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Configuring the GVP Components

Perform these advanced configuration procedures after GVP installation and basic configuration.

Integrating Application Objects

After the Media Control Platform and Call Control Platform Application objects are created and the components are installed, they are integrated with the Resource Manager which acts as a proxy server. SIP devices and VoiceXML or CCXML applications can then make use of media-centric services through the proxy, without having to know the actual location of these resources.

This procedure is optional and required only if you want the Resource Manager to act as a proxy server for outbound requests. To integrate these Application objects with the Resource Manager, you configure the Session Initiation Protocol (SIP) settings.

This procedure describes how to integrate Application objects with the Resource Manager by configuring SIP and secure SIP options.

Procedure: Integrating Application Objects with Resource Manager

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Creating a Connection to a Server

Use the procedure in this section to create connections to:

  • The Message Server—Create a connection in the Media Control Platform, Call Control Platform, Resource Manager, Supplementary Services Gateway, CTI Connector, PSTN Connector, MRCP Proxy, Reporting Server and Policy Server Applications to ensure that component log information reaches the Log database and can be viewed in the Solution Control Interface (SCI).
  • The Reporting Server—Create a connection in the Media Control Platform, Call Control Platform, Resource Manager, PSTN Connector, CTI Connector, Supplementary Services Gateway, and MRCP Proxy Applications to ensure that these components detect the Reporting Server to which they are sending reporting data. Genesys Administrator also requires a connection to Reporting Server to monitor GVP components.
  • SIP Server—Create a connection in the Resource Manager, Supplementary Services Gateway, and PSTN Connector Applications to manage the initiation of outbound calls.
  • MRCP Proxy—Create a connection in the Media Control Platform Application if you are planning to use the proxy to manage MRCPv1 RTSP traffic within the GVP deployment.
  • MRCP Server—Create a connection in the MRCP Proxy Application if you are planning to use the proxy to manage MRCPv1 RTSP traffic within the GVP deployment (in the Media Control Platform Application if you are not deploying the MRCP Proxy).
  • Cisco T-server—Create a connection in the UCM Connector Application to ensure the tenant DBID of the Cisco T-Server is included in Request URI in any SIP INVITE messages sent to the UCM Connector.
  • The SNMP Master Agent—Create a connection in the Media Control Platform, Call Control Platform, Resource Manager, Supplementary Services Gateway, CTI Connector, PSTN Connector, MRCP Proxy, and Reporting Server Applications if you want to capture alarm and trap information.

Procedure: Creating a Connection to a Server

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Provisioning the Speech Resources

The Media Resource Control Protocol (MRCP) speech resources are controlled by the Call Manager Application Program Interface (CMAPI), which opens and closes sessions, and provides the speech recognition and speech synthesis commands that the MRCP Server uses to carry out speech requests.

If the MRCP Proxy is deployed, the configurations in this procedure vary slightly. Therefore, the configurations are described with and without the MRCP Proxy. If you have installed the MRCP Proxy, see also Provisioning the MRCP Proxy on page 250.

Tip
The procedures in this section are required only if you are using Automatic Speech Recognition (ASR) and Text-to-Speech (TTS) speech resources, and have an MRCP Server or MRCP Proxy in your deployment.

This section contains two procedures that create the Speech Resource Applications and assign the MRCP Server or MRCP Proxy to the Media Control Platform.

Procedure: Provisioning Speech Resource Application Objects

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Procedure: Assigning the MRCP Server to the Media Control Platform

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Provisioning the MRCP Proxy

The MRCP Proxy is an optional component, but must be deployed if ASR and TTS usage reporting is required. You can deploy the MRCP Proxy in stand-alone or warm active standby HA mode. The procedures in this section describe the steps for each configuration.

Tip
By design, the MRCP Proxy supports only the NUANCE speech resource.

Procedure: Configuring the MRCP Proxy

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Procedure: Configuring the MRCP Proxy for HA

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Procedure: Adding a Speech Server as Primary or Backup

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Configuring the CTI Connector for Cisco ICM

When you install the CTI Connector, you can select the CTI Framework that is appropriate for your environment Genesys CTI or Cisco ICM. Use the procedure in this section to configure the CTI Connector if you selected Cisco ICM.

Procedure: Configuring the CTI Connector for Cisco ICM Integration

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Provisioning the PSTN Connector

The procedures in this section describe how to configure the mandatory parameters for the Public Switched Telephone Network (PSTN) Connector Application object and the how to integrate the PSTN Connector with SIP Server. There are many more configurable parameters for the PSTN Connector, all of which are optional. For a complete list and description of configuration options, see the Genesys Voice Platform 8.5 User's Guide.

The PSTN Connector component is required if you are planning to migrate from GVP 7.x Voice Communication Server (VCS), or a VoiceGenie (VG) TDM interface to GVP 8.1.2 or later.

This section contains procedures for Configuring the PSTN Connector and Configuring a Trunk DN for the PSTN Connector.

Procedure: Configuring the PSTN Connector

Purpose
To prepare the PSTN Connector to manage inbound and outbound calls for GVP.

Prerequisites

  • All of the GVP components are installed. See Procedure: Using the Deployment Wizard to Install GVP, on page 221.
  • The connections to Message Server, SIP Server, and the SNMP Master Agent are configured in the PSTN Connector Application object. See Procedure: Creating a Connection to a Server, on page 243.
  • SIP Server is installed. See the Voice Platform Solution 8.1 Integration Guide.

Steps

  1. Log in to Genesys Administrator.
  2. On the Provisioning tab, select Environment > Applications.
  3. Select the PSTN Connector Application object that you want to configure.
    The Configuration tab appears.
  4. Click the Options tab, and from the View drop-down list, select the Mandatory Options view.
  5. In the DialogicManager_Route1 and GatewayManager sections, enter the values for the mandatory options as shown in Table 23.
    Table 23: PSTN Connector Mandatory Parameters
    Section Option Value
    DialogicManager_Route1 RouteType Specify one of three route types or call directions for this route, enter:
    • 0 for Inbound
    • 1 for outbound
    • 2 for In/Out

    (See Note below, in this table.)

    Signaling Type Specify one of five signaling types, enter:
    • 0 for T1-ISDN (PRI)
    • 1 for Analog
    • 2 for E1-ISDN (PRI)
    • 3 for T1-RobbedBit
    • 4 for E1-CAS
    Channels Specify the ports for this route by using the format, [<Card>:<PortRange>,<Card>:<PortRange>].

    You can provision more than one board in a route and a partial range of ports in a board for example:

    • 1:1-23
    • 1:1-23,2:1-23
    • 1:1-30,2:1-30
    • 1:1-12,2:1-15
    GatewayManager SIP Destination IP Address Enter the SIP endpoint IP address that will receive SIP calls from the PSTN Connector. (This is the IP address for SIP Server or the

    Resource Manager, depending on your configuration.)

    SIP Destination Port Number Enter the SIP endpoint port number of the server that is configured in SIP Destination IP Address.
    MediaManager Supported Local Codec Type Enter the audio format that is in use on the TDM trunk:
    • 0 - Mulaw
    • 8 - ALaw
    Notes
    • The Inbound & Outbound route type is supported only on ISDN (PRI) lines. If you select one of these route types, ensure that you use a compatible signaling type.
    • If you are using a T1-Robbed Bit or E1-CAS interface, options in the T1rb options group must be configured, specifically the T1rbProtocolFile option, see the component metadata.
    • For JCT boards only, a separate span is required to support ASR or recorded VoiceXML applications in T1-ISDN, E1-ISDN, or E1-CAS environments. The MediaVoxResourceBoard option must be configured with route number that is used for CSP.
    • When JCT boards are used with the PSTN Connector, and the VoiceXML application uses ASR or recording media, not all the spans can be used for call handling. For each span that is configured to take calls, there must be another dedicated span for streaming echo cancelled audio to the Media Control Platform. Therefore, if Route1 is configured to handle calls on span1 (for example, ports=1:1-23), the MediaVoxResourceBoard option under the DialogicManagerRoute1 section should be set to 2. Repeat the same steps if you have configured a DialogicManager_Route2 section.

    This restriction does not apply in the following scenarios:

    • The VoiceXML application is a pure DTMF application (does not use ASR or recording media).
    • The VoiceXML application uses ASR but the JCT board is configured with the T1-Robbed Bit protocol.
  6. Create additional DialogicManager_Route<N> sections as required (for example, you may want to create a section for inbound ports and one for outbound ports):
    1. From the View drop-down list, select either Advanced View (Options) or Advanced View (Annex).
    2. Right-click on the Section column heading and select New.
    3. Enter DialogicManager_Route2 for the Section name, Description for the Option name, and Route2 Information for the Value.
    4. Click OK.
    5. Copy and paste all of the options from the DialogicManager_Route1 and DialogicManager_Route2 sections, modifying the mandatory values as required.
    6. Repeat these steps as required.
    Tip
    Dialogic Circular Buffer Size
    When you configure the PSTN Connector application, set the value for [DialogicManager] DialogicTransferBufferSize to 2048. This specifies the size of the Dialogic Circular buffer which is used for transferring data to the Dialogic firmware. The default value of this parameter on Windows is different, but must be set to 2048 for Linux. If not, the Dialogic may return LOW_WATER_MARK warnings during initial play of the media, which interferes with normal audio play and may result in the prompt being cut off.

    Save the configuration.

End Next

  • Configure the Trunk and Trunk Group DNs for the PSTN Connector. See Procedure: Configuring a Trunk DN for the PSTN Connector.

Procedure: Configuring a Trunk DN for the PSTN Connector

Purpose
To configure SIP Server with a Trunk DN, which points to the PSTN Connector Application object to ensure outbound calls can be routed to a specific PSTN Connector instance.

Summary
You can deploy multiple PSTN Connectors, however, you must ensure that SIP Server routes the outbound call to the same PSTN Connector instance as the inbound call. This procedure includes configuration options to enable this functionality.

Prerequisites

  • The PSTN Connector is installed and configured. See Procedure: Configuring the PSTN Connector, on page 254.

Steps

  1. Log in to Genesys Administrator.
  2. On the Provisioning tab, select Environment > Switching > Switches.
  3. Double-click the switch that you want to configure.
    The Configuration tab appears.
  4. On the DNs tab, select New.
  5. In the General section, enter values for the mandatory fields, selecting Trunk from the Type drop-down list.
  6. In the Switch pane, double-click the Trunk DN you created in Step 4.
  7. On the Options tab, select Advanced View (Annex) from the View drop-down list.
  8. Right-click on the Section column, and select New.
  9. In the New Option dialog:
    • In the Section field, enter TServer.
    • In the Name field, enter contact.
    • In the Value field, enter the IP address and port number of the PSTN Connector separated by a colon for example, 10.10.10.101:5060
  10. 1In the DNs pane, click New.
  11. In the TServer section, add the following options and values:
    • prefix = <xyzz> where <xyzz> represents a number which, if present in the dial string of the outbound call, enables the SIP Server to route the call to the PSTN Connector instance that is configured with this prefix.
    • replace-prefix = <empty String) where <empty String) represents an empty string to ensure that the prefix added by Resource Manager to the destination number string is removed by the SIP Server before the call is forwarded to the same PSTN Connector instance.
  12. 1Save the configuration.

For information about how the PSTN Connector fits into a common VPS deployment architecture, see the Supported Architecture chapter in the Voice Platform Solution 8.1 Integration Guide.

End

Next

  • If required, complete the post-installation activities for the Supplementary Services Gateway. See [[[Procedure: Configuring the Supplementary Services Gateway, on page 259]]].
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