Table Of Contents
Implementing and Configuring the Solution
Implementation
Network Topology
Configuration Task Lists
Collaborative Care Configuration Task List
CallManager Configuration
Regions
Device Pool
Locations
Codec
Cisco IOS Gatekeeper Configuration
Cisco 7985 Devices Configuration
Installing Partner Device Types on CallManager
Polycom VSX-3000/VSX-5000 Device Configuration
Polycom H.323 PVX Configuration
Tandberg T1000 MXP Device Configuration
Cisco Unified Contact Center Express Configuration
Pre-Installation Checklist
UCCX Components
Prompt IVR Codec
Post-Installation Setup Procedures
Accessing the UCCX Administration Functions
Installing License File
Configuring UCCX for CallManager
Configuration of UCCX
Creating Skills
Creating Contact Service Queues (CSQs)
Adding Agent Resources
Assigning Agents to Skills
Uploading Scripts to UCCX
Linking Applications to Scripts
Multiple Route Points
Cisco 7985 Phone Configuration
Audio Settings
Video Settings
Network Settings
Verifying Proper Operation
Determining the System Information
XML Applications for 7985 for IPPA
Polycom PVX Phone Configuration
Polycom VSX-3000 and VSX-5000 Phone Configuration
Audio Settings
Polycom VSX-3000
Polycom VSX-5000
Video Settings
Network Settings
Configuring the VSX System to Use SCCP Protocol
Verifying Proper Operation
Determining the System Information
Firmware Upgrades
Tandberg T1000 MXP Phone Configuration
Audio Settings
Video Settings
Network Settings
Verifying Proper Operation
Firmware Upgrades
Agent Software with UCCX
Configuring Cisco Agent Desktop (CAD)
Installing CAD on the Windows Workstation
Installing IP Phone Agent (IPPA)
Subscribing XML Phones to the IPPA XML Service
Starting the IPPA XML Service on Phone
IPPA Caveats for the Cisco 7985
Cisco IOS Gatekeeper Configuration
QoS Configuration
AutoQoS
Layer 3 Device
Layer 3 Devices
Phone Configuration on CallManager for QoS
CTI Port Configuration on CallManager for QoS
Cisco IOS Gatekeeper QoS
ASA QoS
Traffic Reclassification
QoS Marking Using Cisco Security Agent
QoS Configuration—Not Covered
Access Security
Additionally for Deployment Models 2 and 3
ASA Configuration ACL FW and NAT/PAT Configuration
Sample Configuration from a Cisco ASA
Cisco CallManager Locations
Cisco CallManager SIP Trunk
MPLS VPN
Implementing and Configuring the Solution
This chapter provides configuration and implementation details for Collaborative Care—Language Interpretation Service. For each product component in the solution, we describe the implementation details required to enable voice and video to interwork with CM and UCCX—and thereby support a language skills-based resource. However, the assumption is that in general basic product installations and configuration items not impacting Collaborative Care are not covered and the general product guides and/or SRNDs should be consulted for complete implementation guidelines.
Implementation
The implementation of Collaborative Care involves the integration of several product components. The specific configuration guidelines provided are intended to ensure successful integration. See Chapter 3, "Solution Features and Components" for information about the product components and software dependencies before proceeding with implementation.
Network Topology
Figure 5-1 provides an overview of the network topology that is described in the implementation details. Refer to Figure 5-1 for a frame of reference as you implement each product.
Figure 5-1 Network Topology
Configuration Task Lists
Configuration of Collaborative Care Language Interpretation Service is a multiple-step process that involves network infrastructure and Unified Communications configuration. The implementation details may vary or require additional steps based on the deployment model implemented. This section provides a checklist of the functionality that must be implemented. The following section provide a step-by-step guide to configuration details. The implementation details focus on the attributes required to activate the services. For basic component installation and configuration guides, refer back to the product specifications. Site-to-site connectivity and business-to-business (B2B) connectivity for MPLS VPN are not covered.
Collaborative Care Configuration Task List
•
Key items to configure on Cisco CallManager:
–
Region to support voice and video
–
Locations to allocate bandwidth limits for voice and video
–
Device Pool to define devices that support video and voice
–
Cisco IOS Gatekeeper for H.323 support
–
Phone device configuration
•
Cisco Unified Contact Center Express:
–
Configuration of UCCX
–
RmCm, Rm_JTAPI
–
Call Manager configuration tasks for UCCX Agent configuration
–
Creating Skills
–
Creating CSQs
–
Adding Agent Resources
–
Assigning Agents to Skills
–
Script installation
–
Creating Pilot JTAPI Triggers, defining CTI Ports
•
Choose an IP Endpoint (Cisco 7985, Polycom PVX, Polycom VSX-3000, Polycom VSX-5000, Tandberg T1000 MXP):
–
Audio settings
–
Video settings
–
Network settings
–
XML applications if using Cisco 7985
•
Agent software install and configuration:
–
CAD
–
IPPA
•
Cisco IOS Gatekeeper Configuration if using Polycom PVX
•
QoS provisioning across all components
•
Access Switch Security
•
Deployment Model 2 (DM2) and 3 (DM3) specifics:
–
ASA ACL FW and NAT/PAT
–
ASA QoS
–
Locations for DM2 and DM3
–
SIP Trunk
CallManager Configuration
Regions
Regions are configured to provide a boundary for the codec bandwidth and video bandwidth that can be used per call instance. If an endpoint attempts to make a call beyond the limits defined in the region, the call is adjusted back to the values defined in the region.
To configure regions:
Step 1
Choose System > Region.
Step 2
Select Add New to create a new region that is used for all endpoints that are used for Collaborative Care. This includes both the clinician IP Video endpoint and the agent IP Video Endpoint.
Step 3
Provide a Name for the region, such as Video Region.
Step 4
Under Audio Codec, select G.722 as the codec that is used for the RTP voice communication.
Step 5
Under Video Call Bandwidth, select the radio button for kbps and enter 768. The value should always be a derivative of 56kbps or 64kbps. For this solution, 768 kbps is used for video calls.
Note
Regions are used to associate to a Device Pool.
Device Pool
Device pools are defined and associated to each device type to define a series of system-level definitions. The video region defined in the previous section is used as the region defined in the new device pool created for the video devices.
To configure device pools:
Step 1
Choose System > Device Pool.
Step 2
Select Add New to create a new device pool used for all IP Video devices used for Collaborative Care.
Step 3
Provide a Device Pool Name such as Video Device Pool.
Step 4
Under Cisco Unified CallManager Group, select the group used or choose Default.
Step 5
Under Date/Time Group, select CMLocal or a unique group if defined.
Step 6
Under Region, use the new Region defined in the previous step, such as Video Region
Step 7
Under Softkey Template, select Standard User.
Step 8
Under SRST Reference, select disable.
Step 9
For the remaining options, select the options required for the specific configuration in which this solution is installed.
Note
This device pool defined is used for all SCCP, H.323, and SIP devices.
Locations
Locations are used for Call Admission Control such that the link speed supporting connections between sites has enough bandwidth to support a video call. Using locations is a static non-topology aware method that is best used for fairly simple network topologies. For this example, deployment model 1 is used as the example where the main campus location is the hospital and a branch office location is the clinic.
To configure locations:
Step 1
Choose System > Locations.
Step 2
Select Add New to create a location. See the design section for methods to build locations.
Step 3
Under Name, give a name for this location (for example, Hospital_Main_Campus).
Step 4
Under Audio Calls Information, enter unlimited if bandwidth is not an issue or a value as a derivative of 64kbps. An example would be 512 kbps, which would support 8 simultaneous G.722 voice calls.
Step 5
Under Video Calls Information, enter None if no video calls are allowed, Unlimited if bandwidth is a non issue, or enter a value in kbps that is a derivative of the bandwidth used for video calls, which is 768kbps. An example would be 2304, which would support three simultaneous video calls inclusive of the voice bandwidth.
Step 6
Under RSVP Setting, choose No Reservation as RSVP is not used.
To create the location for the branch office, repeat steps 1-2 then proceed with the following:
Step 1
Under Name, give a name for this location (for example, Clinic_Branch_1).
Step 2
Under Audio Calls Information, enter unlimited if bandwidth is not an issue or a value as a derivative of 64kbps. An example would be 512 kbps, which would support 8 simultaneous G.722 voice calls.
Step 3
Under Video Calls Information, enter None if no video calls are allowed, Unlimited if bandwidth is a non issue, or enter a value in kbps that is a derivative of the bandwidth used for video calls, which is 768kbps. An example would be 2304, which would support three simultaneous video calls inclusive of the voice bandwidth.
Step 4
Under RSVP Setting, choose No Reservation as RSVP is not used.
Note
Consult with the network architect to choose the value for the bandwidth values to enter for audio and video calls. This value should be the bandwidth that should be allocated on the data connection for this service. Also, in general, the links should not be provisioned for more that 75-80% of the total link speed.
Codec
The phone configuration for the codec used in CallManager is a global setting that allows for the use of the G.722 codec.
To configure the codec:
Step 1
Choose System > Enterprise Parameters.
Step 2
Under Enterprise Parameters Configuration, set Advertise G.722 Codec to Enabled.
Cisco IOS Gatekeeper Configuration
As CallManager works as a H.323 VOIP-GW in a H.323 network, a Cisco IOS Gatekeeper must be defined inside CallManager. CallManager then registers as a VOIP-GW to this Gatekeeper.
To configure Cisco IOS Gatekeeper:
Step 1
Choose Device > Gatekeeper.
Step 2
Select Add New.
Step 3
Under Host Name/IP Address, either enter the hostname of the GK configured in the DNS Server or manually enter the IP address of the Gatekeeper.
Step 4
Under Description, enter a name that describes the Gatekeeper, for example, Hospital A Gatekeeper.
Step 5
Under Registration Request Time to Live, leave the default value of 60. This value represents a timer for a keep alive message between the CallManager H.323 Endpoint and the Gatekeeper.
Step 6
Under Registration Retry Timeout, leave the default value of 300. This value represents a timer for how often to retry registration in the event of a time out.
Cisco 7985 Devices Configuration
Ensure that the Cisco 7985 image loaded in CallManager is consistent with the image listed in the Software Release Table 3-4. If not, retrieve this image from CCO and upload to the CallManager. Once the image is loaded onto CallManager, the phone automatically downloads the image after the configuration steps for the Cisco 7985 Phone are completed.
The following steps outline how to define a Cisco 7985 for a clinician or an interpretation agent on the Cisco CallManager. Additional steps are also required on the phone itself that are described in a later section.
To configure Cisco 7985 on CallManager:
Step 1
Choose Device > Phone.
Step 2
Select Add New.
Step 3
Under Phone Type, select Cisco 7985 and click Next.
Step 4
Enter the MAC Address as seen the Cisco 7985 Phone (to obtain the MAC address, press the Settings key on the 7985 phone, then press 2 Network Settings, and then 2 MAC address). Enter the MAC address without the colons or periods.
Step 5
Under Device Pool, select the device pool defined to support video endpoints (for example, Video Device Pool).
Step 6
Under Phone Button Template, choose Standard 7985.
Step 7
Under Location, choose the unique location that has been defined where this IP Video Phone resides. (for example, Clinic_Branch_1 or Hospital_Main_Campus).
Step 8
The radio button for Retry Video Call as Audio should be checked to ensure that video calls without enough bandwidth as defined in the location pool can fallback to voice-only for the call.
Step 9
The radio button for Allow Control of Device from CTI should be checked if the 7985 is being provisioned for an interpretation agent only. If the phone is used for a clinician, this option should be unchecked. This would be a global setting for the phone.
Step 10
Under the Association Information when a Line number is defined with a Directory Number, each line has the option Allow Control of Device from CTI. For an agent phone that may have multiple extensions, the global setting should not be set in Step 8. Instead, the setting can be checked in the Directory Number Information section for the specific Directory Number to allow control from CTI.
Step 11
For the remaining parameters used to define a phone, following the configuration recommendations as outlined in the CM 5.1 Administration Guide:
•
Cisco Unified CallManager Release 5.1(1) New and Changed Information Guide http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008073ee44.html
•
Cisco Unified CallManager Administration Guide, Release 5.0(4) http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_book09186a008066fa60.html
Installing Partner Device Types on CallManager
To configure partner endpoints on CallManager, a Signed device file is required. This device file comes from either Tandberg or Polycom as a Signed File. When getting this file from the vendor, make sure to get the device file that is compliant with CallManager 5.1. Before configuring any partner device, perform the following steps.
To load device files on CallManager:
Step 1
Load the Signed File on a FTP server that is accessible from this CallManager.
Step 2
Go to the Navigation bar, choose Cisco Unified OS Administration, and select Go.
Step 3
Choose Software Upgrade > Install/Upgrade.
Step 4
Under Source, select Remote Filesystem.
Step 5
Under Directory, enter the subdirectory to the root FTP directory if necessary.
Step 6
Under Remote Server, enter the IP Address for the FTP server.
Step 7
Under Remote User, enter the user name of the FTP account.
Step 8
Under Remote Password, enter the password of the FTP account.
Step 9
Under Options/Upgrades, select the Signed File that contains the partner device information (for example, cmterm-PolycomVideoDevice-SCCP.cop.sgn or cmterm-T1001-sccp.cop.sgn).
Step 10
Click Next and the install procedure proceeds.
After these steps are performed, a confirmation of a successful install soon appears. The device option should then appear in the phone configuration under Phone Type if the installation was successful.
Note
This procedure is not required for the Polycom H.323 PVX endpoint.
Polycom VSX-3000/VSX-5000 Device Configuration
Ensure that the Polycom VSX3000 and VSX5000 image loaded on the phone is consistent with the image listed in the Software Release Table 3-5. If not, retrieve this image from Polycom and follow the instructions in Polycom VSX-3000 and VSX-5000 Phone Configuration.
The following outlines the steps to define a Polycom VSX-3000 or VSX-5000 device as a clinician endpoint on the Cisco CallManager.
Note
The VSX-3000 and VSX-5000 cannot be used as an agent endpoint due to the lack of CAD support.
To configure Polycom VSX-3000 or VSX-5000 on CallManager:
Step 1
Choose Device > Phone.
Step 2
Select Add New.
Step 3
Under Phone Type, select Polycom Video Endpoint and click Next.
Note
This option is the same for both VSX-3000 and VSX-5000.
Step 4
Under MAC Address, enter the value as seen from the device. See the Polycom Phone Configuration for details on how to retrieve the MAC address. Enter the MAC address without the:.
Step 5
Under Description, enter a text description to describe the usage of the phone.
Step 6
Under Device Pool, select the device pool defined to support video endpoints (for example, Video Device Pool).
Step 7
Under Phone Bottom Template, select Standard Polycom Video Endpoint.
Step 8
Under Common Phone Profile, select Standard Common Phone Profile.
Step 9
Under Location, choose the unique location that has been defined where this IP Video Phone resides. (for example, Clinic_Branch_1 or Hospital_Main_Campus).
Step 10
The Radio button for Retry Video Call as Audio should be checked to ensure that video calls without enough bandwidth as defined in the location pool can fallback to voice-only for the call.
Step 11
Proceed with defining a unique Directory Number for each line the phone uses.
Step 12
For the remaining parameters used to define a phone, following the configuration recommendations as outlined in the CM 5.1 Administration Guide:
•
Cisco Unified CallManager Release 5.1(1) New and Changed Information Guide http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008073ee44.html
•
Cisco Unified CallManager Administration Guide, Release 5.0(4) http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_book09186a008066fa60.html
Polycom H.323 PVX Configuration
The following outlines the steps to define a Polycom H.323 PVX endpoint as a clinician endpoint on the Cisco CallManager. This additional configuration step is required for H.323 endpoints in addition to the Gatekeeper configurations and endpoint configurations. Additional steps are also required on the phone application that is described in a later section.
Note
Polycom PVX is only used for a clinician endpoint.
To configure Polycom PVX on CallManager:
Step 1
Choose Device > Phone.
Step 2
Select Add New.
Step 3
Under Phone Type, select H.323 Client and click Next.
Step 4
Under Device Name, enter a unique device name that represents this PVX endpoint.
Step 5
Under Description, enter a text description that describes this endpoint.
Step 6
Under Device Pool, select the device pool defined to support video endpoints (for example, Video Device Pool).
Step 7
Under Common Phone Profile, select Standard Common Phone Profile.
Step 8
Under Location, choose the unique location that has been defined where this IP Video Phone resides. (for example, Clinic_Branch_1 or Hospital_Main_Campus).
Step 9
Under Signaling Port, enter 1720, which is the RAS port number for H.323 registration.
Step 10
The Radio button for Retry Video Call as Audio should be checked to ensure that video calls without enough bandwidth as defined in the location pool can fallback to voice only for the call.
Step 11
The Radio button for Wait for Far End H.245 Terminal Capability Set should be checked.
Step 12
The Radio button for Media Termination Point Required should be unchecked since the MTP could have negative impacts to the video stream.
Step 13
Under Gatekeeper Name, pull down and select the Gatekeeper that has been defined to support H.323 devices.
Note
The Gatekeeper should be defined before the H.323 Clients are defined.
Step 14
Under E.164, enter the E.164 number for the PVX endpoint.
Note
The E.164 address entered here should be identical to the Polycom PVX entry for E.164.
Step 15
Under Technology Prefix, enter the value for the gw-type-prefix as entered on the Gatekeeper (example #1*).
Step 16
Under Zone, enter the value as defined on the Gatekeeper for the Gatekeeper name (for example, HosA-gk).
Step 17
For the remaining parameters used to define a H.323 client, following the configuration recommendations as outlined in the CM 5.1 Administration Guide:
•
Cisco Unified CallManager Release 5.1(1) New and Changed Information Guide http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008073ee44.html
•
Cisco Unified CallManager Administration Guide, Release 5.0(4) http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_book09186a008066fa60.html
Tandberg T1000 MXP Device Configuration
Ensure that the Tandberg T1000 MXP image loaded on the phone is consistent with the image listed in the Software Release Table 3-6. If not, retrieve this image from Polycom and follow the instructions in Tandberg T1000 MXP Phone Configuration.
The following outlines the steps to define a Tandberg T1000 MXP device as a clinician endpoint on the Cisco CallManager.
Note
The Tandberg T1000 MXP cannot be used as an agent endpoint due to the lack of CAD support.
To configure Tandberg T1000 MXP on CallManager:
Step 1
Choose Device > Phone.
Step 2
Select Add New.
Step 3
Under Phone Type, select TANDBERG Video Endpoint and click Next.
Step 4
Under MAC Address, enter the value as seen from the device. See the TANDBERG Phone Configuration for details on how to retrieve the MAC address. Enter the MAC address without the:.
Step 5
Under Description, enter a text description to describe the usage of the phone.
Step 6
Under Device Pool, select the device pool defined to support video endpoints (for example, Video Device Pool).
Step 7
Under Phone Bottom Template, select Standard Tandberg Video.
Step 8
Under Common Phone Profile, select Standard Common Phone Profile.
Step 9
Under Location, choose the unique location that has been defined where this IP Video Phone resides. (for example, Clinic_Branch_1 or Hospital_Main_Campus).
Step 10
The Radio button for Retry Video Call as Audio should be checked to ensure that video calls without enough bandwidth as defined in the location pool can fallback to voice-only for the call.
Step 11
Proceed with defining a unique Directory Number for each line the phone uses.
Step 12
For the remaining parameters used to define a phone, following the configuration recommendations as outlined in the CM 5.1 Administration Guide:
•
Cisco Unified CallManager Release 5.1(1) New and Changed Information Guide http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008073ee44.html
•
Cisco Unified CallManager Administration Guide, Release 5.0(4) http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_book09186a008066fa60.html
Cisco Unified Contact Center Express Configuration
The Cisco Unified Contact Center Express (UCCX) was formerly known as the IP Contact Center (IPCC) Express product and has since been renamed. The installation process for UCCX is documented in the Cisco Customer Response Solutions Installation Guide found on Cisco Connection Online. It is recommended to consult this installation guide as it is beyond the scope of this document to describe the installation instructions in detail.
Starting in UCCX version 4.5, the first version of UCCX to support Cisco Unified CallManager 5.x, the term CRS (Customer Response Center) is used to refer to a single server running UCCX. In the IP Contact Center releases previous to UCCX 4.5, a cluster represented two CRS servers. The concept of a cluster however remains and during the installation process you must define the name of the cluster to which this installation of UCCX belongs.
Before beginning the installation process, review the checklist found in the Cisco Customer Response Solutions Installation Guide.
Pre-Installation Checklist
•
Review the deployment guidelines for the Cisco CRS components that you are installing and the server on which you are installing.
•
Make sure that the server on which you are installing is an approved server.
•
Review the guidelines for ensuring that your server operates most efficiently.
•
Install, configure, and start Cisco CallManager.
•
Install the Cisco-provided Windows 2000 Server operating system on the server on which you are installing.
•
Connect the server on which you are installing to the network.
•
Register your Cisco CRS purchase and obtain your license files. (If you have not yet obtained your license files, you can still install Cisco CRS, but you cannot run Cisco CRS applications.)
•
Obtain MS SQL Server 2000 from Cisco, if you plan to install it as part of the Cisco CRS installation procedure.
•
Obtain the general information that you must provide during installation and setup.
•
Review the installation notes.
•
Disable virus scanning and the Cisco Security Agent (CSA) service on the server on which you are installing.
During the installation process there are a few recommended settings that you may want to consider for your installation. These are described in the following sections.
UCCX Components
During the installation phase, it is recommended to install each of the items shown in Figure 5-2. These include the CRS Engine, CRS Agent Data store, CRS Config Data store, CRS Historical Data store, and Microsoft SQL Server. It is optional to select the Monitoring and Recording options.
Figure 5-2 UCCX Components Installation Recommendations
Prompt IVR Codec
It is recommended to select the G.711 Codec which is used to play prompts to callers. This codec is required if you decide to install Voice Recognition or Recording/Monitoring services post installation or in the future.
Figure 5-3 Prompt IVR Codec
Post-Installation Setup Procedures
There are two initial setup procedures that you must perform before you can access the complete set of Cisco CRS Administration features:
•
Cluster Setup—Activates Cisco CRS license files, collects information about Cisco CallManager Administrative XML Layer (AXL) and JTAPI providers, and establishes a Cisco CRS administrator. You must perform this procedure one time for a cluster.
•
Server Setup—Enables specific Cisco CRS components that run on the Cisco CRS server.
If you later need to update information that you specify during a setup procedure, you can use Cisco CRS Administration to make changes. For more information, refer to Cisco Customer Response Solutions Administration Guide.
Accessing the UCCX Administration Functions
There are two methods that you can use to gain access to the administration functions:
•
Directly from the console of the UCCX server:
On the Cisco UCCX server, choose Start > Programs > Cisco CRS Administrator> Application Administrator.
•
From a web browser on a PC that has IP connectivity to the newly-installed UCCX server:
Use the following URL: http://<servername>/AppAdmin. The PC must meet the following requirements to support the user interface:
–
Operating system—Windows 2000 Professional or Windows XP Professional
–
Browser—Microsoft Internet Explorer 6.x
–
Disable pop-up blockers
Installing License File
Once UCCX has been installed, you must next install the license file which is part of the cluster setup post installation steps.
To install the license file:
Step 1
On the Cisco CRS Administrator Setup screen, click Setup.
You see the License Information page.
Step 2
In the License File field, enter the path and name of a Cisco CRS license file or of a ZIP file that contains multiple license files.
Step 3
Click Next. The License Information page appears again.
Step 4
Take one of these actions:
•
If you wish to enter multiple license files that you have not put into a single ZIP file, enter the path and name of another license file in the License File field and then click Next. The license information page appears again. Repeat this process until you have entered all license files.
•
If you have entered the name of a ZIP file that contains multiple license files or have entered all your license files, leave the License File field blank and click Next.
The CallManager Configuration screen appears. This page displays the IP address of the primary AXL server provider that is configured for Cisco CRS and lets you specify backup AXL server providers. It also lets you specify the IP addresses of JTAPI providers and RmCm providers that are configured for Cisco CallManager. See the next section for information on completing these steps.
Configuring UCCX for CallManager
Once you have installed the License file, you see the CallManager Configuration screen:
The AXL Service must be configured and running on the Unified CallManager server, but first, this needs to be configured on UCCX.
In the AXL Service Provider Configuration area:
Step 1
If there are other available AXL service providers that Cisco CRS should use to access Cisco CallManager if the primary AXL service provider fails, move the IP address of each backup AXL service provider that you want from the Available AXL Service Providers list box to the Selected AXL Service Providers list box. Click the left arrow to move the selected IP address from the Available AXL Service Providers list box.
Step 2
If there is more than one item in the Selected AXL Service Providers list box, make sure that the IP address of the primary CallManager running the AXL service appears at the top of the list, followed by the backup CallManagers running the AXL service providers in the order that they should be used if an AXL service failover occurs. Use the up arrow or the down arrow next to the Selected AXL Service Providers list box to specify the order of these servers.
Step 3
In the User Name and Password fields, enter the Cisco CallManager user name and password, if you want to change the information that appears in these fields.
The JTAPI (Java Telephony API) protocol is used by UCCX to communicate with CallManager. It is through this communication protocol that CallManager notifies UCCX of an inbound call and how UCCX likewise informs CallManager of the agent to which the call should be routed.
In the JTAPI Provider Configuration area:
Step 1
Move the IP address of up to two CTI managers from the Available CTI Managers list to the Selected CTI Managers list box. Click the left arrow to move the selected IP address from the Available CTI Managers list box.
Step 2
If there is more than one item in the Available CTI Managers list box, make sure that the primary CTI manager appears at the top of the list, followed by the backup CTI manager. Use the up-arrow or the down-arrow next to the Selected CTI Managers list box to specify the order of these servers.
Step 3
In the User Prefix field, enter the prefix of the JTAPI provider that you want to create in Cisco CallManager. Cisco CRS automatically appends the node ID to this prefix and creates the JTAPI provider in Cisco CallManager.
Step 4
In the Password and Confirm Password fields, enter and confirm the password for the JTAPI provider.
The Resource Manager-Contact Manager, otherwise known as RmCm Subsystem, is broken down into two components. The RM or Resource Manager manages resources such as agents as to their state, Not-Ready, Ready, Working, Logged out, etc. The Contact Manager subsystem provides management of queues and participates in routing calls to agent resources.
In the RmCm Provider Configuration area:
Step 1
Move the IP address of up to two CTI managers from the Available CTI Managers list to the Selected CTI Managers list Box. Click the left arrow to move the selected IP address from the Available CTI Managers list box.
Step 2
If there is more than one item in the Available CTI Managers list box, make sure that the primary CTI manager appears at the top of the list, followed by the backup CTI manager. Use the up-arrow or the down-arrow next to the Selected CTI Managers list box to specify the order of these servers.
Step 3
In the User ID field, enter ID of the RmCm provider that you want to create in Cisco CallManager.
Step 4
In the Password and Confirm Password fields, enter and confirm the Password for the RmCm provider.
Configuration of UCCX
For deployment model 3, the UCCX infrastructure is managed by the Language Interpretation Service and therefore this section is not applicable for the health care provider.
Creating Skills
Agents are assigned one or more skills for which they are responsible. The skills assigned to the agents can each have a specific level of competence specified.
To create skills:
Step 1
From the CRS Administration menu bar, choose Subsystems/RmCm. The UCCX Configuration web page opens, displaying the RM JTAPI Provider area.
Step 2
On the UCCX Configuration navigation bar, click the skills hyperlink. The UCCX Configuration skills summary web page opens to display the Skill Name (customer-definable label assigned to an agent), if configured.
Step 3
Click the Add a New Skill hyperlink.
Step 4
Enter the Skill Name as shown below, then click the Add button.
Creating Contact Service Queues (CSQs)
Unified Contact Center Express uses a grouping mechanism for callers requiring similar services. In our case, these CSQs would map to language. These CSQs are what is referenced within the script which is executing on the UCCX platform and interacting with the caller.
To create CSQs:
Step 1
Enter the CSQ Name as shown in the example, then press Next.
Step 2
Once Next has been selected, you are asked to assign the skills that are used by the agents for this set of callers. Since our example is a 1:1 mapping, the CSQ-Spanish queue contains only one skill, Spanish. The Resource Selection Criteria should be set to Longest Available as shown. This selects the agent that has been available the longest. There are a number of other agent selection methods that can be used.
Adding Agent Resources
Agent resources are first defined as users in Unified CallManager. On the "End User" configuration page, there is an option that is used to specify the UCCX Extension. This drop down displays the extension assigned to the user, and if this option is not specified, the userid is not exported to Unified Contact Center Express. This drop down is shown below.
Note
IP Contact Center (IPCC) has been recently renamed to UCCX, so references toIPCC in the Unified CallManager are referring to the IPCC suite of applications which is inclusive of UCCX.
Once the agent has been identified in Unified CallManager as an UCCX agent resource, the agent should now be present in the UCCX system which is found under Subsystem/RmCm/Resources.
The name of the resource shown in Unified Contact Center Express (UCCX) is the last name of the end user as defined in Unified CallManager and not the userid of the end user.
Assigning Agents to Skills
To assign skills to the agent resources, select the Resource Name as shown on the RmCm/Resources menu (shown above). The following screen is displayed.
Select one or more skills and press the left arrow to move that skill into the Assigned Skills column. Each skill selected can have a unique competence level, depending on the skill level of the translator, or for other business reasons. One example is to reduce the calls that may interrupt a translator that has other primary responsibilities, but within the health care organization is used as overflow in the event that the full time interpreters are not available.
Uploading Scripts to UCCX
Scripts are developed using the CRS Editor tool which is included with Unified Contact Center Express. Once the script has been created, it must be uploaded to the UCCX system. These scripts are stored in the Repository Data Store (RDS) database, along with other information such as prompts, grammars, and various document files on the UCCX System.
From the UCCX Administration menu, choose Applications/Script Management.
The Script Management page opens as shown below.
Clicking on the Script Files directory, followed by Upload New Scripts, opens an upload dialogue box that allows you to upload the script file(s) from your local machine.
Once the upload has been performed, you are presented with a dialogue box confirming the upload. If you are replacing a script that is already in use, the dialogue box lets you refresh the script. If you do not refresh the script, the old copy is used by all applications and subsystems within the UCCX system.
There are two script refresh options:
•
Individual script refresh
•
Bulk script refresh
Note
If a large number of VRU (Voice Response Unit) scripts are configured for your system, the Upload a New Script and Refresh Scripts operations can take a long time to complete. These tasks can also result in high CPU utilization and hence should be performed with caution, especially during peak times of system usage.
Linking Applications to Scripts
Once the scripts are uploaded, applications must be created that define a JTAPI trigger (or pilot number) and map it to a script. To create an application, select Applications/Application Management/Add a New Application as shown below.
When adding a new application that uses a script, select Cisco Script Application and press Next.
Specify the maximum number of sessions that this application is allowed to support. This selection directly correlates to the number of CTI ports that have been defined for the UCCX system to use. In this example, up to 10 callers can be interacting with the script at any one time. Next, select the Script name from the script drop down menu. Any variables that have been defined in the script as a parameter are listed below. Typical variables that are defined with the parameters are prompts, retry counts, and delay seconds. This allows you to override the variable without having to edit the script. In most cases, you won't need to change this. Next select the Add button shown above
Once the application has been added, you are presented with the opportunity to define JTAPI triggers. Select Add new trigger from the menu as shown below.
The added trigger can be one of two types:
•
Traditional IVR DTMF-based trigger
•
An http-based trigger that is executed from within a web browser session that the clinician would initiate
In this example, we choose a standard JTAPI trigger that is used when the clinician dials the number specified for the trigger.
On the next screen, you are required to enter values for a number of fields. Specifying the directory number that the end user dials creates the JTAPI trigger. The CTI Route Point is automatically configured in Unified CallManager. Think of these as route patterns that CallManager uses to determine that calls to this number are calls to the application being defined in UCCX. The Route Points in CallManager are by default named RP_ + the directory number specified. In this case it would be RP_4560. Once this has been configured, select Add to complete the JTAPI trigger configuration task.
You may confirm that the route point has been automatically added to the Unified CallManager by selecting Device/CTI Route Point. It is strongly recommended not to manually change the CTI route point configuration from within CallManager as it may be overridden or refreshed from UCCX.
Multiple Route Points
It is possible to configure a number of different route points that all map to the same application. This may be useful in this solution deployment in that 4560 could be the main pilot number that provides the user with an audible list of languages to choose from. It may however be advantageous to offer the caller a number of different entry points in the form of separate phone numbers (JTAPI Triggers) for each language (in our example, 4561 for Spanish,4562 for Sign Language, etc.).
When the script executes, a simple check of the dialed phone number or JTAPI trigger used to invoke the application can be performed. This prevents the caller from having to navigate the IVR-based menu system and streamlines access to the language of choice. Furthermore, speed dials can be configured on the endpoint devices that map the text Spanish to extension 4561 and so on.
Cisco 7985 Phone Configuration
Audio Settings
The audio settings of the Cisco 7985 are configured on CallManager in the Region settings. It is recommended to create a Video Region and associate all Video endpoints, Route Points, and SIP trunks to this region. When the endpoint is defined to Callmanager either through auto registration or manually, it is assigned to either the default region or manually assigned to a administrator-defined region. In this way it is possible to have a subset of devices utilize the G.722 codec, which provides for greater audio clarity.
The Cisco 7985 has an integrated microphone and speakers. The Cisco 7985 supports the G.722 audio codec and adopts this setting as part of the configuration process when registering with Unified CallManager.
Video Settings
The Cisco 7985 support SIF video at 352x240 or CIF at 352x288 when using the SCCP Protocol. The selection of the video protocol used between two endpoints is determined during the H.264 video negotiation and defaults to the lowest resolution available on either endpoint.
•
460 (PAL)/470 (NTSC) TV lines
•
NTSC - SIF (352 x 240 pixels)
•
PAL - QCIF (176 x 144 pixels)
•
30 frames per second using H.264 when using 128 kbps (or more) for video
•
Up to 768-kbps IP
•
Camera: 460 (PAL)/470 (NTSC) TV lines
Network Settings
To configure LAN or network-related settings, use the menu option by selecting the Settings button to enter configuration menu options. Then select 2 for Network Configurations. You see the following menu:
Note
When this Menu is entered at first, the settings are locked. To unlock the settings to allow configuration, enter the sequence * * # on the 7985 keypad. Look at the keypad on the top right and the lock should be as shown in image shown above.
Use the following procedure:
Step 1
Use item 2, MAC Address to provision the phone inside CallManager.
Step 2
Item 19, SW Port Configuration—Set to Auto Negotiate.
Step 3
Item 20, PC Port Configuration—Set to Auto Negotiate.
Step 4
Item 18, Alternate TFTP Server is No.
Step 5
Determine if you are or are not using DHCP and follow the appropriate steps below:
If using DHCP:
a.
Item 17 DHCP Enabled, set this to Yes.
b.
Once connected to the access switch, DHCP finds the DHCP Server and the following fields are populated by the DHCP Server.
–
DHCP Server
–
IP Address
–
IP Subnet Mask
–
TFTP Server 1
–
TFTP Server 2 if alternate is defined
–
Default Router
–
DNS Server 1
–
DNS Server 2, 3, 4, 5 if applicable
–
Operational VLAN ID
If not using DHCP:
a.
Item 17 DHCP Enabled, set this to NO.
b.
Manually enter the following values as assigned by the Network IT:
–
DHCP Server
–
IP Address
–
IP Subnet Mask
–
TFTP Server 1
–
TFTP Server 2 if alternate is defined
–
Default Router
–
DNS Server 1
–
DNS Server 2, 3, 4, 5 if applicable
–
Operational VLAN ID
Verifying Proper Operation
To verify that the Cisco 7985 has successfully registered with CallManager, a phone number appears at the top right corner on the main screen. To get more detail on the status, select the Settings key and choose option 5 Status. To verify the Network settings, select 3 - Network Statistics.
Determining the System Information
To determine the firmware, revision, mac address, and other system details, select Settings and choose option 4. You should see the following:
Note
Some models show a Boot Load ID of S01655 1.4, 2005-06-13 which works as well.
XML Applications for 7985 for IPPA
See Agent Software with UCCX.
Polycom PVX Phone Configuration
This section provides configuration details on configuring the Polycom PVX to work in the Collaborative Care solution.
•
Registration to a gatekeeper
•
Setup for audio and video options
•
Network settings
For installation, general information, PC platform requirements, camera/microphone recommendations, and detailed product guidelines on Polycom PVX, refer to the product page for Polycom PVX:
http://www.polycom.com/products_services/1,,pw-7953,00.html
After completing installation and registration of the Polycom PVX, follow this procedure:
Note
Only the options essential to configure Polycom PVX to work as a clinician endpoint are shown.
Step 1
Chose General > General.
•
In the User Name, enter a field that is used as the unique H.323-ID for this endpoint. The E-mail address should follow the user name with the E-mail domain address.
•
The recommended setting for Video and Audio Device Preference is Use ViaVideo hardware, if present.
Step 2
Choose General > Call Settings—An option to set the maximum duration of a call in duration of minutes. If no maximum is required, then set the value to 0.
Step 3
Choose General > Performance—An option for the performance to allocate to the PVX application. If no other application is running on the tablet PC, then set this for Polycom PVX. If other applications are used on the PC, then set this value to Balanced.
Step 4
Choose Video > Video Protocol.
In this setting, ensure the Enhanced video (H.264) status option is checked to use H.264 video.
Step 5
Choose Video > Advanced.
In this menu select the following to achieve the best video performance:
•
Enable VGA people encoding
•
Enable VGA 30 frames per second
•
Select High for the camera driver's CPU load
Step 6
Choose Audio.
This menu selection depends on the audio output and input device. If the device has a hardware-based echo cancellation, then choose External for Echo Cancellation. If the hardware device does not have that capability, then choose Internal.
Always select Automatic Gain Control to help provide consistent volume control.
Step 7
Chose Network > Connections.
Selection Directly connected to a LAN or behind a fully aware H.323 firewall since the Cisco ASA FW is fully H.323 aware and the CallManager also performs protocol translation to SIP or SCCP depending on the call destination.
Step 8
Choose Network > Ports.
Ensure that H.323 TCP Port is set for 1720.
Unless there are conflicts with the Media Ports, there is no requirement to change the Media Ports.
Step 9
Choose Network > QoS.
Select Increase priority of video conferencing data on your network.
Step 10
Choose Network > Bandwidth.
Under this menu select 768 kbps for both he Default Call Rate and the Maximum Call Rate. All endpoints used are capable of 768 kbps.
Step 11
Choose H.323.
•
Under Gatekeeper, select Specify and enter the IP address of the Gatekeeper.
•
Under H.323 Extension (E.164), enter the directory number as defined also in the Cisco CallManager for the H.323 line number.
Note
The E.164 address in the Cisco CallManager must be consistent with E.164 entered in this menu.
Tip
To detect if the H.323 PVX endpoint is registered, enter this menu to see the status. In this case, it shows the Gatekeeper registration was successful. Going to the Gatekeeper, the endpoint should also show the device registered.
Tip
To get call statistics, choose Stats > Call Statistics to get call history and Stats > Media Statistics to see the type of bearer channels used for voice and video.
Polycom VSX-3000 and VSX-5000 Phone Configuration
Audio Settings
The audio settings of the VSX-3000 and VSX-5000 are configured on CallManager in the Region settings. It is recommended to create a Video Region and associate all Video endpoints, Route Points, and SIP trunks to this region. When the endpoint is defined to CallManager either through auto registration or manually, it is assigned to either the default region or manually assigned to an administrator-defined region. In this way, it is possible to have a subset of devices utilize the G.722 codec, which provides for greater audio clarity.
Polycom VSX-3000
The VSX-3000 has an integrated stereo microphone and speakers. There is not an option to add an external microphone to the endpoint. There are however other audio sources that can be used for input from a PC or VCR/DVD for example. The VSX-3000 supports the G.722 wideband audio codec and adopts this setting as part of the configuration process when registering with Unified CallManager.
Polycom VSX-5000
The VSX-5000 requires the use of an external microphone and speaker. Typically the VSX-5000 is placed on the top of a video monitor that has either S-Video or VGA/XGA video output. Stereo outputs are supplied and can be connected to an external speaker system either on the video monitor or external to speakers located in the room.
Video Settings
Both the VSX-3000 and VSX-5000 support SIF video at 352x240 or CIF at 352x288 when using the SCCP protocol. The selection of the video protocol used between two endpoints is determined during the H.264 video negotiation and defaults to the lowest resolution available on either endpoint.
•
NTSC 30fps at 56Kbps-2Mbps
•
PAL 24fps at 56Kbps-2Mbps
•
ITU based full screen Pro-Motion TM
•
H.263 interlaced video (60/50 fields full screen video for NTSC/PAL)
•
SIF (352 x 240), CIF (352 x 288)
•
QSIF (176 x 120), QCIF (176 x 144)
Network Settings
The network settings and general configuration menus are the same for both the VSX-3000 and VSX-5000 devices. To configure LAN or network-related settings, use the remote control to select System/Admin Settings/LAN Properties. Any of the following items can be changed in this menu and are shown below.
•
Hostname
•
IP Address
•
Domain Name
•
DNS Server
•
Default Gateway
•
Subnet Mask
•
WINS Server
•
WINS Resolution
•
LAN Speed & Duplex
Configuring the VSX System to Use SCCP Protocol
Using the remote control, select System/Admin Settings/Network/SCCP Settings and configure the following:
•
Auto Discover TFTP Address—This option allows DHCP to configure the IP address of the TFTP server, which is typically Cisco CallManager. This information in the form of an IP address is delivered to all DHCP hosts via DHCP Option 150.
•
TFTP Server Address—If you do not enable the auto discovery option for CallManager as described above, you must manually configure the IP address of the CallManager here. Up to three addresses can be specified and are used in priority order until the VSX system is able to register with CallManager.
When the VSX system has registered with CallManager, the directory number that was assigned to the device is displayed directly below the self video image. Additionally the CallManager Address is displayed when you navigate to System/Admin Settings/Network/SCCP.
Verifying Proper Operation
To verify that the VSX system (3000 or 5000) has successfully registered with CallManager, select Diagnostics/System Status found under the System menu option. If registered, you see a green up-arrow indicating that the VSX system has registered to CallManager.
Determining the System Information
To determine the firmware, revision, Mac address, and other system details, select System/System Information. You see the following fields:
•
System Name
•
Model Number
•
Serial Number
•
IP Video Number
•
System Software
•
Mac Address
•
Boot UI Version
•
IP Address
•
CallManager Name
•
Call Manager Version
•
DSCP Information
Firmware Upgrades
You can download a new version of firmware directly from Polycom's website. The upgrade process is straightforward through the use of the Softupdate application.
To update your software via the Internet:
Step 1
Using a web browser, go to http://www.polycom.com/videosoftware and log in to the Polycom Resource Center. You may need to set up a PRC account if you do not already have one.
Step 2
Navigate to your product page for your specific product. Refer to the release notes for information about the latest software version.
Step 3
Review the Upgrading Polycom Video Software documentation available for your VSX system for more detailed information on how to obtain a software key code and helpful hints in using the SoftUpdate program.
Step 4
Download the VSX Series SoftUpdate file in.zip format.
Step 5
Once you have obtained your Software Key Code from Polycom, you can execute the SoftUpdate program on your PC provided that it has IP connectivity to the VSX system.
Step 6
Once the application starts, it asks you for the IP address of the VSX system to upgrade, as well as the Software Key Code. In addition, you are asked for your remote access password as configured on the VSX system. This password can be changed via System/Admin Settings/General Settings/Security. Once supplied, the upgrade process should begin.
Note
Do not interrupt the upgrade process; if you do, the system may be unusable.
Tandberg T1000 MXP Phone Configuration
Audio Settings
The audio settings of the T1000 MXP when using SCCP is configured in the CallManager Region settings. It is recommended to create a Video Region and associate all Video endpoints, Route Points, and SIP trunks to this region. When the endpoint is defined to CallManager either through auto registration or manually, it is assigned to either the default region or manually assigned to an administrator-defined region. In this way, it is possible to have a subset of devices utilize the G.722 codec, which provides for greater audio clarity.
Video Settings
The T1000 MXP supports H.264 video at up to 30fps at the following video resolutions:
•
SIF (352 x 240)
•
CIF (352 x 288)
The selection of the video protocol used between two endpoints is determined during the H.264 video negotiation and defaults to the lowest resolution available on either endpoint.
Network Settings
The network settings and general configuration menus for the Tandberg T1000 MXP are navigated using the remote control included with each T1000. To configure the LAN or network-related settings, select Settings/Network Settings. Any of the following items can be changed in this menu and are shown below.
•
DNS Settings
•
IP Settings:
–
IP Assignment DHCP or Static
–
If Static, IP Address, Subnet Mask, Default Gateway
–
Ethernet Speed 10/Half, 10/Full, 100/Half, 100/Full or Automatic
–
IP Access password used for remote access to the T1000 MXP
•
TFTP Settings—This setting is used to specify the IP Address of the CallManager. It can be automatically obtained via DHCP or the DHCP option 150 can be overridden by the manual configuration as shown.
–
Alternate TFTP Server (Yes/No)—This option allows DHCP to configure the IP address of the TFTP server, which is typically Cisco CallManager. This information in the form of an IP address is delivered to all DHCP hosts via DHCP Option 150.
–
TFTP Server Address—If you enable the Alternate TFTP Server field, you can specify up to two TFTP Server addresses. These are used in priority order until the T1000 MXP is able to register with CallManager.
Verifying Proper Operation
To verify that the T1000 MXP has successfully registered with CallManager, select System Status found under the Settings/View Status menu option. If registered to CallManager, you see the word Active next to the CallManager 1 field shown in the lower portion of the second screen below.
You see the following information:
•
Active IP Address
•
Active IP Subnet Mask
•
Active Gateway
•
Boot Load ID
•
Software Version
•
Serial Number
•
MAC Address
•
Ethernet Speed
•
VLAN ID
•
Various WLAN settings (not used in this solution)
•
CallManager 1
Firmware Upgrades
Unlike other Cisco endpoints, the T1000 does not download its firmware from the CallManager. Before starting the upgrade, make sure that you have obtained a Release Key from Tandberg. The release key is unique to each T1000 MXP and can only be obtained directly from Tandberg technical support or a Tandberg reseller.
The system can be upgraded through FTP or a Web browser (IE 6.0+) interface.
Web Interface Upgrade Method
To upgrade using the web interface, point the browser to the IP address of the T1000 MXP system. To obtain the IP address of the system, use the System Status menu as shown above. Once you have entered this address, you should be prompted with the system web menu. Select the System Configuration Tab and then select upgrade.
Enter the Release key provided by Tandberg or your Tandberg reseller. Once entered, press the Install Software button. The next menu should be displayed, provided that the serial number dependent release key that you entered is valid for the T1000 MXP. At this point, simply enter the image filename and path or use the browse button to navigate to the location of the file.
Pressing the Install button begins the install process.
FTP Upgrade Method
Step 1
Copy the new firmware to a folder on your local hard disk.
Step 2
From a DOS window, navigate to the folder where the new firmware is stored.
Step 3
Enter ftp <ip address of the T1000 MXP>.
Step 4
Type in the Release Key as the UserID. You should have obtained this from Tandberg or your Tandberg reseller.
Step 5
Type in your password; the default is TANDBERG.
Step 6
Type bin and press enter.
Step 7
Type put <firmware filename> and press Enter.
Step 8
The upload should start.
Step 9
Once completed, close your FTP session and restart the T1000 to boot with the new firmware.
Agent Software with UCCX
Configuring Cisco Agent Desktop (CAD)
Configuring an agent in Unified CallManager is as simple as adding an end user and then assigning an endpoint device to the account. The directory numbers assigned to the lines on the phone are automatically displayed in the primary extension field. It is critical to also assign an UCCX Extension to the end user account to trigger the export of that user to UCCX. Note that after the renaming of IPCC to UCCX, Unified CallManager still refers to UCCX as IPCC.
The Cisco Agent Desktop (CAD) is a Windows-based client that allows the agent to interact with the UCCX system. The agent is able to login to UCCX through CAD and change their state as necessary in order to signal to UCCX whether it should direct calls to this particular agent.
Note
If the agent switches between using CAD and IP Phone Agent (IPPA), the length of the agent userid should be kept to a minimum. For more detailed information, see Installing IP Phone Agent (IPPA).
Endpoints that use CAD must be associated not only to the primary agent/end user, but also to a special application user account that was created in Unified CallManager during the initial install of UCCX. This user account is listed under User Management/Application User in CallManager and has the form xx-RmCm, where xx was a two character prefix specified at the time of the UCCX installation.
This xx-RmCm application user account needs to be associated with each endpoint that uses CAD. This enabled UCCX to query the status of the endpoint using the JTAPI protocol. Without this association, CAD does not fully initialize and end its agent logon process when it attempts to query CallManager for the device status (register, on-hook, off-hook, etc.).
To associate the endpoint devices, simply move the MAC address of the endpoints from the Available Device section into the associated section as shown below. All devices that use CAD or IPPA must be in the lower section as shown, and not in the upper section.
Installing CAD on the Windows Workstation
To install Cisco Agent Desktop:
Step 1
Open your web browser and access the Cisco Unified Contact Center Express User web page at http://servername/appuser. Replace server name with the host name or IP address of the UCCX server name or IP address.
Step 2
The Unified Contact Center Express system user authentication window appears. At the prompt, enter your user name and password and then click Log On. The Welcome window appears.
Step 3
Click the UCCX Downloads hyperlink. The Download Page window appears.
Step 4
Click the Cisco UCCX Agent Desktop hyperlink. Install Manager starts and displays the Welcome window. Click Next.
Step 5
The Installation Server: Location dialog box appears. Enter the host name or IP address and port number of the UCCX Web Server and click Next. This may be already populated for you; if so, leave the fields as shown. The host name or IP address is the same one you used to access the UCCX web interface.
Step 6
Select the version of CAD you wish to install and click Next. The License Agreement dialog box appears. Click Yes to accept the End User License Agreement.
Step 7
The Choose Destination Location dialog box appears. Accept the default destination folder or click Browse to navigate to another destination folder, then click Next.
Step 8
The Start Copying Files dialog box appears. Click Next to start the installation. Install Manager installs the application you chose.
Installing IP Phone Agent (IPPA)
IP Phone agent is an XML application that can be used by the agents to identify themselves to the UCCX system. It is resident on Unified Contact Center Express system and is invoked by subscribing an XML-enabled endpoint to the IPPA Application.
Step 1
The first step is to create the service in Unified CallManager. Phone Services are located under Device/Device Settings/Phone Services as shown below.
Step 2
Select the Add New button and enter values in the following fields as shown:
•
Service Name: IPPA
•
ASCII Service Name: IPPA
•
Service Description: IPPA
•
Service URL: http://<UCCX server name or IP>:6293/ipphone/jsp/sciphonexml/IPAgentInitial.jsp
Step 3
Then press the Save button at the bottom. An update successful message should be displayed.
Subscribing XML Phones to the IPPA XML Service
To subscribe a phone to the newly created IPPA XML Service:
Step 1
Select the device as found under Device/Phone located in CallManager. Once you have located the phone to which you want to subscribe the IPPA service, select the Subscribe/Unsubscribe option located on the right as shown below.
Step 2
From the drop down menu, select the IPPA XML Service which you created on CallManager previously, then select Next.
Step 3
You may override the previously configured fields if necessary. This may be necessary to more accurately describe the IPPA function or to distinguish between test and production versions of IPPA that may reside on different Unified Contact Center Express servers.
Step 4
Once you have made the changes (if desired) to the service names, select Subscribe to complete the subscription process.
Starting the IPPA XML Service on Phone
Step 1
By pressing the Services button on the phone, a list of subscribed services should be displayed. One of these services should be IPPA as defined to Unified CallManager in the preceding section.
Step 2
Upon IPPA startup, the user is requested to login using their End User account as defined in CallManager. This userid field may be somewhat cumbersome to enter using the telephone keypad. It is recommended to keep the userid somewhat short and in lower case if at all possible. Once the user enters the ID, Password, and Extension they are presented with the IPPA main screen.
Step 3
Notice that the user is logged on, but in a Not Ready state. In this state, UCCX knows that the agent is online, but not ready to accept calls. The agent would next needs to press the State button and change their state to Ready as shown below.
Step 4
When an inbound call is received and answered by the agent, IPPA displays the call status and changes the state of the agent to Talking. Upon ending the call, the user is typically placed in the Ready state again unless UCCX has been configured to transition the user into Work state. The Work state is used to allow the agent to wrap up work that may be necessary for documentation purposes post call.
Step 5
If during the call, the agent wishes to change their status to Work, they may do so by selecting the Stats menu option on IPPA. This changes their status to Working so that after the call they can perform any wrap up tasks before accepting the next call.
IPPA Caveats for the Cisco 7985
The 7985 requires that the DNS which has been configured on the phone either manually or via DHCP must be capable of resolving the fully qualified hostname of the Unified CallManager. In other words, if the domain name passed to the 7985 during DHCP configuration is hospital.org and the Unified CallManager hostname is CM1, the DNS server must resolve CM1.hospital.org to the IP address of the CallManager from the viewpoint of the 7985.
If NAT/PAT is being used between the CallManager and the 7985, the DNS server must return the NATed IP address of the CallManager and not its real address. If there are agents inside a firewall performing NAT on the IP address of the CallManager, that DNS server must respond to the DNS request using the actual IP address of the CallManager.
If this is not done, the Services URL button does not function. You can determine if this is the case by selecting the Settings button on the 7985 and selecting HTML Settings. If the URL fields are empty for Directories, Services, etc., then the 7985 was unable to resolve the fully qualified hostname of the CallManager passed to it during its initial boot up sequence. During this sequence, the 7985 requests its configuration file named SEP+MACAddress.cnf.xml via TFTP. This file contains a field called Process NodeName which is the hostname of the CallManager. This field does not change when you change the CallManager Server name via System/Server from a hostname to an IP address.
Figure 5-4 IPPA Caveat for the Cisco 7985
Cisco IOS Gatekeeper Configuration
To support H.323 devices in this solution, all devices register to the Cisco IOS Gatekeeper. The two devices focus on the Cisco CallManager and the Polycom PVX. Each of these devices is configured to register with the Gatekeeper. The Gatekeeper keeps track of the availability of these devices to receive calls based on its registration table. The registration table has directory numbers for each Polycom PVX endpoint that is registered. The Cisco CallManager is also registered to the Gatekeeper. Calls made from the Polycom PVX are routed to the Cisco CallManager, which then routes the call and provide protocol translation between endpoint types.
Configuration for the gatekeeper is simple:
zone local <gatekeeper name> cisco.com <IP address of Gatekeeper, recommend to use
loopback address>
gw-type-prefix #1* default-technology
Once CallManager and Polycom PVX register, the following registration table is shown on the gatekeeper. The Polycom PVX registers as a terminal with a unique E164 address and H.323 ID. The Cisco CallManager registers as a VOIP-GW and is a catch all to direct all calls towards with the exception for the E164 address that are registered to the Gatekeeper. With the given example, 3301 and 3302 resolve to the IP address of the registered endpoint while all other numbers resolve to the Cisco CallManager functioning as a VOIP-GW.
GATEKEEPER ENDPOINT REGISTRATION
================================
CallSignalAddr Port RASSignalAddr Port Zone Name Type Flags
--------------- ----- --------------- ----- --------- ---- -----
172.21.52.98 33005 172.21.52.98 32787 HosA-gk VOIP-GW
H.323-ID: RasAggregator_#1*_HosA-gk_1
Voice Capacity Max.= Avail.= Current.= 0
10.21.153.204 1720 10.21.153.204 1719 HosA-gk TERM
10.21.153.205 1720 10.21.153.205 1719 HosA-gk TERM
Total number of active registrations = 3
QoS Configuration
AutoQoS
AutoQoS is a powerful tool to rapidly apply and deploy the QoS model to support the traffic class required for this solution. AutoQoS Enterprise introduced the support for video traffic. For Cisco IOS routers, AutoQos Enterprise detects and provisions for up to ten classes of traffic:
•
Voice
•
Interactive-Video
•
Streaming-Video
•
Call-Signaling
•
Transactional Data
•
Bulk Data
•
Routing
•
Network Management
•
Best Effort
•
Scavenger
The AutoQoS Enterprise feature consists of two configuration phases, completed in the following order:
•
Auto Discovery (data collection)—Uses NBAR-based protocol discovery to detect the applications on the network and performs statistical analysis on the network traffic.
•
AutoQoS template generation and installation—Generates templates from the data collected during the Auto Discovery phase and installs the templates on the interface. These templates are then used as the basis for creating the class maps and policy maps for your network. After the class maps and policy maps are created, they are then installed on the interface.
For devices that do not support the AutoQoS feature, the implementation section provides some configuration steps to classify traffic accordingly using a DIFFSERV QoS model.
Layer 3 Device
The following should be applied on devices that are Layer 3-aware inside the LIS or hospital network. The key elements are
•
Class map to group voice/video traffic
•
Class map to group signalling traffic
•
Policy map to assign a percentage to the queue
•
Apply the policy map on the interface
Apply these to points of congestion within the network or at the edge of the network. Follow the QoS design rules as provided in the SRND for Enterprise QoS.
Layer 3 Devices
class-map match-all BEARER-TRAFFIC
class-map match-any CallSignaling
policy-map EGRESS-INTERFACE
priority percent 33 ! Choose percentage based on traffic model allocated to voice
bandwidth percent 5 ! Choose percentage based on traffic model allocated to call
signalling
interface GigabitEthernet0/0
description connection to Catalyst 6509E
ip address 172.21.61.65 255.255.255.240
service-policy output EGRESS-INTERFACE
Phone Configuration on CallManager for QoS
Step 1
Choose System > Enterprise Parameters.
Step 2
Under Enterprise Parameters Configuration:
•
DSCP for Phone Configuration should be set to CS3(preference 3) DSCP (011000)
•
DSCP for CallManager to Device Interface should be set to CS3(preference 3) DSCP (011000)
Step 3
Choose System > Service Parameters.
Step 4
Under Server, select the server for the desired configuration.
Step 5
Under Service, select Cisco CallManager.
Step 6
Under Cluster Parameters (System - QOS):
•
Priority Class should be set to Normal Priority
•
DSCP for Audio Calls should be set to EF DSCP (101110)
•
DSCP for Video Calls should be set to AF41 DSCP (100010)
CTI Port Configuration on CallManager for QoS
These are mainly the default setting already defined in Call Manager for DSCP settings.
Step 1
Choose System > Service Parameters.
Step 2
Under Server, select the server for the desired configuration.
Step 3
Select the Advance option to see the hidden options.
Step 4
Under Service, select Cisco CTIManager.
Step 5
Under Clusterwide Parameters (System - QOS):
•
DSCP for ICCP Protocol Links should be set to CS3(preference 3) DSCP (011000)
•
DSCP IP CTIManager to Application should be set to CS3(preference 3) DSCP (011000)
Cisco IOS Gatekeeper QoS
The Gatekeeper uses Cisco IOS methods to map H.323 RAS traffic. Once this traffic is mapped, a DSCP bits are set to CS3 through a policy-map. This policy-map is then applied to the Gatekeepers outbound interface to police all H.323 RAS traffic to ensure the proper DSCP is set.
class-map match-all ras_signaling
description class map for H.323 RAS traffic
interface GigabitEthernet0/0
description To HospitalA-Campus Shared 3750
ip address 172.21.52.101 255.255.255.240
service-policy output set-qos
ASA QoS
The ASA is along the path for traffic flowing from the enterprise to the WAN and the WAN to the enterprise. This setting should be applied to each EGRESS interface to allow for priority queueing for voice, video and signalling traffic.
interface GigabitEthernet0/0
ip address 172.21.52.114 255.255.255.252
interface GigabitEthernet0/1
ip address 172.21.52.78 255.255.255.240
class-map inspection_default
match default-inspection-traffic
service-policy global_policy global
service-policy VOIP-POLICY interface outside
service-policy VOIP-POLICY interface inside
Traffic Reclassification
To comply to the QoS Model used for this solution, some application data requires reclassification to meet the DSCP markings recommended. To implement this model, this setting should be applied to the first Layer 3 QoS-enabled device to remark the DSCP settings for:
•
UCCX messages to CallManager
•
UCCX JTAPI messages to CAD
•
UCCX JTAPI messages to IPPA running on the Cisco 7985
•
CAD JTAPI messages to UCCX
•
IPPA JTAPI messages to UCCX
•
H.323 messages from the Polycom PVX application
The following is the configuration for all JTAPI messaging:
Note
For a complete list of ports used by the UCCX system, refer to the Cisco Contact Center Port Utilization Guide:
http://www.cisco.com/en/US/products/sw/custcosw/ps1846/products_installation_and_configuration_guides_list.html
ip access list extended IPPA-CAD
permit tcp any any eq 42027 ! CAD IPPC Gateway PG
permit tcp any any eq 59020 ! CAD Chat Service
permit tcp any any eq 59000 ! GIOP
permit tcp any any eq 59003 ! Stat Service
permit tcp any any eq 59004 ! Enterprise Server
permit tcp any any eq 65432 ! LRM Services CORBA port
permit tcp any any eq 37350 ! VPN Autodiscover
permit tcp any any eq 59028 ! CORBA for Cisco Supervisor Desktop
permit tcp any host <UCCX Host> eq 8080 ! IPPA Servelet running under TomCat webserver
permit tcp any any eq 59010 ! IPPA JSP Client
permit tcp any any eq 38983 ! LDAP Directory Services for CAD, CSD, CDA
permit udp any any eq 59010 ! IPPA server CORBA port and VoIP client's to-agent
monitoring port
permit tcp any any eq 6293 ! Web Administrators - required for system maintance
class-map match-all IPPA-CAD
match access-group name IPPA-CAD
policy-map IPPA-CAD-Signaling
For H.323 Polycom PVX reclassification, apply the following configuration and apply to the EGRESS interface.
ip access-list extended H.323-Signaling
permit tcp any any range 1718 1720
class-map match-all H.323-Signaling
match access-group name H.323-Signaling
policy-map H.323-Signaling
QoS Marking Using Cisco Security Agent
Beginning in CSA release 5.0, host generated application traffic can be classified and marked using the Differentiated Services Code Point (DSCP) standard. This feature is especially useful when applications running on a host do not self classify their traffic using DSCP, or when access layer switches are not capable.
The Cisco Security Agent uses a centralized server that is used to distribute security and QoS policy to the CSA agents within the administrative domain. This administrator tool is referred to as the CSA MC. Within the CSA MC interface, the network administrator can create and assign QoS and security policies as needed.
The advantage of using CSA to classify and mark traffic generated by the host applications is that it does so at the point of traffic insertion into the network. This approach allows all downstream devices to honor the DSCP based QoS policy as close to the edge as possible. Since the policy is not self administered by the end user, QoS policy consistence can be obtained across the network, again through CSA's centralized approach to policy control.
Some examples of host-based applications that the Cisco Collaborative Care—Language Interpretation Service employs but which do not mark or mark traffic at all or in some cases mark it incorrectly are shown below.
•
Polycom PVX
•
Cisco Agent Desktop
The configuration of such marking and classification is documented in the Implementing Trusted Endpoint Quality of Service Marking document: http://www.cisco.com/application/pdf/en/us/guest/products/ps6786/c1225/ccmigration_09186a00805b6a81.pdf
In order to complete the configuration of the necessary QoS markings for these applications within CSA MC, the following steps are necessary:
Step 1
Create a Static Application Class.
This is used to identify the application by name (Polycom PVX in this example), such as **\vvsys.exe. The ** indicates any path that the executable has been launched from.
Step 2
Create a QoS Rule Module.
This creates an instance of a rule to be applied to the Application Class just created. An application class can have a number of different rules.
Step 3
Add rule to Rule Module.
This allows the assignment of the rule by type of rule—in this case a QoS DSCP attribute, as we want CSA to reclassify traffic generated based on port number for the specific application.
Step 4
Create Access Control Rule.
This assigns the desired DSCP QoS markings to the rule module just created.
In summary, the CSA MC tool allows you to identify the application name and a set of rules that should be applied to traffic being generated by the specified application. The application names for Polycom PVX and Cisco Agent Desktop are shown in Table 5-1.
Table 5-1 Application Names
Application Name
|
Process Name
|
Polycom PVX version 8.x
|
Vvsys.exe
|
Cisco Agent Desktop version 4.5(2)
|
Agent.exe
|
For detailed information about the proper QoS markings required for an end-to-end QoS implementation for this solution, please see the QoS section of this document.
QoS Configuration—Not Covered
MPLS VPN configurations are not described as these setting are provided as part of the managed service from the service provider.
SIP PSTN Gateway requires QoS settings for the traffic on the IP leg of the call. These can be defined using class-maps that map both signaling and voice bearer traffic.
Access Security
Access security has several features. The three features recommended for Collaborative Care are:
•
Port Security—Applied to each interface that is user facing
•
DHCP Snooping—Global command
•
Dynamic Arp Inspection (DAI)—Applied to each interface that is user facing
Show below is part of the IOS switch configuration with these three features enabled.
no ip dhcp snooping information option
ip arp inspection vlan 80
spanning-tree extend system-id
vlan internal allocation policy ascending
interface FastEthernet1/0/48
switchport access vlan 80
switchport port-security maximum 10
switchport port-security aging time 2
switchport port-security aging type inactivity
Additionally for Deployment Models 2 and 3
For deployment model 2 and 3, the addition of site-to-site communication that traverses two different business operations involves extra configuration that is not required for deployment model 1. This section summarized the configuration details required in both deployment model 2 and 3:
•
ASA configuration for ACL FW and NAT/PAT
•
CallManager Location configuration from Deployment Model 2 and Deployment Model 3
•
CallManager SIP Trunk configuration for CallManager to CallManager connections between sites
ASA Configuration ACL FW and NAT/PAT Configuration
This section provides a sample configuration for the ASA FW to protect traffic entering a site, IP address mapping between public and private addresses using NAT, and IP address overloading using PAT if the number of public IP addresses is less than the number of video calls made from a private IP address. This ASA FW configuration should be applied to each site.
Key steps are:
Step 1
access_list for what is allowed.
Step 2
policy map for type of traffic to inspect for embedded IP address such as the IP endpoint addresses and ports embedded into SIP messages across the trunk.
Step 3
Apply 1 to the interfaces and apply 2 globally.
Step 4
Define the NAT and PAT addresses and ranges.
Sample Configuration from a Cisco ASA
interface GigabitEthernet0/0
ip address 172.21.52.122 255.255.255.252
interface GigabitEthernet0/1
ip address 172.21.52.110 255.255.255.240
access-list acl_out extended permit tcp any any eq H.323
access-list acl_out extended permit udp any any eq 1718
access-list acl_out extended permit udp any any eq 1719
! allow SIP traffic for SIP trunk communication between CallManagers
access-list acl_out extended permit tcp any any eq sip
access-list acl_out extended permit tcp any any eq 8080
access-list acl_out extended permit tcp any any eq ssh
logging buffer-size 1048576
logging buffered debugging
logging asdm informational
no logging message 106023
icmp unreachable rate-limit 1 burst-size 1
asdm image disk0:/asdm512-k8.bin
! PAT table for outside address 10.95.2.21 mapping to internal address in the range of
172.21.52.64/28
global (outside) 1 10.95.2.21 netmask 255.255.255.240
nat (inside) 1 172.21.52.64 255.255.255.240
!NAT table for outside to inside address
static (inside,outside) 172.21.61.45 172.21.52.70 netmask 255.255.255.255
static (inside,outside) 172.21.61.47 172.21.52.66 netmask 255.255.255.255
! applies the acl_out to the outside interface
access-group acl_out in interface outside
route outside 0.0.0.0 0.0.0.0 172.21.52.113 3
timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02
timeout sunrpc 0:10:00 H.323 0:05:00 h225 1:00:00 mgcp 0:05:00 mgcp-pat 0:05:00
timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00
timeout uauth 0:05:00 absolute
http 192.168.1.0 255.255.255.0 management
snmp-server enable traps snmp authentication linkup linkdown coldstart
class-map inspection_default
match default-inspection-traffic
Cisco CallManager Locations
The location definitions vary depending on the deployment model. For deployment model 1, refer to the CallManager configuration section for Locations.
For Deployment Model 2, follow these procedures. Each site should have a uniquely-defined location that groups the phones. Separately, each site should have a location for each link of a SIP trunk. The bandwidth allocated for the voice and video resource pool should be designed to fit the data connection that the SIP trunk uses for site-to-site communication.
Note
Prior to proceeding with this procedure, carefully calculate the total available bandwidth between these two sites and allocate a percentage of that bandwidth to this service.
Step 1
Choose System > Locations.
Step 2
Select Add New to create a location. See the design section for methods to build locations.
Step 3
Under Name, give a name for this location (for example, HospitalA_Main_Campus_Phones).
Step 4
Under Audio Calls Information, enter unlimited if bandwidth is not an issue or a value as a derivative of 64kbps. An example would be 512 kbps, which supports eight simultaneous G.722 voice calls.
Step 5
Under Video Calls Information, enter None if no video call are allowed, Unlimited if bandwidth is a non issue, or enter a value in kbps that is a derivative of the bandwidth use for video calls, which is 768kpbs. An example would be 2304, which supports three simultaneous video calls inclusive of the voice bandwidth.
Step 6
Under the RSVP Setting, choose No Reservation as RSVP is not used.
To create the location for the SIP trunk that connects to another business, repeat steps 1-2 then proceed with the following:
Step 1
Under Name, give a name for this location (for example, SIP_trunk_to_HospitalB).
Step 2
Under Audio Calls Information, enter unlimited if bandwidth is not an issue or a value as a derivative of 64kbps. An example would be 512 kbps, which supports eight simultaneous G.722 voice calls.
Step 3
Under Video Calls Information, enter None if no video call are allowed, Unlimited if bandwidth is a non issue, or enter a value in kbps that is a derivative of the bandwidth use for video calls which is 768kpbs. An example would be 2304, which supports three simultaneous video calls inclusive of the voice bandwidth.
Step 4
Under the RSVP Setting, choose No Reservation as RSVP is not used.
For Deployment Model 3, follow these procedures. This model is limited to control the resource management defined on the CallManager at the LIS. This model is not aware of the CallManager at the various hospitals that this LIS supports. Each site that the LIS supports should have a unique location defined. In this location, the video endpoints and the SIP trunk should be associated with the same location.
Note
Prior to proceeding with this procedure, carefully calculate the total available bandwidth between each site that the LIS supports and allocate a percentage of that bandwidth to this service.
Procedure at LIS CallManager for Hospital A:
Step 1
Choose System > Locations.
Step 2
Select Add New to create a location. See the design section for methods to build locations.
Step 3
Under Name, give a name for this location (for example, HospitalA_Main_Campus_Phones).
Step 4
Under Audio Calls Information, enter unlimited if bandwidth is not an issue or a value as a derivative of 64kbps. An example would be 512 kbps, which supports eight simultaneous G.722 voice calls.
Step 5
Under Video Calls Information, enter None if no video call are allowed, Unlimited if bandwidth is a non issue, or enter a value in kbps that is a derivative of the bandwidth use for video calls which is 768kpbs. An example would be 2304, which supports three simultaneous video calls inclusive of the voice bandwidth.
Step 6
Under the RSVP Setting, choose No Reservation as RSVP is not used.
Procedure at LIS CallManager for Hospital B:
Step 1
Choose System > Locations.
Step 2
Select Add New to create a location. See the design section for methods to build locations.
Step 3
Under Name, give a name for this location (for example, HospitalB_Main_Campus_Phones).
Step 4
Under Audio Calls Information, enter unlimited if bandwidth is not an issue or a value as a derivative of 64kbps. An example would be 960 kbps, which supports 15 simultaneous G.722 voice calls.
Step 5
Under Video Calls Information, enter None if not video call are allowed, Unlimited if bandwidth is a non issue, or enter a value in kbps that is a derivative of the bandwidth use for video calls which is 768kpbs. An example would be 3840, which supports five simultaneous video calls inclusive of the voice bandwidth.
Step 6
Under the RSVP Setting, choose No Reservation as RSVP is not used.
Cisco CallManager SIP Trunk
To support inter-site calls between two CallManagers that are operated by two different businesses, the connections require an IP data connection with a SIP call control protocol between the two sites. This section provides the steps to configure a SIP connection from a CallManager.
Step 1
Choose Device > Trunk.
Step 2
Select Add New.
Step 3
Under Trunk Type, select SIP Trunk.
Step 4
Under Device Protocol, select SIP and click Next.
Step 5
Under Device Name, enter a name that represents the SIP trunk connection to another CallManager (for example, SIP_T_Hospital_B).
Step 6
Under Description, provide a brief description of this connection.
Step 7
Under Device Pool, select the device pool that supports video (for example, Video Device Pool).
Step 8
Under Class Classification, choose Use System Default.
Step 9
Depending on the deployment model:
•
For Deployment Model 2—Under Location, select a location that is uniquely defined and does not include the video endpoints in this location group.
•
For Deployment Model 3—Under Location, select a location includes the video phones for this location.
Step 10
Under Packet Capture Mode, select None.
Step 11
The radio button for Media Termination Point Required should be unchecked.
Step 12
The radio button for Retry Video Call as Audio should be checked.
Step 13
Under SIP Information > Destination Address, enter the IP address of the CallManager with which this SIP trunk communicates.
Step 14
Under SIP Information > Destination Port, enter 5060.
Step 15
Under SIP Information > DTMF Signalling Method, choose OOB and RFC2833 since OOB is key to interworking with UCCX.
Step 16
For the remaining parameters used to define a SIP trunk, following the configuration recommendations as outlined in the CM 5.1 Administration Guide:
•
Cisco Unified CallManager Release 5.1(1) New and Changed Information Guide http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_chapter09186a008073ee44.html
•
Cisco Unified CallManager Administration Guide, Release 5.0(4) http://www.cisco.com/en/US/partner/products/sw/voicesw/ps556/products_administration_guide_book09186a008066fa60.html
MPLS VPN
MPLS VPN configurations are not covered in this document. This setting is determined by the MPLS VPN service from the Managed Provider. Align the MPLS implementation with the design guidelines provided in the QoS design section.