense Deployments

This procedure provides an example of how to set up the three dial peers. The specific names and values used are for illustrative purposes only.

Caution

This procedure is not a substitute for the actual CUBE documentation. It is a tutorial to provide detailed information about configuring CUBE for MediaSense. See your CUBE documentation at http://www.cisco.com/go/cube for the latest information.

Procedure

Step 1

Configure media forking on an inbound dial peer.

Assign a unique name to the inbound dial-peer. In this example, the name is set to '1000'.
```
cube(config)# dial-peer voice 1000 voip
```
The command places you in the dial-peer configuration mode to configure a VoIP dial-peer named '1000'.
Specify the session protocol for this inbound dial-peer as 'sipv2' (this value is not optional).
```
cube(config-dial-peer)# session protocol sipv2
```
This command determines if the SIP session protocol on the endpoint is up and available to handle calls. The session protocols and VoIP layers depend on the IP layer to give the best local address and use the address as a source address in signaling or media or both—even if multiple interfaces can support a route to the destination address.
Specify the SIP invite URL for the incoming call. In this example, we assume that inbound, recordable calls will have six digits. Here, we assign the first three digits as '123' and the last three digits are arbitrarily chosen by the caller (as part of the destination DN being dialed). This command associates the incoming call with a dial-peer.
```
cube(config-dial-peer)# incoming called-number 123...$
```
When using multiple codecs, you must create a voice class in which you define a selection order for codecs; then, you can apply the voice class to apply the class to individual dial-peers. In this example, the tag used is '1'.
```
cube(config-dial-peer)# voice-class codec 1
```
This tag uniquely identifies this codec. The range is 1 to 10000.
If call is transferred, be sure to propagate the metadata to MediaSense.You can do so by enabling the translation to PAI headers in the outgoing header on this dial-peer.
```
cube(config-dial-peer)# voice-class sip asserted-id pai
```
Specify that everything that is going through the inbound dial-peer can be forked. Use the same number that you used to set up global forking (see Set up Global Level). In this example, the number media class is '3'.
```
cube(config-dial-peer)# media-class 3
```
Exit the configuration of this inbound dial-peer.
```
cube(config-dial-peer)# exit
cube(config)#
```

Step 2

Configure the outbound dial-peer.

Assign a unique name to the outbound dial-peer. In this example, the name is set to '2000'.
```
cube(config)# dial-peer voice 2000 voip
```
The command places you in the dial-peer configuration mode to configure a VoIP dial-peer named '2000'.
Specify the session protocol for this outbound dial-peer as 'sipv2' (this value is not optional).
```
cube(config-dial-peer)# session protocol sipv2
```
Specify the destination corresponding to the incoming called number. In this example, it is '123...'.
```
cube(config-dial-peer)# destination-pattern 123...$
```
When using multiple codecs, you must create a voice class in which you define a selection order for codecs; then, you can apply the voice class to apply the class to individual dial-peers. Use the same tag used for the inbound dial-peer. In this example, the tag used is '1'.
```
cube(config-dial-peer)# voice-class codec 1
```
Specify the primary destination for this call. In this example, we set the destination to 'ipv4:10.1.1.10:5060'.
```
cube(config-dial-peer)# session target ipv4:10.1.1.10:5060
```
Exit the configuration of this outbound dial-peer.
```
cube(config-dial-peer)# exit
cube(config)#
```

Step 3

Configure the forking dial-peer.

Assign a unique name to the forking dial-peer. In this example, the name is set to '3000'.
```
cube(config)# dial-peer voice 3000 voip
```
The command places you in the dial-peer configuration mode to configure a VoIP dial-peer named '3000'. Optionally, provide a description for what this dial-peer does using an arbitrary English phrase.
```
cube(config-dial-peer)# description This is the forking dial-peer
```
Specify the session protocol for this forking dial-peer as 'sipv2' (this value is not optional).
```
cube(config-dial-peer)# session protocol sipv2
```
Specify an arbitrary destination pattern with no wildcards. Calls recorded from this CUBE will appear to come from this extension. (In the MediaSense Incoming Call Configuration table, this number corresponds to the address field.) In this example, we set it to '3000'.
```
cube(config-dial-peer)# destination-pattern 3000
```
When using multiple codecs, you must create a voice class in which you define a selection order for codecs; then, you can apply the voice class to apply the class to individual dial-peers. Use the same tag used for the inbound dial-peer. In this example, it is '1'.
```
cube(config-dial-peer)# voice-class codec 1
```

Provide the IP address of one of the MediaSense expansion servers (if available) as a destination for the CUBE traffic. In this example, we use a MediaSense server at IP address 10.2.2.20.

Note

Avoid using the primary or secondary MediaSense servers for this step as these servers carry the CUBE load and it is best to avoid adding load to the database servers.
In Small contact center deployment model , provide the signaling IP address of CUBE(E) adjacency configured in the Perimeta SBC for Mediasense forking. See Add CUBE-MEDIASENSE FORK adjacency

cube(config-dial-peer)# session target ipv4:10.2.2.20:5060

Set the session transport type (UDP or TCP) to communicate with MediaSense. The default is UDP. The transport protocol specified with the session transport command, and the protocol specified with the transport command, must be identical.
```
cube(config-dial-peer)# session transport tcp
```

Configure a heartbeat mechanism to monitor connectivity between end points. A generic heartbeat mechanism allows Cisco Unified Border Element to monitor the status of MediaSense servers or endpoints and provide the option of timing-out a dial-peer if it encounters a heartbeat failure.

Note

If you have configured an alternate dial-peer for the same destination pattern, the call fails over to the next preferred dial-peer. Otherwise, the call is rejected. If you have not configured a fail over dial-peer, then do not configure options-keep alive.

cube(config-dial-peer)# voice-class sip options-keepalive

Prevent CUBE from sending multi part body in INVITE to MediaSense.
```
cube(config-dial-peer)# signaling forward none
```
Exit the configuration of this forking dial-peer.
```
cube(config-dial-peer)# exit
cube(config)#
```
Exit the configuration mode.
```
cube(config)# exit
cube#
```
Save your CUBE configuration.
```
cube# copy run start
```

Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking for SCC Deployment Models

Sequence	Task	Done?
1	Setup Global Level
2	Dial-Peer Level Setup
3	Set Up CUBE Dial-Peers for Small Contact Center Deployment

Set Up CUBE Dial-Peers for Small Contact Center Deployment

Set Up CUBE Dial-Peers for Small Contact Center Deployment

The inbound dialpeer for MediaSense should be created for each sub customer . Follow the below steps to create the inbound dialpeer:

Procedure

Step 1

Configure media forking on an inbound dial peer.

Assign a unique name to the inbound dial-peer. In this example, the name is set to '1000'.
```
cube(config)# dial-peer voice 1000 voip
```
The command places you in the dial-peer configuration mode to configure a VoIP dial-peer named '1000'.
Specify the session protocol for this inbound dial-peer as 'sipv2' (this value is not optional).
```
cube(config-dial-peer)# session protocol sipv2
```
This command determines if the SIP session protocol on the endpoint is up and available to handle calls. The session protocols and VoIP layers depend on the IP layer to give the best local address and use the address as a source address in signaling or media or both—even if multiple interfaces can support a route to the destination address.
Specify the SIP invite URL for the incoming call. In this example, we assume that inbound, recordable calls will have six digits. Here, we assign the first three digits as '123' and the last three digits are arbitrarily chosen by the caller (as part of the destination DN being dialed). This command associates the incoming call with a dial-peer.
```
cube(config-dial-peer)# incoming called-number 123...$
```
When using multiple codecs, you must create a voice class in which you define a selection order for codecs; then, you can apply the voice class to apply the class to individual dial-peers. In this example, the tag used is '1'.
```
cube(config-dial-peer)# voice-class codec 1
```
This tag uniquely identifies this codec. The range is 1 to 10000.
If call is transferred, be sure to propagate the metadata to MediaSense. You can do so by enabling the translation to PAI headers in the outgoing header on this dial-peer.
```
cube(config-dial-peer)# voice-class sip asserted-id pai
```
Specify that everything that is going through the inbound dial-peer can be forked. Use the same number that you used to set up global forking (see Set up Global Level). In this example, the number media class is '3'.
```
cube(config-dial-peer)# media-class 3
```
Exit the configuration of this inbound dial-peer.
```
cube(config-dial-peer)# exit
cube(config)#
```

Step 2

Configure the forking dial-peer.

Assign a unique name to the forking dial-peer. In this example, the name is set to '3000'.
```
cube(config)# dial-peer voice 3000 voip
```
The command places you in the dial-peer configuration mode to configure a VoIP dial-peer named '3000'. Optionally, provide a description for what this dial-peer does using an arbitrary English phrase.
```
cube(config-dial-peer)# description This is the forking dial-peer
```
Specify the session protocol for this forking dial-peer as 'sipv2' (this value is not optional).
```
cube(config-dial-peer)# session protocol sipv2
```
Specify an arbitrary destination pattern with no wildcards. Calls recorded from this CUBE will appear to come from this extension. (In the MediaSense Incoming Call Configuration table, this number corresponds to the address field.) In this example, we set it to '3000'.
```
cube(config-dial-peer)# destination-pattern 3000
```
When using multiple codecs, you must create a voice class in which you define a selection order for codecs; then, you can apply the voice class to apply the class to individual dial-peers. Use the same tag used for the inbound dial-peer. In this example, it is '1'.
```
cube(config-dial-peer)# voice-class codec 1
```

Provide the IP address of one of the MediaSense expansion servers (if available) as a destination for the CUBE traffic. In this example, we use a MediaSense server at IP address 10.2.2.20.

Note

Avoid using the primary or secondary MediaSense servers for this step as these servers carry the CUBE load and it is best to avoid adding load to the database servers.
In Small contact center deployment model , provide the signaling IP address of CUBE(E) adjacency configured in the Perimeta SBC for Mediasense forking. See Add CUBE-MEDIASENSE FORK adjacency

cube(config-dial-peer)# session target ipv4:10.2.2.20:5060

Set the session transport type (UDP or TCP) to communicate with MediaSense. The default is UDP. The transport protocol specified with the session transport command, and the protocol specified with the transport command, must be identical.
```
cube(config-dial-peer)# session transport tcp
```

Configure a heartbeat mechanism to monitor connectivity between end points. A generic heartbeat mechanism allows Cisco Unified Border Element to monitor the status of MediaSense servers or endpoints and provide the option of timing-out a dial-peer if it encounters a heartbeat failure.

Note

If you have configured an alternate dial-peer for the same destination pattern, the call fails over to the next preferred dial-peer. Otherwise, the call is rejected. If you have not configured a fail over dial-peer, then do not configure options-keep alive.

cube(config-dial-peer)# voice-class sip options-keepalive

Prevent CUBE from sending multi part body in INVITE to MediaSense.
```
cube(config-dial-peer)# signaling forward none
```
Exit the configuration of this forking dial-peer.
```
cube(config-dial-peer)# exit
cube(config)#
```
Exit the configuration mode.
```
cube(config)# exit
cube#
```
Save your CUBE configuration.
```
cube# copy run start
```

Provisioning TDM Gateway for Media Forking

The following section will provide detailed guidelines on how to configure media recording for calls on TDM trunks. CUBE (E), being an integrated platform can provide TDM trunk connectivity and act as a session border controller at the same time.

For this solution to work, calls from the PSTN are looped back to itself thus creating an inbound VoIP SIP leg to the CUBE. It then sends the call to the call agent in the enterprise network creating an outbound VoIP SIP leg. Thus, the gateway is used to terminate the TDM leg, and originate an IP leg towards the call agent.

Note

In this flow ,calls will effectively halve the stated capacity of the router, thus requiring twice as much router capacity for the same number of calls. If you intended to use the full capacity of the router for calls, you will need two routers. It is better to configure two routers for their individual purposes, rather than using both routers for both purposes.

Follow the below sequence to configure TDM gateway

Sequence	Task	Done?
1	Configure translation rule and profile
2	Configure loopback interface
3	Configure media class
4	Configure dial-peers

Procedure

Step 1	Configure translation rule and profile Configure translation rules for the calling number (ANI) or called number (DNIS) digits for a voice call voice translation-rule 1 rule 1 /^966//8966/ voice translation-rule 2 rule 2 /^8966//966/ The first rule defined is rule 1, in which 966 is the pattern that must be matched and replaced, and 8966 is the pattern that is substituted for 966. Configure translation-profile The translation rules replace a sub string of the input number if the number matches the match pattern, number plan, and type present in the rule voice translation-profile prefix translate called 1 voice translation-profile strip translate called 2 Translation profile prefix will add prefix to the called number based on the translation rule 1.Similarly translation profile strip will remove the prefix from the called number based on the translation rule 2.
Step 2	Configure loopback interface interface Loopback0 ip address 1.1.1.1 255.255.255.255
Step 3	Configure media class Configure the media class to determine which calls should be recorded. cube(config-voi-serv)# media class 3 cube(config-voi-serv)# recorder parameter cube(config-voi-serv)# media-recording 20
Step 4	Configure dial-peers Configure the pots dial-peer for incoming PSTN call dial-peer voice 1 pots description Incoming dial peer for PSTN calls translation-profile incoming prefix incoming called-number 9660000001 port 0/2/1:23 Configure the voip dial-peer to loop back the incoming PSTN call. dial-peer voice 8966 voip description To loop incoming PSTN calls back to itself. destination-pattern 89660000001 session protocol sipv2 session target ipv4:1.1.1.1 # loop back Ip address of the TDM gateway session transport tcp voice-class codec 1 dtmf-relay rtp-nte no vad Configure the inbound dial-peer for newly originated SIP call leg dial-peer voice 89660 voip description inbound dial-peer for the newly originated SIP call leg translation-profile incoming strip session protocol sipv2 session target sip-server session transport tcp incoming called-number 89660000001 voice-class codec 1 dtmf-relay rtp-nte no vad Configure the outbound dial-peer for newly originated SIP call dial-peer voice 9660 voip description Outgoing dial peer for looped call to contact center destination-pattern 9660000001 session protocol sipv2 session target ipv4:192.1.10.1 #IP address of CVP server session transport tcp voice-class codec 1 dtmf-relay rtp-nte media-class 3 no vad Configure the outbound dial-peer for forking call leg dial-peer voice 20 voip description Forking leg to MediaSense server. preference 1 destination-pattern 99999 signaling forward none session protocol sipv2 session target ipv4:192.1.9.1 #Ip address of MediaSense server session transport tcp voice-class sip options-keepalive

Cisco Unified SIP Proxy

Install Cisco Unified SIP Proxy
Configure Cisco Unified SIP Proxy Server
Configure Outbound with Cisco Unified SIP Proxy

Install Cisco Unified SIP Proxy
Configure Cisco Unified SIP Proxy Server
Configure Outbound with Cisco Unified SIP Proxy

Install Cisco Unified SIP Proxy

Installation of CUSP
Post Installation Configuration Tool
Obtaining New or Additional Licenses

Installation of CUSP
Post Installation Configuration Tool
Obtaining New or Additional Licenses

Installation of CUSP

Procedure

Step 1	Download all Cisco Unified SIP Proxy 8.5.7 software files.
Step 2	Copy the files to the FTP server.
Step 3	Starting from router EXEC mode, enter the following: `ping <ftp_server_ip_address>`
Step 4	Enter the following and Install the software: `Service-Module 1/0 install url ftp://<ftp_server_ip_address>/cusp-k9.sme.8.5.7.pkg`
Step 5	Enter `Y` to confirm installation.
Step 6	Enter Cisco Unified SIP Proxy Service Module to monitor and complete the installation.

Example of Installation on a Service Module

CUSP#service-nodule SM4/0 inst
CUSP#$ule SM4/0 install url ftp://10.10.10.203/cusp-k9.snc.8.5.7.pkg
Delete the installed Cisco Unified SIP Proxy and proceed with new installation?
[no]:yes
Loading cusp-k9.snc.8.5.7.pkg.install.src !
[OK – 1850/4096 bytes]
cur_cpu: 1862
cur_disk: 953880
cur_nem: 4113488
cur_pkg_name: cusp-k9.sne.8.5.7.pkg
cur_ios_version: 15.2<4>M5,
cur_image_name:c3900e-universalk9-mz
cur_pid: SM-SRE-900-K9
bl_str:
inst_str:
app_str:
key_filename: cusp-k9.sne.8.5.7.key
helper_filename:cusp-helper.sme.8.5.7
Resource check passed…

Post Installation Configuration Tool

Run the command: CUSP#service-module SM 4/0 session to open the first session.

When you open the first session, the system launches the post installation configuration tool, and asks you if you want to start configuration immediately.

Enter the appropriate response, y or n. If you enter n, the system will halt. If you enter "y", the system will ask you to confirm, then begin the interactive post installation configuration process.

The following is an example:

IMPORTANT::
IMPORTANT:: Welcome to Cisco Systems Service Engine
IMPORTANT:: post installation configuration tool.
IMPORTANT::
IMPORTANT:: This is a one time process which will guide
IMPORTANT:: you through initial setup of your Service Engine.
IMPORTANT:: Once run, this process will have configured
IMPORTANT:: the system for your location.
IMPORTANT::
IMPORTANT:: If you do not wish to continue, the system will be halted
IMPORTANT:: so it can be safely removed from the router.
IMPORTANT::

Do you wish to start configuration now (y,n)? yes
Are you sure (y,n)? yes

IMPORTANT::
IMPORTANT:: A configuration has been found in flash. You can choose
IMPORTANT:: to restore this configuration into the current image.
IMPORTANT::
IMPORTANT:: A stored configuration contains some of the data from a
IMPORTANT:: previous installation, but not as much as a backup.
IMPORTANT::
IMPORTANT:: If you are recovering from a disaster and do not have a
IMPORTANT:: backup, you can restore the saved configuration.
IMPORTANT::
IMPORTANT:: If you choose not to restore the saved configuration, it
IMPORTANT:: will be erased from flash.
IMPORTANT::

Would you like to restore the saved configuration? (y,n) n

Erasing old configuration...done.

IMPORTANT::
IMPORTANT:: The old configuration has been erased.
IMPORTANT:: As soon as you finish configuring the system please use the
IMPORTANT:: "write memory" command to save the new configuration to flash.
IMPORTANT::

Enter Hostname
(my-hostname, or enter to use se-10-50-30-125):
Using se-10-50-30-125 as default

Enter Domain Name
(mydomain.com, or enter to use localdomain): cusp

IMPORTANT:: DNS Configuration:
IMPORTANT::
IMPORTANT:: This allows the entry of hostnames, for example foo.cisco.com, instead
IMPORTANT:: of IP addresses like 1.100.10.205 for application configuration. In order
IMPORTANT:: to set up DNS you must know the IP address of at least one of your
IMPORTANT:: DNS Servers.

Would you like to use DNS (y,n)?y

Enter IP Address of the Primary DNS Server
(IP address): 180.180.180.50
Found server 180.180.180.50

Enter IP Address of the Secondary DNS Server (other than Primary)
(IP address, or enter to bypass):

E

Enter Fully Qualified Domain Name(FQDN: e.g. myhost.mydomain.com)
or IP address of the Primary NTP server
(FQDN or IP address, or enter for 10.50.30.1): 10.50.10.1
Found server 10.50.10.1

Enter Fully Qualified Domain Name(FQDN: e.g. myhost.mydomain.com)
or IP address of the Secondary NTP Server
(FQDN or IP address, or enter to bypass):

Please identify a location so that time zone rules can be set correctly.
Please select a continent or ocean.
1) Africa 4) Arctic Ocean 7) Australia 10) Pacific Ocean
2) Americas 5) Asia 8) Europe
3) Antarctica 6) Atlantic Ocean 9) Indian Ocean
#? 2
Please select a country.
1) Anguilla 27) Honduras
2) Antigua & Barbuda 28) Jamaica
3) Argentina 29) Martinique
4) Aruba 30) Mexico
5) Bahamas 31) Montserrat
6) Barbados 32) Netherlands Antilles
7) Belize 33) Nicaragua
8) Bolivia 34) Panama
9) Brazil 35) Paraguay
10) Canada 36) Peru
11) Cayman Islands 37) Puerto Rico
12) Chile 38) St Barthelemy
13) Colombia 39) St Kitts & Nevis
14) Costa Rica 40) St Lucia
15) Cuba 41) St Martin (French part)
16) Dominica 42) St Pierre & Miquelon
17) Dominican Republic 43) St Vincent
18) Ecuador 44) Suriname
19) El Salvador 45) Trinidad & Tobago
20) French Guiana 46) Turks & Caicos Is
21) Greenland 47) United States
22) Grenada 48) Uruguay
23) Guadeloupe 49) Venezuela
24) Guatemala 50) Virgin Islands (UK)
25) Guyana 51) Virgin Islands (US)
26) Haiti
#? 47
Please select one of the following time zone regions.
1) Eastern Time
2) Eastern Time - Michigan - most locations
3) Eastern Time - Kentucky - Louisville area
4) Eastern Time - Kentucky - Wayne County
5) Eastern Time - Indiana - most locations
6) Eastern Time - Indiana - Daviess, Dubois, Knox & Martin Counties
7) Eastern Time - Indiana - Pulaski County
8) Eastern Time - Indiana - Crawford County
9) Eastern Time - Indiana - Pike County
10) Eastern Time - Indiana - Switzerland County
11) Central Time
12) Central Time - Indiana - Perry County
13) Central Time - Indiana - Starke County
14) Central Time - Michigan - Dickinson, Gogebic, Iron & Menominee Counties
15) Central Time - North Dakota - Oliver County
16) Central Time - North Dakota - Morton County (except Mandan area)
17) Mountain Time
18) Mountain Time - south Idaho & east Oregon
19) Mountain Time - Navajo
20) Mountain Standard Time - Arizona
21) Pacific Time
22) Alaska Time
23) Alaska Time - Alaska panhandle
24) Alaska Time - Alaska panhandle neck
25) Alaska Time - west Alaska
26) Aleutian Islands
27) Hawaii
#? 21

The following information has been given:
United States
Pacific Time

Therefore TZ='America/Los_Angeles' will be used.
Is the above information OK?
1) Yes
2) No
#? 1

Local time is now: Mon Apr 5 11:20:17 PDT 2010.
Universal Time is now: Mon Apr 5 18:20:17 UTC 2010.
executing app post_install
executing app post_install done
Configuring the system. Please wait...
Changing owners and file permissions.
Tightening file permissions ...
Change owners and permissions complete.
Creating Postgres database .... done.
INIT: Switching to runlevel: 4
INIT: Sending processes the TERM signal
==> Starting CDP
STARTED: cli_server.sh
STARTED: ntp_startup.sh
STARTED: LDAP_startup.sh
STARTED: SQL_startup.sh
STARTED: dwnldr_startup.sh
STARTED: HTTP_startup.sh
STARTED: probe
STARTED: fndn_udins_wrapper
STARTED: superthread_startup.sh
STARTED: /bin/products/umg/umg_startup.sh

Waiting 49 ...

IMPORTANT::
IMPORTANT:: Administrator Account Creation
IMPORTANT::
IMPORTANT:: Create an administrator account.
IMPORTANT:: With this account, you can log in to the
IMPORTANT:: Cisco Unified SIP Proxy
IMPORTANT:: GUI and run the initialization wizard.

IMPORTANT::

Enter administrator user ID:
(user ID): test
tesEnter password for test:
(password):
Confirm password for test by reentering it:
(password):

SYSTEM ONLINE
cusp-sre-49# show software version
Cisco Unified SIP Proxy version <8.5.7>
Technical Support: http://www.cisco.com/techsupport Copyright <c> 1986-2008 by Cisco Systems,Inc.
Cusp-src-49# show software packages

Installed Packages:
-	Installer <Installer application >  <8.5.7.0>
-	Infrastructure <Service Engine Infrastructure> <8.5.7>
-	Global <Global manifest > <8.5.7>
-	Bootloader <Secondary> <Service Engine Bootloader> <2.1.30>
-	Core <Service Engine OS Core > <8.5.7>
-	GPL Infrastrucutre <Service Engine GPL Infrastructure > <8.5.7>

Obtaining New or Additional Licenses

Required Information
Using the CLI to Install the Cisco Unified SIP Proxy Release 8.5.7 Licenses
Using the Licensing Portal to Obtain Licenses for Additional Features or Applications

Required Information
Using the Licensing Portal to Obtain Licenses for Additional Features or Applications
Using the CLI to Install the Cisco Unified SIP Proxy Release 8.5.7 Licenses

Required Information

Collect the following information before you obtain new or additional CSL licenses:

The SKU for the features that you need. The SKU is used in the ordering process to specify the desired licenses for the Cisco Unified SIP Proxy features that you want.
The Product ID (PID) and the Serial Number (SN) from the device. Together, these form the unique device identifier (UDI). The UDI is printed on a label located on the back of most Cisco hardware devices or on a label tray visible on the front panel of field-replaceable motherboards. The UDI can also be viewed via software using the show license udi command in privileged EXEC mode.

Using the Licensing Portal to Obtain Licenses for Additional Features or Applications

Note

You must have a Cisco.com password to access some of the URLs in the following procedure.

Follow these steps to obtain additional licenses for Cisco Unified SIP Proxy Release 8.5.7 features.

Procedure

Step 1	Go to http://www.cisco.com/web/ordering/root/index.html and choose one of the ordering processes (through partner, Cisco direct, etc.) and order licenses. When you purchase a license, you will receive a product activation key (PAK), which is an alphanumeric string that represents the purchase.
Step 2	To get your license file, return to the Cisco Product License Registration Portal at http://www.cisco.com/web/ordering/root/index.html. When prompted, and enter the PAK and the unique device identifier (UDI) of the device where the license will be installed.
Step 3	Download the license file or receive the license file by email.
Step 4	Copy the license file(s) to a FTP or TFTP server.

Using the CLI to Install the Cisco Unified SIP Proxy Release 8.5.7 Licenses

Follow these steps to install the licenses for Cisco Unified SIP Proxy

Procedure

Step 1

Login to the CLI.

Step 2

Enter license install <URL>, where <URL> is the FTP URL that you copied the license in the previous procedure.

Step 3

Verify the license by entering either show license or show software licenses.

Step 4

Activate the new license by entering license activate.

Step 5

Reload the module by entering reload and confirming that you really want to reload the module.

Note

You cannot remove evaluation licenses.

Configure Cisco Unified SIP Proxy Server

Login to CUSP portal http://<cusp module IP>/admin/Common/HomePage.do and configure the Cisco Unified SIP Proxy server, in the following order:

Required Software	Tasks
Configure CUSP	Configure Cisco Unified SIP Proxy
Configure Gateway	Configure Gateway
Configure Unified CVP	Configure Unified CVP
Configure Unified Call Manager though UCDM	Configure Cisco Unified Communications Manager

Configure Cisco Unified SIP Proxy
Configure Gateway
Configure Unified CVP
Configure Cisco Unified Communications Manager

Configure Cisco Unified SIP Proxy

Perform the following procedures to configure Unified SIP Proxy

Sequence	Done?	Tasks	Notes
1		Configure Networks
2		Configure Triggers
3		Configure Server Groups
4		Configure Route Tables
5		Configure Route Policies
6		Configure Route Triggers

For complete configuration details of Cisco Unified SIP Proxy, see Full Configuration for Cisco Unified SIP Proxy

Table 1 Example CUSP Deployment Details
Server Name	IP Address	FQDN
CUSP	10.10.10.49	cusp.hcsdc1.icm
CVP	10.10.10.10	cvp.hcsdc1.icm
CUCM	10.10.10.30	ccm.hcsdc1.icm
Gateway	10.10.10.180	gw.hcsdc1.icm

Configure Networks
Configure Triggers
Configure Server Groups
Configure Route Tables
Configure Route Policies
Configure Route Triggers
Full Configuration for Cisco Unified SIP Proxy

Configure Networks

Procedure

Step 1	Login to CUSP portal.
Step 2	Navigate to Configure > Networks and click Add.
Step 3	Enter a unique name for the Network. Example:hcs
Step 4	Choose Standard from the TYPE drop-down list.
Step 5	Enable the Allow Outbound Connections.
Step 6	Click Add on the SIP Listen Points tab.
Step 7	Choose newly added Network and select SIP Listen Points tab.
Step 8	Select the IP address of the CUSP, from the IP address drop-down list, See Table 1.
Step 9	Keep the default port 5060.
Step 10	Select the Transport Type as TCP and click Add.
Step 11	Repeat the step 6 to step 8, select Transport Type as UDP and click Add.
Step 12	Disable SIP Record-Route, select and disable all the networks for the CVP that includes callflows.

Configure Triggers

Procedure

Step 1	Login to CUSP Portal.
Step 2	Navigate to Configure > Triggers and click Add.
Step 3	Enter a name for the Trigger and click Add. Example:hcs trigger in
Step 4	Choose the appropriate Trigger conditions from the drop-down lists. Example: Inbound Network, Is exactly, and hcs
Step 5	Click Add.

Configure Server Groups

Procedure

Step 1	Login to CUSP portal.
Step 2	Navigate to Configure > Server Groups > Groups.
Step 3	Enter a name (FQDN) for the Server Group. Example:ccm.hcsdc1.icm
Step 4	Choose global (default) from Load Balancing Scheme drop-down list.
Step 5	Choose hcs from Network drop-down list.
Step 6	Check the Pinging Allowed check-box.
Step 7	Click Add.
Step 8	Select newly added Server Group to add the elements for a respective server group.
Step 9	Select Elements tab and click Add.
Step 10	In <IP Address> text-box, enter the IP address of the Server Group, see Table 1.
Step 11	In Port text-box, enter the port value.
Step 12	Choose tcp from Transport Type drop-down list.
Step 13	In Q-Value text-box, enter the Q-Value as `1.0`.
Step 14	In Weight text-box, enter the weight `10`.
Step 15	Click Add.
Step 16	Repeat the above steps to configure cvp, gateway, ccm server groups.

Configure Route Tables

Table 2 Example Route Table

Key

Description

Host / Server Group (FQDN)

Network

4000

Agent Extension

ccm.hcsdc1.icm

Note

For Small Contact Center deployment model use Perimeta SBC signaling address: 10.10.10.49

hcs

7777

Network VRU label for CVP client

gw.hcsdc1.icm

hcs

8881

Network VRU label for CUCM client

cvp.hcsdc1.icm

hcs

811

Dialed number

cvp.hcsdc1.icm

hcs

912

Post call survey dialed number

cvp.hcsdc1.icm

hcs

9191

Ringtone

gw.hcsdc1.icm

hcs

9292

Error Tone

gw.hcsdc1.icm

hcs

6661111000

Network VRU label for MR client

cvp.hcsdc1.icm

hcs

978

Customer Dialed Number

out.hcsdc1.icm

hcs

Procedure

Step 1	Login to CUSP portal.
Step 2	Navigate to Configure > Route Tables.
Step 3	Click Add.
Step 4	Enter a name for a Route Table, click Add. Example:hcs
Step 5	Select the Route Table to add the rules for a respective route table.
Step 6	Click Add.
Step 7	In the Key text-box, enter key, see Table 1.
Step 8	Choose a Destination from Route Type drop-down list.
Step 9	In Host / Server Group text-box, enter Hostname (FQDN) / IP address, see Table 1.
Step 10	In Port text-box, enter the Port value.
Step 11	Choose an appropriate Transport Type from the drop-down list
Step 12	Choose an appropriate Network from the drop-down list.

Configure Route Policies

Procedure

Step 1	Login to CUSP portal.
Step 2	Navigate to Configure > Route Policies.
Step 3	Click Add.
Step 4	Enter a name for a Route Policy, click Add.
Step 5	Choose a Name from the drop-down list.
Step 6	Choose a Lookup Key Matches from the drop-down list.
Step 7	Choose the Lookup Key from the drop-down lists.
Step 8	Click Add.

Configure Route Triggers

Procedure

Step 1	Login to CUSP portal.
Step 2	Navigate to Configure > Route Triggers.
Step 3	Click Add.
Step 4	Choose a Routing Trigger from the drop-down list.
Step 5	Choose a Trigger from the drop-down list.
Step 6	Click Add.
Step 7	Select newly added Trigger to add trigger condition.
Step 8	Select the Trigger Condition from the drop-down lists.
Step 9	Click Add.

Full Configuration for Cisco Unified SIP Proxy

cusp(cusp)# show configuration active ver
cusp(cusp)# show configuration active verbose
Building CUSP configuration...
!
server-group sip global-load-balance call-id
server-group sip retry-after 0
server-group sip element-retries udp 2
server-group sip element-retries tls 1
server-group sip element-retries tcp 1
sip dns-srv
 enable
 no naptr
 end dns
!
no sip header-compaction
no sip logging
!
sip max-forwards 70
sip network hcs standard
 no non-invite-provisional
 allow-connections
 retransmit-count invite-client-transaction 3
 retransmit-count invite-server-transaction 5
 retransmit-count non-invite-client-transaction 3
 retransmit-timer T1 500
 retransmit-timer T2 4000
 retransmit-timer T4 5000
 retransmit-timer TU1 5000
 retransmit-timer TU2 32000
 retransmit-timer clientTn 64000
 retransmit-timer serverTn 64000
 tcp connection-setup-timeout 0
 udp max-datagram-size 1500
 end network
!
sip overload reject retry-after 0
!
no sip peg-counting
!
sip privacy service
sip queue message
 drop-policy head
 low-threshold 80
 size 2000
 thread-count 20
 end queue
!
sip queue radius
 drop-policy head
 low-threshold 80
 size 2000
 thread-count 20
 end queue
!
sip queue request
 drop-policy head
 low-threshold 80
 size 2000
 thread-count 20
 end queue
!
sip queue response
 drop-policy head
 low-threshold 80
 size 2000
 thread-count 20
 end queue
!
sip queue st-callback
 drop-policy head
 low-threshold 80
 size 2000
 thread-count 10
 end queue
!
sip queue timer
 drop-policy none
 low-threshold 80
 size 2500
 thread-count 8
 end queue
!
sip queue xcl
 drop-policy head
 low-threshold 80
 size 2000
 thread-count 2
 end queue
!
route recursion
!
sip tcp connection-timeout 30
sip tcp max-connections 256
!
no sip tls
!
sip tls connection-setup-timeout 1
!
trigger condition hcs_trigger_in
 sequence 1
  in-network ^\Qhcs\E$
  end sequence
 end trigger condition
!
trigger condition hcs_trigger_out
 sequence 1
  out-network ^\Qhcs\E$
  end sequence
 end trigger condition
!
trigger condition mid-dialog
 sequence 1
  mid-dialog
  end sequence
 end trigger condition
!
accounting
 no enable
 no client-side
 no server-side
 end accounting
!
server-group sip group ccm.hcsdc1.icm hcs
 element ip-address 10.10.10.31 5060 tcp q-value 1.0 weight 10
 element ip-address 10.10.10.131 5060 tcp q-value 1.0 weight 10
 failover-resp-codes 503
 lbtype global
 ping
 end server-group
!
server-group sip group cvp.hcsdc1.icm hcs
 element ip-address 10.10.10.10 5060 tcp q-value 1.0 weight 10
 failover-resp-codes 503
 lbtype global
 ping
 end server-group
!
server-group sip group gw.hcsdc1.icm hcs
 element ip-address 10.10.10.180 5060 tcp q-value 1.0 weight 10
 failover-resp-codes 503
 lbtype global
 ping
 end server-group
!
route table hcs
 key 4000 target-destination ccm.hcsdc1.icm hcs
 key 77777 target-destination gw.hcsdc1.icm hcs
 key 8881 target-destination cvp.hcsdc1.icm hcs
 key 91100 target-destination cvp.hcsdc1.icm hcs
 end route table
!
policy lookup hcs_policy
 sequence 100 hcs request-uri uri-component user
  rule prefix
  end sequence
 end policy
!
trigger routing sequence 1 by-pass condition mid-dialog
trigger routing sequence 3 policy hcs_policy condition hcs_trigger_out
trigger routing sequence 4 policy hcs_policy condition mid-dialog
trigger routing sequence 5 policy hcs_policy condition hcs_trigger_in
!
server-group sip ping-options hcs 10.10.10.49 4000
 method OPTIONS
 ping-type proactive 5000
 timeout 2000
 end ping
!
server-group sip global-ping
sip cac session-timeout 720
sip cac hcs 10.10.10.10 5060 tcp limit -1
sip cac hcs 10.10.10.131 5060 tcp limit -1
sip cac hcs 10.10.10.180 5060 tcp limit -1
sip cac hcs 10.10.10.31 5060 tcp limit -1
!
no sip cac
!
sip listen hcs tcp 10.10.10.49 5060
sip listen hcs udp 10.10.10.49 5060
!
call-rate-limit 200
!
end
cusp(cusp)#

Configure Gateway

Create a Sip-Server with the CUSP IP
Create a Dial-Peer

Create a Sip-Server with the CUSP IP
Create a Dial-Peer

Create a Sip-Server with the CUSP IP

sip-ua
 retry invite 2
 retry bye 1
 timers expires 60000
 timers connect 1000
 sip-server ipv4:10.10.10.49:5060
 reason-header override

Create a Dial-Peer

dial-peer voice 9110 voip
 description Used for CUSP
 preference 1
 destination-pattern 911T
 session protocol sipv2
 session target sip-server
 session transport tcp
 voice-class codec 1
 dtmf-relay rtp-nte
 no vad

Configure Unified CVP

Configure SIP Proxy
Configure SIP Server Groups
Configure Call Server

Configure SIP Proxy
Configure SIP Server Groups
Configure Call Server

Configure SIP Proxy

Procedure

Step 1	Login to Unified Customer Voice Portal.
Step 2	Navigate to Device Management > SIP Proxy Server, click Add New.
Step 3	Enter the IP Address, Hostname. Select Cisco Unified SIP Proxy from Device Type drop-down list .
Step 4	Click Save.

Configure SIP Server Groups

Procedure

Step 1	Login to Unified Customer Voice Portal.
Step 2	Navigate to System > SIP Server Groups, click Add New.
Step 3	Enter the FQDN name, IP Address, Port, Priority, Weight of CUSP and click Add.
Step 4	Click Save.

Configure Call Server

Procedure

Step 1

Login to Unified Customer Voice Portal.

Step 2

Navigate to Device Management > Call Server.

Step 3

Select Call Server > Click Edit > Click SIP tab.

Step 4

Select Yes to enable Outbound Proxy Server.

Step 5

Enter Outbound SRV domain name / Server Group Name (FQDN), click Save and Deploy.

Note

As CUSP provides centralized dialed plan , delete the existing Dialed number patterns.

Configure Cisco Unified Communications Manager

Login to the Unified Communications Domain Manager administration interface and perform the following steps to complete a route configuration toward the Unified CUSP server.

Add Trunk to CVP
Add Trunk to CUSP

Add Trunk to CVP
Add Trunk to CUSP

Add Trunk to CVP

Procedure

Step 1

Login to Cisco Unified Communication Domain Manager as provider, reseller or customer admin.

Step 2

Ensure that hierarchy is set to the node where Unified Communication Manager is configured.

Step 3

Navigate to SIP Trunks:

For provider or reseller administrator Device Management > CUCM > SIP Trunks
For customer administrator Device Management > Advanced > SIP Trunks

Step 4

Click Add to create SIP trunk.

Step 5

Perform the following, In Device Information tab:

Choose required IP address from CUCM drop-down list that you want to add SIP trunk.
Enter a unique SIP trunk name in Device Name field.
Choose Device Pool from the drop-down list.
Check Run On All Active Unified CM Nodes check-box.

Step 6

Goto SIP Info tab and perform the following:

Click Add icon in Destination panel.
Enter destination IP address of CVP Address IPv4 field.
Change Port to 5090.
Enter Sort Order to prioritize multiple destinations.
Note
Lower sort order indicates higher priority.
Choose newly added SIP Trunk Security Profile from the drop-down list.
Choose sip profile from the drop-down list.

Repeat this step to add another trunk.

Step 7

Click Save.

Add Trunk to CUSP

Procedure

Step 1

Login to Cisco Unified Communication Domain Manager as provider, reseller or customer admin.

Step 2

Ensure that hierarchy is set to the node where Unified Communication Manager is configured.

Step 3

Navigate to SIP Trunks:

For provider or reseller administrator Device Management > CUCM > SIP Trunks
For customer administrator Device Management > Advanced > SIP Trunks

Step 4

Click Add to create SIP trunk.

Step 5

Perform the following, In Device Information tab:

Choose required IP address from CUCM drop-down list that you want to add SIP trunk.
Enter a unique SIP trunk name in Device Name field.
Choose Device Pool from the drop-down list.
Select Run On All Active Unified CM Nodes check-box.

Step 6

Goto SIP Info tab and perform the following:

Click Add icon in Destination panel.
Enter destination IP address of CUSP in Address IPv4 field.
Change Port, if required.
Enter Sort Order to prioritize multiple destinations.
Note
Lower sort order indicates higher priority.
Choose newly added SIP Trunk Security Profile from the drop-down list.
Choose sip profile from the drop-down list.

Repeat this step to add another trunk.

Step 7

Click Save.

Configure Outbound with Cisco Unified SIP Proxy

Configure Unified CCE
Configure Gateway
Configure Cisco Unified SIP Proxy for IVR based Campaign

Configure Unified CCE
Configure Gateway
Configure Cisco Unified SIP Proxy for IVR based Campaign

Configure Unified CCE

Procedure

Step 1	Select Start > All Programs > Cisco Unified CCE Tools > Peripheral Gateway Setup.
Step 2	Click Add under Instance Component, then click Outbound Dialer to add the dialer.
Step 3	On the Outbound Dialer properties page, ensure that the SIP radio button is selected and then click Next.
Step 4	In the SIP Dialer Name text box, enter the SIP dialer name exactly as it is configured in the Dialer Tool under Configuration Manager.
Step 5	In SIP Server Type, ensure that (CUSP)/(CUBE) is selected.
Step 6	Enter CUSP IP in the SIP Server text box and click Next.
Step 7	In the Campaign Manager Server text box, enter Unified CCE DataserverA /RoggerA side IP address.
Step 8	In the CTI Server A text box, enter A side CTIOS server IP Address; in the CTI Server Port A text box, enter 42027 as the port number.
Step 9	In the CTI Server B text box, enter B side CTIOS server IP address; in the CTI Server Port B text box, enter 43027 as the port number.
Step 10	Keep all other fields as default and click Next. In the following window, click Next to complete the install.

Configure Gateway

dial-peer voice 811 voip
 description ******To CUCM*****
 destination-pattern 811T
 session protocol sipv2
 session target sip-server
 voice-class codec 1
 voice-class sip rel1xx supported "100rel"
 dtmf-relay rtp-nte h245-signal h245-alphanumeric
 no vad
!

 sip-ua
retry invite  2
retry bye 1
timers expires 60000
timers connect 1000
sip-server dns:out.hcsdc1.icm
reason header override
permit hostname dns:out.hcsdc1.icm

Configure Cisco Unified SIP Proxy for IVR based Campaign

Procedure

Step 1

Login to CUSP portal.

Step 2

Navigate to Configure > Route Tables.

Step 3

Click the existing route table.

Example:HCS.

Step 4

Select the Route Table to add the rules for a respective route table.

Step 5

Click Add.

Step 6

In Key text-box, enter key, 8881.

Step 7

Choose Destination from Route Type drop-down list.

Step 8

In Host / Server Group text-box, enter Hostname (FQDN) / IP address of CVP.

Example:cvp.hcsdc1.icm

Step 9

In Port text-box, enter the Port value.

Step 10

Choose an appropriate Transport Type from the drop-down list.

Step 11

Choose an appropriate Network from the drop-down list.

Note

As CUSP provides centralized dial plan management you can directly route the IVR call to CVP.

Avaya PG

Follow the below procedures for 4000 and 12000 agent deployment model:

Create Golden Template for Avaya PG
Configure Avaya PG

Create Golden Template for Avaya PG
Configure Avaya PG
Add Avaya PG
Setup Avaya PG
Translation Route for Avaya

Create Golden Template for Avaya PG

Follow this sequence of tasks to create the golden template for Avaya PG.. After each task, return to this page to mark the task "done" and continue the sequence:

Sequence	Tasks	Notes
1	Download HCS-CC_11.0(1)_CCDM-CCE-CVP_vmv9_v1.0.ova.	Follow the procedure Open Virtualization Format Files .
2	Create the virtual machine for the Unified CCE Avaya PG	Follow the procedure Create Virtual Machines.
3	Install Microsoft Windows Server	Follow the procedure Install Microsoft Windows Server 2012 R2 Standard Edition.
4	Install Antivirus Software	Follow the procedure Install Antivirus Software.
5	Install the Unified Contact Center Enterprise	Follow the procedure Install Unified Contact Center Enterprise.
6	Convert the virtual machine to a template.	Follow the procedure Convert the Virtual Machine to a Golden Template.

After you create all golden templates, you can run the automation process (Automated Cloning and OS Customization). After you run the automation process, you can configure the Avaya PG server on the destination system. See Configure Avaya PG.

Configure Avaya PG

This section explains the configuration procedures you must perform for the Avaya PG:

Sequence	Tasks	Notes
1	Configure Network Cards	Follow the procedure Configure Network Cards.
2	Verify the Machine in Domain	Follow the procedure Verify the Machine in Domain.
3	Configure Unified CCE Encryption Utility	Follow the procedure Configure Unified CCE Encryption Utility.
4	Add Avaya PG from Configuration Manager	Follow the procedure Add Avaya PG.
5	Setup Avaya PG	Follow the procedure Setup Avaya PG.
6	Configure CTI server	Follow the procedure Configure CTI Server.
7	Configure CTI OS server	Follow the procedure Configure CTI OS Server.
8	Configure Avaya ACD	Follow the procedure in section 2 of ACD Configuration.
9	Verify Cisco Diagnostic Framework Portico	Follow the procedure Verify Cisco Diagnostic Framework Portico.
10	Cisco SNMP Setup	Follow the procedure Cisco SNMP Setup.

Add Avaya PG

Complete the following procedure to add an Avaya PG using Unified CCE Configuration Manager.

Procedure

Step 1	Login to Unified CCE Admin Workstation server and navigate to Start > Cisco Unified CCE Tools > Administration Tools > Configuration Manager.
Step 2	Choose Tools > Explorer Tools and open PG Explorer in Configuration Manager window.
Step 3	Click Add PG and enter the following values in Logical Controller pane. Enter Avaya_PG_XX , where XX is the Avaya PG number, in the Peripheral Name field. Choose Avaya (Definity) in the Client Type field .
Step 4	Click Peripheral and enter the following values in Peripheral tab. Choose None in the Default Desk Settings field. Check Enable post routing.
Step 5	Click Routing Client tab and enter a name for Routing client.
Step 6	Click Save and Close .

Setup Avaya PG

Procedure

Step 1	Choose Start > All Programs > Cisco Unified CCE Tools > Peripheral Gateway Setup.
Step 2	Click Add in the Instance Components pane, and choose Peripheral Gateway
Step 3	Select the following in the Peripheral Gateway Properties dialog box: Check Production Mode. Check Auto Start System Startup. Check Duplexed Peripheral Gateway. Choose an appropriate PG from PG node Properties ID drop-down list. Select the appropriate side (Side A orSide B) accordingly. Under Client Type pane, add Avaya (Definity) to the selected types. Click Next.

Add PIM1 (Avaya PIM)

Procedure

Step 1

Enter the logical controller ID in the Peripheral Gateway Configuration pane.

Step 2

Select EAS-PHD Mode and check Using MAPD check-box in the Avaya (Definity)ECS Setting pane.

Step 3

Click Add, in the Peripheral Interface Manager pane.

Step 4

Select Avaya(Definity) and PIM1, click OK.

Step 5

Check Enabled in Avaya(Definity) ECS PIM Configuration dialog box.

Step 6

Enter the peripheral name in the Peripheral Name field.

Step 7

Enter the peripheral id in the Peripheral ID field.

Step 8

Check CMS Enabled and enter port number in Port number to listen on field, in Call Management System (CMS) Configuration pane

Step 9

Check Host1 as Enabled in the CVLAN/MAPD Configuration pane.

Step 10

Enter Hostname of ASAI link, check configured ASAI link number for Monitor ASAI links and Post-Route ASAI links

Step 11

Click OK and click Next.

Step 12

Select the preferred side in the Device Management Protocol Properties dialog-box.

Step 13

Click Next.

Step 14

Enter the PG Private Interfaces and the PG Public (Visible) Interfaces in the Peripheral Gateway Network Interfaces dialog box.

Step 15

Click the QoS button in the private interfaces section for Side A and check the Enable QoS check-box and click OK.

This step applies only to Side A.

Step 16

Click the QoS button in the public interfaces section for Side A and check the Enable QoS check-box and click OK.

This step applies only to Side A.

Step 17

Click Next and Finish.

Note

Do not start Unified ICM/CCNodeManager until all ICM components are installed.

Translation Route for Avaya

A translation route is a temporary destination for a call that allows call information to be delivered with the call. Network Blind Transfer is used to return the destination label to the originating CVP routing client.

Configure Unified CCE

Configure Unified CCE

Enable Network Transfer Preferred
Create Service
Configure Translation Route
Configure Script

Enable Network Transfer Preferred
Create Service
Configure Translation Route
Configure Script

Enable Network Transfer Preferred

Perform the below steps for Avaya, CVP and CUCM PIMs:

Procedure

Step 1	In Unified CCE Admin Workstation Server, navigate to Start > Cisco Unified CCE Tools > Administration Tools > Configuration Manager Displays Configuration Manager window.
Step 2	Expand Tools > Explorer Tools > PG Explorer.
Step 3	Choose appropriate PG from the list and expand the PG.
Step 4	Choose appropriate PIM from the list.
Step 5	Goto Routing Client tab, check Network Transfer Preferred check box.

Create Service

Procedure

Step 1	Log in to Unified CCDM portal as a tenant or sub customer.
Step 2	Select Resource Manager.
Step 3	Select the folder from the left hand side panel that you want to create service.
Step 4	Select Service from Resource drop-down list.
Step 5	Enter Name.
Step 6	Select appropriate Avaya peripheral from Peripheral drop-down list.
Step 7	Select Advanced tab, choose Cisco_Voice from Media Routing Domain drop-down list.
Step 8	Click Save.

Configure Translation Route

Procedure

Step 1

In Unified CCE Admin Workstation Server, navigate to Start > Cisco Unified CCE Tools > Administration Tools > Configuration Manager Displays Configuration Manager window.

Step 2

Expand Tools > Explorer Tools > Translation Route Explorer.

Step 3

In Translation Route tab:

Enter Name.
Choose DNIS in Type drop-down list.

Step 4

Click Add Route.

Step 5

In Route tab:

Enter Name
Choose newly created service from Service drop-down list.

Step 6

Click Add Peripheral Target

Step 7

In Peripheral Target tab:

Enter DNIS
Note
DNIS should be same as label.
Choose Network Trunk Group from drop-down list.

Step 8

Click Add Label.

Step 9

In Label tab:

Choose Routing Client from drop-down list.
Enter Label.
Note
Post route VDN should be created as label for CVP routing client

Step 10

Click Save.

Configure Script

Following illustration explains to configure scripts.

Figure 3. Configure Scripts

Cisco Virtualized Voice Browser

Create Golden Template for Cisco Virtualized Voice Browser
Configure Unified CVP
Configure Cisco Virtualized Voice Browser

Create Golden Template for Cisco Virtualized Voice Browser
Configure Unified CVP
Configure Cisco Virtualized Voice Browser

Create Golden Template for Cisco Virtualized Voice Browser

Follow this sequence of tasks to create the golden template for Voice Browser. After each task, return to this page to mark the task "done" and continue the sequence:

Sequence	Done?	Tasks	Notes
1		Download VB_11.0_vmv8_v2.5.ova	See Open Virtualization Format Files.
2		Create the virtual machine for Cisco Virtualized Voice Browser.	Follow the procedure that is documented in, Create Virtual Machines.
3		Install Cisco Virtualized Voice Browser.	Follow the procedure for installing VOS applications for golden templates. See Install Unified Communications Voice OS based Applications.
4		Convert the virtual machine to a Golden Template.	Follow the procedure Convert the Virtual Machine to a Golden Template.

After you create all golden templates, you can run the automation process (Automated Cloning and OS Customization). After you run the automation process, configure Cisco Virtualized Voice Browser. See Configure Cisco Virtualized Voice Browser.

Configure Unified CVP

Add Cisco Virtualized Voice Browser
Associate Dialed Number Pattern

Add Cisco Virtualized Voice Browser
Associate Dialed Number Pattern

Add Cisco Virtualized Voice Browser

Procedure

Step 1	Login CVP operation console.
Step 2	Navigate to Device Management > Gateway.
Step 3	Enter IP Address and Hostname of unified Voice Browser.
Step 4	Keep the default trunk option in Group ID field.
Step 5	Enter Username and Password.
Step 6	Enter Enable Password.
Step 7	Keep default option in Port field.
Step 8	Click Sign in.
Step 9	Click Save.

Associate Dialed Number Pattern

Procedure

Step 1	Login CVP Operation Console.
Step 2	Select System > Dialed Number Pattern.
Step 3	Select the Dialed Number Pattern from the list that you want to associate.
Step 4	From Route to Device drop-down list, select Virtualized Voice Browser IP.
Step 5	Click Save.
Step 6	Click Deploy.

Configure Cisco Virtualized Voice Browser

Access Virtualized VB Administration Web Interface
Access Virtualized VB Serviceability Web Page
Add a SIP Trigger
Configure Agent Greeting
Configure Whisper Announcement
Configure ASR and TTS
Configure Courtesy Callback for Cisco VVB

Access Virtualized VB Administration Web Interface
Access Virtualized VB Serviceability Web Page
Add a SIP Trigger
Configure Agent Greeting
Configure Whisper Announcement
Configure ASR and TTS
Configure Courtesy Callback for Cisco VVB

Access Virtualized VB Administration Web Interface

The web pages of the Virtualized VB Administration web interface allow you to configure and manage the Virtualized VB system and its subsystems.

Use the following procedure to navigate to the server and log in to Vitualized VB Administration web interface.

Procedure

Step 1

Open the Cisco Virtualized Voice Browser Administration Authentication page from a web browser and enter the following case-sensitive URL:https://<servername>/appadmin

In this example, replace <servername> with the hostname or IP address of the required Virtualized VB server.

Displays Security Alert dialog box.

Step 2

Login Cisco Virtualized VB Administration using your credentials.

Note

If you are accessing Virtualized VB for the first time, enter the Application User credentials that you specified during installation of the Virtualized VB.
For security purposes, Cisco Virtualized VB Administration logs out after 30 minutes of inactivity.
Virtalized VB Administration detects web-based cross-site request forgery attacks and rejects malicious client requests. It displays the error message, “The attempted action is not allowed because it violates security policies.”

Step 3

Import the license file and click Next to configure. DisplaysComponent Activation page.

Step 4

After all the components status shows Activated, click Next. DisplaysSystem Parameters Configuration page.

Step 5

Choose codec from the drop-down list and click Next. Displays Language Confirmation page.

Step 6

Choose Language from the drop down list and appropriate options.

Step 7

Click Next.

Access Virtualized VB Serviceability Web Page

The Vitrualized VB Serviceability is used to view alarm and trace definitions for Virtualized VB services; start and stop the Virtualized VB Engine; monitor Virtualized VB Engine activity and to activate and deactivate services. After you log in to Cisco Virtualized VB Administration web page, you can access Virtualized VB Serviceability:

From Navigation drop-down list, or
From Web Browser, enter: https://<server name or IP address>/uccxservice/.

Add a SIP Trigger

Follow the below steps to add a SIP trigger:

Procedure

Step 1	Log in to Cisco Virtualized Voice Browser Administration page.
Step 2	Select Subsystems > SIP Telephony > SIP Triggers.
Step 3	Click Add New.
Step 4	In Directory Information tab, enter Directory Number.
Step 5	Select Language from the drop-down list.
Step 6	Select Application Name from the drop-down list.
Step 7	Optional, click Show More to associate the trigger for ASR.
Step 8	In Override Media Termination field, select Yes option.
Step 9	Move required dialog groups between Select Dialog Groups and Available Dialog Groups.
Step 10	Click Add or Update to save the changes.

Configure Agent Greeting

Configure Whisper Announcement

Procedure

Step 1	Login Voice Browser Administration page.
Step 2	Navigate to Application > Application Management.
Step 3	Ensure ringtone application is listed and associated with the trigger `919191*`.

What to Do Next

Configure ASR and TTS

Virtualized Voice Browser supports ASR and TTS through two subsystems. Follow the below procedure to configure ASR and TTS subsystems:

Configure ASR Subsystem
Configure TTS Subsystem

Configure ASR Subsystem
Configure TTS Subsystem

Configure ASR Subsystem

ASR subsystem allows user to choose options through IVR:

Procedure

Step 1	Log in to Cisco Virtualized Voice Browser Administration page.
Step 2	Select Subsystems > Speech Servers > ASR Servers
Step 3	Click Add New.
Step 4	In Server Name field, enter hostname or IP address.
Step 5	Enter Port Number.
Step 6	Select Locales from the drop-down list and click Add Language.
Step 7	Check Enabled Languages check-box.
Step 8	Click Add.

Configure TTS Subsystem

TTS subsystem converts plain-text (UNICODE) into IVR.

Procedure

Step 1

Log in to Cisco Virtualized Voice Browser Administration page.

Step 2

Select Subsystems > Speech Servers > TTS Servers

Step 3

Click Add New.

Step 4

In Server Name field, enter hostname or IP address.

Step 5

Enter Port Number.

Step 6

Select Locales from the drop-down list and click Add Language.

Step 7

Check Enabled Languages check-box.

Step 8

Select Gender from the below options:

Male
Female
Neutral

Note

Select at least one gender for each enabled language.

Step 9

Click Add.

Note

Click Update to modify the existing configuration.

Configure Courtesy Callback for Cisco VVB

Procedure

Step 1	Log in to Cisco Virtualized Voice Browser Administration page.
Step 2	Select Application > Application Management.
Step 3	Select Comprehensive from the list.
Step 4	Ensure Comprehensive application is associated with the trigger 777777777*

What to Do Next

Configure courtesy callback for gateway, Unified CVP and Unified CCE, see Configure Courtesy Callback

Bias-Free Language

Results

Chapter: Install and Configure Optional Cisco Components

Install and Configure Optional Cisco Components

SPAN-Based Monitoring

Install SPAN based Silent Monitoring

SPAN-Based Silent Monitoring Configuration

Configurations for SPAN from Gateway

Silent Monitor Service Clusters

Configurations for SPAN from Call Manager

Unified CCE AW-HDS-DDS

Cisco RSM

Create Golden Template for Cisco Remote Silent Monitoring

Install the JTAPI Client

Install the Cisco RSM Server

Configuring SNMP Traps for Cisco RSM

Configure SNMP Agent in MIB

Configure Cisco RSM

Configure Cisco RSM for 500 and 1000 Agent Deployment

Configure RSM

Set RSM Configuration Settings for 500 and 1000 Agent Deployment

Configure JTAPI Client Preferences

Edit Registry Settings

Configure Gateway

Set Up the VXML Gateway

Configure Unified CVP

Upload RSM Prompts

Integrate the CVP Call Flow

Call Flow Deployment

Configure Unified CCE

Set the Agent Target Rule

Create the Supervisor Login Account

Create Routing Script for RSM

Configure Unified Communication Manager

Configure Simulated Phone

Create Simphone Device Dependencies

Create Simphone Device

Set Up the Login Pool Simphone

Configure Cisco RSM for 4000 Agent Deployment

Set RSM Configuration Settings for 4000 and 12000 Agent Deployment

Configure Cisco RSM for 12000 Agent Deployment

Configure Cisco RSM for Small Contact Center Deployment

Set RSM Configuration Settings for Small Contact Center Deployment

Configure Cisco RSM for A-Law Codec

Configure RSM

Configure Gateway

Configure Unified CVP

Configure Unified Communications Manager

Configure Service Parameters

Cisco MediaSense

Create Golden Template for Cisco MediaSense

Configure Cisco MediaSense

Cisco MediaSense Primary

Configure Cisco MediaSense Primary

Complete Setup for Primary Server

Configure Incoming Call

Cisco MediaSense Secondary

Add Secondary Node

Configure Cisco MediaSense Secondary

Complete Setup for Secondary Server

Configure MediaSense Forking

Provisioning Cisco Unified CM for Cisco MediaSense BIB Forking

Configure Device

Configure End User

Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking

Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking for HCS Deployment Models

Setup Global Level

Dial-Peer Level Setup

Set Up CUBE Dial-Peers for MediaSense Deployments

Provisioning Cisco Unified Border Element for Cisco MediaSense CUBE Forking for SCC Deployment Models

Set Up CUBE Dial-Peers for Small Contact Center Deployment

Provisioning TDM Gateway for Media Forking

Cisco Unified SIP Proxy

Install Cisco Unified SIP Proxy

Installation of CUSP

Example of Installation on a Service Module

Post Installation Configuration Tool

Obtaining New or Additional Licenses

Required Information