Configuration Guide for Cisco Unified Customer Voice Portal, Release 10.0(1)
Gateway Configuration
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Gateway Configuration

Gateway Configuration

Configure Gateway

Procedure
    Step 1   Log in to Operations Console and click Device Management > Gateway.

    The Find, Add, Delete, Edit Gateways window opens.

    Step 2   Click Add New.
    Note   

    To use an existing Gateway as a template for configuring a new Gateway, select a Gateway from the list of available Gateways and click Use As Template and perform Steps 3 to 5.

    Step 3   Click the General tab, enter the field values, and click Save. See General Settings.
    Step 4   (Optional)Click the Device Pool tab, enter the field values, and click Save. See Add or Remove Device From Device Pool.
    Step 5   Click Save.
    Step 6   (Optional)If the call control client placed the Correlation ID in a GTD parameter other than uus.dat, specify the following parameters to configure a gateway to enable incoming UUI to be used as the Correlation ID.
                                conf t
                                application
                                service <your-cvp-service-name>
                                param use-uui-as-corrid Y (Refer to Note 1)
                                param correlation-gtd-attribute XXX (Refer to Note 2)
                                param correlation-gtd-instance N (Refer to Note 2)
                                param correlation-gtd-field YYY (Refer to Note 2)
                                dial-peer voice 123 pots
                                service <your-cvp-service-name>
                            

    Gateway Settings

    General Settings

    After adding an IOS Gateway, you can execute a subset of IOS Gateway commands on the Gateway from the Operations Console.

    The Ingress Gateway is the point at which an incoming call enters the Unified CVP solution. It terminates Time Division Multiplexing (TDM) phone lines on one side and implements VoIP on the other side. It also provides for sophisticated call routing capabilities at the command of other Unified solution components. It works with SIP and also supports Media Gateway Control Protocol (MGCP) for use with Unified CM.

    The VXML Gateway hosts the IOS voice browser, the component which interprets VXML pages from either the Unified CVP IVR service or the VXML Server, plays .wav files and Text-to-Speech (TTS), inputs voice and Dual Tone Multi Frequency (DTMF), and sends results back to the VXML requestor. It also mediates between Media Servers, Unified CVP VXML Servers, ASR and TTS Servers, and the interactive voice response (IVR) service.

    You can deploy the Ingress Gateway separately from the VXML Gateway, but in most implementations they are the same: one Gateway performs both functions. Gateways are often deployed in farms, for centralized deployment models. In Branch deployment models, one combined Gateway is usually located at each branch office.

    An Egress Gateway is typically used in Call Director model to provide access to a call center automatic call distributor (ACD) or third-party IVR.

    To configure General settings on a Gateway, on the General tab, enter the field values, as listed in the following table:

    Table 1 Unified ICM—General Tab Configuration Settings

    Field

    Description

    Default

    Value

    Restart Required

    IP Address

    The IP address of a Unified ICM Server

    None

    Valid IP address

    No

    Hostname

    The name of the Unified ICM Server

    None

    Valid DNS name. It includes alphanumeric characters and a dash.

    No

    Description

    Additional information of the Unified ICM Server

    None

    Up to 1024 characters

    No

    Device Admin URL

    The URL for the Unified ICM Web configuration application.

    None

    Valid URL

    No

    Activate Gateway Configuration

    Activate the gateway configuration by entering these commands:

    Procedure
      Step 1   call application voice load CVPSelfService
      Step 2   call application voice load HelloWorld

      Add Gateway to Device Pool

      Configure Gateway Settings for Standalone Call Flow Model

      After you configure a gateway through Operations Console, configure settings on the gateway.

      Procedure
        Step 1   All Versions: Transfer the following script, configuration, and .wav files using the Operations Console or through the Unified CVP CD:
        • CVPSelfService.tcl

          Note    This file contains a gateway configuration example.
        • CVPSelfServiceBootstrap.vxml

        • critical_error.wav

        1. Select Bulk Administration > File Transfer > Scripts and Media.
        2. From the Select device type drop-down list, select Gateway.
        3. Select the required file from the Available list, and click the right arrow to move the device to the Selected list.
        4. Click Transfer.
          Note    Ensure to check the transfer status after you click Transfer, because sometimes transfer may fail.
        Step 2   All Versions: Perform Step 2 of the Configure VXML Server Standalone Call Flow Model procedure.

        Example: Gateway Settings for Standalone Call Flow Model

        The first part of the following example provides the basic configuration for setting a VoiceXML Standalone gateway:

        • Applies a timestamp to debugging and log messages

        • Turns on logging

        • Turns off printing to the command line interface console

        • Sends RTP packets

        • Configures ASR/TTS Server

        • Configures gateway settings

        The last part (application) of this example provides the following information:

        • Standalone Service settings for hello_world application on the VXML Server

        • Service requirements for configuring self-service call flow models

        service timestamps debug datetime msec localtime
        service timestamps log datetime msec localtime
        !
        service internal
        logging buffered 99999999 debugging
        no logging console
        !
        ip cef
        !
        voice rtp send-recv
        
        ip host tts-en-us <IP of TTS or MRCP Server>
        ip host asr-en-us <IP of ASR or MRCP Server>
        
        voice class codec 1
        codec preference 1 g711ulaw
        codec preference 2 g729r8
        
        voice service voip
        signaling forward unconditional
        h323
        !
        gateway
        timer receive-rtcp 6
        !
        ip rtcp report interval 3000
        !
        ivr prompt memory 15000
        ivr prompt streamed none
        ivr asr-server rtsp://asr-en-us/recognizer
        ivr tts-server rtsp://tts-en-us/synthesizer
        
        mrcp client timeout connect 10
        mrcp client timeout message 10
        mrcp client rtpsetup enable
        rtsp client timeout connect 10
        rtsp client timeout message 10
        vxml tree memory 500
        http client cache memory pool 15000
        http client cache memory file 500
        http client connection timeout 60
        http client response timeout 30
        http client connection idle timeout 10
        
        application
        service hello_world flash:CVPSelfService.tcl
        param CVPPrimaryVXMLServer <ip address>
        param CVPBackupVXMLServer <ip address>
        param CVPSelfService-port 7000
        param CVPSelfService-SSL 0
        -OR-
        param CVPSelfService-port 7443
        param CVPSelfService-SSL 1
        param CVPSelfService-app HelloWorld
        
        service CVPSelfService
        flash:CVPSelfServiceBootstrap.vxml
        !
                                

        Note


        The optional param CVPSelfService-SSL 1 line enables HTTPS.
        Important: Calls may be rejected with a 403 Forbidden response if Toll Fraud security is not configured correctly. The solution is to add the IP address as a trusted endpoint, or else disable the IP address trusted list authentication altogether using the voice service voip -> "no ip address trusted authenticate" configuration entry.

        Example: Dial-Peer for Standalone Call Flow Model

        The following example provides the configuration for an incoming Pots and VoIP call for the VXML Server (standalone) call flow model:


        Note


        VXML Server (Standalone) supports an incoming call with a TDM through a T1 port only. Using an FXS port is not supported.
                                    
        dial-peer voice 8 pots                        
         description Example incoming POTS dial-peer calling HelloWorld VXML 
        
        Server app                        
         service hello_world                          
         incoming called-number <your DN pattern here>        
         direct-inward-dial
        
                                    
        dial-peer voice 800 voip                            
         description Example incoming VOIP dial-peer calling HelloWorld VXML 
        
        Server app                            
         service hello_world                            
         incoming called-number 800.......                            
         voice-class codec 1                            
         dtmf-relay rtp-nte                            
         no vad                            
        !
                                

        Configure Gateway Settings for Comprehensive Call Flow Model

        Procedure
          Step 1   Install the IOS image on the Ingress Gateway.

          For detailed information, see the Cisco IOS documentation.

          Step 2   Transfer the following script, configuration, and .wav files to the Ingress gateway through the Operations Console or the Unified CVP product CD:
          • bootstrap.tcl

          • handoff.tcl

          • survivabilty.tcl

          • bootstrap.vxml

          • recovery.vxml

          • ringtone.tcl

          • cvperror.tcl

          • ringback.wav

          • critical_error.wav

          Step 3   Configure the Ingress Gateway base settings.
          Step 4   Configure the Ingress Gateway service settings.
          Step 5   Configure an Ingress Gateway incoming Pots Dial-peer.
          Step 6   For SIP without a Proxy Server , complete the following steps:
          1. If you are using DNS query with SRV or A types from the gateway, configure the gateway to use DNS.

            Also, if you are using DNS query with SRV or A types from the gateway, use CLI as shown below:

            Note   

            Generally, a non-DNS setup is: sip-server ipv4:xx.xx.xxx.xxx:5060 .

            ip domain name pats.cisco.com
            ip name-server 10.86.129.16
            sip-ua
            sip-server dns:cvp.pats.cisco.com
            OR:
            ipv4:xx.xx.xxx.xxx:5060
            
          2. Configure the DNS zone file for the separate DNS server that displays how the Service (SRV) records are configured.
            Note   

            SRV with DNS can be used in any of the SIP call flow models, with or without a Proxy server. Standard A type DNS queries can be used as well for the calls, without SRV, but they lose the load balancing and failover capabilities.

            See DNS Zone File Configuration for Call Director Call Flow Model.

          Step 7   For SIP with a Proxy Server, if you are using the DNS Server, you can set your SIP Service as the Host Name (either A or SRV type).
          You can also configure the Gateway statically instead of using DNS. The following example shows how both the A and SRV type records could be configured:
          ip host cvp4cc2.cisco.com 10.4.33.132
          ip host cvp4cc3.cisco.com 10.4.33.133
          ip host cvp4cc1.cisco.com 10.4.33.131
          
          For SIP/TCP:
          ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc3.cisco.com
          ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc2.cisco.com
          ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc1.cisco.com
          
          For SIP/UDP:
          ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc3.cisco.com
          ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc2.cisco.com
          ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc1.cisco.com
          
          Note   

          The DNS Server must be configured with all necessary A type or SRV type records.

          See the SIP Devices Configuration and the Operations Console Online Help, Managing devices > Configuring a SIP Proxy Server for details.

          Step 8   Transfer files to the VXML Gateway using Step 2.
          Step 9   Configure the VXML Gateway base settings.
          Step 10   Configure the VXML Gateway service settings.
          Step 11   If using ASR and TTS Servers, specify IP addresses for those servers for each locale using the applicable name resolution system for the Gateway (DNS or "ip host" commands).
          Note   

          If ASR and TTS use the same server, the MRCP server might allocate one license for the ASR session and a second license for the TTS section. If you are hosting both ASR and TTS on the same speech server, you must select the ASR/TTS use the same MRCP server option in the IVR Service configuration tab in the Operations Console and follow the instructions in the step below.

          Do one of the following:  

          • If you are using ACE, the server name is configured to the virtual IP (VIP) of the Call Server on ACE. For more information, see the Configure High Availability for Unified CVP section.

          • The primary and backup servers must be configured. If using name resolution local to the Gateway (rather than DNS) specify:

            ip host asr- <locale> <ASR server for locale>

            ip host asr- <locale>-backup <backup ASR server for locale>

            ip host tts- <locale> <TTS server for locale>

            ip host tts- <locale>-backup <backup TTS server for locale>

            Example for English US, use:

            ip host asr-en-us 10.86.129.215

          Step 12   If you want the ASR and TTS to use the same MRCP server option, you must configure the gateway as follows.
          1. In the IVR Service in the Operations Console, select the ASR/TTS use the same MRCP server option.
          2. Add the following two host names to the gateway configuration:
            • ip host asrtts- <locale> <IP Address Of MRCP Server>

            • ip host asrtts- <locale> -backup <IP Address Of MRCP Server>

              Where the locale might be something like en-us or es-es, resulting in asrtts-en-us or asrtts-es-es.

          3. Change the 'ivr asr-server' and 'ivr tts-server' lines as follows for MRCPV1:
            • ivr asr-server rtsp://asr-en-server/recognizer

            • ivr tts-server rtsp://tts-en-server/synthesizer

          4. Change the 'ivr asr-server' and 'ivr tts-server' lines as follows for MRCPV2:
            • ivr asr-server sip:asr@10.78.26.103
            • ivr tts-server sip:tts@10.78.26.103
          Step 13   Configure the speech servers to work with Unified CVP.
          Caution   

          The Operations Console can only manage speech servers installed on Windows, not on Linux. If the speech server is installed on Linux, the server cannot be managed.

          To ensure that the speech servers work with Unified CVP, you must make the following changes on each speech server as part of configuring the Unified CVP solution.

          If you are using Nuance SpeechWorks MediaServer (SWMS), the configuration file is osserver.cfg. If you are using Nuance Speech Server (NSS), the configuration file is NSSserver.cfg.

          Make the following changes to the Nuance configuration file:

          • Change: server.resource.2.url VXIString media/speechrecognizer

            To: server.resource.2.url VXIString recognizer

          • Change: server.resource.4.url VXIString media/speechsynthesizer

            To: server.resource.4.url VXIString synthesizer

          • Change: server.mrcp1.resource.3.url VXIString media/speechrecognizer

            To: server.mrcp1.resource.3.url VXIString /recognizer

          • Change: server.mrcp1.resource.2.url VXIString media/speechsynthesizer

            To: server.mrcp1.resource.2.url VXIString media/synthesizer

          • Change: server.mrcp1.transport.port VXIInteger 4900

            To: server.mrcp1.transport.port VXIInteger 554

          If you are using Nuance Speech Server 5 and Nuance Vocalizer for Network 5, make changes to configuration files for each application. Make the following changes to the Nuance Speech Server 5 configuration file (NSSserver.cfg):

          • Change: server.mrcp1.resource.3.url VXIString media/speechrecognizer

            To: server.mrcp1.resource.3.url VXIString /recognizer

          • Change: server.mrcp1.resource.2.url VXIString media/speechsynthesizer

            To: server.mrcp1.resource.2.url VXIString /synthesizer

          • Change: server.mrcp1.transport.port VXIInteger 4900

            To: server.mrcp1.transport.port VXIInteger 554

          • Change: server.mrcp1.transport.dtmfPayloadType VXIInteger 96

            To: server.mrcp1.transport.dtmfPayloadType VXIInteger 101

          • Uncomment the following: server.rtp.dtmfTriggerLeading VXIInteger 0

            If you are using the Nuance Vocalizer for Network 5 TTS System, the following configuration files will need to be updated:

            <install path>\Nuance Vocalizer for Network 5.0\config\ttsrshclient.xml

          • Change: <ssml_validation>strict</ssml_validation>

            To:<ssml_validation>warn</ssml_validation>

            <install path>\Nuance Vocalizer for Network 5.0\config\ttssapi.xml

          • Change: <ssml_validation>strict</ssml_validation>

            To: <ssml_validation>warn</ssml_validation>

          If you are using Nuance Speech Server 10.0, make the following changes to the Nuance configuration file (NSSserver.cfg - C:\Program Files (x86)\Nuance\Speech Server\Server\config):

          • Change: server.mrcp1.resource.3.url VXIString media/speechrecognizer

            To: server.mrcp1.resource.3.url VXIString /recognizer

          • Change: server.mrcp1.resource.2.url VXIString media/speechsynthesizer

            To: server.mrcp1.resource.2.url VXIString /synthesizer

          • Change: server.mrcp1.transport.port VXIInteger 4900

            To: server.mrcp1.transport.port VXIInteger 554

          • Change: server.mrcp1.transport.dtmfPayloadType VXIInteger 96

            To: server.mrcp1.transport.dtmfPayloadType VXIInteger

          Make the following change to the Baseline.xml file C:\Program Files\Nuance\Recognizer\config

          Change: <ssml_validation>strict</ssml_validation>

          To:<ssml_validation>warn</ssml_validation>.

          Step 14   Configure SIP-Specific Actions.

          On the Unified CM server, CCMAdmin Publisher, configure SIP-specific actions:

          1. Create SIP trunks:
            • If you are using a SIP Proxy Server, set up a SIP trunk to the SIP Proxy Server.

            • Add a SIP Trunk for the Unified CVP Call Server.

            • Add a SIP Trunk for each Ingress gateway that will send SIP calls to Unified CVP that might be routed to Unified CM.

            Select Device > Trunk > Add New and add the following:

            • Trunk Type: SIP trunk

            • Device Protocol: SIP

            • Destination Address: IP address or host name of the SIP Proxy Server (if using a SIP Proxy Server). If not using a SIP Proxy Server, enter the IP address or host name of the Unified CVP Call Server.

            • DTMF Signaling Method: RFC 2833

            • Do not check the Media Termination Point Required checkbox.

            • If you are using UDP as the outgoing transport on Unified CVP, also set the outgoing transport to UDP on the SIP Trunk Security Profile.

          2. Add route patterns for outbound calls from Unified CM devices using a SIP Trunk to the Unified CVP Call Server. Also, add a route pattern for error DN.
            Note   

            CVP solution does not support 100rel. On the SIP profile for the Trunk, confirm that SIP Rel1xx Options are disabled.

            For warm transfers, the call from Agent 1 to Agent 2 does not typically use a SIP Trunk, but you must configure the CTI Route Point for that dialed number on the Unified CM Server and associate that number with your peripheral gateway user (PGUSER) for the JTAPI gateway on the Unified CM peripheral gateway. An alternative is to use the Dialed Number Plan on Unified ICME to bypass the CTI Route Point.

          3. Select Call Routing > Route/Hunt > Route Pattern > Add New.
            • Route Pattern: Specify the route pattern; for example: 3xxx for a TDM phone that dials 9+3xxx and all Unified ICME scripts are set up for 3xxx dialed numbers.

            • Gateway/Route List: Select the SIP Trunk defined in Step 2.

          4. If you are sending calls to Unified CM using an SRV cluster domain name, configure the cluster domain name.
            • Select: Enterprise Parameters > Clusterwide Domain Configuration.

            • Add the Cluster fully qualified domain name: FQDN.

          For detailed instructions about using Unified CM and the CUSP Server, see the Cisco Unified SIP Proxy Server documentation.

          Step 15   (Optional)Configure the SIP Proxy Server.

          From the CUSP Server Administration web page (http://<CUSP server>/admin):

          1. Configure the SIP static routes to the Unified CVP Call Server(s), Unified CM SIP trunks, and Gateways.
            Configure the SIP static routes for intermediary transfers for ring tone, playback dialed numbers, and error playback dialed numbers.
            Note   

            For failover and load balancing of calls to multiple destinations, configure the CUSP Server static route with priority and weight.

            See the SIP Devices Configuration and SIP Dialed Number Pattern Matching Algorithm for detailed information.

          2. Configure Access Control Lists for Unified CVP calls.
            • Select Proxy Settings > Incoming ACL.

            • Set address pattern: all

          3. Configure the service parameters.

            Select Service Parameters, and set the following:

            • Add record route: off

            • Maximum invite retransmission count: 2

            • Proxy Domain and Cluster Name: if using DNS SRV, set to the FQDN of your Proxy Server SRV name.

          4. Write down the IP address and host name of the SIP Proxy Server. You need this information when configuring the SIP Proxy Server in Unified CVP.
          5. If using redundant SIP Proxy Servers (primary and secondary or load balancing), decide whether to use DNS server lookups for SRV records or non-DNS based local SRV record configuration.

            The Comprehensive call flow model with SIP calls will typically be deployed with dual CUSP Servers for redundancy. In some cases, you might want to purchase a second CUSP Server. Regardless, the default transport for deployment will be UDP. Make sure you always set the AddRecordRoute setting to Off with CUSP Servers.

            Configure the SRV records on the DNS server or locally on Unified CVP with an .xml file (local xml configuration avoids the overhead of DNS lookups with each call). 

          Step 16   Configure Peripheral Gateways (PGs).

          On the NAM, ICM Configuration Manager, PG Explorer tool, configure a peripheral gateway (PG) for the Unified CVP. Configure a PG for each Unified CVP Call Server as follows:

          In the tree view pane, select the applicable PG.

          Logical Controller tab:

          • Client Type: VRU

          • Name: A name descriptive of this PG

            For example: <location>_A for side A of a particular location

          Peripheral tab:

          • Peripheral Name: Descriptive name of this Unified CVP peripheral

            For example: <location>_<cvp1> or <dns_name>

          • Client Type: VRU

          • Select: Enable Post-routing

          Advanced tab:

          • Select the name of the Unified CVP VRU from the Network VRU field drop-down list.

            For example: cvpVRU

          Routing Client tab:

          • Name: By convention, use the same name as the peripheral

          • Client Type: VRU

          • If you are in a Unified ICMH environment and configuring the CICM, then do the following:

            • Do not select the Network Transfer Preferred checkbox

            • Routing client: INCRP NIC


          Ingress and VoiceXML Gateway Configuration Examples

          Example Gateway Settings for Comprehensive Call Flow Model
          The first part of the following example provides the basic configuration for setting an Ingress gateway:
          • Applies a timestamp to debugging and log messages

          • Turns on logging

          • Turns off printing to the command line interface console

          • Sends RTP packets

          • Configures gateway settings

          The last part of this example provides the following:
          • Allows SIP to play a .wav file that enables caller to hear message from critical_error.wav

          • Performs survivability

          • Enables SIP to play ringtone to caller while caller is being transferred to an agent

          • Logs errors on the gateway when the call fails

          • Defines requirements for SIP Call Server


            Note


            CVP solution does not support 100rel. It can be disabled on the dial-peer level or on a global level under the voice service VoIP section.


          service timestamps debug datetime msec localtime
          service timestamps log datetime msec localtime
          !
          service internallogging buffered 99999999 debuggingn
          no logging console
          !
          ip cef
          !voice rtp send-recv
          !
          voice service voip
          signaling forward unconditional 
          sip 
          min-se 360 
          header-passing
          !voice class codec 1
          codec preference 1 g711ulaw
          codec preference 2 g729r8
          !
          application
          service cvperror flash:cvperror.tcl
          ! 
          service cvp-survivability flash:survivability.tcl
          ! 
          service ringtone flash:ringtone.tcl
          ! 
          service handoff flash:handoff.tcl
          !gateway
          timer receive-rtcp 4
          !
          ip rtcp report interval 2000
          !sip-ua
          retry invite 2
          timers expires 60000
          sip-server ipv4:<IP of CUSP server or Call Server>:5060
          reason-header override
          !
          VoiceXML: Example Gateway Settings for Comprehensive Call Flow Model
          The first part of the following example provides the basic configuration for setting a VoiceXML gateway:
          • Applies a timestamp to debugging and log messages

          • Turns on logging

          • Turns off printing to the command line interface console

          • Sends RTP packets

          • Configures ASR/TTS Server

          • Configures gateway settings

          The last part of this example provides the following:
          • Initiates the VoiceXML leg

          • Initiates the switch leg of the call

          • Plays a .wav file that enables caller to hear message from critical_error.wav

          • Logs errors on the gateway when the call fails

          service timestamps debug datetime msec
          service timestamps log datetime msec
          service internal
          logging buffered 99999999 debugging
          no logging console
          ip cef
          no ip domain lookup
          ip host tts-en-us <IP of TTS or MRCP Server>
          ip host asr-en-us <IP of ASR or MRCP Server>
          voice rtp send-recv
          !
          voice service voip
          signaling forward unconditional
          sip
          min-se 360
          header-passing
          voice class
          codec 1 codec preference 1 g711ulaw
          codec preference 2 g729r8
          !
          ivr prompt memory 15000
          ivr prompt streamed none
          ivr asr-server rtsp://asr-en-us/recognizer
          ivr tts-server rtsp://tts-en-us/synthesizer
          mrcp client timeout connect 10
          mrcp client timeout message 10
          mrcp client rtpsetup enable
          rtsp client timeout connect 10
          rtsp client timeout message 10
          vxml tree memory 500
          http client cache memory pool 15000
          http client cache memory file 500
          http client connection timeout 60
          http client response timeout 30
          http client connection idle timeout 10
          gateway
          timer receive-rtcp 6
          !
          ip rtcp report interval 3000
          application
          service new-call flash:bootstrap.vxml
          service cvperror flash:cvperror.tcl
          service handoff flash:handoff.tcl
          service bootstrap flash:bootstrap.tcl
          param cvpserverss1 1
          !

          Note


          The optional param cvpserverss1 1 line enables HTTPS.


          Configure Gateway Settings for Call Director Call Flow Model

          Procedure
            Step 1   Perform Steps 1 to 4 of the Configure Gateway Settings for Comprehensive Call Flow Model procedure.
            Step 2   Configure the Ingress Gateway:
            1. Configure the Ingress Gateway dial-peer for the Unified CVP Call Server.
            2. Configure a dial-peer for ringtone and error.
            3. If you are using a Proxy Server, configure your session target in the outbound dial peer to point to the Proxy Server.
            4. If you are using the sip-server global configuration, then configure the sip-server in the sip-ua section to be your Proxy Server and point the session target of the dial-peer to the sip-server global variable.
            Note   

            Make sure your dial plan includes this information. You will need to see the Dial plan when you configure the SIP Proxy Server for Unified CVP.

            The SIP Service voip dial peer and the destination pattern on the Ingress Gateway must match the DNIS in static routes on the SIP Proxy Server or Unified CVP Call Server.

            See the SIP Devices Configuration and SIP Dialed Number Pattern Matching Algorithm for detailed information.

            Step 3   For SIP without a Proxy Server, complete the following steps:
            1. If you are using DNS query with SRV or A types from the gateway, configure the gateway to use DNS.

              See the Operations Console online help for detailed instructions. If you are using DNS query with SRV or A types from the gateway, use the gateway configuration CLI as shown below:

              Non-DNS Setup:

              sip-ua
              sip-server ipv4:xx.xx.xxx.xxx:5060
              !
              

              DNS Setup:

              ip domain name patz.cisco.com
              ip name-server 10.10.111.16
              !
              sip-ua
              sip-server dns:cvp.pats.cisco.com
              !
              
            2. Configure the DNS zone file for the separate DNS server that displays how the Service (SRV) records are configured.
              Note   

              SRV with DNS can be used in any of the SIP call flow models, with or without a Proxy server. Standard A type DNS queries can be used as well for the calls, without SRV, but they lose the load balancing and failover capabilities.

              See the DNS Zone File Configuration for Call Director Call Flow Model for more information.

            Step 4   For SIP with a Proxy Server, use one of the following methods:
            Note   

            You can configure the Gateway statically instead of using DNS.

            The following example shows how both the A and SRV type records could be configured:

            ip host cvp4cc2.cisco.com 10.4.33.132
            ip host cvp4cc3.cisco.com 10.4.33.133
            ip host cvp4cc1.cisco.com 10.4.33.131
            

            For SIP/TCP:

            ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc3.cisco.com
            ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc2.cisco.com
            ip host _sip._tcp.cvp.cisco.com srv 50 50 5060 cvp4cc1.cisco.com
            

            For SIP/UDP:

            ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc3.cisco.com
            ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc2.cisco.com
            ip host _sip._udp.cvp.cisco.com srv 50 50 5060 cvp4cc1.cisco.com
            
            Note   

            The DNS Server must be configured with all necessary A type or SRV type records.

            See the SIP Devices Configuration.

            If you are using the DNS Server, you can set your SIP Service as the Host Name (either A or SRV type).

            Step 5   On the Unified CM server, CCMAdmin Publisher, complete the following SIP-specific actions:
            1. Create SIP trunks.
              • If you are using a SIP Proxy Server, set up a SIP trunk to the SIP Proxy Server.

              • Add a SIP Trunk for the Unified CVP Call Server.

              • Add a SIP Trunk for each Ingress gateway that will send SIP calls to Unified CVP that might be routed to Unified CM.

              To add an SIP trunk, select Device > Trunk > Add New and use the following parameters:

              • Trunk Type: SIP trunk

              • Device Protocol: SIP

              • Destination Address: IP address or host name of the SIP Proxy Server (if using a SIP Proxy Server). If not using a SIP Proxy Server, enter the IP address or host name of the Unified CVP Call Server.

              • DTMF Signaling Method: RFC 2833

              • Do not check the Media Termination Point Required check box.

              • If you are using UDP as the outgoing transport on Unified CVP, also set the outgoing transport to UDP on the SIP Trunk Security Profile.

              • Connection to CUSP Server: use 5060 as the default port.

            2. Add route patterns for outbound calls from the Unified CM devices using a SIP Trunk to the Unified CVP Call Server. Also, add a route pattern for error DN.

              Select Call Routing > Route/Hunt > Route Pattern > Add New

              Add the following:

              • Route Pattern: Specify the route pattern; for example: 3XXX for a TDM phone that dials 9+3xxx and all Unified ICME scripts are set up for 3xxx dialed numbers.

              • Gateway/Route List: Select the SIP Trunk defined in the previous substep.

              Note   

              For warm transfers, the call from Agent 1 to Agent 2 does not typically use a SIP Trunk, but you must configure the CTI Route Point for that dialed number on the Unified CM server and associate that number with your peripheral gateway user (PGUSER) for the JTAPI gateway on the Unified CM peripheral gateway. An alternative is to use the Dialed Number Plan on Unified ICME to bypass the CTI Route Point.

            3. If you are sending calls to Unified CM using an SRV cluster domain name, select Enterprise Parameters > Clusterwide Domain Configuration and add the Cluster fully qualified domain name FQDN.
            Step 6   (Optionally) Configure the SIP Proxy Server.
            1. Configure the SIP static routes to the Unified CVP Call Servers, Unified CM SIP trunks, and Gateways.

              Configure the SIP static routes for intermediary transfers for ringtone, playback dialed numbers, and error playback dialed numbers.

              Note   

              For failover and load balancing of calls to multiple destinations, configure the CUSP server static route with priority and weight.

            2. Configure Access Control Lists for Unified CVP calls.

              Select Proxy Settings > Incoming ACL.

              Address pattern: all

            3. Configure the service parameters.

              Select Service Parameters, then set the following:

              • Add record route: off

              • Maximum invite retransmission count: 2

              • Proxy Domain and Cluster Name: if using DNS SRV, set to the FQDN of your Proxy Server SRV name

            4. Write down the IP address and host name of the SIP Proxy Server. (You need this information when configuring the SIP Proxy Server in Unified CVP.)
            5. If using redundant SIP Proxy Servers (primary and secondary or load balancing), then decide whether to use DNS server lookups for SRV records or non-DNS based local SRV record configuration.
              Note   

              If a single CUSP Server is used, then SRV record usage is not required.

               

              Configure the SRV records on the DNS server or locally on Unified CVP with a .xml file (local xml configuration avoids the overhead of DNS lookups with each call). 

              Note   

              See the Local SRV File Configuration Example for SIP Messaging Redundancy section for details.

              The Call Director call flow model with SIP calls will typically be deployed with dual CUSP servers for redundancy. In some cases, you might want to purchase a second CUSP server. Regardless, the default transport for deployment will be UDP; make sure you always disable the record-route in a CUSP server as this advanced feature is not supported in Contact Center deployments.

              For the required settings in the Unified CM Publisher configuration, see the Cisco Unified SIP Proxy documentation.

            Step 7   Configure the PGs for the switch leg.

            On Unified ICME, ICM Configuration Manager, PG Explorer tool:

            1. Configure each peripheral gateway (PG) to be used for the Switch leg. In the tree view pane, select the applicable PG, and set the following:
              1. Logical Controller tab:

                • Client Type: VRU

                • Name: A name descriptive of this PG

                  For example: <location>_A for side A of a particular location

              2. Peripheral tab:

                • Peripheral Name: A name descriptive of this Unified CVP peripheral

                  For example: <location>_<cvp1> or <dns_name>

                • Client Type: VRU

                • Select the check box: Enable Post-routing

              3. Routing Client tab:

                • Name: By convention, use the same name as the peripheral.

                • Client Type: VRU

              For more information, see the ICM Configuration Guide for Cisco ICM Enterprise Edition.

            2. Configure a peripheral for each Unified CVP Call Server to be used for a Switch leg connected to each peripheral gateway.

            Configure Gateway Settings for VRU-Only Call Flow Model: Type 8

            Procedure
              Step 1   Using the Unified CVP Operations Console or the Unified CVP product CD, transfer the following script, configuration, and .wav files to the VoiceXML Gateway used for the VRU leg. Perform Step 2 of the Configure Gateway Settings for Comprehensive Call Flow Model procedure.
              Step 2   Configure the VXML gateway base settings.
              Step 3   Configure the VXML gateway service settings.
              Step 4   Configure the ICM service.

              Using the Operations Console, select Device Management > CVP Call Server > ICM tab. On each Unified CVP Call Server, configure the ICM Service by specifying the following required information:

              1. VRU connection port number.

                Set the VRU Connection Port to match the VRU connection Port defined in ICM Setup for the corresponding VRU peripheral gateway (PIM).

              2. Maximum Length of DNIS.

                Set the maximum length DNIS to a number which is at least the length of the translation route DNIS numbers.

                Example: if the Gateway dial pattern is 1800******, the maximum DNIS length is 10.

              3. Call service IDs: New Call and Pre-routed.

                Enter the new and pre-routed call service IDs. Configure the ports for both groups according to the licenses purchased, call profiles, and capacity by completing the required fields on this tab.

              4. Trunk group IDs: New Call and Pre-routed.
                • Enter the new and pre-routed call trunk group IDs

                • Configure the group number for the Pre-routed Call Trunk group. The group number must match the trunk group number in the Network Trunk group used for the translation route

                • Configure the number of ports according to the licenses purchased and capacity

                • Configure each of the numbers used for translation routes. (The "New Call" group is not used since the calls are being sent to the VRU (Unified CVP) after some initial processing by the NIC/Unified ICME)

              5. Dialed numbers used in the translation route.

                Add the dialed numbers in the DNIS field.

              6. Check the default values of the other settings and change, if desired.

              VoiceXML Gateway Configuration Examples

              Example Gateway Settings for Type 8 Call Flow Model
              The first part of the following example provides the basic configuration for setting a VoiceXML gateway:
              • Applies a timestamp to debugging and log messages

              • Turns on logging

              • Turns off printing to the command line interface console

              • Sends RTP packets

              • Configures ASR/TTS Server

              • Configures gateway settings

              The last part of this example provides the following:
              • Initiates the VoiceXML leg

              • Plays a .wav file that enables caller to hear message from critical_error.wav

              • Logs errors on the gateway when the call fails

              service timestamps debug datetime msec
              service timestamps log datetime msec
              service internal
              logging buffered 99999999 debugging
              no logging console
              ip cef
              no ip domain lookup
              ip host tts-en-us <IP of TTS or MRCP Server>
              ip host asr-en-us <IP of ASR or MRCP Server>
              voice rtp send-recv
              !
              voice service voip
              allow-connections h323 to h323
              signaling forward unconditional
              h323
              sip
              min-se 360
              header-passing
              voice class codec 1
              codec preference 1 g711ulaw
              codec preference 2 g729r8
              !
              ivr prompt memory 15000
              ivr prompt streamed none
              ivr asr-server rtsp://asr-en-us/recognizer
              ivr tts-server rtsp://tts-en-us/synthesizer
              mrcp client timeout connect 10
              mrcp client timeout message 10
              mrcp client rtpsetup enable
              rtsp client timeout connect 10
              rtsp client timeout message 10
              vxml tree memory 500
              http client cache memory file 500
              http client connection timeout 60
              http client response timeout 30
              http client connection idle timeout 10
              gateway
              timer receive-rtcp 6
              !
              ip rtcp report interval 3000
              application
              service new-call flash:bootstrap.vxml
              service cvperror flash:cvperror.tcl
              service handoff flash:handoff.tcl
              
              Example of Dial-peer for ICM VRU Label for Type 8 Call Flow Model
              The following example provides the configuration for an ICM VRU label dial-peer for the Type8 Unified CVP VRU-Only call flow model:
              dial-peer voice 777 voip
               description ICM VRU label
               service bootstrap
               voice-class codec 1
               incoming called-number <your sendtovru label pattern here> 
               dtmf-relay rtp-nte
               no vad
               !

              Configure Gateway Settings for VRU-Only: Type 7

              Procedure
                Step 1   Transfer the following script, configuration, and .wav files to the VoiceXML Gateway used for the VRU leg, using the Unified CVP Operations Console. Perform Step 2 of the Configure Gateway Settings for Comprehensive Call Flow Model procedure.
                Step 2   Configure the VoiceXML gateway base settings.
                Step 3   Configure the VoiceXML gateway service settings.
                Step 4   Configure the ICM Service for each Call Server.

                In the Operations Console, select Device Management > CVP Call Server > ICM tab. For each Unified CVP Call Server, configure the ICM Service by specifying the following required information:

                1. VRU connection port number.

                  Set the VRU Connection Port to match the VRU connection Port defined in ICM Setup for the corresponding VRU peripheral gateway (PIM).

                2. Set the maximum length DNIS to the length of the Network Routing Number.

                  Example: if the Gateway dial pattern is 1800******, the maximum DNIS length is 10.

                3. Call service IDs: New Call and Pre-routed.

                  Enter the new and pre-routed call service IDs. Configure the ports for both groups according to the licenses purchased, call profiles, and capacity by completing the required fields on this tab

                4. Trunk group IDs: New Call and Pre-routed.

                  Enter the new and pre-routed call trunk group IDs. Configure the group number for the Pre-routed Call Trunk group. The group number must match the trunk group number in the Network Trunk group used for the translation route.

                  Configure the number of ports according to the licenses purchased and capacity. Configure each of the numbers used for translation routes. (The "New Call" group is not used since the calls are being sent to the VRU (Unified CVP) after some initial processing by the NIC/Unified ICME.)

                5. Check the default values of other settings and change, if desired.

                VoiceXML Gateway Configuration: Example Gateway Settings for Type 7

                The first part of the following example provides the basic configuration for setting a VoiceXML gateway:

                • Applies a timestamp to debugging and log messages

                • Turns on logging

                • Turns off printing to the command line interface console

                • Sends RTP packets

                • Configures ASR/TTS Server

                • Configures gateway settings

                The last part of this example provides the following:

                • Initiates the VoiceXML leg

                • Plays a .wav file that enables caller to hear message from critical_error.wav

                • Logs errors on the gateway when the call fails

                  service timestamps debug datetime msec
                  service timestamps log datetime msec
                  service internal
                  logging buffered 99999999 debugging
                  no logging console
                  ip cef
                  no ip domain lookup
                  ip host tts-en-us <IP of TTS or MRCP Server>
                  ip host asr-en-us <IP of ASR or MRCP Server>
                  voice rtp send-recv
                  !
                  voice service voip
                   allow-connections h323 to h323
                   signaling forward unconditional
                   h323
                   sip
                   min-se 360
                   header-passing
                  voice class codec 1
                   codec preference 1 g711ulaw
                   codec preference 2 g729r8
                  !
                  ivr prompt memory 15000
                  ivr prompt streamed none
                  ivr asr-server rtsp://asr-en-us/recognizer
                  ivr tts-server rtsp://tts-en-us/synthesizer
                  mrcp client timeout connect 10
                  mrcp client timeout message 10
                  mrcp client rtpsetup enable
                  rtsp client timeout connect 10
                  rtsp client timeout message 10
                  vxml tree memory 500
                  http client cache memory pool 15000
                  http client cache memory file 500
                  http client connection timeout 60
                  http client response timeout 30
                  http client connection idle timeout 10
                  gateway
                   timer receive-rtcp 6
                  !
                  ip rtcp report interval 3000
                  application
                   service new-call flash:bootstrap.vxml
                   service cvperror flash:cvperror.tcl
                   service handoff flash:handoff.tcl
                   service bootstrap flash:bootstrap.tcl
                   !

                The following example provides the configuration for an ICM VRU label dial-peer for the Type 7 Unified CVP VRU-Only call flow model:

                dial-peer voice 777 voip
                 description ICM VRU label
                 service bootstrap
                 voice-class codec 1
                 incoming called-number <your sendtovru label pattern here>
                 dtmf-relay rtp-nte
                 no vad
                 !

                Transfer Script and Media File to Gateway

                Transfer a single script or media file at a time from the Operations Console.

                Procedure
                  Step 1   Log in to the Operations Console and from the Device Management menu, select the type of server to which to transfer the script file.

                  Example:

                  To transfer a script or a media file to a Gateway, select Device Management > Gateway..

                  The Find, Add, Delete, Edit window lists any servers that have been added to the Operations Console.

                  Step 2   Select a server by clicking the link in its Hostname field or by clicking the radio button preceding it and then clicking Edit.
                  Step 3   Select File Transfer in the toolbar, and then click Scripts and Media.

                  The Scripts and Media File Transfer page appears, listing the host name and IP address for the selected device. Script and Media files currently stored in the Operations Server database are listed in the Select From available Script Files drop box.

                  Step 4   If the script or media file is not listed in the Select From Available Script Files drop box:
                  1. Click Select a Script or Media File from Your Local PC.
                  2. Enter the file name in the text box or click Browse to search for the script or media file on the local file system.
                  Step 5   If the script or media file is listed in the Select From Available Script Files drop box, select the script or media file.
                  Step 6   Click Transfer to send the file to the device.