Table Of Contents
Cisco Unified Border Element for Contact Center Solutions
Cisco Unified Communications Manager
Cisco Unified Customer Voice Portal Call Server
Cisco Unified Intelligent Contact Management Enterprise
Cisco Unified Customer Voice Portal VoiceXML Studio
Cisco Unified Customer Voice Portal VoiceXML Server
Nuance Automatic Speech Recognition/Text-To-Speech Server
Centralized Model: SIP-to-H.323 Protocol Translation
Distributed Model: SIP-to-H.323 Protocol Translation
Cisco Unified Communications Manager-Originated Calls
Cisco Unified Border Element for Contact Center Solutions
First Published: December 17, 2007, OL-14676-01
Last Updated: February 25, 2009Contents
Introduction
This document provides a sample configuration for contact center solutions using Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway software). Cisco Unified Border Element facilitates connectivity among independent Cisco Unified Communications, VoIP, and video networks. Cisco Unified Border Element also provides session border controller (SBC) functions such as services demarcation, call admission control (CAC), protocol translation, and security.
This document describes the components used in the network, and provides sample device configurations that have been tested for the described features. Use this document for deploying a Cisco Unified Border Element contact center solution.
Prerequisites
Requirements
Ensure that you are using the Cisco IOS Release 12.4(15)XY software before you attempt this configuration. The contact center solution using Cisco Unified Border Element described in this document was tested using the Cisco IOS Release 12.4(15)XY software. See the "Caveats" section for more information.
Components Used
The information in this guide is based on the software and hardware versions identified and described in the following sections:
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Cisco Unified Presence Server
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The information in this document was created through the use of the devices in a specific lab environment. All of the devices used in this document started with a cleared (default) configuration. If your network is live, make sure that you understand the potential impact of any command.
Cisco IOS VoiceXML Browser
A Cisco Integrated Services Router (Cisco ISR) and a Cisco AS5000XM are used in this solution for Cisco IOS VoiceXML browser services.
Cisco Unified Presence Server
Cisco Unified Presence Server is used in this solution only as a Session Initiation Protocol (SIP) proxy. The presence features are not configured or utilized. The SIP proxy function provides efficient and accurate routing of both presence and general SIP messaging through the enterprise. Any standards-based SIP proxy server should be able to provide the same functionality as the Cisco Unified Presence Server.
The following Cisco Unified Presence Server versions are used in this solution:
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Cisco Unified Communications Manager
Cisco Unified Communications Manager is an integral part of IP-based contact centers where the agents are using Cisco Unified IP Phones. Cisco Unified Communications Manager is not required if the contact center is time-division multiplexing (TDM)-based. The solution discussed in this document uses SIP trunks, H.323 trunks, and Cisco Unified Communications Manager version 5.0(1).
Cisco Unified Border Element
Cisco Unified Border Element is fully interoperable with the Cisco Unified Communications Manager and the Cisco Unified Communications Manager Express. The Cisco Unified Border Element serves as a feature-rich demarcation point for connecting enterprises to service providers over Unified Communications trunks, including SIP trunks. When interoperating with the Cisco Unified Customer Voice Portal, the Cisco Unified Border Element plays the principal role of receiving calls through provisioned SIP trunks and routing the calls to agent phones. For interactive voice response (IVR) treatment, the Cisco Unified Border Element can also provide a Voice Extensible Markup Language (VoiceXML) gateway. The Cisco Unified Communications Manager-originated calls are routed through Cisco Unified Border Element in both H.323-to-H.323 and SIP-to-SIP deployments. The Cisco IOS Release 12.4(15)XY software was used on Cisco Unified Border Element. More features are supported by later Cisco IOS Releases.
Cisco Gatekeeper
The gatekeeper is a network element used by H.323 gateways for call routing. It is an optional component in this solution. However, most H.323 installations incorporate an H.323 gatekeeper for dial plan configuration and bandwidth management.
One of the scenarios for using a gatekeeper with Cisco Unified Customer Voice Portal is for mapping specific dialed numbers to specific Cisco Unified Customer Voice Portal Call Servers or specific Cisco Unified Customer Voice Portal VoiceXML gateways. Another scenario for using a gatekeeper with Cisco Unified Customer Voice Portal is for routing the transfer of callers from a VoiceXML gateway port to a Cisco Unified IP Phone.
Cisco Unified Customer Voice Portal Call Server
The Cisco Unified Customer Voice Portal Call Server implements a SIP back-to-back user agent (B2BUA), H.323 service, and IVR service. The Cisco Unified Customer Voice Portal Call Server is a central component in a Cisco Unified Intelligent Contact Management Enterprise-integrated Cisco Unified Customer Voice Portal implementation. It is located logically between the Cisco IOS gateway and Cisco Unified Intelligent Contact Management Enterprise's Voice Response Unit (VRU) Peripheral Gateway (PG). The Cisco Unified Customer Voice Portal Call Server is not a required component for Standalone deployment models where call routing is not a desired. Cisco Unified CVP Server 4.0(2) was used.
Cisco Unified Intelligent Contact Management Enterprise
Cisco Unified Intelligent Contact Management Enterprise is a mandatory component for advanced call control requiring, for example, IP switching and transfer to agents. Cisco Unified Intelligent Contact Management Enterprise provides call center agent management capabilities and call switching capabilities. This solution uses Cisco Unified Intelligent Contact Management Enterprise version 7.1.2.
Cisco Unified Customer Voice Portal VoiceXML Studio
The Cisco Unified Customer Voice Portal VoiceXML Studio is an Integrated Development Editor that is based on the open-source Eclipse framework. Cisco Unified Customer Voice Portal VoiceXML Studio provides a graphical user interface for designing complex IVR scripts, as well as a palate containing a rich set of standard scripting elements. This solution uses Cisco Unified Customer Voice Portal VoiceXML Studio version 4.0(2).
Cisco Unified Customer Voice Portal VoiceXML Server
The Cisco Unified Customer Voice Portal VoiceXML Server delivers external VoiceXML documents to the Cisco IOS gateway. Voice applications are written using the Cisco Unified Customer Voice Portal VoiceXML Studio and then deployed to the Cisco Unified Customer Voice Portal VoiceXML Server.
As calls invoke external VoiceXML, the Cisco Unified Customer Voice Portal VoiceXML Server builds VoiceXML pages dynamically according to the contents of the deployed application. The Cisco Unified Customer Voice Portal VoiceXML Server interacts only with the Cisco IOS gateway. The gateway requests the VoiceXML document from the Cisco Unified Customer Voice Portal VoiceXML Server upon command from the Cisco Unified Customer Voice Portal Call Server.
This solution uses Cisco Unified Customer Voice Portal VoiceXML Server version 4.0(2).
Nuance Automatic Speech Recognition/Text-To-Speech Server
The Nuance Automatic Speech Recognition (ASR)/Text-To-Speech (TTS) Server provides speech recognition services and text-to-speech services for the Cisco IOS VoiceXML gateway. Communication between the ASR/TTS server(s) and the Cisco IOS VoiceXML gateway uses Media Resource Control Protocol (MRCP). The Cisco Unified Border Element-Cisco Unified Customer Voice Portal interoperability scenarios used MRCP version 2 and the following Nuance versions:
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Cisco Unified SIP Proxy
Cisco Unified SIP Proxy can be used in this solution as a SIP proxy and as an alternative to Cisco Unified Presence Server. Cisco Unified SIP Proxy runs on a network module that deploys on the Cisco 3800 Series ISR platform. Cisco Unified SIP Proxy eliminates the need for a standalone SIP proxy server. It is highly configurable, has flexible routing and normalization policies, is highly scalable and also supports features like percentage routing and time of day routing. This solution uses Cisco Unified SIP Proxy version 1.1. The Cisco 3800 Series ISR router hosting Cisco Unified SIP Proxy was running Cisco IOS Release 12.4(22)T.
Conventions
Refer to Cisco Technical Tips Conventions for information on document conventions.
Background Information
This document describes two models of contact center solutions that include Cisco Unified Border Element: centralized and distributed. In the centralized model, VoiceXML runs on a separate Cisco Integrated Services Router (Cisco ISR) or Cisco AS5000XM independent of Cisco Unified Border Element. In the distributed model, VoiceXML runs on the same Cisco IOS gateway on which Cisco Unified Border Element runs.
To configure a contact center solution with Cisco Unified Border Element, you should understand the following concepts:
Call Flows
This section describes the following call flows for the distributed and centralized models of the contact center solutions using Cisco Unified Border Element:
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Centralized Model: SIP-to-H.323 Protocol Translation
Figure 1 depicts the centralized model, with H.323-to-SIP protocol translation. In Figure 1, the SIP trunk is brought into the enterprise without upgrading Cisco Unified Communications Manager or Cisco Unified Customer Voice Portal to SIP. That is, the topology shown in Figure 1 can include versions of Cisco Unified Customer Voice Portal earlier than version 4.0 that do not support SIP and versions of Cisco Unified Communications Manager earlier than version 5.0 that do not support SIP trunking.
Figure 1 SIP-to-H.323 Protocol Translation Centralized VoiceXML and Cisco Unified Border Element
In the centralized model, Cisco Unified Border Element is located centrally in the data center, together with Cisco Unified Communications Manager and Cisco Unified Customer Voice Portal. Cisco Unified Border Element translates SIP from the service provider's network to H.323 for Cisco Unified Communications Manager and Cisco Unified Customer Voice Portal. VoiceXML runs on a router separate from that of the Cisco Unified Border Element. The Cisco Unified Border Element router is dedicated to terminating the SIP trunk from the service provider.
Distributed Model: SIP-to-H.323 Protocol Translation
Figure 2 depicts the distributed model, with H.323-to-SIP protocol translation. As Figure 2 shows, the SIP trunk is brought into the enterprise without upgrading either Cisco Unified Communications Manager or Cisco Unified Customer Voice Portal to SIP. That is, the topology shown in Figure 2, can include versions of Cisco Unified Customer Voice Portal earlier than version 4.0 that do not support SIP and versions of Cisco Unified Communications Manager earlier than version 5.0 that do not support SIP trunking.
Figure 2 SIP-to-H.323 Protocol Translation in the Distributed Model
In the distributed model, Cisco Unified Border Element is located in the remote or branch office which the SIP trunk enters. Cisco Unified Border Element translates SIP from the service provider's network to H.323 for Cisco Unified Communications Manager and Cisco Unified Customer Voice Portal. VoiceXML runs on the same router as the Cisco Unified Border Element.
Cisco Unified Communications Manager-Originated Calls
Figure 3 shows the call flow for calls originated by Cisco Unified Communications Manager:
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Figure 3 Cisco Unified Communications Manager-Originated Calls: SIP-to-SIP or H.323-to-H.323
Note these points about the call flow in Figure 3:
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Feature Summary
The features described in this section were tested as part of the solution configuration.
Cisco IOS VoiceXML Browser
Applications written in VoiceXML provide access through a voice browser to content and services over the telephone, just as HTML provides access through a web browser running on a PC. The universal accessibility of the telephone and its ease of use make VoiceXML applications a powerful alternative to HTML for accessing the information and services of the World Wide Web.
The Cisco IOS VoiceXML feature provides a platform for interpreting VoiceXML documents. When a telephone call is made to the Cisco IOS VoiceXML-enabled gateway, VoiceXML documents are downloaded from the Cisco Unified Customer Voice Portal servers, providing content and services to the caller, typically in the form of prerecorded audio in an IVR application. You can access online business applications over the telephone that provide, for example, stock quotes, sports scores, or bank account balances.
VoiceXML brings the advantages of web-based development and content delivery to voice applications. VoiceXML is similar to HTML in its simplicity and in its presentation of information. The Cisco IOS VoiceXML feature is based on the VoiceXML 2.1 W3C Candidate Recommendation (June 13, 2005) and is designed to provide web developers great flexibility and ease in implementing VoiceXML applications.
The Cisco IOS VoiceXML browser is central to this solution. Depending on the model being used, the VoiceXML content is interpreted by the Cisco IOS VoiceXML browser on the Cisco Unified Border Element, in the distributed model, or on a separate Cisco IOS VoiceXML gateway, in the centralized model. Both models were tested.
DTMF Interworking
Dual tone multifrequency (DTMF) is important in the contact center because many interactions require the caller to enter DTMF input using the phone keypad. DTMF can be carried out-of-band in the signaling path or in-band in the bearer path. Various DTMF methods exist, and it is important for any solution to interwork reliably among these methods.
The following DTMF relay methods were tested:
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Transcoding
The transcoding feature enables the Cisco Unified Border Element to link two networks using dissimilar codecs. Transcoding may be required in order to conserve bandwidth across a WAN link. For example, G.711 u-law packets consume more bandwidth than G.729r8 packets, and you may need to perform this conversion before traversing a bandwidth-limited WAN link. Transcoding may be required also in cases in which certain applications support only a certain codec type.
In the contact center space, the agents are typically G.711 u-law-or G.729r8-based. However, calls that are inbound via the SIP trunk may be using a different codec. The G.711 u-law, G.729r8, and internet Low Bitrate Codec (iLBC) are the common types. Calls using different codecs than the agents use would necessitate transcoding on the Cisco Unified Border Element.
Transcoding of G.711 u-law to G.729r8 codecs has been tested and is supported. The iLBC codecs are supported also.
GTD Pass-Through
Generic Transparency Descriptor (GTD) is a mechanism for representing telephony signaling messages and parameters in a generic fashion, in particular ISDN User Part (ISUP) messages and parameters. Mapping of every individual ISUP message and parameter to an equivalent H.323 or SIP message and parameter is not viable. The GTD offers a solution in which the ISUP messages and parameters can be represented in a generic fashion and transported by the underlying call signaling messages to each node transited by the call. GTD is supported only across SIP networks that support Multipurpose Internet Mail Extension (MIME) header encapsulation.
In the contact center space, the ISDN user-to-user information (UUI) element has special significance and needs to be passed to the Intelligent Contact Manager (ICM) Enterprise script. The UUI data is passed via the Cisco IOS gateway in the SIP INVITE when signaling forwarding is configured. The INVITE contains the extra section of GTD ISDN variables, and the Cisco Unified Customer Voice Portal SIP subsystem parses the inbound body to send these across the GED-125 API in the NEW CALL message in the "UserToUser" field.
The testing described in this guide focused only on ensuring that the GTD information received on the inbound leg on the Cisco Unified Border Element was passed to the outbound leg via the Session Description Protocol (SDP) of the outbound INVITE. No attempt was made to verify that the UUI data actually reached the ICM script, and no attempt was made to determine how it was used by the script.
HTTPS
With the Secure HTTP (HTTPS) feature, you can use a secure socket interface to load VoiceXML application documents and scripts and play audio prompt files from the HTTP server. In data exchanges between the HTTP server and the client, data is encrypted by one party before it is sent over the network and then decrypted by the other party when it is received. The encryption protection makes eavesdropping much more difficult to decipher, if not impossible.
HTTPS does affect the gateway's performance because of the additional overhead involved in processing secure transactions.
HTTPS was tested for two standard models:
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Both the Cisco Unified Customer Voice Portal VoiceXML Server and the Cisco Unified Customer Voice Portal Call Server support HTTPS connections.
Standalone Model
The Cisco Unified Customer Voice Portal standalone deployment model provides IVR services only. Intelligent contact center integration is not included, although it is possible to transfer the call to an arbitrary destination, without call context and without queuing. The standalone model requires Cisco Unified Customer Voice Portal VoiceXML servers and Cisco IOS gateways, but does not include the Cisco Unified Customer Voice Portal Call Server or any Cisco Unified ICM Enterprise components. IVR applications support either DTMF only or a combination of DTMF and ASR and TTS.
In addition to testing basic functionality of the standalone model, two types of transfers were also tested: blind and bridged. VoiceXML blind transfers result in the call being bridged to an egress voice gateway or a VoIP endpoint, and the Cisco Unified Customer Voice Portal VoiceXML server releases all subsequent call control. VoiceXML-bridged transfers result in the call being bridged to an egress voice gateway or a VoIP endpoint, but the VoiceXML server retains call control so that it can return a caller to an IVR application or transfer the caller to another termination point. VoiceXML blind and bridged transfers are invoked using the transfer element in Cisco Unified Customer Voice Portal VoiceXML Studio. VoiceXML transfers transfer the call to any dial peer that is configured in the gateway.
Caveats
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Configure
This section provides the information you need to configure the features described in this document.
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Network Diagram
This document uses this network setup.
Figure 4 Contact Center Solution with Cisco Unified Border Element Network Topology
Configurations
This document uses these configurations:
Distributed Model
The following sections provide configurations of the key devices in the contact center solution using Cisco Unified Border Element distributed model in which VoiceXML runs on the same Cisco IOS gateway as Cisco Unified Border Element.
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Cisco Unified Border Element
The following is the configuration of the Cisco Unified Border Element in the distributed model. Significant sections of this output are shown in bold type for emphasis.
CUBE# show running-configBuilding configuration...Current configuration : 16233 bytes!! Last configuration change at 19:05:36 PST Fri Sep 28 2007!version 12.4service timestamps debug datetime msec localtimeservice timestamps log datetime msec localtimeno service password-encryption!hostname Router!boot-start-markerno boot startup-testboot-end-marker!logging buffered 10000000no logging consoleenable password lab!!!resource-pool disableno aaa new-modelclock timezone PST -7voice-card 2dsp services dspfarm!voice-card 3dsp services dspfarm!ip cefip domain name example.comip host gtd-uut 10.2.175.5ip host nuance-host 10.1.36.102ip host nuance-pchittap.example.com 10.1.36.102ip host tts-server 10.2.174.53ip host asr-server 10.2.174.53ip host tts-en-us 10.2.174.53ip host asr-en-us 10.2.174.53ip name-server 10.1.34.1!!multilink bundle-name authenticated!voice rtp send-recv!voice service voiph323emptycapability! This command enables the Cisco Unified Border Element! to process calls for both H.323 call legs.allow-connections h323 to h323allow-connections h323 to sipallow-connections sip to h323! This command enables the Cisco Unified Border Element! to process calls between SIP endpoints.allow-connections sip to sip! This command is required if GTD parameters are to be passed between in and out legs.signaling forward unconditionalsiprel1xx disableheader-passing!!voice class uri NUANCE sippattern NUANCE@nuance-host!!voice class codec 1codec preference 1 g711ulawcodec preference 2 g729r8!!voice translation-rule 1rule 1 /6505552.../ /71006/!!voice translation-profile myprofiletranslate called 1!!ivr prompt memory 15000ivr prompt streamed none!applicationservice collect_transfer_bridged flash:SelfService.tclparamspace english language enparamspace english index 0paramspace english location flashparam SelfService-port 7000paramspace english prefix enparam SelfService-app collect_transfer_bridgedparam PrimaryVXMLServer 10.1.34.2!service collect_transfer flash:SelfService.tclparamspace english index 0paramspace english language enparamspace english location flashparam SelfService-app collectdigits_transferparamspace english prefix enparam SelfService-port 7000param PrimaryVXMLServer 10.1.34.2!service new-call flash:bootstrap.vxmlparamspace english index 0paramspace english language enparamspace english location flash:paramspace english prefix en!service Secure flash:SelfService.tclparam SelfService-SSL 1param PrimaryVXMLServer 10.1.34.2paramspace english index 0paramspace english language enparamspace english location flashparamspace english prefix enparam SelfService-port 7443param SelfService-app Test_GD_DTMF_60sec!service SelfService flash:SelfServiceBootstrap.vxmlparamspace english language enparamspace english index 0paramspace english location flash:paramspace english prefix en!service ringtone flash:ringtone.tclparamspace english index 0paramspace english language enparamspace english location flashparamspace english prefix en!service Error flash:Error.tclparamspace english language enparamspace english index 0paramspace english location flashparamspace english prefix en!service bootstrap flash:bootstrap.tclparamspace english language enparamspace english index 0paramspace english location flashparam serverssl 1paramspace english prefix en!service handoff flash:handoff.tclparamspace english index 0paramspace english language enparamspace english location flash:paramspace english prefix en!mrcp client timeout connect 20vxml tree memory 128vxml version 2.0!!crypto pki trustpoint myCAenrollment terminalrevocation-check crl! This command is required only if you are using HTTPS.crypto pki trustpoint myCallServerenrollment terminalrevocation-check crlre!!crypto pki certificate chain myCAcertificate ca 13EDDCFA00000000000330820409 308202F1 A0030201 02020A13 EDDCFA00 00000000 03300D06 092A864886F70D01 01050500 300E310C 300A0603 55040313 03435650 301E170D 30373032...16CB009E E2F58EC0 DF92BC96 32FD483B 7C3B1D8E 230B7E89 44491F9C F65895277E13EFB2 AB9BE2B2 9EB94203 B9quitcrypto pki certificate chain myCallServercertificate ca 1946D58200000000000430820414 308202FC A0030201 02020A19 46D58200 00000000 04300D06 092A864886F70D01 01050500 300E310C 300A0603 55040313 03435650 301E170D 30373037...6545A91F 324C57A2 2A51AFCC 1DBA696D DCD7D941 1939D8CC FD8082F3 AFE07C647EA094FA 3163160F E624A634 B809CB7E 93DBDB21 BEBA1FEBquit!archivelog confighidekeys!!interface GigabitEthernet0/0ip address 10.1.34.1 255.255.0.0duplex fullspeed 100negotiation autono keepalive!interface GigabitEthernet0/1ip address 10.3.29.250 255.255.0.0duplex fullspeed 100negotiation autono keepalive!interface Serial0/0no ip addressshutdownclock rate 2000000no fair-queue!interface Serial0/1no ip addressshutdownclock rate 2000000!ip route 10.2.0.0 255.255.0.0 1.1.0.1ip route 10.4.0.0 255.255.0.0 1.1.0.1ip route 10.86.129.31 255.255.255.255 172.16.146.1ip route 172.16.0.0 255.0.0.0 172.16.146.1ip route 192.255.254.0 255.255.255.0 10.1.0.1!no ip http serverno ip http secure-server!!snmp-server community cisco RW!sccp local GigabitEthernet0/0sccp ccm 10.1.34.1 identifier 1 version 5.0.1sccp!sccp ccm group 1associate ccm 1 priority 1associate profile 1 register MTP000bbecadcf1keepalive retries 5! This block of commands is required for transcoding on the Cisco Unified Border Element! only and only on Cisco 5xxx platforms. Cisco 3xxx platforms do not require this! configuration.switchover method immediateswitchback method immediate!dspfarm profile 1 transcodecodec g711ulawcodec g711alawcodec g729ar8codec g729abr8codec g729r8maximum sessions 144associate application SCCP!!!dial-peer voice 200 voipdescription Incoming dial-peer for all callsincoming called-number 6505552...dtmf-relay rtp-nteno vad!dial-peer voice 600 voipdescription Send call to 3825 ISRshutdowndestination-pattern 6505552001session protocol sipv2session target ipv4:10.1.34.20dtmf-relay rtp-ntecodec g711ulawno vad!dial-peer voice 700 voipdescription Send Call to Cisco Unified Presence Server! The DNIS is translated to 71006 before sending call to CUPS.translation-profile outgoing myprofiledestination-pattern 6505552...session protocol sipv2session target ipv4:10.1.34.7dtmf-relay rtp-ntecodec g711ulawno vad!!dial-peer voice 9292 voipdescription SIP Errorservice Errorincoming called-number 92Tcodec g711ulaw!dial-peer voice 9191 voipdescription SIP Ringtoneservice ringtoneincoming called-number 91Tcodec g711ulaw!dial-peer voice 888999 voipservice bootstrapincoming called-number 888999T! The dial-peer tag "888999" corresponds to the LABEL configured on the ICM. This! label is known as the "VRU transfer label" and is returned to the CUCVP call server! (and eventually to the CUBE via the CUPS) by the ICM for VRU treatment.dtmf-relay rtp-ntecodec g711ulaw!dial-peer voice 100 voipdescription calls VoiceXML server applicationservice test_gd_dtmfshutdownincoming called-number 71075dtmf-relay rtp-ntecodec g711ulaw!!sip-ua!!gatekeepershutdown!!telephony-servicesdspfarm units 5sdspfarm transcode sessions 20sdspfarm tag 1 MTP000bbecadcf1! This command is required only for transcoding on Cisco AS5xxx series routers.max-ephones 6ip source-address 10.1.34.1 port 2000max-conferences 4 gain -6call-forward system redirecting-expandedtransfer-system full-consult!ss7 mtp2-variant Bellcore 0ss7 mtp2-variant Bellcore 1ss7 mtp2-variant Bellcore 2ss7 mtp2-variant Bellcore 3!line con 0exec-timeout 0 0logging synchronousstopbits 1line aux 0stopbits 1line vty 0 4password lablogin!scheduler allocate 10000 400ntp clock-period 17179937ntp update-calendarntp server 10.3.29.249endStatic Routes on Cisco Unified Presence Server
Figure 5 shows the static routes that are configured on the Cisco Unified Presence Server in the distributed model.
Figure 5 Static Routes on Cisco Unified Presence Server
In Figure 5 note the following destination patterns:
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Cisco Unified Communications Manager Configuration
On Cisco Unified Communications Manager, configure the SIP trunks to the Cisco Unified Presence Server as shown in Figure 6. The agent phones should be associated with the appropriate configured application user.
Figure 6 Cisco Unified Communications Manager Express Configuration
Figure 7 shows the SIP information configuration on Cisco Unified Communications Manager.
Figure 7 Cisco Unified Communications Manager Express Configuration
Centralized Model
The following are the configurations of the key devices in the contact center solution using the Cisco Unified Border Element centralized model in which VoiceXML runs on a separate Cisco IOS gateway that is independent of Cisco Unified Border Element.
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Cisco Unified Border Element
The following is the configuration of the Cisco Unified Border Element in the centralized model. Significant sections of this output are shown in bold type for emphasis.
VoiceXMLGateway# show running-configBuilding configuration...Current configuration : 7201 bytes!version 12.4service timestamps debug datetime msecservice timestamps log datetime msecno service password-encryption!hostname VoiceXMLGateway!boot-start-markerboot system flash:c3825-adventerprisek9-mz.124-15.T1.fc2boot-end-marker!!card type command needed for slot 1logging buffered 1000000enable password lab!no aaa new-modelclock timezone PST -7no network-clock-participate slot 1ip cef!!!!ip domain name example.comip host nuance-host 10.1.36.102ip host nuance-pchittap.example.com 10.1.36.102ip host tts-en-us 10.2.174.53ip host asr-en-us 10.2.174.53ip name-server 10.1.34.20!multilink bundle-name authenticated!voice-card 0no dspfarm!voice-card 1no dspfarm!!!!voice service voipallow-connections sip to sipsignaling forward unconditionalsiprel1xx disableheader-passing!!voice class uri NUANCE sippattern NUANCE@nuance-host!!ivr prompt memory 15000ivr prompt streamed all!applicationservice new-call flash:bootstrap.vxmlparamspace english language enparamspace english index 0paramspace english location flash:paramspace english prefix en!service SelfService flash:SelfServiceBootstrap.vxmlparamspace english index 0paramspace english language enparamspace english location flash:paramspace english prefix en!service Error flash:Error.tclparamspace english index 0paramspace english language enparamspace english location flashparamspace english prefix en!service bootstrap flash:bootstrap.tclparamspace english language enparamspace english index 0paramspace english location flashparam serverssl 1paramspace english prefix en!service handoff flash:handoff.tclparamspace english language enparamspace english index 0paramspace english location flash:paramspace english prefix en!!vxml tree memory 128vxml version 2.0!crypto pki trustpoint myCallServerenrollment terminalrevocation-check crl! This command is required only if using HTTPS.!crypto pki certificate chain myCallServercertificate ca 1946D58200000000000430820414 308202FC A0030201 02020A19 46D58200 00000000 04300D06 092A864886F70D01 01050500 300E310C 300A0603 55040313 03435650 301E170D 30373037<snip>6545A91F 324C57A2 2A51AFCC 1DBA696D DCD7D941 1939D8CC FD8082F3 AFE07C647EA094FA 3163160F E624A634 B809CB7E 93DBDB21 BEBA1FEBquit!!archivelog confighidekeys!!interface GigabitEthernet0/0ip address 10.3.29.252 255.255.0.0shutdownduplex fullspeed 100media-type rj45!interface GigabitEthernet0/1ip address 10.1.34.20 255.255.0.0duplex fullspeed 100media-type rj45!ip route 10.2.0.0 255.255.0.0 10.1.0.1ip route 10.3.29.251 255.255.255.255 10.1.0.1ip route 192.255.254.0 255.255.255.0 10.1.0.1!!no ip http serverno ip http secure-server!!control-plane!!dspfarm profile 1 transcodeshutdown!dial-peer voice 888999 voipservice bootstrapincoming called-number 888999Tdtmf-relay rtp-ntecodec g711ulaw!sip-ua!line con 0exec-timeout 0 0logging synchronousstopbits 1line aux 0stopbits 1line vty 0 4password mypasswordloginline vty 5 15password mypasswordlogin!scheduler allocate 20000 1000ntp clock-period 17179969ntp server 10.3.29.249!webvpn cef!endStatic Routes on Cisco Unified Presence Server
Figure 8 shows the static routes that are configured on the Cisco Unified Presence Server in the centralized model.
Figure 8 Static Routes on Cisco Unified Presence Server
The 888999* entry points to the VoiceXML gateway whose IP address is 10.1.34.20 and not to the Cisco Unified Border Element. The VRU leg is established with the Cisco IOS VoiceXML gateway, that is, VoiceXML runs on a separate Cisco IOS gateway.
Cisco Unified Communications Manager Configuration
The Cisco Unified Communications Manager configuration for the centralized model is the same as that described for the distributed model in the "Cisco Unified Communications Manager Configuration" section.
Calls originated by Cisco Unified Communications Manager can use Cisco Unified Border Element and need the configuration on Cisco Unified Communications Manager shown in Figure 9 and Figure 10.
In Figure 9, Cisco Unified Border Element is added as an H.323 Cisco IOS gateway to the Cisco Unified Communications Manager for H.323-to-H.323 calls.
Figure 9 Cisco Unified Communications Manager
Figure 10 shows the route that is required on the Cisco Unified Communications Manager to route the calls through Cisco Unified Border Element.
Figure 10 Cisco Unified Communications Manager
For SIP-to-SIP calls, a SIP trunk needs to be created that points to the Cisco Unified Border Element gateway as the destination address, and a route pattern needs to be created to route the calls through the SIP trunk.
Verify
Use this section to confirm that your configuration work properly. To display and verify your configuration, use the following show commands:
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See the Cisco IOS Voice Command Reference for more information.
Troubleshoot
Use this section to troubleshoot your configuration.
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Use the following debug commands to troubleshoot your configuration:
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This command displays all the SIP service provider interface (SPI) messages. It traces the SIP messages exchanged between the gateway and other User Agents (UAs) and Cisco Unified Presence Server.
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This command displays all debugging messages for the HTTP client.
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This command displays all debugging messages for MRCP operations.
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These commands monitor HTTPS connections only. They do not produce any output for HTTP connections.
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Use this command to troubleshoot a VoiceXML application. The output from this command can be verbose.
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The command traces the execution path through the call control application programming interface (CCAPI), which serves as the interface between the call session application and the underlying network-specific software. You can use the output from this command to understand how calls are being handled by the voice gateway.
This command shows how a call flows through the system. At this debug level, you can see the call setup and teardown operations that are performed on the telephony and network call legs.
Related Information
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