85gvp Us

Genesys Voice Platform 8.
User’s Guide
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Document Version: 85gvp_us_12-2013_v8.5.001.00
Table of Contents
List of
Procedures ................................................................................................................. 11
Preface ................................................................................................................. 13
About Genesys Voice Platform................................................................ 13
Intended Audience................................................................................... 14
Making Comments on This Document .................................................... 14
Contacting Genesys Customer Care....................................................... 14
Document Change History ...................................................................... 15
Chapter 1 Introduction........................................................................................... 17
About GVP............................................................................................... 17
GVP Components............................................................................... 17
IVR Profiles......................................................................................... 20
GVP MIBs ........................................................................................... 20
Genesys Administrator ............................................................................ 20
GVP Identifiers and SIP Headers ............................................................ 22
Session Identifiers .............................................................................. 22
Application Identifiers.......................................................................... 23
Part 1 Provisioning GVP........................................................ 27
Chapter 2 Configuration and Provisioning Overview......................................... 29

Configuring GVP...................................................................................... 29
Configuring GVP Processes in Genesys Administrator...................... 30
Task Summary: Configuring GVP............................................................ 33
Chapter 3 Configuring Common Features........................................................... 37

Configuring SIP Communication and Routing ......................................... 38
Enabling Secure Communication ............................................................ 42
Configuring MCP, MRCPv2, CCP, CTIC, and RM for Secure SIP
Transport ........................................................................................ 44
User’s Guide 3
Procedures that Support Enabling Secure Communication ............... 46
Enabling IPv6 Communication ................................................................ 53
Enabling Conference Services ................................................................ 58
Configuring Reporting.............................................................................. 59
Configuring Logging ................................................................................ 62
Configuring SNMP................................................................................... 68
Configuring Client-Side Connections....................................................... 68
Customizing SIP Responses ................................................................... 74
Configuring Session Timers and Timeouts .............................................. 76
Resource Manager Session Timers.................................................... 76
Additional Timeouts ............................................................................ 79
Chapter 4 Configuring the Resource Manager.................................................... 83

Task Summary: Configuring the Resource Manager............................... 83
Important Resource Manager Configuration Options .............................. 84
Configuring Logical Resource Groups..................................................... 89
Chapter 5 Configuring Policy Server ................................................................... 97

Task Summary: Configuring Policy Server .............................................. 97
Important Policy Server Configuration Options ....................................... 98
Chapter 6 Provisioning IVR Profiles................................................................... 103

Provisioning IVR Profiles for GVP ......................................................... 103
IVR Profile Configuration Options.......................................................... 109
Operational Parameter Management and Self-Service Applications .... 123
IVR Profile Configuration for GVPi ........................................................ 123
IVR Profile Configuration for Cisco ICM ................................................ 127
Mapping IVR Profiles to DID Numbers .................................................. 128
DID Group Bulk Operations Wizard....................................................... 131
Data Retention Policy Wizard................................................................ 134
IVR Profile Configuration for Tenants .................................................... 136
Chapter 7 Configuring the Media Control Platform .......................................... 141

Task Summary: Configuring the Media Control Platform....................... 142
Enabling ASR and TTS ......................................................................... 146
Enabling Outbound Dialing.................................................................... 149
Media Server Markup Language ........................................................... 153
Important Media Control Platform Configuration Options ...................... 154
Important MRCP Server Configuration Options .................................... 190
4 Genesys Voice Platform 8.5

Chapter 8 Configuring the MRCP Proxy ............................................................ 195
Task Summary: Configuring the MRCP Proxy....................................... 195
Task Summary: Configuring the MRCP Proxy for HA ........................... 196
Important MRCP Proxy Configuration Options...................................... 198
Chapter 9 Configuring the Call Control Platform.............................................. 209

Task Summary: Configuring the Call Control Platform .......................... 209
Important Call Control Platform Configuration Options.......................... 212
Chapter 10 Configuring the CTI Connector ......................................................... 225

Configuring the CTI Connector.............................................................. 225
Important CTI Connector Configuration Options ................................... 229
Cisco ICM Messages and Data Formats ............................................... 235
Interaction Data Formats .................................................................. 237
CTIC (Genesys) and Treatments........................................................... 238
Invalid Treatment Types.................................................................... 238
Music Treatment ............................................................................... 239
PlayAnnounce & PlayAnnounceAndDigits Treatments..................... 239
VoiceXML Call Reporting .................................................................. 240
Multiple Trunk Group ID support for CTI Connector (ICM) .................... 240
CTI Connector (ICM) and ECC Variables .............................................. 241
CTIC (ICM) Parameter Notes ................................................................ 242
TrunkGroupID ................................................................................... 242
eccvariablelist and eccSessionIdVarname ......................................... 242
Chapter 11 Configuring the Supplementary Services Gateway ........................ 245

Task Summary: Configuring the Supplementary Services Gateway ..... 245
Important Supplementary Services Gateway Configuration Options..... 246
Call Progress Detection......................................................................... 251
Chapter 12 Configuring the PSTN Connector ..................................................... 255

Task Summary: Configuring the PSTN Connector ................................ 255
Important PSTN Connector Configuration Options ............................... 256
Chapter 13 Configuring the Fetching Module and Squid Proxy........................ 265

Task Summary: Configuring the Fetching Module and Squid................ 265
Important Fetching Module Configuration Options ................................ 266
Configuring the Squid Caching Proxy.................................................... 268
User’s Guide 5
Chapter 14 Configuring the Reporting Server..................................................... 271
Task Summary: Configuring the Reporting Server ................................ 271
Configuring Reporting, by Granularity ................................................... 273
Configuring Database Retention Policies .............................................. 274
Important Reporting Server Configuration Options ............................... 276
Controlling Access to Reporting Services ............................................. 283
Chapter 15 Configuring GVP in Multi-Site Environments .................................. 287

Overview................................................................................................ 287
Configuring the Site Folder.................................................................... 288
Part 2 Monitoring GVP ......................................................... 291
Chapter 16 Reporting Overview............................................................................ 293

Reports—Using GA vs. Using GAX....................................................... 293
Generating a Report with GA ................................................................ 294
Generating a Report with GAX .............................................................. 298
GAX Report Generation Table............................................................... 300
Report Groups ....................................................................................... 303
Call Detail Record Browsing ............................................................. 303
Dashboard ........................................................................................ 303
Operational Reporting....................................................................... 303
Service Quality Reporting ................................................................. 305
Voice Application Reporting .............................................................. 305
Report Filters ......................................................................................... 306
Chapter 17 Voice Platform Dashboards............................................................... 311

Overview................................................................................................ 311
Call Dashboard...................................................................................... 314
IVR Profile Utilization ........................................................................ 315
Component Utilization....................................................................... 316
Tenant Utilization .............................................................................. 318
SQ Latency Dashboard ......................................................................... 319
Fetch Dashboard ................................................................................... 321
SSG Dashboard .................................................................................... 322
SSG IVR Profile Utilization ............................................................... 323
SSG Component Utilization .............................................................. 323
SSG Tenant Utilization...................................................................... 324
PSTNC Dashboard................................................................................ 325
CTIC Dashboard.................................................................................... 326

Chapter 18 Real-Time Reports.............................................................................. 329
Overview................................................................................................ 329
Active Call Browser ............................................................................... 329
Chapter 19 Historical Reports............................................................................... 337

Overview................................................................................................ 337
IVR Profile Call Arrivals ......................................................................... 338
Component Call Arrivals........................................................................ 340
Tenant Call Arrivals................................................................................ 342
Media Service Call Arrivals.................................................................... 343
IVR Profile Call Peaks ........................................................................... 344
Component Call Peaks.......................................................................... 347
Tenant Call Peaks.................................................................................. 349
Media Service Call Peaks...................................................................... 351
MCP VXML Call Arrivals........................................................................ 352
MCP VXML Call Peaks.......................................................................... 352
ASR/TTS Usage .................................................................................... 353
ASR/TTS Usage Peaks ......................................................................... 354
Media Services Usage and GVP Ports Peaks....................................... 355
Historical Call Browser .......................................................................... 358
Per-Call IVR Actions Report ............................................................. 363
Chapter 20 Service Quality Reports ..................................................................... 367

Overview................................................................................................ 367
SQ Call Failures .................................................................................... 367
SQ Failure Summary ............................................................................. 370
SQ Latency Summary ........................................................................... 372
Chapter 21 Voice Application Reports................................................................. 377

Overview................................................................................................ 377
VAR Call Completion Summary............................................................. 377
VAR IVR Action Summary ..................................................................... 380
VAR Last IVR Action.............................................................................. 382
Part 3 Appendixes................................................................ 385
Appendix A Module and Specifier IDs................................................................... 387

Media Control Platform.......................................................................... 387
Next Generation Interpreter Module ID and Specifiers ..................... 408
User’s Guide 7
Genesys Voice Platform Interpreter Module ID and Specifiers ......... 409
Call Control Platform ............................................................................. 411
Connection, Dialog, or Conference Events....................................... 412
Media Controller Events.................................................................... 414
Log_4 (INFO) Events ........................................................................ 416
Resource Manager ................................................................................ 417
CTI Connector ....................................................................................... 422
CTI Adaptor ...................................................................................... 423
CTI Client.......................................................................................... 425
Supplementary Services Gateway ........................................................ 426
PSTN Connector ................................................................................... 428
Dialogic Manager.............................................................................. 428
Gateway Manager ............................................................................ 429
Media Manager................................................................................. 431
PSTN Connector............................................................................... 432
Fetching Module .................................................................................... 432
Appendix B Media Control Platform Reference Information............................... 435

Audio and Video File Formats ............................................................... 435
Audio-Only Formats—Play ............................................................... 435
Video-Only Formats—Play ............................................................... 438
Combined Audio and Video Formats—Play .......................................... 438
Audio-Only Formats—Record........................................................... 439
Video-Only Formats—Record........................................................... 441
Combined Audio and Video Formats—Record................................. 441
Dynamic Media Control Platform Parameters ....................................... 442
CPA Configuration Options That Can be Overwritten ........................... 443
SIP Headers .......................................................................................... 445
Handling Error Responses for Outbound Calls ..................................... 449
VAR Metrics........................................................................................... 450
Appendix C Tuning Call Progress Detection ........................................................ 455

Call Progress Detection......................................................................... 455
Supported North American SIT Tones .............................................. 456
Tone Definition .................................................................................. 456
Answering Machine Detection .......................................................... 460
Beep Detection ................................................................................. 462
Continuous Tone Detection............................................................... 463
Appendix D SIP Response Codes.......................................................................... 465

SIP Responses to Inbound Calls........................................................... 465

Appendix E Device Profiles.................................................................................... 475
Device Profile Usage ............................................................................. 475
Sending SDP .................................................................................... 475
Inbound Usage Examples................................................................. 475
Outbound Usage Examples .............................................................. 481
Receiving SDP.................................................................................. 483
Configuring Device Profiles ................................................................... 484
Device Profile Configuration File ...................................................... 484
Customizing Device Profiles ............................................................. 489
Default Device Profiles .......................................................................... 491
Appendix F VAR API ............................................................................................... 495

Overview................................................................................................ 495
VAR Records ......................................................................................... 495
VoiceXML <log> Extensions.................................................................. 497
Appendix G Video Support ..................................................................................... 503

Overview................................................................................................ 503
Supported Protocols and Specifications ................................................ 503
Video Features ...................................................................................... 504
VoiceXML Features .......................................................................... 504
Advanced Features........................................................................... 506
Appendix H Custom Log Sinks .............................................................................. 509

Overview................................................................................................ 509
Log Sink Interface.................................................................................. 509
Building and Linking the Library ............................................................ 514
Appendix I SSG HTTP Interface............................................................................ 515

Creating Outbound Requests ................................................................ 515
HTTP Request .................................................................................. 515
HTTP Response ............................................................................... 518
Querying Outbound Request Status...................................................... 523
HTTP Request .................................................................................. 523
HTTP Response ............................................................................... 524
Canceling Outbound Requests.............................................................. 529
HTTP Request .................................................................................. 530
HTTP Response ............................................................................... 531
SSG Database Queue Clearing During a Restart ................................. 534
Single HTTP POST (Create/Query/Cancel) .......................................... 534
User’s Guide 9
Asynchronous Result Notification.......................................................... 535
Result Notification on Success ......................................................... 535
Result Notification on Failure............................................................ 537
Root Page Access ............................................................................ 541
HTTP XML Schema............................................................................... 542
Request Schema .............................................................................. 542
Response Schema ........................................................................... 549
Appendix J Network Partitioning Configuration Options ................................... 555

Configuration Options and Protocols..................................................... 555
Appendix K SIP Customizable Headers and Parameters .................................... 559

Abstract Information from Incoming SIP Messages .............................. 559
sip.in.invite.headers .......................................................................... 560
sip.in.invite.params ........................................................................... 560
Session Variables for VXML .................................................................. 560
vxmli.session_vars............................................................................ 561
Media Control Platform ..................................................................... 561
Next Generation Interpreter .............................................................. 561
Supplements Related Documentation Resources ................................................... 565
Document Conventions ...................................................................... 569
Index ............................................................................................................... 571

List of Procedures
Viewing or modifying GVP configuration parameters . . . . . . . . . . . . . . . 30
Creating an SSL private key and certificate . . . . . . . . . . . . . . . . . . . . . . 46
Creating an SSL key and self-signed certificate for use with IIS . . . . . . 47
Creating Security Certificates for TLS Interactions. . . . . . . . . . . . . . . . . 49
Configuring the Fetching Module for HTTPS . . . . . . . . . . . . . . . . . . . . . 51
Configuring logical resource groups . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
Configuring Resource Group capabilities, preferences, and AOR . . . . . 94
Creating the resource access point for Recording Server . . . . . . . . . . . 95
Provisioning IVR Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 103
Mapping IVR Profiles to DIDs. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129
Using the DID Group Bulk Operations Wizard . . . . . . . . . . . . . . . . . . . 131
Using the Data Retention Policy Wizard. . . . . . . . . . . . . . . . . . . . . . . . 134
Provisioning ASR and TTS resources . . . . . . . . . . . . . . . . . . . . . . . . . 146
Modifying the Squid Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . 268
Enabling HTTPS for Reporting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 283
Configuring a Site folder by using Genesys Administrator . . . . . . . . . . 288
Generating a Report Using Genesys Administrator . . . . . . . . . . . . . . . 294
Generating a report using GAX . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 299
Filtering the Voice Platform Dashboard with GA . . . . . . . . . . . . . . . . . 312
Generating the Active Call Browser Report with GA . . . . . . . . . . . . . . 330
Generating the IVR Profile Call Arrivals Report with GA . . . . . . . . . . . 338
Generating the Historical Call Browser Report. . . . . . . . . . . . . . . . . . . 358
Generating the Per-Call IVR Actions Report . . . . . . . . . . . . . . . . . . . . 364
Generating the SQ Call Failures Report with GA . . . . . . . . . . . . . . . . . 368
Generating the SQ Failure Summary Report . . . . . . . . . . . . . . . . . . . . 370
Generating the SQ Latency Summary Report . . . . . . . . . . . . . . . . . . . 372
Generating the VAR Call Completion Summary Report with GA . . . . . 378
Generating the VAR IVR Action Summary Report . . . . . . . . . . . . . . . . 380
Generating the VAR Last IVR Action Report . . . . . . . . . . . . . . . . . . . . 382
Provisioning Device Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 490
User’s Guide 11
Preface
Welcome to the Genesys Voice Platform 8.5 User’s Guide. This document
provides detailed information about configuring, provisioning, and monitoring
Genesys Voice Platform (GVP) and its components.
This document is valid for the 8.5 release of this product.
Note: For versions of this document created for other releases of this
product, visit the Genesys Customer Care website, or request the
Documentation Library DVD, which you can order by e-mail from
Genesys Order Management at [email protected].
This preface contains the following sections:

 About Genesys Voice Platform, page 13

Intended Audience, page 14

Making Comments on This Document, page 14
 Contacting Genesys Customer Care, page 14

Document Change History, page 15
For information about related resources and about the conventions that are
used in this document, see the supplementary material starting on page 559.
About Genesys Voice Platform

GVP is a group of software components that constitute a robust, carrier-grade
voice processing platform. GVP unifies voice and web technologies to provide
a complete solution for customer self-service or assisted service.
In the Voice Platform Solution (VPS), GVP 8.5 is fully integrated with the
Genesys Management Framework. GVP uses Genesys Administrator, the
standard Genesys configuration and management Graphical User Interface
(GUI), to configure, tune, activate, and manage GVP processes and GVP voice
and call control applications. GVP interacts with other Genesys components
and can be deployed in conjunction with other solutions, such as Enterprise
User’s Guide 13
Intended Audience
Routing Solution (ERS), Network Routing Solution (NRS), and

Network-based Contact Solution (NbCS).
Note: GVP is a scalable solution with flexible configuration and deployment

options.
Intended Audience
This document, primarily intended for system integrators and administrators,
assumes that you have a basic understanding of:
• Computer-telephony integration (CTI) concepts, processes, terminology,
and applications.
• Network design and operation.
• Your own network configurations.
You should also be familiar with the Genesys Framework architecture.
Making Comments on This Document

If you especially like or dislike anything about this document, feel free to
e-mail your comments to [email protected].
You can comment on what you regard as specific errors or omissions, and on
the accuracy, organization, subject matter, or completeness of this document.
Please limit your comments to the scope of this document only and to the way
in which the information is presented. Contact your Genesys Account
Representative or Genesys Customer Care if you have suggestions about the
product itself.
When you send us comments, you grant Genesys a nonexclusive right to use or
distribute your comments in any way it believes appropriate, without incurring
any obligation to you.
Contacting Genesys Customer Care

If you have purchased support directly from Genesys, please contact Genesys
Customer Care.
Before contacting Customer Care, please refer to the Genesys Care Program
Guide for complete contact information and procedures.

Document Change History

This is the first release of the Genesys Voice Platform 8.5 User's Guide. In
future releases of this document, this section will list topics that are new or that
have changed significantly since the first release of this document.
User’s Guide 15

Chapter
1 Introduction
Genesys Voice Platform (GVP) is a software suite that integrates call
processing, reporting, management, and application servers with Voice over IP
(VoIP) networks, to deliver Web-driven dialog and call control services to
callers.
This chapter introduces the GVP components and Genesys Administrator, the
GUI for configuring and managing GVP. It contains the following sections:
 About GVP, page 17

Genesys Administrator, page 20

GVP Identifiers and SIP Headers, page 22
About GVP
This section describes the GVP component applications and other objects in a
GVP configuration:
• GVP Components
• IVR Profiles (see page 20)
• GVP MIBs (see page 20)
GVP Components
GVP comprises the following components:
• Resource Manager—Functions as a SIP Proxy that controls access and
routing to all resources in a GVP deployment. It also functions as a SIP
Registrar, and monitors the health of GVP resources in the deployment. Its
functions include:
 Allocates and monitors resources.

Manages sessions.

Selects services.
User’s Guide 17
Chapter 1: Introduction About GVP

Enforces policies.
• Policy Server—Provides validation and resolution of GVP-specific
business rules to Genesys Administrator through an HTTP interface with
Management Framework. Its functions include:

Manages and validates Direct Inward Dialing (DID) numbers.

Provides static analysis and validation of Resource Manager tenant and
IVR policies.
• Media Control Platform—Provides media-centric services to other GVP
components, and to third-party gateways that use GVP services. The
Media Control Platform is responsible for the execution of supported Voice
Extensible Markup Language (VoiceXML) applications. Its functions
include:

Initiates, answers, transfers, and disconnects calls.

Plays audio and Text-to-Speech (TTS) prompts.

Handles Automatic Speech Recognition (ASR) and dual tone
multi-frequency (DTMF) inputs.
 Provides conference services.
• MRCP Proxy—Acts as a proxy for all MRCPv1 Real-Time Streaming
Protocol (RTSP) resource traffic within a GVP deployment. Its functions
include:
 Provides resource management for the MRCPv1 speech resource
traffic.

Provides load balancing for MRCPv1 speech resources.

Processes periodic updates from Management Framework for its
Applications and resources.
 Sends ASR and TTS usage data for tenants, IVR Profiles, or the entire
deployment to the Reporting Server.
• Call Control Platform—Provides call control capability in accordance
with the supported W3C Call Control Extensible Markup Language
(CCXML) standard. The Call Control Platform is optional in a GVP
deployment. It operates as a SIP Back-to-Back User Agent (B2BUA) for
requests to and from GVP components. Its functions include:

Accepts, rejects, and redirects calls, including handling call setup
information to enable intelligent routing and selective answering.
Call-handling capabilities include supervised transfer, whispering, and
call hold.

Creates outbound calls through third-party gateways.
 Uses Media Control Platform services to initiate VoiceXML dialogs,
start conferences, and perform implicit transcoding.

Provides multi-party conference support with moderator and floor
control capabilities.

Provides personal assistant services, such as dialing from a personal
address book or corporate directory, managing personal appointments,
and managing voicemail and e-mail.

Chapter 1: Introduction About GVP
For information about creating CCXML applications that use Call Control
Platform capabilities, see the Genesys Voice Platform CCXML Reference.
• Reporting Server—Stores and summarizes data and statistics submitted
by Reporting Clients to provide near real-time reports by hour, day, week,
and month. Reporting Clients on the Resource Manager, Media Control
Platform, and Call Control Platform send call detail records (CDRs),
Metrics, and Operational Reporting (OR) statistics to the Reporting Server.
The Reporting Server provides an XML web services interface that is used
by Genesys Administrator to obtain GVP reporting information. The XML
web services interface is also accessible to any HTTP client, providing
customers with access to GVP reporting outside of Genesys Administrator.
For information on the real-time and historical reports, see “Monitoring
GVP” on page 291.
• CTI Connector—Provides additional computer telephony integration
(CTI) functionality by connecting to the IVR Server, which is part of the
larger Genesys Suite, through Media Control Platform. As a result, CTI
Connector remains in the call path to receive and pass call data between
IVR Server and the Media Control Platform. Its functions include:
 Routes calls to Universal Routing Server (URS).

Call processing treatments.

Transfers user data
 Transfers through CTI

Remote switch transfers

Receives statistical data
 Performs user/interaction data operations.
• Supplementary Services Gateway—Provides call processing services to
the application layer through HTTP. Its functions include:
 Outbound MCP session initiation.
• PSTN Connector—Provides connectivity to traditional telephony
environments such as Public Switched Telephone Networks (PSTN)
switches. Its functions include:
 Media Gateway functionality (uses Dialogic boards on the TDM side).

Call Progress Analysis.
 All transfers including various switch specific network transfers.
For detailed information about how the GVP components perform their
functions, see Chapter 3, “How GVP Works” in the Genesys Voice
Platform 8.5 Deployment Guide.
For information about the Genesys Voice Platform and component
architecture, see Chapter 2, “GVP Architecture” in the Genesys Voice Platform
8.5 Deployment Guide.
User’s Guide 19
Chapter 1: Introduction Genesys Administrator
IVR Profiles
Voice Extensible Markup Language (VoiceXML) and Call Control Extensible
Markup Language (CCXML) are the application-level languages that are used
to construct voice and call control applications that control the interaction
between the external user and the GVP software.
Voice and call control applications are configured as IVR Profile objects in
Genesys Administrator. The IVR Profiles define how requests received by the
VPS are translated into concrete service requests GVP which components
within the deployment can execute.
GVP MIBs
The MIB Installation Package (IP) contains the Management Information Base
(MIB) files that GVP uses to support Simple Network Management Protocol
(SNMP).
For general information about SNMP in a GVP deployment, see Chapter 2,
“GVP Architecture” in the Genesys Voice Platform 8.5 Deployment Guide. For
detailed information about the MIBs, see the Genesys Voice Platform
Troubleshooting Guide.
Genesys Administrator
Genesys Administrator is a Web-based user interface for the management and
configuration of Genesys components.
Use Genesys Administrator to deploy, configure, provision, and monitor GVP.
Figure 1 shows a typical Genesys Administrator page.

Chapter 1: Introduction Genesys Administrator
Figure 1: Genesys Administrator
To access Genesys Administrator in your Genesys deployment, go to the

following URL:
http://<Genesys Administrator host>/wcm
More Information
• For information about installing Genesys Administrator, see the
Framework 8.5 Genesys Administrator Deployment Guide.
• For general information about using Genesys Administrator, see the online
Framework 8.5 Genesys Administrator Help.
• For information about using Genesys Administrator to configure and
provision GVP Application objects and IVR Profiles, see Procedure:
Viewing or modifying GVP configuration parameters, on page 30 and
Procedure: Configuring logical resource groups, on page 90.
• For information about using Genesys Administrator to monitor GVP and
view reports, see Part 2 of this manual, starting on page 291.
Note: The Genesys Administrator’s enable and disable features for its
objects has no impact on the GVP objects.
User’s Guide 21
Chapter 1: Introduction GVP Identifiers and SIP Headers
GVP Identifiers and SIP Headers

This section explains two important categories of identifiers that are used in
GVP:
• Session Identifiers
• Application Identifiers (see page 23)
Session Identifiers
There are three types of session identifiers that are used to track, co-ordinate,
and report on GVP sessions. Table 1 describes the session identifiers and the
SIP extension headers in which the ID information is captured.
Table 1: GVP Session Identifiers and SIP Headers
Session ID Description SIP Header
Genesys CallUUID The Universally Unique Identifier (UUID) X-Genesys-CallUUID

that T-Server or SIP Server generates for The Resource Manager (SIP Proxy)
the customer interaction. and GVP components (User Agents)
propagate this header, without
changes, in all SIP messages.
GVP Session ID The 128-bit Globally Unique Identifier X-Genesys-GVP-Session-ID

(GUID), and other GVP specific If the header does not already exist
parameters separated by semicolons that in the SIP request, the Resource
identifies a call session for a particular Manager inserts the header in
GVP resource. The GUID is generated by: requests that it forwards. The
Resource Manager when it creates a new Resource Manager and GVP
session in response to a new SIP INVITE components propagate this header in
request. all subsequent SIP messages for the
Media Control Platform or Call Control session.
Platform when it initiates a new session for
an outbound call.
It is passed in the following format:
<GUID>[;gvp.rm.cti-call=1][;gvp.rm.da
tanode=<datanode>][;gvp.rm.tenant-id=
<tenant info>]

Table 1: GVP Session Identifiers and SIP Headers (Continued)
Session ID Description SIP Header
GVP Component ID The ID that the GVP component generates ccxml-session-id

to identify the call leg. For the Call Control Platform only,
The component correlates the Component when it sends a request to initiate an
ID with the GVP Session ID, and logs the outbound session, but the Resource
correlation for correct call detail records Manager session has not been
(CDRs). assigned yet. This enables a newly
For more information on the CDR reports, created CCXML session to make
see “Monitoring GVP” on page 291. multiple SIP requests before it has
received a response to any of them.
Application Identifiers
There are two kinds of applications in a GVP deployment:
• The GVP components or processes, which exist as Application objects in
the Genesys Configuration Layer.
• The VoiceXML and CCXML applications, which exist as IVR Profile
objects in the Genesys Configuration Layer.
Table 2 describes the identifiers that GVP uses for both kinds of applications,
and the SIP extension headers in which the ID information is captured.
Table 2: GVP Application Identifiers and SIP Headers
Application ID Description SIP Header
Application DBID The DBID that the Configuration Layer Not Applicable
assigns to the GVP component Application
object.
This ID is used internally by Reporting
Server to link accumulated call data and
summary data with specific GVP
components.
User’s Guide 23
Table 2: GVP Application Identifiers and SIP Headers (Continued)
IVR Profile name The user-defined name that was assigned to • gvp-tenant-id parameter in the
the IVR Profile when it was created. SIP Request-URI—The Resource
Note: For backwards compatibility: Manager uses this parameter, if it
is present, to identify the voice
If gvp-tenant-id = [TenantX], Resource or call control application for a
Manager assumes that the associated new session.
tenant for the call is TenantX.
• gvp.rm.tenant-id parameter in
If gvp-tenant-id = IVRAppY, Resource the X-Genesys-GVP-Session-ID
Manager assumes that the associated extension SIP header—The
IVR Profile for the call is IVRAppY. If Resource Manager inserts this
the X-Genesys-gsw-ivr-profile-id parameter in the header before it
header is also present, it is used to forwards the initial session
determining the IVR Profile and the request.
Request URI parameter denotes the
• X-Genesys-gsw-ivr-profile-name
tenant.
If gvp-tenant-id = [TenantX].IVRAppY,
Resource Manager assumes that the
associated tenant for the call is TenantX,
and the IVR Profile is IVRAppY. If the
X-Genesys-gsw-ivr-profile-id header
is also present, it is ignored.
IVR Profile DBID The DBID that the Configuration Layer X-Genesys-RM-Application-dbid
assigns to the IVR Profile. The Resource Manager adds this
This ID is used internally by Reporting header to the initial INVITE request
Server to link call level and summary data to a resource. Resources log the
to a specified IVR Profile. DBID in their CDRs to the
Reporting Server.
IVR Profile ID The ID that Resource Manager uses to map X-Genesys-gsw-ivr-profile-id

a new RM session to the IVR Profile.
Campaign ID for The ID that Resource Manager uses to X-Genesys-gsw-session-dbid

MSML assign an outbound campaign-id value in
order to establish a SIP session from the
same campaign on the same Media Server.
Note: The campaign-id is not mandatory for
all of the outbound modes. If an msml
service request is received without the
campaign-id, Resource Manager will route
the request to any available Media Server.

Table 2: GVP Application Identifiers and SIP Headers (Continued)
Gateway Header for The header that Resource Manager uses to X-Genesys-GVP-Trunk-Prefix
PSTN Connector inform SIP Server of the identity of the
PSTN Connector from which the call
originated.
Importing and Exporting Configuration Server Data

When data is exported from Configuration Server, and then imported back
with or without modification, be aware that the DBIDs of existing
configuration objects (such as GVP Application processes and IVR Profiles)
are not preserved when imported or exported. In these cases:
• Reporting Server will not be able to correlate historical data with the new
IDs, and GVP reports will not display older data.
• GVP components that use CCILib may encounter problems, because they
will no longer receive updates for objects for which they registered to
receive updates under the old IDs. Restarting Configuration Server will fix
this problem.
User’s Guide 25

Part
1 Provisioning GVP
This part of the Guide provides information about Genesys Voice Platform
(GVP) configuration and provisioning that you perform on the Provisioning
tab of Genesys Administrator.
This information appears in the following chapters:
• Chapter 2, “Configuration and Provisioning Overview,” on page 29
• Chapter 3, “Configuring Common Features,” on page 37
• Chapter 4, “Configuring the Resource Manager,” on page 83
• Chapter 5, “Configuring Policy Server,” on page 97
• Chapter 6, “Provisioning IVR Profiles,” on page 103
• Chapter 7, “Configuring the Media Control Platform,” on page 141
• Chapter 8, “Configuring the MRCP Proxy,” on page 195
• Chapter 9, “Configuring the Call Control Platform,” on page 209
• Chapter 10, “Configuring the CTI Connector,” on page 225
• Chapter 11, “Configuring the Supplementary Services Gateway,” on
page 245
• Chapter 12, “Configuring the PSTN Connector,” on page 255
• Chapter 13, “Configuring the Fetching Module and Squid Proxy,” on
page 265
• Chapter 14, “Configuring the Reporting Server,” on page 271
• Chapter 15, “Configuring GVP in Multi-Site Environments,” on page 287
User’s Guide 27
:

Chapter
2 Configuration and
Provisioning Overview
This chapter provides an overview of the tasks to configure Genesys Voice
Platform (GVP) components and provision GVP. It contains the following
sections:

Configuring GVP, page 29
 Task Summary: Configuring GVP, page 33
Note: This guide assumes that you have deployed a basic GVP as described
in the Genesys Voice Platform 8.5 Deployment Guide. For more
information about installing GVP components and providing the basic
connections, see the Deployment Guide
Configuring GVP
The GVP components are configured as Application objects in the Genesys
Configuration Layer. To deploy the Voice Platform Solution (VPS), you must
create and configure the required Application objects in Genesys
Administrator. For information about creating and deploying the GVP
Applications, see the Genesys Voice Platform 8.5 Deployment Guide.
To process calls in GVP 8.5, you must provision the IVR Profiles in Genesys
Administrator. To trigger the execution of a particular VoiceXML or CCXML
application when an incoming call is received, map the IVR Profile to a DN
range. For more information about provisioning IVR Profiles and, if required,
mapping them to DNs, see Chapter 6 on page 103.
User’s Guide 29
Chapter 2: Configuration and Provisioning Overview Configuring GVP
Warning! Genesys recommends, that in a multi-tenancy deployment, the

service provider or enterprise manager of GVP should be the only
user that can manage DID groups or define Tenants. Tenants
should not be given access to edit their own configurations
because of a potential conflict in numbering and naming
uniqueness required by GVP. However, Tenant users can be
assigned to read their configurations and reports.
Configuring GVP Processes in Genesys Administrator

The following procedure describes how to configure GVP Application and IVR
Profile objects in Genesys Administrator. For more information about using
Genesys Administrator, see the online Framework 8.5 Genesys Administrator
Help.
Procedure:
Viewing or modifying GVP configuration parameters
Purpose: To describe the general method for using Genesys Administrator to

view or modify configuration options in GVP Application and IVR Profile
objects.
Prerequisites
• The Application or IVR Profile object has been created as described in the
Genesys Voice Platform 8.5 Deployment Guide. In particular, for GVP
Application objects, the Application was created from an Application
Template into which metadata had been imported.
• You are logged in to Genesys Administrator. To access Genesys
Administrator, go to the following URL:
Start of procedure
1. In Genesys Administrator, go to the Options tab of the object that you want
to configure:
• For a component Application, go to the Provisioning > Environment
> Applications > <Component Application> > Options tab.
• For an IVR Profile, go to the Provisioning > Voice Platform > IVR
Profile > Options tab.
Figure 2 shows the Options tab.

Figure 2: Genesys Administrator Options Tab
For each configurable parameter, the Options tab displays the following
information:
• The option display name.
• The configuration section that contains the option.
• The configuration option name provided by the template.
• The current option value, either user-defined or default.
2. You can change the display in a number of ways:
• To sort the information in ascending or descending order by column,
click the column header, and then select the desired sort order option
from the drop-down list.
• To show or hide a column, click any column header, select the Columns
submenu from the drop-down list, and then select or clear check boxes
in the Columns list to show or hide columns.
3. To change an option setting:
a. Double-click the Value of the option that you want to change.
You can view the same information for all configuration options in the
Genesys Voice Platform 8.5 Configuration Options Reference.
b. Enter the new value in the Value field.
User’s Guide 31
4. To save your changes, click Save and then Apply.

5. To update the metadata descriptions (from an updated Templates XML
file), click Reload. This reloads the metadata file without affecting
configured option values.
6. To view the Help documentation for an option, click the blue arrow to the
left of the option name.
End of procedure
Table 3 provides information about the options in the rptui configuration
section of the default Application object. Table 3 provides parameter
descriptions as well as the default parameter values that are preconfigured in
the default Application object.
The default Application object is created automatically and is always
available when you start Genesys Administrator.
Table 3: Reporting UI Configuration Options—default Application
Option Name Description Valid Values and Syntax
Minimum Dashboard The minimum refresh interval, in seconds, to An integer greater than zero.
Refreshing Interval refresh the Voice Platform Dashboard data. Default value: 10
Maximum Number of The maximum number of IVR Profiles and/or An integer range greater than
Items in the Components that can be filtered at one time. zero.
Dashboard Default value: 50
Daylight Saving The daylight savings time difference, in hours <HH>:<mm>

Hours and minutes, to apply to timestamps to adjust to where:
local time.
• <HH> indicates hours.
When specifying the value, use leading zeros if
• <mm> indicates minutes.
necessary.
Default value: 01:00
Reporting Server The timeout, in seconds, for communications Any positive integer.
HTTP Timeout between GVP Reports and the Reporting Server. Default value: 30 (seconds)
If your deployment experiences frequent
timeouts, increase this value.

Chapter 2: Configuration and Provisioning Overview Task Summary: Configuring GVP
Table 3: Reporting UI Configuration Options—default Application (Continued)
Show Local Time Specifies whether GVP Reports will display date • true (1)—Date and time
and time values in local time, rather than in values will display in
Greenwich Mean Time (GMT), which is the local time.
default format that Reporting Server returns. • false—Date and time
To display local time in reports, set this option to values will display in
true (1) and specify a timezone offset (see GMT.
Timezone Offset). Default value: true
Timezone Offset The time offset, in hours and minutes, that will be <s><HH>:<mm>
applied to convert GMT to local time (see Show where:
Local Time), in the time zone where GVP reports
will be accessed. • <s> is either + (plus) or -
(minus), to indicate
Dates and times in all GVP reports will be whether the time should
converted. be added or subtracted.
When specifying the value, use leading zeros if • <HH> indicates hours.
necessary.
• <mm> indicates minutes.
Default value: -08:00
Task Summary: Configuring GVP

Task Summary: Configuring and Provisioning GVP provides an overview of
the tasks to implement full GVP functionality in your deployment.
Task Summary: Configuring and Provisioning GVP
Objective Related Procedures and Actions
Set up connections, SIP See:

communications, and routing • “Configuring SIP Communication and Routing” on page 38.
between the Resource Manager and
all the other GVP components. • For secure communications, see “Enabling Secure
Communication” on page 42.
See also component-specific requirements:
• For the Media Control Platform, see Table Task Summary: on
page 142.
• For the Call Control Platform, see Table Task Summary: on
page 209.
• For the CTI Connector, see “Configuring the CTI Connector”
on page 225.
User’s Guide 33
Task Summary: Configuring and Provisioning GVP (Continued)
Provision the resources for the See “Configuring Logical Resource Groups” on page 89.
Resource Manager.
Provision the IVR Profiles. See Chapter 6 on page 103.

• For additional specific configuration for GVPi IVR Profiles,
see
• To modify legacy VoiceXML applications to work with GVPi,
see the Genesys Voice Platform Application Migration Guide.
(Optional) Deploy Policy Server • Enables Genesys Administrator to manage DID Groups
and Resource Manager policies.
Enable GVP Reporting. • Configure the options in the ems configuration section of the
Resource Manager, Media Control Platform, Call Control
Platform, Fetching Module, and CTI Connector Application
objects. For more information, see “Configuring Reporting”
on page 59.
• Configure the Reporting Server (see Chapter 14 on page 271).
• If required, configure access control for Reporting services
(see “Controlling Access to Reporting Services” on
page 283).
• On the Monitoring tab of Genesys Administrator, verify that
the Voice Platform view appears in the navigation panel. If
necessary, modify the default (Configuration Manager)
Application configuration to enable GVP reports to be
displayed in Genesys Administrator. For more information,
see “” on page 285.
(Optional) Enable Automatic Speech See “Enabling ASR and TTS” on page 146.
Recognition (ASR) and To configure the MRCP Proxy, see Chapter 8 on page 195.
Text-to-Speech (TTS).
(Optional) Enable conferencing. See “Enabling Conference Services” on page 58.

Task Summary: Configuring and Provisioning GVP (Continued)
(Optional) Configure individual In general, see the remaining chapters in the Provisioning part of
components to customize or enable this guide. More specifically, to customize:
GVP features. • Logging behavior, see “Service Quality Analysis (SQA)” on
page 61.
• Session behavior and performance, see “Configuring Session
Timers and Timeouts” on page 76.
• Messaging, see “Configuring SNMP” on page 68 and
Table 100 on page 466.
• Call Control Platform device profiles, see “Configuring
Device Profiles” on page 484.
• Caching behavior, see “Configuring the Squid Caching
Proxy” on page 268.
User’s Guide 35

Chapter
3 Configuring Common
Features
This chapter describes how to implement functionality that is shared across all
the components in a Genesys Voice Platform (GVP) deployment. It contains
the following sections:

Configuring SIP Communication and Routing, page 38
 Enabling Secure Communication, page 42

Enabling IPv6 Communication, page 53

Enabling Conference Services, page 58
 Configuring Reporting, page 59

Configuring Logging, page 62

Configuring SNMP, page 68
 Configuring Client-Side Connections, page 68

Customizing SIP Responses, page 74
 Configuring Session Timers and Timeouts, page 76
This chapter describes selected configuration options (parameters) that are
common to GVP components. Later chapters similarly highlight important
configuration options that are more component-specific.
Note: Configuration options and parameters are one in the same, and these
terms are used interchangeably throughout the chapter.
The configuration option tables in this chapter provide parameter descriptions,

and also the default parameter values that are preconfigured in the GVP
Application objects. For information about all the available configuration
parameters, see the Genesys Voice Platform 8.5 Configuration Options
Reference.
Configure the options in Genesys Administrator on the Provisioning >
Environment > Applications > <GVP Component> > Options tab. For detailed
User’s Guide 37
Chapter 3: Configuring Common Features Configuring SIP Communication and Routing
steps to configure option settings, see Procedure: Viewing or modifying GVP

configuration parameters, on page 30.
Note: The configuration options for GVP processes are complex and provide
a great deal of flexibility. Deploying GVP as described in the Genesys
Voice Platform 8.5 Deployment Guide provides a fully functional,
basic GVP deployment, with the minimum number of customizations
required for GVP to operate in your environment. Before performing
additional customizations, ensure that you review the configuration
options and fully understand their implications.
Configuring SIP Communication and

Routing
Task Summary: Configuring SIP Communications and Routing summarizes
the steps and parameters to configure the transport and routing mechanisms for
SIP messaging within the GVP deployment.
Task Summary: Configuring SIP Communications and Routing
For each Resource Manager, Media Configure the sip.transport.<x> options (see page 40). Note
Control Platform, Call Control the following:
Platform, and CTI Connector • For the Resource Manager, specify separate transports for SIP
Application in your deployment, proxy, registrar, and monitoring purposes.
configure the SIP transports for the
• The lowest <x> in a set of sip.transport.<x> options
supported transport protocols.
indicates the preferred default protocol. By default, User
Datagram Protocol (UDP) is the preferred protocol for all
components (sip.transport.0).
• To make TCP the preferred protocol, either reorder (by
renaming) the respective sip.transport parameters, or else
remove the sip.transport.0 UDP parameter so that the
sip.transport.1 TCP parameter is the lowest numerically
defined sip.transport.<x> and thus, becomes the default.

Task Summary: Configuring SIP Communications and Routing (Continued)
For each Resource Manager, Media • For secure SIP (SIPS) communications, specify a transport for
Control Platform, Call Control TLS. For more information, see “Enabling Secure
Platform, and CTI Connector Communication” on page 42.
Application in your deployment, • For setting IP DiffServ (ToS) field in the outgoing SIP
configure the SIP transports for the messages, specify the ToS parameter:
supported transport protocols. sip.transport.<x>.tos
(continued)
Note: If you change the default preferred protocol for the Call
Control Platform, you must complete additional steps on the
CCXML application side, to ensure that the Request-URI
specifies the correct protocol. For more information, see
page 211.
Configure the route set and routing • For each Media Control Platform and Call Control Platform
table for outbound calls. Application in your deployment, configure the sip.routeset
or sip.securerouteset option.
• For each Resource Manager, Media Control Platform, and
Call Control Platform Application in your deployment,
configure the required sip.route.dest.<n> entries.
Verify settings that determine Review and, if necessary, modify the options that control such
behavior in relation to the SIP stack. parameters as number of threads, size of the Maximum
Transmission Unit (MTU) of the network interfaces, and number
of connections:
• For the Resource Manager, the relevant options are in the
proxy configuration section.
• For the Media Control Platform and Call Control Platform,
the relevant options are in the sip configuration section.
Table 4 provides information about important SIP communications and routing

options. It includes parameter descriptions as well as the default parameter
values that are preconfigured in various configuration sections in the Resource
Manager, Media Control Platform, and Call Control Platform Application
objects.
The default values for the sip.transport.<n> parameter in the Resource
Manager, Media Control Platform, Call Control Platform, and CTI Connector
Applications are summarized in Table 4.
For information about all the available configuration options, see the Genesys
Voice Platform 8.5 Configuration Options Reference.
User’s Guide 39
Table 4: Default SIP Transports
Component Application Section.Option Name Default Value
Resource Manager proxy.sip.transport.0 transport0 udp:any:5060
proxy.sip.transport.1 transport1 tcp:any:5060
proxy.sip.transport.2 transport2 tls:any:5061 cert=

$InstallationRoot$\config\x509_cert
ificate.pem key=$InstallationRoot$\
config\x509_private_key.pem
Note: If host names are used in your deployment, the

proxy.sip.transport.[x] must be configured with that hostname—for
example, proxy.sip.transport.0 = transport0
udp:myhostname.com:5060.
registrar.sip.transport.0 transport0 udp:any:5062
registrar.sip.transport.1 transport1 tcp:any:5062
registrar.sip.transport.2 transport2 tls:any:5063 cert=

monitor.sip.transport.0 transport0 udp:any:5064
monitor.sip.transport.1 transport1 tcp:any:5064
monitor.sip.transport.2 transport2 tls:any:5065 cert=

subscription.sip.transport.0 transport0 udp:any:5066
subscription.sip.transport.1 transport1 tcp:any:5066
subscription.sip.transport.2 transport2 tls:any:5067 cert=


Table 4: Default SIP Transports (Continued)
Media Control Platform sip.transport.0 transport0 udp:any:5070

Note: If all sip.transport.x
sip.transport.1 transport1 tcp:any:5070
values are empty, UDP, TCP,
and TLS transports will all be sip.transport.2 transport2 tls:any:5071 cert=
enabled, and will listen from $InstallationRoot$\config\x509_cert
ports 5070, 5070, and 5071, ificate.pem key=$InstallationRoot$\
respectively, on any network config\x509_private_key.pem
interface.
MRCP V2 Client sip.transport.0 transport0 udp:any:7080

sip.transport.1 transport0 udp:any:7080
and TLS transports will all be sip.transport.2 transport2 tls:any:7081
enabled, and will send MRCP type=TLSv1
session request from ports
7080, 7080, and 7081,
respectively, on any network
interface.
Call Control Platform sip.transport.0 transport0 udp:any:5068

sip.transport.1 transport1 tcp:any:5068
enabled, and will listen from cert=$InstallationRoot$/config/x509
ports 5068, 5068, and 5069, _certificate.pem
respectively, on any network key=$InstallationRoot$/config/x509_
interface. private_key.pem
mediacontroller.sipsproxy The address of SIP Secure Proxy for

outbound SIP requests. Specify in this
format: 10.10.30.205:5071
Default: $LocalIP$:5061
mediacontroller.bridge_ Address of sip secure bridge server

sips_server
Default: $LocalIP$:5061
mediacontroller. If this flag is set to true, all the outbound

sipsecure sip requests would be in SIP Secure
protocol. Note that the hints attribute of
the CCXML elements that initiates an
outbound request can overwrite this
configuration. Possible Values:
1 - true
0 - false (default)
User’s Guide 41
Chapter 3: Configuring Common Features Enabling Secure Communication
Table 4: Default SIP Transports (Continued)
CTI Connector sip.transport.0 transport0 udp:any:5080

Note: If all sip.transport.x sip.transport.1 transport1 tcp:any:5080
enabled, and will listen from cert=$InstallationRoot$/config/x509
ports 5080, 5080, and 5081 _certificate.pem
respectively, on any network key=$InstallationRoot$/config/x509_
interface. private_key.pem
PSTN Connector GatewayManager.

UserAgentAddr
GatewayManager.
UserAgentPort
Enabling Secure Communication

Task Summary: Enabling SIPS, HTTPS, and SRTP in GVP summarizes the
steps and parameters to set up your GVP deployment to use Secure Socket
Layer (SSL) technology for secure SIP (SIPS), secure HTTP (HTTPS), and
secure RTP (SRTP) communication.
Note: Although, the GVP components support SIPS, the Genesys SIP Server
does not. Before you enable SIPS in your GVP deployment, contact
your Genesys Sales Representative for more information.

Task Summary: Enabling SIPS, HTTPS, and SRTP in GVP
Set up GVP to use SIPS for call 1. If required, generate and deploy the SSL private key and
control messaging. certificate (see Procedure: Creating an SSL private key and
certificate).
2. On the Resource Manager, Media Control Platform, Call
Control Platform, and CTI Connector Applications, specify
the SIP transport for TLS, including the additional parameters
for the certificate and key (see information about the
sip.transport.<x> option on pages 38, and the default values
in Table 4 on page 40).
3. On the Media Control Platform and Call Control Platform
Applications, specify secure routing for outbound calls (see
information about the sip.securerouteset option on page 40
and the sip.route.dest.<n> option on page 40).
4. On the Call Control Platform Applications, specify secure SIP
proxy to generate dialogs by using TLS calls (see information
about the mediacontroller.sipsproxy option on page 41.
5. Modify the CCXML applications, as required, to ensure that
the Request-URI specifies TLS as the transport protocol.
Set up the Fetching Module to use 1. Generate and deploy the SSL private key and certificate. For
HTTPS. information about creating a self-signed certificate, see
Procedure: Creating an SSL key and self-signed certificate for
use with IIS, on page 47.
2. On the Media Control Platform or Call Control Platform
Applications in your deployment, configure the https_proxy,
and the ssl_* parameters in the fm section.
3. If Squid is deployed, modify the Squid configuration file, if
necessary, to configure “safe” and SSL ports, and to enforce
SSL (see Procedure: Modifying the Squid Configuration, on
page 268). Also, if the HTTPS connection is to tunnel through
Squid or another HTTP proxy, configure the https_proxy
parameter in the fm section.
Verify that timeout settings are Given the additional processing time and lags associated with
suitable for your deployment. SSL encryption/decryption and handshakes, reconsider the
following settings in particular:
• For the Fetching Module, iproxy.connect_timeout (default is
5 seconds).
• For the Media Control Platform, timeouts in the sessmgr and
sip sections.
User’s Guide 43
Task Summary: Enabling SIPS, HTTPS, and SRTP in GVP (Continued)
Enable SRTP for the media channel On the Media Control Platform Application:
between the Media Control Platform 1. Specify the required mode (accept-only, offer, or
and the remote endpoint. offer_strict) in the mpc.srtp.mode parameter. By default,
SRTP is not enabled.
2. If necessary, modify the default values for the encryption and
authentication algorithms (the cryptographic suites), and for
the session parameters that the Media Control Platform will
advertise in the SDP crypto attribute:
mpc.srtp.cryptomethods
mpc.sessionparams
mpc.sessionparamsoffer
Enable SRTP for the media channel On the MRCPv2 Application that represents the third-party
between the MRCPv2 server and the MRCP server for ASR or TTS, verify and, if required, modify
Media Control Platform. settings for the following options:
provision.vrm.client.TlsCertificateKey
provision.vrm.client.TlsPrivateKey
provision.vrm.client.TlsPassword
Create security certificates to enable On Windows and Linux install and configure security certificates
the Supplementary Services Gateway to enable interactions over TLS. See Procedure: Creating
to interact with SIP Server over Security Certificates for TLS Interactions, on page 49.
secure ports.
If necessary, set up the Reporting See “Enabling HTTPS for Reporting” on page 283.
Server to use HTTPS.
Configure Genesys Administrator to • In Genesys Administrator, on the Provisioning >

use HTTPS to access Reporting Environment > Applications > default > Options tab, set
Server web services, for the GVP the value of rptui.enablehttps to true.
reports that are displayed in the • Ensure that the web server for Reporting Server is configured
Monitoring > Voice Platform view. to enable HTTPS.
Note: Observe standard security practices to ensure that you protect the
security of SSL private keys, SSL certificates, and configured user
names and passwords—for example, ensure that they are stored on
secure hosts, and do not create them over a network.
Configuring MCP, MRCPv2, CCP, CTIC, and RM for Secure SIP

Transport
This section offers configuration examples that enable secure communication.

MCP
Inbound Inbound supports both sips: and transport=TLS schemas. Examples:
INVITE
sips:[email protected]:5071;voicexml=http://000.00.000.00/testca
se/gvp8/hellotransfer.vxml;aai=N/A SIP/2.0
INVITE
sip:[email protected];transport=TLS;voicexml=http://000.00.000.0
0/testcase/gvp8/hellotransfer.vxml;aai=N/A SIP/2.0
Transfer Transfer can make calls with both and “transport=tls”. Examples:
<transfer name="newcall" dest="sips:[email protected]:5071"

bridge=
sent: INVITE sips:[email protected]:5071 SIP/2.0
sent: REFER sips:[email protected]:5071 SIP/2.0
<transfer name="newcall"
dest="sip:[email protected]:5071;transport=tls"bridge=
sent: INVITE sip:[email protected]:5071;transport=tls SIP/2.0
sent: REFER sip:[email protected]:5071;transport=tls SIP/2.0
MRCPv2
Nuance Speech Server 6 supports only the sips: schema. Example:
sips:mresources@[MRCP server IP]:[port]
Nuance Speech Server 5 supports the transport=tls schema. Example:
sip:mresources@[MRCP server IP]:[port] ; transport=TLS
Call Control Platform (CCP)

Inbound CCP supports sip secure schema sips: and transport=TLS
The example below applies to both:
INVITE
sips:[email protected]:5069;ccxml=http://000.00.000.00/testcase/g
vp8/dialog.ccxml;
INVITE
sip:[email protected]:5069;transport=TLS;ccxml=http://000.00.000.
00/testcase/gvp8/dialog.ccxml;
User’s Guide 45
CTI Connector
When CTI Connector receives an incoming call on a secure channel, it will use
only a secure channel to make the outbound call.
For example:
INVITE sips:[email protected]:5080 SIP/2.0
INVITE sip:[email protected]:5080;transport=TLS SIP/2.0
Note: We have not tested CTI Connector behavior by modifying the
mediacontroller.sipsecure parameter. I will check with QA about this and
revert back to you.
Resource Manager
When the Resource Manager receives an incoming call on a secure channel, it
will use only a secure channel to make the outbound call.
Procedures that Support Enabling Secure Communication

The following procedures support the tasks outlined in Task Summary:
Enabling SIPS, HTTPS, and SRTP in GVP:
• Procedure: Creating an SSL private key and certificate
• Procedure: Creating an SSL key and self-signed certificate for use
with IIS, on page 47
• Procedure: Creating Security Certificates for TLS Interactions, on page 49
• Procedure: Configuring the Fetching Module for HTTPS, on page 51
Procedure:
Creating an SSL private key and certificate
Purpose: To illustrate how to create and deploy the private key and SSL
certificate that are used for SIPS and HTTPS authentication.
Perform this procedure for each Resource Manager, Media Control Platform,
and Call Control Platform in your deployment.
Prerequisites
• The OpenSSL Toolkit (openssl) or other SSL tool is available.
You can download the OpenSSL Toolkit for Windows from Shining Light
Productions at the following URL:
http://www.shininglightpro.com/products/Win32OpenSSL.html
For more information about OpenSSL, see http://www.openssl.org/.

Start of procedure
1. Generate the private key:
• For a password-protected key, execute the following command:
openssl genrsa -aes128 -out x509_private_key.pem 2048
• For a non-password-protected key, execute the following command:
openssl genrsa -out x509_private_key.pem 2048
2. Generate the certificate.
The following example of the required command creates a certificate with
file name x509_certificate.pem, which expires in 1095 days:
openssl req -new -x509 -key x509_private_key.pem -out
x509_certificate.pem -days 1095
For information about additional supported parameters, see the openssl
Manual page on the OpenSSL web site (http://www.openssl.org/).
3. Install the certificate and key.
The default GVP configuration assumes that the file names and paths are
as follows:
• For the certificate:
$InstallationRoot$\config\x509_certificate.pem
• For the private key:
$InstallationRoot$\config\x509_private_key.pem
End of procedure
Next Steps
• If required, modify the sip.transport.<x> configuration option for TLS to
update the parameters for the certificate path, key path, and password (if
applicable).
Procedure:
Creating an SSL key and self-signed certificate for use
with IIS
Purpose: To illustrate how to use the OpenSSL Toolkit to create a private key
and self-signed SSL certificate request, to enable HTTPS connections to the
IIS web server for Fetching Module communications.
Prerequisites
• The OpenSSL Toolkit (openssl) has been installed, with default settings.
You can download the OpenSSL Toolkit for Windows from Shining Light
Productions at the following URL:
User’s Guide 47
http://www.shininglightpro.com/products/Win32OpenSSL.html
For more information about OpenSSL, see http://www.openssl.org/.
Start of procedure
1. Set up the openssl directories and files:
a. (Optional, but recommended) Add C:\OpenSSL\bin to your system path
(Control Panel > System > Advanced > Environment Variables >
System Variables).
b. Create a working directory—for example, C:\ssl.
c. Create the directory structure and files required by openssl:
• Directories: keys, requests, and certs
• Files: database.txt and serial.txt—these are empty (zero-byte)
text files
To create the directories and files manually, execute the following
commands at the C:\ssl> UNIX prompt:
md keys
md requests
md certs
copy con database.txt
^Z
copy con serial.txt
01
^Z
2. Set up a Certificate Authority (CA):
a. At the C:\ssl> prompt, execute the following command to create a
1024-bit private key:
openssl genrsa -des3 -out keys/ca.key 1024
b. At the C:\ssl> prompt, execute the following command to create the
CA certificate:
openssl req -config openssl.conf -new -x509 -days 1001 -key
keys/ca.key -out certs/ca.cer
The following certificate is created:
c:\ssl\certs\ca.cer
3. Create an IIS Certificate Request (certreq.txt).
For more information, see the Microsoft Knowledge Base article number
228821, which is available from Microsoft Technical Support
(http://support.microsoft.com).
4. Sign the Certificate Request.
a. Copy the certreq.txt file into C:\ssl\requests.

b. At the C:\ssl> prompt, execute the following command to sign the

request:
C:\ssl>openssl ca -policy policy_anything -config openssl.conf
-cert certs/ca.cer -in requests/certreq.txt -keyfile keys/ca.key
-days 360 -out certs/iis.cer
5. Install the new certificate under IIS.
For more information, see the Microsoft Knowledge Base article number
228836, which is available from Microsoft Technical Support
(http://support.microsoft.com).
The secure web server is now accessible from any web browser, using
SSL.
End of procedure
Next Steps
• Create security certificates for TLS interactions (if required). See
Procedure: Creating Security Certificates for TLS Interactions.
Procedure:
Creating Security Certificates for TLS Interactions
Purpose: To create security certificates to enable the Supplementary Services

Gateway to interact with SIP Server by using TLS through secure ports.
Summary
The Security Pack on Linux provides the components, such as shared libraries
and an example of a Certification Authority (CA), that are used to generate
certificates and to deploy them on the installed GVP components.
Start of procedure
On Windows 1. Import the ca_cert.pem file to the Trusted Root Certificate Authorities
folder.
2. Import the <serial_#>_<host_name>_cert.pfx file to the Personal folder of
the certificate service.
On Linux 3. Install the Security Pack for Linux.
4. Configure the environment variable that corresponds to Linux and specify
the path to Security Pack libraries. For example, export
LD_LIBRARY_PATH=/home/svetar/Security/Linux/SecurityPack_810
The certificate is generated and added to the application.
On Windows and 5. In Genesys Administrator, assign the certificate to the host machine and
Linux the host application.
User’s Guide 49
6. Create a secure port for the SIP Server Application.

For a detailed description of the configuration that is required to assign the
certificate to the host machine, application, and port, see Chapter 18 in the
Genesys8.5 Security Deployment Guide.
7. Assign the certificate to the secure port.
SIP Server will listen on both the default and the secure port.
8. In Genesys Administrator, in the Supplementary Services Gateway
Application, create a connection to SIP Server.
9. In the Connection Info dialog box, edit the properties to configure the
ports to which the Supplementary Services Gateway will connect to SIP
Server. See Figure 3 on page 50.
For a detailed description of how to install and configure security certificates,
see Chapters 16-18 in the Genesys Security Deployment Guide
Figure 3: Connection Info Dialog Box in Genesys Administrator
End of procedure
Next Steps
• Configure the Fetching Module (pwproxy) to access files over HTTPS (see
Procedure: Configuring the Fetching Module for HTTPS).
Note: The following procedure is applicable for pre Genesys Voice Platform
8.1.2 only deployments.

Procedure:
Configuring the Fetching Module for HTTPS
Purpose: To modify the default Fetching Module configuration to enable

secure HTTPS communications.
Perform this procedure on each Fetching Module in your deployment.
Prerequisites
• The SSL certificate and key have been created and installed under IIS.
For information about creating a self-signed certificate, see Procedure:
Creating an SSL key and self-signed certificate for use with IIS, on
page 47.
Start of procedure
1. Create the PEM certificate file:
a. Using a text editor, create a new file, proxy_client.pem. You can store
the file under any directory on the Fetching Module server.
b. Open the ca.key file created by openssl (see Step 2 on page 48).
c. Copy all the lines from ca.key into the new proxy_client.pem file.
d. Press Enter to create one blank line at the end of the text.
e. Open the iis.cer file created when you signed the certificate request
(see Step 4 on page 48).
f. Copy all the lines from iis.cer starting from -----BEGIN
CERTIFICATE---- and ending with -----END CERTIFICATE-----,
inclusive, into the bottom of the proxy_client.pem file.
The final file will look similar to Figure 4.
User’s Guide 51
-----BEGIN RSA PRIVATE KEY-----

Proc-Type: 4,ENCRYPTED
DEK-Info: DES-EDE3-CBC,70D8F72D9BB079C3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-----END RSA PRIVATE KEY-----
-----BEGIN CERTIFICATE-----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-----END CERTIFICATE-----
Figure 4: Sample PEM Certificate File
2. Save the file.

3. In Genesys Administrator, on the Provisioning > Environment >
Applications > <Fetching Module> > Options tab, modify the Fetching
Module configuration:
a. Verify that the value of the fm.https_proxy parameter is empty
(disabled—encrypted pages will not be cached).
b. Configure the following options in the iproxy configuration section:
• ssl_key_passwd = <Your private key passphrase>

Chapter 3: Configuring Common Features Enabling IPv6 Communication
• ssl_key = <Local path to your proxy_client.pem file>

• ssl_cipher_list = TLSv1
For information about other SSL-related configuration options for the
Fetching Module, see the Genesys Voice Platform 8.5 Configuration
Options Reference (options beginning with ssl_ in the iproxy section).
c. Click Save or Apply to save the configuration changes.
4. Restart the Fetching Module application.
End of procedure
Next Steps
• If required, modify the Squid configuration file to identify the “safe” ports
for HTTP and SSL requests, to identify the ports to be used for SSL
connections, and to deny access to non-SSL connections.
For more information, see Procedure: Modifying the Squid Configuration,
on page 268
Enabling IPv6 Communication

Table 5 contains a list of the configuration options that you should be aware of
if you intend to support Internet Protocol version 6 (IPv6) communication in
your environment. These options can be used to customize your GVP
configuration.
Notes: Although, the GVP components supports IPv6, the Genesys SIP
Server and Cisco T-Servers do not. Before you enable IPv6 in your
GVP deployment, contact your Genesys Sales Representative for
more information.
GVP components support non-linked-local IPv6 addresses only.
When using IPv6, do not use linked-local addresses.
Tip: When using IPv4-mapped IPv6 addresses, be aware that not all
operating platforms function in the same way. While IPv4-mapped IPv6
addresses might function on a Windows platform, they might not
function on a Linux platform.
User’s Guide 53
Table 5: Configuration Options that Support IPv6
Component Configuration option Description
Resource Manager • [proxy] sip. Use this option to specify the preferred IP version
preferred_ipversion when a destination address resolves into multiple
• [registrar] sip. IP addresses that use different IP versions. The
preferred_ipversion first IP address processed that matches the
preferred IP version is used. However, if a
• [monitor] sip.
sip.transport is not defined for the preferred
preferred_ipversion
version, a defined version that matches one of the
• [subscription] sip. processed IP addresses is used. Valid values are
preferred_ipversion ipv4 and ipv6.
• [proxy] sip. Use these options to define the transport layer for
transport.<n> the SIP stack and the network interfaces that are
• [registrar] sip. used to process SIP requests.
transport.<n>
• [monitor] sip.
transport.<n>
• [subscription] sip.
transport.<n>
• [proxy] sip.transport. Use these options to specify the sent-by field of

localaddress_ipv6 the Via header and the hostport part of the
• [registrar] sip. Contact header in an outgoing SIP message if an
transport. IPv6 transport is used The value must be a host
localaddress_ipv6 name or domain name. (Configuration of this
option is not required; it can work with the default
• [monitor] sip.
transport.localaddress value.)
_ipv6
transport.localaddress
_ipv6

Table 5: Configuration Options that Support IPv6 (Continued)
Resource Manager • [proxy] sip.route. Use these options to specify the transport that is
(continued) default.udp.ipv6 defined in the sip.transport.x configuration
• [registrar] sip.route. option, where x is the value of this option. These
default.udp.ipv6 options are used when there are no IPv6 UDP
routes found.
• [monitor] sip.route.
default.udp.ipv6
route.default.udp.ipv6
• [proxy] sip.route. Use these options to specify the transport that is

default.tcp.ipv6 defined in the sip.transport.x configuration
• [registrar] sip.route. option, where x is the value of these options.
default.tcp.ipv6 These options are used when there are no IPv6
TCP routes found.
default.tcp.ipv6
route.default.tcp.ipv6
• [proxy] sip.route. Use these options to specify the transport that is

default.tls.ipv6 defined in the sip.transport.x configuration
• [registrar] sip. option, where x is the value of this option. These
route.default. options are used when there are no IPv6 TLS
tls.ipv6 routes found.
default.tls.ipv6
route.default.tls.ipv6
Note: For a complete description of these Resource Manager options and their
default values, see Table 13 on page 85.
Media Control [mpc] Use this option to specify the preferred IP

Platform preferredipinterface interface to use (IPv4 or IPv6) when SDP is being
negotiated. This option value sets the root
connection attribute in SDP answers and sets the
connection attribute in SDP offers.
• [sip] route.default. Use these options to specify the default IPv6 route
udp.ipv6 for UDP, TCP, or TLS. The number denotes the
• [sip] route.default. transport that is defined in the sip.transport.x
tcp.ipv6 configuration option.
• [sip] route.default.
tls.ipv6
User’s Guide 55
Media Control [sip] transport.x Use this option to define the transport layer for
Platform the SIP stack and the network interfaces that are
(continued) used to process SIP requests.
[sip] preferred_ipversion Use this option to specify the preferred IP version

when a destination address resolves into multiple
IP addresses that use different IP versions. The
first IP address processed that matches the
processed IP addresses is used. Valid values are
ipv4 and ipv6.
[sip] transport. Use this option to specify the sent-by field of the
localaddress_ipv6 Via header and the hostport part of the Contact
header in an outgoing SIP message if an IPv6
transport is used The value must be a host name
or domain name. (Configuration of this option is
not required; it can work with the default value.)
Note: For a complete description of these Media Control Platform options and
their default values, see Table 23 on page 156.
Call Control [ccxmli] basichttp.recv. Use this option to specify the IPv6 address or host
Platform host.ipv6 name on which the basic HTTP event I/O
processor will listen for HTTP requests on IPv6
network interface.
[ccxmli] createsession. Use this option to specify the IPv6 address or host
recv.host.ipv6 name on which the session creation event I/O
processor will listen for HTTP requests on IPv6
network interface.
[ccxmli] createsession. Use this option to specify the preferred IP version

recv.accessuri that will be used in the create session access URI
session.ioprocessors[“createsession”].
[mediacontroller] sdp. Use this option to specify the host part of the
localhost.ipv6 local host IPv6 address that is used in SDP.
[sip] basichttp.recv. Use this option to specify the preferred IP version

accessuri that will be used in basic HTTP access URI
session.ioprocessors[“basichttp”].

Call Control • [sip] route.default. Use these options to specify the default IPv6 route
Platform udp.ipv6 for UDP, TCP, or TLS. The number denotes the
(continued) • [sip] route.default. transport that is defined in the sip.transport.x
tls.ipv6
[sip] transport.x Use this option to define the transport layer for
the SIP stack and the network interfaces that are
used to process SIP requests.

ipv4 and ipv6.
[sip] transport. Use this option to specify that the sent-by field
localaddress_ipv6 of the Via header and the hostport part of the
Contact header in the outgoing SIP message will
be set to this value if a IPv6 transport is used.
(Configuration of this option is not required; it
can work with the default value.)
Note: For a complete description of these Call Control Platform options and their
default values, see Table 26 on page 213
CTI Connector [sip] transport.x Use this option to define the transport layer for
the SIP stack and the network interfaces that are
used to process SIP requests.

ipv4 and ipv6.
User’s Guide 57
Chapter 3: Configuring Common Features Enabling Conference Services
CTI Connector • [sip] route.default. Use these options to specify the default IPv6 route
(continued) udp.ipv6 for UDP, TCP, or TLS. The number denotes the
• [sip] route.default. transport that is defined in the sip.transport.x
tls.ipv6
[sip] transport. Use this option to specify that the sent-by field
localaddress_ipv6 of the Via header and the hostport part of the
Contact header in the outgoing SIP message will
be set to this value if a IPv6 transport is used.
(Configuration of this option is not required; it
can work with the default value.)
Note: For a complete description of these CTI Connector options and their
default values, see Table 27 on page 230.
Supplementary [Common] enable-ipv6 Use this option to enable an IPv6 communication

Services Gateway between the SSG and SIP Server.
Note: For a complete description of this Supplementary Services Gateway option

and its default value, see Table 30 on page 247.
Enabling Conference Services

Task Summary: Configuring Conferencing summarizes the steps and
parameters to configure the GVP deployment to provide conference service.
Notes: You can set values for options such as conference reserve, maximums
for number of conferences and participants, and conference
capabilities can be set at the level of the resource group, the resource,
and the IVR Profile, in order of override priority. These parameters
are significant in determining how the Resource Manager handles a
particular request for conference service.
Genesys recommends that, before you modify options, you carefully
review the descriptions for all contexts (resource group, individual
resource, and IVR Profile). For more information about the
configuration options, see the remaining chapters in the Provisioning
section of this guide, and the Genesys Voice Platform 8.5
Configuration Options Reference

Chapter 3: Configuring Common Features Configuring Reporting
Task Summary: Configuring Conferencing
Assign a conference resource to a See “Configuring Logical Resource Groups” on page 89:
logical resource group that provides 1. Create or modify a logical resource group for the Resource
conference service. Manager, where the value of the <logical resource
Note: For MCP only. group>.service-types option includes conference.
2. Set the general conference maximums for the resource group
(see the description of the confmaxsize and confmaxcount
options).
3. If the resource has not already been added to the Resource
Manager connections, add it. For more information, see the
chapter about postintstallation activities in the Genesys Voice
Create an IVR Profile for conference Set the following required parameter:
service. • gvp.service-prerequisite.conference-id
Also consider the following IVR Profile options, which
determine whether and how conference service will be provided:
• gvp.general.application-confmaxsize
• gvp.general.service-type
• gvp.policy.conference-allowed
• gvp.policy.conference-capability-requirements
• gvp.policy.conference-usage-limit and
conference-usage-limit-per-session
For more information, see “IVR Profile Configuration Options”
on page 109.
Verify that conference-related • For the Media Control Platform, review the options in the
settings on the Media Control conference section.
Platform and Call Control Platform • For the Call Control Platform, verify the settings for the
are suitable. Default Conference device profile, and the options in the
mediacontroller configuration sections.
(Optional) Customize the SIP On the Resource Manager, customize the value of the
response codes and Resource rm.conference-sip-error-respcode option.
Manager behavior in the event of an For more information, see Table 100 on page 466.
error.
Configuring Reporting
This section describes important parameters for GVP Reporting, which you
configure in the ems section of the Resource Manager, Media Control Platform,
User’s Guide 59
Call Control Platform, Fetching Module, CTI Connector, UCM Connector,

MRCP Proxy, Supplementary Services Gateway, and PSTN Connector
Application objects.
For general information about Reporting in a GVP deployment, see Chapter 3,
“How GVP Works” in the Genesys Voice Platform 8.5 Deployment Guide.
For information about additional configuration options, see the Genesys Voice
Platform 8.5 Configuration Options Reference.
Except where otherwise indicated, all changes to ems configuration options
take effect after you restart the component Application.
Table 6 summarizes the default values for the various logs and metrics filters
for the component Application objects.
Note: Genesys recommends that you do not modify the log filter settings.
Table 6: Default Log and Metrics Filters
Com- Option
ponent Name (in
ems
Section)*
logconfig. logconfig. logconfig. metrics- metrics- metricsconfig.

DATAC MFSINK TRAPSINK config. config. TRAPSINK
DATAC MFSINK
MCP 0-2,4|*|* *|*|* 0-4|*|* * 0-16,18-41,43 Not Applicable

,52-56,72-74,
76-81,127-129
,130,132-141
CCP 0-2|*|* 0-3,5|*|* *|*|* * 1000-1001,100 *

3-1005,1007-1
016,1019-1021
,1024,1027-10
36,1039-1045,
1048-1050,105
2-1054,1056,1
058-1062
CTIC Not *|*|* *|*|* Not Not Not Applicable

Applicable Applicable Applicable
SSG Not *|*|* *|*|* Not Not Not Applicable

PSTNC Not *|*|* *|*|* Not Not Not Applicable


Table 6: Default Log and Metrics Filters (Continued)
Com- Option
ponent Name (in
ems
Section)*
logconfig. logconfig. logconfig. metrics- metrics- metricsconfig.

DATAC MFSINK TRAPSINK config. config. TRAPSINK
DATAC MFSINK
MRCP Not *|*|* *|*|* Not Not Not Applicable

Proxy Applicable Applicable Applicable
RM Not *|*|* *|*|* Not Not Not Applicable

Note: The MCP, CCP, MRCPP, and RM components must have a connection
to the Reporting Server application in order to collect metrics, CDR
and OR data.
Service Quality Analysis (SQA)

Table 7 describes the parameters required for Service Quality reporting in the
Media Control Platform.
Note: Service Quality reports apply to NGi VoiceXML applications, and are
found in Genesys Administrator. GVP 8.1.5 and thereafter are
NGi-only platforms unless you run MCP 8.1.4 to incorporate support
for GVPi applications.
Table 7: Service Quality Advisor Parameters
Option Name Description Valid Values and

Syntax
SQA Enable Flag Specifies whether to perform Service Quality • True

Analysis. • False
Default value: True
Inbound Reject Failure Specifies which in call reject reason codes that, A pipe (|)separated list.
Codes when encountered, do not mark the call as a Default value: decline
failure.
User’s Guide 61
Chapter 3: Configuring Common Features Configuring Logging
Table 7: Service Quality Advisor Parameters (Continued)
Option Name Description Valid Values and

Syntax
Outbound Reject Failure Specifies which out call reject reason codes that, A pipe (|)separated list.
Codes when encountered, do not mark the call as a Default value:
failure. busy|decline|fax|noanswe
r|hangup
Call Reject Latency Specifies the maximum time, in milliseconds, to Any integer.
Threshold determine whether the call reject latency is Default value: 3000
considered a failure because it falls below the
threshold.
Audio Gap Latency Specifies the largest audio gap allowed while Any integer.
playing audio to the customer. Default value: 2000
Cumulative Response Specifies the maximum threshold, in Any integer.

Latency Threshold milliseconds, before playing a prompt after Default value: 4000
customer interaction.
Inter Prompt Latency Specifies the maximum time, in milliseconds, Any integer.
Threshold before playing a prompt after playing a previous Default value: 4000
prompt when no customer interaction has taken
place.
First Prompt Outbound Specifies the maximum threshold, in Any integer.

Latency Threshold milliseconds, before playing a prompt on an Default value: 3000
outbound call.
First Prompt Inbound Specifies the maximum threshold, in Any integer.

Latency Threshold milliseconds, before playing a prompt on an Default value: 3000
inbound call.
Call Answer Latency Specifies the maximum time, in milliseconds, to Any integer.
Threshold determine whether the call answer latency is Default value: 3000
considered a failure because it falls below the
threshold.
SQA Batch Size Specifies the number of SQA messages to queue An integer in the range of
before sending them to the Reporting Server. 1–5000.
Default value: 5000
Configuring Logging
Table 8 describes the most commonly customized options for logging. Table 9
on page 67 summarizes the default values for these options in GVP. The

options are in the log configuration section of each GVP component

Application.
Configure the options for each component in Genesys Administrator on the
Provisioning > Environment > Applications > <GVP Application> > Options
tab. For the detailed steps to configure option settings, see Procedure: Viewing
or modifying GVP configuration parameters, on page 30.
Changes take effect immediately.
The Application Templates do not expose all the logging parameters that are
standard in Genesys applications; therefore, the Options tab and its metadata
(which is also described in the Genesys Voice Platform 8.5 Configuration
Options Reference) therefore do not describe all the parameters that determine
the logging behavior of GVP applications. For more information about the
additional, standard logging options, see the Log Section in the chapter about
common configuration options in the Framework 8.5 Configuration Options
Reference Manual.
User’s Guide 63
Table 8: Selected Configuration Options—log Section
all A comma-separated list of the output • stdout—Log events are

destinations to which the Application (GVP sent to the Standard
process) sends all log events. output (stdout).
Setting log.verbose to all and this parameter • stderr—Log events are
(log.all) to network enables an application to sent to the Standard error
send log events of the Standard, Interaction, output (stderr).
and Trace levels to Message Server. With this • network—Log events are
setting, Debug-level log events are not sent to sent to Message Server,
Message Server, and they are not stored in the which can reside
Log Database. anywhere on the network.
Message Server stores the
standard A comma-separated list of the output log events in the Log
destinations to which the application (GVP Database.
process) sends log events of the Standard level. • memory—Log events are
sent to the memory output
interaction A comma-separated list of the output
on the local disk. This is
destinations to which the application (GVP
the safest output in terms
process) sends log events of the Interaction of application
level and higher (that is, Standard and performance.
Interaction levels).
• <filename>—Log events
trace A comma-separated list of the output are stored in a file with
destinations to which the application (GVP the specified name. The
process) sends log events of the Trace level and default path for the file is
higher (that is, Standard, Interaction, and the working directory of
Trace levels). the application.
debug A comma-separated list of the output

destinations to which the application (GVP
process) sends log events of the Debug level and
higher (that is, Standard, Interaction, Trace,
and Debug levels).
Enable 6.x Specifies whether the application uses 6.x • true—The log of the
Compatibility Log output logic. level specified by Log
Output Priority Output options is sent to
the specified output.
• false—The log of the
level specified by Log
Output options and higher
levels is sent to the
specified output.

Table 8: Selected Configuration Options—log Section (Continued)
expire (Applicable only if log output is configured to • false—No expiration.

be sent to a log file.) All generated segments
Specifies the criteria for determining when log are stored.
files (segments) expire and are deleted. • <number>
file|<number>—A
number in the range of 1–
100 that specifies the
maximum number of log
files to store.
• <number> day—A
number in the range of 1–
100 that specifies the
maximum number of days
before log files are
deleted. (This value is not
applicable for Reporting
Server.)
Log Message Format The log record header format that an • short—The Application
Application uses when writing logs in the log uses compressed headers
file. when writing log records
Using compressed log record headers improves in the log file.
application performance and reduces the size of • full—The Application
the log file. When message_format=short uses complete headers
(compressed headers): when writing log records
in the log file.
• A header of the log file or the log file
segment contains information about the Log record examples:
application (such as the application name, • Full format:
application type, host type, and time zone), 2002-05-07T18:11:38.19
whereas individual log records within the 6 Standard localhost
file or segment do not contain this cfg_dbserver
information. GCTI-00-05060
• Log message priority is abbreviated to Std, Application started
Int, Trc, or Dbg (instead of Standard, • Short format:
Interaction, Trace, or Debug, 2002-05-07T18:15:33.95
respectively). 2 Std 05060
• The message ID does not contain the prefix Application started
GCTI or the application type ID.
User’s Guide 65
Log Segmentation (Applicable only if log output is configured to • false—No segmentation

be sent to a log file.) is allowed.
Specifies the mode of measurement and • <number> KB|<number>—
maximum size for a log file segment. If the The maximum segment
current log segment exceeds the size set by this size, in kilobytes. The
option, the file is closed and a new log file is minimum segment size is
created. 100 KB.
• <number> MB—The
maximum segment size,
in megabytes.
• <number> hr—The
number of hours for the
segment to stay open. The
minimum time period is 1
hour. (This value is not
applicable for Reporting
Server.)
Time Generation for The system by which an application calculates • <number> local—The
Log Messages. the log record time when a log file is generated. time of log record
The time is converted from the time in seconds generation is expressed as
since the Epoch (00:00:00 UTC, January 1, a local time, based on the
1970). time zone and any
seasonal adjustments.
Time zone information of
the application's host
computer is used.
• <number> utc—The time
of log record generation is
expressed as Coordinated
Universal Time (UTC).

Time Format for Log The format in which the log file presents the • time—The time string is
Messages time at which time the application generated the the HH:MM:SS.sss format
log record. (hours, minutes, seconds,
milliseconds).
• locale—The time string
is formatted according to
the locale of the system.
• ISO8601—The date in the
time string is the ISO
8601 format. Fractional
seconds are given in
milliseconds.
Example:
2001-07-24T04:58:10.123
Verbose Level Specifies the minimum level of log events that • all—All log events.
will be generated. • debug—Log events of all
In descending order of priority, the log event levels (same as all).
levels are: • trace—Log events of the
• Standard Standard, Interaction, and
• Interaction Trace levels.
• Trace • interaction—Log events
• Debug of the Standard and
Interaction levels.
• standard—Log events of
the Standard level only.
• none—No output will be
generated.
Table 9 provides the default values for those options in the log configuration
section that are commonly customized.
Table 9: Default Values for Selected log Options
Option Default Value

Name
RM MCP CCP FM RS SSG CTIC PSTN
all ../logs/ ../logs/ ../logs/ ../logs/ Empty ../logs/ ../logs/ ../logs/

ResourceMgr MCP ccp fm SSG CTIConne PSTNConnec
ctor tor
debug ../logs/ ../logs/ ../logs/ ../logs/ /logs/ ../logs/ ../logs/ ../logs/

ResourceMgr MCP ccp fm rs.log SSG CTIConne PSTNConnec
ctor tor
User’s Guide 67
Chapter 3: Configuring Common Features Configuring SNMP
Table 9: Default Values for Selected log Options (Continued)
Option Default Value

Name
RM MCP CCP FM RS SSG CTIC PSTN
expire 20 (files) 10 (files) 10 (files) 10 false 7 days 20 7 days

(files) (files)
interaction ../logs/ ../logs/ ../logs/ ../logs/ Empty ../logs/ ../logs/ ../logs/

ctor tor
message_ short short short short full short short short

format
segment 10000 KB 10000 KB 10000 KB 10000 KB 10 MB 10000 KB 10000 KB 10000 KB
standard ../logs ../logs/ ../logs/ ../logs/ stdout ../logs/ ../logs/ ../logs/

/ResourceMg MCP_standar ccp_standar fm SSG CTIConne PSTNConnec
r d d ctor tor
time- time ISO8601 ISO8601 time time ISO8601 time ISO8601

format
trace ../logs/ ../logs/ ../logs/ ../logs/ Empty ../logs/ ../logs/ ../logs/

ctor tor
verbose standard interaction interaction standard trace standard standard standard
Configuring SNMP
Table 10 describes the option for snmp task timeout.
Table 10: Selected Configuration Options—snmp Section
SNMP Task Timeout Specifies the maximum time interval, in Any integer greater than zero.
milliseconds, that SNMP waits for a new Default Value: 100
task.
Configuring Client-Side Connections

GVP components provide the ability to configure the server-side ports in a
UDP or a TCP connection. This includes situations where the GVP
components act as the server (for example, Call Control Platform’s SIP service

Chapter 3: Configuring Common Features Configuring Client-Side Connections
port), so that modification of the corresponding configuration results in a

different listening port. It also includes situations where the GVP components
acts as the client (for example, Call Control Platform’s connection to Reporting
Server), so that modification of the port option results in an attempt to connect
to a different remote port.
Table 11 describes the connections for each GVP component.
Table 11: Client Connections
Component Connection Purpose
Call Control Configuration Server —TCP dynamically Allows the CCP to receive
Platform (CCP) configured by the OS configuration data and updates from
the Configuration Server.
Message Server—TCP dynamically Sends logs to the Message Server if

configured by the OS sink logging is turned on.
Local Control Agent (LCA)—TCP Allows CCP to send status

dynamically configured by the OS information to the Solution Control
Server (SCS).
SIP—UDP Allows the CCP to provide SIP

option Service
sip.transport.x
Default value: 5068
SIP—UDP/TCP dynamically configured
by the OS
Reporting Server—TCP dynamically Allows the CCP to send sink, OR

configured by the OS and CDR information to the
Reporting Server.
Call Control HTTP—TCP dynamically configured by Allows the CCP to fetch pages and
Platform (CCP) the OS HTTP messaging.
(continued)
SNMP—TCP dynamically configured by Allows the CCP to connect to SNMP
the OS Master Agent
User’s Guide 69
Table 11: Client Connections (Continued)
Media Control Configuration Server—TCP dynamically Allows the MCP to receive

Platform (MCP) configured by the OS configuration data and updates from

Local Control Agent (LCA)—TCP Allows the MCP to send status

Server (SCS).
SIP—UDP Allows the MCP to provide SIP

option Service.
sip.transport.x
Default value: 5070
SIP—TCP dynamically configured by the
OS
Reporting Server—TCP dynamically Allows the MCP to send logging,

configured by the OS OR, and CDR information to the
Reporting Server.
HTTP—TCP dynamically configured by Allows the MCP to fetch pages and

the OS HTTP messaging.
PageCollector—TCP dynamically Allows the legacy interpreter (GVPi)

configured by the OS with ports ranging to fetch pages from the Web Server.
from 1024 to 65535.
SNMP—TCP dynamically configured by Allows the MCP to connect to

the OS SNMP Master Agent
MRCPv2—TCP/UDP Allows local non-security SIP

option: messages.
mrcpv2client.sip.localport
Default value: 7080

Media Control RTP—UDP Allows the MCP to receive media

Platform (MCP) options: data.
(continued)
• mcp.rtp.portlow
• mcp.rtp.porthigh
Default value: 10000-65535
RTP for ASR/TTS UDP Allows the MCP to perform

options: recognition and play synthesized
text.
• mtinernal.rtp_min_port
• mtinternal.rtp_max_port
Default value: 20000-30000
Reporting Server Configuration Server—TCP dynamically Allows the RS to receive

(RS) configured by the OS configuration data and updates from

Local Control Agent (LCA)—TCP Allows the RS to send status

Server (SCS), and to receive High
Availability instructions.
DBMS (Oracle or MSSQL) Allows the RS to connect to an

Oracle or MSSQL database for data
storage. There can be up to seven
connections.
Resource Manager Configuration Server—TCP dynamically Allows the RM to receive

(RM) configured by the OS configuration data and updates from

Local Control Agent (LCA)—TCP Allows the RM to send status

Server (SCS).
SIP—TCP/UDP dynamically configured Allows the RM to provide SIP

by the OS Service.
User’s Guide 71
Resource Manager Reporting Server—TCP dynamically Allows the RM to send logging, OR,
(RM) configured by the OS and CDR information to the
Reporting Server.
Cluster—TCP dynamically configured by Allows RM to monitor the primary

the OS and backup connections for High
Availability configuration.
Supplementary Configuration Server—TCP dynamically Allows the SSG Connector to receive

Services Gateway configured by the OS configuration data and updates from
(SSG) the Configuration Server.

Local Control Agent (LCA)—TCP Allows the SSG to send status

Server (SCS).
HTTP Client port—TCP Allows the SSG to post the call

option results to the Notification URL that
is specified in the call request sent by
[fm]portrange the Trigger Application.
Default value: Empty
T-Lib Port—TCP dynamically configured Allows the SSG to send and receive
by the OS T-lib messages from SIP Server.

CTI Connector Configuration Server—TCP dynamically Allows the CTI Connector to receive
(CTIC) configured by the OS configuration data and updates from
Local Control Agent (LCA)—TCP Allows the CTI Connector to send

dynamically configured by the OS status information to the Solution
Control Server (SCS).
SIP—UDP Allows the CTI Connector to provide

option SIP Service.
sip.transport.x
Default value: 5080
SIP—TCP dynamically configured by the
OS
IVR Server—TCP dynamically configured Allows the CTI Connector to receive

by the OS with ports in a range is messages from IVR Server.
determined by the configuration parameter
IVRSCClientPortRange in the
corresponding IVR Server section.
PSTN Connector Configuration Server—TCP dynamically Allows the PSTN Connector to

(PSTNC) configured by the OS receive configuration data and
updates from the Configuration
Server.

Local Control Agent (LCA)—TCP Enables the PSTN Connector to send

dynamically configured by the OS status information to the Solution
Control Server (SCS).
SIP—UDP Enables the PSTN Connector to

option: provide SIP Service
• GatewayManager.LocalSIPPort
Default value: 5060
User’s Guide 73
Chapter 3: Configuring Common Features Customizing SIP Responses
MRCP Proxy Configuration Server—TCP dynamically Enables the MRCPP to receive

(MRCPP) configured by the OS configuration data and updates from

Local Control Agent (LCA)—TCP Enables the MRCPP to send status

Server (SCS).
Reporting Server—TCP dynamically Enables the MRCPP to send OR log

configured by the OS. information to the Reporting Server.
ASR and TTS speech engines—TCP Enables MRCPP to send RTSP

dynamically configured by the OS. requests and receive responses.
Policy Server (PS) Configuration Server—TCP dynamically Enables the PS to receive

configured by the OS configuration data and updates from
Message Server—TCP dynamically Enables PS to send logs to the

configured by the OS Message Server if sink logging is
turned on.
Local Control Agent (LCA)—TCP Enables PS to send status

Server (SCS).
HTTP—TCP dynamically configured by Enables PS to receive incoming

the OS policy queries from Genesys
Default value: 8090 Administrator.
Customizing SIP Responses

This section lists the configuration options that enable you to customize the
SIP responses and alarms that the Resource Manager, Media Control Platform,
Call Control Platform, and CTI Connector signal for certain events and
conditions.
For more information about the SIP response codes that are generated, and
how the following configuration options relate to them, see Table 100 on
page 466. For more information about all the configuration options, see the
Genesys Voice Platform 8.5 Configuration Options Reference.

Chapter 3: Configuring Common Features Customizing SIP Responses
To customize GVP behavior in response to error conditions and other events,

consider the following options.
• Resource Manager:

rm.conference-sip-error-respcode

rm.options_response_contenttype

rm.options_response_msg_body

rm.resource-no-match-respcode

rm.resource-unavailable-respcode

rm.suspend-mode-respcode

<gateway resource group>.noresource-response-code
• IVR Profile:

gvp.policy.announcement-call-forbidden-response-code

gvp.policy.announcement-call-forbidden-set-alarm

gvp.policy.announcement-usage-limit-exceeded-set-alarm
 gvp.policy.ccxml-usage-limit-exceeded-respcode

gvp.policy.ccxml-usage-limit-exceeded-set-alarm

gvp.policy.conference-forbidden-respcode
 gvp.policy.conference-forbidden-set-alarm

gvp.policy.conference-usage-limit-exceeded-respcode

gvp.policy.conference-usage-limit-exceeded-set-alarm
 gvp.policy.dialing-rule-forbidden-respcode

gvp.policy.dialing-rule-forbidden-set-alarm

gvp.policy.external-sip-forbidden-respcode
 gvp.policy.external-sip-forbidden-set-alarm

gvp.policy.external-sip-usage-limit-exceeded-set-alarm

gvp.policy.inbound-usage-limit-exceeded-set-alarm

gvp.policy.inbound-call-forbidden-response-code

gvp.policy.msml-call-forbidden-response-code

gvp.policy.msml-call-forbidden-set-alarm
 gvp.policy.msml-usage-limit-exceeded-set-alarm

gvp.policy.outbound-call-forbidden-respcode

gvp.policy.outbound-call-forbidden-set-alarm
 gvp.policy.outbound-usage-limit-exceeded-set-alarm
 gvp.policy.outbound-call-forbidden-response-code
 gvp.policy.transfer-forbidden-respcode

gvp.policy.transfer-forbidden-set-alarm

gvp.policy.usage-limit-exceeded-respcode
 gvp.policy.usage-limit-exceeded-set-alarm
 gvp.policy.voicexml-dialog-forbidden-respcode

gvp.policy.voicexml-dialog-forbidden-set-alarm

gvp.policy.voicexml-usage-limit-exceeded-respcode
 gvp.policy.voicexml-usage-limit-exceeded-set-alarm
User’s Guide 75
Chapter 3: Configuring Common Features Configuring Session Timers and Timeouts
• Media Control Platform:


sip.sendalert

sip.copyunknownheaders
• Call Control Platform:

ccpccxml.defaultrejectcode

ccpccxml.sip.send_progressing

session.copy_unknown_headers

sip.copyunknownheaders

sip.OPTIONS.header.Accept

sip.OPTIONS.header.Accept-Encoding

sip.OPTIONS.header.Accept-Language

sip.OPTIONS.header.Allow
• CTI Connector:
 sip.copyunknownheaders
To further customize the SIP response codes for specific situations, use the
<hints> attribute of the <redirect> and <reject> tags—the responseCode
property of the hints object specifies the response code to be used.
Configuring Session Timers and Timeouts

This section describes two kinds of timeouts that determine the length of a
session and affect responses to service requests:
• The session inactivity timers, expiry timers, and timeouts that the Resource
Manager uses to manage sessions (see “Resource Manager Session
Timers”).
• Additional timeouts that are set at the resource level, or specified in SIP or
HTTP requests (see “Additional Timeouts” on page 79).
Resource Manager Session Timers

Table 12 summarizes the configuration options that determine the session
timers that the Resource Manager uses to manage sessions, in the order in
which the Resource Manager applies them. Configure these options on the IVR
Profile, Tenant, or Application (Media Control Platform, Call Control
Platform, Resource Manager) objects, as applicable for your deployment.
The Resource Manager adds a Session-Expires header to initial INVITE
requests if one is not present, and if the request does not contain the timer
option in the Supported header. The value of the Session-Expires header is the
configured value of the applicable session inactivity timer.

Table 12: Session Timer Configuration Options
Environment Tenant/ IVR Profile
gvp.general. The timeout interval, in seconds, for the SIP Any positive integer.
sip.sessiontimer session that is executed for the IVR Profile. If Default value: Empty
the Resource Manager receives no SIP
messages associated with this call leg within the
timeout interval, it considers the call leg to have
ended.
For the call leg associated with the IVR Profile
for this tenant, the value that you configure for
this sip.sessiontimer option overrides session
expiry timeouts that are set at the level of the
resource and the Resource Manager.
PSTN Connector
GatewayManager Section
Enable Session Timer Specifies whether the session timers is enabled • True
for a call session. • False
Default value: True
Session Timer Interval The time interval, in seconds, for which a call An integer in the range of 90
(secs) session must be refreshed before it expires. to 86400.
Default value: 1800
Session Minimum The minimum time interval, in seconds, for An integer in the range of 90
Timer Interval (secs) which a call session must be refreshed. to 1800.
Default value: 90
Session Timer Refresh Specifies which user agent is to initiate the • 0—Local
refresh of a call session. • 1—Remote
Default value: 0
User’s Guide 77
Table 12: Session Timer Configuration Options (Continued)
Resource Manager
proxy Section
Min SE The minimum value of the Session-Expires Any unsigned integer.

header, in seconds, that the Resource Manager Default value: 90
will accept.
If an incoming SIP request contains a
Session-Expires header with a value that is less
than sip.min_se, the Resource Manager rejects
the INVITE request with a 422 (Session
interval too small) response.
Changes take effect: After restart.
Session Expires The default timeout interval, in seconds, for An integer in the range of
Media Control Platform or Call Control 90–3600.
Platform sessions. If no re-INVITES are sent or Default value: 1800
received within the timeout period, the session
expires.
If a different timeout has been set for a
particular VoiceXML or CCXML application, it
overrides the value of this sip.sessionexpires
option.
Resource Manager
proxy Section
Min SE The minimum value of the Session-Expires Any unsigned integer.

header, in seconds, that the Resource Manager Default value: 90
will accept.
If an incoming SIP request contains a
Session-Expires header with a value that is less
than sip.min_se, the Resource Manager rejects
the INVITE request with a 422 (Session
interval too small) response.
Session Timeout The timeout interval, in seconds, for each SIP Any unsigned integer.
session (call leg) that the Resource Manager Default value: 1800
handles.
If a different timeout has been set for a
particular resource or XML application, it
overrides the Resource Manager session expiry
timeout for the applicable session.

Table 12: Session Timer Configuration Options (Continued)
Resource Manager
registrar Section
Registrar Max Expiry The maximum expiry time, in seconds, of this An integer in the range of
Time registrar. If the client requests an expiry time 60–7200.
greater than this value, this sip.registrar. Default value: 60
maxexpirytime value is returned.
Registrar Min Expiry The minimum expiry time, in seconds, of this An integer in the range of
Time registrar. If the client requests an expiry time 60–7200.
smaller than this value, the request is rejected, Default value: 60
with this value in the Min-Expires header.
Additional Timeouts
The following timeouts and timers are also important for GVP behavior:
• Resource Manager, Media Control Platform, and Call Control Platform:

dproxy.sip.timer.ci_proceeding—The timeout for the client INVITE.
The timer starts after a 1xx response is received for a client INVITE. If
a final response is not received before the timer expires, the SIP
session and dialog is destroyed without further notice to the UAS. This
timer should be greater than the connect timeout of the outbound call
(depending on how the outbound call is initiated, the connect timeout
can be specified in the transfer tag, or in the remdial command).
Otherwise, the dsip.timer.ci_proceeding timer will trigger before the
connect timeout occurs, which overrides the connect timeout. (The
default is 120000 ms, or 120 seconds.)
• Resource Manager:

sip-timer_B—The timeout Resource Manager uses when selecting a
resource for messaging. If no 1xx provisional response is received with
this timeout, Resource Manger considers this resource unreachable,
and attempts to select another resource.
• Media Control Platform:

mpc.rtp.timeout—The timeout for the RTP/RTCP stream. (The default
is 60000 ms.)
 msml.cpd.beeptimeout—The timeout for the default beep. (The default
is 30 seconds.)

msml.cpd.postconnecttimeout—The timeout for the CPD postconnect.
(The default is 30 seconds.)

msml.cpd.preconnecttimeout—The timeout for CPD preconnect. (The
default is 30 seconds.)
User’s Guide 79

sessmgr.acceptcalltimeout—The timeout for the platform to accept
inbound calls, after alerting is issued. (The default is 30000 ms.)

sessmgr.maxincalltime—The maximum call time for inbound calls.
(The default is 0—disabled.)

sip.hfdisctimer—The timeout to terminate a SIP hookflash transfer.
(The default is 5000 ms.)

sip.referxferwaitbye - After a REFER transfer, timeout to wait for
BYE message from the remote end before sending BYE to disconnect
the call. If it is zero, it will send BYE right after a NOTIFY/200 is
received. If it is non-zero, it will wait for the configured timeout (in
milliseconds) before sending the BYE. Values are specified in
millisecond. (The default is 0 - send BYE immediately).

stack.connection.timeout—The connection timeout for the MRCP
Client stack to establish a TCP connection to the MRCP server. (The
default is 10000 ms.)

vrm.client.timeout—The connection timeout for the MRCP Client to
receive a response from the MRCP server. (The default is 10000 ms.)

vxmli.default.connecttimeout—The default value of the
connecttimeout attribute for bridge or consultation transfers, if not
provided. (The default is 30000 ms.)
 vxmli.initial_request_fetchtimeout—The fetch timeout for the
initial VoiceXML page. (The default is 30000 ms.)

vxmli.max_script_time—The maximum time allowed for each script
or ECMAScript expression to be executed. (The default is 2000 ms.)
• Call Control Platform:

ccxmli.fetch.timeout—The default timeout for the fetch of the initial
page to be completed. (The default is 30 seconds.)
• Supplementary Services Gateway:

ssg.ReqAccOnResourceDNErrTimeoutSecs—The timeout for the
Supplementary Services Gateway to reject new requests from tenant
applications if the Supplementary Services Gateway fails to register a
Resource DN with SIP Server. (The default is 900 seconds).
 ssg.ReqAccOnSIPSConnErrTimeoutSecs—The default timeout for the
Supplementary Services Gateway to reject new requests from tenant
applications if the Supplementary Services Gateway fails to connect to
SIP Server. (The default is 900 seconds).
• CTI Connector:
 ctic.connectcalltimeout—The default timeout that CTI Connector
waits for an outbound call to connect. (The default is 6000 ms)

IServer_Sample.keepaliveresptimeout—The timeout that
CTI Connector waits for a response from IVR Server. (The default is 3
seconds.)

IVRSC.scriptidfetchtimeout—The time to wait for a response to fetch
the script id from URS. (The default is 5000 ms.)

• PSTN Connector:

GatewayManager.XferConnectTimeoutMSec—The time to wait for the
transfer result after issuing a blind or consult transfer request. (The
default is 60000 ms)
• MRCP Proxy:

timeout.back_in_service—The time to wait for the server to be put
back into service after it encounters errors such as timeout or TCP
connection error.

timeout.barge_in_occurred—The timeout for a barge-in to occur.

timeout.clean_loop—The time interval in which idle sessions are
cleaned, as determined by timeout.max_idle configuration option.

timeout.close_session—The time it takes for a Close-Session request
to expire.

timeout.control—The time it takes for a CONTROL message to expire.

timeout.define_grammar—The time it takes for a DEFINE-GRAMMAR
message to expire.
 timeout.get_params—The time it takes for a GET-PARAMS message to
expire.

timeout.get_result—The time it takes for a GET-RESULT message to
expire.

timeout.get_server_info—The time to wait to get a response to a
Get-Server-Info request (ping) before timing out.
 timeout.lca_calibrate—The time to wait (at connect or re-connect)
before an application mode query is sent to the LCA. After this timeout
expires the query can be sent.

timeout.max_idle—The maximum amount of time a session can be
idle before it is terminated.

timeout.open_session—The time it takes for an Open-Session to
expire.
 timeout.pause—The time it takes for a PAUSE message to expire.

timeout.recog_start_timers—The time it takes for a
RECOGNITION-START-TIMERS message to expire.
 timeout.recognize—The time it takes for a RECOGNIZE message to
expire.

timeout.reconnect_interval—The time to wait before a reconnect
attempt is made, if the TCP connection is not yet established with the
MRCP server.

timeout.resume—The time it takes for a RESUME message to expire.
 timeout.set_params—The time it takes for a SET-PARAMS message to
expire.
 timeout.speak—The time it takes for a SPEAK message to expire.

timeout.stop—The time it takes for a STOP message to expire.
User’s Guide 81

connection.timeout—The time it takes before a connection times out
when the SRM MRCPv1 and MRCPv2 stack is attempting to establish
a TCP connection to the server.

timeout—The maximum amount of time that SNMP can wait for a
new task.

Chapter
4 Configuring the Resource

Manager
This chapter describes the Resource Manager configuration requirements for
your Genesys Voice Platform (GVP) deployment. It contains the following
sections:

Task Summary: Configuring the Resource Manager, page 83
 Important Resource Manager Configuration Options, page 84

Configuring Logical Resource Groups, page 89
Task Summary: Configuring the Resource

Manager
Task Summary: Configuring the Resource Manager summarizes the
configuration steps and options to implement Resource Manager functionality
in your GVP deployment.
Task Summary: Configuring the Resource Manager
Set up the Resource Manager to See “Configuring SIP Communication and Routing” on page 38.
function as SIP Proxy, SIP Registrar, To secure SIP communications between the Resource Manager
SIP Notifier, and resource monitor and the other GVP components, ensure that you specify a
and manager. transport for the Transport Layer Security (TLS) protocol and a
secure routeset for outbound calls.
Provision GVP resources. See “Configuring Logical Resource Groups” on page 89.
User’s Guide 83
Chapter 4: Configuring the Resource Manager Important Resource Manager Configuration Options
Task Summary: Configuring the Resource Manager (Continued)
Configure the IP DiffServ (ToS). Set the SIP packet’s ToS using
[sip]transport.[n].tos
See “Configuring SIP Communication and Routing” on
page 38”.
Provision IVR Profiles. See Chapter 6 on page 103.
Configure conferencing. See “Enabling Conference Services” on page 58.
Configure reporting. See “Configuring Reporting” on page 59.
Customize logging. See “Configuring Logging” on page 62.
Customize session management See “Configuring Session Timers and Timeouts” on page 76.
behavior and performance. See also the parameters in the proxy section that specify
parameters such as the number of threads and connections.
Customize client-side See “Configuring Client-Side Connections” on page 68.

communication ports.
Customize Resource Manager See “Configuring SNMP” on page 68 and Table 100 on
messaging. page 466.
Important Resource Manager

Configuration Options
This section describes the key configuration options that you either must or
may want to customize.
Environment > Applications > <Resource Manager> > Options tab. For the
detailed steps to configure option settings, see Procedure: Viewing or
modifying GVP configuration parameters, on page 30.
Except where otherwise indicated, all changes to Resource Manager
parameters take effect after you restart the Resource Manager.
The Resource Manager configuration options are in the following
configuration sections:
• cluster—Parameters that determine the High Availability behavior.
• gvp—Parameters that monitor the health and status of the network
interfaces and bonding drivers.
• ems—Parameters that the determine Reporting behavior. (See Table 6 on
page 60.)

• log—Parameters the determine the behavior for Management Framework

logging. (See “Configuring Logging” on page 62.)
• monitor—Parameters that support the Resource Manager in its role as
manager of GVP resources.
• proxy—Parameters that determine the behavior of the Resource Manager
in its role as SIP Proxy and session manager.
• registrar—Parameters that determine the behavior of the Resource
Manager in its role as SIP Registrar.
• rm—Parameters that determine the behavior of the Resource Manager in its
role as manager of GVP services.
• snmp—Parameters that determine the behavior of SNMP. (See
“Configuring SNMP” on page 68.)
• subscription—Parameters that control the SUBSCRIBE/NOTIFY behavior.
Table 13 provides information about important Resource Manager parameters
that are not described in Chapter 3 on page 37. Table 13 provides parameter
descriptions, and also the default parameter values that are preconfigured in the
Resource Manager Application object.
For information about all the available configuration options for the Resource
Manager, see the Genesys Voice Platform 8.5 Configuration Options
Reference.
Table 13: Selected Resource Manager Configuration Options
cluster Section
Election Timer Specifies the interval, in milliseconds, in which An integer in the range of
this node waits for a response from its remote 1000–10000
members. If there is no response within this time, Default value: 3000
the local Resource Manger becomes the active
node.
FailOver Batch Specifies the path to the fail over script. Install dir /bin/nbl.bat
Script Default value:
$InstallationRoot$/bin/nb
l.bat
Heartbeat Interval Specifies the interval, in milliseconds, for which An integer in the range of
the members of the cluster check each other’s 2000–60000
health status. Default value: 2000
User’s Guide 85
Table 13: Selected Resource Manager Configuration Options (Continued)
Cluster Standby This parameter has effect only if cluster.virtual-ip • True

Mode parameter is non-empty. If true, this parameter • False
mandates that the RM cluster run in hot-standby
Default value: False
redundancy mode where call data is replicated to
other RM node for SIP dialog redundancy.
Otherwise, this parameter mandates that the RM
cluster be run in warm-standby redundancy mode
where SIP dialog redundancy is not supported, but
the new requests will be handled by the other
healthy RM node
Members List of ID's of the members in the cluster Default value: 1 2

(unsigned integers 1 to 32 delimited by space). For
NLB, the ID's correspond to the unique host
identifier (priority) number specified for each of
the NLB cluster machines.
Members 1 Describes the IP and TCP port on which the <first member IP
member ID 1 can be reached. The format is address>:<first member
IP:Port where IP and Port specifies the IP and port cluster communication
where this RM node can be reached for cluster port>
communication Default value: Empty
Members 2 Describes the IP and TCP port on which the <second member IP
member ID 2 can be reached. The format is address>:<second member
IP:Port where IP and Port specifies the IP and port cluster communication
where this RM node can be reached for cluster port>
communication Default value: Empty
My Member ID Indicates this cluster manager instance's member An integer.

ID (select one among the ID's listed in Default value: Empty
cluster.members
Cluster Virtual IP If non-empty, this parameter indicates that this An integer.
Address RM node is part of a RM cluster and the value Default value: Empty
specifies the virtual IP address of the RM cluster
this RM node is part of. In stand-alone mode, this
parameter must be left empty.
Cluster HA Mode Specifies which cluster the Resource Manager • none—stand alone RM
instances are configured. • active-standby—RM in
cluster
• active-active—external
load balancer
Default value: none

rm Section
FIPS Mode Enabled Enables FIPS mode in RM. True

False
Changes take effect:
start/restart
SIP Header for DNIS The header from which the Resource Manager • To
will retrieve the DNIS, to identify which IVR • Request-Uri
Profile to use.
• History-Info
Ensure that the value you specify is consistent
Default value: History-Info
with Media Gateway behavior, so that the INVITE
messages that SIP Server forwards to the Resource
Manager have the DNIS information in the
expected header.
• If the value of this parameter is History-Info
but there is no History-Info header in the SIP
INVITE, the Resource Manager picks up the
DNIS from the To header.
• If the value of the specified header in the SIP
INVITE is not a valid DNIS, the Resource
Manager cannot map the SIP request to an IVR
Profile, and it defaults to the next behavior to
select the IVR Profile.
• If the P-Called-Party-ID header is present, SIP
Header for DNIS is ignored.
Changes take effect: Immediately.
Default Resource Specifies the port capacity that is assigned to each Any unsigned integer.
Port Capacity physical resource. Default value: 500
Note: This parameter can be overridden by the
MaxPorts configuration during post installation
activities. For more information, see the Genesys
Voice Platform 8.5 Deployment Guide.
monitor Section
SIP Resource The interval, in milliseconds, at which the Any unsigned integer.
OPTIONS Interval Resource Manager sends OPTIONS messages to a Default value: 5000
healthy resource, to determine whether the
resource is alive.
User’s Guide 87
SIP Unavailable The interval, in milliseconds, at which the Any unsigned integer.
Resource OPTIONS Resource Manager sends OPTIONS messages to a Default value: 5000
Interval dead resource to determine whether the resource is
alive.
SIP Release Specifies how to handle new incoming calls that • True
Conference Resource are joining the conference if the conference • False
on Failure resource goes offline.
Default value: True
If set to True, all conference sessions are released,
and the new incoming calls are routed to the next
available resource.
If set to False, all conference sessions are
released, and the new incoming calls will receive
an error.
proxy Section
Preferred IP Version Specifies the preferred IP version when multiple • ipv4

Used in SIP Proxy IP addresses with different IP versions are • ipv6
resolved from a destination address. The first
Default value: ipv4
address from the list with the preferred IP version
is used. However, if there the sip.transport.x
configuration option is not defined with the
preferred version, other version are used.

Chapter 4: Configuring the Resource Manager Configuring Logical Resource Groups
IP Type of Service Specifies the IP differentiated services field (ToS) Range: 0-255
for SIP Transport to set in all outgoing SIP packets over the SIP Examples:
transport.
• 0—Disabled
Notes:
• 16—IPTOS LOWDELAY
• For Windows Server 2003, the ToS must be (0x10)
enabled in the registry. See
• 32—IPTOS PREC
http://support.microsoft.com/kb/248611
PRIORITY (0x20)
• For Windows Server 2008/2012, the ToS
configuration is not supported. It must be • 64—IPTOS PREC
CRITICAL (0x40)
configured at the OS level. You can define per
executable and per port, and what type of • 184—DiffServ EF
DiffServ bits to set on the outgoing packets (Expedited Forward
using the QoS policy defined in the following 0xBB)
article. Default value: 0
http://technet.microsoft.com/en-us/library/
cc771283.aspx
• For all Operating Systems, when the SIP/RTP
packets are sent across different subnets, the
router may reset the DiffServ bits in the IP
header even though it was set by MCP.
Record Route Host Specifies the host to be used for the Record-Route <hostname or IP address>
when an INVITE is forwarded when Resource
Manager is in stand-alone mode. The value
specified can either be configured as an IP
address, or FQDN. If the value is empty, the IP
address of the outgoing transport is used.
Configuring Logical Resource Groups

For each type of service that GVP provides (VoiceXML, CCXML,
Conference, MSML, Announcement, Recording Server), you must create and
configure a logical resource group (LRG) that the Resource Manager will use
as its resource pool. You must create a resource group for each type of service,
even if there is only one resource available to provide that service (in other
words, if the group has a single member).
Use the Genesys Administrator Resource Group Wizard (Provisioning >
Voice Platform > Resource Groups) to create, modify, or view settings for the
resource group and to specify the resources that belong to each group. For
detailed information about using the Resource Group Wizard, see Procedure:
Configuring logical resource groups.
User’s Guide 89
You identify the actual resource hosts and applications in the connections that
you configure for the Resource Manager. In addition, you must create a remote
access point for the recordingserver resource that points to the Resource
Manager. See Procedure: Creating the resource access point for Recording
Server, on page 95.
For more information about logical resource groups, see the chapter about
post-installation activities in the Genesys Voice Platform 8.5 Deployment
Guide.
The following procedure provides the detailed steps to use the Resource Group
Wizard.
Procedure:
Configuring logical resource groups
Purpose: To create or modify the property information that is shared by a

logical group of GVP resources managed by a particular Resource Manager.
Unless otherwise indicated, changes to logical resource group configurations
take effect immediately.
Prerequisites
• The GVP Application objects have been installed, as described in the
Genesys Voice Platform 8.5 Deployment Guide.

Start of procedure
1. In Genesys Administrator, go to the Provisioning > Voice Platform >
Resource Groups panel (see Figure 5).
Figure 5: The Resource Groups Panel
2. Do one of the following to invoke the wizard:

• To create a new group, click New.
• To modify the configuration parameters for an existing group, select
the group name and click Edit.
• To delete an existing group, click Delete.
For more information on how to use the Resource Groups Wizard, see the
End of procedure
Next Steps
• If required, configure the noresource-response-code option in the <gateway
resource group> section of the Resource Manager Application object.
The default behavior for the Resource Manager with regard to gateway
resources is not to retry failed requests. To configure the Resource
Manager to automatically retry other resources in a gateway resource
group, specify the required SIP failure response codes in the
noresource-response-code option. This option does not appear in the
Resource Management Wizard, you configure it on the Provisioning >
Environment > Applications > <Resource Manager> > Options tab.
• Manually set the capabilities and the preferences for the Logical Resource
Group (see page 94).
Table 14 provides information about the Resource Manager parameters for
logical groups.
User’s Guide 91
Table 14: Logical Group Section Configuration Options
Group Type Specifies the type of logical resource group. • Media Control Platform
• Call Control Platform
• Gateway
• CTI Connector
• Recording Server
CTI Usage Specifies whether the Resource Manager will • Always off—Resource
use CTI Connector for a Gateway logical group Manager does not use
resource. CTI Connector, and
proceeds with the call
using DNIS-IVR Profile
mapping.
• Always On—Resource
Manager will not map the
call.
• Based on DID lookup—
Resource Manager
performs the IVR Profile
lookup for the call and
forwards it to
CTI Connector with the
CTI service parameters
configured in the IVR
Profile.
Port Capacity Specifies the port capacity of all resources in • Default value: 500
this logical resource group combined. Individual
resource port capacity will be ignored.
Note: The port capacity option is available for
the Recording Server resource group only when
parallel-forking is used as the load balancing
scheme.
Load Balancing The distribution algorithm that the Resource • Round-robin

Scheme Manager will use to select a resource within this • Least used
logical resource group.
• Least percent
Note: The parallel forking option value is
• parallel forking
available for the Recording Server LRG only.
Default value: Round-robin

Table 14: Logical Group Section Configuration Options (Continued)
Max Conference Size The maximum number of participants allowed An integer in the range of 0–
(For MCP) in a conference. 32.
Max Conference Count The maximum number of conferences allowed Any integer.
(For MCP) for a resource.
Monitoring Method The method that the Resource Manager will use • SIP OPTIONS—Resource
to determine whether the physical resources that Manager will use SIP
belongs to the logical resource group are alive OPTIONS messages.
and healthy. • None—Resource Manager
will not monitor resource
health. It assumes that
resources in this group are
always alive.
Default value: SIP OPTIONS
Geo Location The geographical location of the resource. A character string.

(Optional)
Capability The list of values supported by the resources A character string.

(Manually configured) within the logical resource that corresponds
to the SIP INVITE capability. This is read
from the configuration parameter capability
of a logical resource, and the parameter value
has the following syntax:
[cap_NameA]=[cap_ValueA1],…,[cap_ValueAm
];
[cap_NameB]=[cap_ValueB1],…,[cap_ValueBn]
;…;
[cap_NameM]=[cap_ValueM1],…,[cap_ValueMi]
Note: The set of [cap_NameX] must be unique.
Preference Specifies which logical resource group has Any positive integer.
(Manually configured)
preference. If there are two resource groups
with difference preference numbers, the
logical resource group with lower number is
considered first for all calls.
Address of Record Specifies the list of contacts for the physical [sip|sips]:<address>:
(AOR) resource. This parameter must be configured <port>.
manually in order for Resource Manager to
(Manually
support static routing (see Procedure:
configured)
Configuring Resource Group capabilities,
preferences, and AOR, on page 94)
User’s Guide 93
The following procedure provides the detailed steps to manually configure the
Logical Resource Group capabilities, preferences and AOR.
Procedure:
Configuring Resource Group capabilities, preferences,
and AOR
Purpose: To configure capabilities, preferences, and AOR for the logical

resource group.
Prerequisites
• The GVP Application objects have been installed, as described in the
• The Logical Resource Group has been created.
Start of procedure
1. In Genesys Administrator, go to the Provisioning > Environment >
Business Units/Sites
2. Select the appropriate tenant.
3. Select the MCP Application, and click Edit.
4. Add the Capability, Preference, and AOR parameters to the gvp.lrg
section. For more information on how to add options using Genesys
Administrator, see the Framework 8.5 Genesys Administrator Help.
See Table 14 for descriptions of these parameters.
End of procedure
Note: When a SIP Server HA pair is configured as a gateway resource for

the Resource Manager, the sip-address configuration option in the
T-Server section of both SIP Servers in the pair must be configured to
point to the virtual-ip for the HA pair. This value is used by SIP Server
to build the Via and the Contact headers in SIP messages.

Procedure:
Creating the resource access point for Recording
Server
Purpose: To provide an overview of the steps to configure logical recording

server Application, to provide a presence for the recording server in the
Genesys Configuration Layer.
Summary
You must create a separate Application for each recording server in your
deployment. The Application type is Resource Access Point.
For detailed information about importing Application Templates and
metadata, and creating Applications from the templates, see Appendix A in
the Genesys Voice Platform 8.5 Deployment Guide, which describes the
pre-installation activities
Prerequisites
• The Resource Manager Installation Package (IP) is available.
Start of procedure
1. Create the Recording Server Application object.
a. Import the required Application Template from the Resource Manager
Installation Package (IP). For example,
VP_CallRecordingServer_814.apd.
b. On the Provisioning > Environment > Applications tab, create and
name the new Resource Access Point, based on the Application
Template.
Configure 2. In the gvp.rm section, on the Provisioning > Environment > Applications
Resource Access > <Recording Server> > Options tab, configure the following options:
Points:
• aor=sip[s]:<host|ip>:<port>
• port-capacity=500
• redundancy-type=active
Note: The host and port number in the aor configuration option, is
populated automatically when the LRG group wizard is used to
create the recordingserver resource group.
User’s Guide 95
3. In the provision section, ensure the default value of 1 is retained for the
recording-server configuration option.
4. Save the configuration.
End of procedure
Next Steps
• No further steps are required.

Chapter
5 Configuring Policy Server

The Genesys Voice Platform (GVP) Policy Server component is used by
Genesys Administrator for the validation and resolution of GVP-specific
business rules. It is a stand-alone Java process that connects to Management
Framework through an HTTP interface.
This chapter provides information about configuring Policy Server in the
following sections:
 Task Summary: Configuring Policy Server, page 97

Important Policy Server Configuration Options, page 98
Task Summary: Configuring Policy Server

Task Summary: Configuring Policy Server summarizes the tasks that are
required to implement Policy Server functionality in your GVP deployment.
Task Summary: Configuring Policy Server
Create a connection to Policy Server In Genesys Administrator, open the Configuration Manager
in the Configuration Manager Application and on the Configuration tab, add Policy Server to
Application. the Connections.
Note: Policy Server can be deployed with Genesys Administrator
and Configuration Manager and later releases only.
Customize client side See “Configuring Client-Side Connections” on page 68.

User’s Guide 97
Chapter 5: Configuring Policy Server Important Policy Server Configuration Options
Important Policy Server Configuration

Options
Configure the options on Genesys Administrator on the Provisioning >
Environment > Applications > <Policy Server> > Options tab. For the
The configurable Policy Server parameters are in the following configuration
sections:
• https—Parameters that determine how security is implemented for Policy
Server.
• https_key—Parameters that specifies the optional security key password.
• log —Parameters that determine the logging behavior.
• reporting—Parameters that determines how data from Policy Server is
reported.
Table 15 provides information about important Policy Server parameters that
are not described in Chapter 3 on page 37. Table 15 provides parameter
the Policy Server Application object.
Unless indicated otherwise, all changes take effect immediately.
For a complete list of Policy Server configuration options and their
descriptions, see the Genesys Voice Platform 8.5 Configuration Options
Reference.
Table 15: Selected Policy Server Configuration Options
https Section
SSL Keystore Path Specifies the path to the keystore file, which Any string of characters.
will be used for all the HTTPS connectors. Default value:
${user.home}/.keystore
Changes take effect: start/restart
SSL Keystore Specifies the password for the keystore file. Any string of characters.
Password Default value: Empty

Table 15: Selected Policy Server Configuration Options (Continued)
SSL Keystore Type Specifies the type of keystore, which defines Any string of characters.
the supported file format for the security Default value: JKS
implementation.
SSL Certificate Specifies the name of the SSL algorithm that Any string of characters.
Algorithm will be used for the configured keystore. Default value: SunX509
HTTPS Protocol Specifies the cryptographic protocol to use. Select one of five option
values—SSL, SSLv2, SSLv3,
TLS, or TLSv1
Default value: TLS
Secure Random Specifies the name of the RNG (Random Any string of characters.
Algorithm Number Generator) algorithm. Default value: Empty
For more information about the RNG, see the Changes take effect: start/restart
JDK JavaDoc for class
java.security.SecureRandom
Security Provider Specifies the name of Java security provider. Any string of characters.
For more information about the security Default value: None
provider, see the JDK JavaDoc for class Changes take effect: start/restart
java.security.Provider
Client Authentication Specifies the HTTPS client authentication Select one of three option
Requirement requirements. values—none, required, or
If this option is set to: preferred.
• none—No certificate is requested; Default value: Empty

Client-side authentication is disabled. Changes take effect: start/restart
• required—A certificate is requested and
the server will require a valid, non-empty
certificate response to establish the
connection. (Works for BIO connector type
only.)
• preferred—A certificate is requested, but
the server will still establish the connection
if the certificate response is empty.
User’s Guide 99
HTTPS Connector Specifies the type of Jetty connector that will Select one of two option
Type be used. values—1 or 2.
If this option is set to: Default value: 2
• 1 <NIO>—Non-blocking NIO connector. Changes take effect: start/restart
• 2 <BIO>—Blocking BIO connector.
For more information about these connectors,
see Jetty's JavaDoc for class
org.mortbay.jetty.security.SslSelectChan
nelConnector.
https_key Section
SSL Key Password Specifies the optional key password for the Any string of characters.
HTTPS configuration. Default value: Empty
log Section
Verbose Level Determines whether or not a log output is Select one of several log event
created. If it is, this option specifies the levels.
minimum level of log events that are Default value: standard
generated.
Any one of the following log event levels can
be selected as the value for this option
(starting with the highest priority level):
standard, interaction, trace, debug, all,
or none.
For a description of the log events that are
logged for each level, see Table 8 on page 64.
Output for Level All Specifies the outputs to which an application A string of characters.
sends all log events. Default value: Empty
The log output types must be separated by a
comma when more than one output is
configured.
Output for Level Specifies the outputs to which an application A string of characters.
Standard sends the log events of the Standard level. Default value: stdout

Interaction sends the log events of the Interaction level Default value: Empty
and higher, which means, more than one
output is configured—standard and
interaction levels.
Trace sends the log events of the Trace level and Default value: Empty
higher, which means, more than one output is
configured—standard, interaction, and
trace levels.
Debug sends the log events of the Debug level and Default value: logs/ps.log
higher, which means, more than one output is
configured—standard, interaction, trace,
and debug levels.
Log Segmentation Specifies the segmentation limit for a log file. A string of characters.
Sets the mode of measurement, along with the Default value: 10MB
maximum size.
If the current log segment exceeds the size set
by this option, the file is closed and a new one
is created.
For a complete description of the option values
for log segmentation, see Table 8 on page 64.
Log Expiration Determines whether or not the log files A string of characters.
expires. If they do, this option sets the Default value: false
measurement for determining when they
expire, along with the maximum number of
files (segments) or days before the files are
removed.
for log expiration, see Table 8 on page 64.
Log Messages Format Specifies the format of log record headers that Select one of two option
an application uses when writing logs in the values—short or full.
log file. Using compressed log record headers Default value: full
improves application performance and reduces
the log file's size.
For a complete description of each option
value, see Table 8 on page 64.
User’s Guide 101

Time Format for Log Specifies how to represent the time when an Select one of three option
Messages application generates log records in a log file. values—time, locale, or
For a complete description of each option ISO8601.
value, see Table 8 on page 64. Default value: time
reporting Section
PS Service IP Specifies the interface IP address that will be A string of characters.

Address used to bind the Policy Server service. Default value: Empty
PS Service Hostname Specifies the hostname that will be used to A string of characters.
access Policy Server. Default value: Empty
PS Service Port Specifies the port on which Policy Server will An integer greater than 0.
receive reporting requests. Default value: 8090
PS Service Protocol Specifies the type of communication protocol Select one of two option
that Policy Server will use to service reporting values—http or https.
requests. Default value: http
Basic HTTP Specifies the user name that Policy Server uses A string of characters.
Authentication User to perform basic HTTP authentication. Default value: Empty
Name
Basic HTTP Specifies the password that Policy Server uses A string of characters.
Authentication to perform basic HTTP authentication. Default value: Empty
Password

Chapter
6 Provisioning IVR Profiles

IVR Profiles are the Voice Extensible Markup Language (VoiceXML), Call
Control Extensible Markup Language (CCXML), Conference, and
Announcement applications that control interactions with external customers.
This chapter describes how to provision IVR Profiles for Genesys Voice
Platform (GVP). It contains the following sections:

Provisioning IVR Profiles for GVP, page 103

IVR Profile Configuration Options, page 109
 Operational Parameter Management and Self-Service Applications,
page 123

IVR Profile Configuration for GVPi, page 123

IVR Profile Configuration for Cisco ICM, page 127
 Mapping IVR Profiles to DID Numbers, page 128

DID Group Bulk Operations Wizard, page 131

Data Retention Policy Wizard, page 134
 IVR Profile Configuration for Tenants, page 136
Provisioning IVR Profiles for GVP

The summary procedure in this section provides an overview of the steps to
provision IVR Profiles for GVP.
Procedure:
Provisioning IVR Profiles
Purpose: To set up GVP so that it uses specified VoiceXML or CCXML

applications.
User’s Guide 103

Chapter 6: Provisioning IVR Profiles Provisioning IVR Profiles for GVP
Prerequisites
Start of procedure
1. Use the IVR Profile Wizard in Genesys Administrator to create or edit an
IVR Profile object. For more information on how to use the wizard, see the
2. On the Provisioning > Voice Platform > IVR Profile tab of Genesys
Administrator, launch the IVR Profile Wizard.
3. On the Service Type page:
a. Enter the name of the IVR Profile.
b. Select the type of service that the IVR Profile requires. This sets the
gvp.general.service-type parameter. The possible values are:
• VoiceXML
• CCXML
• Conference
• Announcement
c. Complete the remainder of the wizard as applicable for your
deployment. Depending on the type of service you have selected in
Step b, the wizard presents different subsequent pages.
• For VoiceXML, go to Step 4 on page 104.
• For CCXML, go to Step 5 on page 106.
• For Conference, go to Step 6 on page 106.
• For Announcement, go to Step 7 on page 107.
4. For VoiceXML:
a. On the Service Properties page:
i. Enter the Initial Page URL. This sets the initial-page-url
parameter.
ii. Enter the Alternate Page URL. This sets the alternatevoicexml
parameter.
iii. Enter the Default Properties Page URL. This sets the
default-properties-page parameter.
iv. Select the type of VoiceXML interpreter. This sets the
voicexml.gvp.appmodule parameter. For more information on the
interpreters, see Chapter 7, “Configuring the Media Control
Platform,” on page 141.
v. Enter the Toll Free Number. This sets the toll-free-number
parameter.

vi. Enter the Virtual Reporting Object 1 and Virtual Reporting

Object 2. This sets the VirtualReportingTag1, and
VirtualReportingTag2 parameters.
b. On the Usage Limits page, if required, enter the maximum number of
concurrent sessions. This sets the usage-limits parameter.
c. On the IVR Capabilities page:
i. If you want outbound calls with bridged or consultation transfers,
allow outbound calls. This sets the outbound-call-allowed
parameter.
ii. If you want midcall transfers (blind or consultation), allow
transfers. This sets the transfer-allowed parameter.
iii. If MSML requests are not to be processed, set msml-allowed to
false.
iv. Select how to use the gateway. This sets the use-same-gateway
parameter.
d. On the CTI Parameters page:
i. If CTI Connector is required to interface with IVR Server, select
Require CTI Interaction. This sets the cti-allowed parameter.
ii. If midcall blind transfers (starts in the application) are allowed
(transfer-allowed is set to true) and you want the transfers to be
performed through the IVR Server, select Transfer on CTI. This
sets the cti.TransferOnCTI parameter.
iii. Enter the default agent. This must be a DN that is configured in the
Configuration Server database. This sets the cti.DefaultAgent
parameter.
e. If you allowed outbound calls and/or transfers, on the Dialing Rules
page:
i. Select to accept or reject the rule expression.
ii. Enter the Regular Expression.
iii. Click Add as New Rule.
Repeat Steps i to iii for each Dialing Rule that you require. This sets the
gvp.policy.dialing-rules parameter.
For more information on Dialing Rules, see the
“gvp.policy.dialing-rules Section” on page 119.
f. On the Policies page, add the SQ Notification Threshold. This sets the
error.notification.threshold parameter.
For more information on VoiceXML IVR Profile parameters, see Table 16
on page 109.
User’s Guide 105

5. For CCXML:
i. Enter the Initial Page URL. This sets the initial-page-url
parameter.
ii. Enter the Virtual Reporting Object 1 and Virtual Reporting
b. On the Usage Limits page, if required:
i. Enter the maximum number of concurrent sessions. This sets the
usage-limits parameter.
ii. Enter the usage limits for each service. This sets the
<service>-usage-limit parameter.
iii. Enter the usage limits per session. This sets the
<service>-usage-limit-per-session parameter.
c. On the IVR Capabilities page, if required:
i. Enable conferencing. This sets the conference-allowed parameter.
ii. Enable outbound calling. This sets the outbound-call-allowed
parameter.
iii. Enable transfers. This sets the transfer-allowed parameter.
iv. Enable VoiceXML dialogs. This sets the voicexml-dialog-allowed
parameter.
v. Select how to use the gateway. This sets the use-same-gateway
parameter.
d. If you allowed outbound calls and/or transfers, on the Dialing Rules
page:
i. Select to accept or reject the rule expression.
ii. Enter the Regular Expression.
iii. Click Add as New Rule.
Repeat Steps i to iii for each Dialing Rule required. This sets the
gvp.policy.dialing-rules parameter.
For more information on Dialing Rules, see “gvp.policy.dialing-rules
Section” on page 119.
e. On the Policies page, add the SQ Notification Threshold. This sets the
For more information on CCXML IVR Profile parameters, see Table 16 on
page 109.
6. For Conference:
i. Enter the conference ID. This sets the conference-id parameter.
ii. Enter the maximum conference size, if required. This sets the
application-confmaxsize.

iii. Enter the Virtual Reporting Object 1 and Virtual Reporting

c. On the Policies page, add the SQ Notification Threshold. This sets the
For more information on Conference IVR Profile parameters, see Table 16
on page 109.
7. For Announcement:
a. On the Service Properties page;
i. Enter the VoiceXML page that is used to play announcements. This
sets the announcement-url parameter.
ii. Enter the content-type, if required.
iii. Enter the repeat count, if required.
iv. Enter the delay amount, if required.
v. Enter the duration, if required.
vi. Enter the Virtual Reporting Object 1 and Virtual Reporting
c. On the Policies page, add the SQ Notification Threshold. This sets the
For more information on Announcement IVR Profile parameters, see
8. On the Context Services page, enter a username and password for context
services authentication.
The password is masked when it is entered into the password field.
9. When you have completed the required configuration in the wizard, click
Finish.
The IVR Profile object now displays in the list on the Provisioning >
Voice Platform > IVR Profile tab in Genesys Administrator.
User’s Guide 107

10. Modify the IVR Profile to capture the required configuration parameters
that are not set with the wizard.
The Resource Manager uses the IVR Profile name that you specify to
identify the context of the session. For more information, see “Application
Identifiers” on page 23.
For detailed information about the IVR Profile configuration options, see
“IVR Profile Configuration Options” on page 109.
For detailed information about configuring IVR Profiles for GVPi, see
“IVR Profile Configuration for GVPi” on page 123.
End of procedure

Chapter 6: Provisioning IVR Profiles IVR Profile Configuration Options
IVR Profile Configuration Options

The IVR Profile configuration options determine the type of service the IVR
Profile will provide, and also its operating parameters.
Configure the IVR Profile parameters in Genesys Administrator:
• Configure the parameters in the gvp.service-parameters section and, if
applicable, in the dbmp section on the Provisioning > Voice Platform >
IVR Profile > <IVR Profile> > Options tab.
• Configure all the other IVR Profile parameters on the Provisioning >
Voice Platform > IVR Profile > <IVR Profile> > Option tab.
For more information about using Genesys Administrator to add or modify
configuration sections and options, see Procedure: Viewing or modifying GVP
Table 16 describes the IVR Profile configuration options.
Notes: All changes to IVR Profile configuration options take effect with the
next session that uses the IVR Profile.
The alarm and response codes are not independent policies. Resource
Manager will look for the corresponding alarm and/or response code
parameters from the matched tenant/profile only.
Table 16: IVR Profile Configuration Options
gvp.general Section
Conf Max Size (For Conference only) The maximum number of Any unsigned integer.
participants in the conference. Default value: 20
This setting does not override conference size
maximums that are configured for the Resource
Manager logical group or the conference resource
itself.
Service Type The default type of service that the IVR Profile • ccxml
provides. • conference
The default service type does not preclude the use • voicexml
of other service types within the application as
• announcement
well.
Toll Free Number The toll free number that is used by this IVR A string of characters.
Profile. Default value: Empty
User’s Guide 109

Table 16: IVR Profile Configuration Options (Continued)
SIP Session Timer The timeout value, in seconds, for the SIP session Any unsigned integer.
Interval that it executed for this IVR Profile. If the Default value: Empty
Resource Manager receives no SIP messages
associated with this call leg within the timeout
interval, it considers the call leg to have ended.
For the call leg that is associated with this IVR
Profile, the value of this sip.sessiontimer
parameter overrides session expiry timeouts that
are set at the level of the tenant, the Resource
Manager, and the resource.
Virtual Reporting Specifies the Virtual Reporting Object 1 and A string of characters.
Tag 1 Virtual Reporting Object 2. Default value: Empty
Virtual Reporting These parameters enable you to query and
Tag 2 correlate call data with custom parameters based
on business needs.
For example, if you are running a certain
campaign, you may want to associate calls with a
virtual reporting object value of “Last Campaign
2009” and later query call data by that, not having
to deal with which DIDs, IVR Profiles or platform
instances were utilized for that campaign.
gvp.log Section
metricsfilter The filter that determines which metrics (for calls <FilterID1>[,<FilterID2>,
that are made to this IVR Profile) will be ...]
forwarded to the Reporting Server. where:
If this parameter is set, the value will override the • <FilterID> is a single
default DATAC filter for the component for sessions Metric ID or a range of
that execute under this IVR Profile. This overrides Metric IDs. For the valid
the default parameter that is set at the platform Metric IDs, see the
level. Genesys Voice
The Resource Manager passes this property value Platform 8.5 Metrics
to the component in a SIP custom header. Reference.
The wildcard character (*)
means “all”.

gvp.policy Section
<Service> Allowed Specifies whether a Resource Manager session is • True

allowed to use the selected service. Valid values • False
for <Service> are:
Default value: True
• Announcement
• ccxml
• Conference
• cpd
• MSML
• Outbound Call
• Treatment
• VoiceXML Dialog
CTI Allowed Specifies whether calls that use this IVR Profile • True—Use
use CTI Connector to interface with IVR Server, CTI Connector
or send directly to Media Control Platform (MCP) • False—Use MCP or CCP
or Call Control Platform (CCP).
Default value: True
Transfer Allowed Specifies whether a Resource Manager session is • True

allowed to perform a transfer by using a SIP REFER • False
request within the existing SIP session.
Default value: True
Allow Burst Usage. Specifies whether burst usage for an application is • True
allowed for various usage-based policies. • False
Note: This parameter applies to individual objects Default value: False
(tenant/profile) only; it’s not applicable for
hierarchical use.
Raise Alarm for Specifies whether to raise an alarm when the burst • True
Exceeding Burst limit has been exceeded. • False
Limit
Dialing Rule Based The SIP response code that is sent in the SIP • <sipcode>;<desc>
Rejection Response response when a call is rejected because of a • <sipcode>
Code dialing rule (gvp.policy.dialing-rules.
where:
rule-<n>).
• <sipcode> is an integer in
the range of 400–699.
• <desc> is any string.
Default value: 403
User’s Guide 111

Raise Alarm for Specifies whether an alarm will be raised for the • True
Dialing Rule Based corresponding policy violation. • False
Rejection
Usage Limits Note: GVP has the ability to set policy limits for Level 1, Level 2 and Level 3
simultaneous use of specific services, and also usage limit alarms as well.
Usage Limits The number of times that a Resource Manager Any unsigned integer.
session can be concurrently in use in the context Default value: Empty
of any IVR Profile.
<service> level2 Specifies the number of times a Resource Any integer

Usage Limit (burst Manager session can concurrently be in use in the Default value: Empty
limit) context of the IVR Profile for level2 burst.
Valid values for <service> are:
• announcement
• ccxml
• conference
• inbound
• msml
• outbound
• voicexml
<service> level3 Specifies the number of times a Resource Any integer

Usage Limit (burst Manager session can concurrently be in use in the Default value: Empty
limit) context of the IVR Profile for level3 burst.
• announcement
• ccxml
• conference
• inbound
• msml
• outbound
• voicexml

<service> Usage The maximum number of concurrent service Any unsigned integer.
Limit sessions that are permitted for this IVR Profile Default value: Empty
• announcement
• ccxml
• conference
• inbound
• msml
• outbound
• voicexml
<service> Usage The number of times that the specified service Any unsigned integer.
Limit per session may be invoked in the context of this instance of a Default value: Empty
Resource Manager session. Valid values for
<service> are:
• announcement
• ccxml
• conference
• msml
• voicexml
<service> Usage The SIP response code that is sent in the SIP • <sipcode>;<desc>
Limit Exceeded response when a request for a service is rejected • <sipcode>
Response Code because the usage limits for that service
where:
(gvp.policy.<service>-usage-limit or gvp.
policy.<service>-usage-limit-per-session) • <sipcode> is an integer in
have been reached. Valid values for <service> the range of 400–699.
are: • <desc> is any string.
• ccxml Default value: 503
• conference
• voicexml
• outbound
• inbound
• announcement
• msml
User’s Guide 113

Usage Limit The SIP response code that is sent in the SIP • <sipcode>;<desc>
Exceeded Response response when a call is rejected because the usage • <sipcode>
Code limits that are specified in the following
where:
configuration options have been reached:
• <sipcode> is an integer in
• gvp.policy.usage-limits
the range of 400–699.
• gvp.policy.outbound-usage-limit
• <desc> is any string.
• gvp.policy.inbound-usage-limit
Default value: 480
(Temporarily unavailable)
<service> Not corresponding policy violation. Valid values for • False
Allowed <service> are:
• ccxml
• conference
• outbound-call
• transfer
• voicexml-dialog
• announcement
• msml
<service> Usage corresponding policy violation. Valid values for • False
Limit Exceeded <service> are:
• ccxml
• conference
• inbound
• outbound
• voicexml
• announcement
• msml

<service> Forbidden Specifies the error response code to send when the <sipcode>;description
Response Code <service> is not allowed. Valid values for or
<service> are:
<sipcode>
• ccxml
Default value: 403
• conference
For more information on SIP
• inbound Response Codes, see
• outbound Appendix D, “SIP Response
• voicexml Codes,” on page 465.
• announcement
• msml
Disable Video Specifies whether to disable video. • True—Disable video

• False—Enable video
Disable <Codec> Specifies whether to disable <Codec> • True—Disable

transcoding. The values for <Codec> are: transcoding
• G729 • False—Enable
• AMR-NB transcoding
• AMR-WB Default value: False
User’s Guide 115

mcp-asr-usage- Specifies whether there will be one Voice • per-call

mode Resource Management (VRM) session for the • per-utterance
entire call, or whether a separate VRM session
Default value: per-call
will be opened for each recognition request.
A single session for the entire call
(mcp-asr-usage-mode = per-call) means that
each call may have multiple recognition sessions.
If this parameter is set not to enable a single
session for the entire call (mcp-asr-usage-mode =
per-utterance), each VRM session is closed
when the recognition request completes, either
successfully or unsuccessfully (such as no match).
Therefore, each call may have multiple VRM
sessions.
The Resource Manager passes this value to the
Media Control Platform in a Request-URI
parameter. The value of this parameter overrides a
similar parameter that is set for the Media Control
Platform overall (asr.load_once_per_call), if the
settings are not consistent. See the description of
the asr.load_once_per_call parameter on
page 156 for more information about the
implications of this setting.
MCP Send/Receive Specifies whether a Media Control Platform is • True

Enabled allowed to perform <send> and <receive> • False
extensions. The Resource Manager passes this
Default value: True
value to the Media Control Platform in a
Request-URI parameter.
MCP Specifies the number of subdialogs allowed in a An unsigned integer.

max-subdialog-depth VoiceXML call. Default value: Empty

<service> Capability A list of name-value pairs that specify the <cap_NameA>=<cap_ValueA>

Requirement capabilities that are required when the specified [;<cap_NameB>=<cap_
<service> is invoked in the context of this IVR ValueB>;...]
Profile. where:
Valid values for <service> are: • <cap_NameX> is the name
• ccxml of the capability.
• conference • <cap_ValueX> is a
• voicexml comma-separated list of
values.
• msml
Example:
• announcement
lang=en-US;grammar=grxml,
Items in the name-value pair list are separated by a gsl
semicolon (;). The value side of each name-value
pair can itself be a comma-separated list of Default value: Empty
capabilities. Each set of values must be unique.
The Resource Manager will direct interactions to a
resource group only if the resource group
capabilities exactly match the capability
requirements that are specified in this option (see
the <Logical Group>.capability option in
Table 14 on page 92).
User’s Guide 117

Use Same Gateway (For gateway service only) Specifies whether • always—The Resource
outbound calls to a gateway must use the same Manager must forward
gateway that the Resource Manager session is the request to exactly the
currently using. same gateway resource
already associated with
A gateway resource becomes associated with a
the Resource Manager
Resource Manager session when (a) the Resource
session, or else the
Manager session is not already associated with
request fails.
another gateway resource and (b) one of the
following occurs: • preferred—The
Resource Manager first
• The Resource Manager receives a request from tries to forward the
a gateway resource. request to the gateway
• The Resource Manager receives a request for a resource already
gateway service and allocates it in accordance associated with the
with the load-balancing scheme for the group. Resource Manager
session, but tries other
gateways if the first
request fails.
• indifferent—The
Resource Manager
chooses a gateway in
accordance with
load-balancing scheme
for the group.
Default value: always
Prediction Factor Specifies the ratio of agent calls to customer calls Any integer range between
from Outbound Contact Server (OCS) for a 0.33—1.0.
campaign to minimize bridging when multiple Default value: 0.5
MCPs are present in the environment.

gvp.policy.dialing-rules Section
rule-<n> For each <n>, this parameter specifies a dialing <rule-type>;<regex>

rule that the Resource Manager will use to where:
determine if an address towards a gateway is
allowed. • <rule-type> is either a or
r, where a = allow and
<n> is a positive integer in the range of 1–10000. r = reject.
The rules are applied in rule number order. • <regex> is a regular
Example: expression.
To reject outbound calls to 911, allow calls to Default value: Empty
toll-free numbers, and reject calls to long-distance
numbers, specify the following set of rules:
gvp.policy.dialing-rules.rule-1: r,911
gvp.policy.dialing-rules.rule-2: a,1800*
gvp.policy.dialing-rules.rule-6: r,1*
Note: Resource Manager will search to match a
dialed number with the rejected expressions first.
Then, it will then search to match the dialed
number with the accepted expressions. If not
found, the dialed number is accepted.
User’s Guide 119

gvp.policy.call-info Section
rule-<n> For each <n>, this parameter specifies a set of <type>;<entity>;<regex>;<

DNIS, ANI, and User-Agent rules as obtained action>
from the SIP message. where:
<n> is a positive integer in the range of 1–100. • <type> is either a, r, or s,
The rules are applied in rule number order. When where:
a rule is matched, subsequent rules are ignored. a = accept.
Notes: r = reject.
These rules apply only to inbound calls. s = script play.
DNIS and User-Agent rules are applicable for the • <entity> is either ani,
Tenant. dnis, or ua, where:
ANI based rules are applicable for the IVR ani = ANI.
Profile.
dnis = DNIS.
Example:
ua = SIP User Agent.
rule-1 = r;ani;408666$;503
• <regex> is a regular
rule-2 = expression.
s;dnis;^[0-9]234$;voicexml,http://www.gvp.c
om/play-error.vxml.
• <action> is set based on
the <type> value, as
follows. If:
<type> = a, <action> is
empty
<type> = r, <action> is
<response-code>,<respo
nse-text>
<type> = s, <action> is
<service-type>,<url>
gvp.policy.speech-resources Section
ASR default engine Specifies the default ASR engine to use. <vendor>?Protocol
Example: Nuance?MRCPv2 where:
• <vendor> is a supported
ASR Vendor.
• Protocol is either MRCPv1
or MRCPv2

TTS default engine Specifies the default TTS engine to use. <vendor>?Protocol
Example: Realspeak?MRCPv2 where:
• <vendor> is a supported
TTS Vendor.
• Protocol is either MRCPv1
or MRCPv2
ASR/TTS default Specifies the default ASR or TTS language. An alphanumeric string.
language Default value: Empty
gvp.service-parameters Section
<service>.<param- Each parameter that you create in this section <value-type>, <value>
name> takes the form of a pair of strings that determine where <value> is any string
whether a Request-URI parameter called and <value-type> is:
<param-name>, with a value specified in <value>,
will be included in forwarded SIP requests. • undefined—The SIP
Request-URI parameter
Valid values for <service> are: with the specified
• announcement <param-name> will not be
• ccxml in the forwarded request
(even if the parameter
• conference was already in the
• cti incoming request).
• gateway • fixed—The parameter
<param-name>=<value>
• voicexml
will be in the SIP
The Resource Manager will apply this parameter Request-URI.
to a SIP request only if the specified <service> is
• default—If the SIP
invoked by the SIP request.
Request-URI parameter
• Setting the <value-type> to undefined deletes with name <param-name>
the <param-name> parameter from the incoming already exists, it will be
SIP request. left unmodified in the SIP
• Setting the <value-type> to fixed overrides Request-URI, but if the
the <param-name> parameter value in the incoming request does not
incoming SIP request. already include the
parameter, the parameter
• Setting the <value-type> to default provides a
<param-name>=<value>
default value for the <param-name> parameter
will be added to the SIP
in the outgoing SIP request, if the
Request-URI.
<param-name> parameter does not already exist.
User’s Guide 121

gvp.service-prerequisite Section
Alternate VoiceXML (For voicexml service only) The URL to an Any valid URL.
URL alternative initial page that the Media Control Default value: Empty
Platform will use if the request to the Initial
Page URL fails. Before it forwards the service
request, the Resource Manager inserts this
information as the value of the
gvp.alternatevoicexml SIP parameter.
Announcement URL (Mandatory for announcement service) Specifies Any valid URL.
the play parameter. Default value: Empty
Conference Id (Mandatory for conference service) The Any alphanumeric string,

conference identifier. without spaces.
Before it forwards the service request, the Default value: Empty
Resource Manager replaces the user part of the
SIP Request-URI with conf=<conference-id>.
The Resource Manager uses the conference-id to
ensure that it routes all requests for the same
conference to the same conference resource, even
if the requests originate from different Resource
Manager sessions.
Default Properties (For voicexml service only) The URL to a page Any valid URL.
Page that contains the default properties and handlers. Default value: Empty
Before it forwards the service request, the
Resource Manager inserts this information as the
value of the gvp.defaultsvxml SIP parameter.
Initial Page URL (Mandatory for voicexml and ccxml services) The Any valid URL.
URL of the initial page that is to be invoked. Default value: Empty
value of the voicexml or ccxml SIP parameter.
gvp.context-services-authentication Section
Context Service Specifies the username that will be used for An alphanumeric string.
Username context services authentication. Default value: Empty
Context Service Specifies the password that will be used for An alphanumeric string.
Password context services authentication. Default value: Empty

Chapter 6: Provisioning IVR Profiles Operational Parameter Management and Self-Service Applications
OPM Section
transaction object Specifies the transaction or list object DBID that Any string of integers
ID will be referenced during runtime of this profile. (except 0 [zero]).
Operational Parameter Management and

Self-Service Applications
Genesys offers a means of controlling the behavior (logic) of routing and
VoiceXML applications, using a common method called Operational
Parameter Management (OPM). Through a web interface, a customer, business
manager, or any third party who has access permission, can modify the value
of a set of custom parameters. When the routing or VoiceXML application
executes in servicing a call, its behavior can be changed based on OPM
parameter values. The parameters, possible values and logic to handle them is
designed by the application developer (routing strategy or GVP VoiceXML
application), and the set of parameters used within an application(s) are
configured for end-user access, and eventually accessed, using the Genesys
Administrator Extension (GAX) interface.
An end-user—a managed services administrator, a business marketing person,
or even a hosted services customer—can be given security access to specific
OPM parameter set(s) via the web. Using dropdown menus, data entry, or radio
buttons, the end-user can change the behavior of an application to match their
needs without ever knowing how the application was created or where it is
executed.
Once configured in GAX, an OPM set can be assigned to a routing strategy or
configured within a GVP IVR Profile, or even both. GVP supports up to two
parameter sets per IVR Profile (application) and is configured as a setting in
the IVR Profile. When the VoiceXML application executes, the GVP
interpreter fetches the named parameter set for use as operational variables
within the application.
IVR Profile Configuration for GVPi

Configure the IVR Profile parameters in Genesys Administrator:
• Configure the parameters in the gvp.service-parameters section and, if
applicable, in the dbmp section on the Provisioning > Voice Platform >
IVR Profile > <IVR Profile> > Options tab.
User’s Guide 123

Chapter 6: Provisioning IVR Profiles IVR Profile Configuration for GVPi
• Configure all the other IVR Profile parameters on the Provisioning >
Voice Platform > IVR Profile > <IVR Profile> > Option tab.
For more information about using Genesys Administrator to add or modify
configuration sections and options, see Procedure: Viewing or modifying GVP
Table 17 section describes the specific IVR Profile configuration options for
GVPi.
Note: All changes to IVR Profile configuration options take effect with the
next session that uses the IVR Profile.
Table 17: IVR Profile Configuration Options for GVPi
App Module Name Specifies which interpreter that Media Control • VXML-LGVP—GVPi
Platform is to use. • VXML-NG—Next
Generation
Default value: VXML-NG
Enable Debugging Specifies whether to enable debugging. • True—Disable debugging

• False—Enable
debugging
ASR Platform Used Specifies the ASR speech engine that is used for Comma separated string.
for Recognition voice recognition. Default value: Empty
Enable Record Specifies whether to record utterance files. • True—Record utterance

Utterance files
• False—Do not record
utterance files
Default value: True
Bad XML Page Specifies the URL to which unsuccessfully parsed Any URL.
Hook VoiceXML pages are written. Default value: Empty
Note: This option is available only if the
voicexml.gvpi.$adn-flag$ (Enable Debugging)
option is set to 1.

Table 17: IVR Profile Configuration Options for GVPi (Continued)
Call Trace Hook Specifies the URL to which call trace information Any URL.
is written. Default value: Empty
option is set to 1.
CTI - End Call when Specifies whether to end a call if an agent hangs • True
Agent Hangs up up when using a route request through IVR Server. • False
CTI - Reroute Specifies how long the platform waits before Any integer range from 3–
Timeout ending the call if the Agent leg ends without 60.
initiating a ReRoute. This is only applicable when Default value: 60
CTI - End Call when Agent Hangs up is set to
False.
Debug Hook Specifies the URL to which debugging Any URL.

information is written. Default value: Empty
option is set to 1.
Default Language The default language for the VoiceXML A string.

application. Default value: en_US
IVR Timeout The timeout interval, in seconds, for the initial An integer range from 0 –
page url to execute before trying the alternate page 300
url. Default value: 0
Dial out Number Specifies the phone number for calls to transfer to Any integer.
if there is an error in the IVR Profile. Default value: Empty
CPA Timeout The timeout interval, in seconds, for call progress Any unsigned integer.
analysis. Default value: Empty
Dump Fetched Specifies whether to move fetched VoiceXML • True—Move files to the
Pages pages to the <mcp installation>\tmp folder. temp folder
• False—Do not move files
to the temp folder
User’s Guide 125

Enable setting Specifies whether to enable the last utterance with • True—Enable
application.lastresult dtmf nomatch when using an on-board DTMF • False—Disable.
$.utterance with recognizes.
Default value: True
DTMF nomatch If set to False, when ASR is disabled, and the
result application throws a nomatch, the
application.lastresult$.utterance parameter
is not populated with invalid digits.
Transfer Option Specifies the type of SIP transfer. • SipRefer

Note: If transfer type is set to 1—Signal Channel, • ReferWithReplace
this option must be set to SipRefer, • AttCourtesy
ATTCourtesy, ATTConsultative, or
• ATTConsultative
ATTConference. If transfer type is set to 2—Signal
Channel, this option must be set to • ATTConference
ReferWithReplaces, or empty. • AttOOBCourtesy
• ATTOOBConsultative
• ATTOOBConference
Default value: Not set.
Transfer Type Specifies whether the transfer that is requested is a • 1—Signal Channel—
blind transfer (one step), or a bridge and Blind
consultation transfer (two step). • 2—Signal Channel—
Bridge and consultation
Default value: 2—Signal
Channel
Transfer Connect Specifies the script to execute when establishing a Any URL.
Url transfer using the AT&T switch. Default value: Empty
Reclaim Code Specifies the sequence of DTMF tones to use Any valid DTMF sequence.
(ATT only) when removing the caller from hold during an Default value:*7
ATTConference transfer, after the conference with
the Agent is completed.
Transfer Connect Specifies whether to enable or disable Transfer • True—Enable

Connect functionality. Works with Transfer • False—Disable.
Connect Script. This parameter is used for
backward compatibility.
Transfer Connect Specifies the script to use if the Transfer Connect A string of characters.
Script parameter is enabled. This parameter is used for Default value: Empty
backwards compatibility.

Chapter 6: Provisioning IVR Profiles IVR Profile Configuration for Cisco ICM
Trap Hook Specifies the URL for sending SNMP traps. Any URL.
TTS Vendor Specifies the Text-to-Speech (TTS) vendor that is Comma separated string.
used. Default value: Empty
TTS Gender Specifies the voice gender that is used for TTS. • Male
• Female
Default value: Male
gvp.service-prerequisite Section
Alternate The URL to an alternative initial page that the Any valid URL.
VoiceXML URL Media Control Platform will use if the request to Default value: Empty
the Initial Page URL fails.
value of the gvp.alternatevoicexml SIP
parameter.
Initial Page URL The URL of the initial page that is to be invoked. Any valid URL.
Before it forwards the service request, the Default value: Empty
value of the voicexml or ccxml SIP parameter.
IVR Profile Configuration for Cisco ICM

Table 18 describes the options that must be configured in the IVR Profile to
support the CTI Connector and Cisco Intelligent Contact Management (ICM)
call flows through NGI.
Table 18: IVR Profile Configuration Options to Support CTIC/ICM
ICM Service ID A static, unique service ID that indicates the A string of characters.
ICM service that is associated with this IVR Default value: fixed
Profile.
User’s Guide 127

Chapter 6: Provisioning IVR Profiles Mapping IVR Profiles to DID Numbers
Table 18: IVR Profile Configuration Options to Support CTIC/ICM (Continued)
Script Mapping Specifies whether the ICM routing script for the Choose one of two option
IVR Profile is chosen, based on the Toll-Free values: fixed,TFN or
Number (TFN) or the DNIS. For example, the fixed,DNIS
ICM script can be specified by using the DNIS. Default value: fixed,TFN
If so, the default value would be a TFN and the
applicable values could be either a TFN or the
DNIS.
Use Bridge Transfer When enabled, specifies that a BRIDGE transfer Choose one of two option
will be invoked by the CTI Connector to values: fixed, TRUE or
connect the caller to agent. When disabled, (by fixed, FALSE.
default) a BLIND transfer is triggered. This Default value: fixed,TRUE
feature is applicable for Service Control
Interface only, when the CONNECT message is
received with TransferHint flag set to false.
CTI Default Agent Specifies the default agent number to which the A string of characters.
Number CTI Connector will send a transfer (to an agent) Default value: fixed
if an ICM CONNECT message is sent with the
label type set to DEFAULT.
gvp.general Section
Toll Free Number A unique identifier for the ICM script which is String of integers.
an attribute that is configured in the IVR Default value: 0-9
profile. The Resource Manager sends the
toll-free number attribute in the Request-URI
that is sent to the CTI Connector in the
tollfreenum format
gvp.policy Section
CTI Allowed Specifies that this IVR Profile is enabled for • True—Enable
CTI functionality through the CTI Connector. • False—Disable.
Mapping IVR Profiles to DID Numbers

DID numbers are the DNs that are obtained from Dialed Number Identification
Service (DNIS).
The Resource Manager can be configured so that it obtains DNIS information
from SIP Server (see rm.sip-header-for-dnis on page 87).

• If your GVP configuration includes a mapping of IVR Profiles to DIDs,

the Resource Manager will use the DNIS information to determine which
IVR Profile to invoke for the session.
• If you do not map IVR Profiles to DIDs, the Resource Manager will use
the default IVR Profile that you specify for the tenant.
All options described create the DID mappings as sections in the Tenant’s
Annex tab. Genesys recommends that you use Genesys Administrator’s IVR
Profile Wizard to map the DIDs.
The following procedure describes how to create mapping rules.
Procedure:
Mapping IVR Profiles to DIDs
Purpose: To associate IVR Profiles with DIDs so that the Resource Manager
can use DNIS information to invoke the required GVP services.
Prerequisites
• The IVR Profiles have been created if the DID Group is to be mapped to a
specific IVR Profile.
Note: DID Groups can be created without IVR Profiles, and assigned to
IVR Profiles at a later time.
For more information about creating an IVR Profile, see the chapter about
Guide. For more information about configuring the IVR Profile, see “IVR
Profile Configuration Options” on page 109.
Start of procedure
1. Go to the Provisioning > Voice Platform > DID Groups.
2. In the menu bar of the tab, click New.
The Property screen displays (see Figure 6).
User’s Guide 129

Figure 6: Specifying a DID Mapping
3. In the Name text box, enter the name of the DID Group.
4. From the IVR Profile drop-down list, select the required IVR Profile.
5. In the Click DIDs box, click Add.
The Add/Edit DID dialog box displays (see Figure 7).
Figure 7: Add/Edit DID
6. Enter the DID.

7. Click OK.
Continue entering the DIDs until finished.
Note: The DIDs can be entered as either a single DID (for example,
100), a range of DIDs (for example, 100-199), or a DID prefix (for
example, 100*).
8. To edit or delete existing DIDs from the group, select the DID you want to
change, and click either Edit or Remove.

Chapter 6: Provisioning IVR Profiles DID Group Bulk Operations Wizard
9. Click Save.
Note: To add, edit or delete multiple DIDs, see “DID Group Bulk
Operations Wizard” on page 131.
End of procedure
DID Group Bulk Operations Wizard

The DID Group Bulk Operations Wizard allows you to add, delete, and move
multiple DIDs among DID groups. You can also create new DID groups with
the wizard. The following procedure describes how to use the DID Group Bulk
Operations Wizard.
Procedure:
Using the DID Group Bulk Operations Wizard
Purpose: To add, delete, and move DIDs using the DID Group Bulk
Operations Wizard.
Prerequisites
• The IVR Profiles have been created.
Start of procedure
1. Go to the Provisioning > Voice Platform > DID Groups.
2. Optionally, select the DID Group you want to change.
3. In the Tasks panel, click Bulk Operations Wizard to invoke the wizard.
4. After reading the introduction, click Next to start the wizard.
User’s Guide 131

5. Select the appropriate operation.

a. Select Add to add multiple DIDs to the selected group, or to create new
DID groups, then click Next.
i. Click Browse to select the CSV file containing the list of new DIDs
.
Note: The uploaded file must be a CSV file with the following
columns (in the given order):
• DID
• DID Group
• Tenant
A DID is either a single DID (in the form <did>), a range of
DIDs (in the form <start>-<end>) or a DID prefix (in the form
<prefix>*). Lines of text that don’t match these patterns are
considered invalid.
A DID Group is the name of either an existing group or a new
group that is to be created. This column is optional and
defaults to the selected DID Group (if one was selected during
the launching of the wizard).
A Tenant is the name of the tenant that owns (or will own) the
specified DID Group. This column is optional and defaults to
the current tenant.
Because DID Group and Tenant columns are optional, a flat
list of DIDs can be uploaded (instead of a CSV) for
addition/moving into of DIDs into the selected (in the DID
group list) DID group.
ii. Click Next.

iii. Review the Confirmation screen for a summarization of the
operation.
Note: The summary includes the counts for invalid DIDs, valid
DIDs, and DIDs that currently belong to other DID Groups.
iv. Click Finish.

b. Select Move to move the specified DIDs to a selected group, then click
Next.
i. Click Browse to select the file containing the list of new DIDs.
Note: The uploaded file must be a CSV file with the following
columns (in the given order):
• DID
• DID Group
• Tenant
A DID is either a single DID (in the form <did>), a range of
DIDs (in the form <start>-<end>) or a DID prefix (in the form
<prefix>*). Lines of text that don’t match these patterns are
considered invalid.
A DID Group is the name of either an existing group or a new
group that is to be created. This column is optional and
defaults to the selected DID Group (if one was selected during
the launching of the wizard).
A Tenant is the name of the tenant that owns (or will own) the
specified DID Group. This column is optional and defaults to
the current tenant.
ii. Click Next.

operation.
iv. Click Finish.

c. Select Delete to remove the specified DIDs from the selected group,
then click Next.
i. Click Browse to select the file containing the list of new DIDs.
Note: The file must be a text file containing a list of DID numbers.
Each line is either a single DID (for example, 100), a range of
DIDs (for example, 100-199), or a DID prefix (for example,
100*). Lines of text that do not match these patterns are
ignored.
ii. Click Next.
User’s Guide 133

Chapter 6: Provisioning IVR Profiles Data Retention Policy Wizard

operation.
iv. Click Finish.

You can download these generated CSV files for diagnostic or auditing
purposes.
End of procedure
Data Retention Policy Wizard

The Data Retention Policy Wizard enables data retention policies for data
retained by the Reporting Server for Tenants and IVR Profiles easily. The
wizard allows the step-by-step configuration for the data retention policies for
call detail reporting, operational reporting, VAR, and service quality data.
Retention policies can also be configured in the Reporting Server application
in Genesys Administrator. For more information, see “Configuring Database
Retention Policies” on page 274.
Procedure:
Using the Data Retention Policy Wizard
Purpose: To create data retention policies using the Data Retention Policy
Wizard.
Prerequisites
• The IVR Profiles have been created.

Chapter 6: Provisioning IVR Profiles Data Retention Policy Wizard
Start of procedure
1. Go to the Provisioning > Voice Platform > IVR Profiles.
2. Under the Tasks panel, select Configure Data Retention Policies to
invoke the wizard.
The Introduction screen appears.
3. Click Next.
4. If you do not want to use the default values, on the CDR Data Retention
screen, enter the required duration for Call Log Events, and CDR.
Note: The value configured for Call Log Events must be less than the
value configured for CDR for the Call Log Events value to be valid.
5. Click Next.
6. If you do not want to use the default values, on the OR Data Retention
screen, enter the required durations for the following:
• Five-minute Summaries
• Thirty-minute Summaries
• Hourly Summaries
• Daily Summaries
• Weekly Summaries
• Monthly Summaries
7. Click Next.
8. If you do not want to use the default values, on the VAR Data Retention
screen, enter durations for the following:
• Daily Summaries
9. Click Next.
10. If you do not want to use the default values, on the SQ Data Retention
screen, enter the required durations for the following:
• Daily Summaries
11. Click Next.
User’s Guide 135

Chapter 6: Provisioning IVR Profiles IVR Profile Configuration for Tenants
12. If you do not want to use the default values, on the Latency Data
Retention screen, enter durations for the following:
• Daily Summaries
13. Click Next, and then Finish to save the values.
For more information on the default values, see “Configuring Database
Retention Policies” on page 274.
End of procedure
IVR Profile Configuration for Tenants

The Tenant configuration options determine how the Resource Manager will
use IVR Profiles in the tenant’s environment.
Setting options at the tenant level sets values that are inherited as defaults by
the IVR Profiles. You can override these settings for individual IVR Profiles
by setting different values for the equivalent options in the IVR Profile.
Configure the tenant parameters in Genesys Administrator on the Provisioning
> Environment > Tenants > <tenant> > Options > Advanced View (Annex)
tab.
All changes to these parameters take effect with the next session that uses the
IVR Profile.
gvp.general Section
Table 19 describes the parameters in the gvp.general section. These
parameters specify general configuration information for the Resource
Manager in the tenant’s environment.

Table 19: General Tenant Configuration Options
Parameter Name Description Valid Values and Syntax
gvp.general Section
Default Application (Mandatory) The default IVR Profile for a request <IVR Profile>
to the Resource Manager. The Resource Manager where <IVR Profile> is the
uses the default IVR Profile if the incoming name of the IVR Profile that
request does not contain information to map the you assigned when you
request to an application. created the IVR Profile
object.
SIP Session Timer The timeout value, in seconds, for the SIP session Any positive integer.
Interval that executes for this IVR Profile. If the Resource Default value: Empty
Manager receives no SIP messages associated
with this call leg within the timeout interval, the
Resource Manager considers the call leg to have
ended.
For the call leg associated with this IVR Profile,
the value of this sip.sessiontimer parameter
overrides session expiry timeouts that are set at
the level of the Resource Manager, but may be
overridden by the sip.sessiontimer setting for
the IVR Profile.
For more information about how the Resource
Manager uses expiry timeouts to manage sessions,
see “Resource Manager Session Timers” on
page 76.
gvp.policy Section
The parameters in this section are identical to the configuration parameters in
the gvp.policy section of the IVR Profile object. These parameters enable you
to configure policies for the Resource Manager—for example, to specify
which requests the Resource Manager will allow, or to attach certain
Request-URI parameters to send to the endpoint to enable or disable particular
features. You can also specify Reporting Server tenant parameters in the
gvp.policy section.
For more information about the configuration options in the gvp.policy
section, see Table 16 on page 109.
User’s Guide 137

gvp.policy.sqa Section
The parameters in this section are determine the Reporting Server service
quality behavior for the tenant. Table 20 describes the import options in the
gvp.policy.sqa section.
Table 20: Service Quality Tenant Configuration Options
gvp.policy.sqa Section
Threshold for If the percentage of successful calls for an IVR An integer in the range of
Service Quality Profile falls below this threshold during a service -1–100.
Notification quality period, a notification is generated. Default value: -1
If set to -1, the Reporting Server default value is
used.
For more information on how to change the default

sqa.error.notification.threshold value for the Tenant in Genesys
Administrator, see the Framework 8.5 Genesys Administrator Help.
The parameters in this section are identical to the configuration parameters in
the gvp.service-parameters section of the IVR Profile object. The Resource
Manager uses these values to add, modify, or delete Request-URI parameters in
the SIP requests that it forwards.
For more information about the configuration options in the
gvp.service-parameters section, see Table 16 on page 109.
gvp.policy.<child-tenant> Section
The parameters in this section are those policies that Resource Manager must
enforce for a child tenant on behalf or a parent tenant. For more information
about the configuration options in the gvp.policy parameters section, see
gvp.dn-groups Section
Table 21 describes the parameters in the gvp.dn-groups section. These
parameters specify the mapping for DID groups in an Hierarchical
Multi-Tenancy (HMT) environment. The Resource Manager obtains DID
group information from this section.

Table 21: DID Groups Tenant Configuration Options
gvp.dn-groups Section
DN Group Name The name of each parameter represents the name • An individual DID, for
of the DID Group. example, 1000
The value contains the list of DIDs. The value is • A consecutive block of
a comma-separated string in which each value DIDs, for example,
represents one of the following: 1000-1999
• An individual DID • A prefix DIO with *
suffix, for example,
• A consecutive block of DIDs
1800555*
• A prefix DID with a * suffix
• Any number of the
above combinations
separated by commas for
a DID Group.
gvp.dn-group-assignments Section
Table 22 describes the parameters in the gvp.dn-group-assignments section.
These parameters specify the IVR Profile mapping for DID Group assignments
in an HMT environment.
Table 22: DID Group Assignments Tenant Configuration Options
gvp.dn-group-assignments Section
DN Group Name The DBID of the IVR profile to which the DID A positive integer.
group is mapped. The value must be a positive
integer.
Any tenant in the hierarchy can define DID
Groups. Each group can contain:
• An individual DN.
• A consecutive block of DNs.
• A prefix DN string (for example, 1234*).
Corresponding to these DID Groups, each tenant
can contain DID group assignments to specify the
target IVR Profile for each group. Only those IVR
Profiles that are at the current tenant level can be
assigned.
User’s Guide 139


Chapter
7 Configuring the Media

Control Platform
The Genesys Voice Platform (GVP) Media Control Platform (MCP)
component provides media-centric services. This chapter provides information
about the configuration of the Media Control Platform and, if required, the
provisioning of the resources for Automatic Speech Recognition (ASR) and
Text-to-Speech (TTS). This chapter contains the following sections:

Task Summary: Configuring the Media Control Platform, page 142
 Enabling ASR and TTS, page 146

Enabling Outbound Dialing, page 149

Media Server Markup Language, page 153
 Important Media Control Platform Configuration Options, page 154

Important MRCP Server Configuration Options, page 190
User’s Guide 141

Chapter 7: Configuring the Media Control Platform Task Summary: Configuring the Media Control Platform
Task Summary: Configuring the Media

Control Platform
Task Summary: Configuring the Media Control Platform summarizes the
configuration steps and options to implement Media Control Platform
functionality in your GVP deployment.
Task Summary: Configuring the Media Control Platform
Integrate the Media Control Platform 1. Point the Media Control Platform to the Resource Manager as
with the Resource Manager. the SIP Proxy server, and define the properties for SIP
communications. Key configuration options are:
 sip.transport.x

sip.routeset or sip.securerouteset
2. To secure SIP communications, ensure that you specify a
transport for the Transport Layer Security (TLS) protocol and
a secure routeset for outbound calls.
For additional, relevant configuration options, see
“Configuring SIP Communication and Routing” on page 38.
3. If you if intend to use the Call Recording Solution through
third-party recording servers, configure the following option
so that it points to the Resource Manager’s IP address and
port:

vrmrecorder.sip.routeset
(Optional) Secure the media channel 1. Enable Secure Real-time Transport Protocol (SRTP) by
between the Media Control Platform specifying the required mode (accept-only, offer, or
and the remote endpoint. offer_strict) in the mpc.srtp.mode parameter. By default,
SRTP is not enabled.
2. If necessary, modify the default values for the encryption and
authentication algorithms (the cryptographic suites) and
session parameters that the Media Control Platform will
advertise in the SDP crypto attribute:
mpc.srtp.cryptomethods
mpc.sessionparams
mpc.sessionparamsoffer
If required for your deployment, See “Enabling ASR and TTS” on page 146.
provision the third-party Media
Resource Control Protocol (MRCP)
servers for ASR and TTS.

Task Summary: Configuring the Media Control Platform (Continued)
Configure the IP DiffServ (ToS) and Set the RTP/RTCP packet and the SIP packets ToS using:
RTP/RTCP. [mpc] rtp.tos
[mpc] rtcp.tos
[sip] transport.[n].tos
See “Configuring SIP Communication and Routing” on page 38.
Configure logging. See “Configuring Logging” on page 62.
Tune Media Control Platform • Configure appropriate maximums and timeouts for your
performance. deployment. Consider the following options, in particular:
vxmli.cache.document.max_count (default is 50)
vxmli.cache.document.max_size (default is 1000000 bytes)
vxmli.max_num_documents (default is 2000)
vxmli.initial_request_fetchtimeout (default is 30000 ms)
vxmli.max_num_sessions (default is 10000)
• If your deployment includes ASR and TTS, consider the
following options, which affect the MRCP Client behavior:
vrm.client.timeout (default is 10000 ms)
stack.connection.timeout (default is 10000 ms)
• See also “Configuring Session Timers and Timeouts” on
page 76.
• It is usually not necessary to modify the default settings for
the media processing behavior of the Media Server (mpc and
mtinternal configuration sections). However, review buffer
and packet size related mediamgr.* and rtp.* options in the
mpc configuration section, to verify that they are optimal for
your deployment (especially if using video).
Customize Media Control Platform • Review and, if necessary, modify the configuration options in
behavior in relation to VoiceXML the vxmli configuration section (see the Genesys Voice
applications. Platform 8.5 Configuration Options Reference). Some of the
important NGI vxmli options are described in Table 23 on
page 156. Some of the important GVPi options are described
in Table 23.
• For Next Generation Interpreter (NGI), consider also the
parameters in the sip configuration section that specify what
parts of SIP messages are exposed to the VoiceXML
application (for example, in.invite.headers and
in.invite.parameters). For the list of SIP headers that are
known to GVP, see Table 96 on page 445.
User’s Guide 143

Customize Media Control Platform • Verify that settings for the following configuration options,
for Inband DTMF detection. which are required for DTMF detection, are suitable for your
deployment. Consider the following parameters:
mpc.rtp.dtmf.receive
mpc.rpt.dtmf.send
mpc.dtmf.detectedge
mpc.dtmf.maxsilence
mpc.dtmf.minduration
mpc.fcr.defaultdtmfhandling
mpc.record.defaultdtmfhandling
Customize Media Control Platform • Verify that the general CPA settings are suitable for your
for Call Progress Analysis deployment. Consider the following parameters:
cpa.maxpreconntime
cpa.maxpostconntime
cpa.maxbeepdettime
cpa.keptdur_before_statechange
cpa.priority_normal_machinegreetingdur
cpa.priority_normal_voicepausedur
cpa.priority_normal_maxvoicesigdur
cpa.priority_voice_machinegreetingdur
cpa.priority_voice_voicepausedur
cpa.priority_voice_maxvoicesigdur
cpa.priority_machine_machinegreetingdur
cpa.priority_machine_voicepausedur
cpa.priority_machine_maxvoicesigdur
cpa.faxdur
cpa.voice_range_db
cpa.maxrings
cpa.voice_level_db
cpa.preconn_tones_det_mode

Customize Media Control Platform • Re-define standard tones or add new custom tones if
for Call Progress Analysis necessary for your deployment:
(Continued) cpa.fax
cpa.ringback
cpa.busy
cpa.fastbusy
cpa.sit_nocircuit
cpa.sit_vacantcircuit
cpa.sit_operatorintercept
cpa.sit_reorder
cpa.custom1
cpa.custom2
cpa.custom3
cpa.custom4
cpa.tone[1-10].segment[1-3].f[1-2]min
cpa.tone[1-10].segment[1-3].f[1-2]max
cpa.tone[1-10].segment[1-3].ontimemin
cpa.tone[1-10].segment[1-3].ontimemax
cpa.tone[1-10].segment[1-3].offtimemin
cpa.tone[1-10].segment[1-3].offtimemax
Customize Media Control Platform • For NGI, if required, set the following configuration option to
for video recording. enable I-frame request during video recording:
mpc.rtp.request_iframe
Customize Media Control Platform • For PSTN Connector, if installed, the following configuration
for PSTN Connector prompt play options can be adjusted for your requirements:
support. mpc.playremoteflushtimeout
mpc.playremoteeodtimeout
mpc.rtp.prefilltime
Customize client-side See “Configuring Client-Side Connections” on page 68

Customize Media Control Platform See “Enabling Outbound Dialing” on page 149.
for Outbound dialing.
behavior and performance.
Customize Media Control Platform See “Configuring SNMP” on page 68 and Table 100 on
User’s Guide 145

Chapter 7: Configuring the Media Control Platform Enabling ASR and TTS
Customize Media Control Platform Create a VoIP Service DN and set the contact option to the
for SIP Server and MSML. Resource Manager IP address in the T-Server section.
For more information, see the Framework 8.5 SIP Server
Deployment Guide.
Customize Media Control Platform Consider the following configuration parameters:

for DTMF transmission method. mcp.rtp.dtmf.send
mcp.sdp.map.orign.[n].dtmftype

jitter buffer. mpc.rtp.recvaudiobuffersize
mpc.rtp.recvvideobuffersiz,
mpc.rtp.dejitter.delay
mpc.rtp.dejitter.timeout

for VoIP metrics reporting. mpc.voipmetrics.enable
sip.voipmetrics.localhost
sip.voipmetrics.registration
sip.voipmetrics.remoteserver
sip.voipmetrics.routeset
Enabling ASR and TTS

The following procedure describes how to create and configure MRCP server
Applications, to provision ASR and TTS speech resources for the GVP
deployment.
Procedure:
Provisioning ASR and TTS resources
Purpose: To provide an overview of the steps to configure logical GVP

MRCPv1 or MRCPv2 speech server Applications, to provide a presence for
third-party speech engines in the Genesys Configuration Layer.
Repeat this procedure as required to create the necessary Application objects.
You must create a separate Application for each third-party speech server in
your deployment. The Application type is Resource Access Point.

Prerequisites
• The Media Control Platform Installation Package (IP) is available.
Start of procedure
1. Create the MRCPv1 or MRCPv2 Application object.
a. Import the required Application Template from the Media Control
Platform Installation Package (IP).
The following Application Templates are available:.
• MRCPv1_ASR_IBM • MRCPv2_ASR_NUANCE
• MRCPv1_ASR_NUANCE • MRCPv2_TTS_NUANCE
• MRCPv1_ASR_TELISMA • MRCPv2_ASR
• MRCPv1_TTS_IBM • MRCPv2_TTS
• MRCPv1_TTS_NUANCE •
• MRCPv1_ASR •
• MRCPv1_TTS •
Note: The generic templates can be used for those vendors that are not
listed above.
b. Import metadata into the Application Template.

c. On the Provisioning > Environment > Applications tab, create and
name the new Resource Access Point, based on the required
Application Template.
For detailed information about importing Application Templates and
metadata, and creating Applications from the templates, see Appendix A
in the Genesys Voice Platform 8.5 Deployment Guide, which describes the
pre-installation activities.
Configure Resource Access Points:

2. If you are configuring the resource access points for the MRCP Server,
omit Step b. If you are configuring them for the MRCP Proxy, omit Step a:
MRCP Server a. On the Provisioning > Environment > Applications > <MRCP Server>
> Options tab, configure the options in the provision section.
• vrm.client.resource.type—Enter the resource type (ASR or
TTS).
User’s Guide 147

• vrm.client.resource.name—Enter a name for the speech resource.

(See template names in Step 1).
Note: MRCP resources of the same type (ASR or TTS) can be

assigned the same resource name. However, a resource name
that is used by a ASR resource cannot be used by a TTS
resource.
• vrm.client.resource.uri—Enter the MRCP Server’s RTSP URI.

For example, rtsp://<MRCPServer Host IP>:<MRCPServer
Port>/<suffix>
• vrmproxy.ping_interval—Enter a value for the interval between
pings (to the MRCP Proxy). (Not required if you have not
deployed the MRCP Proxy.)
MRCP Proxy b. On the Provisioning > Environment > Applications > <MRCP Proxy> >
Options tab, configure the options in the provision section.
• vrm.client.resource.type—Enter the resource type (ASR or
TTS).
• vrm.client.resource.name—Enter a name for the speech resource.
• vrm.client.resource.uri—Enter the MRCP Proxy’s RTSP URI.
For example, rtsp://<MRCPP Host IP>:<MRCPP Port>/<suffix>
For other important configuration options that you may need to be modify,
see “Important MRCP Server Configuration Options” on page 190.
Configure TTS Vendor-Specific Parameters:

3. If required, configure the TTS vendor-specific parameters that will be sent
in SET-PARAM requests:
a. In the provision section on the Provisioning > Environment >
Applications > <MRCP Server> > Options tab, add a new parameter,
vrm.client.TTSVendorSpecific.xxxxxx.
This defines one arbitrary TTS vendor-specific parameter to be sent to
the MRCP server.
b. Define as many vendor-specific keys as you require for the desired
vendor-specific key-value pairs, using the following format:
vrm.client.TTSVendorSpecific.param<n>=value<n>
c. Click Save or Apply to save the MRCP server configuration.
Creating the Connections:

MCP/ MRCP 4. If you are deploying the Media Control Platform without the MRCP Proxy:
Server • On the Provisioning > Environment > Applications > <Media Control
Platform> > Configuration tab, create the connection between the
Media Control Platform and the MRCP Server.

Chapter 7: Configuring the Media Control Platform Enabling Outbound Dialing
MCP/ MRCPP/ 5. If you are deploying the Media Control Platform with the MRCP Proxy:
MRCP Server • On the Provisioning > Environment > Applications > <Media Control
Platform> > Configuration tab, create the connection between the
Media Control Platform and the MRCP Proxy.
• On the Provisioning > Environment > Applications > <MRCP Proxy> >
Configuration tab, create the connection between the MRCP Proxy
and the MRCP Server.
For more details, see the procedure to assign the MRCP Proxy to the Media
Control Platform and the MRCP Server to the MRCP Proxy in the chapter
about post-installation activities in the Genesys Voice Platform 8.5
Deployment Guide.
End of procedure
Enabling Outbound Dialing

One of the most useful features in GVP 8.5 is the ability to initiate outbound
calls in an asynchronous manner through remote dialing and either configured
bi-directional or outbound channels.
Making a Call
You can use the telnet interface to connect to the preconfigured remote dialing
port (default 6999) to place outbound calls. The following example shows the
outbound request with a VoiceXML page:
pw@galahad 379>
pw@galahad 379> telnet localhost 6999
Trying 127.0.0.1...
Connected to localhost.
Escape character is ‘^]’.
PW RemoteDial>
call 4167360905 4167362012
http://www.genesyslab.com/helloworld.vxml 0001 Test
!CALL_SENT 1: telno:4167360905 dnis:4167362012
url:http://www.genesyslab.com/helloworld.vxml
uuidata:Test
PW RemoteDial>
!CALL_STATUS 1: CONNECTED: Line is connected.
PW RemoteDial>
!CALL_DROP 1 41: USER_END: User hung up call. (time spent
was 41 secs) (protocol reason: [DlgcChannel] User
hangup)
PW RemoteDial>
You can also use the command-line interface to make an outbound call. This
interface provides a number of useful commands. These include:
call <telno> <ani> <url> <refno> [uuidata] [defaults]
User’s Guide 149

[parameter_list]
The call command initiates an outbound call to the specified telephone number
(<telno>). The <telno> parameter can accept up to 1023 characters. This can
either be a sip: uri, or a tel: uri. If not specified, a tel: uri is assumed. If a
tel:uri is used, the defaultgw configuration option in the sip section must be
configured to point to a device that can handle the SIP call (for example, a
media gateway), so that the call can be forwarded. When connected, the
VoiceXML page referred to by the specified URL (<url>) is attached to the
call.
The value of the <ani> parameter is displayed in the CDR as the local.uri. If
the value of <ani> is a SIP URI, it is used in the From header of the SIP INVITE
request. If the value of <ani> is not a SIP URI, it is converted into a SIP URI,
and used in the From header of the SIP INVITE request.
The <platform ANI> parameter can accept up to 32 characters. The actual
number of ANI digits that can be delivered on PSTN depends on the
network—for example, the maximum number on ISDN T1 is 15.
The reference number (<refno>) parameter is a user-supplied identifier that is
used to associate status replies with the call initiation, and is unique for each
active call. This reference number must be an integer between 0 and
2147483647.
There are three other optional parameters that can be specified:
• [uuidata]—The user-to-user information element.
• [defaults]—The default VoiceXML page.
• [parameter_list]—The name value pair separated by the pipe (|)
character that is passed from the interface to the call manager.
The gvp.appmodule in the parameter list is used to specify the Next Generation
Interpreter used to execute the vxml page.
Note: GVPi does not support remote dialing.
You can also specify all the parameters before the gvp.appumodule parameter,
or specify the dash (-) character for the default value. For example, if
you wish to specify the parameter list, but not the uuidata and the defaults file,
use the following command:
PW RemoteDial> call 4167366493 2323
http://205.150.90.12/developer/main/cgi-bin/index.cgi
1223 - - NWNAME=dtiB1T21|NUMBERINGPLAN=0
Note: The size of <uuidata> cannot exceed 254 characters.
Other valid commands within the interface are:

• cc—Clears all message counters.

• dump—Toggles debug information.

• e, q, or x —Exit the command-line interface.
• setto <unit_type> <num_units>—Sets the timeout for the call setup.
• timelimit <seconds>—Sets the maximum number of seconds for the call.
The default is 604800 seconds.
• end <refno>—Ends the call at the Remote Dial interface.
• analysis y, n, or a—Enables dialogic call analysis.
• scall—Displays the calls for this session.
• scount—Displays the counters for this session.
• show—Displays the current settings. For example:
!SETTINGS: max_calls:500 timeout_length:120
timeout_unit:s call_analysis:enabled time_limit:604800
• ? or h—Displays the help information the interface.
Status Messages
Once the call has been placed with the CALL_SENT notification, there are two
possible status messages returned.
Note: The <refno> returned in the status message must match the one the
following:
• !CALL_SENT <refno>—telno:<telno> dnis:<dnis> url:<url>
uuidata:<uuid> defaults_file:<defaults>
parameter_list:<parameter_list>
• !SOCKET_ERROR <refno>— Socket not found
• !NO_REFNO—No reference number
• !INVALID_REFNO <refno>—Invalid reference number
• !TOO_MANY_CALLS <refno>—Too many calls in progress
• !INVALID_TELNO <refno>—Incorrect telephone number
• !INVALID_URL <refno>—Incorrect URL format
• !INVALID_UUIDATA <refno>—Incorrect UUIDATA format
• !INVALID_DEFAULTFILE—Incorrect DEFAULTFILE format
• !INVALID_PAIRLIST—Incorrect PAIRLIST format.
• !CALL_FAILED <refno>—telno:<telno> dnis:<dnis> url:<url>
uuidata:<uuid> defaults_file:<defaults>
parameter_list:<parameter_list>
In all these cases, except for CALL_SENT, there will be no further status
returned for this call attempt.
1. The call connects successfully:

!CALL_STATUS <refno> followed by one of the following:
User’s Guide 151


CONNECTED—Connected successfully.

MACHINE—Answering machine detected.

UNKNOWN_STATUS <status number>—Status is unknown.
The call is then dropped with the following message:
!CALL_DROP <refno> <timespent> <network disconnect reason>: <one of
the disconnect reason listed below:> <protocol reason: protocol
disconnect string>

USER_END—The caller disconnected the call.

APPL_END—The VoiceXML application disconnected the call.

TIMELIMIT_END—The timelimit of call is reached.

UNKNOWN_REASON <internal disconnect reason number>—The reason
for the disconnect is unknown.
2. The call does not connect and is dropped immediately with a !CALL_DROP
message instead of a !CALL_STATUS message.
!CALL_DROP <refno> <timespent> <network disconnect reason>:
<drop_status>. < protocol reason: protocol disconnect string>
where <drop_status> is one of the following:

MACHINE—Answering machine detected.

VXML_DECLINE—The VoiceXML Interpreter declined the call.
 BUSY—The connection is busy.

NO_ANSWER—There is no answer in <num_units> <unit_type>.

NO_RESOURCES—There are no free channels or media resources.
 CALL_FAILED—The call failed.

URLTIMEOUT—The fetch URL timed out.

BADURI—The URI type is invalid.
 NOAUTH—The network denied the call.

SHUTTINGDOWN—The Interpreter is shutting down.
 NETWORKTIMEOUT—The network timed out.
 BADDEST—The destination number is invalid.

UNSUPPORTED_URL—The URL is not supported.
 INVALID_TELNO—The telephone number is invalid.

USER_END—The caller disconnected the call.

UNKNOWN_REASON <internal disconnect reason number>—The reason
for the disconnect is unknown.
The <network disconnect reason>, and the <protocol disconnect string>
are returned by the callmanager to give more information about the reason
the call was dropped.

Chapter 7: Configuring the Media Control Platform Media Server Markup Language
Media Server Markup Language

GVP uses the Media Server Markup Language (MSML) application module to
control many different types of services for the Media Control Platform.
In the current release of SIP Server (up to version 7.6), media interactions can
be initiated with different components. For IVR interactions, SIP Server
integrates with GVP using VoiceXML applications, and for simple media
operations and conferencing, SIP Server integrates with the Stream Manager.
Simple media operations (conferencing, announcements, and simple dialog
management) are controlled using SIP, NETANN (RFC 4240), and extensions
to the NETANN protocol.
With more complex media requirements, such as media switching, a full
featured media server language is required to provide control over the media
channel. MSML support has been introduced to allow for such complex media
interactions. The use of MSML allows the Media Control Platform to provide
advanced media services to other Genesys components. For example, Call
Progress Detection services can be provided to Genesys offerings such as the
Outbound Contact Solution, and for GVP itself.
The MSML functionality on the MCP is designed to operate with the
Supplementary Services Gateway, and the Outbound Contact Server through
the SIP Server.
The MSML interface offers the following functionality:
• Performing Call Progress Detection (CPD) on calls.
• Launching VoiceXML applications with the NGI after performing CPD
operations.
• Launching VoiceXML applications with the NGI without performing CPD
operations.
• Playing audio and video prompts.
• Recording, including dual-channel (audio) call recording.
• DTMF collection.
• Conferencing.
• Bridging calls without using a VoiceXML application.
For the complete Genesys Media Server 8.5 supported MSML specification,
see the Genesys Voice Platform 8.5 Deployment Guide, Appendix B.
User’s Guide 153

Chapter 7: Configuring the Media Control Platform Important Media Control Platform Configuration Options
Important Media Control Platform

might want to customize.
Environment > Applications > <Media Control Platform> > Options tab. For
the detailed steps to configure option settings, see Procedure: Viewing or
The configurable Media Control Platform parameters are in the following
• asr—Session Manager parameters determine specific configuration for
ASR behavior.
• calllog—Parameters determine call recording file management.
• conference—Parameters determine the default behavior of the Conference
application module, for NETANN-initiated conference calls.
• cpa—Parameters determine call progress analysis (cpa) type detection.
• email—Parameters enable you to configure e-mail address information for
maintained e-mail messages.
• ems (see Table 6 on page 60)—Parameters determine Reporting behavior
for call detail records (CDRs) and metrics.
• fm—Parameters determine file fetching and caching behavior for NGI.
• log (see “Service Quality Analysis (SQA)” on page 61)—Parameters
determine behavior for Management Framework logging.
• mpc and mtmpc—Parameters determine the default media processing and
transport behavior of the Media Processing Component (MPC), or Media
Server.
• mtinternal—Parameters determine the behavior of the Internal Media
Transport application module, which is responsible for managing internal
media transmission between the Media Server and the ASR and TTS
speech engines. This internal data transmission uses RTP.
• msml—Parameters determine Media Server Markup Language (MSML)
functionality.
• Netann—Parameters determine default behavior for the NETANN Prompt
Announcement application module.
• remdial—Parameters determine remote dialer behavior.
• sessmgr—Parameters determine call control and platform-level behavior of
the Call Manager API (CMAPI) application modules that are loaded at
startup.

Note: Genesys recommends that you do not modify the default values,
unless you are an advanced user who needs to use special CMAPI
applications for your deployment.
• sip—Parameters integrate the Media Control Platform with the SIP Proxy
(the Resource Manager). These parameters determine the behavior of the
SIP Line Manager application module, and configure the supported
transport interfaces.
• snmp (see “Configuring SNMP” on page 68)—Parameters that determine
SNMP behavior.
• stack—Parameters relate to the MRCP stack and determine the way the
Media Control Platform manages connections to the external MRCP
server.
• tts—Parameters determine specific configuration for TTS behavior.
• vrm—Parameters determine the behavior of the MRCP Client. These
parameters relate to the Voice Resource Management (VRM), or Speech
Resource Management (SRM), module.
• vxmli—Parameters determine the behavior of the Next Generation
Interpreter (NGI).
Table 23 provides information about important Media Control Platform
parameters that are not described in Chapter 3 on page 37. Table 23 provides
parameter descriptions as well as the default parameter values that are
preconfigured in the Media Control Platform Application object.
Unless indicated otherwise, all changes take effect on restart.
For information about all the available configuration options for the Media
Control Platform, see the Genesys Voice Platform 8.5 Configuration Options
Reference.
For information about configuring multiple Media Control Platforms, see
“Deploying Multiple Media Control Platforms” in the Genesys Voice Platform
8.5 Deployment Guide.
User’s Guide 155

Table 23: Selected Media Control Platform Configuration Options
asr Section
ASR Load once per Specifies whether there will be one VRM session • True—Single session not
call for the entire call, or whether a separate VRM enabled.
session will be opened for each recognition • False—Single session
request. enabled.
A single session for the entire call Default value: False (only
(load_once_per_call = 1) means that each call one VRM session for the
may have multiple recognition sessions. entire call)
If this parameter is set not to enable a single
session for the entire call (load_once_per_call =
0), each VRM session is closed when the
recognition request completes, either successfully
or unsuccessfully (such as no match). Therefore,
each call may have multiple VRM sessions.
Having multiple VRM sessions in a call may
improve the efficiency of ASR server license
usage. However, be aware of the following
possible consequences:
• There will be longer delays on speech
barge-in.
• Some recognizer servers delete saved
utterance data after each VRM session. In
these cases, the VoiceXML application cannot
refer to the saved utterance file after the
recognition session.
ASR Engine Default Specifies the default ASR Engine resource when <resourcename>
using Request URI. [?<protocol>]
Changes take effect: Immediately For example,
SPEECHWORKS?MRCPv1
callmgr Section
FIPS Mode Enabled Enables FIPS mode in MCP. True

False
start/restart

Table 23: Selected Media Control Platform Configuration Options (Continued)
conference Section
Conference Specifies the maximum number of participants Any integer greater than or
Participant Limit allowed for a conference that is initiated by a equal to 0.
conferencing application. Default value: 0
If this option is set to 0, the number of
participants allowed is unlimited and depends on
the machine resource limits.
Conference Highest Specifies the number of highest inputs that will A range of integers.
Input be used for mixing output. Default value: 3
If this option is set to 0, all inputs are used.
Conference Video Specifies the type of video output for • single

Output Type conferences. • mixed
• If set to single, a single stream output is Default value: single
enabled, where the video stream from one
conference participant is sent to each
conference participant.
• If set to mixed, a video mixed output is
enabled, where the video streams from
multiple conference participants are combined
into one frame and sent to each participant.
cpa Section
The CPA Method Specifies the supported Outbound CPA method. • NONE—Disable CPA for
Used for Outbound Changes take effect: Immediately. outbound calls.
Calls • AUDIOCODES—CPA using
AudioCodes gateway.
• PSTNC—CPA using PSTN
Connector.
• NATIVE—CPA using
Native CPA.
Default value: NONE
User’s Guide 157

Supported Gateway A space-separated list of supported Gateway CPA • AMD —Answering

CPA Events events. If NONE is selected, CPA events will not be Machine Detection
sent to the application. • CPT —Call Progress
Changes take effect: Immediately. Tones Detection
• FAX —Fax Machine
Detection
• PVD—Positive Voice
Detection
• PTT —Push to Talk
Events
Default value: AMD CPT FAX
PVD
Outbound Calls with Specifies the initial CPA state when using Native • preconnect —Detection
Native CPA - Initial CPA. starts as soon as the call is
State Changes take effect: Immediately. initiated.
• postconnect —Detection
starts when the call is
connected.
Default value: preconnect
Outbound Call with Specifies whether the CPA algorithm ignores the • True —Ignore call
Native CPA - Ignore call connect event. connect event.
Call Connect Events Note: This parameter is valid only if Outbound • False —Use the call
Calls with Native CPA - Initial State is set to connect event.
preconnect. Default value: False
fm Section
HTTP Proxy Specifies the HTTP proxy to use for HTTP <host:port>
requests. Default value:
localhost:3128
HTTPS Proxy Specifies the HTTPS proxy to use for HTTPS <host:port>
requests. Default value: Empty

Outgoing Interface Specifies the network interface IP address that is Any string of characters.
used for outgoing HTTP requests. If this Default value: Empty
configuration option has an empty value, the
Media Control Platform automatically selects the
network interface it will use.
If the Squid HTTP proxy is used, it must be
configured to accept HTTP requests from the
interface that is specified. Otherwise, by default,
it accepts HTTP requests from the local host only.
No Cache URL Specifies that documents fetched from a URL Any comma delimited list of
Substring containing one of the substrings in this list should characters.
not be cached. Any substring listed in this comma Default value:
delimited list, will not be cached. cgi-bin,jsp,?
Maxage for Local Specifies, in seconds, how long to cache local file Any integer.
File for. If set to 0, local files will not be cached. Default value: 60
Maximum Cache Size The total maximum size, in bytes, of all cached Any integer.
files. Default value: 50,000,000
Maximum Cache Specifies the maximum size, in bytes, of each Any integer.
Entry Size cache entry. Default value: 500,000
Maximum Cache Specifies the maximum number of entries that Any integer.
Entry Count can be stored in cache. Default value: 1000
Maximum Specifies the maximum number of times to Any string of characters.

Redirections follow the Location header in the HTTP Default value: 5
response.
If set to 0, HTTP redirection is disabled.
Enable 100-Continue Specifies whether to enable the Expect: • 0—Disable

header 100-continue header in HTTP 1.1 requests. • 1—Enable
Default value: 0
SSL Certificate Specifies the certificate file name. Any string of characters.
User’s Guide 159

SSL Certificate Type Specifies the format of the certificate file name. • PEM—Privacy Enhanced
Mail
• DER—Distinguished
Encoding Rules
Default value: PEM
SSL Key Specifies the private key file name. Any string of characters.
SSL Key Type Specifies the format of the key file name. • PEM—Privacy Enhanced
Mail
Encoding Rules
Default value: PEM
SSL Key Password Specifies the password required in order to use Any string of characters.
the SSL Key. Default value: Empty
SSL Version Specifies the Secure Socket Layer version to use. • 0—Automatically detect
version
• 1—Force TLSv1
• 2—Force TLSv2
• 3—Force TLSv3
Default value: 0
Verify Peer Specifies whether to verify the peer’s certificate. • 0—Do not verify
Certificate Note: SSL CA Info or SSL CA Path must also be • 1—Verify
set in order for this parameter to take affect. Default value: 0
SSL CA Info Specifies the file name to use for verifying peer Any string of characters.
certificate. Default value: Empty
SSL CA Path Specifies the path to the directory holding the Any string of characters.
peer certificates. Default value: Empty
Note: This directory must be created using the
openssl c_rehash utility.
SSL Random File Specifies the random initial value used to Any string of characters.
Seed generate the first number of the SSL key. Default value: Empty

SSL Verify Host Specifies how the common name from the peer • 0—Do not verify
certificate is to be verified during the SSL • 1—Check existence only
handshake.
• 2—Make sure that it
matches provided host
name
Default value: 0
SSL Cipher List Specifies the list of ciphers to use for the SSL Any string of characters.
Connection. Default value: Empty
mpc Section
Append Rejected Specifies whether GVP will advertise all • 0—GVP will not
Codecs supported codecs when it generates a Session advertise all supported
Description Protocol (SDP) answer or SDP offer. codecs.
Even if codecs are rejected or not presented in the • 1—GVP will advertise all
caller’s SDP message, the platform will still supported codecs.
support receiving these codecs. The platform will Default value: 0
not send for the SDPs unless a payload is
presented by the caller.
Codecs A space-separated list of the codecs that • amr

correspond to the platform capabilities advertised • amr-wb
with SDP.
• g722
The list controls which codecs the Media Control
• g726
Platform offers to the remote party, for media
sent from the remote party to GVP. • g729
Changes take effect: Immediately. • gsm
• h263
• h263-1998
• h264
• pcmu
• pcma
• telephone-event
• tfci
Default value: pcmu pcma
g726 gsm h263 h263-1998
h264 telephone-event
User’s Guide 161

Codec Preference Specifies whether remote or local preferences • l—Local preferences will
will be used to interpret the list of accepted be used.
codecs. • r—Remote preferences
• Local preferences means that the effective will be used.
accept list is the locally configured accept list, Default value: r
filtered to include only those capabilities also
offered by the remote entity.
• Remote preferences means that the effective
accept list is the list of formats offered by the
remote entity, filtered to include only those
entries also on the locally configured list.
<codec> maxptime If the MCP is offering the SDP, or answering the • 0

SDP where the offer does not have the maxptime, • 10
the maxptime attribute will be set according to
• 20
this configuration.
• 30
If this configuration does not exist, or is disabled,
the maxptime attribute will not be sent unless the • 40
SDP offer had the maxptime attribute. In the case • 60
where other codecs in the SDP also specify • 80
maxptime, the configuration of the codec listed
before this codec will take precedence. • 100
Note: 0 = Disabled
<codec > ptime Specifies the duration, in milliseconds, and • 0

arrival interval of one RTP packet. For example, • 10
if the AMR codec is sent at 20 ms ptime, then one
• 20
AMR RTP packet contains 20 ms duration of
audio data. Also, this 20 ms ptime packet must be • 30
sent every 20 ms in order to supply audio data • 40
continuously. • 60
If the remote SDP does not specify the ptime • 80
attribute, this option is used as the transmission
• 100
rate of this codec.
Note: 0 = Disabled

<codec > ptime If this option is disabled, the SDP ptime attribute
(continued) is not sent to the remote SDP unless the SDP
offer had the SDP ptime attribute.
Display name values for <codec> ptime:
• AMR
• AMR-WB
• G.722
• G.726-32
• G.729
• GSM 6.10
• G.711 A-law (PCMA)
• G.711 -law (PCMU)
• RFC2833 DTMF (Telephone-Event)
Notes: AMR, AMR-WB and GSM 06.10 do not

support the 10 and 30 millisecond durations.
G.726-32 does not support the 80 and 100
millisecond durations.
The packet interval does not always conform to
the ptime for the bridging, conferencing, or ASR
operations. However, packet size always
conforms to the ptime.
Maximum and Specifies the accuracy of the minimum and A positive integer.
Minimum Frequency maximum tone frequencies (in Hz) for CPA. The
of Segments options names are configured as follows:
mpc.cpa.tone<m>.segment<n>.f1min
mpc.cpa.tone<m>.segment<n>.f2min
mpc.cpa.tone<m>.segment<n>.f1max
mpc.cpa.tone<m>.segment<n>.f2max
Where <m> ranges from 1 to 10 for each tone, and
<n> ranges 1 to 3 for each segment.
Default Audio The default audio format for the Call Manager. • ALAW
Formats • ULAW
Default value: ULAW
User’s Guide 163

SRTP Mode The mode of operation with regard to Secure • none—No SRTP support.
Real-Time Transport Protocol (SRTP). The Media Control
For offer mode: Platform will ignore the
crypto attribute in SDP
• If the other side ignores SRTP, the platform offers.
will fall back to non-SRTP mode.
• accept_only—SRTP is
• If a previously negotiated m-line is used in a supported for SDP offers
reoffer or if the far end requests an offer, and sent to the Media Control
that m-line did not have SRTP negotiated, Platform, but the platform
SRTP will not be added. will not add SRTP to
• If the far end reoffers and adds SRTP to a m-lines in outgoing offers
previously negotiated m-line, SRTP will be that did not previously
negotiated. contain it.
• offer—SRTP is
supported for SDP offers
sent to the Media Control
Platform, and will be
included in all outgoing
SDP offers.
• offer_strict—The
Media Control Platform
accepts SRTP received in
the offer, and sends a
crypto line in its own
offer, but will fail if the
answer does not contain a
valid crypto line.
• offer_selectable—Two
media lines are offered for
each media type, one with
crypto, one without. If
both media lines are
accepted, all RTP is sent
and received through the
crypto line.
Default value: none

RTP Send Mode Specifies the output mode for outgoing RTP • continous—Audio
streams. silence is sent when there
Notes: is no data to send.
• vad—RTP transmissions
• Continuous mode applies only for G.711
stop when there is no data
mulaw, G.711 alaw, AMR, and AMR-WB
audio codecs. to send.
• Continuous mode does not apply in the Default value: vad

following scenarios:

When a bridge transfer is in progress

When RTP data is sent to ASR speech
engines.

When the RTP data contains video.
IP Type of Service for Specifies the IP differentiated services field Range: 0-255
RTP/RTCP (ToS) to set in all outgoing RTP/RTCP packets. Examples:
Notes: • 0—Disabled
• For Windows Server 2003, the ToS must be • 16—IPTOS LOWDELAY
enabled in the registry. See (0x10)
• 32—IPTOS PREC
• For Windows Server 2008/2012, the ToS PRIORITY (0x20)
configuration is not supported. It must be
configured at the OS level. You can define per • 64—IPTOS PREC
executable and per port, and what type of CRITICAL (0x40)
DiffServ bits to set on the outgoing packets • 184—DiffServ EF
using the QoS policy defined in the following (Expedited Forward
article. 0xB8)
https://technet.microsoft.com/en-us/library Default value: 0
/hh831689.aspx
Maximum Record Specifies the maximum file size, in bytes, An integer range of 0–
File Size reached before the recording is stopped. 4,000,000,000.
If this option is set to 0, disables this limit. Default value: 0
Note: The recorded file may exceed this limit by
a few hundred bytes depending on the codec and
container chosen.
User’s Guide 165

SDP Origin Name Specifies the origin to match in the SDP. If the <FQDN or IP
Map [n] origin specified by this parameter matches the Address>/[session name
SDP, the DTMF type and confgain specified by content]
DTMF Send Type [n] and Conference gain [n]. Default value: Empty
n= 0 to 9
DTMF Send Type [n] Specifies the DTMF type to use when SDP • SIPINFO
for SDP Origin Name Origin Name Map [n] matches the SDP of the • INBAND
Map[n] call.
Default value: INBAND
n=0 to 9
Conference Gain [n] Specifies the input gain percentage to apply for An integer range of 0–1000
for SDP Origin Name the SDP matching connection when joining a Default value: 100
Map [n] conference.
n= 0 to 9
RTP De-Jitter Delay Specifies the duration, in milliseconds, of buffer An integer range of 0–10000
time to allow for RTP packet inter-arrival Default value: 0
dejittering. This translates to an initial delay
before the packets are dispatched.
If set to 0, inter-arrival detector is disabled.
RTP De-Jitter Specifies the length of time, in milliseconds, that An integer range of 0–1000
Timeout the RTP packets are to wait for the missing RTP Default value: 200
packet. Once the timeout expires, the packets are
dispatched without the missing packet.
RTP/RSTP/RSTP Specifies the ports for MPC to use. A character string with
RTP Port Range Note: The Media Control Platform allocates local possible values of 1030 to
RTP port in a round-robin manner starting from 65535.
the lowest port specified, and starting from the Default value: 10000-65535
lowest port again when the highest port is
reached.
Local RTSP/RTP Specifies the where the RTSP interface is located. <IP Address>
Address Default value: $LocalIP$
RTP Audio Buffer Specifies the size of the buffer to be used for Any integer.
Size sending RTP audio data. Default value: Empty

RTP Video Buffer Specifies the size of the buffer to be used for Any integer.
Size sending RTP video data. Default value: Empty
Notes: The higher frame rates and resolutions
require larger values this parameter, but the
default value should be big enough for MCP to
play any frame rates and resolution.
For H263 or H264 video file play, SQCIF, QCIF,
and CIF resolution with 10, 15, and 30 frame
rates have been tested with the default
configuration.
Media Manager Specifies the size of the buffer to be used for Any integer.
Audio Buffer Size sending non-TTS audio data. Default value: 102400
Media Manager Specifies the size of the buffer to be used for Any integer.
Video Buffer Size sending non-TTS video data. Default value: Empty
Notes: The higher frame rates and resolutions
require larger values this parameter, but the
default value should be big enough for MCP to
play any frame rates and resolution.
For H263 or H264 video file play, SQCIF, QCIF,
and CIF resolution with 10, 15, and 30 frame
rates have been tested with the default
configuration.
Transcoders Specifies the list of transcoders that will be used Valid values:
to provide transcoding services. The G.726 • G.722
transcoder is loaded by default.
• G.726
If this option value is set to none, all transcoders
• G.729
are disabled.
• AMR
• GSM
• AMR-WB
• MP3
• H.263
• H.264
• None
Default value: None
User’s Guide 167

VoIP Metrics Specifies whether to collect the metrics defined in • 0—Disable

RFC 3611 for each audio session. • 1—Enable
These metrics are divided into local and remote. Default value: 0
For local metrics, MCP collects some by
exchanging RTCP messages between itself and
the remote party, and some are calculated locally
from ongoing activities.
For remote metrics, the remote party, if it
supports RFC 3611, sends the metrics to MCP
periodically. MCP records these metrics when it
receives an update.
msml Section
Beep Filename Specifies the filename of the beep that is sent $InstallationRoot$/
before the <join> operation. Default value:
Example: file://$InstallationRoot$
file://$InstallationRoot$/audio/ulaw/ /audio/ulaw/default_audio
default_audio/endofprompt.vox /endofprompt.vox
Beep File Time Limit Specifies the time limit, in milliseconds, for the An integer range of 1–10000.
in Join audible beep when played during a <join> Default value: 5000
element.
CPD default Beep Specifies the CPD beep timeout, in seconds, if An integer range of 0–60.
Timeout the <cpd> element is not used in the VoiceXML Default value: 30
application.
Note: Setting this parameter to 0 disables the
functionality.
CPD default Specifies the CPD post-connect timeout, in An integer range of 0–60.
Post-connect Timeout seconds, if the <cpd> element is not used in the Default value: 30
VoiceXML application.
functionality.
CPD default Specifies the CPD pre-connect timeout, in An integer range of 0–60.
Pre-connect Timeout seconds, if the <cpd> element is not used in the Default value: 30
functionality.

Root Directory for Specifies the path to the prompt media root A character string.
Play Media directory. Default value:
file://$InstallationRoot$
Root Directory for Specifies the path to the recording media root A character string.
Record Media directory. Default value:
Root Directory for Specifies the path to the CPD recording root A character string.
CPD Recording directory. Default value:
/Record
File Extension for Specifies the CPD recording file extension that A character string.
CPD Recording determines the MIME-type and extension to use. Default value: .wav
Default Final Silence Specifies the final silence duration, in seconds, in An integer range of 0–10000.
Timeout order to terminate the recording. Default value: 4
Changes take effect: Immediately
MSML INFO Specifies the content -types allowed in a SIP An alphanumeric string of
Allowed INFO messages for the MSML AppModule. space delimited characters.
Content-Types Only the defined content types are processed, Default value:
others are ignored. application/vnd.radisys.m
sml+xml
Default Audio File Specifies the default file extension of the audio A character string.
Extension for Play files used in play prompt or recording. Default value: .wav
Prompt and
Recording
Netann Section
Root Directory for Specifies the path to the prompt media root A character string.
Prompt Media directory. Default value:
$InstallationRoot$/
Root Directory for Specifies the path to the record media root A character string.
Recorded Media directory. Default value:
$InstallationRoot$/record
User’s Guide 169

Send DTMF-Relay Specifies whether to include prompt • Auto

SIP Info Messages announcement services in the SIP header when • True
receiving DTMF.
• False
Default value: Auto
Root Directory for Specifies the path to the recording media root A character string.
Record Media directory. Default value:
$InstallationRoot$/record
/
Maximum Allowed Specifies the maximum amount of silence, in Any integer.

Silence Time During seconds, allowed during a recording. Default value: 0
Recording If set to 0, silence detection is not used.
Maximum Recording Specifies the maximum time, in seconds, allowed Any integer.
Time to record. Default value: 0
If set to 0, the recording time is unlimited.
Default Repeat Times Specifies the default repeat times to be used for A character string.
for Play Netann Netann announcement playback. Default value: forever
Announcement Note: This parameter is not applicable to DTMF
Prompts prompts.
Conference Specifies the recording mode when recording is • mixed—The recorded file
Recording Mode enabled in a conference. format will be specified
by request with audio
from all participants
mixed into a single file.
• pcap—One pcap format
file will be created for
each participant.
Default value: mixed
List of H.263 Video Specifies, in a comma separated list, H.263 video A character string.
Formats formats that are used for selecting H.263 video Default value: QFIC=2
files to play.
H.263 video formats are:
SQCIF=1 to 6
QCIF=1 to 6
CIF=1 to 6
CIF4=1 to 6
CIF16=1 to 6

remdial Section
Remdial Port Specifies the port used for remote dialing. An integer in the range of
1025–65535.
Default value: 6999
Remdial Max Calls Specifies the maximum number of concurrent Any integer greater than
remdial calls. zero.
Default value: 500
Remdial Max Client Specifies the maximum number of remdial clients Any integer greater than
Sockets allowed to connect to the interface. zero.
Default value: 64
Remdial Telnet Mode Specifies the telnet Operating System mode. • Auto—Mode
automatically selected
based on the OS.
• RAW—Windows OS mode.
• Normal—Linux OS mode.
Default value: Auto
sip Section
(Note: For additional important options in this configuration section, see also “Configuring SIP
Communication and Routing” on page 38.)
Default Blind The default transfer method for SIP, for blind • HKF—Hookflash
Transfer transfers. • REFER—REFER-based
transfer
• BRIDGE—Bridge-based
transfer
• REFERJOIN—Consultative
REFER transfer
• MEDIAREDIRECT—Media
redirect transfer
• ATTCOURTESY—AT&T
In-band Courtesy transfer
• ATTCONSULT—AT&T
In-band Consult transfer
• ATTCONFERENCE—AT&T
In-band Conference
transfer
User’s Guide 171

Default Blind • ATTOOBCOURTESY—AT&T

Transfer (continued) Out-of-Band Courtesy
transfer
• ATTOOBCONSULT—AT&T
Out-of-Band Consult
transfer
• ATTOOBCONFERENCE—
AT&T Out-of-Band
Conference transfer
Default value: REFER
Default Bridge The default transfer method for SIP, for • BRIDGE—Bridge-based
Transfer bridge-type transfers. transfer
redirect transfer
Default value: BRIDGE
Default Consultation The default transfer method for SIP, for • HKF—Hookflash
Transfer consult-type transfers. • BRIDGE—Bridge-based
transfer
REFER transfer
redirect transfer
In-band Conference
transfer
Out-of-Band Consult
transfer
• ATTOOBCONFERENCE—
AT&T Out-of-Band
Conference transfer
Default value: REFERJOIN

Default Gateway The default gateway host and port that will be <Host name or IP
used for SIP calls (transfer, call, or remote dial) to address>:<SIP port>
a telephone, if the destination address does not Default value: Empty
specify a gateway.
If this parameter is not specified, telephony calls
that do not specify a gateway in the destination
address will fail.
Example:
If sip.defaultgw=pstn-gw.voiceplatform.
com:5060 and a SIP call is placed to telephone
number 123456789, the SIP Line Manager
translates the destination address to
sip:123456789@default-gw, and the call is
routed to port 5060 on host pstn-gw.
voiceplatform.com.
Default Host The default host and port that the Media Control <Host name or IP
Platform will use for SIP calls (transfer, call, or address>:<SIP port>
remote dial), if the destination address does not Default value: Empty
contain a host name or IP address.
If this parameter is not specified, calls that do not
specify a host in the destination address will fail.
Example:
If sip.defaulthost=voiceplatform.com:5060
and a SIP call is placed to address sip:1234@, the
destination address is translated to:
sip:[email protected]:5060
Defer Out Alerting Enables early media for an outbound call, by • 0—CallOutAlerting will
specifying whether the CallOutAlerting response not be deferred.
to the session manager will be deferred until the • 1—CallOutAlerting will
media session is initialized and registered. be deferred.
If enabled, the session manager can start Default value: 0
performing media operations on the channel as
soon as the session manager receives the
CallOutAlerting notification.
User’s Guide 173

DNIS Correlation ID The length of the correlation ID, within the A non-negative integer.
Length user-id portion of the DNIS. The correlation ID Default value: 0 (no
is the portion of the user-id that will be stripped, correlation ID)
in order to isolate the DNIS.
Note: In the special case where the correlation ID
is all of the user-id, the ampersand character
(@) will also be stripped away from the DNIS,
because @<hostname> does not make sense.
DNIS Correlation ID The offset that specifies where the correlation ID Any integer.
Offset starts, within the user-id portion of the DNIS. A negative value indicates
The correlation ID is the portion of the user-id that the offset is from the
that will be stripped, in order to isolate the DNIS. right.
Default value: 0 (no offset)
Enable Send/Receive Enables the sending and receiving of SIP INFO • True—VoiceXML
Events messages for VoiceXML application usage. applications are enabled
This parameter does not affect SIP INFO messages to send and receive SIP
INFO messages.
used for other purposes (for example, DTMF).
• False—VoiceXML
applications cannot send
and receive SIP INFO
messages.
Default value: True
Enable SDP answer Specifies whether to send an SDP answer in the • True—The MCP includes
in provisional reliable provisional response if the INVITE the SDP answer.
response contains an SDP offer. • False—The MCP does
Note: Applies only if Enable Reliable not include the SDP
Provisional Responses is set to Supported or answer.
Required, or if Send Alert is set to 2. Default value: True
Enable Reliable Specifies whether to allow the SIP stack to send • 0—Disabled
Provisional reliable 101-199 provisional responses. • 1—Supported
Responses If set to 1, the 100rel extension is included in the • 2—Required
header of the outbound INVITE request giving
Default value: 0
the remote end the option to send the reliable
provisional response.
If set to 2, the MCP includes the 100rel extension
in the Require header of the outbound INVITE
forcing the remote end, that supports PRACK, to
send the reliable provisional response.

HF Disconnect Type The timeout value, in milliseconds, to terminate a Any non-negative integer.
SIP hookflash transfer. Default value: 5000
• If sip.hftype=0 (wait for disconnection), the
transfer is treated as failed if a BYE is not
received from the remote end before this
timeout expires.
• If sip.hftype=1 (force disconnection), the
transfer is always treated as successful. If a
BYE is not received from the remote end before
this timeout expires, then a BYE will be sent
from the local end.
HF Prefix The SIP hookflash transfer dialing prefix. A string that contains one or
Examples: more of the following
characters: 0–9,! * none
• sip.hfprefix=none means the dial string is
exactly as specified in the transfer. Default value: !
• sip.hfprefix=! means dial a hookflash.
• sip.hfprefix=*8, means dial *8 followed by
two pause durations.
HF Stop Dial The digits to dial to stop a hookflash transfer. A string that contains one or
Dialing the digits specified in this parameter will more of the following
abort a multi-phase hookflash. The connection is characters: 0–9 !
switched back to the original caller. Default value: !
Hook Flash Transfer Specifies the type of hookflash transfer for SIP. • 0—Wait for
Type disconnection.
• 1—Force disconnection.
Default value: 0
User’s Guide 175

Customer Inbound The list of header names from incoming <SIP <Header1> [<Header2>...]
<SIP request> request> messages that will be exposed to the where <HeaderX> is:
Headers VoiceXML application, where <SIP request> is
• A header name—Each
one of:
specified header name
• BYE will be exposed.
• INFO • *—All header names will
• INVITE be exposed.
The names of the exposed headers appear in the • none—No header names
application in the following format: will be exposed. If any
sip.invite.<headername>=<value>
other value is specified
alongside none, none is
ignored.
Example: From To Via
Default values:
• For BYE requests: Reason
• For INFO and INVITE
requests: *
Custom Inbound The list of header names from incoming INVITE <Header1> [<Header2>...]
Invite Parameters requests whose parameters will be exposed to the where <HeaderX> is:
The exposed parameter values appear in the specified header name
application in the following format: will be exposed.
sip.invite.<headernam>.<paramname>=<value> • none—No header names
will be exposed. If any
alongside none, then none
is ignored.
Default value: RequestURI
INFO Request The content type of outgoing SIP INFO messages A string indicating the
Content-Type that correspond to VoiceXML application <log> content type.
events. Default value:
A VoiceXML application can trigger the sending application/text
of a SIP INFO message by using the <log> tag
with dest="callmgr". Call Manager will then
send a SIP INFO message to the remote end. The
content of the SIP INFO message is the content of
the <log> tag.

Local RTP Address The Media Control Platform IP address to <IP address>
advertise for Real-time Transport Protocol (RTP). Default value: Empty
With multicast or proxied systems, you may need (which causes the local IP
to specify what IP address to advertise in the SDP address to be determined
description for a session. By default, the IP automatically)
address of the local system is retrieved by
performing a standard gethostname(). However,
if your system is multi homed or behind a
firewall, use this parameter to control the IP
address that is advertised.
P-Asserted-Identity Specifies whether the P-Asserted-Identity • 0—Do not use

Header header is used as the ANI if it is included in the P-Asserted-Identity for
incoming SIP INVITE, and its value is exposed ANI
with the session.connection.remote.uri • 1—Use
session variable. If not, the From header is used. P-Asserted-Identity for
ANI
Default value: 1
P-Called-Party-ID Specifies whether the P-Called-Party-ID header • 0—Do no use

Header is used as the DNIS if it is included in the P-Called-Party-ID for
incoming SIP INVITE, and its value is exposed DNIS
with the session.connection.local.uri session • 1—Use
variable. If not, the To header is used. P-Called-Party-ID for
DNIS
Default value: 1
P-Alcatel-CSBU Specifies the P-Alcatel-CSBU header value of the A character string.

Header Value 200OK response to the initial incoming INVITE if Default value:
the request contains this header. If this parameter fb=notransfer;
is an empty string, no header is set.
dtmf_auto=on
User’s Guide 177

Custom Outbound The list of header names from outgoing <SIP <Header1> [<Header2>...]
<SIP request> request> messages that will be exposed to the where <HeaderX> is:
Headers VoiceXML application, for customization. <SIP
request> is one of:
• INFO will be exposed.
• INVITE • *—All header names will
• REFER be exposed.
The customized names of the exposed headers • none—No header names
appear in the application in the following format: will be exposed. If any
sip.invite.<headername>=<value>
ignored.
Example: From To Via
Default value: *
Custom Outbound The list of header names from outgoing <SIP <Header1> [<Header2>...]
<SIP request> Params request> messages whose parameters will be where <HeaderX> is:
exposed to the VoiceXML application, for
customization. <SIP request> is one of:
• INVITE will be exposed.
• REFER • none—No header names
The exposed parameter values appear in the will be exposed. If any
application in the following format: other value is specified
sip.invite.<headername>.<paramname>=
<value> ignored.
Default value: RequestURI
Route Set Specifies the route set for non-secure SIP Any string of characters.
outbound calls. If defined, this route set is Default value: Empty
inserted as the ROUTE header for all outgoing
calls and forces the MCP to send the SIP
messages through this defined route set. Each
element in the routeset must be separated by
commas. For example,
sip.routeset=<sip:p1.example.com;lr>,<sip:
p2.domain.com;lr>
Note: This parameter does not apply to SIP
REGISTER messages.

Secure Route Set Specifies the route set for secure SIP outbound Any string of characters.
calls. Secure SIP calls must specify the sips Default value: Empty
scheme or tls transport parameters. If defined,
this route set is inserted as the ROUTE header for
all outgoing calls and forces the MCP to send the
SIP messages through this defined route set. Each
element in the routeset configuration option
must be separated by commas. For example,
sip.securerouteset=<sips:p1.example.com;lr
>,<sips:p2.domain.com;lr>
REGISTER messages.
SIP Static Route List Specifies, in a pipe delimited list, the static route Any string of characters.
groups. Each route group contains a list, Default value: Empty
separated by commas, of IP addresses. Within the
route group, each IP address may substitute each
other as an alternate route destination if sending a
SIP request to one of the IP address that fails. For
example,
10.0.0.1,10.0.0.2|10.0.10.1,10.0.10.2
specifies two static route groups, and each group
specified two routes that are alternate to each
other.
Use Original Specifies how the Media Control Platform will • 0—The gateway specified
Gateway in Outbound determine which gateway to use for an outbound in sip.defaultgw or
Call call or transfer, if the destination address does not sip.defaulthost will be
contain a host name or IP address. used.
Example: • 1—The gateway of the
inbound call will be used.
If sip.outcalluseoriggw=1 and the inbound call
came from a gateway with host name 3000, the Default value: 1
call will be placed to one of the following:
• tel://3000
• sip:3000@—The ampersand character (@) is
required to delimit the user part from the host
part of the address.
Refer Transfer Hold Specifies whether to put the originating caller on • 0—Original caller will
hold (Invite hold) before the Media Control not be put on hold.
Platform sends the REFER message for a REFER or • 1—Original caller will be
REFERJOIN transfer. put on hold.
Default value: 1
User’s Guide 179

Refer Transfer Retry Specifies the action to take if the caller (or its • 0—Disabled
REFER on the Media Gateway) cannot handle the REFER • 1—Enabled
Outbound Leg request.
Default value: 0
If the Caller (or its Media Gateway) cannot
handle the REFER request, the transfer will fail.
When failed, MCP will send REFER with
Replaces to the Agent instead (hoping the Agent
can establish direct connection to the Caller,
when the Caller cannot do so).
Registration The settings for registering the Media Control <registration-server>

Platform with the SIP Registrar. <register-as>
<requested-expiry>
You can configure the system to register with one <username> <password>
or more SIP registration servers on the network. [<routeset>]
To specify more than one registration entry,
separate the entries with a pipe (|). where:
The Media Control Platform will attempt to • <registration-server>

is the host and port of the
register with all defined registration entries, and
Resource Manager or
will periodically reregister as required (in
other SIP registration
accordance with the <requested-expiry>
server.
parameter). The Media Control Platform will
de-register when it shuts down.
Example: • <register-as> is the SIP

ResourceManager.yourdomain.com:5064 identity of the Media
[email protected] 60 - -|proxy2.yourdomain.
Control Platform.
com:5064 [email protected] 60 user password • <requested-expiry> is
means that the Media Control Platform will the duration of
register with the Resource Manager as SIP user registration, in seconds.
[email protected], and with another SIP proxy as <username> is the user name
SIP user [email protected], with authentication when authentication is
user name (user), and password (password). required by the server. This
may or may not be the same
as <register-as>. A dash (–
) indicates that no user name
is needed. If username=–
(dash) and the server requests
authentication, Anonymous is
used.

Registration • <Password> is the

(continued) password associated with
the authentication user
name. To specify an
empty string, use a dash
(–).
• <Routeset> is a
comma-separated list of
the servers that the
REGISTER messages will
go through. If a route set
is not defined, the
REGISTER messages will
be sent directly to the
<registration-server>.
Send Alert The SIP response for alerting and intermediate • 0—No SIP response
provisional responses. • 1—Send 180 RINGING
Changes take effect: Immediately response
• 2—Send 183 Session
Progress response with
SDP information
Default value: 1
INFO Allowed A space-delimited list of the content types that <Content type1>[<Content
Content-Type are allowed to be passed up to the VoiceXML type2>...]
application level in a SIP INFO message. Any where <Content typeN> is:
content types that have not been defined will be
• An alphanumeric string—
ignored.
Defines the content type.
• An empty string—Allows
all content to be passed
upstream.
Default value:
application/text
User’s Guide 181

Transfer Methods A space-separated list of the supported transfer • HKF—Hookflash

methods for SIP. • REFER—REFER-based
transfer
REFER transfer
redirect transfer
• none—No transfer
methods for SIP
• ATTCOURTESY—AT&T
In-band Courtesy transfer
In-band Conference
transfer
• ATTOOBCOURTESY—AT&T
out-of-band courtesy
transfer
out-of-band consult
transfer
Transfer Methods • ATTOOBCONFERENCE—

AT&T out-of-band
consult transfer
Default value: REFER
REFERJOIN MEDIAREDIRECT
ATTCOURTESY ATTCONSULT
ATTCONFERENCE
ATTOOBCOURTESY
ATTOOBCONSULT
ATTOOBCONFERENCE

VoiceXML URL Specifies whether VoiceXML URLs in SIP • 0—VoiceXML URLs will
INVITE INVITE messages will be accepted, thereby not be accepted.
bypassing the normal method of selecting a • 1—VoiceXML URLs will
VoiceXML application on the basis of DNIS be accepted.
mapping.
Default value: 1
If vxmlinvite is enabled, the originator of a SIP
call can specify the initial VoiceXML URL that
will be fetched for the session. To implement this
functionality, the originator of the SIP call must
encode the Request-URI in the following special
form:
"sip:dialog.vxml.<URL>@host.com"
where the <URL> portion is encoded (for example,
%3A).
Warning Headers Specifies whether the Media Control Platform • 0—The Media Control
will send warning headers. Platform will send
Changes take effect: Immediately. warning headers only
when it receives an error
response.
• 1—The Media Control
Platform will always send
warning headers, if there
are any.
• 2—The Media Control
Platform will never send
warning headers.
Default value: 0
Transfer Copy A space-delimited list of the headers to be copied <Header1> [<Header2>...]

Headers from inbound call INVITE requests to outbound where <HeaderX> is:
call INVITE requests for the same VoiceXML
session (in other words, for bridged and Release
specified header will be
Link Transfer [RLT] calls).
copied.
The headers are re-scanned for the re-INVITE (the • *—All headers will be
outbound call INVITE request), so changes that copied, including
have been made to the values of the headers unknown headers.
during the inbound call leg are applied on any
outbound calls made within the call session. • none—No headers will be
copied.
Default value: *
User’s Guide 183

IP Type of Service for Specifies the IP differentiated services field Range: 0-255
Transport (ToS) to set in all outgoing SIP packets over the Examples:
SIP transport.
• 0—Disabled
Note: This configuration parameter is not valid
• 16—IPTOS LOWDELAY
on Windows 2008 and 2012 operating systems.
(0x10)
• 32—IPTOS PREC
PRIORITY (0x20)
• 64—IPTOS PREC
CRITICAL (0x40)
• 184—DiffServ EF
(Expedited Forward
0xB8)
Default value: 0m
Transport Instance 0 Specifies the transport layer for the SIP stack and A string.
the network interfaces that are used to process Default value: Empty
SIP requests.
This option uses the following format:
sip.transport.x = transport_name type:
ip:port [parameters]
Where:
• transport_name—Is any string.
• type—Is UDP, TCP, or TLS.
• ip—Is the IP address of the network interface
that accepts incoming SIP messages.
• port—Is the port number where the SIP stack
accepts incoming SIP messages.
• [parameters]—Defines any extra SIP
transport parameters. This is used for
LMSIP2.

Transport Instance 0 If ip is an IPv6 address, [] must be used.

(continued) To define a transport to listen to all IPv4
interfaces, use any or any4 for ip. To define a
transport to listen to all IPv6 interfaces, use any6
for ip.
For example:
cert=[cert path and filename]—The path and
the filename of the TLS certificate to be used.
Applicable to SIPS only and mandatory if SIPS is
used.
key=[key path and filename]—The path and
the filename of the TLS key to be used.
Applicable to SIPS only and mandatory if SIPS is
used.
type=[Type of secure transport]—The type of
secure transport to be used. Value can be TLSv1,
SSLv2, SSLv3, SSLv23. Default value is SSLv23.
Applicable to SIPS only and optional.
password=[password]—The password associated

with the certificate and key pair. Required only if
key file is password protected. Applicable to
SIPS only and optional.
cafile=[CA cert path and filename]—The
path and the filename of the certificate to be used
to verify the peer. The same certificate that is
specified in the cert=[cert path and filename]
parameter can be used as the value here if only
one certificate is preferred.
verifypeer=true—Turns on TLS mutual
authentication. Mandatory for TLS mutual
authentication.
verifydepth=[max depth for the certificate
chain verification]—Sets the maximum depth
for the certificate chain verification. The
recommended value is 1 for the default Genesys
certificate that is provided. Applicable to TLS
mutual authentication only.
User’s Guide 185

Transport Instance 0 The default transport is the smallest non-empty

(continued) ID. If all transport.x values are empty, the
UDP, TCP, and TLS transports are all enabled
and ports 5060, 5060 and 5061 respectively,
listen on any network interface. TLS transport
uses the certificate, x509_certificate.pem,
and key, x509_private_key.pem, in the config
directory and UDP is the default transport.
Transport Instance 1 See the description for Transport Instance 0. A string.

Transport Instance 2 See the description for Transport Instance 0. A string.

Preferred IP version Specifies the connection timer bucket depth. A numeric string.
to be used in SIP If this parameter is set to a higher value, the Default value: 3000
initial memory usage increases, but the allocation
of run-time memory for high loads is prevented,
thereby enhancing performance and stabilizing
memory at a lower mark. The default for this
section is set to the maximum value of 3000 for
performance reasons.
Local Transport IPv6 Specifies whether or not the • true

Address sip.transport.localaddress configuration • false
option contains an SRV domain name.
Default value: false
• If this option is set to true, the port part is not
automatically generated by the SIP stack.
• If this option is set to false, the port of the
outgoing transport is used together with the
host name that is specified by the
sip.transport.localaddress option.

sessmgr Section
Send SDP in INVITE Specifies whether to send the caller’s last SDP to • True
for Media Redirect the called party for Media Redirect calls. • False
For NGI applications: Default value: True
• If this option is set to true, if a call has
connectwhen specified as answered, MCP will
send the caller’s last SDP in the re-INVITE and
the ACK message.
• If this option is set to false, MCP will not
send the caller’s last SDP in the re-INVITE.
For GVPi applications:
• If this option is set to true, and the transfer is
a 2 leg transfer, if a call has connectwhen
specified as answered, MCP will send the
caller’s last SDP in the re-INVITE and the ACK
message.
Note: For backward compatibility with legacy

devices, which always require SDP in INVITE,
Media Control Platform in the default
configuration does not fully conform to RFC3264
when performing a Media Redirect Transfer. In
particular, MCP will send a SDP offer in INVITE
and will send an updated SDP in the ACK. If full
offer/answer behavior is desired, and legacy
devices are not involved, Genesys recommends
setting this parameter to False.
Accept Call Timeout Specifies the time, in milliseconds, to wait after Any integer.
an alert is issued when the application module Default value: 30000
does not accept the inbound call before
disconnecting it.
tts Section
TTS Engine Default Specifies the default TTS Engine resource when <resourcename>
using Request URI. [?<protocol>]
Changes take effect: Immediately Default value: Empty
For example, REASPEAK?MRCPv2.
User’s Guide 187

vrm Section
FIPS Enabled Specifies whether to enable FIPS mode in MRCP • true

Proxy. When FIPS mode is enabled, only FIPS • false
140-2 approved ciphers and algorithms can be
used in SSL connections.
Native DTMF Specifies the Maxage, in milliseconds, that a An integer value of -1

Grammar Maxage native DTMF recognizer uses to fetch an external indicates the server's maxage
grammar. value is used.
Default value: -1
Native DTMF Specifies the Maxstale, in milliseconds, that a An integer value of -1

Grammar Maxstale native DTMF recognizer uses to fetch an external indicates the server's
grammar. A value of -1 indicates to use the maxstale value is used.
server's maxstale value.</ Default value: -1
SRM Default The timeout interval, in milliseconds, for the An integer in the range of 1–
Response Timeout MRCP client to wait for a response from the 60000.
MRCP server. Default value: 10000
If no response is received within this timeout
period, the request is deemed to have failed.
SRM Ping Frequency The interval, in milliseconds, at which the MRCP An integer in the range of 1–
Client pings each MRCP server that has been 3000000.
provisioned. Default value: 30000
The MRCP DESCRIBE method is used as a ping
message.
SRM Ping Timeout The timeout interval, in milliseconds, for the An integer in the range of 1–
MRCP client to wait for a ping response from the 6000000.
MRCP server. Default value: 60000
If no response is received within this timeout
period, the MRCP server is considered to be
unavailable. The MRCP Client disconnects from
the server, and then periodically tries to
re-establish a connection, at a retry interval
specified in the client.ping.frequency
parameter.
Genesys recommends setting the
client.ping.timeout value to twice the value of
the client.ping.frequency parameter.

Universals Grammar The URI convention that the NGI uses to specify builtin:grammar/
URI the universals grammars. universals
Default value:
builtin:grammar/
universals
vxmli Section
Release ASR Engines Specifies that for successful transfers, without • True
on Transfer speech grammars loaded, the interpreter will • False
release all open ASR engines.
Default value: True
Strict Grammar Mode Specifies whether the NGI will follow the • True
VoiceXML specification strictly when handling • False
the grammar element.
The default value (false) means that the NGI
will ignore the mode attribute for an external
grammar.
Enable Real Time Enables real-time debugging for the platform. • True
Debugging • False
Initial Request The HTTP method to use for the initial request. • GET
Method • POST
Default value: GET
Maximum Subdialog Specifies the maximum number of dialogs that An integer range of 1–1000.
Depth are allowed in a VoiceXML session. The depth Default value: 50
increments when a subdialog is entered, and the
depth decrements when a subdialog is returned.
Maximum bytes of Specifies the maximum number of bytes that are An integer range of 0–2 GB.
total saved temp files allowed for the total saved temp files per session. Default value: 100 MB
per session If the limit is exceeded, saving the temp files is
disabled for the applicable session.
Maximum Size of Specifies the maximum size (in bytes) that is An integer range of 0–1 GB.
VXML Document allowed for a VoiceXML document. If the limit is Default value: 0
exceeded, the interpreter will generate a
error.badfetch event.
Note: Setting the parameter to 0 disables the
functionality.
User’s Guide 189

Chapter 7: Configuring the Media Control Platform Important MRCP Server Configuration Options
Maximum Size of Specifies the maximum size (in bytes) that is An integer range of 0–1 GB.
Script File allowed for a script file. If the limit is exceeded, Default value: 0
the interpreter will generate a error.badfetch
event.
functionality.
Maximum Size of Specifies the maximum size (in bytes) that is An integer range of 0–1 GB
XML/JSON data allowed for XML or JSON data. If the limit is Default value: 0
exceeded, the interpreter will generate a
error.badfetch event.
functionality.
Enable External Specifies whether the application can access • True

Messaging Within messaging. • False
VoiceXML Note: If set to False, executing a <send> or Default value: True
<receive> will result in an
error.unsupported.send event or
error.unsupported.receive event.
Transfer Allowed Specifies whether dialog-initiated transfers are • True

allowed. • False
Default value: True
Userdata Prefix The string that, when used as a prefix in a SIP Any string.
header, identifies userdata variables. Default value: X-Genesys-
Important MRCP Server Configuration

Options
This section describes important configuration options that you either must or
Environment > Applications > <MRCP Server> > Options tab. For the detailed
The configurable MRCP server options are in the provision configuration
section. Table 24 provides information about these options. Table 23 provides

preconfigured in the MRCPv1_ASR, MRCPv1_TTS, MRCPv2_ASR, and

MRCPv2_TTS Application objects.
All changes take effect on restart.
For information about all the available configuration options for the MRCP
servers, see the Genesys Voice Platform 8.5 Configuration Options Reference.
Table 24: Selected MRCP Server Configuration Options
ASR and TTS
New MRCP (For MRCPv2 only) Specifies whether the MRCP • True
Connection Per Client will create a new connection to the ASR or • False
Session TTS server for each MRCP session setup.
Default value: True
Vendor Name The name of the speech resource vendor. <vendor_name>

Speech Resource The URI to the speech resource. • For MRCPv1 ASR:
URI rtsp://<MRCP server
IP>:<port>/media/speec
hrecognizer
• For MRCPv1 TTS:
rtsp://<MRCP server
IP>:<port>/media/speec
hsynthesizer
• For MRCPv2:
sip:mresources@<MRCP
server IP>:<port>
ASR Only
ASR Resource Specifies whether or not the MCP reserves an • true (enabled)
Reservation ASR resource prior to accepting the call. This • false (disabled)
resource is available until the resource is explicitly
released, or until the end of the call. The call is
rejected if the resource is not successfully
reserved.
User’s Guide 191

Table 24: Selected MRCP Server Configuration Options (Continued)
Disable Hotword Specifies whether or not the platform treats • true—Recognition-based

Recognition recognition-based barge-in as speech-based barge-in will be treated as
barge-in. speech-based.
Set this parameter to true for all ASR servers that • false—
do not support recognition-based barge-in. Recognition-based
barge-in will not be
treated as speech-based.
HotKey Base Path The HTTP fetchable location for the hotkey /mcp/$AppName$/grammar/co
grammars. The value of this parameter is mmon/hotkey
concatenated with the IP address of the Media Default value: Empty
Control Platform to form a fetchable location for
hotkey grammars.
The <vendor name> in the path must be the same
as the vendor name that is specified in
vrm.client.resource.name on page 190.
HotKey Local Path The local path for the hotkey grammars on the $InstallationRoot$/
Media Control Platform. The MRCP Client uses grammar/<vendor name>/
the HotKeyBasePath to translates this address to hotkey
the appropriate URI, which is sent to the ASR Default value: Empty
servers.
Enable Silence Specifies whether to send silence audio during an • True—Send silence audio
Filling ASR recognition session pause period. to the MRCP server.
• False—Does not send
silence audio to the
MRCP server.
Default value: True
TTS Only
TTS Resource Specifies whether or not the MCP reserves an TTS • true (enabled)
Reservation resource prior to accepting the call. This resource • false (disabled)
is available until the resource is explicitly
released, or until the end of the call. The call is
rejected if the resource is not successfully
reserved.

Note: Media Control Platform supports various MRCP vendors, and

provides select vendor specific templates that have pre-populated
parameters for your convenience. Some of those parameters may not
have defaults listed, as the so-called default values are provided by
the templates and not by software (in the event a parameter is
manually deleted)
User’s Guide 193


Chapter
8 Configuring the MRCP

Proxy
The Genesys Voice Platform (GVP) Media Resource Control Protocol
(MRCP) Proxy component acts as proxy for all MRCPv1 traffic, residing
between the Media Control Platforms and the MRCPv1 resources.
This chapter provides information about configuring the MRCP Proxy in the
following sections:

Task Summary: Configuring the MRCP Proxy, page 195

Task Summary: Configuring the MRCP Proxy for HA, page 196
 Important MRCP Proxy Configuration Options, page 198
Task Summary: Configuring the MRCP

Proxy
Task Summary: Configuring the MRCP Proxy summarizes the configuration
tasks that are required to implement MRCP Proxy functionality in your GVP
deployment.
User’s Guide 195

Chapter 8: Configuring the MRCP Proxy Task Summary: Configuring the MRCP Proxy for HA
Task Summary: Configuring the MRCP Proxy
Complete the prerequisites. 1. Create the ASR and TTS speech resource Application objects
If you have not already done so, see the Procedure: Provisioning
ASR and TTS resources, on page 146.
2. In the Media Control Platform Application, create a server

connection to the MRCP Proxy.
To create server connections, see the procedure in Chapter 7 of
the Genesys Voice Platform 8.5 Deployment Guide.
Configure the MRCP Proxy 1. Configure the client-side connections.

Application See “Configuring Client-Side Connections” on page 68.
2. Create the server connections to:


The ASR and TTS speech resource access points that will be used
by this proxy.

The Reporting Server

The SNMP Master Agent (optional)
To create server connections, see the procedure in Chapter 7 of the
3. Configure the [vrmproxy] uri option with URI that the Media
Control Platform uses to contact the MRCP Proxy.
If the MRCP Proxy and Media Control Platform are installed on
the same host, retain the default value for this option. Otherwise,
configure the host part with the actual IP address of the MRCP
Proxy.
Task Summary: Configuring the MRCP

Proxy for HA
Task Summary: Configuring the MRCP Proxy for HA summarizes the
configuration tasks that are required to implement MRCP Proxy for High
Availability (HA).

Chapter 8: Configuring the MRCP Proxy Task Summary: Configuring the MRCP Proxy for HA
Task Summary: Configuring the MRCP Proxy for HA
Complete the prerequisites. 1. Ensure that the Solution Control Server (SCS)
Application is configured to support HA licenses:
For a description of how to create and configure the
license files, see the Framework 8.5 Deployment Guide
and the Framework 8.5 Management Layer User’s
Guide.
Note: To support HA mode, you must ensure that the
latest versions of Management Framework and LCA are
installed. In addition, the Solution Control Server (SCS)
must have an HA license. If the SCS is not licensed, it
cannot provide HA functionality.
2. Create the ASR and TTS speech resource Application

objects
If you have not already done so, see the Procedure:
Provisioning ASR and TTS resources, on page 146.
3. In the Media Control Platform Application, add a

connection to the primary MRCP Proxy.
To create server connections, see the procedure in
Chapter 7 of the Genesys Voice Platform 8.5
Deployment Guide.
Configure the primary MRCP Proxy 1. Create the server connections to:
Application 
The ASR and TTS speech resource access points that
will be used by this proxy.

The Reporting Server

The SNMP Master Agent (optional)
To create server connections, see the procedure in
Chapter 7 of the Genesys Voice Platform 8.5 Deployment
Guide.
2. Configure the [vrmproxy] uri option with URI that

points to the MRCP Proxy.
If the MRCP Proxy and Media Control Platform are
installed on the same host, retain the default value for
this option. Otherwise, configure the host part with
the actual IP address of the MRCP Proxy.
3. In the Server Info section, add the backup MRCP

Proxy in the Backup Server field.
User’s Guide 197

Chapter 8: Configuring the MRCP Proxy Important MRCP Proxy Configuration Options
Task Summary: Configuring the MRCP Proxy for HA (Continued)
Configure the backup MRCP Proxy Complete the same steps as you did for the primary MRCP
Application Proxy.
See “Configure the primary MRCP Proxy Application”,
(Steps 1 and 2 only) in this table.
Note: The connections must be the same for both the
primary and backup proxy.
Important MRCP Proxy Configuration

Options
Environment > Applications > <MRCP Proxy> > Options tab. For the detailed
The configurable MRCP Proxy parameters are in the following configuration
sections:
• vrmproxy—Parameters that contain connection information for MRCP
Proxy such as, IP address and port number, and application session timers.
• stack—Parameters that determine the connection timeouts, trace behavior,
and RTSP port ranges.
• ems—Parameters that determine Reporting behavior.
• log—Parameters that determine the behavior for Management Framework
logging.
• snmp—Parameters that determine the behavior of SNMP.
Table 25 provides parameter descriptions as well as the default parameter
values that are preconfigured in the MRCP Proxy Application object.
Unless indicated otherwise, all changes take effect immediately.
For a complete list of MRCP Proxy configuration options and their
descriptions, see the Genesys Voice Platform 8.5 Configuration Options
Reference.

Table 25: Selected MRCP Proxy Configuration Options
ems Section
MF Sink Metrics Specifies the metrics that are delivered to the MF A string of characters in the
Filter Sink. format of a comma-separated
An asterisk in the value (*) indicates that all list of values or ranges. A
metrics will be sent to the sink. Alternatively, metric value must be
5-8,50-55,70,71 indicates that metrics with IDs between 0 and 141 inclusive,
5,6,7,8,50,51,52,53,54,55,70 and 71 will be sent but values '*' and blank are
to the MF sink. also allowed.
Default value: *
MF Sink Log Filter Specifies how the log messages that are sent to A pipe-delimited range or
the MF sink are controlled. string of characters for log
The values between pipes can be in the format: levels, module IDs and
m-n,o,p (for example, 0-4, 5,6). The wildcard specifier IDs in the format:
character * (asterisk) can also be used to indicate levels|moduleIDs|specifie
all valid numbers. For example, *|*|* indicates rIDs (repeated if necessary).
that all log messages should be sent to the sink. Default value: *|*|*
Alternatively, 0,1|0-10|*|4|*|* indicates that
CRITICAL(0) and ERROR(1) level messages with
module IDs in the range 0-10 will be sent to the
sink; as well as all INFO(4) level messages.
Persistent DB File for Specifies the full path of the local database file A character string.
CDR Data that is used to locally persist data for CDRs. Default value:
cdrQueue_rm.db
start/restart
Persistent DB File for Specifies the full path of the local database file A character string.
OR Data that is used to locally persist data for Operational Default value:
Reporting. orsQueue_rm.db
start/restart
CDR Batch Size Specifies the number of CDR messages that can An integer between 1-5000
be queued up by the Reporting Client before they inclusive.
are sent to the Reporting Server.
Default value: 500
Larger batch sizes (for example, 50 records)
lessen bandwidth constraints, at the cost of
start/restart
making and sending CDR data at larger intervals.
User’s Guide 199

Table 25: Selected MRCP Proxy Configuration Options (Continued)
OR Batch Size Specifies the number of OR messages that can be An integer between 1-5000
queued up by the Reporting Client before they inclusive.
are sent to the Reporting Server.
Default value: 500
start/restart
OR Reporting Specifies the interval, in seconds, between the An integer between 1-299
Interval accumulation of operational reports that are inclusive.
submitted to the Reporting Server.
Default value: 60
start/restart
Maximum Records in Specifies the maximum number of data items to An integer greater or equal
the Persisted Local the local database for CDR reporting. to -1.
DB File for CDR Queuing occurs either when the Reporting Server Default value: -1
Data is unavailable, or when data is provided to the
client faster than the Reporting Server can
start/restart
consume it.
The default value -1 indicates an unlimited
number of records are allowed. A value of 0
indicates that no records are persisted locally and
data is discarded if the Reporting Server is
unavailable.
Maximum Records in Specifies the maximum number of data items to An integer greater or equal
the Persisted Local the local database for CDR reporting. to -1.
DB File for OR Data Queuing occurs either when the Reporting Server
Default value: -1
is unavailable, or when data is provided to the
client faster than the Reporting Server can
start/restart
consume it.
The default value -1 indicates an unlimited
number of records are allowed. A value of 0
indicates that no records are persisted locally and
data is discarded if the Reporting Server is
unavailable.

TLS Certificate for Specifies the file name of the TLS certificate in A string of characters.
Reporting Client PEM format. The certificate is required to make Default value:
the connection to the Reporting Server $InstallationRoot$/config
(ActiveMQ) over TLS. /MRCPPROXY_EMStoMFLogID.
txt
start/restart
log Section
Verbose Level Determines whether or not a log output is created. Select one of several log
If it is, this option specifies the minimum level of event levels.
log events that are generated. Default value: standard
Any one of the following log event levels can be
selected as the value for this option (starting with
the highest priority level): standard,
interaction, trace, debug, all, or none.
For a description of the log events levels, see
Table 8 on page 64.
Output for Level All Specifies the outputs to which an application A string of characters.
sends all log events. The log output types must be Default value:
separated by a comma when more than one ../logs/MRCPProxy
output is configured.
Log events are sent to the Standard output
(stdout).
Standard sends the log events of the Standard level. Default value:
Log events are sent to the Standard output ../logs/MRCPProxy
(stdout).
Interaction sends the log events of the Interaction level and Default value:
higher, which means, more than one output is ../logs/MRCPProxy
configured—standard and interaction levels.
(stdout).
User’s Guide 201

Trace sends the log events of the Trace level and higher, Default value:
which means, more than one output is ../logs/MRCPProxy
configured—standard, interaction, and trace
levels.
(stdout).
Debug sends the log events of the Debug level and Default value:
higher, which means, more than one output is ../logs/MRCPProxy
configured—standard, interaction, trace,
and debug levels.
(stdout).
Log Segmentation Specifies whether or not there is a segmentation A string of characters.

limit for a log file. If there is, this option sets the Default value: 10000
mode of measurement, along with the maximum
size.
If the current log segment exceeds the size set by
this option, the file is closed and a new one is
created.
for log segmentation, see Table 8 on page 64.
Log Expiration Determines whether or not log files expire. If A string of characters.
they do, this option value sets the measurement Default value: 20
for determining when they expire, along with the
maximum number of files (segments) or days
before the files are removed.
Keep Startup Log File Specifies whether or not a startup segment of the A string of characters.
log, containing the initial T-Server configuration, Default value: false
is kept. If it is, this option value can be set to true
or to a specific size. Changes take effect:
start/restart
If this option value is set to true, the size of the
initial segment will be equal to the size of the
regular log segment that is defined by the
segment option. If this option value is set to
false (segmentation is turned off), the value of
this option will be ignored.

Message File Specifies the file name of application-specific log A string of characters.
events. The name must be valid for the operating Default value: Empty
system on which the application is running. The
option value can also contain the absolute path to
the application-specific *.lms file. Otherwise, an
application looks for the file in its working
directory.
Log Messages Format Specifies the format of log record headers that an Select one of two option
application uses when writing logs in the log file. values—short or full.
Using compressed log record headers improves Default value: short
application performance and reduces the log file's
size.
For a complete description of each option value,
see Table 8 on page 64.
Time Generation for Specifies the system in which an application Select one of two option
Log Messages calculates the log record time when a log file is values—local or utc.
generated. The time is converted from the time in Default value: local
seconds since the Epoch (00:00:00 UTC, January
1, 1970).
For a complete description of each option value,
see Table 8 on page 64.
Time Format for Log Specifies how to represent the time when an Select one of three option
Messages application generates log records in a log file. values—time, locale, or
For a complete description of each option value, ISO8601.
see Table 8 on page 64. Default value: ISO8601
User’s Guide 203

Enable Printing Specifies whether the application attaches Select one of two option
Extended Attributes extended attributes, if any exist, to a log event values—true, or false.
that it sends to log output. Typically, log events Default value: false
of the Interaction log level and audit-related log
events contain extended attributes.
Note: When this option is set to true, audit
capabilities are enabled, but performance is
negatively affected.
Genesys recommends that you enable this option
for Solution Control Server (SCS) and
Configuration Server when audit tracking is used.
For other applications, see Genesys Combined
Log Events Help to find out whether an
application generates Interaction-level and
audit-related log events; If it does, enable the
option when testing new interaction scenarios
only.
Check Point Interval Specifies how often the application generates a An integer.
check point log event to divide the log into Default value: 1
sections of equal time.
By default, the application generates this log
event every hour. Setting the option to 0 prevents
the generation of check-point events.
Memory Snapshot Specifies the name of the file to which the A string of characters.
File Name application regularly prints a snapshot of the Default value: Empty
memory output, if configured to do so. The new
snapshot overwrites the previously written data.
If the application terminates abnormally, this file
contains the latest log messages.
Note: Memory output is not recommended for
processors with a CPU frequency lower than 600
MHz.
Memory Output Specifies the buffer size for log output to the A string of characters.
Buffer Size memory, if configured. Default value: Empty
This option value can be configured in kilobytes
(KB), minimum 128 KB, or megabytes (MB),
maximum 64 MB.

Folder for Temporary Specifies the folder, including the full path, in A string of characters.
Network Log Output which an application creates temporary files that Default value: Empty
File are related to network log output.
If the option value is changed while the
application is running, the change does not affect
the network output that is currently open.
Enable 6.x Specifies whether the application uses 6.x output Select one of two option
Compatibility Log logic. values—true or false.
Output Priority For a complete description of each of the option Default value: false
values, see Table 8 on page 64.
snmp Section
SNMP Task Timeout Specifies the maximum amount of time, in Any integer value greater
milliseconds, that SNMP waits for a new task. than zero (0).
Default value: 100
stack Section
MRCP Connection Specifies the connection timeout, in milliseconds, Any integer value.
Timeout for SRM MRCPv1 and MRCPv2 stack to Default value: 10000
establish a TCP connection to the server.
start/restart
Enable MRCP Stack Specifies whether or not to enable the STACK Boolean: True/False
Debug Trace DEBUG message. Default value: True
start/restart
RTSP Port Range for Specifies the port range of the RTSP stack that is A string of characters.
MRCPv1 Client used by the MRCPv1 client. Default value: 10000-11999
start/restart
vrmproxy Section
FIPS Mode Enabled Enables FIPS mode in MRCP Proxy. • True

• False
Default value: True
start/restart
User’s Guide 205

MRCP Proxy Contact Specifies the full Real-Time Streaming Protocol String of characters in RTSP
RTSP URI URI that is used by the MRCPv1 clients to URI format.
contact this proxy. Default value:
The MRCP Proxy listens for TCP connections at rtsp://$LocalIP$:11000/mr
the port that is specified by the URI. If port is not cpproxy
specified in the URI, default port 11000 is Changes take effect:
assumed. If the MRCP Proxy is deployed on a immediately
host separate from the Media Control Platform,
the default value must be changed to the IP of the
MRCP Proxy.
Error Recovery Time Specifies the timeout, in milliseconds, for a Any integer value.
for Speech Resource server to be put back in service after it encounters Default value: 10000
errors, such as timeouts or TCP connection
errors. Changes take effect:
immediately
Barge-In Timeout Specifies the timeout, in milliseconds, for the Any integer value.
BARGE-IN to occur. Default value: 10000
immediately
Session Clean Specifies the interval, determined by Any integer value.

Interval timeout.max_idle parameter, to clean idle Default value: 60000
sessions.
immediately
Close Session Specifies the timeout, in milliseconds, of Any integer value.

Timeout Close-Session requests. Default value: 10000
immediately
Control Timeout Specifies the timeout, in milliseconds, of CONTROL Any integer value.
messages. Default value: 10000
immediately
Define-Grammar Specifies the timeout, in milliseconds, of Any integer value.

Timeout DEFINE-GRAMMAR messages. Default value: 10000
immediately

Get-Params Timeout Specifies the timeout, in milliseconds, of Any integer value.

GET-PARAMS messages. Default value: 10000
immediately
Get-Result Timeout Specifies the timeout, in milliseconds, of Any integer value.

GET-RESULT messages. Default value: 10000
immediately
Get-Server-Info Specifies the timeout, in milliseconds, to get a Any integer value.

Timeout response to Get-Server-Info requests (ping). Default value: 10000
immediately
Session Max Idle Specifies the maximum session idle time. Any integer value.
Timeout Sessions that exceed this idle time are terminated. Default value: 180000
immediately
Open-Session Specifies the timeout, in milliseconds, of open Any integer value.

Timeout sessions. Default value: 10000
immediately
Pause Timeout Specifies the timeout, in milliseconds, of PAUSE Any integer value.
immediately
Recognition-Start- Specifies the timeout, in milliseconds, of Any integer value.

Timers Timeout RECOGNITION-START-TIMERS messages. Default value: 10000
immediately
Recognize Timeout Specifies the timeout, in milliseconds, of Any integer value.

RECOGNIzE messages. Default value: 10000
immediately
User’s Guide 207

TCP Re-connect Specifies the interval, in milliseconds, between Any integer value.
Interval connection attempts, when the TCP connection Default value: 10000
with the MRCP server is not yet established.
immediately
Resume Timeout Specifies the timeout, in milliseconds, of RESUME Any integer value.
immediately
Set-Params Timeout Specifies the timeout, in milliseconds, of Any integer value.

SET-PARAMS messages. Default value: 10000
immediately
Speak Timeout Specifies the timeout, in milliseconds, of SPEAK Any integer value.
immediately
Stop Timeout Specifies the timeout, in milliseconds, of STOP Any integer value.
Changes take immediately

Chapter
9 Configuring the Call Control

Platform
This chapter provides information about how to configure the Call Control
Platform and how to provision the device profiles in your Genesys Voice
Platform (GVP) deployment. It contains the following sections:

Task Summary: Configuring the Call Control Platform, page 209
 Important Call Control Platform Configuration Options, page 212
Task Summary: Configuring the Call

Control Platform
Task Summary: Configuring the Call Control Platform summarizes the
configuration steps and options to implement Call Control Platform
Task Summary: Configuring the Call Control Platform
Specify use of the sips: schema Set the platform-level configuration option
mediacontroller.sipsecure to true, to specify that all calls
initiated by CCP via the tags <createcall>, <dialogprepare>,
<dialogstart>, <createconference> and <redirect> are
initiated in the sips: schema.
User’s Guide 209

Chapter 9: Configuring the Call Control Platform Task Summary: Configuring the Call Control Platform
Task Summary: Configuring the Call Control Platform (Continued)
Control SIP Secure Mode using the Use the CCXML hints attribute in the tags listed above to
hints Attribute override the platform-level configuration. Examples:
To enable SIP Secure:
<var name="hints" expr="new Object()"/>
<assign name="hints.sipsecure" expr="'1'"/>
To disable SIP Secure:
<assign name="hints.sipsecure" expr="'0'"/>
…
<dialogstart src="'helloworld.vxml'"
connectionid="in_connectionid" dialogid="dialogid"
hints="hints"/>
Control SIP Secure Mode Using the Use the dest attribute in the <createcall> and <redirect> tags
dest Attribute to override the corresponding hints. When using dest for this
purpose requires that you also use the sips: schema. Examples:
<createcall dest="'sips:[email protected]:5071'"
hints="hints"/>
<redirect dest="'sips:[email protected]:5071'"
hints="hints"/>
Integrate the Call Control Platform Point the Call Control Platform to the Resource Manager as the
with the Resource Manager and SIP Proxy server and interim target of media service requests,
Media Control Platform. and define the properties for SIP communications. Key
configuration options are:
• mediacontroller.sipproxy
• mediacontroller.bridge_server (see page 218)
• sip.transport.x (see page 41)
• sip.routeset or sip.securerouteset (see page 41)
For additional, relevant configuration options and actions, see
“Configuring SIP Communication and Routing” on page 38 and
“Enabling Secure Communication” on page 42.

Chapter 9: Configuring the Call Control Platform Task Summary: Configuring the Call Control Platform
(Required only if you made TCP or Modify the CCXML applications so that the Request-URI for
TLS the preferred default transport any endpoints includes the transport=TCP or transport=TLS
protocol [see page 38]) parameter.
Ensure that the Request-URI header • Use CCXML hints in the <createcall>, <dialogprepare>,
in SIP requests specifies the required <dialogstart>, and <createconference> tags. For example:
transport protocol.
<assign name="hints.requesturi" expr="new Object()"/>
<assign name="hints.requesturi.transport"
expr="'tcp'"/>
<dialogstart src="'file:///C:\Program
Files\GCTI\gvp\VP Media Control Platform
8.1\MCP_80\helloaudio.vxml'" hints="hints"/>
Note: The W3C CCXML Specification, Draft 29, specifies that
the hints attribute in <createcall>, and <createconference> be
an ECMAScript object. There is no such specification for
<dialogprepare>, however, Genesys recommends that hints are
not passed as any other primitive data type for any of these three
hints attributes.
Ensure that the Call Control Platform Verify and, if necessary, modify the device profiles that have
can interact with all other SIP been provisioned. For more information, see “Configuring
devices in your deployment. Device Profiles” on page 484.
Enable PRACK support. Configure CCP for Reliable Provisional Responses, specifically:
• sip.prack.support
Configure the IP DiffServ (ToS). Set the SIP packet’s ToS using
[sip]transport.[n].tos
See “Configuring SIP Communication and Routing” on
page 38”.
User’s Guide 211

Chapter 9: Configuring the Call Control Platform Important Call Control Platform Configuration Options
Tune Call Control Platform Configure appropriate maximums and timeouts for your
performance. deployment. Consider the following options, in particular:
• ccxmli.max_num_documents (default is 6000)
• ccxmli.num_session_processing_threads (default is 5)
• ccxmli.max_num_sessions (default is 6000)
• ccxmli.max_conn_per_session (default is 100)
• ccxmli.max_dialog_per_session (default is 100)
• ccxmli.max_conf_per_session (default is 100)
See also “Configuring Session Timers and Timeouts” on page 76.

behavior and performance.
Customize Call Control Platform See “Configuring SNMP” on page 68 and Table 100 on
Important Call Control Platform

Environment > Applications > <Call Control Platform> > Options tab. For
the detailed steps to configure option settings, see Procedure: Viewing or
Except for some ems options, all changes to Call Control Platform options take
effect immediately.
The Call Control Platform configuration options are in the following
• ccpccxml—Parameters determine the behavior of the Call Control Platform
in relation to the CCXML applications (for example, whether transfers
through dialogs are allowed).
• ccxmli—Parameters determine the behavior of the CCXML Interpreter (for
example, the HTTP port and URL for the IOProc function; maximums for
the number of sessions, documents, and per-session conferences,
connections, dialogs, processing threads, and so on).


• fm—Parameters determine file fetching behavior.
• mediacontroller—Parameters integrate the Call Control Platform, through
the Resource Manager, with the Media Control Platform, which acts as a
bridge server for call transfers and conferences.
• session—Parameters determine the behavior of the Call Control Platform
during sessions (for example, whether unknown headers will be copied
into forwarded SIP messages).
• sip—Parameters integrate the Call Control Platform with the SIP Proxy
(the Resource Manager).
• snmp (see “Configuring SNMP” on page 68)—Parameters determine
SNMP behavior.
Table 26 provides information about important Call Control Platform
parameters that are not described in Chapter 3 on page 37. Table 26 provides
preconfigured in the Call Control Platform Application object.
For information about all the available configuration options for the Call
Reference.
Table 26: Selected Call Control Platform Configuration Options
ccpccxml Section
FIPS Mode Enabled Enables FIPS mode in CCP. True

False
start/restart
Default CCXML The URI for the default CCXML application. <URI path to file>
Default value:
\config\default.ccxml
User’s Guide 213

Table 26: Selected Call Control Platform Configuration Options (Continued)
Send SIP Progressing Specifies whether CCP is to send the 180 SIP • 0—The 180 response is
response with the <accept> tag for all incoming sent when the <send> tag
calls. is called.
• 1—The 180 response is
sent immediately after
sending the 100 Trying
message.
Default value: 0
ccxmli Section
BasicHTTP Receive Specifies whether or not a descriptive text will be • true

- Show Error Body returned in the response body when an HTTP • false
failure response is given for a request to the Basic
HTTP Event I/O Processor.
If this option is set to true, it is enabled and a
descriptive text will be returned.
BasicHTTP Receive Specifies the IPv4 address or host name on which String
- Host for IPv4 the basic HTTP event I/O processor will listen for Default value: Empty
network HTTP requests on IPv4 network interface.
If the value of this option is an empty string, the
system listens on all available IPv4 network
interfaces. If the host name is specified, the first
IPv4 address in the resolved list is used.
BasicHTTP Receive Specifies the IPv6 address or host name on which String
- Host for IPv6 the basic HTTP event I/O processor will listen for Default value: Empty
network HTTP requests on IPv6 network interface.
interfaces. If the host name is specified, the first
CreateSession The IPv4 address or host name on which the String

Receive Host for session creation event I/O processor will listen for Default value: Empty
IPv4 network HTTP requests on IPv4 network interface.
interface. If the host name is specified, the first

CreateSession Specifies the IPv6 address or host name on which String

Receive Host for the session creation event I/O processor will listen Default value: Empty
IPv6 network for HTTP requests on IPv6 network interface.
interface. If the host name is specified, the first
Preferred IP Version Specifies the preferred IP version that will be used • ipv4
Used in BasicHTTP in basic HTTP access URI • ipv6
Access URI session.ioprocessors["basichttp"].
Default value: ipv4
Preferred IP version Specifies the preferred IP version that will be used • ipv4
Used in in the create session access URI • ipv6
CreatesSession session.ioprocessors["createsession"].
Default value: ipv4
Access URI
Save CCXML Files Specifies whether fetch request, response, and • TRUE
Save Script Files data for each CCXML or ECMAScript file that is • FALSE
fetched and processed in a session will be saved to
Default value: FALSE
disk.
This feature is convenient for debugging CCXML
applications, particularly when CCXML pages are
dynamically generated during a session.
fm Section
HTTP Port Range Specifies the local port range that will be used for String
HTTP requests. If this parameter is not specified, Default value: Empty
the CCP allows the operating system choose the
local port.
HTTP Proxy Specifies the HTTP proxy to use for HTTP <host:port>
requests. Default value:
localhost:3128
HTTPS Proxy Specifies the HTTPS proxy to use for HTTPS <host:port>
requests. Default value: Empty
User’s Guide 215

Outgoing Interface Specifies the network interface IP address that is Any string of characters.
used for outgoing HTTP requests. If this Default value: Empty
configuration option has an empty value, the
Media Control Platform automatically selects the
network interface it will use.
If the Squid HTTP proxy is used, it must be
configured to accept HTTP requests from the
interface that is specified. Otherwise, by default, it
accepts HTTP requests from the local host only.
No Cache URL Specifies that documents fetched from a URL Any comma delimited list of
Substring containing one of the substrings in this list should characters.
not be cached. Any substring listed in this comma Default value:
delimited list, will not be cached. cgi-bin,jsp,?
Maxage for Local Specifies, in seconds, how long to cache local file Any integer.
File for. If set to 0, local files will not be cached. Default value: 60
Maximum Cache The total maximum size, in bytes, of all cached Any integer.
Size files. Default value: 50,000,000
Maximum Cache Specifies the maximum size, in bytes, of each Any integer.
Entry Size cache entry. Default value: 500,000
Maximum Cache Specifies the maximum number of entries that can Any integer.
Entry Count be stored in cache. Default value: 1000
Maximum Specifies the maximum number of times to follow Any integer.

Redirections the Location header in the HTTP response. Default value: 5
If set to 0, HTTP redirection is disabled.
Enable Cookie Specifies whether to enable HTTP cookie. • 0—Disable

• 1—Enable
Default value: 1
Enable 100-Continue Specifies whether to enable the Expect: • 0—Disable

header 100-continue header in HTTP 1.1 POST requests. • 1—Enable
Default value: 0
SSL Certificate Specifies the certificate file name. Any string of characters.

SSL Certificate Type Specifies the format of the certificate file name. • PEM—Privacy Enhanced
Mail
Encoding Rules
Default value: PEM
SSL Key Specifies the private key file name. Any string of characters.
SSL Key Type Specifies the format of the key file name. • PEM—Privacy Enhanced
Mail
Encoding Rules
Default value: PEM
SSL Key Password Specifies the password required in order to use the Any string of characters.
SSL Key. Default value: Empty
SSL Version Specifies the Secure Socket Layer version to use. • 0—Automatically detect
version
• 1—Force TLSv1
• 2—Force TLSv2
• 3—Force TLSv3
Default value: 0
Verify Peer Specifies whether to verify the peer’s certificate. • 0—Do not verify
Certificate Note: SSL CA Info or SSL CA Path must also be • 1—Verify
set in order for this parameter to take affect. Default value: 0
SSL CA Path Specifies the path to the directory holding the peer Any string of characters.
certificates. Default value: Empty
Note: This directory must be created using the
openssl c_rehash utility.
SSL Random File Specifies the random initial value used to generate Any string of characters.
Seed the first number of the SSL key. Default value: Empty
User’s Guide 217

SSL Verify Host Specifies how the common name from the peer • 0—Do not verify
certificate is to be verified during the SSL • 1—Check existence only
handshake.
• 2—Make sure that it
matches provided host
name
Default value: 0
SSL Cipher List Specifies the list of ciphers to use for the SSL Any string of characters.
Connection. Default value: Empty
mediacontroller Section
Address of bridge The Resource Manager IP address. The Call <IP address>
server Control Platform sends requests to the Resource Default value: Empty
Manager to find a bridging server to use when two
endpoints cannot be joined because of media
bridging limitations (implicit conference and
transcoding).
The bridge server must be capable of:
• Sending media to multiple endpoints.
• Sending and receiving from distinct endpoints.
• Performing transcoding.
Device Profile of The name of the device profile to use with the <Device profile name>
Bridge Server configured bridge server. Default value: Default
For information about configuring device profiles, Conference
see “Configuring Device Profiles” on page 484.

Full Audio Codec A space-separated list of the <media type> codecs <payload>|<codec>|<MIME-t
Full Video Codec that get set in the SDP in an initial offer when ype>|<rate>|<number of
there is no media bridge. In other words, the channels>
media line that will be used to create a connection Default values:
less SDP.
• Audio—
0|pcmu|audio/basic|800
0|1
8|pcma|audio/x-alaw-ba
sic|8000|1
9|g722|audio/g722|8000
|1
23|g726-16|audio/g726-
16|8000|1
22|g726-16|audio/g726-
16|8000|1
2|g726-32|audio/g726|8
000|1
125|g726-40|audio/g726
-40|8000|1
18|g729|audio/g729|800
0|1
3|gsm|audio/x-gsm|8000
|1
• |1105|AMR|audio/AMR|8000
|1
Full Audio Codec A space-separated list of the <media type> codecs 112|AMR-WB|audio/AMR-W
that get set in the SDP in an initial offer when B|16000|1
Full Video Codec
there is no media bridge. In other words, the 101|telephone-event|no
(continued)
media line that will be used to create a connection ne|8000|1
less SDP. (continued) • Video—
34|h263|video/H263|90000|
1
99|h263-1998|video/H263-1
998|90000|1
113|H264|video/H264|90000
|1
Inbound allowed The default allowed media types for an inbound • dynamic
Media call. All inbound calls will be limited to this set of • audio video
media types in terms of SDP exchange.
• audio
Note: If set to dynamic, the media type is
• video
determined from the capability SDP of the
inbound call. If capability SDP is not available, it Default value: dynamic
defaults to audio and video.
User’s Guide 219

SDP localhost Specifies the host part of the local host IPv4 String
address that is used in SDP. Default value: $LocalIP$
SDP localhost IPv6 Specifies the host part of the local host IPv6 String
address that is used in SDP. Default value: $LocalIPv6$
SIP Proxy The Resource Manager address-of-record (AOR). <Resource Manager IP

The Resource Manager is the SIP Proxy that the address>:<SIP port>
Call Control Platform uses for outbound SIP Default value: Empty
requests.
Unknown headers A space-separated list of the unknown headers A string specifying a list
allowed for a SIP that can be sent in an outgoing SIP message. header names.
Message Genesys recommends that you allow the Default value: Warning
following headers: Reason
• Reason
• Warning
Note: Specifying a wildcard (*) means that all
unknown headers are allowed; therefore, the
wildcard should be the only value in the field—
for example, sip.allowknownheaders = *.
sip Section
Enable Reliable Specifies whether to allow the SIP stack to send • 0—Disable
Provisional reliable provisional responses (100-199). • 1—Supported
Responses If set to 1, PRACK is supported, and the 100rel • 2—Required
extension is included in the Supported header of
Default value: 0
the outbound INVITE request.
If set to 2, PRACK is required, and the 100rel
extension is included in the Require header of the
outbound INVITE request.
Default IPv4 route Specifies the default IPv4 route for UDP. The Numeric
for UDP number denotes the transport that is defined in the Default value: Empty
sip.transport.x configuration option, where x
is the value of this parameter and is used when no
IPv4 UDP routes are found.

Default IPv6 route Specifies the default IPv6 route for UDP. The Numeric
for UDP number denotes the transport that is defined in the Default value: Empty
IPv6 UDP routes are found.
If this parameter is not set, the first IPv6 UDP
transport found in the sip.transport.x becomes
the default.
Default IPv4 route Specifies the default IPv4 route for TCP. The Numeric
for TCP number denotes the transport that is defined in the Default value: Empty
IPv4 TCP routes are found.
Default IPv6 route Specifies the default IPv6 route for TCP. The Numeric
for TCP number denotes the transport that is defined in the Default value: Empty
IPv6 TCP routes are found.
If this parameter is not set, the first IPv6 TCP
the default.
Default IPv4 route Specifies the default IPv4 route for TLS. The Numeric
for TLS number denotes the transport that is defined in the Default value: Empty
IPv4 TLS routes are found.
Default IPv6 route Specifies the default IPv6 route for TLS. The Numeric
for TLS number denotes the transport that is defined in the Default value: Empty
IPv6 TLS routes are found.
If this parameter is not set, the first IPv6 TLS
the default.
User’s Guide 221

Local Transport IPv4 Specified the sent-by field of the Via header and String
Address the hostport part of the Contact header in the Default value: Empty
outgoing SIP message will be set to this value if a
IPv4 transport is used. The value must be a host
name or domain name.
If this option value is left empty the outgoing
transport's actual IP and port is used for the Via
and Contact headers.
Note: If the domain name that is used in the SRV
record query is specified, the
sip.transport.localaddress.srv configuration
option must be set to true to prevent the port
part from being automatically generated by the
SIP stack.
Local Transport IPv6 Specifies that the sent-by field of the Via header String
Address and the hostport part of the Contact header in Default value: Empty
the outgoing SIP message will be set to this value
if a IPv6 transport is used. The value must be a
host name or domain name.
Note: If the domain name that is used in the SRV
record query is specified, the
sip.transport.localaddress.srv configuration
option must be set to true to prevent the port
part from being automatically generated by the
SIP stack.
Local Transport Specifies whether or not the • TRUE

Address Contains sip.transport.localaddress configuration • FALSE
SRV Domain Name option contains an SRV domain name.
Default value: FALSE
• If this option value is set to true, the port part
is not automatically generated by the SIP
stack.
• If this option value is set to false, the
outgoing transport's port is used, together with
the host name that is specified by the
sip.transport.localaddress configuration
option.

Preferred IP version Specifies the local TCP port range to be used for String
Used in SIP SIP transport. If this parameter is not specified, Default value: Empty
the CCP allows the operating system to choose
the local port.
Route Set Specifies the route set for non-secure SIP Any string of characters.
outbound calls. If defined, this route set is inserted Default value: Empty
as the ROUTE header for all outgoing calls and
forces the MCP to send the SIP messages through
this defined route set. Each element in the
routeset must be separated by commas. For
example,
sip.routeset=<sip:p1.example.com;lr>,<sip:p
2.domain.com;lr>
REGISTER messages.
groups. Each route group contains a list, separated Default value: Empty
by commas, of IP addresses. Within the route
group, each IP address may substitute each other
as an alternate route destination if sending a SIP
request to one of the IP address that fails. For
example,
10.0.0.1,10.0.0.2|10.0.10.1,10.0.10.2
other.
User’s Guide 223

IP Type of Service Specifies the IP differentiated services field (ToS) Range: 0-255
for RTP/RTCP to set in all outgoing RTP/RTCP packets. Examples:
Notes: • 0—Disabled
• For Windows Server 2003, the ToS must be • 16—IPTOS LOWDELAY
enabled in the registry. See (0x10)
• 32—IPTOS PREC
• For Windows Server 2008/2012, the ToS PRIORITY (0x20)
configuration is not supported. It must be
configured at the OS level. You can define per • 64—IPTOS PREC
executable and per port, and what type of CRITICAL (0x40)
DiffServ bits to set on the outgoing packets • 184—DiffServ EF
using the QoS policy defined in the following (Expedited Forward
article. 0xB8)
https://technet.microsoft.com/en-us/library/ Default value: 0
hh831689.aspx
For other SIP section parameters, see Table 4 on page 40.

Chapter
10 Configuring the
CTI Connector
The Genesys Voice Platform (GVP) provides two modes of CTI deployment
that provide access to Genesys Management Framework functionality—
Genesys CTI through IVR Server and Cisco CTI through Intelligent Contact
Management (ICM).
This chapter provides information about how to configure the CTI Connector
to function in each of the two deployment modes. It contains the following
sections:

Configuring the CTI Connector, page 225

Important CTI Connector Configuration Options, page 229
 Cisco ICM Messages and Data Formats, page 235

CTIC (Genesys) and Treatments, page 238

Multiple Trunk Group ID support for CTI Connector (ICM), page 240
 CTI Connector (ICM) and ECC Variables, page 241

CTIC (ICM) Parameter Notes, page 242
Configuring the CTI Connector

When you install the CTI Connector, you can select the CTI Framework
(Genesys or Cisco) that is appropriate for your environment. This section
includes task summaries for each type of CTI deployment
Task Summary: Configuring the CTI Connector for Genesys CTI summarizes
the configuration steps and options to implement CTI Connector functionality
in your Genesys CTI deployment.
User’s Guide 225

Chapter 10: Configuring the CTI Connector Configuring the CTI Connector
Task Summary: Configuring the CTI Connector for Genesys CTI
Integrate the CTI Connector with the Point the CTI Connector to the IVR Server. The key
IVR Server. configuration section is IVRServer_Sample (see Page 229).
For more information on integrating GVP with IVR Server, see
the Voice Platform Solution 8.x Integration Guide.
Enable CTI Transfers. See “Provisioning IVR Profiles for GVP” on page 103.
Ensure that the CTI Connector can Verify and, if necessary, modify the device profiles that have
interact with all other SIP devices in been provisioned. For more information, see “Configuring
your deployment. Device Profiles” on page 484.
Note: The error response that is forwarded by SIP Server to
GVP from the agent by must always be the same (603 Decline).
To ensure this happens, set the sip-busy-type configuration
option value to 2 on the Trunk to which the error response will
be sent.
Customize client side See “Configuring Client-Side Connections” on page 68.

Task Summary: Configuring the CTI Connector for Cisco CTI summarizes the
configuration steps and options to implement CTI Connector functionality in
your Cisco CTI deployment.
Task Summary: Configuring the CTI Connector for Cisco CTI
Integrate the CTI Connector with Cisco When you are installing the CTI Connector, and you select
ICM. the Cisco ICM, the Service Control Interface is initialized by
default.
1. If you want to use the Call Routing Interface (CRI), in
the CTI Connector Application, configure the
[icmc].ICMInterface option with the CRI value.
2. Configure the ICM Trunk Group ID:

In the CTI Connector Application, configure the
[icmc].TrunkGroupID option with the applicable value.

Task Summary: Configuring the CTI Connector for Cisco CTI (Continued)
Configure the listener ports for the Single Tenant Environments

VRU-PGs You can configure the TenantName configuration option for
a specific tenant, to enable the CTI Connector to handle
calls for that tenant only, and reject inbound calls from all
other tenants. (See Specify the Tenant name. in this table.)
The CTI Connector supports multiple VRU-PG connections
for a single tenant, and you can specify a comma-separated
list of listener ports, one for each VRU-PG.
• In the CTI Connector Application’s Tenant1
configuration section, change the value of the Ports
configuration option, as required.
A TrunkGroupID (TG ID) is a list of listener port numbers,
separated by commas, on which CTIConnector waits for a
TCP connection from the Cisco VRU-PG. Optionally, the
TG IDs supported by the PIMs can also be configured here.
The TG IDs can be listed for a particular PIM separated by
an ampersand (&).
For example: 6000:1&2,7000,8000:3&4
In this example, 6000 supports Trunk Group IDs 1 and 2,
7000 does not specify the TG IDs it supports and 8000
supports TG IDs 3 and 4.
Notes:
• Valid range for Trunk Group IDs is 0-65535.
• Same TG IDs should not be mentioned by more than one
PIM; Trunk Group IDs must be unique across all the
PIMs.
• The value mentioned as the default TG ID under the
ICMC section should not be specified by any of the PIMs
as a supported Trunk Group.
User’s Guide 227

Configure the listener ports for the Multi-tenant Environments

VRU-PGs (continued) The CTI Connector supports multi-tenant configurations and
multiple VRU-PG connections for each tenant. For
multi-tenant environments:
• Copy the Tenant1 section and rename it for each
additional tenant. For example, Tenant2, Tenant3,
Tenant4.
• Ensure each section has a TenantName option configured
with a valid Tenant's name in Configuration Server. If
there is no value in the TenantName field, the
CTI Connector processes requests from all tenants that
do not have a TenantName configured.
For example, if there are three tenants and Tenant 1 is
configured as follows:
[T1]
TenantName=T1
but Tenant 2 does not have a TenantName configured, it
will cater to requests received for Tenant 2 and Tenant 3.
• For each newly created tenant, in the CTI Connector
Application’s Tenantx configuration section, change
the value of the Ports configuration option, as required.
For example, 8000, 9000, 10000.
• TrunkGroupID is set to the value of the parameter
gvp.rm.resource-req.TrunkGroupID received as a RURI
param in the incoming sip INVITE message. If the
parameter is not received, then you can set the value
here. Default is 1.
Note: Ensure that there are no duplicate ports configured
across all tenants.
Specify the Tenant name. In the CTI Connector Application’s Tenant1 section, enter
the tenant name for value of the TenantName configuration
option.
• For each newly created tenant, in the CTI Connector
Application’s Tenantx configuration section, change the
value of the Ports configuration option, as required. For
example, 8000, 9000, 10000.

Chapter 10: Configuring the CTI Connector Important CTI Connector Configuration Options
Ensure that the CTI Connector can Verify and, if necessary, modify the device profiles that have
interact with all other SIP devices in your been provisioned. For more information, see “Configuring
deployment. Device Profiles” on page 484.
Note: The error response that is forwarded by SIP Server to
GVP from the agent by must always be the same (603
Decline). To ensure this happens, set the sip-busy-type
configuration option value to 2 on the Trunk to which the
error response will be sent.
Customize client side communication See “Configuring Client-Side Connections” on page 68.
ports.
Important CTI Connector Configuration

Options
Environment > Applications > <CTI Connector> > Options tab. For the
The configurable CTI Connector parameters are in the following configuration
sections:
• ctic—Parameters determine CTI Connector behavior.
• icmc—Parameters that enable CTI Connector/Intelligent Contact
Management (ICM) functionality.
• IServer_Sample—Parameters determine IVR Server information and
properties in the deployment.
• ivrsc—Parameters required for controlling CTI Connector integration
with IVR Server.
User’s Guide 229

• mediacontroller—Parameters required for controlling the B2BUA

framework that CTI Connector uses.
• sip—Parameters required for defining the SIP protocol level attributes for
the Media Controller embedded SIP Stack.
Table 27 provides information about important CTI Connector parameters that
are not described in Chapter 3 on page 37. Table 27 provides parameter
the CTI Connector Application object.
For information about all the available configuration options for the Media
Reference.
Table 27: Selected CTI Connector Configuration Options
ctic Section
Default DNIS Specifies the default DNIS if the IVR Server does Any string of characters.
not provide the DNIS. Default value: Empty
Note: This option is applicable only when IVR
Server is configured in behind-the-switch-mode
Disable CTI Specifies whether to collect CDRs for CTI • True

Connector’s CDR Connector. • False
Update
Fetch DNIS From Specifies whether the CTI Connector it to receive • True
IVR Server the DNIS from IVR Server. • False
Note: This option is applicable only when IVR Default value: False
Server is configured in behind-the-switch-mode.
FIPS Mode Enabled Enables FIPS mode in CTIC. True

False
start/restart

Table 27: Selected CTI Connector Configuration Options (Continued)
IVRPort Base Index Specifies the starting IVR port number. Each port Any integer value.
number increments by one after this is set. Default value: -1
If this parameter is set to -1, CTI Connector will
not generate an IVR Port, instead it will take the
port base on DNISIndicator.
Notes: Use this option when the port information
is unavailable. IVR Server uses this port number
to pass the DNIS information. For deployments
using multiple CTI Connectors, the port range
must be distinct.
This parameter is applicable when IVR Server is
deployed in-front-of-switch mode.
Max IVRPorts Specifies the maximum number of IVR ports that Any integer value.
CTI Connector uses. Default value: 2000
Notes: Use this option when the port information
is unavailable.
This parameter is applicable when IVR Server is
deployed in-front-of-switch mode.
CTI Framework Specifies which CTI framework to use for CTI • IVRServerClient—The
functionality. Genesys IVR Server
• CiscoICMClient—The
Cisco ICM
Default value:
IVRServerClient
icmc Section
Trunk Group ID The Trunk Group ID information that is sent to Any integer value.
the ICM for every call through the Voice Default value: 0
Resource Unit-Peripheral Gateway (VRU-PG) to
report ICM metrics.
ECC Variables CTI Connector registers the configured list of A string of characters.
ECC variable names with ICM through the initial Default value: Empty
REGISTER_VARIABLES message. The ECC variable
names must be separated by commas.
User’s Guide 231

ECC SessionID CTI Connector sends the SessionID to ICM eccSessionIdVarName

Variable Name through this variable. Example: userECCVar1 Default value:
The variable name must be specified in the ECC userSessionId
Variables list. If not, the SessionID will not be Takes effect at start/restart.
sent in the NEW_CALL message.
By default, the variable is set to userSessionId
and the SessionID is sent through userSessionId.
If it is empty, the SessionID is not sent in the
NEW_CALL message.
Use Translation Label This value indicates to CTIC whether the translation-routed-call
incoming call is translation-routed or normal. Default value: false
Set to true for Type 8 Network VRU Takes effect at start/restart.
deployment.
DNIS mapping This parameter value indicates which field from DNISIndicator
attribute from RUN_SCRIPT_REQ message should be used for Valid values: ScriptID, 1-10
RUN_SCRIPT_REQ fetching the DNIS value.
Default value:
message
None (blank)
Takes effect after restart
ICM Interface to Use Specifies the interface that is used by the CTI • Service Control
Connector to communicate with ICM. By default, Interface—0
it communicates with ICM by using the Service • Call Routing Interface
Control Interface (SCI). (CRI)/Event Data Feed
(EDF)—1
Default value: 0
ivrsc Section
Customer IVR Specifies the list of IVR Servers that A string of characters.
Servers List CTI Connector uses. Default value:
IServer_Sample;
Fetch Script ID from Specifies the user defined key value from Any integer value.
URS Genesys Framework. Default value: 0
Script ID Key Name Specifies the key name configured in URS that is A string of characters.
used in the UdataGet message for IVR Server Default value: Empty
Client.
Note: This parameter is applicable when IVR
Server is set in behind-the-switch mode.

IServer_Sample Section
IVR Client Name Specifies the IVR Group Name that is configured A string of characters.
in Genesys Administrator. Default value: Empty
IVR Server Host IP Specifies the host name of the IVR Server. <Host name or IP address>
Address Default value: Empty
IVR Server Specifies the gli_server_address port of the Any integer.

Communication Port IVR Server application as configured in Genesys Default value: Empty
Administrator.
mediacontroller Section
Default IP version in Specifies the default IP version that will be used • ipv4
SDP in the SDP message, and applied to the initiated • ipv6
SDP offer to the endpoint.
Default value: ipv4
Local IPv6 Address Specifies whether or not the sent-by field of the String.
for SDP Via header and the hostport part of the Contact Default value: Empty
header in the outgoing SIP message is set to this
value if a IPv6 transport is used. The value must
be a host or domain name.
If the domain name that is used in the SRV record
query is specified, the
sip.transport.localaddress.srv option must
be set to true to prevent the port part from
being automatically generated by the SIP stack.
SIP Proxy Specifies The address of SIP Proxy for outbound String.
SIP requests, in the following format: Default value:
10.10.30.205:5070 $LocalIP$:5080
sip Section
Contact Header User Specifies the Contact Header name generated by A string of characters.
Name the platform. Default value: CTIConnector
User’s Guide 233

groups. Each route group contains a list, Default value: Empty
separated by commas, of IP addresses. Within the
route group, each IP address may substitute each
other as an alternate route destination if sending a
SIP request to one of the IP address that fails. For
example,
10.0.0.1,10.0.0.2|10.0.10.1,10.0.10.2
other.
Tenant1 Section
Ports
Peripheral Gateway • Specifies a list of listener port numbers, Any string of characters.
Communication Port separated by commas on which the CTI Default value: 9000
Numbers Connector waits for TCP connections from
the Cisco VRU-PG. For example:
6000,7000,8000.
• TrunkGroupID is set to the value of the
parameter
gvp.rm.resource-req.TrunkGroupID received
as a RURI parameter in the incoming sip
INVITE message. If the parameter is not
received, then you can set the value here.
Default is 1.
Tenant Name Specifies the name of the tenant. Any string of characters.
Load Balancing Between IVR Servers

This feature enables the configuration between two or more IVR Servers to
support the same set of IVR Server Clients. The IVR Server Client uses a
round-robin methodology for distributing calls between IVR Servers. Load
balancing is supported for both In-Front-of-the-Switch and
Behind-the-Switch-modes.

Chapter 10: Configuring the CTI Connector Cisco ICM Messages and Data Formats
Cisco ICM Messages and Data Formats

ICM’s Voice Resource Unit (VRU) uses the GED-125 interface, which is
divided into two sections—the communications interface and applications
interface. The communications interface defines conventions and protocols
necessary to establish, maintain, and terminate data communications between
the ICM and the GVP. The applications interface defines messages that allow
GVP to exchange call processing information with the ICM.
Table 28 lists the ICM messages that are supported by the CTI Connector for
each interface and how they are used.
Table 28: CTI Connector-Supported ICM Messages
ICM messages
Connection Management
OPEN_REQ GVP <- ICM
OPEN_CONF GVP -> ICM
CLOSE_REQ GVP <- ICM
CLOSE_CONF GVP -> ICM
HEARTBEAT_REQ GVP <- ICM
HEARTBEAT_CONF GVP -> ICM
FAILURE_CONF GVP -> ICM
FAILURE_EVENT GVP -> ICM
SCI Initialization sequence
INIT_SERVICE_CTRL_REQ GVP <- ICM
INIT_SERVICE_CTRL_CONF GVP -> ICM
INIT_SERVICE_CTRL_DATA GVP -> ICM
INIT_SERVICE_CTRL_END GVP -> ICM
REGISTER_VARIABLES GVP -> ICM
EDF Initialization sequence
INIT_DATA_REQ GVP <- ICM
INIT_DATA_CONF GVP -> ICM
FAILURE_CONF GVP -> ICM
User’s Guide 235

Table 28: CTI Connector-Supported ICM Messages (Continued)
ICM messages
INIT_VRU_DATA_EVENT GVP -> ICM
INIT_DATA_END_EVENT GVP -> ICM
SCI messages
NEW_CALL GVP -> ICM
RUN_SCRIPT_REQ GVP <- ICM
RUN_SCRIPT_RESULT GVP -> ICM
CONNECT GVP <- ICM
RELEASE GVP <- ICM
CANCEL GVP <- ICM
DIALOGUE_FAILURE_CONF GVP -> ICM
DIALOGUE_FAILURE_EVENT GVP -> or ICM

<-
EVENT_REPORT GVP -> ICM
CONNECT_TO_RESOURCE GVP <- ICM
RESOURCE_CONNECTED GVP -> ICM
CRI messages
ROUTE_REQUEST_EVENT GVP -> ICM
ROUTE_SELECT GVP <- ICM
ROUTE_END_EVENT GVP -> ICM
ROUTE_END GVP <- ICM
EDF events
DELIVERED_EVENT GVP -> ICM
CALL_CLEARED_EVENT GVP -> ICM
Mid-call events using EDF
SET_CALL_VARIABLES GVP -> ICM

Interaction Data Formats

ICM can send or receive call-related data to or from the CTI Connector in ICM
messages, that is used to make routing decisions. For example, ICM can
determine which call control instruction to send to CTI Connector, based on
Caller Entered Digits (CED) information that GVP passes to ICM. Similarly,
ICM can send call data that VoiceXML applications use to play specific
instructions to the caller.
Three kinds of constructs are used to exchange this call data with the IVR:
CED, Call Variables and ECC variables. The data and variables information is
passed within GVP in the SIP messages that are exchanged between the CTI
Connector and the Media Control Platform. Depending on the call flow, the
following SIP messages are used to exchange this information:
• INVITE, REFER—Interaction data is passed in the custom SIP header in the
following format:
X-Genesys-ICM_CED: 109
X-Genesys-ICM_CallVar1=<xx>
X-Genesys-ICM_CallVar6=<yy>
X-Genesys-ICM_ECC_user<variablename>=<value>
• INFO, BYE—Interaction data in passed in the SIP message body. The
content type application/x-www-form-urlencoded;charset=utf-8 is passed
in the following format:
ICM_CED=1&ICM_CallVar1=<xx>&ICM_CallVar7=<yy>&ICM_ECC_user<variable
name>=<value>
Table 29 contains a description of the format and the method that is used to
exchange interaction data between the Media Control Platform and the CTI
Connector.
Table 29: ICM Constructs Format in SIP Messages
ICM Construct Naming format in SIP Description

messages
Caller Entered Digits ICM_CED ICM identifies the caller entered digit
field by a tag identifier and not by name.
The name is used between the VoiceXML
application, Media Control Platform, and
CTI Connector only to make it user
friendly and readable.
User’s Guide 237

Chapter 10: Configuring the CTI Connector CTIC (Genesys) and Treatments
Table 29: ICM Constructs Format in SIP Messages (Continued)
ICM Construct Naming format in SIP Description

messages
Call variables ICM_CallVar<N> ICM refers to call variables by their

numbered tags and not by the call variable
name. The name is used between
VoiceXML application, the Media
Control Platform, and CTI Connector
only to make call variables more user
friendly and readable.
ECC variables ICM_ECC_user<variable name> ECC variable names must begin with
user. To avoid name clash with other
ECC variables that are currently in use, as
a best practice, Genesys recommends that
you add the company name to the ECC
variable name. For example,
user<variable name>
The CTI Connector submits the
user<variable name> part only to ICM
as the ECC variable name.
CTIC (Genesys) and Treatments

This section describes how CTI Connector (Genesys) handles various
treatments and treatment types. For brevity, CTI Connector (Genesys) is
abbreviated to CTIC(G).
Invalid Treatment Types

CTIC(G) sends TreatStatus as NotStarted to IServer if:
• CTIC(G) receives an invalid treatment type.
• CTIC(G) receives a PlayAnnounce treatment request containing PROMPT, for
the following cases:
 if the PROMPT option DigitsNumber is present in the TreatCall request.

if the PROMPT option iUser_Ann_ID is present in the TreatCall request.

if the TreatCall request contains multiple prompts.
Note: CTIC(G) does not support PROMPT service for the Genesys Legacy
Interpreter, or for the PlayAnnounce&Collect treatment type.

Chapter 10: Configuring the CTI Connector CTIC (Genesys) and Treatments
Music Treatment
CTIC(G) sends a NETANN-style INVITE to MCP as follows:
INVITE sip:annc@<RM-IP-Addr>:5080;DURATION=10;play=http://172.24.129.55:8080
/Test/Resources/Prompts/m.vox;gvp.netann.reportvxml=true SIP/2.0
PlayAnnounce & PlayAnnounceAndDigits Treatments

Upon receiving PlayAnnounce and PlayAnnounceAndDigits, CTIC(G) creates an
INVITE message in NETANN format with the voicexml parameter set to the value
of the pre-configured CTIConnector parameter PlayAnnouncePath or
PlayAnnounceAndDigitsPath—the path to the pre-canned VXML applications
supplied by MCP for handling these treatments.
• If the CTIC(G) parameters are not set, then voicexml=SCRIPTURL uses the
SCRIPTURL provisioned in the IVR profile.
• The INVITE message for PlayAnnounce, when the PlayAnnouncePath
parameter is set, is created in this format:
INVITE sip:[email protected]:5080;LANGUAGE=English(US);MSGID=2;
MSGTXT=In%20the%20process%20of%20playing%app;ScriptID=PlayAnnounce;
voicexml=http://localhost/treatments/PlayAnn.vxml SIP/2.0
• The INVITE message for PlayAnnounce, when PlayAnnouncePath
parameter is not set, is created in this format:
voicexml=SCRIPTURL SIP/2.0
• The INVITE message for PlayAnnounceAndDigits, when the
PlayAnnounceAndDigitsPath parameter is set, is created in this format:
INVITE sip:[email protected]:5080;LANGUAGE=English(US);MAX_DIGITS=1;
MSGID=2;MSGTXT=In%20the%20process%20of%20playing%app%20and%20Digits;
ScriptID=PlayAnnounceAndDigits;
voicexml=http://localhost/treatments/PlayAnnDigits.vxml SIP/2.0
• The INVITE message for PlayAnnounceAndDigits, when the
PlayAnnounceAndDigitsPath parameter is not set, is created in this format:
INVITE sip:[email protected]:5080;LANGUAGE=English(US);MAX_DIGITS=1;
MSGID=2;MSGTXT=In%20the%20process%20of%20playing%app%20and%20Digits;
ScriptID=PlayAnnounceAndDigits;voicexml=SCRIPTURL SIP/2.0
CTIC(G) supports the PROMPT service for the PlayAnnounce treatment type.
CTIC(G) expects one of the following options to be present in the TreatCall
request:
• TEXT—The ASCII text to pronounce using text-to-speech technology (if
supported by the IP equipment).
The INVITE message for PlayAnnounce, when TEXT parameter is set, is
created in this format:
INVITE sip:[email protected]:5060;TEXT="<ASCII text to
pronounce>";ScriptId=PlayAnnounce;voicexml=file://../treatments/PlayAnn.vxml
SIP/2.0
User’s Guide 239

Chapter 10: Configuring the CTI Connector Multiple Trunk Group ID support for CTI Connector (ICM)
• IntegerID—ID of a message to play. CTIC uses NETANN "annc" to play

the provided prompt that is identified by "IDintegerID".
The INVITE message for PlayAnnounce, when the ID parameter is set, is
created in this format:
INVITE sip:[email protected]:5070;play=announcement/<IntegerID> SIP/2.0
If CTIC receives a PlayAnnounce treatment request from IServer containing
PROMPT as well as MSGTXT, then MSGTXT is used for the announcement and
PROMPT-related parameters are discarded.
• The INVITE message for PlayAnnounce in this case is created in this
format:
voicexml=SCRIPTURL SIP/2.0
VoiceXML Call Reporting

You can configure CTIC(G) to send an indication to Media Control Platform,
to report music or an announcement, as a VoiceXMLcall to the Reporting
Server, in the CDR.
To enable this behavior, add gvp.netann.reportvxml=true as an RURI parameter
in the outgoing SIP INVITE message to Resource Manager.
Multiple Trunk Group ID support for CTI

Connector (ICM)
In a type 2 or type 8 ICM deployment, the incoming call to ICM is assigned a
particular PIM before it is sent to the Voice Response Unit (GVP), and must be
routed to the appropriate CTI Connector instance that is connected to the
already-selected PIM.
Each Trunk Group ID (TGID) is allocated a specific set of trunks and only the
calls targeted for that particular trunk group will share these trunks. If a call
comes in without specifying any trunk group then such calls will continue to
use trunks from the common pool.
There must be an LRG configured for each CTI Connector with the capability
parameter defined as TrunkGroupID=TG1,TG2
...where TG1, Tg2 is a comma-separated list of TG IDs that are supported
by the CTIC application under this LRG.
The CTIC (ICM) instance must be configured with the correct corresponding
PIM port along with all the trunk groups supported by that PIM.
CTIC supports multiple TGIDs configuration through the Ports parameter in the
respective Tenant section. List the TGIDs can be listed for the PIM along with
TCP listener ports in this format:

Chapter 10: Configuring the CTI Connector CTI Connector (ICM) and ECC Variables
port#:[TGID&TGID];port#:[TGID&TGID];
In this example: 6000:1&2;7000;8000:3&4, the PIM listening on port 6000
supports TGIDs 1 and 2, the PIM listening on port 7000 has no TGID specified,
and the PIM on port 8000 supports TGIDs 3 and 4.
Set the maximum value of TGID with the configuration parameter
MaxTrunkGroupID (section ICMC). Default = 65535.
In ICM, each TGID is associated with one set of DNISes having a unique
prefix. Whenever a call is received by ICM for a particular TGID, then it picks
the DNIS from the corresponding DNIS set and places the call to GVP with the
selected DNIS.
So, when a call is received at SIP-Server, it will have only DNIS information
but not the actual TGID of the call. GVP must identify a TGID for an
incoming call for the provided DNIS. To extract the actual TGID, create a DN
of the type trunk on the SIP switch for each TGID.
The Trunk DN should be configured with the following parameters.
[TServer]prefix=”Unique DNIS Prefix”
[TServer]contact=”sip:<RM-IPAddr>:<RM Port>
[TServer]request-uri=”<uniqueDNISPrefix>@<RM-IPAddr>:<RM
Port>;gvp.rm.resource-req.TrunkGroupID=<TGID>
CTI Connector (ICM) and ECC Variables

CTI Connector (ICM) reads ECC variables via the parameter
[ICMC]eccvariablelist and registers them with ICM through the initial
REGISTER_VARIABLES message.
Configure ECC variable names with their tag values as a comma separated
string, in this format: variable_name:tag_value,variable_name:tag_value
Specifying Tag Values

Consider this example:
userECCVar1:6010,userECCVar2:6011,userECCVar3,userECCVar4
The first and second variables include a tag value and the third and fourth
variable do not.
• For each ECC variable, CTIC (ICM) generates a tag value if you don’t
include one.
• Generated tag values begin with the start tag value configured for the
parameter [ICMC]eccStartTagValue. Default: 5010. This value is
incremented by 1 for each subsequently generated tag value.
• If the variable eccStartTagValue is empty, CTIC (ICM) generates these tag
values:
for userECCVar3: 5010
User’s Guide 241

Chapter 10: Configuring the CTI Connector CTIC (ICM) Parameter Notes

• If the variable eccStartTagValue is 6051, CTIC (ICM) generates these tag
values:
Specifying the ECC Variable for the Session ID

Beginning with GVP release 8.1.6, you can specify the ECC variable that
CTIC (ICM) uses to pass the Session ID to ICM. Configure the parameter
[ICMC]eccSessionIdVarName.
Specifying this particular ECC Variable is optional. Certain conditions apply to
specifying and to not specifying it:
If You Specify
The variable name that you specify in the parameter eccSessionIdVarName
must also be specified in the ECC Variables list. If it is not, or if
eccSessionIdVarName is specified but empty or null, then the Session ID is not
sent in the NEW_CALL message.
If You Do Not Specify

If you omit the parameter eccSessionIdVarName, then the Session ID is sent
through userSessionId. This method is backward compatible with pre-8.1.6
releases. The default tag value for userSessionId is 5000.
CTIC (ICM) Parameter Notes

TrunkGroupID
TrunkGroupID is set to the value of the gvp.rm.resource-req.TrunkGroupID,
received as a RURI param in the incoming sip INVITE message.
If gvp.rm.resource-req.TrunkGroupID is not received, then the value of
TrunkGroupID parameter configured in CTIC application is set in the field.
eccvariablelist and eccSessionIdVarname

These parameters are optional. The GVP 8.1.6 application template defines
their default values, for compatibility with previous versions:
eccvariablelist=userSessionId:5000
eccSessionIdVarName=userSessionId

You can add or modify the ECC variable names as needed. See the “CTI
Connector (ICM) and ECC Variables” on page 241.
User’s Guide 243


Chapter
11 Configuring the
Supplementary Services
Gateway
The Genesys Voice Platform (GVP) Supplementary Services Gateway (SSG)
component provides managed initiation of outbound sessions and queuing
functionality to accept a batch of outbound session creation requests. It also
provides result notifications for requests, including batch requests, that are
received from the Trigger Application to determine whether a particular
outbound call has succeeded or failed.
This chapter contains the following sections:

Task Summary: Configuring the Supplementary Services Gateway,
page 245
 Important Supplementary Services Gateway Configuration Options,
page 246

Call Progress Detection, page 251
Trigger Applications interact with the Supplementary Services Gateway
through HTTP. For more information on the HTTP Interface and the HTTP
XML schema, see Appendix I, “SSG HTTP Interface,” on page 515.
Task Summary: Configuring the

Supplementary Services Gateway
Task Summary: Configuring the Supplementary Services Gateway (SSG)
summarizes the configuration steps and options to implement SSG
User’s Guide 245

Chapter 11: Configuring the Supplementary Services Gateway Important Supplementary Services Gateway
Task Summary: Configuring the Supplementary Services Gateway
Configure the Supplementary The key configuration sections are HTTP, Tenant1 and SSG (see
Services Gateway to initiate page 246).
outbound calls.
Enable Reporting Server to poll the Install LCA and SNMP Master Agent on the Supplementary
Supplementary Services Gateway Services Gateway server.
data.

Install and configure security Create a security certificate on Windows and Linux to enable the
certificates. Transport Layer Security (TLS) connection between the
Supplementary Services Gateway and SIP Server.
For installation and configuration procedures, see Chapters 16-18
in the Genesys Security Deployment Guide.
Important Supplementary Services

Gateway Configuration Options
Environment > Applications > <Supplementary Services Gateway> > Options
tab. For the detailed steps to configure option settings, see Procedure: Viewing
or modifying GVP configuration parameters, on page 30.
The configurable SSG parameters are in the following configuration sections:
• Common—Parameters that determine whether or not IPv6 communication is
used between the SSG and SIP Server.
• fm—Parameters that determine the behavior of the HTTP Client.
• HTTP—Parameters that determine the embedded HTTP server behavior.
• SSG—Parameters that determine SSG behavior.
• Tenant1—Parameters that determine Tenant association to Trunk Group
and Routing Point for multi tenancy support.
• ems—Parameters that determine Reporting behavior for call detail records
(CDRs) and metric. (See Table 6 on page 60.)

• log—Parameters that determine behavior for Management Framework

logging. (See “Service Quality Analysis (SQA)” on page 61.)
Table 30 provides information about important Supplementary Services
Gateway parameters that are not described in Chapter 3 on page 37. Table 30
provides parameter descriptions as well as the default parameter values that are
preconfigured in the Supplementary Services Gateway Application object.
For information about all the available configuration options for the
Supplementary Services Gateway, see the Genesys Voice Platform 8.5
Configuration Options Reference.
Table 30: Selected SSG Configuration Options
Common Section
Enable IPv6 for SIP Specifies whether the IPv6 communication • 1 (true)
Server connection between SSG and SIP Server is enabled or • 0 (false)
disabled.
Default value: 0
fm Section
HTTP Proxy Specifies the HTTP Proxy that will be used by A string.
the Fetching Module. Default value: Empty
HTTP Section
HTTPS Certificate Specifies the name of the HTTPS Server Any string of characters.
File Name Certificate file. Default value:
$InstallationRoot$/config
/x509_certificate.pem
HTTPS Cert Key File Specifies the name of the HTTPS Server Any string of characters.
Certification Key file. Default value:
$InstallationRoot$/config
/x509_private_key.pem
HTTPS Cert Specifies the password to access the Certificate Any string of characters.
Password Key file. Default value: Empty
(optional)
Secure Protocol Specifies the name of the secure protocol and • SSLv23
Version version. • SSLv3
• SSLv2
• TSLv1.
Default value: SSLv23
User’s Guide 247

Table 30: Selected SSG Configuration Options (Continued)
Default HTTP Page Specifies the default HTTP page that SSG uses to Any string of characters.
service incoming outbound requests. Default value: SSG
HTTP Port Specifies the port that SSG uses to receive HTTP Any integer.
requests from trigger applications. Default value: 9800
HTTPS Port Specifies the port that SSG uses to receive Any integer.
HTTPS requests from trigger applications. Default value: 9801
SSG Section
FIPS Mode Enabled Enables FIPS mode in SSG. True

False
start/restart
Request Batch Size Specifies the number of requests that can be A string of characters.
fetched from the database into memory in a given Default value: TotalPorts
cycle.
If set to TotalPorts, SSG uses the GVP total port
capacity received from the SIP Server
EventResourceInfo as the batch limit.
If set to AvailPorts, SSG uses the current
available port capacity received from the SIP
Server EventResourceInfo as the batch limit.
Otherwise, you can configure any integer value
as a string for the batch limit.
Clean Interval Specifies the time (in seconds) that determines An integer in the range of
the frequency in which SSG removes expired or 30–900.
completed requests from the database. Default value: 180
Queue Low Specifies when to activate the next fetch cycle An integer in the range of
Watermark from the database. The algorithm uses this value 1—99.
to calculate the percent of the total batch limit. Default value: 25
When the in-memory queue falls below this
number, the next fetching cycle starts.

Initiated Call Retry Specifies whether the Initiated Call requests are • 0 (Do Not Retry Initiated
Flag retried when the SSG process starts back up. Requests)
• 1 (Retry Initiated
Requests)
• 2 (Purge New And
Initiated Requests)
• 3 (Purge All Requests
Without Notification)
Default value: 1
Max DB Connection Specifies the maximum number of database Any integer.

Pool Size connections SSG will use. Default value: 7
Min DB Connection Specifies the minimum number of database An integer.

Pool Size connections SSG will use. Default value: 3
Maximum Attempts Specifies the upper bound of the MaxAttempts Any integer.
Limit parameter that the trigger application is allowed Default value: 25
to use in the HTTP requests to SSG.
Time to Live Limit Specifies the upper bound of the TimeToLive Any integer.
parameter, in minutes, that the trigger application Default value: 1440
is allowed to use in the HTTP requests to SSG.
Application Slot Specifies the number of records that are allotted • Proportionate—Divide
Calculation for an application in each database fetch. the batch limit among the
applications in the same
ratio as their pending
requests.
• Equal—Divide the batch
limit among the
applications equally.
Default value:
Proportionate
Equal Priority Specifies the priority that is given to new and old • True—Give equal
Between Old and requests for applications. priority.
New Note: If this option is set to True, increased • False—Do not give equal
database fetches can result in performance priority.
degradation. Default value: False
User’s Guide 249

Port Load Factor Specifies the number of outbound calls that SSG An integer in the range of 1–
initiates at a time. The value that is specified is a 100.
percentage of the current GVP available port Default value: 100
capacity.
Next Retry Interval Specifies the retry interval, in seconds, for call if An integer in the range of 1–
temporary internal errors occur. 65536.
Default value: 10
Resource DN Specifies the frequency, in seconds, that SSG will Any integer.
Registration Failure attempt to re-register the Resource DN with SIP Default value: 120
Recovery Interval Server if the Resource DN registration is lost
during runtime, or fails at SSG start up.
SIPS Connection Specifies the frequency, in seconds, that SSG will Any integer.
Failure Recovery attempt to connect to SIP Server if the connection Default value: 120
Interval to SIP Server is lost during runtime, or fails at
start up.
Request Acceptance Specifies the timeout interval, in seconds, that Any integer.
Time-Out on SSG rejects requests from trigger applications if Default value: 900
Resource DN DN registration with SIP Server fails.
Registration Failure
Request Acceptance Specifies the timeout interval, in seconds, that Any integer.
Time-Out on SIPS SSG rejects requests from trigger applications if Default value: 900
Connection Failure the connection with SIP Server fails.
Max Calls/Sec to SIP Specifies the maximum number of calls per An integer in the range of 1–
Server second that the Supplementary Services Gateway 200.
initiates with SIP Server. Default value: 30
Tenant1 Section
Trunk Group DN Specifies the DN for outbound calls to use if Any string of characters.
using the Next Generation Interpreter. The Trunk Default value: Empty
Group DN (TGDN) option value is used as the
Tenant name.
Routing Point DN Specifies the DN for outbound calls to use if Any string of characters.
using the Legacy GVP Interpreter, or Default value: Empty
CTI Connector.
Name of Access Specifies which access group controls Digest Any string of characters.
Group Authentication enabling for the tenant. Default value: Empty

Chapter 11: Configuring the Supplementary Services Gateway Call Progress Detection
Note: If SSG is to support multiple tenants, copy the Tenant1 section to

Tenant2, Tenant3, etc and appropriately configured. See the
Framework 8.5 Genesys Administrator Help for information on how
to copy sections.
Call Progress Detection

There are three distinct call flows that SIP Server uses for making outbound
calls.
• CPD is not performed
• CPD is performed on the Media Gateway
• CPD is performed on the Media Server
The Supplementary Services Gateway does not know if Call Progress
Detection (CPD) is performed at the Media Gateway or at the Media Server.
CPD can be started with the first media packets received, or only after the call
is connected.
SIP Server selects a CPD provider (Media Gateway or Media Server), and sets
the appropriate CPD mode. If it is configured for both providers, the Media
Gateway takes precedence as the CPD provider. SIP Server is also responsible
for receiving the CPD result from the CPD provide, and pass it to the
Supplementary Services Gateway through the corresponding T-Event.
SIP Server sends the following CPD results to the Supplementary Services
Gateway:
• Pre-connect phase

GeneralError
 Busy
 NoAnswer

SitDetected
 SitInvalidnum

SitVacant

SitIntercept
 SitUnknown

SitNocircuit

SitReorder
• Post-connect phase

Unknown

AnsweringMachineDetected
 FaxDetected,

Voice
User’s Guide 251

The Supplementary Services Gateway applies any of the following CPD

algorithms during a call:
• no_progress_detection
• no_am_detection
• positive_am_detection
• voice_priority_detection (full_positive_am_detection)
• am_priority_detection (accurate_am_detection)
• telephony_preset
The Trigger Application passes CPD control parameters to the Supplementary
Services Gateway in the create request. All CPD attributes are optional, and if
they are not available in the CreateRequest, the SIP Server uses its default
configuration. Table 31 describes the various CPD attributes in the
CreateRequest:
Table 31: CPD Attributes
Attribute Description
record Specifies whether the CPD part of the call should be

recorded.
This is mapped to extension record in
TMakePredictiveCall API.
• True —Record CPD
• False —Do not record CPD
preconnect Specifies when to start CPD.

This value is mapped to the cpd-on-connect extension in
TMakePredictiveCall request.
• True—Start CPD when the first RTP packet is
received.
• False—Start CPD when the call is connected.
rnatimeout Specifies the timeout interval (in seconds) for the Ring No
Answer scenario.
This value is passed to SIP Server in the
TMakePredictiveCall request. SIP Server starts the timer
after receiving the 180 Ringing message from the external
party. If the timer expires, and the call is not connected,
SIP Server disconnects the call, and sends the
EventReleased TEvent with the CallState attribute set to
NoAnswer to the Supplementary Services Gateway.

Table 31: CPD Attributes (Continued)
postconnecttimeout Specifies the timeout interval (in seconds or milliseconds)

for the post connect scenario.
This value is passed to SIP Server in the
TMakePredictiveCall request. SIP Server starts the timer
starts when the outbound call is connected. If the timer
expires without a call result detected, SIP Server sends the
EventEstablsihed TEvent with the CallState attribute set
to Unknown to the Supplementary Services Gateway.
detect Specifies the action that SSG is to take with the outbound
call when CPD is detected.
• None (default)—Do not request CPD. Start the IVR as
soon as call is connected. This maps to
no_progress_detection.
• All—Start the IVR regardless of the detection result
(VOICE/MACHINE/FAX). This maps to
full_positive_am_detection.
• Voice—Start the IVR only if the detection result is
VOICE. The call is re-attempted if the detection result
is MACHINE. It is not re-attempted if the detection
result is FAX. This maps to
full_positive_am_detection.
• AM—Start the IVR only if the detection result is
MACHINE. The call is re-attempted if the detection
result is VOICE. It is not re-attempted if the detection
result is FAX. This maps to accurate_am_detection,
and requires the following setting on SIP Server
(am-detected: connect).
• FAX—Start the IVR only if the detection result is
FAX. The call is re-attempted if the detection result is
VOICE/MACHINE. This maps to no_am_detection,
and requires the following setting on SIP Server
(fax-detected: connect).
• Voice,AM,FAX —Can be combined by using comma
separation (for example, voice,am or am,fax or
voice,am,fax). If any of the comma-separated values
are detected, connect to IVR; otherwise, retry the call.
This maps to full_positive_am_detection.
User’s Guide 253


Chapter
12 Configuring the PSTN

Connector
The Genesys Voice Platform (GVP) PSTN Connector component provides
connectivity to traditional telephony environments, such as Public Switched
Telephone Networks (PSTN), and PBXs or ACDs. It uses Dialogic hardware
and software to interface with the PSTN network for processing Time Division
Multiplexed (TDM) calls.
This chapter contains the following sections:

Task Summary: Configuring the PSTN Connector, page 255

Important PSTN Connector Configuration Options, page 256
Task Summary: Configuring the PSTN

Connector
Task Summary: Configuring the PSTN Connector summarizes the
configuration steps and options to implement PSTN functionality in your GVP
deployment.
User’s Guide 255

Chapter 12: Configuring the PSTN Connector Important PSTN Connector Configuration Options
Task Summary: Configuring the PSTN Connector
Configure the PSTN Connector to The key configuration parameters are:

manage inbound and outbound calls. RouteType
Signaling Type
Channels
SIP Destination IP Address
SIP Destination Port Number
Supported Local Codec Type
See Table 32 for other important PSTN Connector options.
Configure Trunk DN. See the “Post-Installation Configuration of GVP Components”

chapter of the Genesys Voice Platform 8.5 Deployment Guide.
Configure reporting See “Configuring Reporting” on page 59.

Important PSTN Connector Configuration

Options
Environment > Applications > <PSTN Connector> > Options tab. For the
The configurable PSTN Connector parameters are in the following
• MediaManager—Parameters determine media behavior.
• GatewayManager—Parameters determine SIP behavior.
• DialogicManager—Parameters determine Dialogic and CPA behavior.
• DialogicManager_CPD—Parameters determine T-Server and CPD behavior.
• DialogicManager_Route1—Parameters determine the routing through
Dialogic behavior.

• log (see “Configuring Logging” on page 62)—Parameters determine

behavior for Management Framework logging.
Table 32 provides information about important supplementary PSTN
Connector parameters that are not described in Chapter 3 on page 37. Table 32
provides parameter descriptions as well as the default parameter values that are
preconfigured in the PSTN Connector Application object.
For information about all the available configuration options for tee PSTN
Connector, see the Genesys Voice Platform 8.5 Configuration Options
Reference.
Table 32: Selected PSTN Connector Configuration Options
MediaManagerSection
DTMF Payload Type Specifies the payload or encoding type of DTMF Any integer.
packets. Default value: 101
Supported Local Specifies the codec that is used by the TDM • Mulaw
Codec Type trunks. The RTP stream generated by PSTN • Alaw
Connector uses the same codec.
Default value: Alaw
GatewayManagerSection
PSTN Connector SIP Specifies the local SIP port for PSTN Connector Any integer.
Port to use for SIP communication. Default value: 5170
SIP Destination IP Specifies the end point to send SIP calls to when <Host name or IP address>
Address PSTN Connector receives TDM calls and Default value: Empty
translates them to SIP.
SIP Destination Port Specifies the SIP port number of the end point Any integer.
Number that is configured in the SIP Destination IP Default value: Empty
Address parameter.
Enable Session Timer Specifies whether to enable session timers. • True

• False
Default value: True
Session Timer Specifies the time interval, in seconds, for which An integer in the range of
Interval a call session is refreshed. If not set, the session 90–86400
will expire. Default value: 1800
User’s Guide 257

Table 32: Selected PSTN Connector Configuration Options (Continued)
DialogicManager Section
ATT Conference Specifies the duration, in milliseconds, for which Any integer.
Sleep Time Before PSTN Connector must wait before connecting the Default value: 5000
Answer agent, and the caller in an AT&T Conference
Transfer. Set this parameter when whisper to
agent is required, otherwise, set the value to 0.
CPA Failure Timeout Specifies the maximum time, in milliseconds, to Any integer.
wait for positive answering machine detection. Default value: 4000
CPA Max Inter-ring Specifies the maximum time, in milliseconds, to Any integer.
Timeout wait between consecutive ring-backs before Default value: 8000
disconnecting.
CPA Min Inter-ring Specifies the minimum ring duration, in Any integer.
Timeout milliseconds, for answering machine detection. Default value: 1900
CPA Option Specifies whether to choose custom enabled CPA • 0—Enable CPA Detection
parameters. • 1—Enable custom CPA
detection Springware
• 2—Enable custom CPA
detection DMV
Default value: 0
CPA PAMD Option Specifies the level of accurate answering machine • 0—Quick AM detection
detection. • 1—Full AM detection
• 2—Accurate AM
detection
Default value: 2
CPA Qualification Specifies which template PSTN Connector is to • 0—Qualification

Templates use for AM Detection. Template1
• 1—Qualification
Template2
Default value: 0
CPA Start Delay in Specifies the time, in milliseconds, to wait before Any integer.
MSec starting cadence, frequency, or positive voice Default value: 250
detection.

Disable Custom Specifies whether to delete custom tones before • True

Tones before CPA performing CPA. • False
Set this value to True for
Springware boards.
Minimum Download Specifies the minimum amount of data to be Any integer.

Size for Play buffered before starting the play on the TDM Default value: 32768
side.
Ringback Filename Specifies the audio file to use for playing Any string of characters.
ringback tone. The file format must be 8Khz Default value: m12.vox
PCM -law or A-law, and must contain a single
ring with the desired trailing silence.
If this parameter is not configured, the value
configured for AlawIndexFileName or
UlawIndexFileName is used based on the protocol
configured (A-law for E1, and -law for T1).
Default DNIS Value Specifies default DNIS number when the DNIS Any string of characters.
information is not available in behind-the-switch Default value: NoDNIS
configurations.
DialogicManager_CPD Section
IP Address of Specifies the IP address of the primary T-Server. <Host name or IP address>
Primary TServer Default value: Empty
Primary TServer Specifies the primary T-Server’s port number. Any integer.
Listening Port Default value: 0
IP Address of Backup Specifies the IP address of the backup T-Server. <Host name or IP address>
TServer Default value: Empty
Backup TServer Specifies the backup T-Server’s port number. Any integer.
Listening Port Default value: 0
Use TServer to Make Specifies whether to use T-Server to make • True

Calls outbound calls. • False
Note: If set to False, Dialogic will make Default value: False
outbound calls.
User’s Guide 259

FAX2 Tone as Specifies whether to accept FAX2 tones as • True

Answering Machine answering machine. • False
Offhook Delay Specifies the time, in milliseconds, to wait before Any integer.
going off hook. Default value: 100
If negative, go off-hook first, wait the specified
time, then dial.
If positive, dial first, wait the specified time, then
set the channel off-hook.
Note: This parameter is valid only if Use TServer
to Make Calls is set to True.
Postconnect Priority Specifies which application has priority if • TServer

conflicting CPD results are received. • Dialogic
Default value: TServer
Preconnect Priority Specifies which application has priority for • TServer

preconnect CPD events. • Dialogic
Default value: TServer
TServer Reconnect Specifies the time, in milliseconds, to wait before Any integer.
Timeout the reconnecting to the dialer. Default value: 20000
Wait for Offhook Specifies whether to wait for the off-hook • True
Confirmation confirmation event from T-Server before dialing. • False
Note: This parameter is valid only if Offhook Default value: False
Delay is set to a negative value.
DialogicManager_Route1 Section
Enable CPD Library Specifies whether CPD results are to be received • True
from T-Server. • False
If set to False, CPD results are received from Default value: False
Dialogic.
Route Description Specifies the description of the route configured. • Inbound Route
• Outbound Route
• Inbound Outbound Route
Default value: Inbound
Route

Dial Prefix Specifies the number to prepend to the number Any string of characters.
dialed. Default value: 1
Note: This parameter is used only when the
Network Type is PSTN.
Range of Directory Specifies, in a comma or dash separated list, the Any string of characters.
Numbers DN range for the route. Default value: Empty
For example, 101-110,115,120-130.
Note: This parameter is used only if TServer is
used for CPD (Enable CPD Library is True).
ISDN Numbering Specifies the encoding of the Calling/Called • 0x00—Unknown

Plan Party IE Numbering Plan in the outgoing setup. • 0x01—ISDN E.164
Used for outbound ISDN routes.
• 0x02—Telephony E.163
• 0x09—Private
Default value: 0x01
ISDN Numbering Specifies the encoding of the Calling/Called • 0x00—Unknown

Type Party IE Numbering Type in the outgoing setup. • 0x01—International
Used for outbound ISDN routes. Number
• 0x02—National Number
• 0x04—Subscriber
Number
Default value: 0x02
Max Digits to Dial Specifies the number of digits to dial. Any integer.
If Network Type is set to PSTN, then this value Default value: 7
must be 7,10, or 11.
If Network Type is set to Enterprise, then this
parameter can have any value.
If this value is set to 0, there is no maximum
number of digits to dial.
If this value is missing or invalid, the default is
used.
User’s Guide 261

Media Resource Specifies the board number used for Continuous Any integer.
Board to Use for CSP Speech Processing (CSP) when using JCT Default value: 0
boards.
If no value is specified, it defaults to the same
board as used for the network port.
For ISDN JCT boards, a different board must be
configured for CSP other than the network board.
Note: Routes with this parameter configured
must be on a single board.
Network Type Specifies the type of telephony network the route • 0—PSTN
is connected to. • 1—Enterprise
(PBX/ACD)
Default value: 0
New Call Specifies when to collect digits for inbound calls, • 0—Before Answer
Confirmation and when to start CPA for outbound calls. • 1—After Answer
For inbound calls, if set to After Answer, the call Default value: 0
is accepted and answered before collecting the
DNIS. If set to Before Answer, the ANI and
DNIS are collected before the called is answered.
For outbound calls, if set to After Answer, CPA is
started immediately after dialing. If set to Before
Answer, CPA is started after the call is connected.
Note: If you are using the groundstart protocol,
you must set New Call Confirmation to After
Answer.
Max Digits to Specifies the maximum number of digits (ANI + Any integer.
Receive in Overlap DNIS + delimiters) to receive in overlap receive Default value: 0
Receive Mode mode.
Enable ISDN Overlap Specifies whether to enable ISDN Overlap • 0—True

Receive Receive mode. • 1—False
Default value: 0
Channels Specifies the ports used for this route. <Card:PortRange,

Card:PortRange>

Signaling Type Specifies the Route Signaling type. • 0—T1-ISDN PRI

• 1—Analog
• 2—E1-ISDN PRI
• 3—T1-RobbedBit
• 4—E1-CAS
T1-RB ANI DNIS Specifies the character that separates ANI from Any single character.
Delimiter DNIS in the incoming call data. Default value: *
T1-RB ANI/DNIS Specifics which order to receive the ANI and • 0—No ANI/DNIS
Order DNIS. • 1—DNIS only
Note: This parameter is ignored if the signaling • 2—DNIS followed by
protocol is not T1-RobbedBit. ANI
• 3—ANI followed by
DNIS
Default value: 1
T1-RB Protocol File Specifies the Dialogic T1 configuration file to Any string of characters.
use. For example, use us_mf_loop_io for Default value: pdk_dmv
loopback testing.
Note: This parameter is mandatory for T1
robbed-bit signaling.
T1-RB Remove Specifies whether the ANI/DNIS delimiters are to • True

ANI/DNIS Delimiter be removed. • False
Note: This parameter is ignored if the signaling Default value: True
protocol is not T1 robbed-bit signaling.
User’s Guide 263

Two Channel Specifies the type of two channel transfer to use. • Empty
Transfer Type • NortelRLT
• TBCT
• ECTExplicit
• ECTExplicit_AUS
• ECTExplicit_NZ
• ECTExplicit_UK
• QSigPathReplace
Route Type Specifies the type of call direction to use. • Empty

• 0—Inbound
• 1—Outbound
• 2—In/Out
Note: To configure additional routes, copy the DialogicManager_Route1

section to DialogicManager_Route2, DialogicManager_Route3, etc and
configure appropriately. See the Framework 8.5 Genesys
Administrator Help for information on how to copy sections.

Chapter
13 Configuring the Fetching

Module and Squid Proxy
This chapter describes the requirements to configure the Fetching Module and
Third-Party Squid caching proxy in your Genesys Voice Platform (GVP)
deployment.
It contains the following sections:

Task Summary: Configuring the Fetching Module and Squid, page 265

Important Fetching Module Configuration Options, page 266
 Configuring the Squid Caching Proxy, page 268
Task Summary: Configuring the Fetching

Module and Squid
Task Summary: Configuring the Fetching Module and Squid summarizes the
configuration steps and options to configure the Fetching Module and Squid
Caching Proxy in your GVP deployment.
Notes: You do not need to change the default configuration for the Fetching
Module to work.
With GVP version 8.1.2 and higher, the Fetching Module
functionality has been included in the Media Control Platform, and
Squid is an optional component.
User’s Guide 265

Chapter 13: Configuring the Fetching Module and Squid Proxy Important Fetching Module Configuration Options
Task Summary: Configuring the Fetching Module and Squid
Modify the Squid caching proxy See “Configuring the Squid Caching Proxy” on page 268.
configuration, if required for the For more information about configuring Squid, which is an
following reasons: open-source product, see online sources.
• To configure for a second-level
proxy.
• You cannot configure the Web
Server to deliver Expires headers,
and you need to change the Squid
refresh-pattern rules.
• You are following the
recommended practice of denying
access to all ports except those
that you have identified as safe,
but the ports you are using for
HTTP or, if applicable, HTTPS
and SSL are not the ports that are
configured as safe ports and SSL
ports, respectively, in the default
Squid configuration file.
Schedule a task to rotate the Squid See the chapter about post-installation activities in the Genesys
Caching Proxy service logs. Voice Platform 8.5 Deployment Guide.
Important Fetching Module Configuration

Options
This section describes the key configuration options that you may want to
customize.
Environment > Applications > <Fetching Module> > Settings tab of each
Fetching Module Application in your deployment. For the detailed steps to
configure option settings, see Procedure: Viewing or modifying GVP
Except for some ems options, all changes to Fetching Module options take
effect on restart.
Note: If you restart the Fetching Module, you must also stop and then restart
the associated Media Control Platform or Call Control Platform.

Chapter 13: Configuring the Fetching Module and Squid Proxy Important Fetching Module Configuration Options
The Fetching Module configuration options are in the following configuration

sections:
behavior for Management Framework logging.
• iproxy—Parameters determine the behavior of the pwproxy process (the
Fetching Module as an HTTP or HTTPS proxy).
• snmp (see “Configuring SNMP” on page 68)—Parameters determine the
behavior of SNMP.
Table 33 provides information about some important Fetching Module
parameters in the iproxy section. Table 33 provides parameter descriptions as
well as the default parameter values that are preconfigured in the Fetching
Module Application object.
For information about all the available configuration options for the Fetching
Module, see the Genesys Voice Platform 8.5 Configuration Options Reference.
Table 33: Selected Fetching Module Configuration Options
iproxy Section
HTTP Proxy The IP address and port that the HTTP or HTTPS <Proxy IP address>:<port>
HTTPS Proxy proxy will use. Default value:
• If HTTP_proxy is disabled (empty), the pwproxy • HTTP proxy—
will not use an HTTP proxy. 127.0.0.1:3128
• If HTTPS_proxy is disabled (empty), the • HTTPS proxy—Empty
pwproxy will not use an HTTPS proxy.
No Cache URL A space-separated list of substrings that, if <Substring1>

Substring contained in a URL, will ensure that the page is [<Substring2>...]
not cached. Default value: cgi-bin
ssl_* Various options starting with ssl_, to configure

aspects of Secure Socket Layer (SSL)
functioning.
For more information about configuring secure
communications in GVP, see “Enabling Secure
Communication” on page 42.
User’s Guide 267

Chapter 13: Configuring the Fetching Module and Squid Proxy Configuring the Squid Caching Proxy
Configuring the Squid Caching Proxy

In general, the default Squid configuration file should be suitable for most
installations. However, there are three reasons why you might need to modify
the Squid configuration file:
• You need to configure for a second-level proxy.
• You cannot configure your Web Server to deliver Expires headers, and you
wish to change the Squid defaults for the expressions Squid tries to match
in SIP request-URI headers to control refresh behavior.
• You need to configure non-standard “safe” ports or SSL ports for HTTP
and SSL.
By default, the Squid configuration file:

Identifies the following as SSL ports: port 443 563.

Identifies the following as a “safe” port for HTTP: port 80.
 Denies requests to unknown ports (in other words, ports that are not
identified as “safe”).

Denies CONNECT to other than SSL ports.
The following procedure describes how to modify the Squid configuration file.
Procedure:
Modifying the Squid Configuration
Purpose: To modify the configuration file of the caching proxy to enable a

second-level proxy, to specify different refresh-pattern rules for matching
Request-URI expressions, or to enable non-standard “safe” and SSL ports.
Perform this procedure on each Media Control Platform and Call Control
Platform host in your deployment whose behavior you want to modify.
Prerequisites
• You have the required permissions to modify files in the Squid
configuration directory.
Start of procedure
1. Back up the original configuration file in case you need to restore it later.
2. Open the Squid configuration file in a text editor.
c:\squid\etc\squid.conf (Windows)
/usr/local/squid/etc/squid.conf (Linux)

Second-Level 3. To configure for a second-level proxy, add the following lines:

Proxy
cache_peer <parentcache.yourdomain.com> parent <port> 0 noquery
default
acl local-servers dstdomain <yourdomain.com>
acl all src 0.0.0.0/0.0.0.0
never_direct deny local-servers
never_direct allow all
Where:
• <parentcache.yourdomain.com> is the next proxy in the chain.
• <port> is the port number on which the parent cache is listening.
• <yourdomain.com> identifies the domains that should not go through the
parent proxy.
Refresh-Pattern 4. To modify the Squid refresh-pattern rules, add or reorder as many lines as
Rules you require, to specify the refresh-patterns in the order in which you want
Squid to consider them. Use the following format for each line:
refresh_pattern [-i] regex <min> <percent> <max> [<options>]
Where:
• <min> is the amount of time, in minutes, that an object without an
explicit expiry time should be considered fresh. The recommended
value is 0. Any non-negative values may cause dynamic applications
to be erroneously cached unless the application designer has taken the
appropriate actions.
• <percent> is a percentage of the age of the object (where age is the
time since last modification) that an object without an explicit expiry
time will be considered fresh.
• <max> is the upper limit, in minutes, for how long objects without an
explicit expiry time will be considered fresh.
• <options> are one or more of the following:
• override-expire—Enforces min age even if the server sent an
Expires: header. Doing this violates the HTTP standard. Enabling
this feature could make you liable for problems, which it causes.
• override-lastmod—Enforces min age even on objects that were
modified recently.
• reload-into-ims—Changes client no-cache or reload to
If-Modified-Since requests. Doing this violates the HTTP
standard. Enabling this feature could make you liable for problems,
which it causes.
• ignore-reload—Ignores a client no-cache or reload header. Doing
this violates the HTTP standard. Enabling this feature could make
you liable for problems, which it causes.
The default is:
refresh-pattern. 0 20% 4320
User’s Guide 269

Configure “safe” 5. In the ACCESS CONTROLS section:

and SSL ports a. Add or modify access control lines as required to ensure that the
following lines match the applicable port configurations in your
deployment:
• acl Safe_ports port <safe port> #http
• acl Safe_ports port <safe port> #https
• acl SSL_ports port <SSL port>
b. To deny requests to unknown ports (in other words, ports that have not
been identified as “safe”), verify that the following line has not been
commented out or deleted:
http_access deny !Safe_ports
c. To deny connections to other than SSL ports, verify that the following
line has not been commented out or deleted:
http_access deny CONNECT !SSL_ports
6. Save the file.
Execute the 7. Do one of the following to execute the update:
Update • Execute the following command to force a re-read of the configuration
file:
C:\squid\sbin\squid.exe -k reconfigure -n squidNT (windows)
/usr/local/squid/bin/squid - k reconfigure (linux)
• Restart Squid.
Restarting Squid will not affect the Fetching Module. However, if a fetch is
in progress, it may fail.
Note: Changes to the configuration file are not reflected in the running
configuration until you execute this command
End of procedure

Chapter
14 Configuring the Reporting

Server
This chapter provides information about how to configure the Genesys Voice
Platform (GVP) Reporting Server. It contains the following sections:

Task Summary: Configuring the Reporting Server, page 271
 Configuring Reporting, by Granularity, page 273

Configuring Database Retention Policies, page 274

Important Reporting Server Configuration Options, page 276
 Controlling Access to Reporting Services, page 283
Task Summary: Configuring the Reporting

Server
Task Summary: Configuring the Reporting Server summarizes the
configuration steps and options to set up the Reporting Server and to customize
EMS Reporting behavior in your GVP deployment.
Task Summary: Configuring the Reporting Server
Verify directory paths for: If necessary, modify settings for options in the following
• Java Message Service (JMS) for configuration sections:
CDR, OR summary, and call • messaging. In particular, verify the path to the directory that
events reporting. ActiveMQ uses for persistent queuing
• The Atomikos distributed (activemq.dataDirectory).
transactions processing engine.
User’s Guide 271

Chapter 14: Configuring the Reporting Server Task Summary: Configuring the Reporting Server
Task Summary: Configuring the Reporting Server (Continued)
Configure Reporting Server to If required, set the rs.nodb.enabled parameter in the

operate without a database. persistence section to true.
Configure the maximum size of If necessary, modify settings for the

reports for different levels of rs.query.limit.<granularity period> options in the
granularity. reporting configuration section.
For more information, including a summary of the default
maximums, see “Configuring Reporting, by Granularity”.
Configure the maximum size of Call • If necessary, modify the cdr.max-page-size option, to
Detail Record (CDR) and Call configure a suitable value for your deployment, for the
Events reports. maximum number of CDR or metrics records per page The
default is 100.
• Consider also the cdr.max-page-count option, for the
maximum number of pages per report. The default is 10.
Configure database retention Use the Database Retention Policy Wizard (see “Data Retention
policies. Policy Wizard” on page 134) to configure the database retention
policies.
For more information, see “Configuring Database Retention
Policies” on page 274.
Configure Reporting Server behavior See “Important Reporting Server Configuration Options” on
in general. page 276.

(Optional) Configure HTTPS to See “Controlling Access to Reporting Services” on page 283.
secure access to Reporting Services.
Verify that Genesys Administrator If necessary, configure or modify the connection between
displays GVP reports requested from Genesys Administrator and the Reporting Server. For more
the Monitoring > Voice Platform information, see the Genesys Voice Platform 8.5 Deployment
navigation panel. Guide.
If running on Oracle in partitioned Disable the GATHER_STATS_JOB before installing the RS database
mode, optimize performance. to ensure that inaccurate statistics are not associated with the
staging tables. For more information, see the Genesys Voice

Chapter 14: Configuring the Reporting Server Configuring Reporting, by Granularity
Configuring Reporting, by Granularity

Granularity refers to the degree of aggregation in a given summary report. For
example, a request for a Call Peak report at the granularity level of month will
return a peak value for each month in the requested time range. A request for a
Call Peak report at the granularity level of week will return a peak value for
each week in the requested time range.
The Reporting Server supports reporting at the following levels of granularity:
• 5-minute
• 30-minute
• Hour
• Day
• Week
• Month
If the requested time period does not encompass an integral unit that matches
the specified granularity level, then the Reporting Server expands the time to
cover an integral number. For example, if the granularity is day, a request for a
report from 2008/01/01 00:00 – 2008/01/01 14:00 will be expanded to 2008/01/01
00:00 – 2008/01/02 00:00.
The Reporting Server normalizes From and To parameters that specify the time
range in a reporting request, so that they lie on time unit boundaries that match
the granularity level. For example, if the granularity is hour, then the Reporting
Server normalizes the start time and end time of the report so that they point to
the beginning of an hour. In this case, a start or end time request for 11:30 will
be normalized to 11:00.
Ensure that the values that are set for the rs.query.limit.<granularity>
configuration options in the reporting section (see page 282) are appropriate
for your reporting purposes and environment.
Table 34 summarizes the default values for the reporting.rs.query.limit.
<granularity> configuration options. Each of these options specifies the
maximum number of units of a particular aggregation period that will be
included in reports at that aggregation period’s level of granularity. For
example, if you request a 5-minute report covering 2008/11/17–2008/11/19,
the range is truncated to cover the day maximum (2008/11/17 00:00–
2008/11/18 00:00).
User’s Guide 273

Chapter 14: Configuring the Reporting Server Configuring Database Retention Policies
Table 34: Default Maximum Units,

by Granularity Level
Aggregation Period Maximum Number of Units
5 minutes 288 (5-minute periods, equals 1 day)
30 minutes 48 (30-minute periods, equals 1 day)
Hour 168 (hours, equals 1 week)
Day 92 (days)
Week 53 (weeks)
Month 36 (months)
Configuring Database Retention Policies

By default, the database maintenance process runs daily to purge data in
accordance with database retention policies. The database retention policies
are defined in the following options in the dbmp configuration section:
• On the Reporting Server, the rs.db.retention.*.default options—These
set the default retention periods for the GVP deployment overall.
• On the IVR Profile, equivalent rs.db.retention.* options—These
override the default retention periods, for data relating to the specific
VoiceXML or CCXML application.
Table 35 summarizes the default Reporting Server database retention periods
for data at the varying levels of granularity (aggregation periods).
Before you modify the default retention periods, consider your reporting
requirements and the reporting results you expect. Ensure that your default
database retention period settings are consistent with settings for the
reporting.rs.query.limit.<granularity> configuration options, so that data
that you expect to include in reports at various granularity levels is not purged
prematurely from the database.
For information on using the Data Retention Policy Wizard to configure these
policies, see “Data Retention Policy Wizard” on page 134.
Note: When using the default parameter, there is a possibility that one
maintenance execution may be skipped when Day Light Savings
occurs. This will cause missing data on the VAR reports.

Chapter 14: Configuring the Reporting Server Configuring Database Retention Policies
Table 35: Default Database Retention Periods
Type of Data Granularity Option Name in dbmp Section Minimum Default

Valid Value Value, in
(Integer) Days
CDRs N/A rs.db.retention.cdr.default >0 30
Operational 5-minute rs.db.retention.operations.5min.default >0 1

data
30-minutes rs.db.retention.operations.30min.default > 0 7
Daily rs.db.retention.operations.daily.default > 30 90
Hourly rs.db.retention.operations.hourly.default > 0 7
Weekly rs.db.retention.operations.weekly. > 30 364

default (52 weeks)
Monthly rs.db.retention.operations.monthly. > 30 1095

default (36 Months)
VAR summary 5-minute rs.db.retention.var.5min.default >0 1

statistics (Call
Summary and 30-minutes rs.db.retention.var.30min.default >0 7
IVR Action
Daily rs.db.retention.var.daily.default > 30 90
statistics)
Hourly rs.db.retention.var.hourly.default >0 7
Monthly rs.db.retention.var.monthly.default > 30 1095

(36 months)
Weekly rs.db.retention.var.weekly.default > 30 364

(52 weeks)
Service Quality Daily rs.db.retention.latencies.daily.default > 30 90

Latency
Summary data Hourly rs.db.retention.latencies.hourly.default >0 7
Weekly rs.db.retention.latencies.weekly. > 30 364

default (52 weeks)
Monthly rs.db.retention.latencies.monthly. > 30 1095

default (36 Months)
Service Quality N/A rs.db.retention.sq.failures.default >0 365

Failures
User’s Guide 275

Chapter 14: Configuring the Reporting Server Important Reporting Server Configuration Options
Table 35: Default Database Retention Periods (Continued)
Type of Data Granularity Option Name in dbmp Section Minimum Default

Valid Value Value, in
(Integer) Days
Service Quality Daily rs.db.retention.sq.daily.default > 30 90

Summary data
Hourly rs.db.retention.sq.hourly.default >0 7
Weekly rs.db.retention.sq.weekly. > 30 364

default (52 weeks)
Monthly rs.db.retention.sq.monthly. > 30 1095

default (36 Months)
Disabling CDR Storage for Resource Manager and

You can use the Reporting Server configuration option
[cdr]media-service-cdrs.reduce to disable storage of Resource Manager
(RM) and Media Control Platform (MCP) CDRs that have certain media
service types.
Set the option to true to disable storage of these CDRs in the remote database:
• Any RM CDR and MCP CDR with its media service type set to: media,
record or conference.
• Any RM CDR or MCP CDR that has the media service type CPD and
VXML resource flag set to false and media service type set to CPD.
Notes: • RM and MCP must provide the correct media service type in the first
CDR message for a given call session.
• A new VXML field for RM CDRs is set by RM, to satisfy the above
requirement.
• See RS.OR.MS.x for more details about media service types.
Important Reporting Server Configuration

Options
Environment > Applications > <Reporting Server> > Options tab. For the


The configurable Reporting Server parameters are in the following
• agentx—Parameters that determine the behavior of the connection attempts
and delays between the SNMP subagent and the SNMP Master Agent.
• cdr—Parameters determine behavior for processing and reporting on call
detail records (CDRs).
• dbmp—Parameters determine database retention policies (see Table 35 on
page 275).
• https (see “Controlling Access to Reporting Services” on page 283) —
Parameters determine HTTPS settings.
• https_key—Parameters determine HTTPS key settings.
• imdb—Parameters determine database query behavior.
• latency—Parameters determine latency threshold behavior.
logging behavior.
• messaging—Parameters specify the paths for the ActiveMQ JMS broker
that receives Reporting Server messages.
• persistence—Parameters configure behavior for Hibernate interactions
with the database.
• reporting—Parameters determine the number of records that will be
considered for different levels of granularity.
• schedule—Parameters provide the cron expressions for scheduling tasks.
• sqa—Parameters determine service quality behavior.
• transaction—Parameters provide the directory paths for the Atomikos
distributed transactions processing engine.
Table 36 provides information about important Reporting Server parameters
that are not described in Chapter 3 on page 37. Table 36 provides parameter
the Reporting Server Application object.
Except for changes in the dbmp and log sections, all changes take effect on
restart.
For information about all the available configuration options for the Reporting
Server, see the Genesys Voice Platform 8.5 Configuration Options Reference.
User’s Guide 277

Table 36: Selected Reporting Server Configuration Options
agentx Section
Subagent Connection Specifies the maximum connection attempts to be An integer greater than 0.
Attempts made by the SNMP subagent to the SNMP Default value: 0
Master Agent.
If the option value if not set or if the value is less
than or equal to 0, there is no limit on the number
of attempts.
cdr Section
Call Timeout The amount of time, in minutes, until a call is An integer in the range of
considered timed out from the perspective of 1-1440.
VAR and CDR reporting. Default value: 180 (3 hours)
The Reporting Server may receive no CDR
call-termination update because:
• The call was dropped from the platform (for
example, because a component shut down
unexpectedly).
Call Timeout • The Reporting Server is simply not receiving

(continued) updates from the component (for example,
because the network connection is down). The
component queues data that it cannot send to
the Reporting Server, so the Reporting Server
may eventually receive a CDR update for a
call that was previously assumed to be timed
out. In these cases, the Reporting Server will
appropriately update the CDR.
The interval at which the timeout process runs is
configurable (see schedule.quartz.rs.
calltimeout on page 283). The timeout process
uses the value of the call-timeout parameter to
identify calls that have timed out since the
process last ran.
Limit of Disk Storage Specifies a limit to the amount of disk storage String.
for Messages that can be used for messages handled by the Default value: 256 gb
Handled by the ActiveMQ broker.
ActiveMQ Broker
Max Page Count The maximum number of pages that will be An integer in the range of 1–
returned in any given CDR or Call Events report 100.
request. Default value: 10

Table 36: Selected Reporting Server Configuration Options (Continued)
Max Page Size The maximum number of records that will be An integer in the range of
returned in a single page in any given CDR or 1-10000.
Call Events report request. Default value: 100
This limit prevents users from overloading the
system by requesting unreasonably large numbers
of CDRs or metrics in a report.
imdb Section
Max Concurrent CDR The maximum number of concurrently executed An integer in the range of 1–
Queries CDR in-progress queries. 15.
Default value: 3
Max Query Lock The maximum time, in milliseconds, for the An integer in the range of
Timeout real-time query to wait before locking the 100–5000.
in-memory storage. Default value: 1000
latency Section
Threshold Criteria for Specifies the latency threshold, in milliseconds, <Threshold|Percentile>

<given> Latency and the percentile for the given latency. For every Default values:
Service Quality period, the Report Server
calculates the actual latency for the specified • Page Fetch—1500|95
percentile. It that number exceeds the threshold, • Audio Fetch—1000|95
an error is logged. • Grammar Fetch—1000|95
• Day Fetch—2000|95
• JavaScript Fetch—
1000|95
• Page Compile—100|95
• JavaScript Execution—
50|99
• Initial Response—
4000|95
• Call Answer—2000|95
• Call Reject—2000|95
• First Prompt Inbound—
2000|95
User’s Guide 279

• First Prompt
Outbound—2000|95
• Inter Prompt—2000|95
• Cumulative Response—
2000|95
• DTMF Prompt—2000|95
• ASR Input—2000|95
• No Input Response—
2000|95
• Recording Response—
2000|95
• Transfer Response—
2000|95
• MRCP ASR Session
Establish—100|95
• MRCP TTS Session
Establish—100|95
• MRCP ASR Set Params—
100|95
Threshold Criteria for • MRCP ASR Stop—100|95

<given> Latency • MRCP Define Grammar—
(continued) 500|95
• MRCP Recognize—500|95
• MRCP Speak—100|95
• MRCP TTS Set Params—
100|95
• MRCP TTS Stop—100|95

messaging Section
ActiveMQ Broker Specifies the type of connector that will be An integer equal to 0, 1, or 2.
Connectors enabled on the ActiveMQ broker. • 0 = Unencrypted
connections only.
• 1 = SSL connections only.
(Not supported for
GVP 8.1.3 or earlier
clients.)
• 2 = SSL-enabled for
GVP 8.1.4 and earlier
clients, that will connect
in unencrypted mode.
(GVP 8.1.4 and later
clients will connect in
encrypted mod.)
Default value: 0
ActiveMQ JMS Specifies the SSL listening port for the An integer greater than 0.
Broker Port for SSL ActiveMQ JMS broker that receives incoming Default value: 61617
data from Reporting Clients.
ActiveMQ Keystore Specifies the path to the Java Keystore file that A string of characters.
for SSL Private Key contain the cryptographic key and trusted Default value: keystore.ks
and Certificate certificate entries that ActiveMQ broker requires
to provide TLS/SSL support.
ActiveMQ Keystore Specifies the password that is required to open A string of characters.
Password the keystore used by the ActiveMQ broker. Default value: ””
Local Listening Specifies the IP address for the listening port that A string of characters.
Address for the is used by the ActiveMQ broker (an unencrypted Default value: ””
ActiveMQ's Broker connector).
Local Listening Specifies the IP address for the TLS (encrypted) A string of characters.
Address for the listening port that is used by the ActiveMQ Default value: ””
ActiveMQ's Broker broker (an unencrypted connector).
(TLS)
User’s Guide 281

persistence Section
Read-Only Mode Specifies read-only mode for the Reporting • true

Server. The Reporting Server does not write to • false
the remote database, but continues to support
report queries.
• If this option value is true, read-only mode is
enabled.
• If this option value is false, standard
read-write mode is enabled
reporting Section
Maximum configured The maximum number of <granularity> periods See Table 35 on page 275.
units of <time that are included in any report with a granularity Example:
period> of <granularity>, where <granularity> is:
If rs.query.limit.5min=288
• 5min—5-minute periods (1 day), a request for a report
• 30min at 5-minute granularity for
• day the time period 2008/01/01
00:00 – 2008/01/02 12:00
• hour
will be truncated to
• month 2008/01/01 00:00 –
• week 2008/01/02 00:00.
If a reporting request at a particular granularity
level specifies a time range that is greater than the
configured maximum, the request is truncated to
cover the maximum allowed time period, starting
from the From time specified in the request.
Summarization Specifies the buffer time, in minutes, that ensures An integer range from 0–
Buffer Time that the summarization process runs after this 44640.
time has elapsed. Records will be summarized Default value: 60
before this time.
The maximum Specifies the maximum time, in seconds, for Any integer range from 0–
allowed database Reporting Server to query the database before 65535.
query running time cancelling the request. Default value: 60
(in seconds) before
the RS sends cancel
query request to
database server

Chapter 14: Configuring the Reporting Server Controlling Access to Reporting Services
schedule Section
Call Timeout Process The cron schedule for Quartz to execute the Call Default value:
Timeout Process, which is responsible for timing 0 50 * * * ?
out Resource Manager, Media Control Platform,
Call Control Platform, and VAR CDRs, so that
they do not get stuck as open calls in the
database.
By default, the process runs every 50 minutes.
A configurable option specifies the timeout
interval that determines when a call is considered
timed out (see the cdr.call-timeout option).
RS DB Maintenance The cron schedule for Quartz to purge data from Default value:
Process the database, in accordance with data retention 0 0 1 * * ?
policies.
By default, the process runs at 1 a.m. every day.
sqa Section
Minimum Calls for Specifies the minimum number of calls that need Any integer value.
Service Quality to be recorded before the service quality alarm is Default value: 100
issued at the critical level.
Minimum Latency Specifies the minimum number of latency Any integer value.
Measurements for measurements that need to be recorded before a Default value: 100
Threshold Warning warning is logged.
Controlling Access to Reporting Services

GVP 8.1 and above leverages the HTTPS features to secure users access to
Reporting Services.
GVP 8.1 and above does not support selective, role-based access for different
categories of Reporting services.
The following procedure describes how to configure Reporting Server to use
HTTPS.
Procedure:
Enabling HTTPS for Reporting
Purpose: To enable HTTPS for Reporting.
User’s Guide 283

Start of procedure
1. Obtain a server certificate that is signed by a third-party authority (for
example, CAcert, Comodo, or VeriSign).
2. Install the server certificate.
Use the PKCS12Import utility to import the server certificate into the Jetty
keystore with the following command:
java -classpath <jar> org.mortbay.jetty.security.PKCS12Import
<source> <keystore>
Where:
<jar> is the path to the ems-rs.jar file.
<source> is the path to the PKCS12 file that contains the keys and
certificates.
<keystore> is the path to the keystone file where the keys and
certificates are installed.
For example,
java -classpath ems-rs.jar
org.mortbay.jetty.security.PKCS12Import rs_example_com.pkcs12
keystore.jks
Enter input keystore passphrase: secret123
Enter output keystore passphrase: secret123
Alias 0: 1
Adding key for alias 1
3. Configure the Reporting Server application.
a. In Genesys Administrator, go to the Provisioning > Environment >
Applications > <Reporting Server> > Options tab.
b. Under the reporting section, add the following parameters:
hostname = the FQDN of the host to which the server certificate is
assigned
protocol = https
c. Under the https section, modify the following parameters:
https.keystore.path = the path to the keystore file.
https.protocol = SSL.
password = the keystore password.
d. Under the https_key section, set the password parameter to the
keystore password.
e. Click Save.
4. Restart Reporting Server.

5. To test HTTPS authentication, navigate to

https://<FQDN>:8080/ems-rs/components, where <FQDN> is the fully
qualified domain name of the host to which the server certificate is
assigned.
Reporting Server will now service all web service requests through
HTTPS, including those from the Management Framework Reporting UI
(Genesys Administrator).
6. Use the Trusted Root Certification Authorities MMC snap-in to verify that
the certificate is trusted by Windows.
7. Verify that GVP reports are properly displayed in Genesys Administrator
when you request them from the Monitoring > Voice Platform navigation
panel.
End of procedure
User’s Guide 285


Chapter
15 Configuring GVP in
Multi-Site Environments
This chapter describes the requirements to configure Genesys Voice Platform
(GVP) to support multi-site environments.

Overview, page 287

Configuring the Site Folder, page 288
Overview
GVP supports multi-site configurations in large scale environments. Typically,
a single site is represented by a Resource Manager (RM) instance (or a pair of
redundant RM instances), a Reporting Server (RS) instance (or a pair of
redundant RS instances), and a pool of Media Control Platform (MCP)
instances. Within a single site, scalability is limited by the number of call
attempts-per-second (CAPS) supported by the Reporting Server.
GVP scalability has gone beyond a single site and is now scalable across
multiple physical sites. The Resource Manager can facilitate resource-sharing
between sites and consistently enforce usage policies across all sites. The
Reporting Server can generate historical and real-time reports that are filtered
to produce site reporting or system-wide reporting. In addition, SIP Server
instances within the same site can use all of the available resources within the
site.
Site Identification
In GVP multi-site environments, sites are identified by a folder object in
Genesys Management Framework. The folder is created and configured in
Genesys Administrator on the Provisioning tab. For example, Provisioning >
User’s Guide 287

Chapter 15: Configuring GVP in Multi-Site Environments Configuring the Site Folder
Environment > Application > Site folder. The Options tab in the
configuration properties of the Site folder, the Advanced View (Annex)
contains a configuration section called gvp.site. You can add various options
to this section to control resource-sharing, geo-location, and the types and
weight (or number) of calls that can be routed to this site. See Table 37.
Management Framework obtains the site name from the site folder within the
Applications folder. The DBID of the site folder is used as the site ID, so it
must be unique.
The only GVP Applications in the site folder are the Resource Manager and
Reporting Server.
Configuring the Site Folder

Configure the Site folder in Genesys Administrator, by using the options
described in Table 37 and the Procedure: Configuring a Site folder by using
Genesys Administrator.
Table 37: Site Folder Configuration Options
Configuration Option description Valid values

option
contact Specifies the SIP route address for A list of IP

(mandatory) this site. This configuration option is addresses and port
used for call forwarding and site numbers.
monitoring.
geo-location Specifies a list of geographic locations A comma-separated

that are supported by this site. list.
resource-sharing Specifies whether or not resource True or False

sharing is enabled for this site.
weight Specifies the relative weight that is Any unsigned

allocated to this site. integer.
Procedure:
Configuring a Site folder by using Genesys
Administrator
Purpose: To configure a Site folder to support multi-site environments.

Start of procedure
1. In Genesys Administrator, go to Provisioning > Environment >
Applications.
2. In the task bar, click New Folder.
The Folder name dialog box appears.
3. Enter the site folder name and click OK.
4. Right-click on the folder and select Edit.
The Configuration tab appears.
5. On the Options tab, select New.
6. In the Section field enter, gvp.site.
7. In the Name field:
a. Enter contact. This is the SIP route address in the format:
10.10.10:5060
Which represents either the virtual IP and Resource Manager proxy
port (when RM is clustered) or the network interface that RM binds to
and the RM proxy port (when RM is standalone). (This configuration
option is mandatory.)
b. Enter the following additional options, if required:
— weight—Enter a weight. (If a value is not specified, default = 100.)
— geo-location—Enter a location. (Optional, can be left blank.)
— resource-sharing—Enter true or false. (If a value is not
specified, default = true).
8. In the Value field, enter an appropriate value.
9. To add additional options, click Save & New.
10. To move an existing Resource Manager or Reporting Server Application
into the folder:
• Highlight the Application.
• In the task bar, select Move to.
• In the Browse window, select the Site folder you created in Step 2.
11. In the Confirm dialog box, click Yes.
End of procedure
Next Steps
• No further steps are required.
User’s Guide 289


Part
2 Monitoring GVP
This part of the Guide describes the available real-time and historical reports in
This information appears in the following chapters:
• Chapter 16, “Reporting Overview,” on page 293
• Chapter 17, “Voice Platform Dashboards,” on page 311
• Chapter 18, “Real-Time Reports,” on page 329
• Chapter 19, “Historical Reports,” on page 337
• Chapter 20, “Service Quality Reports,” on page 367
• Chapter 21, “Voice Application Reports,” on page 377
User’s Guide 291

:

Chapter
16 Reporting Overview
This chapter describes how to use Genesys Administrator to create real-time
and historical reports. It contains the following sections:

Reports—Using GA vs. Using GAX, page 293

Generating a Report with GA, page 294
 Generating a Report with GAX, page 298

GAX Report Generation Table, page 300

Report Groups, page 303
 Report Filters, page 306
Reports—Using GA vs. Using GAX

Genesys Administrator Extension (GAX) can now generate all reports that are
available in Genesys Administrator (GA), and many new reports that GA does
not offer. Configuration of GVP 8.5 and earlier, as well as the ability to
generate many reports, remains in Genesys Administrator (GA).
Below is a breakdown of reports that you can generate with GAX vs. with GA.
Functionality Exclusive to GAX

Generating these new reports:
• VoiceXML Call Arrivals, VoiceXML Call Peaks, Media Service Call
Arrivals, Media Service Call Peaks.
• Call Durations for VoiceXML, Media Service, and ASR/TTS.
Functionality Exclusive to GA
• Configuring GVP.
User’s Guide 293

Chapter 16: Reporting Overview Generating a Report with GA
Functionality Common to both GAX and GA

• Generating Call Browser Reports: Historical Call Status, In Progress Call
Status.
• Generating VAR reports: VAR Call Completion, VAR IVR Action Usage,
VAR Last IVR Action.
• Generating these Operational and Dashboard reports:

Real-time Call Browser, IVR Profile Call Arrivals, IVR Profile Call
Peaks, Tenant Call Arrivals, Tenant Call Peaks.

Component Call Arrivals (RM, MCP, CCP, PSTNC, CTIC, ASR,
TTS).

Component Call Peaks (RM, MCP, CCP, PSTNC, CTIC, ASR, TTS).

Call Dashboard, SSG Dashboard, Fetch Dashboard, PSTNC, CTIC
Dashboard.
• Generating Service Quality Reports: CallFailures, Call Summary, Latency
Details, Latency Dashboard.
Genesys Administrator is the legacy tool that you use to monitor your
call-center activity; it enables you to analyze call volumes, trends, and the
effectiveness of your voice and call-control applications.
For more information on how to use Genesys Administrator, see the
Framework 8.5 Genesys Administrator Help file.
Warning! The tenant that is defined as the parent becomes the reference
entry point in the tenant hierarchy. The parent tenant with read
permissions can view their child tenants and their configurations
and reports, but cannot view the child tenants below them (their
grandchild tenants).
Generating a Report with GA

The following procedures explain how to generate a report using Genesys
Administrator or Genesys Administrator Extension.
Procedure:
Generating a Report Using Genesys Administrator
Purpose: To generate a report by using Genesys Administrator.
Prerequisites
• The valid URL for Genesys Administrator—for example, http://<Genesys
Administrator host>/wcm/.

• The username and password with the correct permissions for running
reports.
• The name of Genesys Administrator application—for example, default.
Note: The default application object is automatically created, and is

seen when Genesys Administrator is invoked.
• The host name and port of the Genesys Configuration Server.
Start of procedure:
1. In the web browser’s address bar, enter http://<Genesys Administrator
host>/wcm/.
The login to Genesys Administrator dialog box appears.
2. Enter the following parameters:
• User Name
• Password
• Application
• Host Name
• Port
3. Click Login.
The Genesys Administrator screen appears (see Figure 8) with the
Monitoring tab active.
User’s Guide 295

Figure 8: Genesys Administrator
4. From the Navigation panel, select Voice Platform.

The reporting categories are visible (see Figure 9).

Figure 9: Genesys Administrator Voice Platform
5. Select the required report.

6. The filter screen appears with the possible filter criteria (see Figure 10). A
different set of filters appear for each report type.
Figure 10: Configure Filters for the Historical Call Browser Screen
User’s Guide 297

Chapter 16: Reporting Overview Generating a Report with GAX
7. Select the required filters.

For information on how to select the Active Call Browser filters, see
Procedure: Generating the Active Call Browser Report with GA, on
page 330.
8. Click Generate Report.
The Report Results screen appears (see Figure 11).
Figure 11: Example of Historical Call Browser Report Results
9. Click Clear Filters if you want to erase the existing filters.
End of procedure
Generating a Report with GAX

Genesys Administrator Extension, part of the Genesys Framework, is a
web-based graphical user interface (GUI) that provides advanced
administrative and operational functionality targeted primarily providing
access to GVP reporting and Operation Parameter Management.

Chapter 16: Reporting Overview Generating a Report with GAX
Procedure:
Generating a report using GAX
Purpose: To generate a report by using Genesys Administrator Extension

(GAX).
Prerequisites
• The valid URL for GAX—for example, http://<GAX_host>:<port>/gax/
reports.
Start of procedure:
1. In the web browser’s address bar, enter http://<GAX_host>:<port>/gax/.
The GAX login dialog box appears.
2. Complete the following fields:
• User Name
• Password
3. Click Login.
The GAX home screen appears.
4. In the Navigation panel across the top of the home screen, select Reports.
The VP Reporting drop-down menu appears.
5. Select the required report: the list Call Browser, Dashboard, Operational
Report, Service Quality Report, VAR Report.
6. The report screen appears; the filter criteria occupy the left third. A
different set of filters appears for each report type.
7. Select the appropriate filters. Required filters have a red asterisk(*) next to
their name.
Filters for each of the four report types are described in the online help:
• Call Browser Report
• Dashboard Report
• Operation Report
• Service Quality Report
• VAR Report
Filters may appear in these formats:
• Radio button list (examples: Call Status, Report Type)
• Check box list (examples: Query Data From, Media Control Platform
Components, IVR Profiles)
• Drop-down list (examples: Component Type, Call Type)
User’s Guide 299

Chapter 16: Reporting Overview GAX Report Generation Table
• Text entry field (examples: DIDs, Remote URI)

• Pop-up calendar / text entry field (examples: From Date, To Date)
8. Click Generate.
The Report Results window appears and occupies the right two-thirds of the
screen. See the table GAX Report Display Controls in the online help.
End of procedure
Next Steps
• Which choices generate a specific report? See “GAX Report Generation
Table” on page 300.
• What is in each report? See Chapter 19, “Historical Reports,” on page 337.
The following Reporting descriptions appear in GAX online help:
• Overview.
• Applying report filters for each report type (see also: “Report Filters” on
page 306).
• Reading the reports.
GAX Report Generation Table

The table “To Generate a Report, You Must Make These Selections” lists the
menu and filter choices that you must make to generate each report. These
choices are not the only choices available to you—they are the specific choices
that you must make to generate that particular report.
Other filter choices—such as Query Data From*, From Date*,To Date*,
Granularity, Session Type, and others—help you refine the data that each
report presents. Some are mandatory(*). These filters are common to multiple
reports; they are described in the GAX online help and in the table “Report
Filters” on page 306.
Table 38: To Generate a Report, You Must Make These Selections
Report Reports Report Filter Type Item selection

to generate Menu Type selection
selection selection selection
Real-time Call Call Browser In Progress — —

Browser
Historical Call Call Browser Historical — —

Browser

Table 38: To Generate a Report, You Must Make These Selections (Continued)

Call Dashboard Dashboard Call — —

Dashboard
SSG Dashboard Dashboard SSG — —

Dashboard
Fetch Dashboard Dashboard Fetch — —

Dashboard
Connector Dashboard Connector — —

Dashboard Dashboard
SQ Latency Dashboard SQ Latency — —

Dashboard Dashboard
IVR Profile Call Operational Call Arrivals IVR Profile Select IVR Profile(s)
Arrivals Report
IVR Profile Call Operational Call Peaks IVR profile Select IVR Profile(s)
Peaks Report
Component Call Operational Call Arrivals Component Select one: RM, MCP, VXML,
Arrivals Report Media Service, CCP, PSTNC,
CTIC
ASR / TTS Call Operational Call Arrivals Component Select one: ASR, TTS
Arrivals Report
Component Call Operational Call Peaks Component Select one: RM, MCP, VXML,
Peaks Report Media Service, CCP, PSTNC,
CTIC
ASR / TTS Call Operational Call Arrivals Component Select one: ASR, TTS
Peaks Report
Tenant Call Arrivals Operational Call Arrivals Tenant Select Tenant(s)

Report
Tenant Call Peaks Operational Call Peaks Tenant Select Tenant(s)

Report
IVR Profile Call Operational Call IVR Profile Select IVR Profile(s)
Durations Report Durations
Component Call Operational Call Component Select Media Control Platform

Durations Report Durations (MCP) Component(s)
User’s Guide 301

Table 38: To Generate a Report, You Must Make These Selections (Continued)

Tenant Call Operational Call Tenant Select Tenant(s)

Durations Report Durations
SQ Call Failures Service Call Failures — —

Quality (SQ)
Report
Call Summary SQ Report Call — —

Summary
Latency Details SQ Report Latency — —

Details
VAR Call VAR Report Call — —

Completion Completion
VAR IVR Action VAR Report IVR Action — —

Usage Usage
VAR Last IVR VAR Report Last IVR — —

Action Action

Chapter 16: Reporting Overview Report Groups
Report Groups
Reports that you can generate using the GAX-GVP Reporting Plugin fall into
these categories:
Call Detail Record Browsing

Generate real-time and historical reports of status for calls processed by
different components in the GVP deployment.
Dashboard
Generate real-time reports that monitor in-progress calls from the perspective
of IVR Profiles or GVP components.
Operational Reporting
Generate reports on the rate of call arrivals, call durations, and peak call
volume by IVR Profile or GVP component.
Call Duration Operational Reports for MCP VXML and

Media Service
The Operational Report type Call Duration generates total call duration data
for specific time periods.
The total call duration for a given hour is the sum of time spent by every call in
that hour. A call that spans multiple time periods contributes the appropriate
percentage of its duration to each of the time periods. For example, a 3-hour
call that starts at 3:45 spans four different hour periods, and thus contributes—
proportionally—to four call duration records.
Report granularities are HOUR, DAY, WEEK and MONTH.
MCP VXML Duration

MCP VXML Duration reports generate historical call duration summaries for
MCP calls that have the VXML resource usage flag set to used. These reports
contain:
• Total duration of MCP VXML calls per tenant-id.
• Total duration of MCP VXML calls per application-id
• Total duration of MCP VXML calls per component-id
• Total duration of MCP VXML calls for the entire site (single RS
deployment).
User’s Guide 303

Media Service Duration

Media Service Duration reports generate historical call duration summaries for
call sessions that use media services (sessions that have the media-service
CDR parameter set) These reports contain:
• Total duration of MEDIA RM calls per tenant-id and for the entire RS site.
• Total duration of CPD RM calls per tenant-id and for the entire RS site.
• Total duration of RECORDING RM calls per tenant-id and for the entire
RS site.
• Total duration of CONFERENCE RM calls per tenant-id and for the entire
RS site.
• Total duration of TREATMENT MCP (MCP sessions with TREATMENT
and with no VXML) calls per tenant-id and for the entire RS site.
ASR Call Duration

ASR Call Duration reports generate historical call duration summaries for
MCP calls that have the ASR resource usage flag (see RS.MCP.CDR.2) set to
used. These reports contain:
• Total call duration of MCP calls that used an ASR resource for a given
IVR application-id.
tenant-id.
component-id.
• Total call duration of MCP calls that used an ASR resource for the entire
site (single RS deployment).
TTS Call Duration

TTS Call Duration Reporting provides historical call duration summaries for
MCP calls that have the TTS resource usage flag (see RS.MCP.CDR.2) set to
used. These reports contain:
• Total call duration of MCP calls that used an TTS resource for a given IVR
application-id.
• Total call duration of MCP calls that used an TTS resource for a given
tenant-id.
• Total call duration of MCP calls that used an TTS resource for a given
MCP (component-id).
• Total call duration of MCP calls that used an TTS resource for the entire
site (single RS deployment).

Service Quality Reporting

Generate service quality reports of call failures, call summaries, and latency
details for MCP components, from call analysis data provided by the Reporting
Server.
Voice Application Reporting

Generate reports on the logical success and failure rates for calls and IVR
Actions in a given IVR Profile.
Note: VAR reporting data is available only for applications that leverage the
VAR <log> interfaces described in the Reporting Server Functional
Specification.
User’s Guide 305

Chapter 16: Reporting Overview Report Filters
Report Filters
Table 39 describes the filter criteria that you can use to retrieve call detail
records, IVR action data, or summary data.
Table 39: Report Filters
Filter Name Description
Time Range Filters the data by start date, start time, end date, and end
time. The results will display calls that started on or after
the start time and ended before the end time.
Note: Selecting from the Predefined Ranges
automatically populates the Start and End dates with
common time ranges.
• This Hour (granularity = five minutes)
• Today (granularity = hour)
• Yesterday (granularity = hour)
• This Week (granularity = day)
• Last Week (granularity = day)
• This Month (granularity = day)
• Last Month (granularity = day)
The following Time Ranges are for the Active Call
Browser report only.
• Last Five Minutes
• Last Fifteen Minutes
• Last Thirty Minutes
• Last Hour
• Last Day
Default: Today.
Granularity Presents the data at various levels of aggregation:

• Five Minutes
• Thirty Minutes
• Hour
• Day
• Week
• Month
• Six week moving average
Default: Hour

Table 39: Report Filters (Continued)
IVR Profile Filters the data by IVR Profile. You can choose more than
one IVR Profile for some of the reports.
For more information on IVR Profiles, see Chapter 6 on
page 103.
Component Type Filters the data by Component Type. The possible

Components are:
• RM—Resource Manager
• MCP—Media Control Platform
• CCP—Call Control Platform
• CTIC—CTI Connector
• PSTNC—PSTN Connector
Component Filters the data by the component. A component is a

provisioned Resource Manager, MCP, CCP, CTIC, or
PSTNC application. You can choose more than one
component for some of the reports.
Note: All selected components must be of the same type.
Tenant Filters the data by Tenant.
DID Filters the data by DID.
Call Type Filters the data by call type. The possible call types are:
• Inbound—Applicable for MCP and RM components.
• Outbound—Applicable for MCP and RM components.
• Bridged—Applicable for MCP components only.
• New Call—Applicable for CCP components only.
• Createccxml—Applicable for CCP components only.
• External—Applicable for CCP components only.
• Unknown—Applicable for RM components only.
Call Length Filters the data by the length of time, in milliseconds, of

the call. Minimum and maximum durations can be
specified. If only a minimum duration is specified, calls
that exceeded this duration are displayed. If only a
maximum duration is specified, calls that lasted for less
than or equal to this duration are displayed.
User’s Guide 307

Call Failure Type Filters the data by the reason for the failed call. The
possible call failure types are:
• Call answer
• Call reject
• Inbound first prompt latency
• Outbound first prompt latency
• Inter-prompt latency
• Cumulative response latency
• Audio gap latency
• Application error
• System error
• Unknown Failure
ID Filters the data by the call ID. The possible IDs are:
• Session ID—The GVP Component specific ID that is
generated by the component to identify the call leg.
• GVP GUID—The globally unique ID that identifies a
complete interaction with GVP. This ID is generated by
the Resource Manager, and is passed to all the
resources that provide service for the call.
• Genesys UUID—The Genesys CallUUID that is
generated by T-Server or SIP Server.
For more information on these IDs, see Chapter 1,
“Introduction,” on page 17.
Call State Filters the data by Call States. The possible Call States
are:
• Accepted—The call has been received by Resource
Manager, but has not yet landed on a VoiceXML
platform, or been transferred to an agent.
• IVR—The call has landed on a VoiceXML platform.
• Transferring—The call is being transferred to an
agent.
• Transferred—The call was successfully completed.

Call Disposition Filters the data according to the outcome of the call. The
possible call dispositions are:
• Completed in IVR—The call completed in self service.
• Abandoned in Queue—The caller hung up while
waiting in the queue. This is available only for those
call flows that include IVR Server for CTI.
• Transferred to Agent—The call was send to an
agent.
• Rejected—The call cannot be routed to a media
resource.
Call End State Filters the data by Call End State. The possible Call End
States are:
• Application End—The voice application hung up.
• System Error—The call did not end properly.
• Unknown—The platform did not log an end state.
• User End—The caller hung up.
Call Result Filters the date by Call Results. The possible Call Results
are:
• Success—The call was processed successfully.
• Failed—A failure occurred that prevented the call
from being processed properly
• Rejected—The platform rejected the call.
• Unknown—Some unknown reason that the call ended
abruptly.
Virtual Reporting Filters the data by the user defined Virtual Reporting
Object Object.
Remote URI Filters the data by the full URI of the remote party that is
involved in the session.
Note: Accepts the * wildcard.
Local URI Filters the data by the URI of the local service.
Note: Accepts the * wildcard.
Note: Certain filters are only available when a particular component is

selected. For example, Call End State is displayed only when MCP is
selected.
User’s Guide 309

The data on the reports that use the granularity filter are stored in the database
for the length of time that is given for the dbmp.rs.db.retention.cdr.default
parameter. Granularity works with the data reporting limits that are configured
in the Reporting Server. These limits are the maximum amount of data that the
Reporting Server returns based on the which granularity level is selected. The
Report Server options are:
• rs.query.limit.5mins
• re.query.limit.30mins
• rs.query.limit.hour
• rs.query.limit.day
• rs.query.limit.week
• rs.query.limit.month
For more information, see “Configuring Reporting, by Granularity” on
page 273.

Chapter
17 Voice Platform Dashboards

This chapter describes the Genesys Voice Platform (GVP) Dashboards. The
Voice Platform Dashboards display real-time and historical information for
selected IVR Profiles, components, and tenants. It contains the following
sections:

Overview, page 311

Call Dashboard, page 314
 SQ Latency Dashboard, page 319

Fetch Dashboard, page 321

SSG Dashboard, page 322
 PSTNC Dashboard, page 325

CTIC Dashboard, page 326
Overview
The Voice Platform Dashboards display a high-level summary of the current
usage for IVR Profiles and Resource Manager (RM), Media Control Platform
(MCP), Call Control Platform (CCP), Supplementary Services Gateway
(SSG), and PSTN Connector (PSTNC) components. Each dashboard can be
configured to auto-update its display at regular intervals. The data that is
displayed represents current values. How current the data is depends on two
factors:
• In-progress session counts are derived from CDRs. There can be delays in
the delivery of CDRs to Reporting Server. The dashboard reflects the
CDRs that are currently available to Reporting Server.
• Calls this hour/day; Peaks today are derived from operational summary
data. By default operational data is submitted once per minute. The hover
ToolTip indicates how current the reported values are.
The following procedures describe how to filter the dashboard layout.
User’s Guide 311

Chapter 17: Voice Platform Dashboards Overview
Notes: The Voice Platform Dashboard is unrelated to the Dashboard under

Monitoring > Environment. This dashboard displays alarm and log
information for all applications in the environment.
When the Reporting Server is switching from back up mode to
primary mode, the dashboard will display inaccurate data for a short
period of time.
Procedure:
Filtering the Voice Platform Dashboard with GA
Note: In GAX, see the onscreen help about filtering and configuration.
Purpose: To filter the Voice Platform Dashboard using Genesys Administrator.
Prerequisites
• The valid URL for Genesys Administrator—for example, http://<Genesys
Administrator host>/wcm/.
reports.
• The name of Genesys Administrator application—for example, default.
Note: The default application object is automatically created, and is seen

when Genesys Administrator is invoked.
• The host name and port of the Genesys Configuration Server.
Start of procedure:
1. In the web browser’s address bar, enter http://<Genesys Administrator
host>/wcm/.
The login to Genesys Administrator dialog box appears.
2. Enter the following parameters:
• User Name
• Password
• Application
• Host Name
• Port

Chapter 17: Voice Platform Dashboards Overview
3. Click Login.
The Genesys Administrator screen appears (see Figure 8 on page 296) with
the Monitoring tab active.
4. From the Navigation panel, select Voice Platform.
5. Select the required dashboard—for example, Call Dashboard.
The selected dashboard appears (for example, see Figure 12).
Figure 12: The Call Dashboard
The dashboards enable you to do the following:

• Add—Add new IVR Profiles by service type, Components by
component type, or Tenants to the Voice Platform Dashboard.
• Remove—Remove the selected IVR Profile, Component, or Tenant from
the dashboard display.
User’s Guide 313

Chapter 17: Voice Platform Dashboards Call Dashboard
• Remove All—Remove all IVR Profiles, Components, or Tenants from

the dashboard display.
To add IVR Profiles, Components, or Tenants:
a. Click Add. The Select IVR Profiles, the Select Components, or the
Select Tenants dialog box appears.
b. Select one or more IVR Profiles, Components, or Tenants by either
clicking the check boxes, or selecting the individual rows. When
clicking rows, you can also use the Shift/Ctrl keys for multiple
selections.
Note: By default, you can select a maximum of 50 IVR Profiles,

Components, or Tenants at a time. Use the
dashboard.max.filtered.items parameter in the rptui section of
the default application to change this limit. For more information
on configuring options, see “Configuring GVP Processes in
Genesys Administrator” on page 30.
c. Click OK.
To remove IVR Profiles, Component, or Tenants:
a. Select the desired IVR Profiles, Components, or Tenants from the
Dashboard.
b. Click Remove.
To remove all of the IVR Profiles, Components, or Tenants:
• Click Remove All.
6. To enable refreshing for the Voice Platform Dashboard, select the Refresh
every check box, and enter the desired rate in either seconds or minutes.
End of procedure
Note: In order for the GVP Dashboard to display data, you must connect
each GVP application to the SNMP Master Agent application.
Call Dashboard
This section describes the Call Dashboard. It contains the following sections:

IVR Profile Utilization, page 315

Component Utilization, page 316

Tenant Utilization, page 318

IVR Profile Utilization

The IVR Profile Utilization section of the Call Dashboard (see Figure 13)
displays current IVR Profile activity for the current day and time up to and
including the time that is spent viewing the dashboard. The dashboard
pictorially represents the current burst levels for each IVR Profile. They are
flagged with the following colors:
• Green—Level 1
• Yellow—Level 2
• Red—Level 3
The level of the colored bar depicts the utilization progress which is also
displayed in the hover ToolTip.
Note: Usage limits and bursting limits apply to IVR Profile utilization only.
For more information on usage limits and bursting levels see Chapter 14,
“Configuring the Reporting Server,” on page 271.
Figure 13: Call Dashboard—IVR Profile Utilization
Table 40 lists and describes the IVR Profile Utilization columns.
Table 40: IVR Profile Utilization Columns
Column Description
Profile Name The name of the IVR Profile as provisioned in Genesys

Administrator.
In-progress The number of calls, broken down by Resource

Manager call type, that are currently in progress. This
value is as current as the CDRs in the Reporting Server
database.
Note: Select the hyperlinked value to display the
Active Call List report.
User’s Guide 315

Table 40: IVR Profile Utilization Columns (Continued)
Column Description
This Hour The number of calls, broken down by Resource

Manager call type, that were processed in the last hour
including those calls processed at the last updated time
as shown in the Tooltip.
Historical Call Browser Report.
Today The number of calls, broken down by Resource

Manager, that were processed today including those
calls processed at the last updated time as shown in the
tool tip.
Historical Call Browser report.
Daily Peak The greatest number of in-progress calls that are

registered for this IVR Profile.
Note: Select the hyperlinked value to display the IVR
Profile Call Peaks report.
Peak Time The time for which today’s peak calls registered.
VAR % Successful The percentage of calls that had a successful VAR call
result.
VAR % Failed The percentage of calls that had a failed VAR call
result.
VAR % Unknown The percentage of calls that did not have the VAR call
result.
Component Utilization
The Component Utilization section of the Call Dashboard (see Figure 14)
displays current activity for GVP Components (RM, MCP,CCP, SSG, and
PSTNC platforms) for the current day and time up to and including the time
that is spent viewing the dashboard.

Figure 14: Call Dashboard—Component Utilization
Table 41 lists and describes the Component Utilization columns.

Table 41: Component Utilization Columns
Column Description
Component Name The name of the Component as provisioned in Genesys

Administrator.
Note: Component names are listed in alphabetical order
according to the component type (CCP, MCP,
RM,CTIC, PSTNC).
In-progress The number of calls, broken down by call type, that are
currently in progress.
Note: Select the hyperlinked value to display the Active
Call List report.
This Hour The number of calls, broken down by call type, that
were processed in the last hour. This values is current as
of the time that is displayed in the ToolTip.
Today The number of calls, broken down by call type, that

were processed today including those calls that were
processed at the last updated time as shown in the
ToolTip.
User’s Guide 317

Table 41: Component Utilization Columns (Continued)
Column Description

registered for this Component.
Component Call Peaks report.
Tenant Utilization
The Tenant Utilization section of the Call Dashboard (see Figure 15) displays
current activity for tenants for the current day and time up to and including the
time that is spent viewing the dashboard.
Figure 15: Call Dashboard—Tenant Utilization
Table 42 lists and describes the Tenant Utilization columns.
Table 42: Tenant Utilization Columns
Column Description
Tenant Name The name of the Tenant as provisioned in Genesys

Administrator.
In-progress The number of calls, broken down by Resource

Manager call type, that are currently in progress. This
value is as current as the CDRs in the Reporting Server
database.
Active Call List report.
This Hour The number of calls, broken down by Resource

Manager call type, that were processed in the last hour
including those calls processed at the last updated time
as shown in the Tooltip.
Historical Call Browser Report.

Chapter 17: Voice Platform Dashboards SQ Latency Dashboard
Table 42: Tenant Utilization Columns (Continued)
Column Description
Today The number of calls, broken down by Resource

Manager, that were processed today including those
calls processed at the last updated time as shown in the
tool tip.

registered for this IVR Profile.
Note: Select the hyperlinked value to display the IVR
Profile Call Peaks report.
VAR % Successful The percentage of calls that had a successful VAR call
result.
VAR % Failed The percentage of calls that had a failed VAR call
result.
VAR % Unknown The percentage of calls that did not have the VAR call
result.
SQ Latency Dashboard
The SQ Latency Dashboard (see Figure 16) displays service quality latency
data for a set of selected MCP components. The latency data is grouped by
latency category, which contains many latency types. The data is broken down
for comparison by today’s data, this week’s data, and this month’s data. For
more information on SQ Latency data, see Chapter 20, “Service Quality
Reports,” on page 367.
Figure 16: SQ Latency Dashboard
Table 43 lists and describes the SQ Latency Dashboard Columns.
User’s Guide 319

Chapter 17: Voice Platform Dashboards SQ Latency Dashboard
Table 43: SQ Latency Dashboard Columns
Column Description
Period The time of the reporting period.
Call Control The aggregated number of call type latency thresholds.

Possible types are:
• CALL_ANSWER
• CALL_REJECT
Prompt The aggregated number of prompt latency

thrsholds.Possible types are:
• INBOUND_FIRST_PRMOPT
• OUTBOUND_FIRST_PROMPT
• INTERPROMPT
• INITIAL_RESPONSE
Response The aggregated number of response latency thresholds.

Possible types are:
• CUMULATIVE_RESPONSE
• DTMF_INPUT_RESPONSE
• ASR_INPUT_RESPONSE
• NOINPUT_RESPONSE
• RECORDING_RESPONSE
• TRANSFER_RESPONSE
Fetching The aggregated number of fetch latency thresholds.

Possible types are:
• PAGE_FETCH
• AUDIO_FETCH
• GRAMMAR_FETCH
• DATA_FETCH
• JAVA_SCRIPT_FETCH

Chapter 17: Voice Platform Dashboards Fetch Dashboard
Table 43: SQ Latency Dashboard Columns (Continued)
Column Description
Execution The aggregated number of execution latency

thresholds. Possible types are:
• PAGE_COMPILE
• JAVA_SCRIPT_EXECUTION
MRCP The aggregated number of MRCP latency thresholds.

Possible types are:
• MRCP_ASR_SESSION_ESTABLISH
• MRCP_TTS_SESSION_ESTABLISH
• MRCP_ASR_SET_PARAMS
• MRCP_TTS_SET_PARAMS
• MRCP_ASR_STOP
• MRCP_TTS_STOP
• MRCP_DEFINE_GRAMMAR
• MRCP_RECOGNIZE
• MRCP_SPEAK
Fetch Dashboard
The Fetch Performance Dashboard (see Figure 17) displays near-real-time
statistics of the Media Control Platform and Call Control Platform fetching
processes. This dashboard pulls this data from the Reporting Server through
SNMP from the Media Control Platform and the Call Control Platform
components.
Figure 17: Fetch Performance Dashboard
User’s Guide 321

Chapter 17: Voice Platform Dashboards SSG Dashboard
Table 44 lists and describes the Fetch Performance Dashboard columns.
Table 44: Fetch Performance Dashboard Columns
Column Description
Component The name of the component as provisioned in Genesys

Administrator.
Sessions The total number of action sessions per component.
Requests The total number of active fetch requests.
Cache Size The current size, in megabytes, of the cache memory.
HTTP Hit The total number of HTTP cache hits.
File Hits The total number of file cache hits.
Proxy Hit The total number of HTTP proxy cache hits.
Proxy Reval The total number of HTTP proxy cache re-validations.
HTTP Failed The total number of failed HTTP fetches.
File Failed The total number of failed file fetches.
Avg. HTTP Resp The average HTTP response time.
Avg. Proxy Hit The average HTTP proxy cache hit response time.
Avg. Proxy Reval The average HTTP proxy cache re-validation response
time.
Avg. Server Resp The average HTTP server response time.
Avg. Server Reval The average HTTP server re-validation response time.
SSG Dashboard
The SSG Dashboard display near-real-time utilization of the Supplementary
Services Gateway (SSG) components. This dashboard pulls this data from the
Reporting Server through SNMP from the SSG components. The SSG
Dashboard has the following panes:

SSG IVR Profile Utilization, page 323

SSG Component Utilization, page 323

SSG Tenant Utilization, page 324

SSG IVR Profile Utilization

The SSG IVR Profile Utilization dashboard (see Figure 18) displays SSG
usage statistics summarized by IVR profile.
Figure 18: SSG Dashboard—IVR Profile Utilization
Table 45 lists and describes the SSG IVR Profile Utilization columns.
Table 45: SSG IVR Profile Utilization Columns
Column Description
IVR Profile Name The name of the IVR Profile.
Queued The total number of calls that are waiting in queues.
Successful The total number of successful calls.
Failed The total number of calls that failed.
Avg. Time The average time to complete a call.
Avg. Attempts The average number of attempts to complete a call.
SSG Component Utilization

The SSG Component Utilization dashboard (see Figure 19) displays SSG
usage statistics summarized by Component.
Figure 19: SSG Dashboard—Component Utilization
Table 46 lists and describes the SSG Component Utilization columns.
User’s Guide 323

Table 46: SSG Component Utilization Columns
Column Description
Component Name The name of the SSG component.
Active The total number of active calls.
Peak The peak number of active calls.
Total HTTP The total number of HTTP requests.
Total HTTPS The total number of HTTPS requests.
Total Rejected The total number of rejected requests.
SSG Tenant Utilization

The SSG Tenant Utilization dashboard (see Figure 20) displays SSG usage
statistics grouped by tenants.
Figure 20: SSG Dashboard—Tenant Utilization
Table 47 lists and describes the SSG Tenant Utilization columns.
Table 47: SSG Tenant Utilization Columns
Column Description
Tenant Name The name of the tenant.

Chapter 17: Voice Platform Dashboards PSTNC Dashboard
PSTNC Dashboard
The PSTN Connector (PSTNC) dashboard (see Figure 21) displays
near-real-time utilization data of PSTNC components and the PSTN boards
that they manage. This dashboard pulls this data from the Reporting Server
through SNMP from the PSTN Connector components.
Figure 21: PSTNC Dashboard
Table 48 lists and describes the PSTNC dashboard columns.
Table 48: PSTNC Dashboard Columns
Column Description
Component Name The name of the PSTN Connector component.
TDM Inbound Total The total number of TDM inbound calls received.
TDM Outbound The total number of TDM outbound calls attempted.

Total
TDM Inbound The total number of TDM inbound calls received that are
Active currently active.
TDM Outbound The total number of TDM outbound calls attempted that
Active are currently active.
SIP Inbound Total The total number of SIP inbound calls received.
User’s Guide 325

Chapter 17: Voice Platform Dashboards CTIC Dashboard
Table 48: PSTNC Dashboard Columns (Continued)
Column Description
SIP Outbound Total The total number of SIP outbound calls initiated.
SIP Inbound Active The number of SIP inbound calls that are currently active.
SIP Outbound The number of SIP outbound calls that are currently
Active active.
Board Name The name of the PSTN board.

Note: Hover over the board name to display the board
description in the tool tip. The D-channel status is
indicated by the green (up) or grey (down) icon beside the
board name.
Alarms The status of the PSTN board. This is indicated with a

colored icon:
• red—Indicates that the incoming signal is corrupt. On
the Dialogic card, the red LED is lit.
• yellow—Indicates that the network has failed. On the
Dialogic card, the yellow LED is lit.
• blue—Indicates the absence of an incoming signal. On
the Dialogic card, the red and green LEDs are lit.
Note: The lack of an icon means that there are no active
alarms of that color for this board. Multiple icons of
different colors can appear at the same time.
Loss of Sync The PSTN board is not synchronized.
Framing Error The PSTN board has encountered a framing error.
Bipolar Violation The PSTN board has encountered a bipolar violation.
Note: Select the arrow (>) beside the board to view each port. Hover over
each port number to view the individual details of that port.
Right-click a port to reset or disable it.
CTIC Dashboard
The CTI Connector (CTIC) dashboard (see Figure 22) displays near real-time
CTI Connector component and ICM connection statistics, which are polled by
the Reporting Server by using SNMP.

Figure 22: CTIC Dashboard
Table 49 lists and describes the CTIC dashboard columns.
Table 49: CTIC Dashboard Columns
Column Description
Component Name The name of the CTI Connector component.
Active Since The time that the component became active.
Total Active Calls The total number of active calls.
Total Active The total number of active inbound calls.

Inbound Calls
Total Active The total number of active outbound calls.

Outbound Calls
Total Bridged The total number of bridged calls.
Total Completed The total number of completed calls.
Total Completed in The total number of calls completed in IVR.

IVR
Total Failed at CTI The total number of calls failed at CTI.
Total Failed at GVP The total number of calls failed at the Voice Platform.
User’s Guide 327

Table 49: CTIC Dashboard Columns (Continued)
Column Description
Total Default Agent The total number of Default Agent responses received.
Total NewCall The total number of failed NewCall requests.

Failed
Total Queued Calls The total number of queued calls.
Total Route The total number of failed Route requests.

Request Failed
Total Route The total number of Route responses received.

Responses
Connection ID The connection ID of the CTI Connector.
Completed The total number of calls successfully completed on this

port.
Failed The total number of failed calls on this port.
Connection State The connection state of the CTI Connector (either UP or

DOWN).
Connection Time The time at which the connection request from Intelligent
Contact Management (ICM) is accepted by CTI
Connector.
Active The current number of active calls on this port.
PG Address The IP address of the ICM from which the connection

request is accepted.
PG Communication The listening port on which CTIC accepts new connection

Port requests from ICM.
Tenant The name of the tenant for which CTIC is handling the
calls on this port.

Chapter
18 Real-Time Reports
This chapter describes the available reports that display real-time data. It
contains the following sections:

Overview, page 329

Active Call Browser, page 329
Overview
The real-time reports display statistics of the current call that is in progress.
However, real-time data updates are not instantaneous, because there may be a
slight delay while the Media Control Platform (MCP), the Call Control
Platform (CCP), or Resource Manager (RM) sends data to the Reporting
Server.
Call detail records (CDR) and call events are delivered in batches to the
Reporting Server. By default the batch size is 500 CDRs or ten seconds. This
means that a message will be sent either when 500 CDR updates are queued, or
ten seconds has expired, whichever occurs first. You can reconfigure the
system to be more real-time by changing the batch size—for example,
changing it to 1. This means that a CDR update or the call event will be
delivered to the Reporting Server as soon as it is raised by the component.
There are performance implications to changing the batch size. For more
information, see “Configuring the Reporting Server” on page 271.
Active Call Browser

The Active Call Browser report (see Figure 23) displays the list of calls that
currently are being processed by GVP. It also includes any call that the
Reporting Server has not marked as timed out.
User’s Guide 329

Chapter 18: Real-Time Reports Active Call Browser
A call is considered as timed out if the call processing component has

unexpectedly shut down, or if the connection between the call processing
component and the Reporting Server is broken. In either case, the Reporting
Server has not received the update indicating that the call ended. If the
connection is down, the Reporting Server will eventually receive the update;
however, if the processing component unexpectedly shut down, the call will
stay as timed out.
The Reporting Server processes timed out calls once hourly (by default), and
marks calls as timed out when they have been in progress for more than a
configured period of time (by default 3 hours).
The following procedure describes how to generate the Active Call Browser
report.
Procedure:
Generating the Active Call Browser Report with GA
Purpose: To generate the Active Call Browser report by using Genesys

Administrator.
Start of procedure:
1. Follow the instructions to generate a report (see Procedure: Generating a
Report Using Genesys Administrator, on page 294).
2. For Step 7 in those instructions:
a. On the Time Range tab, select the appropriate Time Range (on
page 306).
b. On the Call Info tab, select the appropriate Call Type (on page 308)
and Call State (on page 308).
c. On the IVR Profile tab, specify the IVR Profiles (on page 332) to
include in the report. To modify the list of IVR Profiles, do any of the
following:
i. Too add an IVR Profile, click Add. Select the IVR Profiles from the
list.
ii. To remove an IVR Profile from the list, click Remove.
iii. To remove all IVR Profiles the list. Click Remove All.
d. On the Components tab, select the Component Type (on page 307).To
build a list of Components (on page 307) for which active calls are to be
reported on:
i. To add a component, click Add.
ii. To remove a component, select it and click Remove.

iii. To remove all components, select Remove All.

e. On the Tenants tab, select the Tenant (on page 307).To build a list of
Tenants (on page 307) for which active calls are to be reported on:
i. To add a tenant, click Add.
ii. To remove a tenant, select it and click Remove.
iii. To remove all tenants, select Remove All.
f. On the Other tab, enter the Virtual Reporting Object, the Remote URI,
and/or the Local URI (on page 309).
3. Click Generate Report. Continue from Step 8 on page 298.
Click the specific GVP GUID link to display all component specific call
detail records for that GVP GUID.
End of procedure
Notes: If you do not select an IVR Profile, all IVR Profile data is displayed. If
you do not select a Component, only RM call detail records will
display. You must select the MCP component to view MCP call detail
records.
Data is returned if no filter is selected.
When the backup Reporting Server switches to primary mode, the
Active Call Browser report will show inaccurate data for a short period
of time.
You can select multiple IVR Profiles and multiple Components.
For more information on the details of completed calls, see “Component Call
Peaks” on page 347.
Figure 23: Active Call Browser Report
Table 50 describes the fields for the summary level for the Active Call Browser
report.
User’s Guide 331

Table 50: Active Call List Report Summary Fields
Field Description
GVP GUID The globally unique ID that identifies a complete

interaction with GVP. This ID is generated by the
Resource Manager, and is passed to all the resources that
provide service for the call. For more information on the
GVP Session ID, see “Session Identifiers” on page 22.
Site ID The site ID from which the call originated.
Session ID The GVP Component ID that is generated by the

component to identify the call leg. For more information
on the GVP Component ID, see “Session Identifiers” on
page 22.
Genesys UUID The Genesys CallUUID that is generated by T-Server or

SIP Server. For more information on the Genesys
CallUUID, see “Session Identifiers” on page 22.
IVR Profile The name of the IVR Profile that is selected.
Start Time The start date and start time of the call.
Component The name of the component (RM, MCP, or CCP) that is

selected.
Call Type The type of the call. Valid call types are the following:
• Inbound
• Outbound
• Bridged
• Unknown
• New Call
• Createccxml
• External
Remote URI The SIP address of the calling party.
Local URI The SIP address of the component that received the call.

Table 50: Active Call List Report Summary Fields (Continued)
Field Description
Call State The state of the call. Valid call states are the following:
• IVR—The call is being processed by MCP.
• Accepted—The call has been received by RM, but not
landed on the VoiceXML platform or transferred to an
agent.
• Transferring—The call is being transferred to an agent.
• Transferred—The call was successfully transferred to
an agent.
Profile Usage The number of calls in progress for the given IVR Profile.
This is recorded when the Resource Managers associates
the call with the IVR Profile. The burst level is indicated
with a color bar. The colors green, yellow, and red
correspond to burst levels 1, 2, and 3 respectively.
Tenant Usage The number of calls in progress for the tenant. The burst
level is indicated with a color bar. The colors green,
yellow, and red correspond to burst levels 1, 2, and 3
respectively.
The Active Call List Details report breaks down the detail recorded of the
selected call according to component type (see Figure 24).
Figure 24: Active Call List Details Report
Table 51 describes the fields for the Active Call Browser Details report.
Table 51: Active Call List Details Report Fields
Fields Description
Component Type The type of component. The possible components are:

• RM (Resource Manager)
• MCP (Media Control Platform)
• CCP (Call Control Platform)
User’s Guide 333

Table 51: Active Call List Details Report Fields (Continued)
Fields Description
GVP ID The globally unique ID that identifies a complete

Genesys ID The Genesys CallUUID that is generated by T-Server or


page 22.
IVR Profile The name of the IVR Profile as seen in Genesys

Administrator or Configuration Manager.
Component The name of the Component application as seen in

Genesys Administrator or Configuration Manager.
Start Time The start date and start time of the call.
End Time The end date and end time of the call if the call has
completed.
Call Status The state of the call. Valid call states are the following:
• Completed
• Timed Out
• In Progress
Call Type The type of call. Valid call types are the following:
• Inbound
• Outbound
• Bridged
• Unknown
• New Call
• Createccxml
• External
Remote URI The remote Uniform Resource Identifier.
Local URI The local Uniform Resource Identifier.

Table 51: Active Call List Details Report Fields (Continued)
Fields Description
Call State The state of the call. Valid call states are the following:
agent.
• Transferred—The call was successfully transferred to
an agent.
Dialed Number The number dialed.
Profile Usage The number of calls in progress for the given IVR Profile.
This is recorded when the Resource Manager associates
the call with the IVR Profile. The burst level is indicated
with a color bar. The colors green, yellow, and red
correspond to burst levels 1, 2, and 3 respectively.
Tenant Usage The number of calls in progress for the tenant. The burst
level is indicated with a color bar. The colors green,
respectively.
Session Start Origin The source from which the call started.
Parent Session ID The unique identifier of the parent session.
User’s Guide 335


Chapter
19 Historical Reports
This chapter describes the available historical reports. It contains the following
sections:

Overview, page 337

IVR Profile Call Arrivals, page 338
 Component Call Arrivals, page 340

Tenant Call Arrivals, page 342

Media Service Call Arrivals, page 343
 IVR Profile Call Peaks, page 344

Component Call Peaks, page 347

Tenant Call Peaks, page 349
 Media Service Call Peaks, page 351

MCP VXML Call Arrivals, page 352

MCP VXML Call Peaks, page 352
 ASR/TTS Usage, page 353

ASR/TTS Usage Peaks, page 354
 Media Services Usage and GVP Ports Peaks, page 355
 Historical Call Browser, page 358
Overview
The historical reports display call detail records, call arrival and summary
information over a selected period of time, of the specified IVR Profiles,
Components, and Tenants.
The Historical Call Summary and Historical Peaks reports display the data in
both a pictorial graph and a table. The graph provides the following navigation
features:
• To zoom in, drag from left to right on a selected area.
User’s Guide 337

Chapter 19: Historical Reports IVR Profile Call Arrivals
• To pan around the graph, right-click and drag the graph.

• To restore to normal view (100%), double-click the graph.
• To turn on or off the visibility of the individual series, click the eye icon in
the chart legend.
• Service Quality reports apply to NGi VoiceXML applications, and are
found in Genesys Administrator. GVP 8.1.5 and thereafter are NGi-only
platforms unless you run MCP 8.1.4 to incorporate support for GVPi
applications.
Note: If you are viewing reports in Microsoft Internet Explorer, Genesys

recommends that you use 1280 x 1024 (or greater) screen resolution,
or use Mozilla or Firefox, in order for the graphs to resize and repaint
quickly.
IVR Profile Call Arrivals

The IVR Profile Call Summary report (see Figure 25) lists a summary of call
arrival data that is submitted for each IVR Profile selected, for a given period
of time.
The following procedure describes how to generate the IVR Profile Call
Arrivals report.
Procedure:
Generating the IVR Profile Call Arrivals Report with GA
Purpose: To generate the IVR Profile Call Arrivals report using Genesys
Administrator.
Start of procedure:
2. For Step 7 in those instruction:
page 306) and the Granularity (on page 306).

Chapter 19: Historical Reports IVR Profile Call Arrivals
b. On the IVR Profile tab, specific the IVR Profiles (on page 307) to
following:
list.
End of procedure
To view the matching Historical Peaks data, select the Historical Peaks link
from the Related Reports section of the Tasks panel.
Note: You can select up to a maximum of eight IVR Profiles; however, you
cannot access the IVR Profile Peaks report from the Tasks panel.
Figure 25: IVR Profile Call Arrivals Report
Table 52 describes the fields for the IVR Profile Call Arrivals report.
User’s Guide 339

Chapter 19: Historical Reports Component Call Arrivals
Table 52: IVR Profile Call Arrivals Report Fields
Field Description
Period The date and time of the call for the given granularity
period. Click >> to view the End of Period column.
IVR Profile Name The total number of calls for the given IVR Profile.
IVR Profile Name The total number of inbound calls for the given IVR
(Inbound) Profile.
IVR Profile Name The total number of outbound calls for the given IVR
(Outbound) Profile.
IVR Profile Name The total number of unknown calls for the given IVR
(Unknown) Profile.
Component Call Arrivals

The Component Call Arrivals report (see Figure 26) lists a summary of call
arrival data that is submitted for each component selected, for a given period of
time.
The following procedure describes how to generate the Component Call
Arrivals report.
Procedure:Generating the Component Call Arrivals

Report
Purpose: To generate the Component Call Arrivals report using Genesys

Administrator.
Start of procedure:

Chapter 19: Historical Reports Component Call Arrivals
b. On the Components tab, select the Component Type (on page 307).To
reported on:
End of procedure
To view the matching Component Peaks data, select the Historical Peaks link
Figure 26: Component Call Arrivals Report
Table 53 describes the fields for the Component Call Arrivals report.
User’s Guide 341

Chapter 19: Historical Reports Tenant Call Arrivals
Table 53: Component Call Arrivals Report Fields
Field Description
Component Name The total number of calls for the given IVR Profile.
Component Name The total number of inbound calls for the given IVR
(Inbound) Profile.
Component Name The total number of outbound calls for the given IVR
(Outbound) Profile.
Component Name The total number of unknown calls for the given IVR
(Unknown) Profile.
Tenant Call Arrivals

The Tenant Call Arrivals report (see Figure 27) lists a summary of call arrival
data that is submitted for each tenant selected, for a given period of time.
The following procedure describes how to generate the Tenant Call Arrivals
report.
Procedure:Generating the Tenant Call Arrivals Report
Purpose: To generate the Tenant Call Arrivals report using Genesys

Administrator.
Start of procedure:
b. On the Tenants tab, select the Tenant (on page 307).To build a list of

Chapter 19: Historical Reports Media Service Call Arrivals
End of procedure
To view the matching Tenant Peaks data, select the Historical Peaks link
Figure 27: Tenant Call Arrivals Report Get
Table 54 describes the fields for the Tenant Call Arrivals report
Table 54: Tenant Call Arrivals Report Fields
Field Description
Start of Period The date and time of the call for the given granularity
Tenant Name The total number of calls for the given Tenant.
Tenant Name The total number of inbound calls for the given Tenant.
(Inbound)
Tenant Name The total number of outbound calls for the given Tenant.
(Outbound)
Tenant Name The total number of unknown calls for the given Tenant.
(Unknown)
Media Service Call Arrivals

The Genesys Media Server (GMS) collects, summarizes and reports on session
arrivals for each media service requested by SIP Server over MSML.
User’s Guide 343

Chapter 19: Historical Reports IVR Profile Call Peaks
For each of the services in Table 55, the GMS provides session arrival (total
sessions) for the RS deployment or for each tenant if in a multi-tenant
environment. The time period for each arrival count can be hour or higher
granularity (see Time Period in “Report Filters” on page 306).
The report generates data for the following types of calls:
Table 55: Media Service Call Arrival Reporting
Media Service Report Details
GMS Treatment Total arrivals for 'Treatment' sessions (excluding

Play Application use of VoiceXML)
GMS CPD Total arrivals for 'CPD' sessions
GMS Record Total arrivals for 'Recording' sessions
GMS Conference Total arrivals for 'Conferencing' sessions
GMS Media Total arrivals for 'Media' sessions
This report can be generated for specific tenants or for the entire deployment.
• See “Generating a Report with GA” on page 294 for the procedure steps.
• See “GAX Report Generation Table” on page 300 for the specific choices
to make, to generate this report (and others).
IVR Profile Call Peaks

The IVR Profile Call Peaks report (see Figure 28) provides the peak volume of
calls during a given period of time for a given IVR Profile.
The following procedure describes how to generate the IVR Profile Call Peaks
report.
Procedure:Generating the IVR Profile Call Peaks

Report
Purpose: To generate the IVR Profile Call Peaks report using Genesys
Administrator.

Start of procedure:
b. On the IVR Profile tab, specific the IVR Profiles (on page 307) to
following:
list.
End of procedure
Note: You can select one IVR Profile at a time.
To view the matching IVR Profile Call Arrivals data, select the Historical
Arrivals link from the Related Reports section of the Tasks panel.
The Peaks Volume, which is shown on the graph, counts the peak number of
calls that is observed during the specified time range, according to the selected
granularity level.
User’s Guide 345

Figure 28: IVR Profile Call Peaks Report
Table 56 describes the fields for the IVR Profile Call Peaks report.
Table 56: IVR Profile Call Peaks Report Fields
Field Description
Total The total number of calls for the given peak period.
Inbound The total number of inbound calls for the given peak
period.
Outbound The total number of outbound calls for the given peak
period.
Unknown The total number of unknown calls for the given peak
period.

Chapter 19: Historical Reports Component Call Peaks
Component Call Peaks

The Component Call Peaks report (see Figure 29) provides the peak volume of
calls during a given period of time for a given Component.
The following procedure describes how to generate the Component Call Peaks
report.
Procedure:Generating the Component Call Peaks

Report
Purpose: To generate the Component Call Peaks report using Genesys

Administrator.
Start of procedure:
reported on:
End of procedure
◆ To view the matching Component Call Arrivals data, select the Historical
Note: You can select up to eight components at a time.
granularity level.
User’s Guide 347

Chapter 19: Historical Reports Component Call Peaks
Figure 29: Component Call Peaks Report
Table 57 describes the fields for the Component Call Peaks report.
Table 57: Component Call Peaks Report Fields
Field Description
Total The peak number of concurrent calls reached during the

time period.
period.
period.
period.

Chapter 19: Historical Reports Tenant Call Peaks
Tenant Call Peaks

The Tenant Call Peaks report (see Figure 30) provides the peak volume of calls
during a given period of time for a given Tenant.
The following procedure describes how to generate the Tenant Call Peaks
report.
Procedure:Generating the Tenant Call Peaks Report
Purpose: To generate the Tenant Call Peaks report using Genesys

Administrator.
Start of procedure:
b. On the Tenant tab, select the Tenant (on page 307).To build a list of
i. To add a tenant click Add.
End of procedure
◆
To view the matching Tenant Call Arrivals data, select the Historical
Note: You can select up to eight components at a time.
granularity level.
User’s Guide 349

Chapter 19: Historical Reports Tenant Call Peaks
Figure 30: Tenant Call Peaks Report
Table 58 describes the fields for the Tenant Call Peaks report.
Table 58: Tenant Call Peaks Report Fields
Field Description
Total The peak number of concurrent calls reached during the

time period.
period.

Chapter 19: Historical Reports Media Service Call Peaks
Table 58: Tenant Call Peaks Report Fields (Continued)
Field Description
period.
period.
Media Service Call Peaks

The Genesys Media Server (GMS) collects, summarizes and reports on
concurrent session peaks for each media service requested by SIP Server over
MSML.
For each of the services in Table 59, the GMS provides peaks
(simultaneous/concurrent sessions) with a timestamp of when the peak
occurred, and for each tenant if in a multi-tenant environment. The time period
for each peak can be hours or a finer granularity (see Time Period in “Report
Filters” on page 306).
Table 59: Media Service Call Peak Reporting
Media Service Report Details
GMS Treatments Peak Daily Use of Treatment sessions and peak

timestamp. This may be used as the daily maximum call
parking use of GMS (excluding Play Application use of
VoiceXML).
GMS CPD Peak Daily Use of CPD sessions and peak timestamp. Call
Progress Detection Peaks can be used for outbound
calling.
GMS Record Peak Daily Use of Recording sessions and peak

timestamp. This may be useful for CRQM billing or at
least loading.
GMS Conference Peak Daily Use of Conferencing sessions and peak

timestamp. This may be useful for loading statistics of the
system.
GMS Media Peak Daily Use of Media sessions and peak timestamp.
This is for completeness where the service is not
categorized.
The reference interval is daily peak with a timestamp of the peak.
User’s Guide 351

Chapter 19: Historical Reports MCP VXML Call Arrivals
These peaks are available from Reporting Server, by tenant for reporting in
Genesys Administrator and for Genesys License Reporting Manager (LRM)
access via the Reporting Server web services API.
This report can be generated for specific tenants, or for the entire RS
deployment.
MCP VXML Call Arrivals

The MCP Call Arrivals report provides a summary of MCP session arrivals
using a GVP VoiceXML application (SIP Invite) during a given period of time.
Play Application (VoiceXML) will be monitored and counted as a VoiceXML
session. So VoiceXML session arrivals are total use count of VoiceXML
irrespective of how it was requested. The time period for each arrival count
can be hour or higher granularity (see Time Period in “Report Filters” on
page 306).
This report can be generated for specific tenants, IVR profiles, or MCP
components, or for the entire deployment.
to make, to generate this report (and others)
MCP VXML Call Peaks

The MCP Call Peaks report provides the peak volume of calls using a GVP
VoiceXML application (SIP Invite) during a given period of time. Play
Application (VoiceXML) will be monitored and counted as a VoiceXML
session. So VoiceXML session peaks are the simultaneous total use of
VoiceXML irrespective of how it was requested. The time period for each peak
can be hours or a finer granularity.
This report can be generated for specific tenants, IVR profiles, or MCP
components, or for the entire deployment.

Chapter 19: Historical Reports ASR/TTS Usage
Note: MCP CDR data is used to calculate the VXML Call Peaks. CDR data
that arrives late may be incorporated into hourly VXML peak
statistics, but a late CDR is not incorporated if it arrives late by more
than the operations counts retention parameter
(rs.db.retention.operations.counts.default).
ASR/TTS Usage
The ASR/TTS Usage report provides the overall ASR and TTS usage on calls
on a per-component (ASR or TTS Server), IVR Profile, Tenant, and
deployment basis.
Procedure:Generating the ASR/TTS Usage Report
Purpose: To generate the ASR/TTS Usage report using Genesys Administrator

Extension.
Start of procedure:
reported on:
End of procedure
User’s Guide 353

Chapter 19: Historical Reports ASR/TTS Usage Peaks
Table 60 describes the fields for the ASR/TTS Usage report.

Table 60: ASR/TTS Usage Report Fields
Field Description
Session The number of ASR or TTS sessions that are used during
a time period for a specific ASR Server or TTS Server,
IVR Profile, or tenant.
Session Count The total number of inbound, outbound, and unknown

calls for the given peak period.
ASR/TTS Usage Peaks

The ASR/TTS Usage Peaks report provides the peak ASR and TTS usage on
calls on a per-component (ASR or TTS Server), IVR Profile, or Tenant basis.
Note: GVP does not provide usage peaks for MRCPv2;

GVP does provide usage peaks for MRCPv1.
Procedure:Generating the ASR/TTS Usage Peaks

Report
Purpose: To generate the ASR/TTS Usage Peaks report using Genesys

Administrator.
Start of procedure:
reported on:

Chapter 19: Historical Reports Media Services Usage and GVP Ports Peaks
End of procedure
You can select and modify the list of IVR Profiles and Tenants in the same way
as you did the Components, by clicking on their respective tabs. The Resource
Type drop-down menu is not displayed on the IVR Profiles or Tenants tab.
Table 61 describes the fields for the ASR/TTS Usage Peaks report.
,
Table 61: ASR/TTS Usage Peaks Report Fields
Field Description
Session The peak number of concurrent ASR or TTS sessions that

are used during a time period for a specific ASR Server or
TTS Server, IVR Profile or Tenant.
Session Count The total number of inbound, outbound, and unknown

calls for the given peak period.
Media Services Usage and GVP Ports

Peaks
Resource Manager (RM) and Media Control Platform (MCP), along with SIP
Server, form a Genesys Media Server which is part of the Genesys SIP Server
offering. These same components are partly found in GVP. Consequently, this
section applies to both environments on the GVP platform. When routing
requests the use of a media service via MSML, the RM/MCP can track the
peak simultaneous number of sessions used by a Tenant of that service. A
Tenant can have a single IVR Profile defined for each service type. If the
system is used without multiple Tenants, then the single Tenant is the
Environmental Tenant.
The peak use of a given service (see “Media Service Call Peaks” on page 351)
may occur at a different time than another service. The peak uses are of value
for capacity planning purposes, and in some cases, for billing purposes (for
example, Pay Per Use).
The Services are as follows:
Peak Daily Use of Treatment sessions and peak timestamp

This may be used as the daily maximum call parking and call qualification use
of media services. As this can be a separate saleable and billable item with
User’s Guide 355

Genesys (unless you have GVP ports), it will eventually tracked by the
Genesys License Reporting Manager (LRM) product. Hence this statistic may
be used for pay per use (PPU) tracking by Genesys for customer billing
contracts, via the LRM server. This value is also useful for system capacity
planning. LRM tracks this peak value.
Peak Daily Use of CPD sessions and peak timestamp

Call Progress Detection Peaks might be used for monitoring loads for
outbound call progress detection. This peak tracks the number of simultaneous
sessions doing Call Progress Detection (CPD) using the MCP's CPD
capability. Currently this value is provided for system capacity planning.
Peak Daily Use of Recording sessions and peak timestamp

This peak tracks the number of simultaneous sessions doing recording by a
given tenant. Note that a given call may have many recording legs—caller,
agent, supervisor—such that the recording sessions level will exceed the
number of calls in the system if most calls record. Currently this value is
provided for system capacity planning.
Peak Daily Use of Conferencing sessions and peak timestamp

This peak may be useful for capacity planning and knowing how much load
the system is receiving.
Peak Daily Use of Media sessions and peak timestamp

This peak is for completeness where the service is not categorized, and can
contain special media functions outside of GVP type deployments.
Special Service: Peak Daily Use of VoiceXML = GVP Ports

If a call arrives using a GVP VoiceXML application (SIP Invite), the
VoiceXML session is included in the simultaneous peak use of VoiceXML.
Peak Daily Use of GVP VoiceXML sessions (see “MCP VXML Call Peaks”
on page 352) includes the use of MSML sessions initiated for a PLAY
APPLICATION (VoiceXML) request and not counted as a Treatment session.
Play Application (VoiceXML) is monitored and counted as a VoiceXML
session. So VoiceXML session peaks are the simultaneous total use of
VoiceXML irrespective of how it was requested. This peak is tracked and
measured by LRM using the Reporting Server. It can be a contributing metric
value for Pay Per Use (PPU) for GVP ports.
Daily Peak and Timestamp

The peak value reached on any given day, for any service, are tracked by the
GVP Reporting Server and the data is displayed in the GAX reporting tool.

LRM and PPU

The daily peak values for media services and VoiceXML sessions have a
timestamp for use by LRM. LRM may collect this daily peak, and after a
month of such collections, produce a final monthly peak statistic Pay Per Use
statistic for Genesys entitlement tracking.
At this time, LRM collects only VoiceXML session and Treatment Session
peaks.
Capacity Planning and Resource Allocations

Based on the media service requests for all service types including VoiceXML
sessions, different services may demand more resources and processing than
others. By examining the various service loadings and using the GVP and
Media portion of the Genesys Hardware Sizing Guide, an operational manager
of media services can determine if more MCPs are required to handle a
particular load or all loads.
Important Play Application Limitations

Some call center architects might prefer to use routing to control the
interaction with a caller while still using a VoiceXML application through the
Play Application media service. Several important limitations should be noted:
• GVP ports are required.
• The Play Applications will be reported as individual events for the tenant,
not as one whole call. That is, if the tenant has several applications in
routing using different VoiceXML snippets, they will all be reported
simply in GVP historical reporting as short tenant VoiceXML sessions.
• Routing plus Play Application is NOT a substitute for GVP applications
where session preservation is important: for example, you can not use
speech resources more than once per call, and hence only once in a Play
Application per call.
• If multiple Play Applications are used where back end communication
exists (web services), there will be NO cookie preservation or data
persistence between Play Application requests. This is particularly true for
CTI integrations with third party routing systems; GVP, not Play
Application, should be used.
Best Practices use of Play Application is to use ONE Play Application
VoiceXML for conducting a series of interactions with a caller, including back
end and speech resources, and when that interaction sequence is done, return
the data and call control to routing.
User’s Guide 357

Chapter 19: Historical Reports Historical Call Browser
Historical Call Browser

The Historical Call Browser report (see Figure 31) displays a list of completed
calls. It provides the ability to search for and browse call detail records. These
records represent calls that either completed successfully or eventually were
timed out by the Reporting Server.
A call is considered as timed out if the call processing component has
unexpectedly shut down, or if the connection between the call processing
component and the Reporting Server is broken. In either case, the Reporting
Server has not received the update indicating that the call ended. If the
connection is down, the Reporting Server will eventually receive the update;
however, if the processing component unexpectedly shut down, the call will
stay as timed out.
The Reporting Server processes timed out calls once hourly (by default), and
marks calls as timed out when they have been in progress for more than a
configured period of time (by default 3 hours).
The following procedure describes how to generate the Historical Call
Browser report.
Procedure:
Generating the Historical Call Browser Report
Purpose: To generate the Historical Call Browser report using Genesys

Administrator.
Start of procedure:
page 306).
b. On the Call Info tab:
i. Enter the DID (on page 307).
ii. Select the appropriate Call Type (on page 307).
iii. Select the Call Disposition (on page 309).
iv. Enter the minimum and maximum Call Lengths (on page 307).
v. Enter the Call End State (on page 309).
vi. Enter the Call Result (on page 309).
vii. Select the appropriate ID (on Page 308), and enter its value.

c. On the IVR Profile tab, specific the IVR Profiles (on page 307) to
following:
list.
d. On the Components tab, select the Component Type (on page 307).To
reported on:
e. On the Tenants tab, select the Tenant (on page 307).To build a list of
f. On the Other tab, enter the Virtual Reporting Object, the Remote URI,
and/or the Local URI (on page 309).
End of procedure
Note: You can select multiple IVR Profiles, Components, and Tenants.
User’s Guide 359

Figure 31: Historical Call Browser Report
Table 62 describes the fields for the Historical Call Browser report.
Table 62: Historical Call Browser Report Fields
Field Description
Start Date The start date and time of the call.
End Date The end date and time of the call.
Duration The length (in milliseconds) of the call.
GVP GUID The globally unique ID that identifies a complete


page 22.
Genesys UUID The Genesys CallUUID that is generated by T-Server or


Table 62: Historical Call Browser Report Fields (Continued)
Field Description
IVR Profile The name of the IVR Profile that is selected.
Tenant The name of the Tenant.
Component The name of the application (RM, CCP, MCP) that is

selected.
Call Type The type of call. Valid call types are the following:
• Inbound
• Outbound
• Bridged
• Unknown
• New Call
• Createccxml
• External
Call Status The state of the call. Valid call states are the following:
agent.
Remote URI The remote Uniform Resource Identifier.
Local URI The local Uniform Resource Identifier.
Call Disposition The outcome of the call. Valid call states are the
following:
• Unknown—The outcome was not specified by the
Resource Manager.
• Completed in IVR—The call completed in self service.
• Transferred to Agent
• Abandoned in Queue—The caller hung up while
waiting in the queue.
Note: The Abandoned in Queue disposition is available
only for those call flows that involve IVR Server for CTI.
This column is only visible when the Component type is
RM.
User’s Guide 361

Field Description
Queue Wait Time The amount of time (in milliseconds) that the caller waited
in queue. This value is displayed only for those calls with
the Transferred to Agent or Abandoned in Queue
disposition.
Note: Because calls can be transferred to an agent without
waiting in queue, the Queue Wait Time may display no
values (empty) for the Transferred to Agent disposition.
This column is only visible when the Component type is
RM.
Dialed Number The number that was dialed. This column is visible only
when the Component type is RM.
Site ID The site ID from which the call originated.

(RM only)
Session Start Origin The source from which the call started.
(CCP only)
End Reason The reason for the ending the call.

(CCP only)
Parent Session ID The unique identifier of the parent session.

(MCP only)
SQ The link to the SQ Call Failure Dashboard for the selected

(MCP only) MCP record.
End State The end state of the call. Valid states are:
(MCP only) • Application End—The application hung up.
• Unknown—The MCP did not log an end state.
End Result The end result of the call, as reported by the application.
(MCP only) Valid results are:
• Success
• Failed
Resource Type The type of resources that were used on the call.
(MCP only)

Field Description
Reason The reason for the call.

(MCP only)
Notes Any other notes that are associated with the call.
(MCP only)
Profile Usage The number of calls that are in progress for the given IVR
Profile. This is recorded when the Resource Manager
associates the call with the IVR Profile. The burst level is
indicated with a color bar. The colors green, yellow, and
red correspond to burst levels 1, 2, and 3 respectively.
Note: This column is only visible when the Component
type is RM.
Tenant Usage The number of calls that are in progress for the tenant. The
burst level is indicated with a color bar. The colors green,
respectively.
Note: This column is visible only when the Component
type is RM.
Clicking on a GVP GUID link results in displaying the Historical Call Browser
details that breaks down the selected record according to component type. It
displays the call detail records for all components that were involved in
handling the call. There can be multiple call detail records for each component
if there was more than one leg in the call.
Clicking on a Session ID of an MCP call results in displaying the VAR events
associated with the call.
Expanding an MCP call row reveals a table of custom VAR variables
associated with that call.
Per-Call IVR Actions Report

Within the Historical Call Browser you can generate the Per-Call IVR Action
Report. The Per-Call IVR Actions Report (see Figures 32 and 33) lists the
VAR actions and custom variables handled during a single Media Control
Platform session. The following procedure describes how to generate the
Per-Call IVR Actions Report.
The following procedure describes how to generate the Per-Call IVR Actions
Report.
User’s Guide 363

Procedure:
Generating the Per-Call IVR Actions Report
Purpose: To generate the Per-Call IVR Actions Report using Genesys

Administrator.
Summary
The Per-Call IVR Actions Report displays Media Control Platform data only.
Start of procedure
page 306) or a predefined range and the Granularity (on page 306).
3. On the Component tab, select the Media Control Platform for which you
require data.
4. Click Generate Report.
5. In the IVR Action column, click the arrows in the row for which you
require data.
The reporting data is displayed.
End of procedure

Figure 32: Per-Call IVR Actions Report
The report also provides usage information in the CDR. See Figure 33.
Figure 33: Per-Call IVR Actions Report —Usage Metrics
Table 63 describes the fields for the Per-Call IVR Actions Report.
Table 63: Per-Call IVR Actions Report Fields
Field Description
ASR If ASR is used at any point during the call.
TTS If TTS is used at any point during the call.
User’s Guide 365

Table 63: Per-Call IVR Actions Report Fields (Continued)
Field Description
LOCALREC If a local recording was executed during the call.
MSRREC If a media stream replication recording was executed

during the call.
CONF If conferencing was established during the call.
BRIDGING If bridging was established during the call.
VIDEO If a video connection was established during the call.
CODEC If any transcoding was used for this call (any leg).
VOICEXML If VoiceXML was used during the call.
NATIVECPA If native media server CPD/CPA was used during the call.
GATEWAYCPA If gateway-based CPD/CPA was used during this call.
MSPLAY If a request for MSML <play> was made during the call.
MSCOLLECT If a request for MSML <collect> or <dtmf> was made

during the call.

Chapter
20 Service Quality Reports

This chapter describes the available reports Service Quality data. It contains
the following sections:

Overview, page 367

SQ Call Failures, page 367
 SQ Failure Summary, page 370

SQ Latency Summary, page 372
Overview
The Service Quality tool measures system performance, based on service
quality metrics that impact the caller experience, and includes alarm generation
and detailed reporting. The Service Quality reports display statistics of service
quality metrics and time measurement for specific system tasks. Service
Quality data is for Media Control Platform components only.
There are performance implications to changing the batch size. For more
information, see “Configuring the Reporting Server” on page 271.
SQ Call Failures
Failed Calls A single call can either be a success or a failure from the perspective of Service
Quality, and GVP uses this type of failure to calculate SQ % (the percentage of
calls that are successful). Consider this type, a failed call.
Quality Failures Regardless of whether a single call is a failure or a success, multiple non-fatal
failures may occur during that single call. Consider these quality failures. GVP
uses quality failures to track total failures (a number which can exceed the total
number of calls).
• Failure time represents the exact time that a failure occurred.
User’s Guide 367

Chapter 20: Service Quality Reports SQ Call Failures
• The order column in an SQ Call Failures report represents the order in

which call failures (within a single call) occurred. If a specific call has 10
call failures, then each failure will have an order value somewhere between
1 and 10. Therefore, order is the relationship between multiple failures
within a single call.
The SQ Call Failures report (see Figure 34) displays the list of calls that have
failed service quality.
The following procedure describes how to generate the SQ Call Failures
report.
Procedure:
Generating the SQ Call Failures Report with GA
Purpose: To generate the SQ Call Failures report by using Genesys

Administrator.
Start of procedure:
Report Using Genesys Administrator, on page 294.
a. On the Time Range tab or in the corresponding start/end fields, select
the appropriate Time Range (on page 306).
b. On the Call Info tab, if required select the appropriate Call Failure
Type (on page 308).
c. On the IVR Profile tab, if required specify the IVR Profiles (on
page 307) to include in the report. To modify the list of IVR Profiles,
do any of the following:
list.
d. On the Components tab, if required select the Component Type (on
page 307).To build a list of Components (on page 307) for which SQ
failures are to be reported on:

Chapter 20: Service Quality Reports SQ Call Failures
Note: You do not need to select Call Failure Type, IVR Profiles, or
Components in order to retrieve data.

Click the specific Session ID link to display the historical call data for that
record.
End of procedure
.
Figure 34: SQ Call Failures Report
Table 64 describes the fields for the SQ Call Failures report.
Table 64: SQ Call Failures Report Fields
Field Description
Start Date The start time of the call that failed to meet a defined
minimum latency standard (service quality).
Session ID The Session ID of the call that failed to meet a defined

Component The Component of the call that failed to meet a defined

IVR Profile The IVR Profile of the call that failed to meet a defined
Duration The duration of the call that failed to meet a defined

Order If there is more than one failure for the given call, this is
the chronological order of the failure, starting with the
number 1.
Failure Time The time the call that failed to meet a defined minimum
latency standard (service quality).
User’s Guide 369

Chapter 20: Service Quality Reports SQ Failure Summary
Table 64: SQ Call Failures Report Fields (Continued)
Field Description
Failure Type The reason for the call that failed to meet a defined
SQ Failure Summary
The SQ Failure Summary report (see Figure 35) provides a graphical display
of calls that failed to meet a defined minimum latency standard for sessions that
have ended in each service quality period.
The following procedure describes how to generate the SQ Failure Summary
report.
Procedure:
Generating the SQ Failure Summary Report
Purpose: To generate the SQ Failure Summary report by using Genesys

Administrator.
Start of procedure:
page 306), and granularity
b. On the IVR Profile tab, specify the IVR Profiles (on page 307) to
following:
list.
c. On the Components tab, build a list of Components (on page 307) for
which service quality failures are to be reported on:

Chapter 20: Service Quality Reports SQ Failure Summary
d. On the Tenants tab, build a list of tenants (on page 307) for which
service quality failures are to be reported on:
Select the hyperlinked value in any column to view a detailed histogram of
that column.
End of procedure
Figure 35: SQ Failure Summary Report
Table 65 describes the fields for the SQ Failure Summary report.
Table 65: SQ Failure Summary Report Fields
Field Description
IVR Profile The name of the IVR Profile.
Component The name of the Component.
Failed The total number of calls received that failed a threshold.
Total The total number of calls received.
SQ The service quality percentage.
User’s Guide 371

Chapter 20: Service Quality Reports SQ Latency Summary
Table 65: SQ Failure Summary Report Fields (Continued)
Field Description
Call answer The total number of calls answered.
Call reject The total number of calls rejected.
In. 1st prompt The total number of calls that failed in the first inbound
prompt threshold.
Out. 1st prompt The total number of calls that failed in the first outbound
prompt threshold.
Inter-prompt The total number of calls that failed in the inter prompt
threshold.
Init resp The total number of calls that failed at the initial response
threshold.
Audio gap The total number of calls that failed in the audio gap
threshold.
App.error The total number of calls that failed because of an

application error.
Sys. error The total number of calls that failed because of a system
error.
SQ Latency Summary
The SQ Latency Summary report (see Figure 36) displays the number of calls
that fell below a threshold for sessions that have ended in each service quality
period. The latency thresholds are configured in the Media Control Platform
application under the ems section. For more information on these parameters,
see “Service Quality Analysis (SQA)” on page 61.
The following procedure describes how to generate the SQ Latency Summary
report.
Procedure:
Generating the SQ Latency Summary Report
Purpose: To generate the SQ Latency Summary report by using Genesys

Administrator.

Start of procedure:
page 306), and granularity.
b. On the Components tab, select the Component (on page 307).
Note: Time range and component are mandatory parameters.

Click the double arrow (>>) in each column heading to see the number of
latencies per type.
Select the hyperlinked number to display a histogram of latency threshold
for each latency type.
For information on near-real-time data for SQ Latency statistics, see “SQ
Latency Dashboard” on page 319.
End of procedure
Figure 36: SQ Latency Summary Report
Table 66 describes the fields for the SQ Latency Summary report.

L
Table 66: SQ Latency Summary Report Fields
Field Description
Start of Period The start time of the reporting period.
User’s Guide 373

Table 66: SQ Latency Summary Report Fields (Continued)
Field Description
Call Control The number of calls that fell below the Call Control
threshold. The Call Control thresholds are:
• CALL_ANSWER
• CALL_REJECT
Prompt The number of calls the fell below the Prompt threshold.
The Prompt thresholds are:
• INBOUND_FIRST_PRMOPT
• OUTBOUND_FIRST_PROMPT
• INTERPROMPT
• INITIAL_RESPONSE
Response The number of calls that fell below the Response

threshold. The Response thresholds are:
• CUMULATIVE_RESPONSE
• DTMF_INPUT_RESPONSE
• ASR_INPUT_RESPONSE
• NOINPUT_RESPONSE
• RECORDING_RESPONSE
• TRANSFER_RESPONSE
Fetching The number of calls that fell below the Fetching threshold.
The Fetching thresholds are:
• PAGE_FETCH
• AUDIO_FETCH
• GRAMMAR_FETCH
• DATA_FETCH
• JAVA_SCRIPT_FETCH
Execution The number of delayed calls because of the following

execution reasons:
• PAGE_COMPILE
• JAVA_SCRIPT_EXECUTION

Table 66: SQ Latency Summary Report Fields (Continued)
Field Description
MRCP The number of delayed calls because of the following

MRCP reasons:
• MRCP_ASR_SESSION_ESTABLISH
• MRCP_TTS_SESSION_ESTABLISH
• MRCP_ASR_SET_PARAMS
• MRCP_TTS_SET_PARAMS
• MRCP_ASR_STOP
• MRCP_TTS_STOP
• MRCP_DEFINE_GRAMMAR
• MRCP_RECOGNIZE
• MRCP_SPEAK

The number next to the chart is the average latency.

The light blue bar marks the minimum and maximum latency readings.
 The dashed line marks the average.

The dark blue area around the average marks the standard deviation (1
standard to each side of the dashed line).

The solid short line marks the nth percentile based on the Reporting
Server latency parameters (if an estimate exists).
 Hovering over the chart displays a tooltip that shows the numeric
values depicted in the chart.
User’s Guide 375


Chapter
21 Voice Application Reports

This chapter describes the Voice Application Reports. It contains the following
sections:

Overview, page 377

VAR Call Completion Summary, page 377
 VAR IVR Action Summary, page 380

VAR Last IVR Action, page 382
Overview
The Voice Application Reports display the usability data for applications that
have been divided into logical transactions using the VoiceXML <log> tag.
For more information on Genesys Voice Platform specific log extensions, see
the Genesys Voice Platform 8.x Genesys VoiceXML 2.1 Help file.
VAR Call Completion Summary

The VAR Call Completion Summary report (see Figure 37) displays the
relative frequency with which calls to a given VoiceXML application end in
different states.
The following procedure describes how to generate the VAR Call Completion
Summary report.
User’s Guide 377

Chapter 21: Voice Application Reports VAR Call Completion Summary
Procedure:
Generating the VAR Call Completion Summary Report
with GA
Purpose: To generate the VAR Call Completion Summary report using

Start of procedure:
b. On the IVR Profile tab, select the IVR Profile (on page 307).
c. On the Tenants tab, select the Tenant (on page 307).
End of procedure
To view the matching VAR IVR Action Usage data, select the VAR IVR Action
Usage link from the Related Reports section of the Tasks panel.
Note: The VAR Call Completion Summary report focuses on VoiceXML

applications using VAR; therefore, it displays MCP data only.

Chapter 21: Voice Application Reports VAR Call Completion Summary
Figure 37: VAR Call Completion Summary Report
Table 67 describes the fields for the VAR Call Completion Summary report.
Table 67: Call Completion Summary Report Fields
Field Description
Call Completion The date and time (in yyyy-mm-dd hh:mm:ss format)
Time when the call finished.
End State The end state of the call. Valid states are:
• Application End—The application hung up.
• Unknown—The MCP did not log an end state.
User’s Guide 379

Chapter 21: Voice Application Reports VAR IVR Action Summary
Table 67: Call Completion Summary Report Fields (Continued)
Field Description
End Result The end result of the call, as reported by the application.
Valid results are:
• Success
• Failed
• Unknown—The call end result is not specified by the
application that is using the VAR <log> interface.
For more information on VoiceXML <log> extensions,
see “VoiceXML <log> Extensions” on page 497.
Total Calls The total number of calls that ended for the time duration
(granularity) that is selected.
% Calls The percentage of calls that ended with the particular

combination of result and reason.
Avg. Call Len. (sec) The average length, in seconds, of the call.
VAR IVR Action Summary

The VAR IVR Action Summary report (see Figure 38) displays statistics on
individual IVR Actions that are within the <log> tag in a VoiceXML
application. For more information on the <log> extension, see “VAR Metrics”
on page 450.
The following procedure describes how to generate the VAR IVR Action
Summary report.
Procedure:
Generating the VAR IVR Action Summary Report
Purpose: To generate the VAR IVR Action Summary report using Genesys
Administrator.
Start of procedure:

Chapter 21: Voice Application Reports VAR IVR Action Summary
b. On the IVR Profile tab, specify the IVR Profiles (on page 307) to
following:
list.
c. On the Tenants tab, select the Tenant (on page 307).To build a list of
End of procedure
To view the matching Call Completion Summary data, select the VAR Call
Completion link from the Related Reports section of the Tasks panel
Note: The VAR IVR Action Summary report focuses on VoiceXML

applications using VAR; therefore, it displays MCP data only.
Figure 38: VAR IVR Action Summary Report
Table 68 describes the fields of the VAR IVR Action Summary report.
User’s Guide 381

Chapter 21: Voice Application Reports VAR Last IVR Action
Table 68: VAR IVR Action Summary Report Fields
Field Description
IVR Profile The name of the IVR Profile for which these actions
occurred.
IVR Action The name of the IVR action.
Usage Count The number of times that the IVR action was used.
% Actions The percentage of actions that were successful.

successful
Calls The number of calls that used this IVR action at least
once.
% Calls The percentage of total calls that used this IVR action at
least once.
VAR Last IVR Action

The VAR Last IVR Action report (see Figure 39) displays the details of the last
IVR Actions that were used during the end of a call.
The following procedure describes how to generate the VAR Last IVR Action
report.
Procedure:
Generating the VAR Last IVR Action Report
Purpose: To generate the VAR Last IVR Action report using Genesys
Administrator.
Start of procedure:
b. On the IVR Profile tab, select the IVR Profile (on page 307).
c. On the Tenants tab, select the Tenant (on page 307).

End of procedure
Note: The VAR Last IVR Action report displays MCP data only.
Figure 39: VAR Last IVR Action Report
Table 69 describes the fields for the VAR Last IVR Action report.
Table 69: Last IVR Action Used Report Fields
Field Description
IVR Action The name of the IVR Action.
Last Used in Calls The total number of calls for which the given IVR Action
was the last action that was executed.
% of Calls The percentage of the total number of calls in which this

IVR Action was used.
User’s Guide 383

Table 69: Last IVR Action Used Report Fields (Continued)
Field Description
% Success The percentage of successful calls in which this IVR

Action was used.
% Failed The percentage of failed calls in which this IVR Action

was used.
% Unknown The percentage of calls that used the Last IVR Action in
which the call end result was not specified by the
application that is using the VAR <log> interface.

Part
3 Appendixes
This part of the Guide contains miscellaneous reference information in the
following appendixes:
• Appendix A, “Module and Specifier IDs,” on page 387
• Appendix B, “Media Control Platform Reference Information,” on
page 435
• Appendix C, “Tuning Call Progress Detection,” on page 455
• Appendix D, “SIP Response Codes,” on page 465
• Appendix E, “Device Profiles,” on page 475
• Appendix F, “VAR API,” on page 495
• Appendix G, “Video Support,” on page 503
• Appendix H, “Custom Log Sinks,” on page 509
• Appendix I, “SSG HTTP Interface,” on page 515
• Appendix J, “Network Partitioning Configuration Options,” on page 555
• Appendix K, “SIP Customizable Headers and Parameters,” on page 559
User’s Guide 385

:

Appendix
A Module and Specifier IDs

This appendix lists various internal Genesys Voice Platform (GVP) identifiers
that are required for advanced configuration of EMS Logging and Reporting.
This appendix contains the following sections:
 Media Control Platform, page 387

Call Control Platform, page 411

Resource Manager, page 417
 CTI Connector, page 422

Supplementary Services Gateway, page 426

PSTN Connector, page 428
 Fetching Module, page 432
For detailed information about the metrics (application-level logs) that the
Media Control Platform (MCP) and the Call Control Platform (CCP) generate,
including metric IDs and descriptions, see the Genesys Voice Platform 8.5
Metrics Reference.

Table 70 lists the Media Control Platform Application Module names and IDs.
For the Next Generation Interpreter (NGI), see “Next Generation Interpreter
Module ID and Specifiers” on page 408.
Table 70: Media Control Platform Application Module Names and IDs
Module Name Description or Comment Module ID Specifiers (link)
MTMPC Media Processing Component (MPC) wrapper 47 MTMPC
LMBase Base Line Manager 21 LMBase
LMSIP2 SIP Line Manager 40 LMSIP2
User’s Guide 387

Appendix A: Module and Specifier IDs Media Control Platform
Table 70: Media Control Platform Application Module Names and IDs (Continued)
SESSMGR Call Manager API 28 SESSMGR
CALLSESSION 29 None
SMMAIN Main module in the Media Control Platform 31 SMMAIN
CMUTIL Media Control Platform utility components 33 CMUTIL
APPMODULE Base Application Module 34 APPMODULE
REMDIAL Remote Dial Remdial Application Module 38 REMDIAL
CONFERENCE Conference Application Module 41 CONFERENCE
SQA Any and all SQA logs 43 SQA
MEDIAMGR The Media Manager part of the MPC 176 MEDIAMGR
CONTROL The control layer part of the MPC 177 CONTROL
MEDIA The media layer part of the MPC 178 MEDIA
RTP_INTERFACE The RTP layer of the MPC 179 RTP_INTERFACE
DSP The DSP components 180 DSP
VGULOGMOD_ The main Utility 128 VGULOGMOD_

MAIN MAIN
MTINTERNAL The Internal Media Transport application 130 MTINTERNAL

module
RTSPSTACK The RTSP stack 132 RTSPSTACK
MSML MSML Implementation 27 MSML
NETANN NetAnn Implementation 39 NETANN
LMBASE lmbase 21 LMBASE
VRMMGR 37 None
ADAPTOR MRCP Client Adaptor 106 ADAPTOR
CLIENT MRCP V1 Client 97 VRMCLIENT
MRCPV1STACK MRCP V1 Stack 96 MRCPV1STACK
V2_CLIENT MRCP V2 Client & VRM Recorder 107 MRCPV2CLIENT

Table 70: Media Control Platform Application Module Names and IDs (Continued)
MRCPV2STACK MRCP V2 Stack 105 MRCPV2STACK
DTMFRECO DTMF Recognizer 58 DTMFRECO
Table 71 lists the Media Control Platform specifier names and IDs.
Table 71: Media Control Platform Specifier Names and IDs
Specifier ID Specifier Name
MTMPC
1001 CMLOGMOD_MTMPC_INITFAILED
2001 CMLOGMOD_MTMPC_CONNERROR
3001 CMLOGMOD_MTMPC_ROUTETOOLONG
LMBase
1001 CMLOGMOD_LMBASE_IDGENDIRUNACCBLE
1003 CMLOGMOD_LMBASE_SYSIPNOTRETRVABLE
1004 CMLOGMOD_LMBASE_FAILUPDTEOPENCALLIDFILE
1005 CMLOGMOD_LMBASE_NOTPUTSEQNUMTOCALLIDFILE
2001 CMLOGMOD_LMBASE_RESETCALLIDFILECONTNTINVD
3001 CMLOGMOD_LMBASE_NOMEDIASESSPLAYAUDIO
3002 CMLOGMOD_LMBASE_NOMEDIASESSPLAYDTMF
3004 CMLOGMOD_LMBASE_NOMEDIASESSRECRDAUDIO
3005 CMLOGMOD_LMBASE_NOMEDIASESSSTREAMING
LMSIP2
2001 CMLOGMOD_LMSIP2_RECVUNEXPCTACK
2002 CMLOGMOD_LMSIP2_MEDIAERROR
2003 CMLOGMOD_LMSIP2_ERRSNDINVRESPONSE
2004 CMLOGMOD_LMSIP2_REGISTERTIMEOUT
2005 CMLOGMOD_LMSIP2_REGISTERBADREQUEST
User’s Guide 389

Table 71: Media Control Platform Specifier Names and IDs (Continued)
2006 CMLOGMOD_LMSIP2_REGISTERFORBIDDEN
2007 CMLOGMOD_LMSIP2_REGISTERNOTFOUND
2008 CMLOGMOD_LMSIP2_REGISTERNOTACCEPTABLE
2009 CMLOGMOD_LMSIP2_REGISTEROTHERERROR
2010 CMLOGMOD_LMSIP2_VGSIPERRORNOTIFY
2011 CMLOGMOD_LMSIP2_ERRPARSESDPCONTENT
2012 CMLOGMOD_LMSIP2_REGISTERALGONOTSUPPORTED
2013 CMLOGMOD_LMSIP2_REGISTERAUTHENTICATIONERROR
2014 CMLOGMOD_LMSIP2_NONMATCHINGSIPINFO
2015 CMLOGMOD_LMSIP2_CUSTOMPARAMERROR
3001 CMLOGMOD_LMSIP2_CANTACCEPTNONINVITECALL
3002 CMLOGMOD_LMSIP2_ERRSNDINVITE
3003 CMLOGMOD_LMSIP2_ERRCREATERTPSESS
3004 CMLOGMOD_LMSIP2_ERRCREATEPSTNSESS
3005 CMLOGMOD_LMSIP2_BADDYNAMICPAYLOAD
3006 CMLOGMOD_LMSIP2_BADDTMFRECV
3007 CMLOGMOD_LMSIP2_ZEROCLOCKRATE
4001 CMLOGMOD_LMSIP2_MESSAGE
4002 CMLOGMOD_LMSIP2_PROCDELAY
SESSMGR
1001 CMLOGMOD_SESSMGR_IDGENDIRUNACCBLE
1003 CMLOGMOD_SESSMGR_SYSIPNOTRETRVABLE
1004 CMLOGMOD_SESSMGR_FAILUPDTEOPENCALLIDFILE
1005 CMLOGMOD_SESSMGR_NOTPUTSEQNUMTOCALLIDFILE
1007 CMLOGMOD_SESSMGR_VRMINITFAIL

1008 CMLOGMOD_SESSMGR_CANTINITLICENSEMGR
2001 CMLOGMOD_SESSMGR_ATTEMPTAUDIOCTRLWBARGEIN
2002 CMLOGMOD_SESSMGR_BADFRMTSCPTAUDIO
2003 CMLOGMOD_SESSMGR_BADFRMTSCPTTTS
2004 CMLOGMOD_SESSMGR_BADFRMTSCPTSTRMNG
2016 CMLOGMOD_SESSMGR_OUTCALLNORESOURCE
2017 CMLOGMOD_SESSMGR_TTSMGRLOST
2018 CMLOGMOD_SESSMGR_TRFRTODESTNOTAUTH
2019 CMLOGMOD_SESSMGR_DESTURINOTSUPP
2020 CMLOGMOD_SESSMGR_DESTURIMALFORMED
2021 CMLOGMOD_SESSMGR_STRMMODUNEXPTEVENT
2022 CMLOGMOD_SESSMGR_LOSTASRMGR
2023 CMLOGMOD_SESSMGR_INITCALLSESSWNOLNMGR
2024 CMLOGMOD_SESSMGR_RESETCALLIDFILECONTNTINVD
2026 CMLOGMOD_SESSMGR_ISDNCAUSECODEERR
3001 CMLOGMOD_SESSMGR_UNEXPECTTTSERROR
3002 CMLOGMOD_SESSMGR_EXPIREASRTTSIGNORED
3003 CMLOGMOD_SESSMGR_UNEXPECTCMCALLBILLEVENT
3005 CMLOGMOD_SESSMGR_FAILMEDSTRMRESLT
3006 CMLOGMOD_SESSMGR_APPMODULENOTFOUND
3007 CMLOGMOD_SESSMGR_UNABLETOSENDLOGTOASR
3008 CMLOGMOD_SESSMGR_INVALIDVRMMESSAGE
4001 CMLOGMOD_SESSMGR_INBOUNDDTMF
4003 CMLOGMOD_SESSMGR_NOINOUTLINES
User’s Guide 391

SMMAIN
1001 CMLOGMOD_SMMAIN_VRMDLLLOADFAIL
1002 CMLOGMOD_SMMAIN_VRMSETLOGFAIL
1003 CMLOGMOD_SMMAIN_MAKEVRMFAIL
1004 CMLOGMOD_SMMAIN_CREATEVRMFAIL
1006 CMLOGMOD_SMMAIN_CALLMGRCFGPARAMERR
1007 CMLOGMOD_SMMAIN_LOADTOOMANYCMGRMOD
1008 CMLOGMOD_SMMAIN_FAILCREATECMGRMOD
1009 CMLOGMOD_SMMAIN_LOADTOOMANYDEVICE
1010 CMLOGMOD_SMMAIN_FAILCREATEDEVICE
1011 CMLOGMOD_SMMAIN_FAILINITDEVICE
1012 CMLOGMOD_SMMAIN_LOADTOOMANYMEDTRPT
1013 CMLOGMOD_SMMAIN_FAILCREATEMEDTRPT
1014 CMLOGMOD_SMMAIN_FAILINITMEDTRPT
1015 CMLOGMOD_SMMAIN_LOADTOOMANYLNMGRS
1016 CMLOGMOD_SMMAIN_FAILCREATELNMGR
1017 CMLOGMOD_SMMAIN_FAILINITLNMGR
1018 CMLOGMOD_SMMAIN_SESSMGRAPPMODCFGERR
1019 CMLOGMOD_SMMAIN_LOADTOOMANYAPPMOD
1020 CMLOGMOD_SMMAIN_SESSMGRMODCFGERR
1021 CMLOGMOD_SMMAIN_LOADTOOMANYSESSMOD
1022 CMLOGMOD_SMMAIN_FAILOPENLICENSE
1023 CMLOGMOD_SMMAIN_FAILPARSELICENSE
1024 CMLOGMOD_SMMAIN_MACVALIDERR
1025 CMLOGMOD_SMMAIN_GENINITLICERR

1026 CMLOGMOD_SMMAIN_CANTCREATEVGNETLIB
1027 CMLOGMOD_SMMAIN_CANTINITVGNETLIB
1028 CMLOGMOD_SMMAIN_FAILINITCFGOBJ
1029 CMLOGMOD_SMMAIN_CANTSTARTCMGR
1030 CMLOGMOD_SSMAIN_NOTINBIN
2002 CMLOGMOD_SMMAIN_FAILLOADAPPMODLIB
2003 CMLOGMOD_SMMAIN_FAILINITAPPMOD
2004 CMLOGMOD_SMMAIN_NOVLDAPPMODINLIB
2005 CMLOGMOD_SMMAIN_LIBNODEFMAKEAPPMOD
2006 CMLOGMOD_SMMAIN_VXMLAPPMODNOTLOAD
3001 CMLOGMOD_SMMAIN_FAILSETFDLIMIT
4001 CMLOGMOD_SMMAIN_MCP_STARTED
4002 CMLOGMOD_SMMAIN_MCP_STOPPED
CMUTIL
2001 CMLOGMOD_CMUTIL_TELNUMLONG
2002 CMLOGMOD_CMUTIL_TELNUMINVCHAR
2003 CMLOGMOD_CMUTIL_POSTDIALLONG
2004 CMLOGMOD_CMUTIL_POSTDIALINVCHAR
2005 CMLOGMOD_CMUTIL_CONFLICTEXT
2006 CMLOGMOD_CMUTIL_HUNTGPINVTRUNK
3001 CMLOGMOD_CMUTIL_HUNTGPNONEXISTTRUNK
3002 CMLOGMOD_CMUTIL_CALLREQNONEXISTHUNTGP
3003 CMLOGMOD_CMUTIL_WAITFORDIAL
3004 CMLOGMOD_CMUTIL_ATTRIBLONG
3005 CMLOGMOD_CMUTIL_VALUELONG
User’s Guide 393

APPMODULE
1001 CMLOGMOD_APPMODULE_FAILSTRTWORKNGTHRD
2001 CMLOGMOD_APPMODULE_FAILREGAPP
2002 CMLOGMOD_APPMODULE_FAILREGAPPMOD
3001 CMLOGMOD_APPMODULE_FAILBINDAPP
REMDIAL
2001 CMLOGMOD_REMDIAL_FAILREGREMDLMOD
2002 CMLOGMOD_REMDIAL_CANTCREATESERVERSOCK
2003 CMLOGMOD_REMDIAL_SOCKETERROR
3001 CMLOGMOD_REMDIAL_MAXCALLSWARN
3002 CMLOGMOD_REMDIAL_MAXCLIENTS
3003 CMLOGMOD_REMDIAL_NOACTIVESESS
3004 CMLOGMOD_REMDIAL_MAXCALLSREACHED
CONFERENCE
2001 CMLOGMOD_CONFERENCE_FAILED
2002 CMLOGMOD_CONFERENCE_UNEXPTREASON
4001 CMLOGMOD_CONFERENCE_ESTABLISHED
4002 CMLOGMOD_CONFERENCE_TERMINATED
SQA
4001 CMLOGMOD_SQA_DTMF
4002 CMLOGMOD_SQA_TRANSFERSTART
4003 CMLOGMOD_SQA_TRANSFEREND
4004 CMLOGMOD_SQA_PROMPTTYPE
4006 CMLOGMOD_SQA_RECOGNITIONSTART
4007 CMLOGMOD_SQA_RECOGNITIONEND

4008 CMLOGMOD_SQA_OPENRECORDFILE
4009 CMLOGMOD_SQA_ClOSERECORDFILE
4010 CMLOGMOD_SQA_MEDIAROUTING
4011 CMLOGMOD_SQA_AUDIOGAP
4012 CMLOGMOD_SQA_FIRSTAUDIOPK
4013 CMLOGMOD_SQA_LASTAUDIOPK
427820 CMLOGMOD_SQA_ECMASCRIPT_TIMINGS
427801 CMLOGMOD_SQA_COMPILE_TIME
427802 CMLOGMOD_SQA_FETCH_TIME
MEDIAMGR
2001 MPCLOGMOD_MEDIAMGR_INVALIDMEDIA
2002 MPCLOGMOD_MEDIAMGR_UNEXPECTEDRTSPDISC
2003 MPCLOGMOD_MEDIAMGR_RTSPREQFAIL
2004 MPCLOGMOD_MEDIAMGR_RTSPREPLYERROR
2005 MPCLOGMOD_MEDIAMGR_RTSPRTPERROR
2006 MPCLOGMOD_MEDIAMGR_UNSUPPORTEDVIDFMT
2008 MPCLOGMOD_MEDIAMGR_UNSUPPORTEDAUDCHNLS
2009 MPCLOGMOD_MEDIAMGR_BADAVICHNKSIZE
2010 MPCLOGMOD_MEDIAMGR_MALFORMEDAVIHDR
2011 MPCLOGMOD_MEDIAMGR_RECBUFFISOTOOSMALL
2012 MPCLOGMOD_MEDIAMGR_UNABLETOALLOCMEM
2013 MPCLOGMOD_MEDIAMGR_NOISOTRAK
2014 MPCLOGMOD_MEDIAMGR_BADISOBOXSIZE
2016 MPCLOGMOD_MEDIAMGR_BRANDINCOMPT3GPP
2017 MPCLOGMOD_MEDIAMGR_BADMAJ3GPPBRAND
User’s Guide 395

2018 MPCLOGMOD_MEDIAMGR_ERORISOBOXVALUE
2019 MPCLOGMOD_MEDIAMGR_FAILTOSTARTRECORD
2020 MPCLOGMOD_MEDIAMGR_NOMEDIAINFOOBJECT
3001 MPCLOGMOD_MEDIAMGR_RECFRAMEDISCARD
3002 MPCLOGMOD_MEDIAMGR_UNEXPECTEDRTSPREPLY
3003 MPCLOGMOD_MEDIAMGR_BADISOBOXVALUE
3004 MPCLOGMOD_MEDIAMGR_BADISOBOXTYPE
3005 MPCLOGMOD_MEDIAMGR_MANDISOBOXMISS
3006 MPCLOGMOD_MEDIAMGR_BUFFTOOSMALLTOPARSEISOHDR
3007 MPCLOGMOD_MEDIAMGR_UNSUPPORTEDAUDRATE
3008 MPCLOGMOD_MEDIAMGR_UNSUPPORTEDVIDRATE
CONTROL
1001 MPCLOGMOD_CONTROL_INITVGMEDIAINFOFAILED
1002 MPCLOGMOD_CONTROL_INITDSPCAPFAILED
2001 MPCLOGMOD_CONTROL_INVALIDHRTIMERRES
2002 MPCLOGMOD_CONTROL_SDPPARSEFAILED
3001 MPCLOGMOD_CONTROL_INVALIDCFG
3002 MPCLOGMOD_CONTROL_CONNINITFAILED
3003 MPCLOGMOD_CONTROL_CONNMODIFYFAILED
3004 MPCLOGMOD_CONTROL_SENDDTMFNOTALLOWED
3005 MPCLOGMOD_CONTROL_INVALIDCONFIGPARAM
3006 MPCLOGMOD_CONTROL_EVENTPOOLTHRESHOLDREACHED
3007 MPCLOGMOD_CONTROL_EVENTPOOLTHRESHOLDLOWERED
4001 MPCLOGMOD_CONTROL_DIRECTBRIDGE

MEDIA
2001 MPCLOGMOD_MEDIA_RECORDOPENFAILED
2002 MPCLOGMOD_MEDIA_PLAYBACKOPENFAILED
2003 MPCLOGMOD_MEDIA_TRANSCODINGSUPPRESSED
3001 MPCLOGMOD_MEDIA_ACCESSFAILED
3002 MPCLOGMOD_MEDIA_SINKBUFFERFULL
3003 MPCLOGMOD_MEDIA_SOURCEBUFFERFULL
3004 MPCLOGMOD_MEDIA_PACKETBUFFERFULL
3005 MPCLOGMOD_MEDIA_RTPPACKETTOOLARGE
3006 MPCLOGMOD_MEDIA_BUFFERTOOSMALL
3007 MPCLOGMOD_MEDIA_BRIDGEOBJECTNOTFOUND
3008 MPCLOGMOD_MEDIA_H263SORTEROUTOFPACKET
3009 MPCLOGMOD_MEDIA_SILENCEFILLDISABLED
3010 MPCLOGMOD_MEDIA_SENDDTMFDISABLED
3011 MPCLOGMOD_MEDIA_NORTPSTREAMSENDDTMF
3012 MPCLOGMOD_MEDIA_NORTPSTREAMMEDIATRANSMIT
3013 MPCLOGMOD_MEDIA_ERRORDECODINGRFC2833
RTP_INTERFACE
3001 MPCLOGMOD_RTPIF_INCORRECTTIMEINDEX
3002 MPCLOGMOD_RTPIF_OUTOFSEQUENCEINCOMINGRTP
3003 MPCLOGMOD_RTPIF_INCOMINGRTPDELAY
3004 MPCLOGMOD_RTPIF_ERRORDEFRAMINGPACKET
3005 MPCLOGMOD_RTPIF_UNEXPECTEDPAYLOADTYPE
3006 MPCLOGMOD_RTPIF_ERRORCRYPTOSRTP
3007 MPCLOGMOD_RTPIF_TXRTCPAPPPKTFAIL
User’s Guide 397

3008 MPCLOGMOD_RTPIF_TXRTCPAPPPKTDELAY
DSP
3002 MPCLOGMOD_DSP_NOTRANSCODER
3003 MPCLOGMOD_DSP_CODEC_UNSUPPORTED
4001 MPCLOGMOD_DSP_VIDEOTRANSCODE_START
VGULOGMOD_MAIN
1002 VGLOG_CANT_OPEN_DLL
2003 VGLOG_SOCKET_SEND_FAILED
3003 MPCLOGMOD_DSP_CODEC_UNSUPPORTED
4001 MPCLOGMOD_DSP_VIDEOTRANSCODE_START
7001 VGLOG_TRACE_GENERIC
MTINTERNAL
2001 VGLOG_MTINTERNAL_MINORMAXPORT
2002 VGLOG_MTINTERNAL_MINLARGERTHANMAX
3001 VGLOG_MTINTERNAL_OPENFILEERROR
3002 VGLOG_MTINTERNAL_SENDDATAERROR
3003 VGLOG_MTINTERNAL_WRITEFILEERROR
3004 VGLOG_MTINTERNAL_DISCARDRTPPACKET
RTSPSTACK
2001 VGLOG_RTSP_NEW_FAILED
2002 VGLOG_RTSP_INVALID_CONFIG
2003 VGLOG_RTSP_UNINIT
2004 VGLOG_RTSP_CONSTRUCT_BAD_MSG
2005 VGLOG_RTSP_PARSE_BAD_MSG
2006 VGLOG_RTSP_SOCKET_ERROR

3001 VGLOG_RTSP_SOCKET_EVENT
5001 VGLOG_RTSP_SOCKET_CLOSE
MSML
3001 CMLOGMOD_MSML_CONFIGWARNING
NETANN
2001 CMLOGMOD_NETANN_ERRORPLAYINGMEDIA
3001 CMLOGMOD_NETANN_INVALIDPARAM
3002 CMLOGMOD_NETANN_JOINCONFERENCEFAILED
LMBASE
1001 CMLOGMOD_LMBASE_IDGENDIRUNACCBLE
1003 CMLOGMOD_LMBASE_SYSIPNOTRETRVABLE
1004 CMLOGMOD_LMBASE_FAILUPDTEOPENCALLIDFILE
1005 CMLOGMOD_LMBASE_NOTPUTSEQNUMTOCALLIDFILE
2001 CMLOGMOD_LMBASE_RESETCALLIDFILECONTNTINVD
3001 CMLOGMOD_LMBASE_NOMEDIASESSPLAYAUDIO
3002 CMLOGMOD_LMBASE_NOMEDIASESSPLAYDTMF
3004 CMLOGMOD_LMBASE_NOMEDIASESSRECRDAUDIO
3005 CMLOGMOD_LMBASE_NOMEDIASESSSTREAMING
VRMMGR
NULL
ADAPTOR
2001 VGLOG_CONFIGURATION_ERROR
2002 VGLOG_CLOSE_SESSION_FAIL
2003 VGLOG_STOP_FAIL
2004 VGLOG_LOG_FAIL
User’s Guide 399

2005 VGLOG_LOAD_GRAMMAR_FAIL
2006 VGLOG_ASR_SETPARAMS_FAIL
2007 VGLOG_ASR_RECOGNIZE_FAIL
2008 VGLOG_PROMPTDONE_FAIL
2009 VGLOG_ASR_INTERPRET_FAIL
2010 VGLOG_TTS_GETPARAMS_FAIL
2011 VGLOG_TTS_SETPARAMS_FAIL
2012 VGLOG_TTS_SPEAK_FAIL
2013 VGLOG_TTS_CONTROL_FAIL
2014 VGLOG_TTS_RESUME_FAIL
2015 VGLOG_TTS_PAUSE_FAIL
2016 VGLOG_TTS_BARGEIN_OCCURRED_FAIL
2017 VGLOG_OPEN_SESSION_FAIL
2018 VGLOG_UNKNOWN_MRCPPROTOCOL
2019 VGLOG_PROVISION_HANDLER_FAIL
2020 VGLOG_REDIRECT_FAIL
2021 VGLOG_DTMFINPUT_FAIL
2022 VGLOG_TRAP_ASR_SENDREQUEST_TIMEOUT
2023 VGLOG_TRAP_ASR_SENDREQUEST_FAILURE
2024 VGLOG_TRAP_ASR_RECEIVE_MRCPRESPONSEERR
2015 VGLOG_TRAP_ASR_RECEIVE_MRCPEVENTERR
2026 VGLOG_TRAP_ASR_RECEIVE_SERVERRESPONSEERR
2027 VGLOG_TRAP_TTS_SENDREQUEST_TIMEOUT
2028 VGLOG_TRAP_TTS_SENDREQUEST_FAILURE
2029 VGLOG_TRAP_TTS_RECEIVE_MRCPRESPONSEERR

2030 VGLOG_TRAP_TTS_RECEIVE_MRCPEVENTERR
2031 VGLOG_TRAP_TTS_RECEIVE_SERVERRESPONSEERR
VRMCLIENT
1001 VGLOG_INVALID_ENG_TYPE
1002 VGLOG_INVALID_ENG_URI
1003 VGLOG_INVALID_ENG_ENTRY
1004 VGLOG_INVALID_ENG_IP_PORT
1005 VGLOG_EMPTY_ENG_LIST
1006 VGLOG_ENG_PARSE_ERROR
1007 VGLOG_MISSING_ENG_TYPE_LIST
1008 VGLOG_INVALID_STACK
1009 VGLOG_ENG_TYPE_INIT_ERROR
1010 VGLOG_STACK_INIT_ERROR
1011 VGLOG_REQ_MGR_INIT_ERROR
1012 VGLOG_CONNECTION_MGR_INIT_ERROR
1013 VGLOG_STACK_HDLR_INIT_ERROR
1014 VGLOG_PROVISION_ERROR
1015 VGLOG_INIT_FAILURE
2001 VGLOG_FILE_STAT_ERROR
2002 VGLOG_GRAM_SIZE_ERROR
2003 VGLOG_GRAM_OPEN_ERROR
2004 VGLOG_GRAM_OFFSET_ERROR
2005 VGLOG_MEM_ALLOC_ERROR
2006 VGLOG_GRAM_READ_ERROR
2007 VGLOG_SERVER_CONNECT_ERROR
User’s Guide 401

2008 VGLOG_SERVER_INFO_ERROR
2009 VGLOG_INVALID_PARAM
2010 VGLOG_NO_GRAM_BASE
2011 VGLOG_PING_ERROR
2012 VGLOG_NO_RES_ID
2013 VGLOG_SESSION_STORAGE_ERROR
2014 VGLOG_CHANGE_STATE_ERROR
2015 VGLOG_INVALID_TIMER_EVENT
2016 VGLOG_SESSION_REMOVE_ERROR
2018 VGLOG_INVALID_MSG_ID
2019 VGLOG_UNKNOWN_TIMEOUT
2020 VGLOG_REQUEST_TYPE_FAILURE
2021 VGLOG_TIMER_REMOVE_ERROR
2022 VGLOG_RESPONSE_FAILURE
2023 VGLOG_REQUEST_REMOVE_ERROR
2024 VGLOG_INVALID_REQUEST
2025 VGLOG_SOCKET_DISCONNECT
2026 VGLOG_INVALID_AUDIO_CODEC
2027 VGLOG_SEND_REQUEST_ERROR
2028 VGLOG_STACK_SYSTEM_ERROR
2029 VGLOG_UNIMPLEMENTED_METHOD
2031 VGLOG_LOST_CONNECTION
3001 VGLOG_RECO_ERROR
3002 VGLOG_RECONNECT_SUCCESS
3003 VGLOG_INCORRECT_TTS_MSG_ORDER

3004 VGLOG_INCORRECT_NLSML_FORMAT
3005 VGLOG_ERROR_DECODE_FAILURE
3006 VGLOG_GRAMMAR_NOT_EXIST
3007 VGLOG_GRAMMAR_READING_ERROR
3008 VGLOG_HOTKEY_GRAMMAR_ERROR
MRCPV2CLIENT
1001 VGLOG_FAIL_LOADING_MRCP_MODULE
1002 VGLOG_INVALID_ENG_ENTRY
1003 VGLOG_STACK_INIT_ERROR
1004 VGLOG_REQ_GR_INIT_ERROR
1005 VGLOG_RESOURCE_MGR_INIT_ERROR
1006 VGLOG_FAILED_TO_OPENSESSION
2001 VGLOG_CONFIGURATION_ERROR
2002 VGLOG_CLOSE_SESSION_FAIL
2003 VGLOG_STOP_FAIL
2004 VGLOG_LOG_FAIL
2005 VGLOG_LOAD_GRAMMAR_FAIL
2006 VGLOG_ASR_SETPARAMS_FAIL
2007 VGLOG_ASR_RECOGNIZE_FAIL
2008 VGLOG_PROMPTDONE_FAIL
2009 VGLOG_ASR_INTERPRET_FAIL
2010 VGLOG_TTS_GETPARAMS_FAIL
2011 VGLOG_TTS_SETPARAMS_FAIL
2012 VGLOG_TTS_SPEAK_FAIL
2013 VGLOG_TTS_CONTROL_FAIL
User’s Guide 403

2014 VGLOG_TTS_RESUME_FAIL
2015 VGLOG_TTS_PAUSE_FAIL
2016 VGLOG_TTS_BARGEIN_OCCURRED_FAIL
2017 VGLOG_OPEN_SESSION_FAIL
2018 VGLOG_UNKNOWN_MRCPPROTOCOL
2019 VGLOG_PROVISION_HANDLER_FAIL
2020 VGLOG_REMOVE_SESSION_FAIL
2021 VGLOG_GET_SESSION_DATA_FAIL
2022 VGLOG_SOCKET_DISCONNECT
2023 VGLOG_STACK_SYSTEM_ERROR
2024 VGLOG_REQUEST_TYPE_FAILURE
2025 GVPLOG_NO_RES_ID
2026 VGLOG_CHANGE_STATE_ERROR
2027 VGLOG_SESSION_STORAGE_ERROR
2028 VGLOG_REQ_MGR_INIT_ERROR
2029 VGLOG_STACK_HDLR_INIT_ERROR
2030 VGLOG_NO_GRAMMAR_BASE
2031 VGLOG_SEND_REQUEST_ERROR
2032 VGLOG_MRCPV2_PARSE_BAD_MSG
2033 VGLOG_INITIALIZATION_FAIL
2034 VGLOG_SIPSEND_REQUEST_ERROR
2035 VGLOG_RECEIVE_RESPONSE_ERROR
2036 VGLOG_RESPONSE_FAILURE
2037 VGLOG_INVALID_REQUEST
2038 VGLOG_TIMER_REMOVE_ERROR

2039 VGLOG_OUTSTANDING_CONN_REMOVE_ERROR
2040 VGLOG_INVALID_MSG_ID
2041 VGLOG_REQUEST_REMOVE_ERROR
2042 VGLOG_SERVER_CONNECTION_ERROR
2043 VGLOG_FAIED_TO_GETRESOURCEINFO
2044 VGLOG_INVALID_AUDIO_CODEC
2045 VGLOG_SESSION_INITIATION_ERROR
2046 VGLOG_SESSION_REMOVE_ERROR
2047 VGLOG_INVALID_TIMER_EVENT
2048 VGLOG_UNKNOWN_TIMEOUT
2050 VGLOG_HOTKEY_GRAMMAR_ERROR
2051 VGLOG_FAIL_CREATE_SIP_USER_AGENT
2052 VGLOG_FAIL_TO_INITIATE_SIP_SESSION
2053 VGLOG_SIP_ERROR
3001 VGLOG_INCORRECT_NLSML_FORMAT
3002 VGLOG_INCORRECT_TTS_MSG_ORDER
3003 VGLOG_FAIL_TO_OPEN_FILE
4001 VGLOG_MRCPCLIENT_DEFAULT_ENGINE
4002 VGLOG_GET_LOCALIP_FAILED
4003 VGLOG_VRMCLIENT_TTSREQ
4004 VGLOG_VRMCLIENT_TTSRESP
4005 VGLOG_VRMLIENT_GET_LOCALIP_FAILE
4006 VGLOG_SETPARAM_ERROR
DTMFRECO
1000 DTMF_OUT_OF_MEMORY
User’s Guide 405

1001 FAILED_TO_INIT_DTMF_RECOGNIZER
2000 GRAM_ERROR
2001 INVALID_SESSION_ID
2002 FAILED_TO_ACCESS_GRAMMAR
2003 INVALID_STATE
2004 FAILED_TO_CREATE_SESSION
2005 GRAM_NUMBER_MISMATCH
2006 GRAMMAR_TYPE_ERROR
2007 GRAMMAR_DEFINE_ERROR
2008 EMPTY_GRAM_ID
2009 FAILED_TO_CREATE_JS
2010 GRAM_ERROR_EXCEEDED_MAX_TABLE_SIZE
2011 FAILED_TO_INIT_XML_CONVERTER
2012 SEMANTIC_INTERPRETATION_ERROR
2013 FAILED_TO_CREATE_DTMF_RECOG_THREAD
2014 FAILED_TO_START_DTMF_RECOG
2015 FAILED_TO_FETCH
2016 FAILED_RECOGNITION
2017 GRAM_SYNTAX_ERROR
2018 FAILED_TO_PARSE_GRAM
2019 FAILED_TO_GENERATE_NLSML
3000 GRAM_WARNING
3001 FAILED_TO_CACHE
3002 FAILED_TO_PROCESS_BUFFERED_INPUT
3003 FAILED_TO_CLEAR_BUFFERED_INPUT

3004 FAILED_TO_GENERATE_FLAT_PARSE_LIST
3005 DUPLICATED_RULES
3006 FAILED_TO_ACCESS_RULE_DURING_SI
3007 FAILED_TO_PROCESS_DTMF_INPUT
3008 FAILED_TO_STOP_DTMF_RECOG
3009 FAILED_TO_DELETE_DTMF_SESSION
3010 FAILED_TO_GET_BUFFERED_DTMF
3011 FAILED_TO_DELETE_NOINPUT_TIMER
3012 FAILED_TO_PROCESS_EVENT
3013 FAILED_TO_PROCESS_NOINPUT
3014 FAILED_TO_START_NOINPUT_TIMER
MRCPV1STACK
1001 VGLOG_SOME_MRCPV1_CRITICAL_ALARM
2001 VGLOG_MRCPV1_NEW_FAILED
2002 VGLOG_MRCPV1_INVALID_CONFIG
2003 VGLOG_MRCPV1_UNINIT
2004 VGLOG_MRCPV1_CONSTRUCT_BAD_MSG
2006 VGLOG_MRCPV1_BAD_REQUEST
MRCPV2STACK
2001 VGLOG_MRCPV2_NEW_FAILED
2002 VGLOG_MRCPV2_INVALID_CONFIG
2003 VGLOG_MRCPV2_UNINIT
2004 VGLOG_MRCPV2_CONSTRUCT_BAD_MSG
User’s Guide 407

2006 VGLOG_MRCPV2_BAD_REQUEST
21906 VGLOG_MRCPV2_SOCKET_ERROR
21908 VGLOG_MRCPV2_SOCKET_CLOSE
21909 VGLOG_MRCPV2_SOCKET_EVENT
21910 VGLOG_MRCPV2_INVALID_SESSIONID
21911 VGLOG_MRCPV2_INVALID_METHOD
Next Generation Interpreter Module ID and Specifiers

The Module ID for the NGI application is 192. Table 72 describes the
specifiers for the NGI application module.
Table 72: NGI Specifiers
Specifier ID Specifier Name Level
3501 NGI_LOG_JS_WARNING Warning
3502 NGI_LOG_JS_INFO Info
3503 NGI_LOG_NET_CONNECT_FAILURE Warning
3504 NGI_LOG_NET_CONNECT_FAILURE_INFO Info
3505 NGI_LOG_INITIALIZE_ERROR Warning
3506 NGI_LOG_CREATE_DIALOG_FAILURE Info
3507 NGI_LOG_CONFIGURATION Info
3508 NGI_LOG_INVALID_PROPERTY Info
3509 NGI_LOG_INVALID_SYNTAX Warning
3510 NGI_LOG_UNEXPECTED_WARNING Warning
1000 NGI_LOG_CONVERSION Info
1001 NGI_LOG_CONVERSION_WARNING Warning
1002 NGI_LOG_APPLICATION_ERROR Info (vxml application)
1003 NGI_LOG_APPLICATION_WARNING Warning (vxml application)

Table 72: NGI Specifiers (Continued)
Specifier ID Specifier Name Level
1004 NGI_LOG_SOMETHING_UNEXPECTED Error
1005 NGI_LOG_UNEXPECTED_WARNING Warning
1006 NGI_LOG_INCALL_SETUP_FAILURE Error
1007 NGI_LOG_CREATE_CALL_FAILURE Error
1008 NGI_LOG_NGI_INITIALIZATION_FAILURE Error
1009 NGI_LOG_CONFIGURATION_WARNING Warning
1010 NGI_LOG_RECORDED_FILE_TOO_SMALL Warning
1011 NGI_LOG_FETCH_FAILURE Info
1012 NGI_LOG_FETCH_FAILURE_WARNING Warning
1013 NGI_LOG_GRAMMAR_ERROR Info
1014 NGI_LOG_PROMPT_FETCH_TIMEOUT Error
1015 NGI_LOG_PROMPT_FETCH_ERROR Error
1016 NGI_LOG_FETCH_RESOURCE_TIMEOUT Error
1017 NGI_LOG_FETCH_RESOURCE_ERROR Error
1018 NGI_LOG_PARSE_ERROR Error
Genesys Voice Platform Interpreter Module ID and Specifiers

The Module ID for the GVPi application is 193. Table 73 describes the
specifiers for the GVPi application module.
Table 73: GVPi Specifiers
103 POP_CTRL_TIMING_INFO
150 CONV_CTRL_GET_PAGE
151 CONV_CTRL_NEW_URL_EXCEPTION
309 XML_CREATEACTION_ENTRYTRACE
310 XML_CREATEACTION_EXITTRACE
User’s Guide 409

Table 73: GVPi Specifiers (Continued)
311 XML_PRINTTOKEN_TRACE
312 XML_STACKSIZE_TRACE
320 XML_ACTION_EXEC_TRACE
321 XML_ACTION_EXCEPTION_TRACE
325 XMLPAGE_INDEXFILE
330 XMLPAGE_CREATE_DELETE_TRACE
340 CALL_PERFORMANCE
375 XMLPAGE_CACHE_ACTIONEXEC
380 XML_PAGE_CACHE
381 XML_PAGE_STAT
410 VXML_ACTIONSTATEMAP
440 VXML_CALL_TRACE
441 VXML_GRAMMAR_MATCH
442 VXML_PROMPT_QUEUE
443 VXML_EXEC_CONTEXT
444 VXML_CRDATA_OBJ
445 VXML_TRANS_REC
446 VXML_USER_UTTERANCE
447 VXML_UPDATE_SESS_VARS
448 VXML_GR_REGEXPR
449 VXML_GR_MIN_MAX_TONES
450 VXML_JS_DOM
500 PC_GENERAL
501 PC_DOWNLOAD_TIME
502 PC_UPLOAD_TIME

Appendix A: Module and Specifier IDs Call Control Platform
Table 73: GVPi Specifiers (Continued)
503 PC_STREAM_TIME
504 PC_PREFETCH_IGNORE
505 PC_PREFETCH_ABANDON
506 PC_CLEANUP
507 PC_THRDDATA_CLEANUP
508 PC_SESSION_CLEANUP
509 PC_INTRA_HOST
510 PC_NOTIFICATION
511 PC_HTTP_THRD
517 PC_HTTP_GET
518 PC_HTTP_PUT
520 PC_CONN
521 PC_SESSION
522 PC_COOKIE
Call Control Platform

Table 74 lists the Call Control Platform Application Module names and IDs.
Table 74: Call Control Platform Application Module Names and IDs
Module Module ID
Main Call Control Platform (CCP) application module 151
CCXML Interpreter 152
Media Controller 153
User’s Guide 411

Connection, Dialog, or Conference Events

Table 75 describes the specifiers for the main Call Control Platform module
(Module ID = 151). These events are related to a connection, dialog box, or
conference.

Table 75: CCP Connection, Dialog, or Conference Events
Module ID Specifier ID Description
151 Critical Events
1 Failed to initialize software.
Error Events
257 Inbound connection failure.
258 Media Controller reported error.
Warning Events
514 Inbound connection rejected while in suspended

state.
515 Application did not specify an event name for

<send>.
516 History Info header is malformed.
517 Invalid hints passed.
Info Events
1025 Connection created.
1026 Connection terminated.
1027 Sending 180 Ringing automatically as configured.
1044 Conference created.
1045 Conference terminated.
1046 Dialog created.
1047 Dialog terminated.
1048 Dialog transfer request rejected per configured.
1049 Buffering up join request until ready.
1050 Issuing buffered join requests.
User’s Guide 413

Media Controller Events

Table 76 describes the Media Controller events.
Table 76: CCP Media Controller Events
153 Critical Events
Error Events
257 Device profile entry empty.
258 Inbound call leg offer was rejected by application.
259 Bridging server encountered error.
260 Failure to initialize the Session Factory.
Warning Events
514 Uninitialization encountered problems.
515 Operation issued on the leg failed.
516 SDP generation/processing encountered problems.
517 SIP 491 Glare occurred.
518 Maximum number of retries reached.
519 Maximum number of updates reached.
520 Operation execution failed.
521 NULL Operation added to Transaction.
522 CallTerminate received and state is either

DISCONNECTED or ERROR.
523 The Default SIP Reject Code is < 300
Info Events
1024 Connection timeout.
1025 Dialog Unsupported MIME Type.
1026 Conference created.

Table 76: CCP Media Controller Events (Continued)
153 1027 Conference terminated.

(continued)
1028 Standard Conference creation.
1029 Implicit Conference creation.
1030 Media established.
1031 Media modified.
1032 Media terminated.
1033 SIP-CallID: [<SIP Call-ID>]; SendInvite()

Failed[<returncode>].
1034 SIP-CallID: [<SIP Call-ID>];

SendResponse(<SIP code>) for <SIP method>
1035 SIP-CallID: [<SIP Call-ID>]; SendCancel()

1036 SIP-CallID: [<SIP Call-ID>]; SendRequest(<SIP

method>) Failed[<returncode>].
1037 SIP-CallID: [<SIP Call-ID>]; SendInfo()

1038 SIP-CallID: [<SIP Call-ID>]; SendBye()

1039 SIP-CallID: [<SIP Call-ID>]; SendAck()

1040 Device profile selected.
1041 List of operations in the transaction: <OP1, OP2,

...>.
1042 SIP UserCall not connected.
1043 SIP UserCall error state.
1044 SIP UserCall received a failure response.
1045 SIP UserCall failed sending a message.
User’s Guide 415

Log_4 (INFO) Events

Table 77 describes the CCXML interpreter events at the INFO level.
Table 77: CCXMLI Log_4 INFO Events
152 256 INTR initialization failed.
352 CCXMLI initialization failed.
353 Set property value failed.
1024 INTR initialized.
1025 INTR uninitialized.
1026 A new CCXML session created.
1027 A CCXML session terminated.
1028 Failed to fetch document.
1029 Failed to parse document.
1030 Failed to compile document.
1031 Document initialization failed.
1032 Event not caught by application.
1033 Application log (by <log> tag).
1034 Error event generated by application.
1035 Exceed maximum session limit.
1036 Interpreter already shutting down.
1037 Session is still alive.
1120 CCXMLI initialized.
1122 A new CCXML session created.
1123 A CCXML session terminated.
1124 Failed to fetch document.
1127 Document initialization failed.

Appendix A: Module and Specifier IDs Resource Manager
Table 77: CCXMLI Log_4 INFO Events (Continued)
1128 Event not caught by application.
1129 Application log.
1130 Error event generated by application.
1132 Interpreter already shutting down.
Resource Manager
The Module ID for the Resource Manager application is 148.
Table 78 describes the specifiers for the Resource Manager application
module.
Table 78: Resource Manager Specifiers
Specifier Specifier Name

ID
257 GVPLOG_RM_UNRECOVERABLEERR
513 GVPLOG_RM_CONFIGERR
514 GVPLOG_RM_CCPSS7ERR
515 GVPLOG_RM_SOCKETERR
516 GVPLOG_RM_RESOURCEALLOCERR
517 GVPLOG_RM_CDRINITERR
518 GVPLOG_RM_CDRUNINITERR
519 GVPLOG_RM_CDRRECORDCREATEERR
520 GVPLOG_RM_CDRRECORDDELETEERR
521 GVPLOG_RM_DIALINGRANGEEXCEED
522 GVPLOG_RM_DIALINGTYPEINVALID
523 GVPLOG_RM_DIALINGEXPRINVALID
524 GVPLOG_RM_DNISNOTEXIST
525 GVPLOG_RM_DEFAULTTENTANTNOTFOUND
User’s Guide 417

Table 78: Resource Manager Specifiers (Continued)

ID
526 GVPLOG_RM_REQUESTURITRANSLATIONFAIL
527 GVPLOG_RM_CALLCREATEFAIL
528 GVPLOG_RM_APPPROFILENOTFOUND
529 GVPLOG_RM_TENANTNOTFOUND
530 GVPLOG_RM_DEFAULTIVRPROFILENOTFOUND
531 GVPLOG_RM_DEFAULTSERVICETYPENOTFOUND
532 GVPLOG_RM_MANDATORYURIPARAMNOTFOUND
533 GVPLOG_RM_INVALIDURIPARAM
534 GVPLOG_RM_SERVICEPREREQNOTFOUND
535 GVPLOG_RM_NOMATCHINGSERVICETYPE
536 GVPLOG_RM_NOMATCHINGGWPREFERENCE
537 GVPLOG_RM_CCILIBINVALIDPARAM
538 GVPLOG_RM_CCILIBCONFIGOBJERR
539 GVPLOG_RM_CCILIBRMOBJERR
540 GVPLOG_RM_CCILIBRESOBJNOTFOUND
541 GVPLOG_RM_CCILIBLOGICALRESCREATEFAIL
542 GVPLOG_RM_CCILIBPHYSICALRESCREATEFAIL
543 GVPLOG_RM_CCILIBTENANTNOTFOUND
544 GVPLOG_RM_CCILIBTENANTCREATEFAIL
545 GVPLOG_RM_CCILIBAPPIDNOTFOUND
546 GVPLOG_RM_CCILIBLINKEDRESNOTFOUND
547 GVPLOG_RM_CCILIBPARENTNOTFOUND
548 GVPLOG_RM_CCILIBLOGICALRESGROUPNOTFOUND
549 GVPLOG_RM_CCILIBTENANTCONVERTERROR
550 GVPLOG_RM_CCILIBCAPADDERROR


ID
551 GVPLOG_RM_CCILIBAPPCONVERTERROR
552 GVPLOG_RM_CCILIBINVALIDINPUTARG
553 GVPLOG_RM_RESSESSIONCREATEFAIL
554 GVPLOG_RM_CCILIBAPPCREATEFAIL
555 GVPLOG_RM_CCILIBUPDATEINVALIDCFGOBJ
556 GVPLOG_RM_CCILIBUPDATETENANTNOTFOUND
557 GVPLOG_RM_CCILIBUPDATETENANTPOPULATEFAIL
558 GVPLOG_RM_CCILIBUPDATEAPPNOTFOUND
559 GVPLOG_RM_CCILIBUPDATEAPPPOPULATEFAIL
560 GVPLOG_RM_CCILIBUPDATELOGICALRESNOTFOUND
561 GVPLOG_RM_CCILIBUPDATELOGICALRESADDERR
562 GVPLOG_RM_CCILIBUPDATERESOBJNOTFOUND
563 GVPLOG_RM_CCILIBUPDATEPHYRESCREATEFAIL
564 GVPLOG_RM_CCILIBUPDATEINVALIDOBJ
565 GVPLOG_RM_CCILIBUPDATEINVALIDOBJTYPE
566 GVPLOG_RM_CCILIBUPDATETENANTADDFAIL
567 GVPLOG_RM_CCILIBUPDATEAPPADDFAIL
568 GVPLOG_RM_CCILIBUPDATETENANTREMOVEFAIL
569 GVPLOG_RM_CCILIBUPDATEAPPREMOVEFAIL
570 GVPLOG_RM_CCILIBUPDATETENANTUPDATEFAIL
571 GVPLOG_RM_CCILIBUPDATEAPPLICATIONUPDATEFAIL
572 GVPLOG_RM_REGISTERERROR
574 GVPLOG_RM_POLICYVIOLATIONERROR
575 GVPLOG_RM_GENERIC_ERROR
576 GVPLOG_RM_SUBSCRIPTION_ERROR
User’s Guide 419


ID
577 GVPLOG_RM_POLICYENFORCEMENTVIOLATIONERROR
769 GVPLOG_RM_INVALIDMSG
770 GVPLOG_RM_INVALIDCONFIG
771 GVPLOG_RM_CCPSS7SUBSERFAIL
772 GVPLOG_RM_NETWORKPROBLEM
773 GVPLOG_RM_REQUESTURIPARSEFAIL
774 GVPLOG_RM_OPTIONUSERINFOEXIST
775 GVPLOG_RM_TOHEADERPARSEFAIL
776 GVPLOG_RM_RMSERVICEAGENTBADMSGFORMAT
777 GVPLOG_RM_RMSUSPEND
778 GVPLOG_RM_SIPSERVICESAMEPRECEDENCE
779 GVPLOG_RM_INVALIDCALLTENANTID
780 GVPLOG_RM_FAILEDTOFINDLINKEDTENANT
781 GVPLOG_RM_FAILEDTOFINDLINKEDRESOURCE
782 GVPLOG_RM_LOGICALRESINFONOTFOUND
783 GVPLOG_RM_LOGICALRESPOPULATEFAIL
784 GVPLOG_RM_LOGICALRESSECTIONNOTFOUND
785 GVPLOG_RM_PHYSRESPOPULATEFAIL
786 GVPLOG_RM_TENANTPOPULATEINCOMPLETE
787 GVPLOG_RM_APPINFONOTFOUND
788 GVPLOG_RM_APPPOPULATEINCOMPLETE
789 GVPLOG_RM_DNISEXTRACTFAIL
790 GVPLOG_RM_SETTINGLOGICALRESPROPERTIES
791 GVPLOG_RM_AORNOTFOUND
792 GVPLOG_RM_CAPACITYNOTFOUND


ID
793 GVPLOG_RM_CAPACITYNONUNSIGNED
794 GVPLOG_RM_SETTINGPHYRESPROPERTIES
795 GVPLOG_RM_UPDATELOGICALRESGROUPNOTFOUND
796 GVPLOG_RM_UPDATEPOPULATEPHYRESFAIL
797 GVPLOG_RM_UPDATEFAILGETPHYRES
798 GVPLOG_RM_UPDATEPOPULATELOGICALRESFAIL
799 GVPLOG_RM_UPDATELOGICALRESNOTFOUND
800 GVPLOG_RM_UPDATEPHYRESREMOVED
801 GVPLOG_RM_UPDATETENANTADDED
802 GVPLOG_RM_UPDATEAPPADDED
803 GVPLOG_RM_UPDATELINKEDTENANTREMOVED
804 GVPLOG_RM_UPDATEAPPREMOVED
805 GVPLOG_RM_UDPATELINKEDTENANTUPDATED
806 GVPLOG_RM_UDPATEAPPDATAUPDATED
807 GVPLOG_RM_UPDATEIGNORED
808 GVPLOG_RM_WARNING_BAD_REGEX
809 GVPLOG_RM_SNMP_DISABLED
810 GVPLOG_RM_BURSTAPPBEGIN
811 GVPLOG_RM_BURSTAPPEND
812 GVPLOG_RM_BURSTTENANTBEGIN
813 GVPLOG_RM_BURSTTENANTEND
814 GVPLOG_RM_NETWORKRECOVERY
815 GVPLOG_RM_REDUNDANCYUNDEFINED
816 GVPLOG_RM_UNSUBSCRIBE
817 GVPLOG_RM_SINGLETENANT
User’s Guide 421

Appendix A: Module and Specifier IDs CTI Connector

ID
818 GVPLOG_RM_HAMODE
1025 GVPLOG_RM_CCPSS7STATE
1026 GVPLOG_RM_COMMNOTICE
1027 GVPLOG_RM_CLUSTERNOTICE
1028 GVPLOG_RM_CCPPROXYSTATE
1029 GVPLOG_RM_STARTUP
1030 GVPLOG_RM_SHUTDOWN
1031 GVPLOG_RM_RMSERVICEAGENTSTATUS
1032 GVPLOG_RM_STATUSLOG
1033 GVPLOG_RM_ACTIVEMODE
1034 GVPLOG_RM_STANDBYMODE
1035 GVPLOG_RM_CONFIGINFO
1281 GVPLOG_RM_PROVCHANGE
1282 GVPLOG_RM_CCPSS7NOTIFY
1283 GVPLOG_RM_MODULECONNECTIVITY
1284 GVPLOG_RM_MODULECONFIGMODIF
1285 GVPLOG_RM_CLUSTERINFO
1286 GVPLOG_RM_NEWCALL
1287 GVPLOG_RM_REGISTERINFO
2305 GVPLOG_RM_GENERIC_TRACE
CTI Connector
Table 79 lists the CTI Connector Application Module names and IDs.

Table 79: CTI Connector Application Module Names and IDs
Module Module ID
CTI Adaptor 171
CTI Client 172
CTI Adaptor
Table 80 describes the specifiers for the CTI Adaptor module.
Table 80: CTI Adaptor Specifiers
1501 CTICA_INVALID_SESSION_ERROR
1502 CTICA_MEMORY_ERROR
1503 CTICA_INTERNAL_ERROR
1504 CTICA_UNSUPPORTED_SIP_EVENT_ERROR
1505 CTICA_UNSUPPORTED_MSG_BODY_ERROR
1506 CTICA_INITIALIZATION_ERROR
1507 CTICA_CCLIB_ERROR
1508 CTICA_MC_ERROR
1509 CTICA_SIPCALLERROR_ERROR
1510 CTICA_ENCODE_DATA_ERROR
1511 CTICA_DECODE_DATA_ERROR
1512 CTICA_SIP_STAT_ERROR
1513 CTICA_CALLOBJECT_STAT_ERROR
1514 CTICA_PARSING_ERROR
1515 CTICA_SNMPLIB_ERROR
1516 CTICA_NOTFOUND_ERROR
1517 CTICA_UNKNOWN_ERROR
1518 CTICA_CTICLIENT_ERROR
User’s Guide 423

Table 80: CTI Adaptor Specifiers (Continued)
1519 CTICA_RM_DOWN
1520 CTICA_CONFIG_ERROR
1550 CTICA_CALLFLOW
1551 CTICA_CALLFLOW_QUERYSTRING
1552 CTICA_CALLFLOW_SIPSTAT
1553 CTICA_CALLFLOW_OBJSTAT
1554 CTICA_CALLFLOW_SIP
1555 CTICA_AUTOMATION
1575 CTICA_SNMPDATA

CTI Client
Table 81 describes the specifiers for the CTI Client module.
Table 81: CTI Client Specifiers

ID
1401 CTICC_INVALID_SESSION_ERROR
1402 CTICC_INTERNEL_ERROR
1403 CTICC_UNSUPPORTED_URS_TREATMENT_ERROR
1404 CTICC_SEND_XML_MSG_ERROR
1405 CTICC_UNSUPPORTED_IVR_TREATMENT_ERROR
1406 CTICC_INITIALIZATION_ERROR
1407 CTICC_CCIBLIB_ERROR
1408 CTICC_IVR_SERVER_CONNECTION_ERROR
1409 CTICC_IVR_DATAMSG_ERROR
1410 CTICC_IVR_LOGIN_ERROR
1411 CTICC_INVALID_QUERY_STRING_PARAMETER_ERROR
1412 CTICC_RECV_CALLSTATUS_ERROR
1413 CTICC_GETCALLINFOFORRESP_ERROR
1414 CTICC_INVALID_QUERY_STRING_ERROR
1415 CTICC_CALLERROR_ERROR
1416 CTICC_ENCODE_DATA_ERROR
1417 CTICC_DECODE_DATA_ERROR
1418 CTICC_INVALID_CALL_ERROR
1419 CTICC_IVR_SERVER_TIMEOUT_ERROR
1420 CTICC_SNMPLIB_ERROR
1440 CTICC_IVR_SHUTDOWN_ERROR
1450 CTICC_CALLFLOW
1451 CTICC_CALLFLOW_QUERYSTRING
User’s Guide 425

Appendix A: Module and Specifier IDs Supplementary Services Gateway
Table 81: CTI Client Specifiers (Continued)

ID
1452 CTICC_CALLFLOW_XML
1453 CTICC_CALLFLOW_INFO
1475 CTICC_SNMPDATA
1480 CTICC_CTIC_STARTED
1481 CTICC_CTIC_STOPPED
1482 CTICC_CTIC_STOPPED_GRACEFULLY
1483 CTICC_IVR_SERVER_CONNECTION_UP
Supplementary Services Gateway

The Module ID for the Supplementary Services Gateway is 174.
Table 82 describes the specifiers for the Supplementary Services Gateway
application module.
Table 82: Supplementary Services Gateway Specifiers
ID Specifier Name
20101 SSG_INITIALIZATION_ERROR
20102 SSG_INVALID_PTR_ERROR
20103 SSG_INTERNAL_ERROR
20104 SSG_CCILIB_ERROR
20105 SSG_HTTP_ERROR
20106 SSG_SNMPLIB_ERROR
20107 SSG_CONFIG_OBJECT_ERROR
20111 SSG_DB_CONNECTION_DOWN
20112 SSG_DB_CONNECT_ERROR
20113 SSG_DB_ERROR
20114 SSG_DB_PROCEDURE_FAILED_ERROR

Appendix A: Module and Specifier IDs Supplementary Services Gateway
Table 82: Supplementary Services Gateway Specifiers (Continued)
ID Specifier Name
20121 SSG_HTTP_REQ_STORAGE_ERROR
20122 SSG_REQUEST_PROCESS_ERROR
20123 SSG_SIP_PROCESSING_ERROR
20124 SSG_QUERYSTRING_PARSE_ERROR
20125 SSG_CUSTOM_OBJECT_ERROR
20126 SSG_NOTIFICATION_URL_GET_FAILED
20131 SSG_STARTED
20132 SSG_STOPPED
20133 SSG_SHUTDOWN
20141 SSG_HTTP_QUERYSTRING
20142 SSG_SNMPDATA
20151 SSG_TLIB_INIT_ERROR
20152 SSG_TEVENTS_ERROR
20153 SSG_TLIB_CONN_RECOVERY_ERROR
20154 SSG_TLIB_GENERIC_ERROR
20161 SSG_SIPSERVER_CONTACT_FAILED
20162 SSG_REQUEST_REJECTION_SIPSERVER_NOT_CONNECTED
20163 SSG_SIPSERVER_APPLICATION_NOT_FOUND
20171 SSG_RESOURCE_DN_NOT_REGISTERED
20172 SSG_REQUEST_REJECTION_RESOURCE_DN_NOT_REGISTER
ED
20173 SSG_TENANT_RESOURCE_DN_NOT_AVAILABLE
User’s Guide 427

Appendix A: Module and Specifier IDs PSTN Connector
PSTN Connector
Table 83 lists the PSTN Connector Application Module names and IDs.
Table 83: PSTN Connector Application Module Names and IDs
Module Module ID
Dialogic Manager 138
Gateway Manager 139
Media Manager 140
PSTN Connector 141
Dialogic Manager
Table 84 describes the specifiers for the Dialogic Manager module.
Table 84: Dialogic Manager Specifiers
1001 DLGC_MGR_INIT_ERROR
2001 DLGC_MGR_D_CHAN_STATUS_DOWN
2002 DLGC_MGR_D_CHAN_STATUS_UP
2003 DLGC_MGR_LINK_ERROR
2004 DLGC_MGR_LINK_OK
3001 DLGC_MGR_B_CHAN_STATUS_DOWN
4001 DLGC_MGR_B_CHAN_STATUS_UP
4002 DLGC_MGR_B_CHAN_STATUS_CHANGED

Gateway Manager
Table 85 describes the specifiers for the Gateway Manager module.
Table 85: Gateway Manager Specifiers

ID
1001 GW_MGR_INITIALIZATION_ERROR
2001 GW_MGR_QUERY_PARSE_ERROR
2002 GW_MGR_CALLOBJ_NOT_FOUND
2003 GW_MGR_TDM_HANGUP_ERROR
2004 GW_MGR_UNSUPPORTED_MEDIA
2005 GW_MGR_CODEC_MATCH_ERROR
2006 GW_MGR_INVALID_CALL_STATE
2007 GW_MGR_OTHER_CALLOBJ_NOT_FOUND
2008 GW_MGR_RETRIEVE_MSG_ERROR
2009 GW_MGR_ACTIVATE_MEDIA_ERROR
2010 GW_MGR_DEACTIVATE_MEDIA_ERROR
2011 GW_MGR_ANSWER_CALL_ERROR
2012 GW_MGR_NO_FREE_PORTS_ERROR
2013 GW_MGR_INVALID_DIAL_NUM_ERROR
2014 GW_MGR_DIAL_ERROR
2015 GW_MGR_CREATE_MEDIA_ERROR
2016 GW_MGR_CALLOBJ_CREATE_ERROR
2017 GW_MGR_DNIS_MISSING_ERROR
2018 GW_MGR_ACCEPT_CALL_ERROR
2019 GW_MGR_REFER_NUM_MISSING_ERROR
2020 GW_MGR_UNSUPPORTED_XFER_TYPE_ERROR
2021 GW_MGR_DLGC_BLIND_XFER_UNSUPPORTED_ERROR
2022 GW_MGR_ONE_CHANNEL_XFER_ERROR
User’s Guide 429

Table 85: Gateway Manager Specifiers (Continued)

ID
2023 GW_MGR_REPLACES_CALL_ID_MISSING_ERROR
2024 GW_MGR_TWO_CHANNEL_XFER_ERROR
2025 GW_MGR_DESTROY_MEDIA_ERROR
2026 GW_MGR_BRIDGE_ERROR
2027 GW_MGR_SIP_STACK_INIT_ERROR
2027 GW_CREATE_MSG_ERROR
3001 GW_MGR_STOP_RINGBACK_WARN
3002 GW_MGR_UNEXPECTED_BYE_WARN
3003 GW_MGR_UNEXPECTED_CANCEL_WARN
3004 GW_MGR_CALLER_HUNGUP_WARN
3005 GW_MGR_AGENT_HUNGUP_WARN
3006 GW_MGR_START_RINGBACK_WARN
3007 GW_MGR_MAX_GLARE_RETRIES_DONE_WARN
3008 GW_MGR_MEDIA_STOP_TIMEOUT_WARN
3009 GW_MGR_BLIND_XFER_TIMEOUT_WARN
3010 GW_MGR_ATT_XFER_TIMEOUT_WARN
3011 GW_MGR_TWO_CH_XFER_TIMEOUT_WARN
3012 GW_MGR_UNSUPPORTED_MIB_ATTRIB_WARN
4001 GW_MGR_GLARE_OCCURRED_INFO
4002 GW_MGR_CALLOBJ_DELETED_INFO

Media Manager
Table 86 describes the specifiers for the Media Manager module.
Table 86: Media Manager Specifiers

ID
1001 MEDIA_MGR_INIT_ERROR
2001 MEDIA_MGR_MEDIA_SESSION_CREATE_ERROR
2002 MEDIA_MGR_MEDIA_SESSION_ACTIVATE_ERROR
2003 MEDIA_MGR_MEDIA_SESSION_DEACTIVATE_ERROR
2004 MEDIA_MGR_MEDIA_SESSION_DESTROY_ERROR
2005 MEDIA_MGR_MEDIA_SESSION_NOT_FOUND
2006 MEDIA_MGR_RTP_SESSION_CREATE_ERROR
2007 MEDIA_MGR_RFC2833_HANDLER_CREATE_ERROR
2008 MEDIA_MGR_BUFFER_QUEUE_CREATE_ERROR
2007 MEDIA_MGR_TDM_INIT_MEDIA_ERROR
2010 MEDIA_MGR_TDM_START_MEDIA_ERROR
2011 MEDIA_MGR_DTMF_ENCODE_ERROR
2012 MEDIA_MGR_DTMF_SEND_ERROR
2013 MEDIA_MGR_DTMF_DECODE_ERROR
2014 MEDIA_MGR_TDM_STOP_MEDIA_ERROR
2015 MEDIA_MGR_DTMF_SEND_TO_TDM_ERROR
2016 MEDIA_MGR_RTP_NETWORK_CREATE_ERROR
2017 MEDIA_MGR_RTP_BRIDGE_CREATE_ERROR
2018 MEDIA_MGR_RTP_NETWORK_DESTROY_ERROR
2019 MEDIA_MGR_RTP_BRIDGE_DESTROY_ERROR
2020 MEDIA_MGR_RTP_MODIFY_NETWORK_ERROR
2021 MEDIA_MGR_RTP_JOIN_ERROR
2022 MEDIA_MGR_RTP_PACKET_SEND_ERROR
User’s Guide 431

Appendix A: Module and Specifier IDs Fetching Module
PSTN Connector
Table 87 describes the specifiers for the PSTN Connector module.
Table 87: PSTN Connector Specifiers
1001 PSTNC_CRITICAL_INITIALIZATION
4001 PSTNC_PROC_STARTED
4002 PSTNC_PROC_STOPPED
Fetching Module
The Module ID for the Fetching Module Application is 80.
Table 88 describes the specifiers for the Fetching Module application module.
Table 88: Fetching Module Specifiers
Specifier ID Specifier Name Description
Level: Critical
40000 FMLOG_MEM_ALLOC_FAIL Memory allocation failed for %s.
40001 FMLOG_FM_INIT_FAIL Fetching Module initialization failed.
Level: Error
20020 FMLOG_SESS_OPEN_FAIL Open Session to Fetching Server failed.
20021 FMLOG_CONN_FAIL Connect to Fetching Server failed.
20022 FMLOG_SEND_FAIL Send to Fetching Server failed.
20023 FMLOG_BAD_SESS_ID Invalid session ID.
20006 FMLOG_EMSLOG_INIT_FAIL EMS logging service initialization failed.
20007 FMLOG_BAD_SHMEM_PARAM Invalid shared memory parameter.
20008 FMLOG_SHMEM_NAME_EMPTY Empty shared memory name.
20009 FMLOG_SHSEM_NAME_FAIL Shared semaphore name generation failed.
20010 FMLOG_SHSEM_CREATE_FAIL Shared semaphore creation failed.
20011 FMLOG_SHSEM_LOCK_FAIL Shared semaphore lock failed.

Table 88: Fetching Module Specifiers (Continued)
Specifier ID Specifier Name Description
20012 FMLOG_SHMEM_MAP_FAIL Shared memory map failed for file %s.
20013 FMLOG_SHMEM_ATTACH_FAIL Shared memory attach failed for ID %d.
20014 FMLOG_SHMEM_NAME_FAIL Shared memory name generation failed.
20015 FMLOG_SHMEM_CREATE_FAIL Shared memory creation failed for size %d.
20016 FMLOG_SHMEM_READ_FAIL Unable to read shared-memory.
20017 FMLOG_SHMEM_WRITE_FAIL Unable to write shared-memory.
20018 FMLOG_GET_PIPE_FAIL Failed to get pipe name.
20019 FMLOG_OPEN_PIPE_FAIL Failed to open pipe.
Level: Warning
30000 FMLOG_SESS_CLS_FAIL Close Session to Fetching Server failed.
30003 FMLOG_CLS_PIPE_FAIL Failed to close pipe.
User’s Guide 433


Appendix
B Media Control Platform

Reference Information
This appendix provides miscellaneous reference information about the Media
Control Platform.

Audio and Video File Formats, page 435

Combined Audio and Video Formats—Play, page 438
 Dynamic Media Control Platform Parameters, page 442

CPA Configuration Options That Can be Overwritten, page 443

SIP Headers, page 445
 Handling Error Responses for Outbound Calls, page 449

VAR Metrics, page 450
Audio and Video File Formats

This section provides information about the supported file formats for playing
and recording audio and video media:
• Audio-Only Formats—Play
• Video-Only Formats—Play (see page 438)
• Combined Audio and Video Formats—Play (see page 438)
• Audio-Only Formats—Record (see page 439)
• Video-Only Formats—Record (see page 441)
• Combined Audio and Video Formats—Record (see page 441)
Audio-Only Formats—Play
Table 89 lists the supported audio-only file formats for playing prompts.
User’s Guide 435

Appendix B: Media Control Platform Reference Information Audio and Video File Formats
Table 89: Supported Audio File Formats—Play
Expected File MIME-type File Format Sample Size Encoding

Extension
.vox audio/x-vox Raw audio 8-bit mono G.711 ulaw, G.711 alaw
audio/vox (depends on platform
configuration)
.au audio/au Audio with .au 8-bit mono G.711 ulaw, G.711 alaw,
audio/x-au header PCM, G.726, G.722 (depends
on file header information)
.awb audio/amr-wb Raw audio AMR-WB for encoding
.ulaw audio/basic Raw audio 8-bit mono G.711 ulaw

audio/PCMU
audio/mulaw
.alaw audio/x-alaw-basic Raw audio 8-bit mono G.711 alaw

audio/PCMA
audio/alaw
g.722 audio/g722 Raw audio G.722
.g729 audio/g729 Raw audio G.729
.adpcm24 audio/x-g726-24 Raw audio 24 kb/sec ADPCM (G.726)

audio/g726-24
.adpcm audio/x-g726 Raw audio 32 kb/sec ADPCM (G.726)

audio/x-g726-32
audio/g726
auduio/g726-32
audio/x-adpcm
audio/adpcm
audio/x-adpcm8
.adpcm40 audio/x-g726-40 Raw audio 40 kb/sec ADPCM (G.726)

audio/g726-40
Note: The sample rate is always 8000 Hz except for AMR-WB and G722 which are 16000 Hz. If Media
Control Platform detects a non-8000 Hz audio file, it issues a warning message, and plays the prompt as
if the sampling rate is 8000 Hz. A configurable Media Control Platform parameter,
mpc.mediamgr.strictsamplingrate, enables you to prevent the playing of non-8000 Hz audio files
(except for AMR-WB).

Appendix B: Media Control Platform Reference Information Audio and Video File Formats
Table 89: Supported Audio File Formats—Play (Continued)
Expected File MIME-type File Format Sample Size Encoding

Extension
.pcm8 audio/L8 Raw audio 8-bit unsigned Linear PCM

mono
.pcm16 audio/L16 Raw audio 16-bit signed Linear PCM

mono
.wav audio/wav Audio with .wav G.711 ulaw, G.711 alaw,

audio/x-wav header G.722, G.726, G.729, PCM,
PCM16, GSM 6.10,
MS-GSM, AMR, AMR-WB
(G.722.2), ADPCM (depends
on file header information)
.avi audio/avi Audio stored in G.711 ulaw, G.711 alaw, G.

audio/x-avi AVI container 722, G.726, G.729, PCM,
PCM16, GSM, AMR,
AMR-WB (G.722.2),
ADPCM (depends on file
header information)
.nist audio/wav Audio with NIST 8-bit mono G.711 ulaw, G.711 alaw
audio/x-wav header (depends on file header
information)
.gsm audio/x-gsm Raw audio GSM 6.10

audio/gsm
.msgsm audio/x-ms-gsm Raw Audio MS-GSM

audio/ms-gsm
.amr audio/amr Raw audio AMR
.awb audio/amr-wb Raw audio AMR-WB (G.722.2)
.3gp(p) audio/3gpp Audio stored in AMR, AMR-WB (G.722.2)

3GP container
Note: The sample rate is always 8000 Hz except for AMR-WB and G722 which are 16000 Hz. If Media
Control Platform detects a non-8000 Hz audio file, it issues a warning message, and plays the prompt as
if the sampling rate is 8000 Hz. A configurable Media Control Platform parameter,
mpc.mediamgr.strictsamplingrate, enables you to prevent the playing of non-8000 Hz audio files
(except for AMR-WB).
User’s Guide 437

Appendix B: Media Control Platform Reference Information Combined Audio and Video Formats—Play
Note: The Media Control Platform (MCP) supports V7.00 (2006-06) of

3GPP TS 26.244 for 3GP and ISO/IEC 14496-12 ISO; therefore, the
MCP may not play latter versions of 3GP files correctly.
Video-Only Formats—Play
Table 90 lists the supported video-only file formats for playing prompts.
Table 90: Supported Video File Formats—Play
Expected File MIME-type Sample Rate File Format Encoding

Extension
.263 video/h263 30 fps Raw video h263

video/x-h263 (recommended)
.263 video/h263-1 30 fps Raw video h263-1998

998 (recommended)
.264 video/3gp(p) 30 fps Video stored in h264

video/x-h264 (recommended) 3GP container
Note: The .avi recordings produced by GVP are supported with media
players that can play avi files only without index tables. A list of
known supported media players include Super and ffdshow. A list of
known unsupported media players include QuickTime and Nokia
Media Converter Pro.
Combined Audio and Video Formats—Play

Table 91 lists the supported audio/video file formats for playing prompts.

Table 91: Supported Audio/Video File Formats—Play
Expected File MIME-type Sample Rate File Format Encoding

Extension
.avi video/avi • Audio: 8000 Hz Audio/video • Audio: G.711 ulaw, G.711

video/x-avi • Video: 30 fps stored in AVI alaw, PCM, ADPCM
(recommended) container (depends on file header
information)
• Video: h263, h263-1998,
h264, vp8 (depends on file
header information)
.3gp video/3gp(p) • Audio: 8000 Hz Audio/video • Audio: AMR

• Video: 30 fps stored in 3GP • Video: h263, h263-1998,
(recommended container h264 (depends on file header
for h263) information)
• Video: 29.97
fps
(recommended
for h264)
Audio-Only Formats—Record
Table 92 lists the supported audio-only file formats for recording.
Table 92: Supported Audio File Formats—Record
MIME-type Recorded File Sample Size Encoding File

Format Extension
audio/amr-wb Raw audio AMR-WB .awb

(g722.2)
audio/x-vox Raw audio 8-bit mono G.711 ulaw, G.711 alaw .vox
audio/vox (depends on platform
configuration)
audio/basic Raw audio 8-bit mono G.711 ulaw .ulaw

audio/PCMU
audio/mulaw
Notes:
• Genesys Voice Platform (GVP) 8.1 and above support the 8000 Hz and 16000 Hz audio sampling
rates.
• Genesys Voice Platform (GVP) 8.1 and above do not support .au and .nist file recording.
User’s Guide 439

Table 92: Supported Audio File Formats—Record (Continued)

Format Extension
audio/x-alaw-basic Raw audio 8-bit mono alaw .alaw

audio/PMCA
audio/alaw
audio/g722 Raw audio G.722 .g722
audio/g729 Raw audio G.729 .g729
audio/x-g726-24 Raw audio 24 kb/sec ADPCM (G.726) .adpcm24

audio/g726-24
audio/x-g726 Raw audio 32 kb/sec ADPCM (G.726) .adpcm

audio/x-adpcm
audio/adpcm
audio/x-adpcm8
audio/x-g726-32
audio/g726-32
audio/g726
audio/x-g726-40 Raw audio 40 kb/sec ADPCM (G.726) .adpcm40

audio/g726-40
audio/L8 Raw audio 8-bit unsigned Linear PCM .pcm8

mono
audio/L16 Raw audio 16-bit signed Linear PCM .pcm16

mono
Notes:
rates.

Table 92: Supported Audio File Formats—Record (Continued)

Format Extension
audio/x-wav;codec= Audio with .wav audio_codec: ulaw, alaw, .wav

<audio_codec> header pcm, pcm16, g722, g726,
;rate=<g726_encoding gsm. Default: ulaw or alaw
_ (depends on platform
rate> configuration).
audio/wav;codec= g726_encoding_rate:
<audio_codec> 16 kb, 24 kb, 32 kb, or
;rate=<g726_encoding 40 kb.
_ Default: 32 kb.
rate>
audio/x-gsm Raw audio gsm 6.10 .gsm

audio/gsm
audio/amr Raw audio AMR .amr
audio/3gpp Audio stored in AMR, AMR-WB(G.722.2) .3gp(p)

3GP(P) container
audio/amr-wb Raw audio 16000 hz AMR-WB .awb
Notes:
rates.
Video-Only Formats—Record
Table 93 lists the supported video-only file formats for recording.
Table 93: Supported Video File Formats—Record
MIME-type Recorded File Format Encoding File

Extension
video/h263 Raw video h263 .263
video/h263-1998 Raw video h263-1998 .263
video/h264 Raw video h264 .264
Combined Audio and Video Formats—Record

Table 94 lists the supported audio/video file formats for recording.
User’s Guide 441

Appendix B: Media Control Platform Reference Information Dynamic Media Control Platform Parameters
Table 94: Supported Audio/Video File Formats—Record
MIME-type Recorded File Encoding File

Format Extension
video/avi;codec=<audio_ Audio/video audio_codec: ulaw, alaw, pcm16, .avi

codec>;rate=<g726_ stored in AVI pcm8, adpcm, gsm, g722, g726,
encoding_rate> container g.729, amr, amr-wb (g722.2), none.
;videocodec=<video_codec> Default: ulaw or alaw (depends on
video/x-avi;codec=<audio_ platform configuration).
codec>;rate=<g726_ video_codec: h263.
encoding_rate> Default: h263
;videocodec=<video_codec> g726_encoding_rate: 16 kbps,
24 kbps, 32 kbps, or 40 kbps.
Default: 32 kbps.
video/3gpp;codec=<audio_ Audio/video audio_codec: amr, amr-wb (g.722.2), .3gp(p)

codec> stored in 3GP(P) none.
;videocodec=<video_codec> container Default: amr
video_codec: h263, h264.
Default: h263
Note: Genesys Voice Platform (GVP) 8.1 and above support the 8000 Hz and 16000 Hz audio sampling
rates.
RTSP Server Support

The Media Control Platform supports the following RTSP servers:
• RTP transport for MP3 audio format using RTSP.
• Darwin Streaming Server v 6.0.3-2
• Helix Server (Columbia) (13.0.0.479)
Note: Because of a streaming hint format compatibility issue, MCP recorded

3GP files cannot be played by the Darwin Streaming Server.
Dynamic Media Control Platform

Parameters
This section lists the configuration and service parameters whose values can be
set dynamically for a call session.

Appendix B: Media Control Platform Reference Information CPA Configuration Options That Can be Overwritten
The dynamic value is obtained from the gvp.config.<parameter

name>=<parameter value> parameter in the Request-URI of the establishing SIP
INVITE.
The following configuration options can be set dynamically:
asr Section mpc Section
load_once_per_calla codec
delay_for_dtmf codecpref
log_metrics_to_asr fcr.defaultdtmfhandling
transmitmultiplecodec
sessmgr Section appendrejcodec
maxincalltime rtp.dtmf.receive
ECS_Fallback rtp.dtmf.send
join_fallback record.defaultdtmfhandling
record.start.beep.filename rtp.tos
alert_before_fetch rtcp.tos
mediaswitch_on_alert maxrecordfilesize
acceptcalltimeout disabledcodecs
disabledtranscoders
sip Section rtp.request_iframe
warningheaders
sendalert
sendrecvevents
logmsg
xfercopy.headers
a. The value can also be overridden by the value of the gvp.poli-

cy.mcp-asr-usage-mode parameter of the IVR Profile, which the Re-
source Manager passes to the Media Control Platform as a
Request-URI parameter.
CPA Configuration Options That Can be

Overwritten
Call Progress Analysis (CPA) configuration options can be overwritten by the
gvp.service parameters in the IVR Profile. The IVR Profile service parameter
User’s Guide 443

Appendix B: Media Control Platform Reference Information CPA Configuration Options That Can be Overwritten
must be prefixed by voicexml.gvp.config for VoiceXML services, and by

cpd.gvp.config for MSML services.
For more information about call progress detection and analysis, see the
Table 95: CPA Options That Can Be Overwritten
msml.cpd.beeptimeout mpc.cpa.maxsil_before_beep
msml.cpd.postconnectimeout mpc.cpa.preconn_tones_det_mode
msml.cpd.preconnectimeout mpc.cpa.ontime_ringback_match_percent
mpc.cpa.enable_log_param mpc.cpa.ontime_preconn_match_percent
mpc.cpa.enable_log_result mpc.cpa.silencefilltimeout
mpc.cpa.maxpreconntime mpc.cpa.ringback
mpc.cpa.maxpostconntime mpc.cpa.busy
mpc.cpa.maxbeepdettime mpc.cpa.fastbusy
mpc.cpa.detectstatenolimitbuffdur mpc.cpa.sit_nocircuit
mpc.cpa.keptdur_before_statechange mpc.cpa.sit_vacantcircuit
mpc.cpa.priority_normal_machinegreetingdur mpc.cpa.sit_operatorintercept
mpc.cpa.priority_normal_voicepausedur mpc.cpa.sit_reorder
mpc.cpa.priority_normal_maxvoicesigdur mpc.cpa.fax
mpc.cpa.priority_voice_machinegreetingdur mpc.cpa.custom1
mpc.cpa.priority_voice_voicepausedur mpc.cpa.custom2
mpc.cpa.priority_voice_maxvoicesigdur mpc.cpa.custom3
mpc.cpa.priority_machine_machinegreetingdur mpc.cpa.custom4
mpc.cpa.priority_machine_voicepausedur mpc.cpa.tone1(..10).segment1(..3).f1(..2).min
mpc.cpa.priority_machine_maxvoicesigdur mpc.cpa.tone1(..10).segment1(..3).f1(..2).max
mpc.cpa.faxdur mpc.cpa.tone1(..10).segment1(..3).ontime.min
mpc.cpa.voice_range_db mpc.cpa.tone1(..10).segment1(..3).ontime.max
mpc.cpa.voice_level_db mpc.cpa.tone1(..10).segment1(..3).offtime.min
mpc.cpa.maxrings mpc.cpa.tone1(..10).segment1(..3).offtime.max

Appendix B: Media Control Platform Reference Information SIP Headers
SIP Headers
Table 96 lists the SIP headers that the Media Control Platform recognizes and
uses. You can use values from many of these headers to send and receive data
to and from the VoiceXML or CCXML application in SIP INFO messages.
Do not use the header names in Table 96 for any custom headers, or they will
be ignored.
Table 96: SIP Headers Known to GVP
SIP Header Standard/Specification Description

Name (Section)
Accept RFC 3261 (20.1) When responding to a SIP OPTIONS request, lists all
the content types accepted by the component.
Allow RFC 3261 (20.5) When responding to a SIP OPTIONS request, lists all
the methods supported by the component.
Call-ID RFC 3261 (20.8) Standard support.
Contact RFC 3261 (20.10) Forms the remote request URI in a dialog.
Content-Length RFC 3261 (20.14) Standard support.
Content-Type RFC 3261 (20.15) Supported content types:

• application/dtmf-relay
• application/sdp
• application/text
• application/www-form-urlencoded
• message/sipfrag;version=2.0
• telephone/event
CSeq RFC 3261 (20.16) Standard support.
Diversion draft-levy-sip-diversion (08) Exposed to the application as a read-only redirection

variable if the History-Info header is not available.
Event RFC 33515 Supported event package:

• refer
From RFC 3261 (20.20) Contains the calling party information (ANI). Maps
to the VoiceXML session variable
session.connection.remote.uri.
User’s Guide 445

Table 96: SIP Headers Known to GVP (Continued)

Name (Section)
History-Info RFC 4244 The list of header values that are exposed at the
application layer as the redirection variable. Maps to
the VoiceXML session variable
session.connection.redirect.
• Original Called Number (OCN) is treated as the
first entry in the History-Info header.
• Redirection Reason is treated as a list of all
reasons in the History-Info header values.
Min-Expires RFC 4028 (5) Minimum session timer.
Max-Forwards RFC 3261 (20.22) Standard support.
P-Alcatel-CSBU Sets the header of the 200OK response to the initial

incoming INVITE. This is used for ESS deployments.
P-Asserted- RFC 3325 Provides the calling party information (ANI) if the
Identity From header is anonymous.
If this header exists, its value overrides the From
header as the ANI.
P-Called-Party_I Provides the DNIS if the To header is anonymous.

D If this header exists, its value overrides the To header
as the DNIS.
Privacy RFC 3323 Sets the Presentation Indicator of the VoiceXML

session variable session.connection.redirect.
Reason RFC 3326 If the Reason header is in the BYE message, the reason
text will be available as a read-only variable in the
application.
Record-Route RFC 3261 (20.30, 16.12.1) Specifies the routeset when sending requests within
the dialog.
Refer-To RFC 3515 (2.1) Sets the destination of the transfer request.
Replaces RFC 3891 Sets the dialog to replace for whisper transfer.


Name (Section)
Require RFC 3261 (20.32) Supported option tags:

• 100rel (PRACK not supported)
• timer
If the Media Control Platform receives tags that it
does not understand, it rejects the request with 420
Bad Extension.
Route RFC 3261 (20.34) Sets the next hop address when sending a request.
You can set the value with the application or by
configuration.
If the INVITE request contains Record-Route headers,
Record-Route values override the configured ruttiest
for all requests within the dialog.
RSeq RFC 3262 (7.1) Sent by the User Agent Server (UAS) on a reliable
response.
Rack RFC 3262 (7.2) Sent by the User Agent Client (UAC) to
acknowledge (ACK) a reliable response.
PRACK RFC 3262 Supports reliable provisional responses for a SIP

INVITE.
Session-Expires RFC 4028 (4) Sets the session expiry time and the refresher role.
Subscription RFC 3515 Supported by the REFER method only.

State
Supported RFC 3261 (20.37) Supported option tags:

• 100rel (PRACK not supported)
• timer
To RFC 3261 (20.39) Contains the called party information (DNIS). Maps
to the VoiceXML session variable
session.connection.local.uri.
Unsupported RFC 3261 (20.40) Contains the list of option tags not supported by the
User Agent (UA) when rejecting a call.
Via RFC 3261 (20.42) Standard support.
User’s Guide 447


Name (Section)
Warning RFC 3261 (20.43) Returned by a UAS when it failed to negotiate a

media session or the request contained a malformed
NETANN request. The following warning codes are
used in the following situations:
• 300 - incompatible network protocol
• 301 - incompatible network address
• 302 - incompatible transport protocol
• 303 - incompatible bandwidth
• 304 - unsupported media type
• 305 - unsupported media format
• 306 - unknown attribute not supported
• 307 - unknown parameter was presented
• 399 - malformed request URI (malformed
NETANN request)
X-Genesys-CallU Genesys UUID, generated by SIP Server (or

UID T-Server).
X-Genesys-GVP- GVP Session Identifier, generated by the Resource

Session-ID Manager (for new inbound sessions) or the Media
Control Platform or the Call Control Platform (for
new outbound sessions).
X-Genesys-RM- The DBID of the IVR Profile (in other words, the
Application-dbid VoiceXML or CCXML application).
X-Genesys-GSW The indicator that the SIP Server has received a

-Predictive-Call request to make an outbound call.
X-Genesys-GSW The ID of the IVR Profile.

-IVR-Profile-id
X-Genesys-Outb The indicator that there is user-data attached to the

oundData Outbound call.
X-Genesys-GSW The user-data parameter for CTI Connector

-Session-DBID integration with Supplementary Services Gateway.

Appendix B: Media Control Platform Reference Information Handling Error Responses for Outbound Calls
Handling Error Responses for Outbound

Calls
Table 97 summarizes how the Media Control Platform interprets SIP error
responses that it receives in response to outgoing INVITE requests.
For information about SIP response codes that the Media Control Platform
generates, see Appendix D, “SIP Response Codes,” on page 465.
Table 97: Error Response Handling—Outbound Calls
SIP Response Call End Disconnect Action in VoiceXML Application

Reason (for Reason (to
Code Phrase Metrics) Determine Call/
Transfer Result)
301 Moved baddest CM_DISCREASON_ error.connection.baddestination

Permanently BADDEST event during <transfer>
404 Not Found
410 Gone
484 Address
Incomplete
502 Bad Gateway
604 Does Not Exist

Anywhere
401 Unauthorized noautho CM_DISCREASON_ error.connection.noroute event

OUT_NOAUTH during <transfer>
402 Payment
A <transfer> form value of unknown
Required
is assigned.
403 Forbidden
407 Proxy
Authentication
Required
408 Request noanswer CM_DISCREASON_ noanswer in the <transfer> result

Timeout OUT_NOANSWER
User’s Guide 449

Appendix B: Media Control Platform Reference Information VAR Metrics
Table 97: Error Response Handling—Outbound Calls (Continued)
SIP Response Call End Disconnect Action in VoiceXML Application

Reason (for Reason (to
Code Phrase Metrics) Determine Call/
Transfer Result)
480 Temporarily busy CM_DISCREASON_ busy in the <transfer> result

Unavailable OUT_USERBUSY
486 Busy Here
600 Busy
Everywhere
603 Decline
405 Method Not unsupported CM_DISCREASON_ error.unsupported.transfer.blind/

Allowed UNSUPPORTED consultation/bridge event during
<transfer>
488 Not
Acceptable
Here
501 Not
Implemented
606 Not
Acceptable
503 Service resourcelimit CM_DISCREASON_ error.connection.noresource event

Unavailable OUT_NORESRC during <transfer>
504 Gateway busy CM_DISCREASON_ network_busy in the <transfer>

Timeout OUT_NWBUSY result
No response
All other errors error CM_DISCREASON_ error.connection.noroute event

GENERROR during <transfer>
A <transfer> form value of unknown
is assigned.
VAR Metrics
Table 98 summarizes the metrics that the Media Control Platform generates
when the Next Generation Interpreter (NGI) executes a VAR-specific <log>
tag. The metrics include the PCDATA specified in the <log> element.

Table 98 includes information about the valid syntax and values for the
VAR-specific <log> tag. If the format of the PCDATA for the element does not
conform to the valid syntax, the VAR metric will not be logged.
For more information about using the VAR <log> tag labels (or extensions) in
VoiceXML applications, see the Genesys Voice Platform 8.x Genesys
VoiceXML 2.1 Reference Help.
Formatting Note
Contrary to type conventions in the remainder of this guide, italic text in the
<log> tag syntax indicates placeholders for user-specified values. The angle
brackets are a required part of the VoiceXML syntax.
Table 98: VAR <log> Tags and Metrics
Metric <log> Tag Label Syntax and Valid Values
call_result <log label="com.genesyslab.var.CallResult">result[|reason]</log>

where:
• result is SUCCESS|FAILED|UNKNOWN. (The default is UNKNOWN.)
• reason is an optional string of up to 256 characters that provides a textual reason
for the call result.
Notes
• The result and reason values are not case-sensitive.
• If the developer specifies a call result other than SUCCESS or FAILED, UNKNOWN is
assumed.
• Preceding and trailing white space in the result is ignored.
• The system will truncate reason content beyond 256 characters.
call_notes <log label="label=com.genesyslab.var.CallNotes">notes</log>

where notes are up to 4 KB (4096 bytes) of free-form notes associated with the call.
Notes
• The notes collection cannot be empty.
• The system will truncate content beyond 4 KB.
User’s Guide 451

Table 98: VAR <log> Tags and Metrics (Continued)
ivr_action_start <log label="com.genesyslab.var.ActionStart">actionID[|parentID=PID]</log>

where:
• The actionID is the ID of the VoiceXML application action being started.
• PID is the ID of the parent action, if this action is nested inside some other active
action.
Notes
• The actionID and PID IDs are any valid UTF8 string, to a maximum of
64 characters, that does not contain spaces or pipes.
• Action IDs are case-sensitive.
• White space is ignored.
• An active action is one that has started and not yet ended. If a specified PID is not
the ID of an active action, the reporting infrastructure will ignore the
ivr_action_start metric.
ivr_action_end <log label="com.genesyslab.var.ActionEnd">actionID[|result[|reason]]</log>

where:
• The actionID is the ID of the VoiceXML application action being ended.
• The result value is one of SUCCESS|FAILED|UNKNOWN, indicating the result of the
action. The default is UNKNOWN.
• The reason value is an optional string of up to 256 characters that provides a
textual reason for the action result.
If ActionEnd is not explicitly specified, the Reporting Server implicitly ends actions
under the following circumstances:
• A sibling action (in other words, an action with the same parent) is started.
• The call ends.
If the Reporting Server implicitly ends an action, the value of result is UNKNOWN, and
the value of reason is NULL.
Notes
• The actionID is any valid UTF8 string, to a maximum of 64 characters, that does
not contain spaces or pipes.
• The actionID is case-sensitive.
• The result and reason values are not case-sensitive.
• White space in the metric is ignored.
• If the specified actionID is not the ID of an active action, the reporting
infrastructure will ignore the ivr_action_end metric.
• If the developer specifies an action result other than SUCCESS, FAILED, or UNKNOWN,
the reporting infrastructure will ignore the ivr_action_end metric.

Table 98: VAR <log> Tags and Metrics (Continued)
ivr_action_notes <log label="com.genesyslab.var.ActionNotes">actionID|notes</log>

where:
• The actionID is the ID of the VoiceXML application action.
• The notes variable is a collection of up to 4 KB (4096 bytes) of free-form notes
associated with the action.
Notes
• The actionID is any valid UTF8 string, to a maximum of 64 characters, that does
not contain spaces or pipes. Preceding and trailing white space is ignored.
• The notes collection cannot be empty.
• Any notes content beyond 4 KB will be truncated.
• VoiceXML action notes (ivr_action_notes) may be logged during the specified
action or after it has ended.
custom_var <log label="com.genesyslab.var.CustomVar">name|value</log>

where:
• The name is the name of the custom variable.
• The value is the value of the custom variable.
Notes
• The name variable is any valid UTF8 string, to a maximum of 64 characters, that
does not contain spaces or pipes. Preceding and trailing white space is ignored.
• The value variable is any valid UTF8 string, to a maximum of 256 characters.
White space is significant.
• Custom variables may be specified at any point in a VoiceXML application.
• The reporting infrastructure will allow a maximum of eight (8) custom variables to
be specified for a given call. Any variables logged beyond the maximum will be
ignored.
User’s Guide 453


Appendix
C Tuning Call Progress

Detection
This appendix provides information about how to finely tune the behavior of
Call Progress Detection (CPD) on the Genesys Voice Platform (GVP) to
facilitate diagnostics of unsuccessful calls and better manage contact center
campaigns.

Call Progress Detection, page 455
Call Progress Detection

Call Progress Detection enables detection and identification of telephone
network call progress tones. These tones identify conditions, such as network
congestion, busy conditions, and ringback (alerting). Tone detection can
provide improved diagnostics of the conditions when a caller cannot be
reached successfully. It also provides improved management of call
campaigns, in which calls that receive certain SIT results can be removed from
the campaign, rather than retried.
CPD matches pre-defined tones against the audio that is received from the
network. CPD is typically applied in the pre-connect state, before the call is
answered. In addition, in certain cases, it is useful to continue monitoring for
CPD tones after the connection is made.
The following sections contain information about the supported CPD tones and
how to use GVP configuration options to tune CPD on your platform:
• “Supported North American SIT Tones” on page 456
• “Tone Definition” on page 456
• “Answering Machine Detection” on page 460
• “Beep Detection” on page 462
User’s Guide 455

Appendix C: Tuning Call Progress Detection Call Progress Detection
Supported North American SIT Tones

The Media Control Platform supports North American CPD tones (by default),
which include:
• Fax
• Busy
• Fast Busy/Congestion
• Ringback
• Special Information Tones (SIT), which include:

No circuit

Reorder

Operator Intercept

Vacant circuit
• Ten additional custom tones
Tone Definition
Tones are either predefined, or custom defined in the platform configuration.
Each tone definition includes up to three segments, in which each segment can
contain one or two audio frequency bands that must be present for detection to
occur. The duration of each segment and the pause between segments can also
be configured as part of the tone definition.
For example, the parameters in Table 99 are required to define the SIT No
Circuit tri-tone:
Table 99: SIT ‘No Circuit’ Tone Definition
Segment Frequency 1 Frequency 2 Tone ‘On’ time Tone ‘Off’ time

Min-Max (Hz) Min-Max (Hz) Min-Max (ms) Min-Max (ms)
1 950-1020 0-0 320-440 0-60
2 1400-1450 0-0 320-440 0-60
3 1740-1850 0-0 320-440 0-100
By assigning these values and ranges to the segments of the tone definition,
you are enabling the detection of the SIT No Circuit tri-tone during CPD.
The accuracy of frequency detection is +/- 10 Hz for a signal level that is equal
to the nominal level for the North American Numbering Plan. (See Supplement
2 to ITU.T Recommendation E.180.)
The detection result is included in the MSML fragment that is executing on the
Media Control Platform and is passed to the application in a SIP INFO
message.

Pattern Types
CPD behavior is controlled by a list of pattern types. Each pattern type (see
“Supported North American SIT Tones” on page 456) has an associated
configuration option, in which a list of one or more tone definitions (that are
mapped to the pattern type) is configured. If the configuration option for a
pattern type is not set, detection of that pattern type is disabled.
The list of available tone definition names consists of a set of built-in standard
tones and a number of configurable tones. The built-in tones currently include
North American definitions for ringback, busy, fast busy, fax, and SIT. These
tone definitions are named na_ringback, na_busy, na_sit, and standard_fax.
Preconnect, Postconnect and Custom Tones

You can also configure ten additional custom tone definitions, named tone1
through tone10. These custom tone definitions are the vehicle for localization.
You can localize the custom tones by configuring them to match the local CPD
tone requirements, and then assigning them to tone patterns.
Preconnect Tones Preconnect tones (busy, fast busy, ringback, SIT, and custom defined tones),
can be detected in both preconnect and postconnect states. You can enable the
detection mode by configuring the mpc.cpa.preconn_tones_det_mode option
with a value of 1. By default, this mode is disabled and preconnect tones are
detected in the preconnect state only.
The transition to the connected state (where postconnect detection occurs) can
be triggered by using out-of-band signaling, or in the event that ringback is
used, when ringback stops (assuming the maximum number of rings is not
exceeded).
Fax tone is a postconnect tone and has a treatment different from that of
preconnect tones. As a result, assigning a fax tone to a preconnect tone pattern
or assigning a preconnect tone to the fax pattern (for example, assigning a fax
tone to a busy pattern) is not recommended. If a custom fax tone is required,
any one of the ten custom tones can be used for this purpose.
Postconnect In the postconnect state, the returned result can be fax, human, or answering
Tones machine. Optionally, the Media Control Platform can retain a configurable
amount of media (received before the connection is available) for postconnect
processing. This enables you to control (through the configuration) scenarios
in which the audio path to the caller might be open before a connect signal is
received. Use the mpc.cpa.keptdur_before_statechange configuration option,
in which a time interval (duration) is defined in milliseconds to control this
behavior.
Custom Tones Custom tones are defined by using a set of configuration options for each tone.
The configuration of each of the three segments in a tone includes the
frequency range for one of two frequencies—the on time range for the
segment, and the off time range, which marks the end of the segment.
User’s Guide 457

In the following list, <m> is the custom tone identifier (1 through 10), <n> is the
segment identifier within the tone (1 through 3), the frequencies are identified
as f1 and f2, and the min and max configuration options define a range:
• mpc.cpa.tone<m>.segment<n>.f1min
• mpc.cpa.tone<m>.segment<n>.f1max
• mpc.cpa.tone<m>.segment<n>.f2min
• mpc.cpa.tone<m>.segment<n>.f2max
• mpc.cpa.tone<m>.segment<n>.ontimemin
• mpc.cpa.tone<m>.segment<n>.ontimemax
• mpc.cpa.tone<m>.segment<n>.offtimemin
• mpc.cpa.tone<m>.segment<n>.offtimemax
When f2 values are specified, a second frequency for the segment is implicitly
enabled. When segment2 values are specified, a second segment is implicitly
enabled, and when segment3 values are specified, a third segment is enabled.
These configuration options match the data in the tone definition examples in
Monitoring When the Media Control Platform monitors ringbacks to detect the connection,
Ringbacks the mpc.cpa.maxrings configuration option is used to define an upper limit for
the number of ringbacks. If the number of ringbacks exceeds the value of this
option, a timeout result is returned, and it is assumed that the call was not
answered. The timeout feature is disabled when this option is configured with
a value of 0.
Time Limits The Media Control Platform supports the configuration of time limits for CPA
detection. A timeout result is returned if the timeout interval expires. Three
timeout intervals are supported:
1. Timeout interval for the call to advance from the preconnect to the
connected state (mpc.cpa.maxpreconntime).
2. Timeout interval for the call fax, human or answering machine detection
occurring after the start of the connected state (mpc.cpa.maxpostconntime).
3. Timeout interval for answering machine beep after answering machine
result-type detection (mpc.cpa.maxbeepdettime).
If any of these options are configured with a value of 0, the timeout is
disabled.
Configuration
You can provision CPD in one of two ways:
• Configuring the Media Control Platform—The values that are set in the
[mpc].cpa configuration options enable CPD detection for all calls that
land on this particular Media Control Platform.

• Configuring the IVR Profiles—The values that are specified in the IVR
Profile service parameters enable IVR application-level CPD tuning and
override the Media Control Platform system-wide configuration options.
Calls that use these particular IVR Profiles will have customized
CPD-related configurations.
For example, an option is configured in Configuration Manager as follows:
[mpc].cpa.voice_range_db=25
However, the IVR Profile’s gvp.service-parameters section has the
name/value pair configured as follows:
Name: voicexml.gvp.config.mpc.cpa.voice_range_db
Value: fixed, 20
When Resource Manager routes a VoiceXML application to Media Control

Platform with this particular IVR Profile, it invokes the VoiceXML
application, which includes the mpc.cpa.voice_range_db=20 option as the
session parameter setting in the URI. The Media Control Platform honors the
session parameter setting first, so the human voice detection is based on a
voice range of 20 decibels.
For MSML services, you can set this service parameter in the
gvp.service-parameters section of the default IVR Profile. For example:
Name:cpd.gvp.config.mpc.cpa.voice_range_db
Value: fixed, 20
This configuration will then apply to human voice detection in all MSML
services that are included in the CPD request.
For a complete list of CPA options that can be overwritten, see Table 95, “CPA
Options That Can Be Overwritten,” on page 444.
Tuning
This section contains useful information that can be used to tune CPD
performance.
CPD performance is controlled by tone patterns and tone definitions. To
diagnose CPD issues:
1. Carefully examine the configuration to ensure that the tone patterns
include appropriate tone definitions that are accurately configured for your
locale and telephony network.
2. If the configuration is correct, collect CPD recordings and results, and
analyze them to determine if there are any systemic issues. For example, is
the CPD audio in the recorded files valid?
User’s Guide 459

3. Ensure that the audio cut-through is taking place prior to connect. If not, it
is likely that the only available CPD results are being delivered by the SIP
network itself. For example, if a media gateway is being used to connect to
the TDM network, and audio cut-through does not occur, the only available
results are from the gateway itself.
Logging
You can enable or disable CPD-related information logging by using the
following two configuration options:
• [mpc].cpa.enable_log_param (Value: true or false)
• [mpc].cpa.enable_log_result (Value: true or false)
When the [mpc].cpa.enable_log_param is set to true, the Media Control
Platform logs all of the configuration option values that are used for CPD
detection, whenever CPD detection is initiated by an MSML request, or in a
Metrics Log The log is in metrics log format, which includes the Global Call ID, timestamp,
Format and configuration information. See the following example of the metrics log
format:
max_preconnect_time=30000|max_postconnect_time=20000|max_beep_det_time=
30000|no_limit_timeout=30000|chunks_not_flush_on_state_chg=90000|machin
e_greet_dur=1800|voice_pause_dur=1000|max_voice_signal_dur=800|fax_dura
tion=160|voice_range_db=25|voice_level_db=17.5|max_ring_cnt=9|sil_befor
e_beep=4500|preconnect_tone_det_mode=0|notime_ringback_match_percent=50
|ontime_preconnect_match_percent=60
Tone Setting In addition, the MCP logs the tone-setting information in the metrics log
Information format. See the following example of the tone setting information:
ringbak=tone1|segment=1,f1min=0,f1max=0,f2min=0,f2max=0,ontimemin=20,on
timemax=20,offtimemin=0,offtimemax=0|segment=2,f1min=0,f1max=0,f2min=0,
f2max=0,ontimemin=20,ontimemax=20,offtimemin=0,offtimemax=0|segment=3,f
1min=0,f1max=0,f2min=0,f2max=0,ontimemin=20,ontimemax=20,offtimemin=0,o
fftimemax=0
When the [mpc].cpa.enable_log_result option value is set to true, the Media
Control Platform logs all of the CPA results that are reported by the Media
Control Platform. The CPA result log is in the metrics log format, which
includes the Global Call ID, timestamp, and CPA result. See the following
example of a CPA result:
cpa_result Answering machine detected
Answering Machine Detection

When the call moves to the postconnect state, Answering Machine Detection
(AMD) begins. An answering machine typically delivers a greeting that is
different from that of an actual person, or one that is used by a business. An

answering machine or service often plays a beep tone to prompt the caller to
leave a message, which is then recorded. As a result, a number of parameters
and internal heuristics control assessment of the postconnect media, and drive
a categorization of the entity that is answering the call.
Configuration
The Media Control Platform enables the configuration of three categories to
control AMD behavior:
• Human pause time
• Maximum human voice time
• Answering machine greeting time
Profile Types In addition, the Media Control Platform supports three AMD profiles, which
provide different weightings, based on the expected demographic that is
receiving the calls. For example, when detection is ambiguous:
• Normal profile—Does not favor either human or answering machine
detection.
• Answering machine profile—Favors answering machine detection.
• Human profile—Favors human detection.
These profiles, each defining separate values, can be selected on a per-call
basis by using MSML.
Use the following options to configure the profile types:
• mpc.cpa.priority_normal_machinegreetingdur
• mpc.cpa.priority_normal_voicepausedur
• mpc.cpa.priority_normal_maxvoicesigdur
• mpc.cpa.priority_voice_machinegreetingdur
• mpc.cpa.priority_voice_voicepausedur
• mpc.cpa.priority_voice_maxvoicesigdur
• mpc.cpa.priority_machine_machinegreetingdur
• mpc.cpa.priority_machine_voicepausedur
• mpc.cpa.priority_machine_maxvoicesigdur
Signal-to-Noise The Media Control Platform supports two configuration options related to the
Ratio signal-to-noise ratio, and the level of speech that is detected in the input signal.
• mpc.cpa.voice_range_db—Specifies the minimum dynamic range (ratio of
maximum-to-minimum energy level) for which each section of the
received media is considered to contain an active signal (in decibels).
• mpc.cpa.voice_level_db—Specifies the active voice signal level (in
decibels) relative to the maximum.
User’s Guide 461

Tuning
Use the following configuration options to tune AMD:
• voice_range_db—The signal-to-noise ratio, which indicates voice traffic.
• machinegreetingdur—The duration of time after connection, which
indicates a machine greeting. If the voice signal is longer than this
duration, the input is likely to be considered an answering machine. If the
voice signal is shorter than this, and longer than maxvoicesignal duration,
the result is weighted, based on the profile that is in use.
• maxvoicesigdur—The duration of time after connection, which indicates a
voice signal. If the voice signal is shorter than this duration, the input is
likely to be considered human. If the voice signal is longer than this, and
shorter than machinegreetingdur duration, the result is weighted, based on
the profile that is in use.
• voicepausedur—The amount of silence that indicates the end of AMD.
If you observe: Make the following adjustments:

• Longer human greetings that are Increase the value of the maxvoicesigdur,
being identified as a machine. and machinegreetingdur configuration
options.
• Noisy greetings not identified as Reduce the value of the voice_level_db
human. configuration option.
• Takes too long to connect after Reduce the value of the voicepausedur
human greeting. configuration option.
Beep Detection
The Media Control Platform supports optional answering machine beep
detection. The request for beep detection is passed as part of the MSML
fragment within the request for CPD/AMD. Beep detection takes place as part
of AMD and enables identification of the beep tone that usually follows an
answering machine greeting. This phase of detection, which is indicated by a
period of silence, then a beep, followed by another period of silence, has the
following characteristics:
• A transition from low energy to a period of strong energy in the signal.
• Detection of a beep that has one or two strong frequencies present in the
signal.
• The presence of this signal for a minimum amount of time followed by a
minimum period of silence.
Note: This feature detects any single or dual frequency response in the
signal.

Configuration
The mpc.cpa.maxbeepdettime configuration option controls how long the
Media Control Platform waits for a beep detection result after an answering
machine has been identified. When this option value is set to 0, the timeout
feature is disabled.
The mpc.cpa.maxsil_before_beep configuration option controls the maximum
amount of silence allowable before a beep. If this value is exceeded, beep
detection is abandoned and a silence timeout result is returned.
The minimum on time, and the minimum off time are currently not
configurable.
Tuning
The following configuration option is the only one that impacts beep detection:
maxsil_before_beep—The maximum amount of silence allowable prior to the
detection of a beep.
If you observe: Then adjust:
Delays before connection to the Increase or decrease the beep detection
application, or beeps not being detected timeout.
because the application has already
started.
Continuous Tone Detection

Beginning with release 8.1.6, you can configure CPA to detect a continuous
tone, to satisfy specific—but not universal—conditions. This configuration
provides a mechanism in the Media Server CPD to handle continuous tone
detection, by making the definition of continuous configurable.
The UK, Ireland, UAE and at least 7 other countries use a continuous tone for
the states confirmation, howler, paytone, or number unattainable.
In these environments, where outbound calling applications need to handle the
state number unattainable (which is a continuous tone), set the value of
ontimemax to 0 (zero), to disable checking the ontime against a maximum
value. As a result, any tone lasting longer than the ontimemin value is
considered to be “continuous.”
User’s Guide 463


Appendix
D SIP Response Codes

This appendix lists the Session Initiation Protocol (SIP) responses that
Genesys Voice Platform (GVP) components send or receive in response to
error conditions and other events.
It contains the following section:
 SIP Responses to Inbound Calls, page 465
For information about how the Media Control Platform handles error
responses that it receives for outbound call requests, see Table 97 on page 449.
SIP Responses to Inbound Calls

Table 100 summarizes the SIP response codes, other than the normal 200 OK
responses, that the Resource Manager (RM), Media Control Platform (MCP),
and Call Control Platform (CCP) signal in response to error conditions and
other events during incoming and outbound call setup and processing.
Table 100 also lists the configuration options to customize those responses,
where applicable.
Resource Manager handles SIP responses from other components in
accordance with rules described in the Genesys Voice Platform 8.5
Deployment Guide.
User’s Guide 465

Appendix D: SIP Response Codes SIP Responses to Inbound Calls
Table 100: SIP Response Codes
SIP Response Sent By Situations Configurable Options and

Notes
Code Phrase
100 Trying MCP The immediate response to a

CCP valid INVITE request.
RM
PSTN
Connector
180 MCP The default intermediate sip.sendalert (see page 181)

Ringing response to an INVITE request.
CCP Intermediate response is sent for ccpccxml.sip.send_

all incoming calls, on <accept> progressing
(no media bridge configured).
Depending on configuration:
• Response is sent when <send>
is called.
• Response is sent immediately
after sending 100 Trying.
PSTN The default intermediate

Connector response to an INVITE request.
183 Session MCP The non-default intermediate sip.sendalert (see page 181)
Progress response, which includes SDP
information.
MCP An incoming call is being offered Sent if debugging is enabled on

to the Next Generation the MCP:
Interpreter (NGI) for debugging. vxmli.debug.enabled
The NGI passes the debugger IP
address and port information to
the calling party in the following
SIP headers:
• X-GVP-NGI-DEBUG-IP
• X-GVP-NGI-DEBUG-PORT
Note: The information can also
be sent in INVITE messages.

Table 100: SIP Response Codes (Continued)

Notes
Code Phrase
183 Session CCP Intermediate response sent for If the media bridge requires
Progress incoming calls when a media changes, they are implemented
(continued) bridge has been configured through subsequent SDP
between this bridge and any other updates in re-INVITE, 200 OK,
endpoint. The response includes and 183 messages.
the appropriate SDP content. Note: When a BYE is received
on a SIP dialog that is
associated with an endpoint
while a transition involving
that endpoint is being executed,
any new bridge involving the
endpoint will fail. The
error.connection.
join event is thrown, with an
empty Reason property.
202 MCP A REFER request to initiate an

Accepted outbound call outside of a SIP
dialog is accepted by the
PSTN A REFER request to initiate

Connector outbound call for Dialogic Blind
Transfer or AT&T Out-of-band
Transfers.
3xx [Various] MCP The MCP, acting as a User Agent See Warning header
Server (UAS), failed to negotiate information in Table 96 on
a media session or the NETANN page 445.
request was malformed.
302 Moved CCP The platform is redirecting a call To customize the SIP response
Temporarily in the ALERTING state (<redirect> code for specific situations, use
tag). the <hints> attribute of the
If the CCXML application <redirect> tag—the
specifies a <reason> attribute, the responseCode property of the
reason is included in the text hints object specifies the
portion of the Reason header. response code that is to be
used.
User’s Guide 467


Notes
Code Phrase
400 RM Malformed Request-URI in a

Bad Request REGISTER message.
MCP Malformed Request-URI in an

INVITE message.
CCP The initial CCXML page URI is

malformed.
PSTN Malformed protocol. PSTN

Connector Connector failed to extract the
SDP message from the INVITE.,
or the custom header for the CPA
is incorrect.
403 Forbidden RM The domain name in a REGISTER

request does not match the
configured domain name.
RM Call is rejected because of an IVR Profile:

IVR Profile policy (dialing rule). gvp.policy.dialing-rule-
forbidden-respcode
Note: An equivalent
configuration option also
enables an alarm to be set.
RM Service request is rejected IVR Profile:

because the IVR Profile policy gvp.policy.conference-
does not allow the service in the forbidden-respcode
session.
gvp.policy.external-sip-
forbidden-respcode
gvp.policy.outbound-call-
forbidden-respcode
gvp.policy.transfer-forbidd
en-respcode
gvp.policy.voicexml-dialog-
forbidden-respcode
Note: Equivalent configuration
options also enable an alarm to
be set.


Notes
Code Phrase
404 Not Found RM The Resource Manager could not

match the incoming request to an
IVR Profile.
RM The Resource Manager could not

match the incoming request to a
service.
MCP The Request-URI has conf as the

user part, but does not have a
conf-id parameter.
PSTN The PSTN Connector received an

Connector unauthorized number or invalid
digits from the TDM network.
405 Method Not RM REFER, OPTIONS, SUBSCRIBE, or

Allowed INFO message was sent outside of
an existing SIP dialog.
408 Request RM The Resource Manager does not

Timeout receive a response from the
resource.
RM The Resource Manager has not

received a response from any
resource.
RM An INVITE request specifies a

host for which the Resource
Manager is not responsible. (By
default, no responsible domains
are specified, so that all requests
are accepted.)
PSTN The PSTN Connector did not

Connector receive a response from the TDM
network for an outbound call.
415 Unsupported PSTN The PSTN Connector received an

Media Type Connector unsupported media type.
420 Bad MCP The MCP rejects an incoming

Extension call that requires 100rel (SIP
Provisional Message Reliability).
User’s Guide 469


Notes
Code Phrase
423 Interval Too RM The Resource Manager received

Brief a REGISTER request for a
registration period that fell below
the configured minimum expiry
time.
480 Temporarily RM A conference call fails because of rm.conference-sip-error-

Unavailable insufficient resource port respcode
capacity or because the
conference has reached
maximum size.
RM The Resource Manager is not rm.resource-unavailable-

able to select a resource to respcode
forward the request to, because a
suitable resource is not available.
RM The Resource Manager is not rm.resource-no-match-

able to select a resource to which respcode
to forward the request, because
the deployment does not include
the required resource.
RM Call is rejected because usage IVR Profile:

limits, as specified in the IVR gvp.policy.usage-limit-
Profile policy, have been exceeded-respcode
exceeded.
Note: An equivalent
CCP The platform is rejecting an ccpccxml.defaultrejectcode

incoming connection in the To further customize the SIP
ALERTING state (<reject> tag). response code for specific
If the CCXML application situations, use the <hints>
specifies a <reason> attribute, the attribute of the <reject> tag—
reason is included in the text the responseCode property of
portion of the Reason header. the hints object specifies the
response code that is to be
used.
CCP The platform is not in READY ccpccxml.defaultrejectcode

state, and is therefore rejecting all
INVITE and OPTIONS requests.


Notes
Code Phrase
PSTN There are no ports available for

Connector an outbound call, or there is no
answer from the TDM network.
481 Call Does MCP The platform received a request

Not Exist PSTN that does not match with any
Connector existing dialog.
484 Address PSTN The DNIS is not present in the

Incomplete Connector TO header of the INVITE
message.
487 Request MCP A CANCEL or BYE is received

Terminated before the final response to the
INVITE is sent.
488 Not CCP An endpoint’s SDP capabilities The response includes a

Acceptable cannot be obtained. Warning header with warning
Here code 399, and one of the
following warning text
messages:
• Unable to generate an
offer—The INVITE
contained no SDP.
• Unable to generate an
answer—All other
situations.
User’s Guide 471


Notes
Code Phrase
500 Server RM The Resource Manager received

Internal a REGISTER request from a
Error resource about which it had no
information from Management
Framework.
RM The Resource Manager does not

receive a 2xx response from any
resource (see the Genesys Voice
Platform 8.5 Deployment
Guide).
MCP Unable to create a media session The response includes an

to handle the call. explanatory Warning header
(see Warning header
information in Table 96 on
page 445).
MCP The MCP is unable to fetch or

parse the VoiceXML document.
PSTN The PSTN Connector failed to

Connector make an outbound call.
503 Service RM The Resource Manager is in rm.suspend-mode-respcode

Unavailable suspend mode when a new
session request arrives.
CCP The CCP is unable to create a

CCXML session to handle the
call.


Notes
Code Phrase
503 Service RM Call is rejected because usage IVR Profile:

Unavailable limits for the service, as specified gvp.policy.ccxml-usage-limi
(continued) in the IVR Profile policy, have t-exceeded-respcode
been exceeded.
gvp.policy.conference-usage
-limit-exceeded-respcode
gvp.policy.voicexml-usage-l
imit-exceeded-respcode
Note: An equivalent
MCP A conference cannot be created

because of a resource problem
(for example, failed to join to
conference because of the
conference limit).
MCP The Media Server is not All error responses to an

accepting new calls for a reason INVITE request, if they do not
other than those that are covered involve SDP negotiation, will
by the 500 response. contain a Warning header with
a value of 399 and a
human-readable description of
the error.
MCP The VoiceXML application could

not be fetched or parsed.
CCP The platform is not in READY ccpccxml.defaultrejectcode

state, and is therefore rejecting all
HTTP requests to start a new
CCXML session.
PSTN The PSTN Connected received a

Connector “Service Not Available: from the
TDM network.
603 Declined PSTN The PSTN Connector failed to

Connector send a Redirection FACILITY
message in an AT&T Conference
Transfer.
User’s Guide 473


Notes
Code Phrase
BYE message CCP Bridge failure resulting from the The Reason header value is set
failure of endpoints to negotiate to Application Disconnect.
SDP might cause the CCP to
send SIP BYE messages to the
components involved.
[Configurable] RM The Resource Manager returns a rm.options_response_

response to a SIP OPTIONS contenttype
message. rm.options_response_msg_
body
[Configurable] SIP Server When the Resource Manager Resource Manager:

receives any of the configured <gateway resource
SIP responses from SIP Server group>.noresource-response-
for a request to a gateway code
resource, it will retry the request
on other gateway resources in the
logical resource group.

Appendix
E Device Profiles
This appendix provides details for configuring device profile and summarizes
the settings for the default device profiles that are provisioned for the Call
Control Platform and CTI Connector. It contains the following sections:

Device Profile Usage, page 475

Configuring Device Profiles, page 484
 Default Device Profiles, page 491
Device Profile Usage

The Call Control Platform and the CTI Connector use device profiles to
determine the behavior of the devices with which it interacts, in order to
produce the most appropriate SIP messages. Each call handled by Call Control
Platform or CTI Connector is associated with a device profile. The profile is
used to customize Session Description Protocol (SDP) information sent to the
device.
This section describes the device profile parameters and how each parameter
affects the SDP formation.
Sending SDP
This section describes the Call Control Platform and the CTI Connector
behavior based on device profile configuration when constructing the SDP.
Inbound Usage Examples

This section describes some usage examples for inbound calls.
User’s Guide 475

Appendix E: Device Profiles Device Profile Usage
Offer-less Initial INVITE Without Bridging

Example 1 Figure 40 shows that an SDP offer is generated in a183 response or a 200OK
response with the following settings:
• unjoined-initial-answer-pref equals none or connectionless-sdp
• connectionless-sdp-type equals hold
Caller CCP/CTI Connector
INVITE
200OK/183 (SDP Offer)
Figure 40: Offer-less INVITE for Inbound Call Example 1
This SDP offer contains a connection line with 0.0.0.0 as the IP address.
Example 2 Figure 41 shows an SDP offer that is generated in a 183 response or a 200OK
• connectionless-sdp-type set to non-routable
INVITE

Example 3 Figure 42 shows an SDP offer that is generated in a 183 response or a 200OK
• offer-answer-support equals true
• connectionless-sdp-type equals none
• nomedia-sdp-support equals true
or
• unjoined-initial-answer-pref equals nomedia-sdp
• nomedia-sdp-support equals true
INVITE
This SDP offer does not contain any media lines.

Example 4 Figure 43 on page 478 shows that CCP/CTI Connector is unable to decide
which method to use for generating an SDP offer with the following settings
configured:
• nomedia-sdp-support equals false
User’s Guide 477

INVITE
488 Not Acceptable
A SIP 488 response is generated to terminate the call.
Initial Inbound Offer Without Bridging

Example 1 Figure 44 shows that an SDP answer is generated in a 183 response or a 200OK
INVITE (SDP Offer)
183/200OK (SDP Answer)
Figure 44: Initial Inbound Offer Without Bridging Example 1
This SDP contains a connection line with 0.0.0.0 as the IP address.

• unjoined-initial-answer-pref equals reject-media
or
• unjoined-initial-answer-pref equals none
INVITE (SDP)
183/200OK (SDP)
This SDP response has all media lines in the offer with ports set to 0.
response with the following settings.
• offer-answer-support equals false
• nomedia-SDP-support equals true
or
• unjoined-initial-answer-pref equals nomedia-SDP
• nomedia-SDP-support equals true
or
User’s Guide 479

INVITE (SDP)
183/200OK (SDP)
This SDP response does not contain any media lines.

Example 4 Figure 47 shows that the platform could not determine a valid way to reject the
media based on the device profile information with the following settings:
• nomedia-SDP-support equals false
or
• nomedia-SDP-support equals false

INVITE (SDP)
488
The incoming call is rejected.
Outbound Usage Examples

This section describes some usage examples for outbound calls.
Offer-less Outbound INVITE

Example 1 Figure 48 shows an SDP answer in an ACK response that contains a
connection line with 0.0.0.0 IP address. The configuration settings are as
follows:
• offer-less-invite-support equals true
• unjoined-initial-offer-pref equals offer-less
• unjoined-initial-answer-pref equals connectionless-SDP
User’s Guide 481

CCP/CTI Connector Called Party
INVITE
200OK (SDP)
ACK (SDP)
Figure 48: Offer-less Outbound INVITE Example1
Example 2 Figure 49 shows an SDP offer that is generated in INVITE with the following
settings:
• connectionless-sdp-type equals non-routable
• offer-less-invite-support equals false
• unjoined-initial-offer-pref equals none
or
• connectionless-sdp-type equals non-routable
• unjoined-initial-offer-pref equals connectionless-SDP
.
CCP/CTI Connector Called Party
INVITE
Figure 49: Offer-less Outbound INVITE Example 2
Example 3 The platform cannot generate an outbound INVITE based on the device profile
with the following settings:
• offer-less-invite-support equals false

Example 4 Creating a dialog to Media Server is an exceptional case. The platform will
always set the ports in each media line to 0. This puts the call on hold if the
device supports offer and answer.
Offer-less Early Join

Figure 50 shows an early join of an inbound call to an outbound call. The
outbound initial INVITE is always offer-less regardless of its device profile
setting.
Caller CCP/CTI Connector Called Party
INVITE
INVITE
Figure 50: Offer-less Early Join Example
Receiving SDP
CCP and CTI Connector perform the following verifications when receiving
an SDP offer:
1. If the offer-answer-support parameter equals true, the number of media
lines in the offer can not be less than previously negotiated, or the call will
be terminated.
2. If the offer-answer-support equals false, and the new SDP contains more
media lines than previously received, the new media lines are ignored.
They will not be passed to any other calls that are joined.
CCP and CTI Connector perform the following verification when receiving an
SDP answer:
◆
If the offer-answer-support equals true, the number of media lines in the
SDP answer must be the same as the offer sent, or the call will be
terminated.
User’s Guide 483

Appendix E: Device Profiles Configuring Device Profiles
Configuring Device Profiles

The Call Control Platform and CTI Connector use device profiles to determine
the behavior of the devices with which they interacts, in order to produce the
most appropriate SIP messages.
Device Profile Configuration File

Device profiles are defined in the <Call Control Platform Installation
Directory>\config\ccpccxml_provision.dat and <CTI Connector Installation
Directory>\config\CTIC_Provision.dat files, which are installed when you
install the Call Control Platform, and the CTI Connector.
For the format and syntax of device profile entries in the configuration file, see
Procedure: Provisioning Device Profiles, on page 490.
Device Classes
Table 101 describes the properties that define the device classes
of the device profiles.
Table 101: Device Profile Class Properties
Property Description Valid Values
connectionless-sdp-type Support for Connectionless • hold—Use hold SDP (for example,

SDP. c=IN IP4 0.0.0.0 or
c=IN IP6 0.0.0.0.0.0.0.0).
Indicates the mechanism that
should be used to indicate null • non-routable—Use SDP with a
SDP (in other words, no media non-routable connection (for
streams) to the device. example, c=IN IP4 1.1.1.1 or
c=IN IP6 1.1.1.1.1.1.1.1).
distinct-send-recv-support Support for send-recv media • true—The Device supports a

lines. sendonly media line and a recvonly
Indicates if the offer-answer media line on separate connections.
user agent supports a sendonly • false—The Device does not support
media line and a recvonly a sendonly media line and a
media line on separate recvonly media line on separate
connections. connections.
multiple-recvonly-support Support for multiple recvonly • true—The Device supports

media lines. receiving the recvonly attribute in
Indicates if the offer-answer multiple SDP media lines.
user agent supports receiving • false—The Device does not support
multiple recvonly media lines. receiving the recvonly attribute in
multiple SDP media lines.

Table 101: Device Profile Class Properties (Continued)
nomedia-SDP-support Support for receiving SDP • true—The Device supports

containing no media lines. receiving SDP without media lines.
Indicates if the offer-answer • false—The Device does not support
user agent supports receiving an receiving SDP without media lines.
SDP containing no media lines.
The device answers with an
SDP with no media lines and
remains in a state with no media
connections until a re-INVITE.
offer-answer-support Support for the Offer-Answer • true—The Device supports the

model. offer-answer model.
Indicates if the device supports • false—The Device does not support
the offer-answer model the offer-answer model.
described in RFC 3264 (in other
words, whether the device will
respond to SDP offers with an
answer according to the rules
defined in the RFC).
offer-less-invite-support Support for INVITE requests that • true—The Device supports

do not contain an SDP offer. receiving INVITE requests that do not
Indicates if the device supports contain an SDP offer.
an INVITE request that does not • false—The Device does not support
contain an SDP offer. The receiving INVITE requests that do not
device responds with an SDP contain an SDP offer.
offer in its response to the
INVITE.
recvonly-support Support for recvonly media • true—The Device supports

lines. receiving the recvonly attribute in
Indicates if the offer-answer the SDP media line.
user agent supports the • false—The Device does not support
a=recvonly media line attribute receiving the recvonly attribute in
in SDP messages that it the SDP media line.
receives.
User’s Guide 485

restricts-media-source Support for receiving media • true—The Device supports

only from the User Agent (UA) receiving media from the UA to
to which media is being sent. which media is being sent, and only
from that UA.
Indicates if the offer-answer
user agent is able to receive • false—The Device supports
media from a UA other than the receiving media from the UA to
UA to which it is sending which media is being sent, as well as
media. from other UAs.
sendonly-support Support for sendonly media • true—The Device supports

lines. receiving the sendonly attribute in
Indicates if the offer-answer the SDP media line.
user agent supports the • false—The Device does not support
a=sendonly media line attribute receiving the sendonly attribute in
in SDP messages that it the SDP media line.
receives.
unjoined-initial-answer-pref Indicates the preferred method • connectionless-SDP—If the value

for performing an answer during of connectionless-sdp-type is
an initial INVITE when no anything other than none, a response
bridges have been established. with the specified
connectionless-SDP type will be sent
to the endpoint. If the value of
connectionless-sdp-type is none,
there is no preferred method (this
parameter is treated as if the
preference was none).
• reject-media—If the value of
offer-answer-support is true, a
response with SDP that rejects all
media lines (0 port) will be sent to
the endpoint. Otherwise, there is no
preferred method (this parameter is
treated as if the preference was
none).

unjoined-initial-answer-pref • nomedia-SDP—If the value of

(continued) nomedia-SDP-support is true and
the value of offer-answer-support
is false, a response with no-media
SDP will be sent to the endpoint.
Otherwise, there is no preferred
method (this parameter is treated as
if the preference was none).
• none—No preferred method. Other
device profile parameters determine
the method.
unjoined-initial-offer-pref Indicates the preferred method • offer-less—If the value of

for performing an initial INVITE offer-less-invite-support is
without establishing bridges. true, an INVITE without an offer
will be sent to the endpoint.
Otherwise, there is no preferred
method (this parameter is treated as
if the preference was none).
• connectionless-SDP—If the value
of connectionless-sdp-type is
anything other than none, an INVITE
with the specified
connectionless-SDP type will be sent
to the endpoint. If the value of
connectionless-sdp-type is none,
there is no preferred method (this
parameter is treated as if the
preference was none).
• none—No preferred method. Other
device profile parameters determine
the method.
Unjoined-initial-offer-pref
The unjoined-initial-offer-pref parameter controls the behavior when
generating the initial INVITE for an outbound call if the call is not explicitly
joined to any other calls.
If this parameter is set to none or does not exist in the device profile, the
following methods are used in the order given if enabled in the device profile:
1. offer-less-invite-support
2. connectionless-sdp
If neither method is enabled, the call will fail.
User’s Guide 487

If multiple methods are supported by the device, the

unjoined-initial-offer-pref parameter specifies the method to use.
During software initialization, the parameter has the following verifications,
and if it detects an error, the value of unjoined-initial-offer-pref will
default to none.
• unjoined-initial-offer-pref equals offer-less
Required—offer-less-invite-support equals true
Otherwise—unjoined-initial-offer-pref equals none
• unjoined-initial-offer-pref equals connectionless-sdp
Required—connectionless-SDP-type does not equal none
Otherwise—unjoined-initial-offer-pref equals none
• unjoined-initial-offer-pref equals none
unjoined-initial-offer-pref equals offer-less if
offer-less-invite-support equals true
Otherwise—unjoined-initial-offer-pref equals connectionless-sdp if
connectionless-SDP-type does not equal none
unjoined-initial-answer-pref Configuration
The unjoined-initial-answer-pref parameter controls the behavior when
generating the initial SIP response for an inbound call if the call is not already
joined to another call.
This parameter also applies to an outbound call if the offer-less INVITE
method is used to generate the INVITE. In this case, the received response will
contain an offer, and the unjoined-initial-answer-pref parameter controls
how the SDP answer is generated in the ACK message.
If this parameter is set to none or does not exist in the device profile, the
following methods are used in the order given for answering an SDP offer:
Note: This applies when receiving an offer-less INVITE for an inbound

call.
1. connectionless-sdp-type
2. reject-media (if offer-answer-support equals true)
3. nomedia-sdp-support (if offer-answer-support equals false)
4. Call Rejected with 488
If the unjoined-initial-answer-pref parameter is set to none or does not exist
in the device profile, the following methods are used in the order given for
generating an SDP offer:
1. connectionless-sdp-type

2. nomedia-sdp-support
3. Call Rejected with 488
If multiple methods are supported by the device, the
unjoined-initial-answer-pref specifies the method to use.
During software initialization, this parameter has the following verifications,
and if it detects an error, the value of unjoined-initial-answer-pref will
default to none
• unjoined-initial-answer-pref equals connectionless-sdp
Required—connectionless-SDP-type does not equal none
Otherwise—unjoined-initial-answer-pref equals none
Required—offer-answer-support equals true
• unjoined-initial-answer-pref equals nomedia-sdp
Required—nomedia-sdp-support equals true and offer-answer-support
equals false
Customizing Device Profiles

Call Control Platform and CTI Connector are preprovisioned with a number of
default device profiles, which reflect Genesys’ knowledge of the behavior of
some commonly used SIP devices, including the GVP Media Control
Platform.
Your deployment may require the Call Control Platform or the CTI Connector
interface with a SIP device that is not currently defined in the default device
profile provisioning file. If the SIP device attributes do not match any of the
preprovisioned device profiles, you must create a new device profile, or else
modify an existing one, to match the actual attributes supported by the SIP
device.
• If the SIP request from the unknown device includes the User-Agent
header, or another header that the Call Control Platform or the CTI
Connector can use to identify the device, Genesys recommends that you
create a new device profile.
• If the SIP request does not include headers that can be used for
identification purposes, calls from the unknown device will use one of the
default device profiles (Default Inbound, Default Outbound, Default
Dialog, or Default Conference). In this case, if you wish to support the
unknown device, you must modify parameters in the default device
profile(s).
For example, if an unknown SIP device that does not support
Offer-Answer makes an inbound call, the call will fail unless you change
User’s Guide 489

the offer-answer-support parameter for the Default Inbound device profile

from true to false.
Tip: To verify which device profile was used for a failed call, use the log
files at debug level: Search for SelectProfile, and match the incoming
INVITE to the device profile selection. Then review the parameter values
for that profile to identify the parameters you need to change.
The following procedure describes how to modify the device profile

provisioning file.
Procedure:
Provisioning Device Profiles
Purpose: To modify the ccpccxml_provision.dat or the CTI_Provision.dat

file to enable the Call Control Platform or the CTI Connector to interact with
non-default SIP devices.
Prerequisites
• The Call Control Platform or the CTI Connector has been installed in a
directory for which you have write access permissions.
• You have identified the required attributes for the device profile(s) you
want to create or modify.
Start of procedure
1. Back up the existing provision files, in case you later want to restore the
original settings.
2. Open the <Call Control Platform Installation Directory>\config\
ccpccxml_provision.dat or the <CTI Connector Installation
Directory>\config\
CTIC_Provision.dat file in a text editor.
3. Add or modify device profile entries as required for your deployment.
The format for each device profile entry is the following:
<entry id="<Entry ID>" type="401" name="CCXML Device Profile">
<Precedence>
<Profile Name>
<Device Profile Class Name>
<# of properties>
<Property Name 1> <Property Value 1>
...
<Property Name m> <Property Value m>
<SIP Header Name> <Regex>
</entry>
Where:

Appendix E: Device Profiles Default Device Profiles
• <Entry ID> is an unsigned integer that uniquely identifies the entry.

• <Precedence> is an unsigned integer that indicates the order of priority
in which the Call Control Platform or the CTI Connector will attempt
to assign the device profile. The larger the value, the lower the
precedence (1 is the highest). A value of 0 (zero) indicates default.
Except for 0, precedence values must be unique.
• <Profile Name> is a unique alphanumeric string that identifies the
device profile. Spaces are allowed.
• <Device Profile Class Name> is the class of device profiles to which
this device profile belongs.
• <# of properties> is the number of properties that are defined in the
entry.
• <Property Name x> is a non-empty alphanumeric string that must be
unique within the device profile. Spaces are not allowed.
• <Property Value x> is a non-empty alphanumeric string that specifies
the value of the <Property Name x> property. Spaces are not allowed.
• <SIP Header Name> is an optional parameter that, if defined, specifies
the SIP header from the incoming SIP INVITE that the Call Control
Platform or the CTI Connector will attempt to match, to assign the
device profile for inbound connections. Spaces are not allowed.
• <Regex> is the expression that the Call Control Platform or the CTI
Connector will attempt to match in the specified SIP header from the
incoming SIP INVITE. If <SIP Header Name> is empty, <Regex> is also
empty.
Note: The angle brackets in the first and last lines of each device profile
entry are required characters in the syntax.
4. Save the file.

5. Restart the Call Control Platform or the CTI Connector.
End of procedure
Default Device Profiles

Tables 102 and 103 summarize the default device settings, by profile.
User’s Guide 491

The format of each entry in the profile definition file is:

<entry id="Entry ID" type="401" name="CCXML Device Profile">
Precedence
Profile Name
SIP Device
Number of properties
PropertyA ValueA
PropertyB ValueB
...
User-Agent User-Agent
</entry>
Where:
• The angle brackets are a necessary part of the syntax.
• Italic text indicates placeholders for items that are listed in Tables 102 and
103.
• SIP Device is the Device Profile Class Name.
• User-Agent is the SIP Header Name.
You may optionally add a short description in each property line.
Table 102: Default Device Profile Settings
Item Value for
Cisco Default Default Default Default Audio- Con-

Gateway In- Out- Con- Dialog codes vedia
bound bound ference Gate- Media
way Server
Entry ID 1 2 3 4 5 6 7
Precedence 1 0 0 0 0 2 3
Profile Name Cisco Default Default Default Default Audio- Conv-

Gateway Inbound Outbound Con- Dialog codes edia
ference Gate- Media
way Server
Number of properties 12 10 10 10 10 12 10

Table 102: Default Device Profile Settings (Continued)
Item Value for
Cisco Default Default Default Default Audio- Con-

Gateway In- Out- Con- Dialog codes vedia
bound bound ference Gate- Media
way Server
sendonly-support false true true true true true true

Properties
recvonly-support false true true true true true true
distinct-send-recv- false true true true true false false

support
multiple-recvonly- false true true true true false false

support
restricts-media-source false true true true true false true
connectionless-sdp- hold hold non- non- hold non- hold

type routable routable routable
offer-answer-support false true false true true false false
nomedia-SDP-support false true false false false false true
offer-less-invite- true true true true true true true

support
User-Agent Cisco Inbound Outbound Con- Dialog Audio- Con-

ference codes vedia
Table 103: Default Device Profile Settings
Item
X-Lite Brook- GVP Audio- Eye Kapanga Dialogic

trout MCP codes Beam Media
Snow- MP104 Gateway
shore
Entry ID 8 9 10 11 12 13 15
Precedence 4 5 6 7 8 9 15
Profile Name X-Lite Brooktrout GVP Audio Eye Kapanga Dialogic

Snowshore MCP codes Beam Media
MP104 Gateway
User’s Guide 493

Table 103: Default Device Profile Settings (Continued)
Item
X-Lite Brook- GVP Audio- Eye Kapanga Dialogic

trout MCP codes Beam Media
Snow- MP104 Gateway
shore
Number of properties 12 10 12 10 12 12 12
sendonly-support true false true true true true true

Properties
recvonly-support true false true true true true true
distinct-send-recv- true false true false true false false

support
multiple-recvonly- true false true true true false true

support
restricts-media-source true true true false true true false
connectionless-sdp- non- hold hold hold non- non- non-

type rout- routable routable routable
able
offer-answer-support false true true true false false false
nomedia-SDP-support false false true false false false false
offer-less-invite- true true true false true true true

support
User-Agent X-Lite Brooktrout GVP Audio Eye Kapanga Dialogic

MCP codes Beam Media
MP104 Gateway

Appendix
F VAR API
This appendix describes the Voice Application Reporter (VAR) application
programming interface (API). It contains the following sections:

Overview, page 495

VAR Records, page 495
 VoiceXML <log> Extensions, page 497
Overview
In GVP 8.5, Voice Application Reporter is provided by the GVP Reporting
Server. The Reporting Server provides access to a web service (VAR Reporting
Service) that generates VAR reports. VAR statistics are computed by the
Reporting Server based on the series of events the Media Control Platform
produces while it is executing VoiceXML applications. The platform generates
some of these events when it encounters VAR-specific <log> tag extensions.
VAR Records
The following section describes the information that is contained in the three
main types of VAR records that are stored by the Reporting Server.
VAR Call Detail Records

VAR Call Detail Records (CDRs) are used to record information about
individual MCP sessions. They contain the following VAR specific
information:
• MCP Session ID
• MCP Component ID
• VAR Call Result
User’s Guide 495

Appendix F: VAR API VAR Records
• Call End State


User End—Indicates that the MCP session ended because the caller
hung up first.

Application End—Indicates that the MCP session ended because the
VoiceXML application hung up first.

System Error—Indicates that the MCP session ended because of a
system error.

Unknown—Indicates that the MCP did not log an end state for the
session.

VAR Call Notes logged by the VXML Application.

Access to the corresponding MCP CDR (for additional information).
VAR Call Detail Records are used to generate VAR CDR reports.
VAR Call Summary Records

VAR Call Summary records provide aggregated statistics (specifically call
count and cumulative call length) for MCP sessions with details broken down
by VoiceXML Application ID, Call State, VAR Call Result, and VAR Call
Reason.
The records are used to derive Call Summary statistics which are useful for
showing the number of calls in each call result category. For example, if you
want to know the number of calls that failed for Application ID - 101, or the
different reasons that are associated with these failed calls, these are the
statistics that you retrieve. For more information on viewing Call Summary
records, see “VAR Call Completion Summary” on page 377.
VAR IVR Action Summary Records

GVP 8.5 uses the concept of an IVR Action to refer to a sequence of
VoiceXML executions that has been identified by the application developer.
The Reporting Server processes information about the IVR Actions that are
executed by the IVR Application, and then aggregates the data into summary
records. These records provide information about the IVR Actions that were
processed during a given time period. Theses records are broken down by time
period, application ID, IVR Action ID, IVR Action result, and IVR Action
reason. The total IVR Action count, the total unique call, the total last IVR
Action count (for each result type), the total count (for each result type), and
end state are also stored in these records.
Action Summary records are used to generate IVR Action reports. For more
information on viewing IVR Action Summary records, see “VAR IVR Action
Summary” on page 380.

Appendix F: VAR API VoiceXML <log> Extensions
VoiceXML <log> Extensions

The Media Control Platform and the Reporting Server support extensions to
the VoiceXML <log> tag in order to generate Voice Application Reporter
statistics. These extensions allow application developers to add specific VAR
details to their VoiceXML pages. A given VoiceXML call can have IVR
actions, a call result, notes that are associated with a specific action, and
custom name/value pairs (variables). The extensions are described in the
following sections.
com.genesyslab.var.CallResult
The platform provides an extension to the <log> tag, using
label=com.genesyslab.var.CallResult. This allows application developers to
specify a result for a call using the following format:
<log label="com.genesyslab.var.CallResult">result[|reason]</log>
The following code snippet is an example that uses the
com.genesyslab.var.CallResult element.
<?xml version="1.0" encoding="UTF-8"?>

<vxml version="2.0" xmlns="http://www.w3.org/2001/vxml">
<form id="weather_info">
<block>
Welcome to the weather information service.
</block>
<field name="state">
<prompt>What state?</prompt>
<grammar src="state.grxml" type="application/srgs+xml"/>
<catch event="help">
Please speak the state for which you want the weather.
</catch>
<catch event="exit">
<log label="com.genesyslab.var.CallResult"> FAILED |exit</log>
<exit/>
</catch>
</field>
<field name="city">
<prompt>What city?</prompt>
<grammar src="city.grxml" type="application/srgs+xml"/>
Please speak the city for which you want the weather.
</catch>
<log label="com.genesyslab.var.CallResult"> FAILED |exit</log>
<exit/>
</catch>
</field>
User’s Guide 497

<block>
<log
label="com.genesyslab.var.CallResult">SUCCESS|reported</log>
<submit next="/servlet/weather" namelist="city state"/>
</block>
</form>
</vxml>
The value of the result must either SUCCESS, FAILED or UNKNOWN (default). The
result is not case-sensitive, and preceding or trailing spaces are ignored. If the
VoiceXML application specifies a call result other than those that are
mentioned, or if no call result is specified, the call result is set to UNKNOWN. A
CallResult <log> tag can be used more than once in an application, but only the
one that is processed last will be recorded by the Reporting Server.
The maximum length of the reason is 256 bytes, and any text beyond the limit
will be truncated.
com.genesyslab.var.ActionStart and
com.genesyslab.var.ActionEnd
An IVR Action can be used to define key transaction points within a
VoiceXML application, and associate those transactions with a given action
ID. An Action starts when a <log> tag is executed with the label attribute set to
com.genesyslab.var.ActionStart. The Action ends when a <log> tag is
executed with the label attribute set to com.genesyslab.var.ActionEnd.
The following code snippet shows the syntax to start an IVR Action:
<log
label="com.genesyslab.var.ActionStart">actionID[|parentID=<PID>]</log>
Application developers must be aware of the following:
• The actionID is the ID of the action that is being started. If this action is
nested inside of an active action, the ID of the parent action (PID) must also
be included.
• The actionID and PID can be any valid UTF8 string that does not contain
spaces or pipes, and it is restricted to a maximum of 64 bytes.
• Spaces are ignored.
The following code snippet shows the syntax to end an IVR Action:
<log
label="com.genesyslab.var.ActionEnd">actionID[|result[|reason]]</log>
• The actionID is the ID of the action that is being started.
• The ID can be any valid UTF8 string that does not contain spaces or pipes,
and it is restricted to a maximum of 64 bytes.

• The result must be either SUCCESS, FAILED or UNKNOWN (default). The result is
not case sensitive, and proceeding and trailing spaces are ignored.
• The reason is optional. The maximum length of the reason is 256 bytes,
and any text beyond that limit is truncated.
Note: The Reporting Server will implicitly end actions in certain cases (see
“Implicit End” on page 500.
The following code snippet is an example using the

com.genesyslab.var.ActionStart and the com.genesyslab.var.ActionEnd
elements.
<?xml version="1.0" encoding="UTF-8"?>

<vxml version="2.0" xmlns="http://www.w3.org/2001/vxml">
<form id="weather_info">
<block>
Welcome to the weather information service.
<log label="com.genesyslab.var.ActionStart">action_1</log>
</block>
<field name="state">
<prompt>What state?</prompt>
<grammar src="state.grxml" type="application/srgs+xml"/>
Please speak the state for which you want the weather.
</catch>
<log label="com.genesyslab.var.ActionEnd">action_1|FAILED|user
exited</log>
<log label="com.genesyslab.var.ActionNotes">action_1|ended in
state dialog</log>
<exit/>
</catch>
</field>
<field name="city">
<prompt>What city?</prompt>
<grammar src="city.grxml" type="application/srgs+xml"/>
Please speak the city for which you want the weather.
</catch>
<log label="com.genesyslab.var.ActionEnd">action_1|FAILED|user
exited</log>
<log label="com.genesyslab.var.ActionNotes">action_1|ended in
city dialog</log>
<exit/>
</catch>
</field>
User’s Guide 499

<block>
<log label="com.genesyslab.var.CustomVar">state|<value
expr="state"/></log>
<log label="com.genesyslab.var.CustomVar">city|<value
expr="city"/></log>
<log
label="com.genesyslab.var.ActionEnd">action_1|SUCCESS|Weather
Accessed</log>
<submit next="/servlet/weather" namelist="city state"/>
</block>
</form>
<catch event=".">
<log
label="com.genesyslab.var.ActionEnd">action_1|FAILED|unexpected
event</log>
<exit/>
</catch>
</vxml>
In some cases an ActionEnd and ActionStart labels are ignored by the
Reporting Server if:
• The specified parentID is not the ID of an active Action (an Action that has
started, but that has not yet ended).
• Its actionID is not the ID of an active IVR Action.
• Its result is not one of SUCCESS, FAILED, or UNKNOWN.
Implicit End
An IVR Action starts when a <log> tag with an ActionStart label is executed.
However, if an IVR Action is currently active, starting a new Action will
automatically cause the previously active Action to end (implicit end), unless
the new Action designates the previously active Action as its parent.
Ending an IVR Action will cause the Reporting Server to end all of its child
Actions implicitly. In addition, when a call ends, all active IVR Actions will be
ended implicitly.
Last IVR Action

The last IVR Action represents the last IVR Action that was executed for a
given call. VAR IVR Action summary records allow the Reporting Server to
keep track of the total number of times that a specific IVR Action was
considered the last action that was executed.
If IVR Actions are still active at the end of a call, the most-nested IVR Action
that was still in progress at call end is designated as the last action. For
example, in the following code snippet, action_3 will be designated the last
IVR action because the application exits while the Action is still in progress.
<block>
Testing Last IVR Action

<log label="com.genesyslab.var.ActionStart">action_2|
parentID=action_1</log>
<exit/>
<log
label="com.genesyslab.var.ActionEnd">action_3|SUCCESS|test1</log>
<log
<log
</block>
In the following code snippet, action_1 will be designated the last IVR Action
because the application exits after this Action ends.
<block>
Testing Last IVR Action
<log
<log
<log
<exit/>
</block>
com.genesyslab.var.ActionNotes
The platform provides an extension to the <log> tag that allows application
developers to associate free-form notes with an IVR Action.
The following code snippet shows the syntax for action notes:
<log label="com.genesyslab.var.ActionNotes">actionID|notes</log>
• The actionID is the ID of the action.
• The ID can be any valid UTF8 string that does not contain spaces or pipes,
and it is restricted to a maximum of 64 bytes.
• Notes are limited to 4096 bytes, and cannot be empty. Any content beyond
the limit is truncated.
IVR Action Notes can be logged during or after the specified action is ended.
User’s Guide 501

com.genesyslab.var.CallNotes
developers to associate free-form notes with a call. Call notes are limited to
4096 bytes, and cannot be empty. Any content beyond the limit will be
truncated.
the following code snippet shows the syntax for call notes:
<log label="com.genesyslab.var.CallNotes"> notes</log>
com.genesyslab.var.CustomVar
developers to associate custom name/value pairs with a call.
The following code snippet shows the syntax for custom variables:
<log label="com.genesyslab.var.CustomVar">name|value</log>
• The name is any valid UTF8 string that does not contain spaces or pipes,
and it is restricted to a maximum of 64 bytes. Spaces are ignored.
• The value is any valid UTF8 string, to a maximum of 256 bytes. Spaces are
significant.
• If it is not formatted properly, the custom variable data is logged as a
simple message, and will not impact VAR statistics.
Custom variables can be specified at any point in a VoiceXML application.
You can have a maximum of eight configured custom variables for any given
call by setting the [ems]dc.default.max.custom_vars parameter in the Media
Control Platform Application object in Genesys Administrator. Any custom
variables that are specified beyond the maximum are discarded by the system.

Appendix
G Video Support
This appendix describes the Genesys Voice Platform (GVP) supported video
formats.
 Overview, page 503

Supported Protocols and Specifications, page 503

Video Features, page 504
Overview
GVP includes support for the following video applications:
• Video voicemail
• Video conferencing and conferencing management
• Entertainment applications
Because video support is not defined as part of VoiceXML 2.1, the VoiceXML
tags, <audio> and <record>, are enhanced to allow development of video play
and video record applications—for example, video voicemail.
Supported Protocols and Specifications

GVP supports the following video formats:
• AVI container files with H.263 encoded video and G.711 encoded audio
(8kHz).
• 3GPP container files with H.263 encoded video and AMR/AMR-WB
encoded audio (8kHz).
• 3GPP container files with H.264 encoded video and AMR/AMR-WB
encoded audio (8KHZ/16KHZ).
User’s Guide 503

Appendix G: Video Support Video Features
Table 104 lists the supported MIME types.

Table 104: MIME Types
Format File Extension MIME Type
AVI .avi video/avi;codec or video/x-avi=<audio

codec>;videocodec=<video codec>;
3GP .3gp video/3gpp;codec=<audio codec>;videocodec=<video codec>
RAW .263 video/H263 or video/H263 - 1998 or video/x-h263
.264 video/H264 or video/x-h264
• audio_codec for AVI can be ulaw (g.711 mulaw), alaw (g.711 alaw), g729[b]/g729a[b], AMR-NB,
AMR-WB, adpcm, pcm 16 (singed linear PCM 16 bit, or pcm (unsigned linear PCM 8 bit)
• audio_codecs for 3GP can be amr (AMR-NB or AMR-WB)
• video_codec for AVI can be h263 (h.263) or h263-1998 (h.263+) or VP8
• video_codec for 3GP can be h263 (h.263) or h263-1998 (h.263+) or h264
Notes: H.263 media transport over RTP conforms to Mode A transmission

as defined in RFC2190.
H.263+ media transport over RTP conforms to RFC2429.
H.264 media transport over RTP conforms to RFC3984 bis.
The VP8 transcoder is not required if the file being played contains
VP8 video,and/or the caller has requested VP8 video.
Video Features
GVP supports many features for video recording and playback.
VoiceXML Features
GVP video supports the following VoiceXML features:
• Video file playback (including embedded audio)
• Video file record (including embedded audio)
• Video text overlay
• DTMF recognition
• Speech recognition
• Speech and DTMF barge-in
• Prompt queuing

• Caching
Video Playback
The following snippet of code provides an example of video playback:
<?xml version=“1.0”?>
<vxml version=“2.0” xmlns=“http://www.w3.org/2001/vxml”>
<meta name=“application” content=“Video Playback
Example”/>
<form id=“Welcome”>
<block name=“Hello”>
<audio src=“builtin:prompts/sting.vox”/>
Which trailer would you like to watch, Science Fiction or
Drama?
</block>
<field name=“movie”>
<option> Science Fiction </option>
<option> Drama </option>
<filled>
<if cond=“movie==‘Science Fiction’”>
Here’s the trailer for Science Fiction
<audio src=“harrypotter.avi”/>
<elseif cond=“movie=‘Drama’”/>
Here’s the trailer for Drama
<audio src=“jurassic.avi”/>
</if>
</filled>
</field>
</form>
</vxml>
Video Recording
The following snippet of code provides an example of video recording:
<?xml version=“1.0”?>
<vxml version=“2.0” xmlns=“http://www.w3.org/2001/vxml”>
<meta name=“application” content=“Video Recording
Example”/>
<property name=“caching” value=“safe”/>
<property name=“bargein” value=“false”/>
<property name=“confidencelevel” value=“0.45”/>
<property name=“loglevel” value=“4”/>
<form>
<record name=“video_message” beep=“true” maxtime=“30s”
finalsilence=“5s” dtmfterm=“true”
dest=“RecordedFile/”
User’s Guide 505

type=“video/avi;codec=pcm16;videocodec=h263”>
<prompt> please re cord your message </prompt>
<filled>
Here is your video message <value expr=“video_message”/>
</filled>
</record>
</form>
</vxml>
Video Text Overlay

GVP allows the VoiceXML application to specify the text that will be written
on top of the video output. Each video file can be specified with pieces of text
and other attributes that are used to control how the text is displayed. Each
piece of text and its corresponding attributes are represented as an element in
the videotxt array. The GVP videotextexpr extension attribute can be used to
specify a number of different attributes for the text.
The following snippet of code provides an example of video text overlay:
<block name="setupTxt">
<var name="videotxt" expr="new Array(1)"/>
<assign name="videotxt[0]" expr="new Object()"/>
<assign name="videotxt[0].xoffset" expr="0"/>
<assign name="videotxt[0].yoffset" expr="0"/>
<assign name="videotxt[0].text" expr="'My text on Video.'"/>
<assign name="videotxt[0].fontsize" expr="60"/>
<assign name="videotxt[0].fontwidth" expr="0"/>
<assign name="videotxt[0].fontname" expr="'Courier New'"/>
<assign name="videotxt[0].fontstyle" expr="'Regular'"/>

<assign name="videotxt[0].fontcolor" expr="'ffff00'"/>
<assign name="videotxt[0].bgcolor" expr="'000000'"/>
<audio src="audiofile/superman-12s.avi" gvp:videotextexpr="videotxt"/>
</block>
For more information about the video text overlay feature, see the description
of the videotextexpr attribute of the VoiceXML <audio> element in the
Genesys Voice Platform 8.x VoiceXML Help.
Advanced Features
GVP has many advanced features that work well with video. For more
information, see the “Tutorials” in the Genesys Voice Platform 8.x Genesys
VoiceXML Reference Help file.

VCR Controls
GVP allows the caller to navigate within an audio or video stream using
DTMF keys. Functions include pause, resume, skip forward, skip backward, as
well as other features.
Note: When VCR control is used with video, the video will not be updated
(and appear out-of-sync) until an I-frame is played from the Media
Control Platform.
Advanced Barge-in
GVP allows intelligent prompt playback, and confirmation of what prompts
the caller has heard with the reporting of barge-ins offsets based on time and
marks set in the prompt stream.
Conferencing
Video conferencing can be managed by the CCXML and the Media platforms
in the following ways:
• Full, or half-duplex conference connections (including listen only, send
only, or full bidirectional video and audio).
• Video switching.
• Video pre-select.
• Video based on active (loudest) speaker
• Video mixing (tiled conferencing)
Note: Video mixing is enabled by setting the [conference]

video_output_type configuration option value to mixed. The layout
can be controlled by using the [conference] video_mixer_layouts
configuration option.
The </join> attributes are added for specifying video conferencing behavior:
videoalgorithm=”loudest”|”fixed”|”none”(optional)
Where:

loudest is the video from the active participant.

fixed is a pre-selected video channel.
 none disables video conferencing.
Video mixing is enabled by setting the [conference] video_output_type
configuration option value to mixed. The layout that is used can be controlled
by using the [conference] video_mixer_layouts configuration option.
Full Call Recording

GVP supports recording video interactions on the call.
User’s Guide 507

Media Redirect Transfer

Transfer and media redirect works with video allowing GVP to remain in the
call control path while redirecting media to the appropriate endpoints.
SIP and NETANN Access

GVP supports video access through the SIP and NETANN protocols.

Appendix
H Custom Log Sinks

This appendix describes how to develop a custom log sink that can integrate
with the Genesys Voice Platform (GVP) logging infrastructure. It contains the
following sections:

Overview, page 509

Log Sink Interface, page 509
 Building and Linking the Library, page 514
Overview
The GVP logging infrastructure enables you to develop custom log sinks to
filter and process GVP logs according to your specific requirements. A custom
log sink can receive all log types and call metrics.
On Windows, a custom sink must be a DLL (Dynamically Linked Library); on
Linux, it must be a shared object. The instructions provided in this appendix
focus on writing a log sink for C++ using Microsoft Visual Studio 2005 (C++
2005) for Windows.
Log Sink Interface

A custom log sink must support the proper sink interface so that the logging
infrastructure can load it, and execute the methods correctly. The main DLL
interface (or export) consists of the GetSink() function only.
The following code snippet shows one way to define the function for Visual
Studio 2005:
extern "C"
{
__declspec(dllexport) gvp::IEMSLogSink* GetSink();
}
User’s Guide 509

Appendix H: Custom Log Sinks Log Sink Interface
The GetSink() function returns a pointer to a gvp::IEMSLogSink object. The

following code snippet shows the implementation of the GetSink() function:
gvp::IEMSLogSink* GetSink()
{
return new myCustomLogSink();
}
Where myCustomLogSink() is a constructor for a call that is inherited from the
gvp::IEMSLogSink class.
The following code snippet shows an overview of the abstract
gvp::IEMSLogSink class:
class IEMSLogSink
{
public:
IEMSLogSink() {m_bInitialized = false;}
virtual ~IEMSLogSink() {};
/**
* @return The version of the Log Sink.
*/
virtual const char * GetVersion() const = 0;
/**
* Initializes the Log Sink.
* @param[in] pszName The name assigned to this sink.
* @param[in] llNetworkID The network ID of the current process.
* @param[in] pConfigService - For internal GVP use.
* @param[in] pIEMSLog:: - For internal GVP use.
* @return TRUE (success), FALSE (error).
*/
virtual bool Initialize(const char* pszName, long long
llNetworkID,
void* pConfigService, void* pEMSLogInterface) = 0;
/**
* Uninitialize the Log Sink.
*/
virtual bool Uninitialize() = 0;
/**
* This function can be used to process log messages (or metrics).
* @param[in] uLogLevel The log level (LOG_0 - LOG_5 or METRICS).
* @param[in] uLogID The log ID, including module bits and
specifier bits, or the METRICS ID.
* @param[in] strCallID The caller ID string.
* @param[in] timeValue The current time value.
* @param[in] llOriginalSenderID The network ID of the calling
component.
* @param[in] strData The log message.
* @param[in] uThreadID The thread ID.

* @return TRUE (success), FALSE (error).

*/
virtual bool LogToSink(unsigned int uLogLevel, unsigned int
uLogID,
const char* strCallID, const timeval & timeValue,
long long llOriginalSenderID, const char* strData,
unsigned long uThreadID = 0) = 0;
/**
* For Future use, NOT CURRENTLY USED
*/
virtual bool ExecuteSinkCommand(const char* pszCommand,
list<string> & listCommandParameters, string & strResult) = 0;
/**
*/
virtual bool GetSupportedCommands(list<string> & commandList,
list<string> & descriptionList) = 0;
/**
*/
virtual bool GetSinkHealthData(std::map<std::string, void*> &
healthData, const char* pszAttribute = NULL) = 0;
protected:
bool m_bInitialized;
};
When coding the custom sink, you need to write a new class to inherit from
gvp::IEMSLogSink. The GetSink() function must be written so that it returns an
instance of this new class. The implementation specifics of the
gvp::IEMSLogSink functions in the new class depend on your specific
requirements.
Initialization and Shutdown

When the GVP process starts, it creates the GVP Logging object which will
load all sinks that are specified in the [ems] log_sinks parameter of that
process (MCP, CCP or RM). For example, the MCP would load MYSINK DLL
from the $InstallationRoot$/bin/mySink.dll file (along with the other default
logs sinks) at startup, if the following parameters are configured:
[ems]
log_sinks=MFSINK|DATAC|TRAPSINK|MYSINK
log_dll.MYSINK=$InstallationRoot$/bin/mySink.dll
logconfig.MYSINK = *|*|*
metricsconfig.MYSINK = *
User’s Guide 511

During logging initialization, the mySink.dll’s GetSink() method is called to

create the IEMSLogSink object pointer within the mySink.dll. If successful, the
Initialize() method is then called for the IEMSLogSink object.
In the following initialization example, the pszName parameter is MYSINK, and
the llNetworkID is the unique Genesys Management Framework DBID
associated with the MCP Application.
virtual bool Initialize(const char* pszName, long long llNetworkID,
void* pConfigService, void* pEMSLogInterface);
When the MCP is shut down, the logging system is shut down with the
Uninitialize() method.
The following uninitialized example is called for the IEMSLogSink object:
/**
* Uninitialize the Log Sink.
*/
virtual bool Uninitialize();
Note: You must use the Uninitialize() method to properly shut down your
custom logging sink.
For more information on the [ems] log sink parameters, see Configuring
Reporting, page 59.
LogToSink() Method
The LogToSink() method is called every time a log is sent to the custom sink
through GVP Logging.
Table 105 lists and describes the LogToSink() method parameters.

Table 105: LogToSink() Method Parameters
Parameter Description Valid values and Examples
unsigned int uLogLevel Specifies the type of log. • 0—CRITICAL LOGS

• 1—ERROR LOGS
• 2—WARNING LOGS
• 3—NOTICE LOGS
• 4—INFO LOGS
• 5—DEBUG LOGS
• 6—CALL METRICS
unsigned int uLogID Specifies the unique identifier of the The module represents the code
log. This includes the module module that generated the log
identifier (the 12 most significant message (or 0 if a Call Metric).
bits), and the log specifier (the 20 The specifier bits represent an ID
least significant bits). For metrics associated with the log message
logs, the module ID is always 0. within the given module.
constr char* strCallID Specifies the session ID for the MCP Example string for MCP:
and CCP session. 00020023-100003E9
constr timeval & Specifies the time, in UTC, the log timeval structure
timeValue was created.
long long Specifies the unique Genesys Long long

llOriginalSenderID Management Framework DBID that
is associated with the GVP
application the generates the log
messages.
const char* strData The string containing the entire log • Metrics example string:
message. incall_initiated 13:1
• Regular log example string:
Starting Resource Manager
unsigned lonlog.g Specifies the thread ID of the GVP Unsigned long

uThreadID logging thread that the LogSink()
method uses.
Each time a log is sent to the sink, the above values are passed through the
LogToSink() method. At this point, the custom sink can perform additional
filtering, and process the logs as desired.
User’s Guide 513

Appendix H: Custom Log Sinks Building and Linking the Library
Threading and Performance Issues

GVP logging is performed in a separate thread within each component’s
process. All logging sinks, including custom user defined sinks, carry out their
processing within this single logging thread. This can help minimize the
impact of logging on the performance of each GVP component, especially
when logging sinks block when writing to disk. Nevertheless, be aware that the
length of time a sink takes to process a log directly delays other logging sinks.
Building and Linking the Library

The following section describes how to build and link the library for Linux and
Windows.
Linux
Building and linking the custom shared object on Linux can be done using g++
(GCC). The following example shows what commands and options to use:
g++ -c -g -Wall -D_REENTRANT -D_NO_LARGEFILE64_SOURCE -fPIC

myCustomLogSink.C -o myCustomLogSink.o
g++ -g -Wall -D_REENTRANT -D_NO_LARGEFILE64_SOURCE -fPIC -shared

-Wl, -soname, libMySink.so -o libMySink.so myCustomLogSink.o
-lpthread
You will need to use the –fPIC compiler option to make the code position
independent, and the –shared linker flag to tell the linker that this is a shared
object. After the shared object file is created, it can be used by MCP, CCP and
RM.
Windows
Building and linking on Windows depends on the compiler available. For
Microsoft Visual C++, create a Win32 Application - DLL project. Make sure
that in the property pages of the project, the Configuration Properties >
General >Configuration Type is configured as Dynamic Library (.dll), and
that the Configuration Properties > C/C++ >Code Generation > Runtime
Library is configured as Multi-threaded DLL.

Appendix
I SSG HTTP Interface

Trigger Applications interact with the Supplementary Services Gateway
through HTTP. This Appendix describes the XML schema for HTTP requests
and responses:
• POST is used for outbound creation, query, and cancellation requests.
• GET is used to query the status of a previously submitted request.
• DELETE is used to cancel a previously submitted request.
This Appendix contains the following sections:
 Creating Outbound Requests, page 515

Querying Outbound Request Status, page 523

Canceling Outbound Requests, page 529
 SSG Database Queue Clearing During a Restart, page 534

Single HTTP POST (Create/Query/Cancel), page 534

Asynchronous Result Notification, page 535
 HTTP XML Schema, page 542
Creating Outbound Requests

HTTP POST is the only way in which Trigger Applications create outbound
requests for the Supplementary Services Gateway. The POST body must
conform to the XML request schema that is described in the “Fatal Errors”
section.
HTTP Request
The HTTP POST request URI must contain TenantName as a query string
parameter. The Content-Type of the POST request must be either text/xml or
application/xml. The content can be either a single request, or multiple (bulk)
User’s Guide 515

Appendix I: SSG HTTP Interface Creating Outbound Requests
requests. The Supplementary Services Gateway validates the content with the
defined schema, and inserts create request(s) into the Database for persistence.
The following examples show how to create requests.
Single Create Request Without CPD Parameters

POST /SSG?TenantName=<tenant> HTTP/1.1
Accept: */*
Accept-Encoding: gzip, deflate
Accept-Language: en-us
Content-Length: 990
Content-Type: application/xml
Host: 172.24.129.81:9820
Pragma: no-cache
User-Agent: Mozilla/4.0 (compatible; MSIE 6.0; Windows NT 5.0;
InfoPath.1; .NET CLR 2.0.50727)
Connection: Keep-Alive
<SSGRequest xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
<CreateRequest Token="T7034" MaxAttempts="2" TimeToLive="123s"

IVRProfileName="Application" Telnum="9884719189"
NotificationURL="http://182.123.12.12/DIR/OutURL.xml">
</CreateRequest>
</SSGRequest>
Single CreateRequest with CPD Parameters

Accept: */*
Content-Length: 1123
Host: 172.24.129.81:9820
Pragma: no-cache
<CreateRequest
IVRProfileName="SSGProfile"
NotificationURL="http://172.24.129.86/Vamsi/Web/Outcome.xml"
Telnum="9884719189"
Token="T7034"
MaxAttempts="3"
TimeToLive="12000ms">
<cpd record="false"
postconnecttimeout="6000ms"
rnatimeout="6000ms"

preconnect="true"
detect="all"/>
</CreateRequest>
</SSGRequest>
Bulk CreateRequest with CPD Parameters

Accept: */*
Content-Length: 1788
Host: 172.24.129.81:9820
Pragma: no-cache
<CreateRequest Token="Token7034" MaxAttempts="2
TimeToLive="123s" IVRProfileName="Application"
Telnum="9884719189"
NotificationURL="http://182.123.12.12/DIR/OutURL.xml"
Ani="12345">
<cpd record="false"
rnatimeout="6000ms"
preconnect="true"
detect="all"/>
</CreateRequest>
<CreateRequest Token="Token7035" MaxAttempts="2"TimeToLive="123s"
IVRProfileName="Application" Telnum="9884719189"
NotificationURL="http://182.123.12.12/DIR/OutURL.xml"
Ani="12345"/>
<cpd record="false"
rnatimeout="6000ms"
preconnect="true"
detect="all"/>
</CreateRequest>
</SSGRequest>
Table 106 describes the CreateRequest attributes for HTTP POST.
User’s Guide 517

Table 106: CreateRequest Attributes
Token A unique token generated by the TA for each single or bulk

create request. When the Supplementary Services Gateway
responds to the original CREATE request, the response
contains the token associated with that request. The
Supplementary Services Gateway does not enforce
uniqueness of the token. This attribute is mandatory.
IVRProfileName The name of the IVR profile that will be used for this
outbound call. IVR profiles used in outbound calls are
provisioned in Genesys Administrator and are sent to the
Resource Manager through SIP Server. The Supplementary
Services Gateway does not perform validation on IVR
profiles. This attribute is mandatory.
Telnum The telephone number used to make an outbound call (for

SIP requests, it must be the SIP URI). This attribute is
mandatory.
NotificationURL An encoded URL sent to the Supplementary Services

Gateway from the TA. The Supplementary Services
Gateway uses this URL to send asynchronously notifications
to the TA indicating the success or failure of the outbound
call. This attribute is mandatory.
TimeToLive The length of time (in seconds or milliseconds) that the

request stays alive in persistent storage. If the outbound call
is not initiated within this time period, no further attempts
are made and the Supplementary Services Gateway sends a
Notification URL to the TA indicating that call initiation
failed. This attribute is mandatory.
MaxAttempts The number of times the SSG attempts to place the outbound
call, should it fail. When the maximum number of attempts
is reached, no further attempts are made and the
Supplementary Services Gateway sends a Notification URL
to the TA indicating that call initiation failed. This attribute
is mandatory.
ANI The ANI in the outbound call that is presented to the

external party. This attribute is optional.
HTTP Response
In most cases, the Supplementary Services Gateway responds to Trigger
Applications with a 200 OK message, and the body contains the result (success

or failure) formatted in the XML response schema (see “Result Notification on

Failure” on page 537). The Contents-Type of the response is txt/xml.
Once the Supplementary Services Gateway validates the HTTP POST request,
it generates an internal unique Request ID for each request, and inserts the
requests into the Database. The generated Request IDs and the Tokens are
passed back to the Trigger Application in the response. For any further
communication with the Supplementary Services Gateway (querying status or
cancellation), the Trigger Application must use the Request IDs.
If the Supplementary Services Gateway encounters any failure during the
HTTP POST validation or insertion into DB, it generates a failure response in
the 200 OK. For a bulk POST request, if parsing or validation fails, then the
entire request fails. However, if the parsing and validation succeed, but later
there are operational errors (for example, inserting a specific record failed),
then the response would contain both success and failure indications.
If the Supplementary Services Gateway receives an HTTP POST request with
multiple operations like CREATE, QUERY and CANCEL, and if the
maximum number of database records are reached in the middle of processing,
the Supplementary Services Gateway will execute all the QUERY and
CANCEL operations only. It will insert the records in the database until it
reaches the threshold, and for any remaining records, the Supplementary
Services Gateway sends a failure response with the reason and the reason code.
The following examples show the possible responses to the CreateRequest.
Single CreateRequest Successful Response

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 347
Content-Type: text/xml
Cache-control: no-cache
<? xml version="1.0" ?>

<SSGResponse xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
<ResponseElement ResponseType="SUCCESS" Token="T1001"
RequestID="123435" />
</SSGResponse>
Bulk CreateRequest Successful Response

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 683
<? xml version="1.0" ?>

User’s Guide 519

RequestID="123435"/>
<ResponseElement ResponseType="SUCCESS" Token="T1003
</SSGResponse>
Single/Bulk CreateRequest Failure Response (Entire Request Failed)

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 347
<?xml version="1.0" ?>

<FailureDescription ReasonCode="110" Reason="XML Data Parsing
Failed: <Failure Reason> at LineNumber: <XXX> and
ColumnNumber: <YYYY>”/>
</SSGResponse>
Bulk CreateRequest Mixed Response

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 623

<ResponseElement ResponseType="FAILURE" Token=”T1003”
ReasonCode="120" Reason="DB Insertion failed"/>
<ResponseElement ResponseType="FAILURE" Token=”T1004”
ReasonCode="121" Reason="Maximum DBRecords threshold
reached"/>
</SSGResponse>
Table 107 describes the various HTTP status codes that the Supplementary
Services Gateway generates:

Table 107: SSG Response Attributes
ResponseType • SUCCESS—Sent if the Create request was added to the

database successfully. The token and RequestID are
passed back to the TA in the response.
• FAILURE—Sent if the Create request parsing fails,
mandatory attributes are missing, or it is not added to
the database. The token, Reason code, and Reason are
 If parsing or validation fails, the Failure Description
with the Reason Code and Reason are passed back.
The entire POST request fails.

If parsing or validation succeeds, but other failures
occur (for example, a specific request is not added to
the database), the tokens are passed back to the TA.
Token Received from the TA that is sending the request and is

associated with the Create request.
RequestID A unique internal ID that is generated by the

Supplementary Services Gateway for each Create request.
This attribute is passed back to the TA when requests are
successfully added to the database.
Reason Code, • TenantName missing in RequestURI

Reason 
Reason Code: 101

Reason: TenantName parameter is missing
• TenantName empty in RequestURI


Reason Code: 102

Reason: TenantName value is empty
• Value supplied in TenantName does not match the

Tenant1.TGDN parameter
 Reason Code: 103
 Reason: Invalid TenantName
User’s Guide 521

Table 107: SSG Response Attributes (Continued)
• HTTP POST request parsing failed due to invalid XML

data

Reason Code: 110

Reason: XML Data Parsing Failed:: <Failure
reason> at LineNumber: XXX and ColumnNumber:
YYY
• Database insertion failed


Reason Code: 120

Reason: Failed to insert record into DB
• Maximum number of records in DB reached


Reason Code: 121

Reason: Maximum DBRecords threshohld reached
• Shutdown in progress

Reason Code: 130

Reason: Unable to process the request. Shutdown
in progress
• SIP Server connection down


Reason Code: 150

Reason: SIP Server Connection is not available
• Tenant Resource DN is not in registered state


Reason Code: 151

Reason: Tenant Resource DN for the tenant is
not registered
• HTTP Method is not supported

 Reason Code: 400
 Reason: HTTP Method is not supported
• Internal error

Reason Code: 500

Reason: Unable to process the request due to
Internal Error

Appendix I: SSG HTTP Interface Querying Outbound Request Status
Querying Outbound Request Status

The Trigger Application can make a request to the Supplementary Services
Gateway for the status of a previously submitted outbound request. Both HTTP
GET (single query) and HTTP POST (single/bulk query) are used. For HTTP
POST, the content must conform to the XML request schema that is described
in the “Fatal Errors” section.
HTTP Request
The Tenant Name and RequestID are the only required parameters in the HTTP
GET or HTTP POST query request. The Tenant Name and RequestID must be
passed in the query string of HTTP GET method, or in the XML body of the
HTTP POST. For the HTTP POST, the Content-Type must be either text/xml
or application/xml.
The following examples show the query requests:
HTTP GET Query Request

GET /SSG?RequestID=1234&TenantName=<name>
Accept: */*
Host: 172.24.129.81:9800
Cache-Control: max-stale=0
Single HTTP POST Query Request

POST /SSG?TenantName=<name> HTTP/1.1
Accept: */*
Content-Length: 376
Host: 172.24.129.81:9800
Pragma: no-cache
<QueryRequest RequestID="1231245"/>
</SSGRequest >
User’s Guide 523

Bulk HTTP POST Query Request

POST /SSG?TenantName=<name> HTTP/1.1
Accept: */*
Content-Length: 489
Host: 172.24.129.81:9820
Pragma: no-cache
</SSGRequest >
HTTP Response
In most scenarios, the Supplementary Services Gateway returns HTTP 200 OK
with the content (conforming to the XML response schema as described in the
“Fatal Errors” section) to indicate success or failure. The Content-Type of the
200 OK response is text/xml.
The following examples show the possible responses for querying.
Single Query Successful Response

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 392

<SSGResponse>
RequestID="123435"
TenantName="Environment" IVRProfileName="Application"
Telnum="11011"
NotificationURL="http://182.24.129.82/Dir/Response.asp"
AttemptsMade=4
MaxAttempts=7 TimeToLive=”12000s” TTLRemaining=”3477s”
Status="Waiting to be processed"/>
</SSGResponse>
Bulk Query Successful Response

HTTP/1.1 200 OK

Date: Tue, 03 Jul 2007 13:42:22 GMT

Content-Length: 272

<SSGResponse>
RequestID="123435"
Telnum="11011"
AttemptsMade=4
RequestID="376489"
Telnum="11012"
AttemptsMade=0
</SSGResponse>
Single/Bulk QueryRequest Failure Response (Validation Failure)

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 347

<FailureDescription ReasonCode="110" Reason="XML Data Parsing
Failed:: <Failure Reason> at LineNumber: <XXX> and
</SSGResponse>
Single Query Failure Response :

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 176

<SSGResponse>
<ResponseElement ResponseType="FAILURE" RequestID="1234"
User’s Guide 525

ReasonCode="404"
Reason="RequestID not found in the Database"/>
</SSGResponse>
Bulk Query Mixed Response

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 657

<SSGResponse>
RequestID="123435"
Telnum="11011"
AttemptsMade=4
<ResponseElement ResponseType="FAILURE" RequestID="1234"
ReasonCode="404"
Reason="RequestID not found in the Database"/>
</SSGResponse>

Table 108 describes the Query attributes.

Table 108: Query Attributes
Response Type • SUCCESS—Sent if the Create request was added to the

mandatory attributes are missing, or it is not added to the
database. The token, Reason code, and Reason are

If parsing or validation fails, the Failure Description

Token A unique token that is generated by the TA for each single

or bulk create request. When the Supplementary Services
Gateway responds to the original CREATE request, the
response contains the token that is associated with that
request. The Supplementary Services Gateway does not
enforce uniqueness of the token. This attribute is
mandatory.
Request ID A unique internal ID that is generated by the Supplementary

Services Gateway for each Create request. This attribute is
passed back to the TA when requests are successful added
to the database.
outbound call. IVR profiles that are used in outbound calls
are provisioned in Genesys Administrator and sent to the
Telnum The telephone number that is used to make an outbound call

(for SIP requests, it must be the SIP URI). This attribute is
mandatory.
NotificationURL An encoded URL that is sent to the Supplementary Services

Gateway from the TA. The Supplementary Services
Gateway uses this URL to send asynchronously
notifications to the TA indicating the success or failure of
the outbound call. This attribute is mandatory.
User’s Guide 527

Table 108: Query Attributes (Continued)
MaxAttempts The number of times that the Supplementary Services

Gateway attempts to place the outbound call, should it fail.
When the maximum number of attempts is reached, no
further attempts are made and the Supplementary Services
Gateway sends a Notification URL to the TA that indicate
that call initiation failed. This attribute is optional.
Attempts Made The total number of times that the Supplementary Services
Gateway attempted to place this outbound call.

failed. This attribute is optional.
TTL Remaining The length of time (in seconds or milliseconds) that remain
for the request to stay alive.
Status The state returned of the current request. The valid states
are:
• Waiting to be processed
• Outbound call failed because TTL expired/TTL
expired
• Outbound call in progress
• Outbound call has been completed

Appendix I: SSG HTTP Interface Canceling Outbound Requests
Table 108: Query Attributes (Continued)
Reason Code, • RequestID missing in Request URI

Reason 
Reason Code = 104

Reason = RequestID parameter is missing
• RequestID empty in Request URI


Reason Code: 105

Reason: RequestID value is empty
• RequestID value is not valid


Reason Code: 106

Reason: Invalid RequestID value
• HTTP POST request parsing failed—invalid schema


Reason Code: 110

YYY

Reason Code: 130

in progress
• Supplied Request ID does not match any database

records

Reason Code: 404

Reason: Request not found in the database
• Internal error

Reason Code: 500

Reason:Unable to process the request due to
Internal Error
Canceling Outbound Requests

The Trigger Application can request the Supplementary Services Gateway to
cancel a pending outbound request. The cancel operation succeeds if the
outbound request is not already in progress (for example, a MakeCall request
has been made to SIP Server). For an in progress outbound request, the cancel
operation fails. You can either use the HTTP DELETE (single cancel), or
HTTP POST (single/bulk cancel) to cancel the request. For HTTP POST, the
User’s Guide 529

content must conform to the XML request schema in the “Fatal Errors”
section.
HTTP Request
The TenantName and RequestID are the only parameters required in the HTTP
DELETE or the HTTP POST cancel request. The TenantName and RequestID
must be passed in the query string of the HTTP DELETE or in the XML body
of the HTTP POST. For HTTP POST, the Content-Type must be either
text/xml or application/xml.
The following examples show how to cancel outbound requests.
HTTP DELETE for Cancel

DELETE /SSG?RequestID=1234&TenantName=<name>
Accept: */*
Host: 172.24.129.81:9820
HTTP POST for a Single Cancel

POST /SSG&TenantName=<name>
Accept: */*
Host: 172.24.129.81:9820
Cache-Control: no-cache
Content-Length: 273
<CancelRequest RequestID="1231245"/>
</SSGRequest >
HTTP POST for a Bulk Cancel

POST /SSG&TenantName=<name>
Accept: */*
Host: 172.24.129.81:9820
Cache-Control: no-cache
Content-Length: 381

</SSGRequest >
HTTP Response
For most scenarios, the Supplementary Services Gateway returns the HTTP
200 OK with the content indicating success or failure. When using a bulk cancel
request, some of the cancel requests might succeed while others might fail. The
Content-Type of the 200 OK response is text/xml.
The following examples show how the possible responses to the
CancelRequest.
Single CancelRequest Successful Response

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 120

<ResponseElement ResponseType="SUCCESS" RequestID="1231245"/>
</SSGResponse>
Bulk CancelRequest Mixed Response

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 488

<SSGResponse
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
<ResponseElement ResponseType="SUCCESS" RequestID=”1121245”/>
<ResponseElement ResponseType="FAILURE" RequestID=”1000000”
ReasonCode="106" Reason="RequestID not found"/>
</SSGResponse>
Single and Bulk CancelRequest Failure Response

HTTP/1.1 200 OK
Date: Tue, 03 Jul 2007 13:42:22 GMT
Content-Length: 488
User’s Guide 531


<SSGResponse
xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance">
<FailureDescription ReasonCode="110" Reason=" ”XML Data
Parsing Failed:: <Failure Reason> at LineNumber: <XXX> and
</SSGResponse>
Table 109 describes the various attributes in the SSG cancel response.
Table 109: SSG Cancel Response Attributes
ResponseType • SUCCESS—Sent if the Create request was added to the

mandatory attributes are missing, or it is not added to
the database. The token, Reason code, and Reason are

If parsing or validation fails, the Failure Description

RequestID A unique internal ID that is generated by the

Supplementary Services Gateway for each Create request.
This attribute is passed back to the TA when requests are
successfully added to the database.
Reason Code, • RequestID missing in Request URI

Reason 
Reason Code = 104
 Reason = RequestID parameter is missing
• RequestID empty in Request URI


Reason Code: 105

Reason: RequestID value is empty
• RequestID value is not valid


Reason Code: 106

Reason: Invalid RequestID value

Table 109: SSG Cancel Response Attributes (Continued)
Reason Code, • HTTP POST request parsing failed due to invalid XML
Reason (continued) data

Reason Code: 110

YYY

Reason Code: 130

in progress
• Outbound request identified by RequestID is already in

progress or RequestID not found in the database

Reason Code: 140

Reason: Outbound request already in
progress/RequestId not found in database
• Failed to delete record in DB


Reason Code: 401

Reason: Database Access Error. Unable to
process the request
• Internal error

Reason Code: 500

Reason: Unable to process the request due to
Internal Error
User’s Guide 533

Appendix I: SSG HTTP Interface SSG Database Queue Clearing During a Restart
SSG Database Queue Clearing During a

Restart
The InitiatedCallRetryFlag parameter specifies what the Supplementary
Services Gateway (SSG) does with requests that are present in the database
during a restart.
Table 110: InitiatedCallRetryFlag Settings
Value SSG Action on requests in the database
0 Considers all INITIATED failed and invokes a notification URL

for those requests when they expire.
1 Retries the INITIATED requests. Deletes nothing.
3 Deletes only requests with status NEW (1) and INITITATED (2).
4 Deletes all requests in the database (status can be NEW, INITIATED

or PROCESSED).
Single HTTP POST (Create/Query/Cancel)

The Supplementary Services Gateway supports any combination of create,
query, and cancel in the same HTTP POST request.
The following example shows a single request:
<CreateRequest
Token="T1001"
MaxAttempts="2"
TimeToLive="123"
IVRProfileName="Application"
Telnum="9884719189"
</CreateRequest>
</SSGRequest >
Note: Although TenantName is not required for a query, or cancel request, if

the POST content has a create request, the request URI query string
must contain the TenantName parameter.
The responses that the Supplementary Services Gateway generates contain a

combination of create, query, and cancel responses.

Appendix I: SSG HTTP Interface Asynchronous Result Notification
Asynchronous Result Notification

The Supplementary Services Gateway notifies the Trigger Application of the
final outcome of the outbound request with the Notification URL (which is a
mandatory parameter in the CreateRequest). The Supplementary Services
Gateway uses the HTTP GET on the Notification URL, and it appends certain
query string parameters regarding the status of the outbound request.
Result Notification on Success

The Supplementary Services Gateway supports positive notification to the
Trigger Application when a call between GVP and an external party is
successful. The outbound call is marked as successful when the Supplementary
Services Gateway receives the TreatmentApplied event from SIP Server, and at
that time, the request is marked for deletion.
SSG supports positive notification back to the TA in the case of successful call
establishment between GVP and the external party. For any outbound call,
once SSG receives the TreatmentApplied event from SIP Server, the call is
deemed as successful and SSG invokes the NotificationURL. Note that the
NotificationURL is invoked when the successful request is removed from the
DB during batched deletions.
For example:
http://test.genesyslab.com/trigger/result.asp?Token=T3001&RequestID
=123345&TenantName=Environment&IVRProfileName=Application&Telnum=40
86552367&AttemptsMade=1&MaxAttempts=3&TimeToLive=12000s&TTLRemainin
g=4367s&CallUUID=
8GQE8C05B56SV2HRTTJ8M9RFR8000001&Result=SUCCESS&Status=DestinationB
usy:RingNoAnswer
Table 111 describes the various parameters that Supplementary Services
Gateway adds in the query string of the Notifications during a successful
notification.
User’s Guide 535

Table 111: Successful Notification URL Parameters

mandatory.

passed back to the TA when requests are successfully added
to the database.
TenantName The name of the tenant in the original create request.
are provisioned in Genesys Administrator and are sent to
the Resource Manager through SIP Server. The
Supplementary Services Gateway does not perform
validation on IVR profiles. This attribute is mandatory.

mandatory.

Gateway sends a Notification URL to the TA indicating that
call initiation failed. This attribute is optional.

TTL Remaining The length of time remaining (in seconds or milliseconds)


Table 111: Successful Notification URL Parameters (Continued)
CallUUID The unique identifier that is generated by SIP Server for this
call.
Note: If multiple attempts have been made for this
outbound call, the last call’s CallUUID is included in the
Notification URL.
Result SUCCESS
Status The failure reasons (separated by a colon) of prior attempts

if more than one attempt was made for this call. For
example,
• CallStateNoAnswer:voice
• AnsweringMachineDetected:voice
• FaxDetected:AnsweringMachineDetected
Result Notification on Failure

When an outbound request fails (exceeds the maximum number of attempts, or
TTL expires), the Supplementary Services Gateway sends the Notification
URL to the Trigger Application with the results.
The parameters that are passed in the query string of the Notification URL are
the same as the positive notification scenario. The following table describes the
differences in values, where applicable.
Table 112: Failure Notification URL Parameters

mandatory.

passed back to the TA when requests are successful added
to the database.
TenantName The name of the tenant in the original create request.
User’s Guide 537

Table 112: Failure Notification URL Parameters (Continued)
are provisioned in Genesys Administrator and sent to the

mandatory.

Gateway sends a Notification URL to the TA indicating that
call initiation failed. This attribute is optional.

TTL Remaining The length of time remaining (in seconds or milliseconds)

CallUUID The unique identifier generated by SIP Server for this call.
Note: If multiple attempts have been made for this
outbound call, the last call’s CallUUID is included in the
Notification URL.
Result FAILURE
Status The failure reasons (separated by a colon) of prior attempts

for all attempts that are made for this call. For example,
• voice:MaxAttempts Exceeded
• Unknown External Error:Unknown External Error:TTL
Expired
• FATAL ERROR - InvalidNum
• DestinationBusy:FATAL ERROR - FaxDetected

Fatal Errors
When the Supplementary Services Gateway receives any failure reason, it is
considered a FATAL ERROR, and the request is no longer sent for outbound call
processing. For example, the reason FaxDetected would be considered a FATAL
ERROR when the CPD tag includes detect=voice, am, or voice,am, and the call
lands on a Fax machine.
Status = FATAL ERROR–<Reason from any one from the below list>
• InvalidNum
• CallStateSitVacant
• CallStateSitNocircuit
• FaxDetected
Status = Fatal Error–InvalidNum
Table 113 lists and describes the possible call failure reasons.
Table 113: Call Failure Reasons
Reason Description
Voice Positive human voice is detected.
DestinationBusy The dialed number is busy. The call is retried again for
processing.
CallStateSit Detected A SIT error occurs. If the call has Attempts and TTL
remaining, it is retried for processing.
AnsweringMachine Answering Machine is detected.

Detected
InvalidNum An invalid number is dialed. The call is considered a

FATAL ERROR, and it is not held for processing.
CallStateSit A SIT error occurs. The call is considered a FATAL

Vacant ERROR, and it is not held for processing.
CallStateSit A SIT error occurs. If the call has Attempts and TTL
Intercept remaining, it is retried for processing.
Unknown remaining, it is retried for processing.
CallStateSit A SIT error occurs. The call is considered a FATAL

Nocircuit ERROR, and it is not held for processing.
Reorder remaining, it is retried for processing.
User’s Guide 539

Table 113: Call Failure Reasons (Continued)
Reason Description
FaxDetected Fax Machine is detected.
CallStateUnknown An unknown error occurred. The call is considered as

human voice.
CallStateTransferred If these calls have Attempts and TTL remaining, they

CallStateConferenced are retried for processing.
CallStateGeneralError
CallStateRemote
Release
CallStateNoAnswer
CallStateAll
TrunksBusy
CallStateQueueFull
CallStateCleared
CallStateOverflowed
CallStateAbandoned
CallStateRedirected
CallStateForwarded
CallStateConsult
CallStatePickedup
CallStateDropped
CallState
Droppednoanswer
CallStateCovered
CallStateConverseOn
CallStateBridged
CallStateDeafened
CallStateHeld
UnknownExternal The call failure reason is not known. If this call has
Error Attempts and TTL remaining, it is retried for processing.

Table 113: Call Failure Reasons (Continued)
Reason Description
FATALERROR The call failed due to the following reasons:

• InvalidNum
• CallStateSitVacant
• CallStateSitNocircuit
• FaxDetected
The call is not held for further processing.
MaxAttempts The maximum number of attempts for the request is

Exceeded reached.
TTL Expired The TimeToLive for the request has expired.
Request is Cancelled The request is cancelled by the Trigger Application.

by Trigger Application
Root Page Access

The Supplementary Services Gateway provides Service Description details
when it's root URL is accessed through a web browser. The HTTP interface
returns a listing of all available services when the root page is accessed which
enables the SSG’s HTTP interface self-documented.
The service listing is returned in Web Application Description Language
(WADL) format that details services such as Method, Parameters, and help
information.
The root page can be accessed with any browser in the following format:
http://<Name/IP Address of SSG Machine>:<Http Port>/ (for example,
http://172.24.129.86:9800/).
The following shows and example of the HTTP GET for Service Listing:
GET /
Accept: */*
Host: 172.24.129.81:9800
User’s Guide 541

Appendix I: SSG HTTP Interface HTTP XML Schema
HTTP XML Schema

The Supplementary Services Gateway uses an embedded HTTP server to
initiate outbound calls with third-party Trigger Applications. For more
information on HTTP requests and responses, see the “How the Supplementary
Services Gateway Works” chapter of the Genesys Voice Platform 8.1
Deployment Guide.
Genesys uses the standard HTTP POST, GET, and DELETE methods for
client/server communication which conforms a defined XML schema. The
POST, GET, and DELETE methods are used to send requests and responses
between the server and the client.
Request Schema
<xs:schema xmlns:xs="http://www.w3.org/2001/XMLSchema">
<xs:element name="SSGRequest">
<xs:complexType>
<xs:annotation>
<xs:documentation xml:lang="en">
A single POST body can contain single create/query/cancel or multiple,
and any combination of the three.
It must conform to the XML request schema present in schema directory
under root path.
</xs:documentation>
</xs:annotation>
<xs:sequence>
<xs:element name="CreateRequest" minOccurs="0" maxOccurs="unbounded"
type="CreateRequestDef"/>
<xs:element name="QueryRequest" minOccurs="0" maxOccurs="unbounded"
type="QueryRequestDef"/>
<xs:element name="CancelRequest" minOccurs="0" maxOccurs="unbounded"
type="CancelRequestDef"/>
</xs:sequence>
</xs:complexType>
</xs:element>
<xs:complexType name="CreateRequestDef">
<xs:annotation>
"CreateRequest" tag is used to specify the attributes used for
creating new outbound call requests.
</xs:documentation>
</xs:annotation>
<xs:sequence>
<xs:element name="cpd" minOccurs="0" maxOccurs="1">
<xs:annotation>

"cpd" tag defined in CreateRequest is used for supporting Call Progress

Detection in SSG.
All CPD attributes are optional.
</xs:documentation>
</xs:annotation>
<xs:complexType>
<xs:attribute name="record" use="optional">
<xs:annotation>
Specifies if the CPD part of the call should be recorded.
true or 1: CPD part to be recorded
false or 0: Do not record CPD part
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:boolean"/>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="preconnect" use="optional">

<xs:annotation>
This attribute is used to decide when to start the CPD.
true or 1: CPD is started as soon as the first RTP packet is received.
false or 0: CPD is started when call is connected.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:boolean"/>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="rnatimeout" use="optional">

<xs:annotation>
Timeout to be applied for Ring No Answer scenario.
Unit is in sec (e.g. 30s). If no unit is specified, seconds assumed.
The range, enforced by SSG through XML Schema is 1 to 60 seconds.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:string">
<xs:pattern value="60|[1-9]s|[1-5][0-9]s|60s|[1-9]|[1-5][0-9]"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="postconnecttimeout" use="optional">

<xs:annotation>
Timeout to be applied for postconnect CPD.
User’s Guide 543

Unit is in sec or msec (e.g. 20s or 3000ms). If no unit is specified,

seconds assumed.
The range, enforced by SSG through XML Schema is 1 to 60 seconds.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:pattern
value="60|[1-9]s|[1-5][0-9]s|60s|[1-9]|[1-5][0-9]|[1-9][0-9]{3}ms|[1-5][0-9]{4}ms|60
000ms"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="detect" use="optional">

<xs:annotation>
This attribute provides control to the Trigger Application about
what to do with the outbound call when various types of CPD are detected.
none (default): CPD in not requested at all by the customer. As soon as
call is connected, start IVR.
all: Turn on full CPD. As soon as call is connected, start IVR.
voice: Only if voice is detected, connect to IVR. Any other detection,
retry.
am: Only if answering m/c is detected, connect to IVR. Any other
detection, retry.
fax: Only if fax is detected, connect to IVR. Any other detection, retry.
voice/am/fax can be combined with comma separation (e.g. voice,am or
am,fax or voice,am,fax etc.).
Refer the XML Schema for the combinations
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:enumeration value="none" />
<xs:enumeration value="all" />
<xs:enumeration value="voice" />
<xs:enumeration value="am" />
<xs:enumeration value="fax" />
<xs:enumeration value="voice,am" />
<xs:enumeration value="voice,fax" />
<xs:enumeration value="am,fax" />
<xs:enumeration value="voice,am,fax" />
</xs:restriction>
</xs:simpleType>
</xs:attribute>
</xs:complexType>
</xs:element>
<xs:element name="CustomData" minOccurs="0" maxOccurs="1">

<xs:complexType>
<xs:annotation>
"CustomData" tag defined in CreateRequest is to allow the user to pass
additional key/value pairs to the IVR application.
To add each Key/Value pair, a sub-element "KeyValue" should be added with
attributes
"Key" carrying "KeyName" and
"Value" carrying "Value" for the above KeyName.
</xs:documentation>
</xs:annotation>
<xs:sequence>
<xs:element name="KeyValue" minOccurs="1" maxOccurs="unbounded">
<xs:complexType>
<xs:sequence>
</xs:sequence>
<xs:attribute name="Key" use="required">
<xs:annotation>
"Key" carrying "KeyName"
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:NMTOKEN">
<xs:maxLength value="255"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="Value" use="required">

<xs:annotation>
Value to be provided for the KeyName
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
</xs:complexType>
</xs:element>
</xs:sequence>
</xs:complexType>
</xs:element>
</xs:sequence>
<xs:attribute name="IVRProfileName" use="required">

<xs:annotation>
User’s Guide 545

Name of the Application Profile to be used for an outbound call.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:minLength value="1"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="NotificationURL" use="required">

<xs:annotation>
This URL will be used by SSG to asynchronously notify the
Trigger Application with the result of an outbound call
(success or failure).
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:token">
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="Telnum" use="required">

<xs:annotation>
Telephone Number to make an outbound call.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:pattern value="([a-z0-9A-Z.])*(@)?([a-z0-9A-Z.])*"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="Token" use="required">

<xs:annotation>
The Trigger Application is expected to pass a unique Token with each
create request to SSG.
</xs:documentation>
</xs:annotation>
<xs:simpleType>

</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="MaxAttempts" use="required">

<xs:annotation>
Number of times SSG should attempt to place the outbound call.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:nonNegativeInteger">
<xs:minInclusive value="1"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="TimeToLive" use="required">

<xs:annotation>
Duration the outbound call request can live in the persistent storage.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:pattern
value="[6-9][0-9]s|[1-9][0-9]{2}s|[1-9][0-9]{3}s|[1-9][0-9]{4}s|[6-9][0-9]|[1-9][0-9
]{2}|[1-9][0-9]{3}|[1-9][0-9]{4}"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="ANI" use="optional">

<xs:annotation>
ANI that is passed on in the outbound call to the external party.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:pattern value="([a-z0-9A-Z.])*(@)?([a-z0-9A-Z.])*"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
</xs:complexType>
<xs:complexType name="QueryRequestDef">
<xs:annotation>
User’s Guide 547

"QueryRequest" tag is used to specify the attributes used for

fetching the details of an existing outbound call requests from
SSG's persistence storage.
</xs:documentation>
</xs:annotation>
<xs:attribute name="Token" use="optional">

<xs:annotation>
The Trigger Application is expected to pass the Token with each
query request to SSG that was received in create request.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="RequestID" use="required">

<xs:annotation>
The identifier of the outbound call request whose details needs to be
fetched from SSG's persistent storage when passed in QueryRequest.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
</xs:complexType>
<xs:complexType name="CancelRequestDef">
<xs:annotation>
"CancelRequest" tag is used to specify the attributes used for
cancelling an existing outbound call requests from
SSG's persistence storage.
</xs:documentation>
</xs:annotation>
<xs:attribute name="Token" use="optional">

<xs:annotation>
The Trigger Application is expected to pass the Token with each
cancel request to SSG that was received in create request.

</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="RequestID" use="required">

<xs:annotation>
The identifier of the outbound call request whose details needs to be
deleted from persistent storage when passed in CancelRequest.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
</xs:complexType>
</xs:schema>
Response Schema
<xs:schema xmlns:xs="http://www.w3.org/2001/XMLSchema">
<xs:element name="SSGResponse">
<xs:complexType>
<xs:annotation>
SSG responds to the Trigger Application with 200 OK and the body contains the
result
(either success or failure) formatted in XML response schema.
The response body in 200 OK is single/bulk depending on whether the POST request
was single/bulk.
It must conform to the XML response schema present in schema directory
under root path.
</xs:documentation>
</xs:annotation>
<xs:sequence>
<xs:element name="ResponseElement" minOccurs="0" maxOccurs="unbounded"
type="ResponseElementDef"/>
<xs:element name="FailureDescription" minOccurs="0" maxOccurs="1"
type="FailureDescriptionDef"/>
</xs:sequence>
</xs:complexType>
User’s Guide 549

</xs:element>
<xs:complexType name="ResponseElementDef">
<xs:annotation>
"ResponseElement" tag is used to specify the attributes sent as
response to the received create/query/cancel request.
</xs:documentation>
</xs:annotation>
<xs:attribute name="ResponseType" use="required">

<xs:annotation>
"CreateRequest" tag is used to specify the attributes used for
creating new outbound call requests.
SUCCESS : If the create/quey/cancel request operation was successful.
FAILURE : If the create/query/cancel request parsing failed or some

mandatory attributes missing or DB insertion failed.
If parsing or other validation fails, then FailureDescription
is generated with ReasonCode and Reason.
If parsing and other validations succeed, but there are other
failures (e.g. specific record insertion into db failed),
then the response would contain Token(s) in the ResponseElement(s).
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:pattern value="SUCCESS|FAILURE"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="Token" use="required">

<xs:annotation>
Token provided in the original create request is passed back.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="RequestID" use="optional">

<xs:annotation>
SSG generates an internal unique request ID for each create request.

For query or cancel request operation, RequestID provided is passed back.

</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="TenantName" use="optional">

<xs:annotation>
Name of the Tenant provided in the original create request is passed back.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:string"/>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="IVRProfileName" use="optional">

<xs:annotation>
Name of the IVRProfile provided in the original create request is passed
back.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="NotificationURL" use="optional">

<xs:annotation>
NotificationURL provided in the original create request is passed back.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="MaxAttempts" use="optional">

<xs:annotation>
Maximum Attempts provided in the original create request is passed back..
</xs:documentation>
</xs:annotation>
<xs:simpleType>
User’s Guide 551

</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="AttemptsMade" use="optional">

<xs:annotation>
The Number of times the requested outbound call has been attempted so far.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="TimeToLive" use="required">

<xs:annotation>
TimeToLive provided in the original create request is passed back.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:pattern
value="[6-9][0-9]s|[1-9][0-9]{2}s|[1-9][0-9]{3}s|[1-9][0-9]{4}s"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="TTLRemaining" use="required">

<xs:annotation>
Time remaining for a request out of received TimeToLive.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:pattern
value="[6-9][0-9]s|[1-9][0-9]{2}s|[1-9][0-9]{3}s|[1-9][0-9]{4}s"/>
</xs:restriction>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="ReasonCode" use="optional">

<xs:annotation>
Error code generated by SSG for any failures.

</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:nonNegativeInteger"/>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="Reason" use="optional">

<xs:annotation>
Description of the error for any failures.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:token"/>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="InternalAttempts" use="optional">

<xs:simpleType>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="Status" use="optional">

<xs:annotation>
Current state of the request.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:simpleType>
</xs:attribute>
</xs:complexType>
<xs:complexType name="FailureDescriptionDef">
<xs:annotation>
SSG returns a FailureDescription with appropriate ReasonCode
and the line number of the parsing error when request parsing
fails due to schema validation or missing mandatory parameters.
</xs:documentation>
</xs:annotation>
If the ,
.
<xs:attribute name="ReasonCode" use="required">

<xs:annotation>
Error code generated by SSG for any schema validation or missing parameter
failures.
User’s Guide 553

</xs:documentation>
</xs:annotation>
<xs:simpleType>
</xs:simpleType>
</xs:attribute>
<xs:attribute name="Reason" use="required">

<xs:annotation>
Description of Error code for any schema validation or missing parameter
failures.
</xs:documentation>
</xs:annotation>
<xs:simpleType>
<xs:restriction base="xs:token"/>
</xs:simpleType>
</xs:attribute>
</xs:complexType>
</xs:schema>

Appendix
J Network Partitioning
This appendix describes the Genesys Voice Platform (GVP) configuration
options for each component that are used to support network traffic
partitioning.

Configuration Options and Protocols, page 555
Configuration Options and Protocols

This section provides a list of configuration options that are used by the GVP
components to specify the network interfaces that are used for certain types of
network traffic.
Table 114 lists the configuration options and protocols that are used by the
Media Control Platform.
Table 114: Media Control Platform
Purpose Protocol Configuration Option
SIP Calls SIP [sip] transport.*
RTP [mpc] rtp.localaddr
HTTP Fetch Requests HTTP [fm] interface
Connection to RTSP Server for RTSP RTSP [vrm] rtp.localadddr

Prompt Playback
RTP [mpc] rtsp.rtp.localaddr
User’s Guide 555

Appendix J: Network Partitioning Configuration Options Configuration Options and Protocols
Table 114: Media Control Platform (Continued)
Connection to MRCPv1 Resources RTSP [vrm] rtp.localadddr
RTP [vrm] rtp.localadddr
Connection to MRCPv2 Resources SIP [mrcpv2client] sip.transport.*
MRCPv2 [vrm] client.mrcpv2.localaddr
RTP [vrm] rtp.localaddr
Composer Proprietary via [vxmli] debug.server.ip

TCP
Table 115 lists the configuration options and protocols that are used by the Call
Control Platform.
Table 115: Call Control Platform
SIP Calls SIP [sip] transport.*
HTTP Fetch Requests HTTP [fm] interface
Basic HTTP Event I/O Processor HTTP [ccxmli] basichttp.recv.host
Session Creation Event I/O Processor HTTP [ccxmli] createsession.recv.host
Resource Manager.
Table 116: Resource Manager
SIP Calls SIP [proxy] sip.transport.*, sip.localport,

sip.localsecureport
SIP Monitoring SIP [monitor] sip.transport.*, sip.localport,

sip.localsecureport
SIP Subscriptions SIP [subscription] sip.transport.*, sip.localport,

sip.localsecureport
SIP Registrations SIP [registrar] sip.transport.*, sip.localport,

sip.localsecureport
Cluster Messaging TCP [cluster] virtual-ip, member.1, member.2

MRCP Proxy.
Table 117: MRCP Proxy
RTSP Contact URI RTSP [vrmproxy] uri
RTSP Port Range RTSP [stack] connection.portrange
Table 118 lists the configuration options and protocols that are used by the CTI
Connector.
Table 118: CTI Connector
SIP Calls SIP [sip] transport.*,

[sip] transport.localaddress,
[sip] transport.localaddress_ipv6,
[sip] transport.localaddress.srv
Connection to Cisco ICM for CTI TCP [Tenant1] Ports

Interaction using GED-125 interface
Connection to IVR Server for CTI TCP [IServer_sample] iserveraddr,

interaction using XML interface iserversocket
Supplementary Services Gateway.
Table 119: Supplementary Services Gateway
HTTP Requests Processing HTTP [HTTP] HTTPPort
HTTPS Requests Processing HTTP [HTTP] HTTPSPort,

[HTTP] CertFile,
[HTTP] CertKeyFile,
The configured parameters (above) are the which is the IP address of the Supplementary Services
Gateway host and are used in the HTTP requests as the port and host name.
User’s Guide 557

PSTN Connector.
Table 120: PSTN Connector
SIP Calls SIP [GatewayManager] LocalIPAddress,

[GatewayManager] UserAgentAddr
RTP [MediaManager] LocalHostAddress
Reporting Server.
Table 121: Reporting Server
SIP Calls SIP [GatewayManager] LocalIPAddress,

[GatewayManager] UserAgentAddr
RTP [MediaManager] LocalHostAddress

Appendix
K SIP Customizable Headers

and Parameters
The GVP 8.1.6 Media Control Platform (MCP) supports propagation of SIP
headers, parameters and request URI parameters to the VXML applications for
incoming SIP messages, and customization of SIP headers, parameters and
request URI parameters for outgoing SIP messages.
The MCP can be configured to pass incoming SIP INVITE requests for the
URI’s parameters, headers, and parameters of any headers to the VXML
application as session variables.
The contents of this chapter includes:
 Abstract Information from Incoming SIP Messages, page 559

Session Variables for VXML, page 560
Abstract Information from Incoming SIP

Messages
The configuration parameters sip.in.invite.headers and
sip.in.invite.params can be defined to abstract information from incoming
SIP messages. They will generate variables to be sent from the Media Control
Platform to the VXML Interpreter in Sip.Invite.<headername> and
Sip.Inivte.<headername>.<paramname> formats, respectively.
Sip.Invite.<headername> contains the header value, as well as all its
parameters. Sip.Invite.<headername>.<paramname> contains the value of a
specific header parameter.
User’s Guide 559

Appendix K: SIP Customizable Headers and Parameters Session Variables for VXML
sip.in.invite.headers
Defines the list of headers to expose to the application. This specifies a list of
header names from the incoming INVITE requests, whose values will be
exposed to the application.
For example, sip.in.invite.headers = From To Via. The exposed values'
names will be in the format sip.invite.<headername>=<value>. If this value is
*, then all headers will be exposed. If this value is none, then no headers will be
exposed. none will be ignored alongside other values.
Default: *
sip.in.invite.params
Defines list of parameters to expose to the application. This specifies a list of
header names from the incoming INVITE requests, whose parameter values
will be exposed to the application.
For example, sip.in.invite.params = From To Via. The exposed values’
names will be in for format sip.invite.<headername>.<paramname>=<value>. If
this value is none, then no parameters will be exposed. none will be ignored
alongside other values.
Default: RequestURI
Session Variables for VXML

The configuration parameter session_vars and the Next Generation Interpreter
configuration parameter vxmli.session_vars can be defined to provide the
session variables to the VXML applications.
session_vars
Each session variable entry is composed of three components. The first
component is the session variable name as exposed within VoiceXML. The
second component is the variable name sent back from the Call Manager. The
third component indicates whether the session variable will be included in the
request for the initial page URL.
Default:

session.connection.answeredby|ANSWEREDBY|0|session.connection.uuipr
otocol|UUIPROTOCOL|0|session.connection.redirect|REDIRECT|0|session
.connection.aai|UUIDATA|0|session.connection.local.uri|LOCALURI|1|s
ession.connection.remote.uri|REMOTEURI|1|session.connection.origina
tor|ORIGIN|0|session.connection.channelidref|PSTNCHANNELID|1|sessio
n.connection.protocol.name|PROTOCOLNAME|0|session.connection.protoc
ol.version|PROTOCOLVERSION|0|session.com.voicegenie.consultdata|con
sultdata|1|session.com.voicegenie.instance.parent|PARENT|1|session.
connection.protocol.isup.natureofconnection.si|NatureOfConnection.S
I|0|session.connection.protocol.isup.natureofconnection.cc|NatureOf
Connection.CCI|0|session.connection.protocol.isup.natureofconnectio
n.ec|NatureOfConnection.EC|0|session.connection.protocol.isup.origi
nalcallednumber.num|OriginalCalledNumber.num|0|session.connection.p
rotocol.isup.originalcallednumber.nai|OriginalCalledNumber.NAI|0|se
ssion.connection.protocol.isup.originalcallednumbe...
vxmli.session_vars
Each session variable entry is composed of three components. The first
component is the session variable name as exposed within VoiceXML. The second
component is the variable name sent back from the Call Manager. The
third component indicates either whether the session variable will be included in
the request for the initial page URL (0 = do not include, 1 = include in GET, 2 =
include in POST, 3 = include in GET and POST), or the type of array of the
session variable (6 = associative array, 7 = ???).
Default:
session.connection.local.uri|LOCALURI|1|session.connection.remote.u
ri|REMOTEURI|1|session.connection.originator|ORIGIN|1|session.conne
ction.protocol.name|PROTOCOLNAME|0|session.connection.protocol.vers
ion|PROTOCOLVERSION|0|session.connection.protocol.sip.headers|Sip.I
nvite|6|session.connection.redirect|REDIRECTHEADER|7|session.connec
tion.callidref|CALLIDREF|1|session.com.voicegenie.instance.parent|P
ARENT|1|session.connection.ocn|OCN|1|session.connection.rdnis|RDNIS
|1|session.connection.rreason|RREASON|1
Here is an example configuration for exposing request URI’s paramA, request
URI’s paramB, From header, and To header’s paramC:

sip.in.invite.headers=From
sip.in.invite.params=RequestURI To
Next Generation Interpreter

Vxmli.session_vars=...|session.connection.protocol.sip.invite.from
|Sip.Invite.From|0|session.connection.protocol.sip.invite.requestur
i.paramA|Sip.Invite.RequestURI.paramA|0|session.connection.protocol
.sip.invite.requesturi.paramB|Sip.Invite.RequestURI.paramB|0|sessio
User’s Guide 561

n.connection.protocol.sip.invite.to.paramC|Sip.Invite.To.paramC|0
With the configuration above and the following SIP INVITE message:
INVITE sip:[email protected];paramA=valueA;paramB=valueB
SIP/2.0
Via: SIP/2.0/UDP
205.150.90.207:5060;branch=z9hG4bK0809fb404f9bcd
From: <sip:[email protected]:5060>;tag=9FB30200-
B96C-01D0-5052-C114EBCA0416
To: <sip:[email protected]>;paramC=valueC
Max-Forwards: 70
CSeq: 1 INVITE
Call-ID: 9FB30200-B96C-C781-2A00-
[email protected]:5060
Contact: sip:[email protected]:5060
Content-Length: 190
Content-Type: application/sdp
v=0
o=Cisco-SIPUA 2455 9673 IN IP4 205.150.90.208
s=SIP Call
c=IN IP4 205.150.90.208
t=0 0
m=audio 30400 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
The following session variables will be defined:
Session Variable Value
session.connection.protocol.sip.invite.from <sip:[email protected]
:5060>;tag=9FB30200-B96C-01D0-
5052-C114EBCA0416
session.connection.protocol.sip.invite.requestur i.paramA valueA
session.connection.protocol.sip.invite.requestur i.paramB valueB
session.connection.protocol.sip.invite.to.paramC valueC
The Media Control Platform can also be configured to set outgoing SIP
INVITE or REFER requests’ URI parameters, headers, and parameters of any
headers (limitations) using signaling variables from the VXML application.
This feature is supported for transfers and calls initiated using RemDial, and can
be enabled by configuring sip.out.invite.headers, sip.out.invite.params,
sip.out.refer.headers and sip.out.refer.params. Please see the Media
Control Platform Deployment Guide regarding the details for these parameters.
Below is an example VoiceXML page that will perform configuration that is
described in the above paragraph, using signalvar:
<?xml version="1.0"?>

<vxml version="2.1" xmlns="http://www.w3.org/2001/vxml"

xmlns:gvp="http://www.anexample.com/1111/vxml21-extension">
——<property name="com.genesyslab.externalevents.enable" value="false"/>
——<property name="com.genesyslab.externalevents.queue" value="true"/>
——<form id="form1">
————<var name="callvars" expr="new Object()"/>
————<block>
——————<assign name="callvars['sip.invite.requesturi.parama']" expr="'valueA'"/>
——————<assign name="callvars['sip.invite.requesturi.paramb']" expr="'valueB'"/>
——————<assign name="callvars['sip.invite.headerc']" expr="'valueC'"/>
————</block>
————<transfer destexpr="'sip:[email protected]:5060'" bridge="true" gvp:signalvar="callvars">
——————<filled>
————————<exit/>
——————</filled>
————</transfer>
——</form>
</vxml>
Below is an example Media Control Platform configuration for customizing
request URI’s paramA, request URI’s paramB and HeaderC in outgoing
INVITE messages (for <transfer> involving two call legs and remdial calls):
sip.out.invite.headers=HeaderC
sip.out.invite.params=RequestURI
If the following signaling variables are defined (or the equivalent name/value
list is defined and appended to the remdial call request):
Sip.Invite.RequestURI.paramA=valueA
Sip.Invite.RequestURI.paramB=valueB
Sip.Invite.HeaderC=valueC
Then, the following SIP INVITE message will be generated for the outgoing
call:
INVITE sip:[email protected];paramA=valueA;paramB=valueB
SIP/2.0
Via: SIP/2.0/UDP
205.150.90.207:5060;branch=z9hG4bK0809fb404f9bcd
From: <sip:[email protected]:5060>;tag=9FB30200-
B96C-01D0-5052-C114EBCA0416
Session Variable Value
session.connection.protocol.sip.invite.from
<sip:[email protected]:5060>;tag
=9FB30200-B96C-01D0-5052-
C114EBCA0416
session.connection.protocol.sip.invite.requestur
i.paramA
valueA
session.connection.protocol.sip.invite.requestur
i.paramB
valueB
session.connection.protocol.sip.invite.to.paramC valueC
User’s Guide 563

User’s Guide 545

Appendix K: SIP Customizable Headers and Parameters
To: <sip:[email protected]>
Max-Forwards: 70
CSeq: 1 INVITE
Call-ID: 9FB30200-B96C-C781-2A00-
[email protected]:5060
Contact: sip:[email protected]:5060
HeaderC: valueC
Content-Length: 190
Content-Type: application/sdp
v=0
o=Cisco-SIPUA 2455 9673 IN IP4 205.150.90.208
s=SIP Call
c=IN IP4 205.150.90.208
t=0 0
m=audio 30400 RTP/AVP 0 101
Chapter 4: Network Interfaces 4.1 SIP
66 VoiceGenie 7.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Supplements
Related Documentation
Resources
The following resources provide additional information that is relevant to this
software. Consult these additional resources as necessary.
Management Framework
• Framework 8.5 Deployment Guide, which provides information about
configuring, installing, starting, and stopping Framework components.
• Framework 8.5 Genesys Administrator Deployment Guide, which provides
information on installing, and configuring Genesys Administrator.
• Framework 8.5 Genesys Administrator Help, which provides information
about configuring and provisioning contact center objects by using the
• Framework 8.5 Configuration Options Reference Manual, which provides
descriptions of the configuration options for Framework components.
SIP Server
• Framework 8.5 SIP Server Deployment Guide, which provides information
about configuring and installing SIP Server.
Genesys Voice Platform

• Genesys Voice Platform 8.5 Deployment Guide, which provides
information about installing and configuring Genesys Voice Platform
(GVP).
• Genesys Voice Platform 8.1 Troubleshooting Guide, which provides
troubleshooting methodology, basic troubleshooting information, and
troubleshooting tools.
User’s Guide 565

• Genesys Voice Platform 8.5 SNMP and MIB Reference, which provides
information about all of the Simple Network Management Protocol
(SNMP) Management Information Bases (MIBs) and traps for GVP,
including descriptions and user actions.
• Genesys Voice Platform 8.x Genesys VoiceXML 2.1 Reference Help, which
provides information about developing Voice Extensible Markup
Language (VoiceXML) applications. It presents VoiceXML concepts, and
provides examples that focus on the GVP Next Generation Interpreter
(NGI) implementation of VoiceXML.
• Genesys Voice Platform 8.1 Legacy Genesys VoiceXML 2.1 Reference
Manual, which describes the VoiceXML 2.1 language as implemented by
the Legacy GVP Interpreter (GVPi) in GVP 7.6 and earlier, and which is
supported in the GVP 8.5 release.
• Genesys Voice Platform 8.x Application Migration Guide, which provides
detailed information about the application modifications that are required
to use legacy GVP 7.6 voice and call-control applications in GVP 8.5.
• Genesys Voice Platform 8.5 CCXML Reference Manual, which provides
information about developing Call Control Extensible Markup Language
(CCXML) applications for GVP.
• Genesys Voice Platform 8.5 Configuration Options Reference, which
replicates the metadata available in the Genesys provisioning GUI, to
provide information about all the GVP configuration options, including
descriptions, syntax, valid values, and default values.
• Genesys Voice Platform 8.5 Metrics Reference, which provides
information about all the GVP metrics (VoiceXML and CCXML
application event logs), including descriptions, format, logging level,
source component, and metric ID.
Genesys Voice Platform 8.x Web Services API wiki, which describes the Web
Services API that the Reporting Server supports.
Voice Platform Solution

• Voice Platform Solution 8.x Integration Guide, which provides information
about integrating GVP, SIP Server, and, if applicable, IVR Server.
Composer Voice
• Composer 8.1 Deployment Guide, which provides information about
installing and configuring Composer Voice.
• Composer 8.1 Help, which provides information about using Composer
Voice, a GUI for developing applications based on VoiceXML and
CCXML.

Open Standards
• W3C Voice Extensible Markup Language (VoiceXML) 2.1, W3C
Recommendation 19 June 2007, which is the World Wide Web Consortium
(W3C) VoiceXML specification that GVP NGI supports.
• W3C Voice Extensible Markup Language (VoiceXML) 2.0, W3C
Recommendation 16 March 2004, which is the W3C VoiceXML
specification that GVP supports.
• W3C Speech Synthesis Markup Language (SSML) Version 1.0,
Recommendation 7 September 2004, which is the W3C SSML
• W3C Voice Browser Call Control: CCXML Version 1.0, W3C Working
Draft 29 June 2005, which is the W3C CCXML specification that GVP
supports.
• W3C Semantic Interpretation for Speech Recognition (SISR) Version 1.0,
W3C Recommendation 5 April 2007, which is the W3C SISR specification
that GVP supports.
• W3C Speech Recognition Grammar Specification (SRGS) Version 1.0,
W3C Recommendation 16 March 2004, which is the W3C SRGS
Genesys
• Genesys Technical Publications Glossary, which ships on the Genesys
Documentation Library DVD and which provides a comprehensive list of
the Genesys and CTI terminology and acronyms used in this document.
• Genesys Migration Guide, which ships on the Genesys Documentation
Library DVD, and which provides documented migration strategies for
Genesys product releases. Contact Genesys Customer Care for more
information.
• Release Notes and Product Advisories for this product, which are available
on the Genesys Customer Care website at http://genesyslab.com/support.
Information about supported operating systems and third-party software is
available on the Genesys Documentation website in the following documents:
Genesys Supported Operating Environment Reference Guide
Genesys Supported Media Interfaces Reference Manual
Genesys product documentation is available on the:
• Genesys Customer Care website at http://genesyslab.com/support.
• Genesys Documentation Library DVD, which you can order by e-mail
from Genesys Order Management at [email protected].
User’s Guide 567

Consult these additional resources as necessary:
• Genesys Hardware Sizing Guide, which provides information about
Genesys hardware sizing guidelines for the Genesys 8.x releases.
• Genesys Interoperability Guide, which provides information on the
compatibility of Genesys products with various Configuration Layer
Environments; Interoperability of Reporting Templates and Solutions; and
Gplus Adapters Interoperability.
• Genesys Licensing Guide, which introduces you to the concepts,
terminology, and procedures relevant to the Genesys licensing system.
• Genesys Database Sizing Estimator 7.6 Worksheets, which provides a
range of expected database sizes for various Genesys products.
For additional system-wide planning tools and information, see the
release-specific listings of System-Level Documents on the Genesys
Documentation website

Document Conventions
This document uses certain stylistic and typographical conventions—
introduced here—that serve as shorthands for particular kinds of information.
Document Version Number

A version number appears at the bottom of the inside front cover of this
document. Version numbers change as new information is added to this
document. Here is a sample version number:
80fr_ref_06-2008_v8.0.001.00
You will need this number when you are talking with Genesys Customer Care
about this product.
Screen Captures Used in This Document

Screen captures from the product graphical user interface (GUI), as used in this
document, may sometimes contain minor spelling, capitalization, or
grammatical errors. The text accompanying and explaining the screen captures
corrects such errors except when such a correction would prevent you from
installing, configuring, or successfully using the product. For example, if the
name of an option contains a usage error, the name would be presented exactly
as it appears in the product GUI; the error would not be corrected in any
accompanying text.
Type Styles
Table 122 describes and illustrates the type conventions that are used in this
document.
Table 122: Type Styles
Type Style Used For Examples
Italic • Document titles Please consult the Genesys Migration

• Emphasis Guide for more information.
• Definitions of (or first references to) Do not use this value for this option.
unfamiliar terms A customary and usual practice is one
• Mathematical variables that is widely accepted and used within a
Also used to indicate placeholder text within particular industry or profession.
code samples or commands, in the special case The formula, x +1 = 7
where angle brackets are a required part of the where x stands for . . .
syntax (see the note about angle brackets on
page 570).
User’s Guide 569

Table 122: Type Styles (Continued)
Type Style Used For Examples
Monospace All programming identifiers and GUI Select the Show variables on screen
font elements. This convention includes: check box.
(Looks like • The names of directories, files, folders, In the Operand text box, enter your
teletype or configuration objects, paths, scripts, dialog formula.
typewriter boxes, options, fields, text and list boxes, Click OK to exit the Properties dialog
text) operational modes, all buttons (including
box.
radio buttons), check boxes, commands,
tabs, CTI events, and error messages. T-Server distributes the error messages in
EventError events.
• The values of options.
If you select true for the
• Logical arguments and command syntax.
inbound-bsns-calls option, all
• Code samples. established inbound calls on a local agent
Also used for any text that users must are considered business calls.
manually enter during a configuration or Enter exit on the command line.
installation procedure, or on a command line.
Square A particular parameter or value that is optional smcp_server -host [/flags]

brackets ([ ]) within a logical argument, a command, or
some programming syntax. That is, the
presence of the parameter or value is not
required to resolve the argument, command, or
block of code. The user decides whether to
include this optional information.
Angle A placeholder for a value that the user must smcp_server -host <confighost>
brackets specify. This might be a DN or a port number
(< >) specific to your enterprise.
Note: In some cases, angle brackets are
required characters in code syntax (for
example, in XML schemas). In these cases,
italic text is used for placeholder values.

Index
Symbols A
[] (square brackets). . . . . . . . . . . . . . 570 access control
< > (angle brackets) . . . . . . . . . . . . . 570 configuring, for reporting . . . . . . . . . 283
<log>tag . . . . . . . . . . . . . . 495, 497, 498 accessing Genesys Administrator . . . . . . . 21
com.genesyslab.var.actionend . . . . . . 498 active call list report . . . . . . . . . . . . . 329
com.genesyslab.var.actionnotes . . . . . 501 address of bridge server . . . . . . . . . . . 218
com.genesyslab.var.actionstart . . . . . . 498 agentx configuration section . . . . . . 277, 278
com.genesyslab.var.callnotes . . . . . . . 502 ALAW . . . . . . . . . . . . . . . . . . . . 163
com.genesyslab.var.callresult . . . . . . . 497 all configuration option . . . . . . . . . . . . . 64
com.genesyslab.var.customvar . . . . . . 502 allow burst usage . . . . . . . . . . . . . . 111
syntax . . . . . . . . . . . . . . . . . . . 450 alternate voicexml url . . . . . . . . . 122, 127
<service>.<param- name> . . . . . . . . . . 121 angle brackets . . . . . . . . . . . . . . . . 570
<service>-capability-requirement . . . . . . 117 announcement allowed . . . . . . . . . . . 111
<service>-forbidden-set-alarm . . . . . . . . 114 announcement level2 usage limit . . . . . . 112
<service>-usage-limit-exceeded-respcode. . 113 announcement level3 usage limit . . . . . . 112
<service>-usage-limit-exceeded-set-alarm . 114 announcement url . . . . . . . . . . . . . . 122
<service>-usage-limit-per-session . . . . . . 113 announcement usage limit . . . . . . . . . . 113
answering machine detection . . . 460, 461, 462
app module name . . . . . . . . . . . . . . 124
Numerics append rejected codecs . . . . . . . . . . . 161
application DBID . . . . . . . . . . . . . . . . 23
100 Trying . . . . . . . . . . . . . . . . . . 466 application slot calculation . . . . . . . . . . 249
180 Ringing . . . . . . . . . . . . . . . . . 466
application templates . . . . . . . . . . . . 147
183 Session Progress . . . . . . . . . . . . 466 applications
202 Accepted. . . . . . . . . . . . . . . . . 467 identifiers . . . . . . . . . . . . . . . . . . 23
302 Moved Temporarily . . . . . . . . . . . 467
ASR
3xx SIP response code. . . . . . . . . . . . 467 configuration options . . . . . . . . . . . 191
400 Bad Request. . . . . . . . . . . . . . . 468 enabling . . . . . . . . . . . . . . . . . . 146
403 Forbidden . . . . . . . . . . . . . . . . 468
provisioning resources . . . . . . . . . . 146
404 Not Found . . . . . . . . . . . . . . . . 469 usage mode . . . . . . . . . . . . . . . . 116
405 Method Not Allowed . . . . . . . . . . . 469 asr configuration section . . . . . . . . 154, 156
408 Request Timeout . . . . . . . . . . . . 469
asr default engine . . . . . . . . . . . . . . 120
420 Bad Extension . . . . . . . . . . . . . . 469
asr load once per call . . . . . . . . . . . . 156
423 Interval Too Brief. . . . . . . . . . . . . 470
asr platform used for recognition . . . . . . 124
480 Temporarily Unavailable . . . . . . . . . 470
asr/tts default languages. . . . . . . . . . . 121
481 Call Does Not Exist . . . . . . . . . . . 471
assigning
487 Request Terminated . . . . . . . . . . . 471
default IVR Profile. . . . . . . . . . . . . 137
488 Not Acceptable Here . . . . . . . . . . 471
MRCP server . . . . . . . . . . . . . . . 149
500 Server Internal Error . . . . . . . . . . . 472
asynchronous result notification . . . . . . . 535
503 Service Unavailable . . . . . . . . . . . 472
ATT . . . . . . . . . . . . . . . . . . . . . 126
User’s Guide 571

audience, defining . . . . . . . . . . . . . . 14 call timeout process . . . . . . . . . . . . . 283
audio call trace hook . . . . . . . . . . . . . . . . 125
file formats, play . . . . . . . . . . . . . . 435 calllog configuration section . . . . . . . . . 154
file formats, record. . . . . . . . . . . . . 439 CallUUID . . . . . . . . . . . . . . . . . . . . 22
audio/video Campaign ID for MSML . . . . . . . . . . . . 24
file formats, play . . . . . . . . . . . . . . 438 capabilities
file formats, record. . . . . . . . . . . . . 441 service . . . . . . . . . . . . . . . . . . 117
Audiocodes Gateway (device profile) . . . . 492 ccpccxml configuration section . . . . 212, 213
Audiocodes MP104 (device profile) . . . . . 493 ccpccxml_provision.dat file . . . . . . . . . 484
ccxml allowed . . . . . . . . . . . . . . . . 111
CCXML applications
B and IVR Profiles. . . . . . . . . . . . . . 103
modifying for TLS . . . . . . . . . . . . . . 43
backup tserver listening port . . . . . . . . . 259 triggering . . . . . . . . . . . . . . . . . . 29
bad xml page hook . . . . . . . . . . . . . . 124
CCXML devices . . . . . . . . . . . . . . . 484
barge-in timeout . . . . . . . . . . . . . . . 206 ccxml level2 usage limit . . . . . . . . . . . 112
basic http authentication password . . . . . 102 ccxml level3 usage limit . . . . . . . . . . . 112
basic http authentication user name . . . . . 102
ccxml usage limit. . . . . . . . . . . . . . . 113
basic http receive host for ipv4 network . . . 214 ccxmli configuration section . . . . . . 212, 214
basic http receive host for ipv6 network . . . 214
cdr batch size . . . . . . . . . . . . . . . . 199
basic http receive show error body. . . . . . 214
cdr configuration section . . . . . . . . 277, 278
beep detection . . . . . . . . . . . . . 462, 463
certificate
beep file time limit in join . . . . . . . . . . . 168 creating . . . . . . . . . . . . . . . . . . . 46
beep filename . . . . . . . . . . . . . . . . 168
creating self-signed . . . . . . . . . . . . . 47
brackets default path . . . . . . . . . . . . . . . . . 47
angle. . . . . . . . . . . . . . . . . . . . 570 channels . . . . . . . . . . . . . . . . . . . 262
square . . . . . . . . . . . . . . . . . . . 570
check point interval . . . . . . . . . . . . . 204
Brooktrout Snowshore (device profile) . . . . 493 Cisco Gateway (device profile) . . . . . . . 492
burst . . . . . . . . . . . . . . . . . . . . . 315 clean interval. . . . . . . . . . . . . . . . . 248
bursting. . . . . . . . . . . . . . . . . . . . 315
client authentication requirement . . . . . . . 99
BYE message . . . . . . . . . . . . . . . . 474 client side connection configuration . . . . . . 68
client side connections
C Call Control Platform . . . . . . . . . . . . 69
CTI Connector . . . . . . . . . . . . . . . 73
call completion summary report . . . . . . . 377 Media Control Platform . . . . . . . . . 70, 71
Call Control Platform PSTN Connector . . . . . . . . . . . . . . 73
client side connections . . . . . . . . . . . 69 Reporting Server . . . . . . . . . . . . . . 71
conference configuration options . . . . . . 59 Resource Manager . . . . . . . . . . . . . 71
configuring. . . . . . . . . . . . . . . . . 209 Supplementary Services Gateway . . . . . 72
default SIP transport. . . . . . . . . . . . . 41 close session timeout . . . . . . . . . . . . 206
device profiles . . . . . . . . . . . . . . . 475 cluster configuration section . . . . . . . . . . 84
functions . . . . . . . . . . . . . . . . . . . 18 cluster standby mode . . . . . . . . . . . . . 86
module IDs . . . . . . . . . . . . . . . . 411 codec preference . . . . . . . . . . . . . . 162
provisioning device profiles . . . . . . . . 490 codecs . . . . . . . . . . . . . . . . . . . . 161
specifier IDs . . . . . . . . . . . . . . . . 411 com.genesyslab.var.actionend. . . . . . . . 498
call dashboard . . . . . . . . . . . . . 315, 316 com.genesyslab.var.actionnotes. . . . . . . 501
component utilization . . . . . . . . . . . 316 com.genesyslab.var.actionstart . . . . . . . 498
ivr profile utilization . . . . . . . . . . . . 315 com.genesyslab.var.callnotes . . . . . . . . 502
call info . . . . . . . . . . . . . . . . . . . . 120 com.genesyslab.var.callresult . . . . . . . . 497
call progress detection . . . . . . . . . 456, 457 com.genesyslab.var.customvar . . . . . . . 502
attributes. . . . . . . . . . . . . . . . . . 252 commenting on this document . . . . . . . . . 14
post-connect. . . . . . . . . . . . . . . . 251 common configuration section . . . . . 246, 247
pre-connect . . . . . . . . . . . . . . . . 251 compatible-output-priorityconfiguration option . 64
call progress component utilization . . . . . . . . . 316, 318
detection . . . . . . . . . . 457, 459, 460 Component ID . . . . . . . . . . . . . . . . . 23
call timeout . . . . . . . . . . . . . . . . . . 278 components

Call Control Platform . . . . . . . . . . . . 18 new call confirmation . . . . . . . . . . . 262
configuring in Genesys Administrator . . . . 30 offhook delay . . . . . . . . . . . . . . . 260
CTI Connector . . . . . . . . . . . . . . . . 19 postconnect priority . . . . . . . . . . . . 260
GVP . . . . . . . . . . . . . . . . . . . . . 17 preconnect priority . . . . . . . . . . . . 260
identifiers . . . . . . . . . . . . . . . . . . 23 primary tserver listening port . . . . . . . 259
Media Control Platform . . . . . . . . . . . 18 pstn connector sip port . . . . . . . . . . 257
PSTN Connector . . . . . . . . . . . . . . 19 range of directory numbers . . . . . . . . 261
Reporting Server . . . . . . . . . . . . . . 19 ringback filename . . . . . . . . . . . . . 259
Resource Manager . . . . . . . . . . . . . 17 route description . . . . . . . . . . . . . 260
Supplementary Services Gateway . . . . . 19 route type . . . . . . . . . . . . . . . . . 264
conf max size. . . . . . . . . . . . . . . . . 109 session timer interval . . . . . . . . . . . 257
conference signaling type . . . . . . . . . . . . . . . 263
configuring. . . . . . . . . . . . . . . . . . 59 sip destination ip address . . . . . . . . . 257
events . . . . . . . . . . . . . . . . . . . 412 sip destination port number . . . . . . . . 257
IVR Profile . . . . . . . . . . . . . . . . . . 59 supported local codec type . . . . . . . . 257
services, configuring. . . . . . . . . . . . . 59 t1-rb anidnis delimiter . . . . . . . . . . . 263
conference allowed. . . . . . . . . . . . . . 111 t1-rb anidnis order . . . . . . . . . . . . 263
conference configuration t1-rb protocol file . . . . . . . . . . . . . 263
section . . . . . . . . . . . 59, 154, 157 t1-rb remove anidnis delimeter . . . . . . 263
conference gain for sdp origin name map . . 166 tserver reconnect timeout . . . . . . . . . 260
conference highest inputs . . . . . . . . . . 157 two channel transfer type . . . . . . . . . 264
conference id . . . . . . . . . . . . . . . . . 122 use tsever to make calls . . . . . . . . . 259
conference level2 usage limit . . . . . . . . 112 wait for offhook confirmation . . . . . . . 260
conference level3 usage limit . . . . . . . . 112 configuration options. . . . . . . . 126, 191, 219
conference participant limit . . . . . . . . . . 157 <service>.<param- name> . . . . . . . . 121
conference usage limit . . . . . . . . . . . . 113 <service>-capability-requirement . . . . . 117
conference video output type . . . . . . . . 157 <service>-forbidden-set-alarm . . . . . . 114
configuration . . . . . . . . . . . . . . 461, 463 <service>-usage-limit-exceeded-respcode 113
Configuration Layer . . . . . . . . . . . . . 29 <service>-usage-limit-exceeded-set-alarm 114
configuration option <service>-usage-limit-per-session . . . . 113
backup tserver listening port . . . . . . . 259 address of bridge server . . . . . . . . . 218
channels . . . . . . . . . . . . . . . . . . 262 all . . . . . . . . . . . . . . . . . . . . . . 64
cpa failure timeout . . . . . . . . . . . . . 258 allow burst usage . . . . . . . . . . . . . 111
cpa max inter-ring timeout. . . . . . . . . 258 alternate voicexml url . . . . . . . . . 122, 127
cpa min inter-ring timeout . . . . . . . . . 258 announcement allowed . . . . . . . . . . 111
cpa options . . . . . . . . . . . . . . . . 258 announcement level2 usage limit . . . . . 112
cpa pamd option . . . . . . . . . . . . . . 258 announcement level3 usage limit . . . . . 112
cpa qualification template . . . . . . . . . 258 announcement url . . . . . . . . . . . . . 122
cpa start delay in msec . . . . . . . . . . 258 announcement usage limit . . . . . . . . 113
default dnis value . . . . . . . . . . . . . 259 app module name . . . . . . . . . . . . . 124
disable custom tones before cpa . . . . . 259 append rejected codecs . . . . . . . . . 161
dtmf payload type . . . . . . . . . . . . . 257 application slot calculation . . . . . . . . 249
enable cpd library . . . . . . . . . . . . . 260 asr default engine . . . . . . . . . . . . . 120
enable isdn overlap receive . . . . . . . . 262 asr load once per call . . . . . . . . . . . 156
enable session timer . . . . . . . . . . . 257 asr platform used for recognition . . . . . 124
fax2 tone as answering machine . . . . . 260 asr/tts default languages . . . . . . . . . 121
ip address of backup tserver . . . . . . . 259 bad xml page hook . . . . . . . . . . . . 124
ip address of primary tserver . . . . . . . 259 barge-in timeout. . . . . . . . . . . . . . 206
isdn numbering plan . . . . . . . . . . . . 261 basic http authentication password . . . . 102
isdn numbering type . . . . . . . . . . . . 261 basic http authentication user name . . . 102
max digits to dial. . . . . . . . . . . . . . 261 basic http receive host for ipv4 network . 214
max digits to receive in basic http receive host for ipv6 network . 214
overlap receive mode . . . . . . . . . . 262 basic http receive show error body . . . . 214
media resource board to use for csp . . . 262 beep file time limit in join . . . . . . . . . 168
minimum download size for play . . . . . 259 beep filename . . . . . . . . . . . . . . . 168
network type . . . . . . . . . . . . . . . . 262 call timeout . . . . . . . . . . . . . . . . 278
User’s Guide 573

call timeout process . . . . . . . . . . . . 283 default dnis . . . . . . . . . . . . . . . . 230
call trace hook . . . . . . . . . . . . . . . 125 default gateway . . . . . . . . . . . . . . 173
ccxml allowed . . . . . . . . . . . . . . . 111 default host . . . . . . . . . . . . . . . . 173
ccxml level2 usage limit . . . . . . . . . . 112 default http page . . . . . . . . . . . . . 248
ccxml level3 usage limit . . . . . . . . . . 112 default ipv4 route for tcp . . . . . . . . . 221
ccxml usage limit . . . . . . . . . . . . . 113 default ipv4 route for tls . . . . . . . . . . 221
cdr batch size . . . . . . . . . . . . . . . 199 default ipv4 route for udp . . . . . . . . . 220
check point interval . . . . . . . . . . . . 204 default ipv6 route for tcp . . . . . . . . . 221
clean interval . . . . . . . . . . . . . . . 248 default ipv6 route for tls . . . . . . . . . . 221
client authentication requirement . . . . . . 99 default ipv6 route for udp . . . . . . . . . 221
close session timeout . . . . . . . . . . . 206 default language . . . . . . . . . . . . . 125
cluster standby mode . . . . . . . . . . . . 86 default properties page . . . . . . . . . . 122
cluster virtual ip address. . . . . . . . . . . 86 default resource port capacity. . . . . . . . 87
codec preference . . . . . . . . . . . . . 162 defer out alerting . . . . . . . . . . . . . 173
codec ptime . . . . . . . . . . . . . .162, 163 define-grammar timeout . . . . . . . . . 206
codecs . . . . . . . . . . . . . . . . . . . 161 device profile bridge server . . . . . . . . 218
compatible-output-priority . . . . . . . . . . 64 dial out number . . . . . . . . . . . . . . 125
conf max size . . . . . . . . . . . . . . . 109 dial prefix . . . . . . . . . . . . . . . . . 261
conference allowed . . . . . . . . . . . . 111 dialing rule based rejection
conference gain for sdp origin name map . 166 response code . . . . . . . . . . . . . 111
conference highest inputs . . . . . . . . . 157 disable cti connector’s cdr update . . . . 230
conference id . . . . . . . . . . . . . . . 122 disable hotword recognition . . . . . . . . 192
conference level2 usage limit . . . . . . . 112 dn group name . . . . . . . . . . . . . . 139
conference level3 usage limit . . . . . . . 112 dn group range . . . . . . . . . . . . . . 139
conference participant limit . . . . . . . . 157 dnis correlation id length . . . . . . . . . 174
conference usage limit . . . . . . . . . . 113 dnis correlation id offset. . . . . . . . . . 174
conference video output type . . . . . . . 157 dtmf send type for sdp orign name map . 166
contact header user name. . . . . . . . . 233 dump fetched pages . . . . . . . . . . . 125
context service password . . . . . . . . . 122 ecc variables . . . . . . . . . . . . . . . 231
context service username . . . . . . . . . 122 election timer . . . . . . . . . . . . . . . . 85
control timeout. . . . . . . . . . . . . . . 206 EMS Logging . . . . . . . . . . . . . . . . 63
cpa method used for outbound calls . . . 157 enable 100 continue header . . . . . 159, 216
cpa timeout . . . . . . . . . . . . . . . . 125 enable 6.x compatibility log output priority 205
cpd default beep timeout . . . . . . . . . 168 enable cookie . . . . . . . . . . . . . . . 216
cpd default final silence timeout . . . . . . 169 enable debugging . . . . . . . . . . . . . 124
CPD default Post-connect Timeout . . . . 168 enable external messaging
CPD default Pre-connect Timeout. . . . . 168 within voicexml . . . . . . . . . . . . . 190
create session receive host enable ipv6 for sip server connection . . . 247
for ipv4 network . . . . . . . . . . . . . 214 enable printing extended values . . . . . 204
create session receive host enable real time debugging . . . . . . . . 189
for ipv6 network . . . . . . . . . . . . . 215 enable record utterance. . . . . . . . . . 124
cti allowed . . . . . . . . . . . . . . . . . 111 enable reliable provisional responses 174, 220
cti end call when agent hangs up . . . . . 125 enable sdp answer in
cti reroute timeout . . . . . . . . . . . . . 125 provisional response . . . . . . . . . . 174
cti usage . . . . . . . . . . . . . . . . . . . 92 enable send/receive events . . . . . . . . 174
custom inbound invite parameter . . . . . 176 enable silence filling . . . . . . . . . . . 192
customer iservers list . . . . . . . . .205, 232 equal priority between old and new . . . . 249
daylight saving hours . . . . . . . . . . . . 32 error recovery time speech resource . . . 206
debug . . . . . . . . . . . . . . . . . . . . 64 expire . . . . . . . . . . . . . . . . . . . . 65
debug hook . . . . . . . . . . . . . . . . 125 failover batch script . . . . . . . . . . . . . 85
default application . . . . . . . . . . . . . 137 fetch dnis from ivr server . . . . . . . . . 230
default audio formats . . . . . . . . . . . 163 fetch script id from urs . . . . . . . . . . 232
default blind transfer. . . . . . . . . .171, 172 file extension for cpd recording . . . . . . 169
default bridge transfer . . . . . . . . . . . 172 fips enabled . . . . . . . . . . . . . . . . 188
default ccxml . . . . . . . . . . . . . . . 213 folder for temporary network log output file 205
default consultation transfer . . . . . . . . 172 for conference. . . . . . . . . . . . . . . . 59

for session timers . . . . . . . . . . . . . . 77 log message format . . . . . . . . . . 101, 203
full audio codec . . . . . . . . . . . . . . 219 log segmentation . . . . . . . . . . . 101, 202
get-params timeout . . . . . . . . . . . . 207 max calls/sec to sip server . . . . . . . . 250
get-result timeout . . . . . . . . . . . . . 207 max concurrent cdr queries . . . . . . . . 279
get-server-info timeout . . . . . . . . . . 207 max conference count . . . . . . . . . . . 93
group type . . . . . . . . . . . . . . . . . . 92 max conference size . . . . . . . . . . . . 93
heartbeat interval . . . . . . . . . . . . . . 85 max db connection pool size . . . . . . . 249
hf disconnect type . . . . . . . . . . . . . 175 max ivr ports . . . . . . . . . . . . . . . 231
hf prefix . . . . . . . . . . . . . . . . . . 175 max page count . . . . . . . . . . . . . . 278
hf stop dial . . . . . . . . . . . . . . . . . 175 max page size. . . . . . . . . . . . . . . 279
hook flash transfer type . . . . . . . . . . 175 max query lock timeout . . . . . . . . . . 279
hotkey base path . . . . . . . . . . . . . 192 maxage for local file. . . . . . . . . . 159, 216
hotkey local path . . . . . . . . . . . . . 192 maximum and minimum frequency
http port . . . . . . . . . . . . . . . . . . 248 of segment . . . . . . . . . . . . . . . 163
http port range . . . . . . . . . . . . . . . 215 maximum attempts limit . . . . . . . . . . 249
http protocol . . . . . . . . . . . . . . . . . 99 maximum bytes of total saved
http proxy . . . . . . . . . 158, 215, 247, 267 temp files per session. . . . . . . . . . 189
https cert key file . . . . . . . . . . . . . 247 maximum cache entry count . . . . . 159, 216
https cert password . . . . . . . . . . . . 247 maximum cache entry size . . . . . . 159, 216
https certificate file name . . . . . . . . . 247 maximum cache size . . . . . . . . . 159, 216
https connector type . . . . . . . . . . . . 100 maximum configured units . . . . . . 279, 282
https port . . . . . . . . . . . . . . . . . 248 maximum number of items
https proxy. . . . . . . . . . . . 158, 215, 267 in the dashboard . . . . . . . . . . . . . 32
icm interface to use . . . . . . . . . . . . 232 maximum record file size . . . . . . . . . 165
in.<SIP request>.headers . . . . . . . . . 176 maximum records in persisted local db file for
inbound allowed media . . . . . . . . . . 219 or data . . . . . . . . . . . . . . . . . 200
inbound level2 usage limit . . . . . . . . . 112 maximum records in persisted local db file
inbound level3 usage limit . . . . . . . . . 112 for cdr data . . . . . . . . . . . . . . . 200
inbound usage limit . . . . . . . . . . . . 113 maximum redirections . . . . . . . . 159, 216
info allowed content type . . . . . . . . . 181 maximum size of script file . . . . . . . . 190
info request content type . . . . . . . . . 176 maximum size of vxml document . . . . . 189
initial page url . . . . . . . . . . . . .122, 127 maximum size of xml/json data . . . . . . 190
initial request method . . . . . . . . . . . 189 maximum subdialog depths . . . . . . . . 189
initiated call retry flag . . . . . . . . . . . 249 mcp max subdialog depth. . . . . . . . . 116
interaction . . . . . . . . . . . . . . . . . . 64 mcp send/receive enabled . . . . . . . . 116
ip type of service for sip transport . . . . . . 89 mcp-asr-usage-mode . . . . . . . . . . . 116
ip type of service for transport . . . . . . . 184 members . . . . . . . . . . . . . . . . . . 86
ip type of service rtp/rtcp . . . . . . .165, 224 members 1 . . . . . . . . . . . . . . . . . 86
ivr client name . . . . . . . . . . . . . . . 233 members 2 . . . . . . . . . . . . . . . . . 86
ivr port base index . . . . . . . . . . . . . 231 memory output buffer size . . . . . . . . 204
IVR Profile, for conference . . . . . . . . . 59 memory snapshot file name. . . . . . . . 204
ivr server communication port . . . . . . . 233 message file . . . . . . . . . . . . . . . 203
ivr server host ip address . . . . . . . . . 233 message_format . . . . . . . . . . . . . . 65
ivr timeout . . . . . . . . . . . . . . . . . 125 metricsfilter . . . . . . . . . . . . . . . . 110
keep startup log file . . . . . . . . . . . . 202 mf sink log filter . . . . . . . . . . . . . . 199
limit of disk storage for messages handled by mf sink metrics filter . . . . . . . . . . . . 199
activemq broker . . . . . . . . . . . . . 278 min db connection pool size . . . . . . . 249
list object id . . . . . . . . . . . . . . . . 123 minimum calls for service quality . . . . . 283
load balancing scheme . . . . . . . . . . . 92 minimum dashboard refreshing interval . . 32
local address contains srv domain name . 222 minimum latency measurments for threshold
local listening address for activemq warning . . . . . . . . . . . . . . . . . 283
broker (tls) . . . . . . . . . . . . . . . . 281 monitoring method . . . . . . . . . . . . . 93
local rtp address . . . . . . . . . . . . . . 177 mrcp connection timeout . . . . . . . . . 205
local transport ipv4 address . . . . . . . . 222 mrcp proxy contact address. . . . . . . . 206
local transport ipv6 address . . . . . .186, 222 msml allowed . . . . . . . . . . . . . . . 111
log expiration . . . . . . . . . . . . .101, 202 msml info allowed content types . . . . . 169
User’s Guide 575

my member id . . . . . . . . . . . . . . . . 86 reporting server http timeout . . . . . . . . 32
native dtmf grammar maxage . . . . . . . 188 request acceptance time-out on
native dtmf grammar maxstale . . . . . . 188 resource dn registration failure . . . . . 250
new mrcp connection per session . . . . . 191 request acceptance time-out on
next retry interval . . . . . . . . . . . . . 250 sips connection failure . . . . . . . . . 250
no cache url substring . . . . . . 159, 216, 267 request batch size . . . . . . . . . . . . 248
noresource-response-code . . . . . . 91, 474 resource dn registration failure
open-session timeout . . . . . . . . . . . 207 recovery interval . . . . . . . . . . . . 250
or batch size . . . . . . . . . . . . . . . . 200 resume timeout . . . . . . . . . . . . . . 208
or reporting interval . . . . . . . . . . . . 200 root directory for cpd recording . . . . . . 169
out.<SIP request>.headers . . . . . . . . 178 root directory for play media . . . . . . . 169
outbound call allowed . . . . . . . . . . . 111 root directory for prompt media . . . . . . 169
outbound call with native cpa ignore call root directory for record media . . . . 169, 170
connect events . . . . . . . . . . . . . 158 route set. . . . . . . . . . . . . . . . . . 223
outbound calls with native cpa initial state 158 routeset . . . . . . . . . . . . . . . . . . . 39
outgoing interface . . . . . . . . . . .159, 216 rs db maintenance process . . . . . . . . 283
output for level all . . . . . . . . . . .100, 201 rs.query.limit.<granularity> . . . . . . . . 273
output for level debug . . . . . . . . .101, 202 rtp de-jitter delay . . . . . . . . . . . . . 166
output for level interaction . . . . . . .101, 201 rtp de-jitter timeout . . . . . . . . . . . . 166
output for level standard. . . . . . . .100, 201 rtp send mode. . . . . . . . . . . . . . . 165
output for level trace. . . . . . . . . .101, 202 rtsp port range for mrcpv1client . . . . . . 205
p-asserted-identity header. . . . . . . . . 177 rule-<n> . . . . . . . . . . . . . . . . 119, 120
pause timeout . . . . . . . . . . . . . . . 207 save ccxml files . . . . . . . . . . . . . . 215
pay load factor. . . . . . . . . . . . . . . 250 save script files . . . . . . . . . . . . . . 215
p-called-party-id header . . . . . . . . . . 177 script id key name. . . . . . . . . . . . . 232
persistent db file for cdr data . . . . . . . 199 sdp local host . . . . . . . . . . . . . . . 220
persistent db file for or data . . . . . . . . 199 sdp local host ipv6 . . . . . . . . . . . . 220
prediction factor . . . . . . . . . . . . . . 118 sdp origin name map . . . . . . . . . . . 166
preferred ip version used in basic http secure protocol version . . . . . . . . . . 247
access uri . . . . . . . . . . . . . . . . 215 secure random algorithm . . . . . . . . . . 99
preferred ip version used in create security provider . . . . . . . . . . . . . . 99
session uri . . . . . . . . . . . . . . . . 215 segment. . . . . . . . . . . . . . . . . . . 66
preferred ip version used in SIP . . . . . . 186 send alert . . . . . . . . . . . . . . . . . 181
preferred ip version used in sip . . . . . . 223 send dtmf relay sip info messages . . . . 170
ps service hostname . . . . . . . . . . . 102 send sdp in invite for media redrirect . . . 187
ps service ip address . . . . . . . . . . . 102 send sip progressing . . . . . . . . . . . 214
ps service port . . . . . . . . . . . . . . . 102 service type . . . . . . . . . . . . . . . . 109
ps service protocol . . . . . . . . . . . . 102 session clean interval . . . . . . . . . . . 206
queue low watermark . . . . . . . . . . . 248 session max idle timeout . . . . . . . . . 207
raise alarm for dialing rule set-params timeout . . . . . . . . . . . . 208
based rejection . . . . . . . . . . . . . 112 setting dynamically . . . . . . . . . . . . 443
raise alarm for exceeding show local time . . . . . . . . . . . . . . . 33
burst limit . . . . . . . . . . . . . . . . 111 sip header for dnis . . . . . . . . . . . . . 87
read-only mode . . . . . . . . . . . . . . 282 sip proxy . . . . . . . . . . . . . . . . . 220
reclaim code . . . . . . . . . . . . . . . . 126 sip resource options interval . . . . . . . . 87
recognition-start-timers timeout . . . . . . 207 sip session timer interval . . . . . . . 110, 137
recognize timeout . . . . . . . . . . . . . 207 sip static route list . . . . . . . . . . . . . 223
refer transfer hold . . . . . . . . . . . . . 179 sip unavailable resource options
refer transfer retry refer interval . . . . . . . . . . . . . . . . . . 88
on the outbound leg . . . . . . . . . . . 180 sip.sessiontimer . . . . . . . . . . . . . . . 77
registration. . . . . . . . . . . . . . .180, 181 sip.transport.<x> . . . . . . . . . . . . . . 38
release asr engines on transfer . . . . . . 189 sips connection failure recovery
remdial max calls . . . . . . . . . . . . . 171 interval . . . . . . . . . . . . . . . . . 250
remdial max client sockets . . . . . . . . 171 snmp task timeout. . . . . . . . . . . 68, 205
remdial port . . . . . . . . . . . . . . . . 171 speak timeout . . . . . . . . . . . . . . . 208
remdial telnet mode . . . . . . . . . . . . 171 speech resource uri . . . . . . . . . . . . 191

srm default response timeout . . . . . . . 188 usage limits . . . . . . . . . . . . . . . . 112
srm ping frequency . . . . . . . . . . . . 188 use original gateway in
srm ping timeout. . . . . . . . . . . . . . 188 outbound call . . . . . . . . . . . . . . 179
srtp mode . . . . . . . . . . . . . . . . . 164 use same gateway . . . . . . . . . . . . 118
ssl ca info . . . . . . . . . . . . . . . . . 160 userdata prefix . . . . . . . . . . . . . . 190
ssl ca path . . . . . . . . . . . . . . .160, 217 verbose level . . . . . . . . . . . 67, 100, 201
ssl certificate. . . . . . . . . . . . . .159, 216 verify peer certificate . . . . . . . . . 160, 217
ssl certificate algorithm . . . . . . . . . . . 99 voicexml dialog allowed . . . . . . . . . . 111
ssl certificate type . . . . . . . . . . .160, 217 voicexml level2 usage limit . . . . . . . . 112
ssl cipher list. . . . . . . . . . . . . .161, 218 voicexml level3 usage limit . . . . . . . . 112
ssl key . . . . . . . . . . . . . . . . .160, 217 voicexml url invite . . . . . . . . . . . . . 183
ssl key password . . . . . . . . 100, 160, 217 voicexml usage limit . . . . . . . . . . . 113
ssl key type . . . . . . . . . . . . . .160, 217 warning headers . . . . . . . . . . . . . 183
ssl keystore password . . . . . . . . . . . . 98 configuration section
ssl keystore path . . . . . . . . . . . . . . 98 agentx. . . . . . . . . . . . . . . . . . . 278
ssl keystore type. . . . . . . . . . . . . . . 99 configuration sections
ssl random file seed . . . . . . . . . .160, 217 agentx. . . . . . . . . . . . . . . . . . . 277
ssl verify host . . . . . . . . . . . . .161, 218 asr. . . . . . . . . . . . . . . . . . . 154, 156
ssl version . . . . . . . . . . . . . . .160, 217 calllog . . . . . . . . . . . . . . . . . . . 154
ssl_* . . . . . . . . . . . . . . . . . . . . 267 ccpccxml . . . . . . . . . . . . . . . 212, 213
standard . . . . . . . . . . . . . . . . . . . 64 ccxmli . . . . . . . . . . . . . . . . . 212, 214
stop timeout . . . . . . . . . . . . . . . . 208 cdr. . . . . . . . . . . . . . . . . . . 277, 278
strict grammar mode . . . . . . . . . . . 189 cluster . . . . . . . . . . . . . . . . . . . . 84
summarization buffer time . . . . . . . . . 282 common. . . . . . . . . . . . . . . . 246, 247
supported gateway cpa events . . . . . . 158 conference . . . . . . . . . . . . 59, 154, 157
tcp reconnect interval . . . . . . . . . . . 208 cpa . . . . . . . . . . . . . . . . . . 154, 157
threshold criteria for latency . . . . . . . . 279 ctic . . . . . . . . . . . . . . . . . . 229, 230
time format for log message. . . . . . . . . 67 dbmp . . . . . . . . . . . . . . . . . . . 277
time format for log messages . . . . .102, 203 dialogicmanager . . . . . . . . . . . 256, 258
time generation for log messages . . . . . 203 dialogicmanager_cpd . . . . . . . . . 256, 259
time to live limit . . . . . . . . . . . . . . 249 dialogicmanager_route1 . . . . . . . 256, 260
timezone offset . . . . . . . . . . . . . . . 33 e-mail . . . . . . . . . . . . . . . . . . . 154
tls certificate for reporting client . . . . . . 201 ems . . . . . . . . . . . . . . . . 84, 199, 267
toll free number . . . . . . . . . . . . . . 109 fm . . . . . . . . . . . . . . 158, 215, 246, 247
trace . . . . . . . . . . . . . . . . . . . . . 64 gatewaymanager . . . . . . . . . . . 256, 257
transcoders . . . . . . . . . . . . . . . . 167 gvp . . . . . . . . . . . . . . . . . . . . . 84
transfer allowed . . . . . . . . . . . . 111, 190 gvp.dn-group . . . . . . . . . . . . . . . 138
transfer connect . . . . . . . . . . . . . . 126 gvp.dn-group-assignment . . . . . . . . . 139
transfer connect url . . . . . . . . . . . . 126 gvp.general . . . . . . . . . . . . . . . . 136
transfer copy headers . . . . . . . . . . . 183 gvp.policy . . . . . . . . . . . . . . . 137, 138
transfer methods . . . . . . . . . . . . . 182 gvp.service-parameters . . . . . . . . . . 138
transfer option . . . . . . . . . . . . . . . 126 http . . . . . . . . . . . . . . . . . . 246, 247
transfer type . . . . . . . . . . . . . . . . 126 https . . . . . . . . . . . . . . . . . . 98, 277
transport instance 0 . . . . . . . 184, 185, 186 https_key . . . . . . . . . . . . . . . 100, 277
transport instance 1 . . . . . . . . . . . . 186 icmc . . . . . . . . . . . . . . . . . . 229, 231
transport instance 2 . . . . . . . . . . . . 186 imdb . . . . . . . . . . . . . . . . . . . . 277
trap hook . . . . . . . . . . . . . . . . . 127 iproxy . . . . . . . . . . . . . . . . . . . 267
trunk group id of trunk id. . . . . . . . . . 231 iserver_sample . . . . . . . . . . . . 98, 233
tts default engine . . . . . . . . . . . . . 121 ivrsc . . . . . . . . . . . . . . . . 98, 229, 232
tts engine default . . . . . . . . . . . . . 187 latency . . . . . . . . . . . . . . . . . . 277
tts gender . . . . . . . . . . . . . . . . . 127 log . . . . . . . . . . . . . . . . . . . 100, 201
tts vendor . . . . . . . . . . . . . . . . . 127 mediacontroller . . . 59, 98, 213, 218, 230, 233
universals grammar uri . . . . . . . . . . 189 mediactrller . . . . . . . . . . . . . . . . . 59
unknown headers allowed mediamanager . . . . . . . . . . . . 256, 257
for a sip message . . . . . . . . . . . . 220 messaging . . . . . . . . . . . . . . . . 277
usage limit exceeded response code . . . 114 monitor . . . . . . . . . . . . . . . . . 85, 87
User’s Guide 577

mpc . . . . . . . . . . . . . . . . . .154, 161 context service password . . . . . . . . . . 122
msml. . . . . . . . . . . . . . . . . .154, 168 context service username . . . . . . . . . . 122
mtinternal . . . . . . . . . . . . . . . . . 154 control timeout . . . . . . . . . . . . . . . . 206
mtmpc . . . . . . . . . . . . . . . . . . . 154 Convedia Media Server (device profile) . . . 492
netann . . . . . . . . . . . . . . . . .154, 169 conventions
pagecollector . . . . . . . . . . . . . . . 250 in document . . . . . . . . . . . . . . . . 569
persistence . . . . . . . . . . . . . .277, 282 type styles. . . . . . . . . . . . . . . . . 569
proxy. . . . . . . . . . . . . . . . . . . 85, 88 cpa configuration section . . . . . . . 154, 157
registrar . . . . . . . . . . . . . . . . . . . 85 cpa failure timeout . . . . . . . . . . . . . . 258
remdial. . . . . . . . . . . . . . . . .154, 171 cpa max inter-ring timeout . . . . . . . . . . 258
reporting . . . . . . . . 102, 277, 279, 281, 282 cpa method used for outbound calls . . . . . 157
rm . . . . . . . . . . . . . . . . . . . . 85, 87 cpa min inter-ring timeout . . . . . . . . . . 258
schedule . . . . . . . . . . . . . . . .277, 283 cpa options. . . . . . . . . . . . . . . . . . 258
session . . . . . . . . . . . . . . . . . . 213 cpa pamd option . . . . . . . . . . . . . . . 258
sessmgr . . . . . . . . . . . . . . 43, 154, 187 cpa qualification template . . . . . . . . . . 258
sip . . . . . . 43, 155, 171, 213, 220, 230, 233 cpa start delay in msec . . . . . . . . . . . 258
snmp. . . . . . . . . . . . 198, 205, 230, 247 cpa timeout . . . . . . . . . . . . . . . . . 125
ssg. . . . . . . . . . . . . . . . . . .246, 248 cpd default beep timeout . . . . . . . . . . 168
stack . . . . . . . . . . . . . . . 155, 198, 205 cpd default final silence timeout . . . . . . . 169
subscription . . . . . . . . . . . . . . . . . 85 CPD default Post-connect Timeout . . . . . 168
tenant1 . . . . . . . . . . . . . . . .234, 246 CPD default Pre-connect Timeout . . . . . . 168
transaction. . . . . . . . . . . . . . . . . 277 create session receive host
tts . . . . . . . . . . . . . . . . . . .155, 187 for ipv4 network . . . . . . . . . . . 214
vrm . . . . . . . . . . . . . . . . . .155, 188 create session receive host
vrmproxy. . . . . . . . . . . . . . . .198, 205 for ipv6 network . . . . . . . . . . . 215
vxmli . . . . . . . . . . . . . . . . . .155, 189 creating
Configuration Server data . . . . . . . . . . 25 SSL private key and certificate . . . . . . . 46
Configuring . . . . . . . . . . . . . . . . . . 38 SSL private key and self-signed certificate . 47
configuring cti allowed . . . . . . . . . . . . . . . . . . 111
access control for reporting . . . . . . . . 283 CTI Connector
ASR and TTS . . . . . . . . . . . . . . . 146 client side connections . . . . . . . . . . . 73
Call Control Platform . . . . . . . . . . . 209 configuring . . . . . . . . . . . . 97, 196, 226
client side connections . . . . . . . . . . . 68 default SIP transport . . . . . . . . . . . . 42
conference service . . . . . . . . . . . . . 59 functions . . . . . . . . . . . . . . . . . . 19
CTI Connector . . . . . . . . . . . 97, 196, 226 module IDs . . . . . . . . . . . . . . . . 422
database retention policies . . . . . . . . 274 specifier IDs . . . . . . . . . . . . . . . . 422
device profiles . . . . . . . . . . . . . . . 484 cti end call when agent hangs up . . . . . . 125
DNIS mapping . . . . . . . . . . . . . . . 129 cti reroute timeout . . . . . . . . . . . . . . 125
Fetching Module. . . . . . . . . . . . . . 266 cti usage . . . . . . . . . . . . . . . . . . . . 92
Fetching Module for HTTPS. . . . . . . . . 51 ctic configuration section. . . . . . . . 229, 230
GVP . . . . . . . . . . . . . . . . . . . . . 33 custom
GVP for SIP Server integration . . . . . . 571 device profiles. . . . . . . . . . . . . . . 489
in Genesys Administrator . . . . . . . . . . 30 SIP responses . . . . . . . . . . . . . 74, 91
Media Control Platform . . . . . . . . . . 142 custom inbound invite parameter . . . . . . 176
options in the Genesys Administrator . . . . 31 custom log sinks . . . . . . . . . . . . . . . 509
PSTN Connector . . . . . . . . . . . . . 256 custom tones . . . . . . . . . . . . . . . . 457
Reporting Server . . . . . . . . . . . . . 271 customer iservers list . . . . . . . . . 205, 232
resource groups . . . . . . . . . . . 89, 90, 94
Resource Manager . . . . . . . . . . . . . 83
route set . . . . . . . . . . . . . . . . . . . 39 D
safe ports, Squid . . . . . . . . . . . . . 270
Squid . . . . . . . . . . . . . . . . . . . 268 database, reporting
default retention periods . . . . . . . . . 274
SSL ports, Squid . . . . . . . . . . . . . 270
Supplementary Services Gateway . . . . 246 retention policies . . . . . . . . . . . . . 274
connection events . . . . . . . . . . . . . . 412 daylight saving hours . . . . . . . . . . . . . 32
dbmp configuration section . . . . . . . . . 277
contact header user name . . . . . . . . . . 233
debug configuration option . . . . . . . . . . 64

debug hook. . . . . . . . . . . . . . . . . . 125 configuration section . . . . . 256, 259
default dialogicmanager_route1
database retention periods . . . . . . . . 274 configuration section . . . . . 256, 260
device profiles . . . . . . . . . . . . . . . 475 disable cti connector’s cdr update . . . . . . 230
IVR Profile . . . . . . . . . . . . . . . . . 137 disable custom tones before cpa . . . . . . 259
log filters . . . . . . . . . . . . . . . . . . . 60 disable hotword recognition . . . . . . . . . 192
log option values . . . . . . . . . . . . . . 67 dn group name. . . . . . . . . . . . . . . . 139
metrics filters . . . . . . . . . . . . . . . . 60 dn group range . . . . . . . . . . . . . . . 139
SIP transports . . . . . . . . . . . . . . . . 40 dnis correlation id length . . . . . . . . . . . 174
SSL private key and certificate paths . . . . 47 dnis correlation id offset . . . . . . . . . . . 174
default application . . . . . . . . . . . . 32, 137 DNIS, mapping IVR Profiles . . . . . . . . . 128
default audio formats . . . . . . . . . . . . . 163 document
default blind option . . . . . . . . . . . 171, 172 conventions . . . . . . . . . . . . . . . . 569
default bridge transfer . . . . . . . . . . . . 172 errors, commenting on . . . . . . . . . . . 14
default ccxml . . . . . . . . . . . . . . . . . 213 version number . . . . . . . . . . . . . . 569
Default Conference (device profile) . . . . . 492 dtmf payload type . . . . . . . . . . . . . . 257
default consultation transfer . . . . . . . . . 172 dtmf send type for sdp orign name map . . . 166
Default Dialog (device profile) . . . . . . . . 492 dump fetched pages . . . . . . . . . . . . . 125
default dnis . . . . . . . . . . . . . . . . . . 230 dynamic configuration options . . . . . . . . 443
default dnis value. . . . . . . . . . . . . . . 259
default gateway . . . . . . . . . . . . . . . 173
default host . . . . . . . . . . . . . . . . . . 173 E
default http page . . . . . . . . . . . . . . . 248
ecc variables . . . . . . . . . . . . . . . . . 231
Default Inbound (device profile) . . . . . . . 492
default ipv4 route for tcp . . . . . . . . . . . 221 election timer. . . . . . . . . . . . . . . . . .85
default ipv4 route for tls . . . . . . . . . . . 221 email configuration section . . . . . . . . . 154
ems configuration section . . . . . .84, 199, 267
default ipv4 route for udp. . . . . . . . . . . 220
default ipv6 route for tcp . . . . . . . . . . . 221 EMS Logging
default ipv6 route for tls . . . . . . . . . . . 221 configuration options . . . . . . . . . . . . 63
enable 100 continue header . . . . . . 159, 216
default ipv6 route for udp. . . . . . . . . . . 221
default language . . . . . . . . . . . . . . . 125 enable 6.x compatibility log output priority . . 205
Default Outbound (device profile) . . . . . . 492 enable cookie . . . . . . . . . . . . . . . . 216
default properties page. . . . . . . . . . . . 122 enable cpd library . . . . . . . . . . . . . . 260
default resource port capacity . . . . . . . . 87 enable debugging . . . . . . . . . . . . . . 124
defer out alerting . . . . . . . . . . . . . . . 173 enable external messaging
define-grammar timeout . . . . . . . . . . . 206 within voicexml . . . . . . . . . . . 190
deleting resource groups . . . . . . . . . . . 91 enable ipv6 for sip server connection . . . . 247
device profile bridge server . . . . . . . . . 218 enable isdn overlap receive . . . . . . . . . 262
device profiles enable printing extended values . . . . . . . 204
configuration file . . . . . . . . . . . . . . 484 enable real time debugging . . . . . . . . . 189
configuring. . . . . . . . . . . . . . . . . 484 enable record utterance . . . . . . . . . . . 124
customizing . . . . . . . . . . . . . . . . 489 enable reliable provisional responses . 174, 220
default . . . . . . . . . . . . . . . . . . . 475 enable sdp answer in
properties . . . . . . . . . . . . . . . . . 484 provisional response . . . . . . . . 174
provisioning . . . . . . . . . . . . . .484, 490 enable send/receive options . . . . . . . . . 174
enable session timer . . . . . . . . . . . . . 257
dial out number. . . . . . . . . . . . . . . . 125
enable silence filling . . . . . . . . . . . . . 192
dial prefix . . . . . . . . . . . . . . . . . . . 261
enabling
dialed number mapping . . . . . . . . . . . 128
ASR . . . . . . . . . . . . . . . . . . . . 146
dialing rule based rejection
HTTP Basic Authorization for reporting . . 283
response code . . . . . . . . . . . . 111
outbound dialing . . . . . . . . . . . . . 149
dialing-rules . . . . . . . . . . . . . . . . . 119
reporting . . . . . . . . . . . . . . . . . . 34
dialog
SIPS, HTTPS, SRTP . . . . . . . . . . . . 43
events . . . . . . . . . . . . . . . . . . . 412
SRTP . . . . . . . . . . . . . . . . . . . . 44
Dialogic Media Gateway (device profile) . . . 493
TTS . . . . . . . . . . . . . . . . . . . . 146
dialogicmanager configuration section . 256, 258
Environment tenant
dialogicmanager_cpd
IVR Profile settings . . . . . . . . . . . . 136
User’s Guide 579

session timers . . . . . . . . . . . . . . . . 77 G
equal priority between old and new . . . . . 249
error recovery time speech resource. . . . . 206 gateway
events response to failures . . . . . . . . . . . . . 91
conference. . . . . . . . . . . . . . . . . 412 Gateway header for PSTN Connector . . . . . 25
connection . . . . . . . . . . . . . . . . . 412 gatewaymanager configuration section 256, 257
dialog . . . . . . . . . . . . . . . . . . . 412 Genesys Administrator
media controller . . . . . . . . . . . . . . 414 accessing . . . . . . . . . . . . . . . . . . 21
expire configuration option . . . . . . . . . . 65 configuring HTTPS . . . . . . . . . . . . . 44
expiry timers . . . . . . . . . . . . . . . . . 76 configuring objects . . . . . . . . . . . . . 30
exporting Configuration Server data . . . . . 25 configuring options . . . . . . . . . . . . . 31
Eye Beam (device profile) . . . . . . . . . . 493 described . . . . . . . . . . . . . . . . . . 20
more information . . . . . . . . . . . . . . 21
Settings tab . . . . . . . . . . . . . . . 30, 31
F using . . . . . . . . . . . . . . . . . . . . 30
Genesys CallUUID. . . . . . . . . . . . . . . 22
failover batch script. . . . . . . . . . . . . . 85 Genesys Configuration Layer . . . . . . . . . 29
failures get-params timeout . . . . . . . . . . . . . 207
gateway . . . . . . . . . . . . . . . . . . . 91 get-result timeout . . . . . . . . . . . . . . 207
Resource Manager handling . . . . . . . . 91 get-server-info timeout . . . . . . . . . . . . 207
fax2 tone as answering machine . . . . . . . 260 granularity . . . . . . . . . . . . . . . 273, 306
fetch dashboard . . . . . . . . . . . . . . . 321 group type . . . . . . . . . . . . . . . . . . . 92
fetch dnis from ivr server . . . . . . . . . . . 230 groups, resource . . . . . . . . . . . . 89, 90, 94
fetch script id from urs . . . . . . . . . . . . 232 GVP
Fetching Module component identifiers . . . . . . . . . . . . 23
configuring. . . . . . . . . . . . . . . . . 266 Component ID . . . . . . . . . . . . . . . 23
configuring for HTTPS. . . . . . . . . . . . 51 components . . . . . . . . . . . . . . . . . 17
enabling secure communications . . . . . . 43 configuring . . . . . . . . . . . . . . . . . 33
HTTPS. . . . . . . . . . . . . . . . . . . . 43 enabling ASR and TTS . . . . . . . . . . 146
module IDs . . . . . . . . . . . . . . . . 432 enabling reporting. . . . . . . . . . . . . . 34
specifier IDs . . . . . . . . . . . . . . . . 432 MIBs . . . . . . . . . . . . . . . . . . . . 20
SSL configuration . . . . . . . . . . . . . 267 MRCP speech servers . . . . . . . . . . 146
file extension for cpd recording. . . . . . . . 169 provisioning . . . . . . . . . . . . . . . . . 33
file formats Session ID . . . . . . . . . . . . . . . . . 22
audio, play . . . . . . . . . . . . . . . . . 435 SIP response codes . . . . . . . . . . . 465
audio, record . . . . . . . . . . . . . . . 439 video support . . . . . . . . 455, 503, 555, 559
audio/video, play . . . . . . . . . . . . . 438 gvp configuration section . . . . . . . . . . . 84
audio/video, record . . . . . . . . . . . . 441 GVP MCP (device profile) . . . . . . . . . . 493
video, play . . . . . . . . . . . . . . . . . 438 gvp.config parameter . . . . . . . . . . . . 443
video, record. . . . . . . . . . . . . . . . 441 gvp.dn-group configuration section . . . . . 138
files gvp.dn-group-assignment
device profile configuration . . . . . . . . 484 configuration section . . . . . . . . 139
filters . . . . . . . . . . . . . . . . . . . . . 306 gvp.general configuration section . . . . . . 136
default for logs and metrics . . . . . . . . . 60 gvp.policy configuration section . . . . . . . 137
fips enabled . . . . . . . . . . . . . . . . . 188 gvp.policy. configuration section . . . . . . . 138
fm configuration section . . . 158, 215, 246, 247 gvp.rm.tenant-id parameter . . . . . . . . . . 24
folder for temporary network log output file . 205 gvp.service-parameters configuration section138
font styles GVPi
italic . . . . . . . . . . . . . . . . . . . . 569 module IDs . . . . . . . . . . . . . . . . 409
monospace . . . . . . . . . . . . . . . . 570 specifier IDs . . . . . . . . . . . . . . . . 409
format for log message. . . . . . . . . . . . 67 gvpi . . . . . . . . . . . . . 124, 125, 126, 127
full audio codec. . . . . . . . . . . . . . . . 219 gvp-tenant-id parameter . . . . . . . . . . . . 24
full video codec. . . . . . . . . . . . . . . . 219
H
headers

Session-Expires . . . . . . . . . . . . . . . 76 GVP Component ID . . . . . . . . . . . . . 23
heartbeat interval. . . . . . . . . . . . . . . 85 GVP Session ID. . . . . . . . . . . . . . . 22
hf disconnect type . . . . . . . . . . . . . . 175 IVR Profile . . . . . . . . . . . . . . 24, 108
hf prefix. . . . . . . . . . . . . . . . . . . . 175 IVR Profile DBID . . . . . . . . . . . . . . 24
hf stop dial . . . . . . . . . . . . . . . . . . 175 IVR Profile ID . . . . . . . . . . . . . . . . 24
historical module IDs, listed . . . . . . . . . . . . . 387
call browser report. . . . . . . . . . . . . 358 specifier IDs, listed . . . . . . . . . . . . 387
component call summary report . . . . . . 340 If any transcoding was used for this call . . . 366
component peaks report. . . . . . . . . . 347 IIS and SSL . . . . . . . . . . . . . . . . . . 47
ivr profile report . . . . . . . . . . . . . . 344 imdb configuration section . . . . . . . . . . 277
reports . . . . . . . . . . . . . . . . . . . 337 importing Configuration Server data . . . . . . 25
hook flash transfer type . . . . . . . . . . . 175 in.<SIP request>.headers . . . . . . . . . . 176
hookflash transfer . . . . . . . . . . . . . . 80 inactivity timers . . . . . . . . . . . . . . . . 76
hotkey base path . . . . . . . . . . . . . . . 192 inbound allowed media . . . . . . . . . . . 219
hotkey local path . . . . . . . . . . . . . . . 192 inbound level2 usage limit . . . . . . . . . . 112
HTTP inbound level3 usage limit . . . . . . . . . . 112
Basic Authorization . . . . . . . . . . . . 283 inbound usage limit . . . . . . . . . . . . . 113
http cert key file . . . . . . . . . . . . . . . 247 info allowed content type . . . . . . . . . . 181
http configuration section. . . . . . . . 246, 247 info request content type . . . . . . . . . . 176
http interface . . . . . . . . . . . . . . . . . 515 initial page url . . . . . . . . . . . . . 122, 127
asynchronous result notification . . . . . . 535 initial rerquest method . . . . . . . . . . . . 189
cancelling outbound requests . . . . . . . 529 initiated call retry flag . . . . . . . . . . . . 249
creating outbound request status . . . . . 523 intended audience . . . . . . . . . . . . . . . 14
creating outbound requests . . . . . . . . 515 interaction configuration option . . . . . . . . 64
http xml schema . . . . . . . . . . . . . . 542 ip address of backup tserver. . . . . . . . . 259
http port . . . . . . . . . . . . . . . . . . . 248 ip address of primary tserver . . . . . . . . 259
http port range . . . . . . . . . . . . . . . . 215 ip type of service for sip transport . . . . . . . 89
http protocol . . . . . . . . . . . . . . . . . 99 ip type of service for transport . . . . . . . . 184
http proxy. . . . . . . . . . . 158, 215, 247, 267 ip type of service rtp/rtcp. . . . . . . . 165, 224
http xml schema . . . . . . . . . . . . . . . 542 iproxy configuration section . . . . . . . . . 267
HTTPS IPv6
and Genesys Administrator . . . . . . . . . 44 supported . . . . . . . . . . . . . . . . . . 53
and Reporting Server web server . . . . . . 44 isdn numbering plan . . . . . . . . . . . . . 261
configuring Fetching Module . . . . . . . . 51 isdn numbering type . . . . . . . . . . . . . 261
enabling . . . . . . . . . . . . . . . . . . . 43 iserver_sample configuration section . . 98, 233
setting up Fetching Module . . . . . . . . . 43 italics . . . . . . . . . . . . . . . . . . . . . 569
https cert password. . . . . . . . . . . . . . 247 ivr action list report. . . . . . . . . . . . . . 363
https certificate file name . . . . . . . . . . . 247 ivr action summary report . . . . . . . . . . 380
https configuration section . . . . . . . . 98, 277 ivr client name . . . . . . . . . . . . . . . . 233
https connector type . . . . . . . . . . . . . 100 ivr port base index . . . . . . . . . . . . . . 231
https port . . . . . . . . . . . . . . . . . . . 248 IVR Profile
https proxy . . . . . . . . . . . . . 158, 215, 267 assigning default to tenant . . . . . . . . 137
https_key configuration section . . . . 100, 277 configuring . . . . . . . . . . . . . . . . . 30
configuring DNIS mapping . . . . . . . . 129
database retention policies . . . . . . . . 274
I DBID . . . . . . . . . . . . . . . . . . . . 24
defined . . . . . . . . . . . . . . . . 20, 103
icm interface . . . . . . . . . . . . . . . . . 232
dialed number mapping . . . . . . . . . . 128
icmc configuration section . . . . . . . 229, 231 for conference. . . . . . . . . . . . . . . . 59
identifiers
mapping calls to. . . . . . . . . . . . 29, 128
application DBID. . . . . . . . . . . . . . . 23
metrics filter . . . . . . . . . . . . . . . . 110
Campaign ID for MSML . . . . . . . . . . . 24
name . . . . . . . . . . . . . . . . . 24, 108
for GVP applications. . . . . . . . . . . . . 23
tenant settings . . . . . . . . . . . . . . 136
for GVP components . . . . . . . . . . . . 23
IVR Profile ID . . . . . . . . . . . . . . . . . 24
for GVP sessions . . . . . . . . . . . . . . 22
ivr profile utilization . . . . . . . . . . . . . 315
Gateway header for PSTN Connector. . . . 25
ivr server communication port . . . . . . . . 233
Genesys CallUUID . . . . . . . . . . . . . 22
User’s Guide 581

ivr server host ip address . . . . . . . . . . 233 max query lock timeout . . . . . . . . . . . 279
ivr timeout . . . . . . . . . . . . . . . . . . 125 max subdialog depth. . . . . . . . . . . . . 116
ivrsc configuration section . . . . . 98, 229, 232 maxage for local file . . . . . . . . . . 159, 216
maximum and minimum frequency
of segment . . . . . . . . . . . . . 163
K maximum attempts limit . . . . . . . . . . . 249
maximum bytes of total saved temp
Kapanga (device profile) . . . . . . . . . . . 493 files per session . . . . . . . . . . . 189
keep startup log file . . . . . . . . . . . . . 202
maximum cache entry count . . . . . . 159, 216
maximum cache entry size . . . . . . 159, 216
L maximum configured units . . . . . . . 279, 282
maximum number of items
last ivr action used report . . . . . . . . . . 382 in the dashboard . . . . . . . . . . . 32
latency configuration section . . . . . . . . . 277 maximum record file size . . . . . . . . . . 165
least used load balancing . . . . . . . . . . 92 maximum records in persisted local db file for or
limit of disk storage for messages handled by data . . . . . . . . . . . . . . . . . 200
activemq broker . . . . . . . . . . . 278 maximum records in persisted local db file
list object id. . . . . . . . . . . . . . . . . . 123 for cdr data . . . . . . . . . . . . . 200
load balancing scheme. . . . . . . . . . . . 92 maximum redirections . . . . . . . . . 159, 216
load-balancing maximum size of script file. . . . . . . . . . 190
least used . . . . . . . . . . . . . . . . . . 92 maximum size of vxml document . . . . . . 189
round robin . . . . . . . . . . . . . . . . . 92 maximum size of xml/json file . . . . . . . . 190
local address contains srv domain name . . 222 maximum subdialog depths . . . . . . . . . 189
local listening address for activemq mcp send/receive enabled . . . . . . . . . . 116
broker (tls) . . . . . . . . . . . . . . 281 mcp-asr-usage-mode configuration option . 116
local rtp address . . . . . . . . . . . . . . . 177 Media Control Platform
local transport ipv4 address . . . . . . . . . 222 application modules . . . . . . . . . . . . 387
local transport ipv6 address . . . . . . 186, 222 client side connections . . . . . . . . . 70, 71
log configuration section . . . . . . . . 100, 201 conference configuration options . . . . . . 59
log expiration . . . . . . . . . . . . . . 101, 202 configuring . . . . . . . . . . . . . . . . 142
log message format . . . . . . . . . . 101, 203 data in SIP headers . . . . . . . . . . . . 445
log segmentation . . . . . . . . . . . . 101, 202 default SIP transports . . . . . . . . . . . . 41
logging . . . . . . . . . . . . . . . . . . . . 460 enabling ASR and TTS . . . . . . . . . . 146
logs functions . . . . . . . . . . . . . . . . . . 18
default configuration values . . . . . . . . . 67 module IDs . . . . . . . . . . . . . . . . 387
default filters . . . . . . . . . . . . . . . . . 60 specifier IDs . . . . . . . . . . . . . . . . 389
SRTP . . . . . . . . . . . . . . . . . . . . 44
media controller
M specifier IDs . . . . . . . . . . . . . . . . 414
media controller events . . . . . . . . . . . 414
managing media manager configuration section . . . . 257
sessions . . . . . . . . . . . . . . . . . . . 76 media resource board to use for csp . . . . 262
mapping media server markup language . . . . . . . 153
calls to IVR Profiles . . . . . . . . . . . . 128 mediacontroller configuration
configuring DNIS . . . . . . . . . . . . . 129 section. . . . .59, 98, 213, 218, 230, 233
max calls/sec to sip server . . . . . . . . . . 250 mediactrller configuration section . . . . . . . 59
max concurrent cdr queries . . . . . . . . . 279 mediamanager configuration section . . . . 256
max conference count . . . . . . . . . . . . 93 members . . . . . . . . . . . . . . . . . . . . 86
max conference size . . . . . . . . . . . . . 93 members 1 . . . . . . . . . . . . . . . . . . . 86
max db connection pool size . . . . . . . . . 249 members 2 . . . . . . . . . . . . . . . . . . . 86
max digits to dial . . . . . . . . . . . . . . . 261 memory output buffer size . . . . . . . . . . 204
max digits to receive in memory snapshot file name . . . . . . . . . 204
overlap receive mode . . . . . . . . 262 message file . . . . . . . . . . . . . . . . . 203
max ivr ports . . . . . . . . . . . . . . . . . 231 message_format configuration option . . . . . 65
max page count . . . . . . . . . . . . . . . 278 messaging configuration section. . . . . . . 277
max page size . . . . . . . . . . . . . . . . 279 metrics

default filters . . . . . . . . . . . . . . . . . 60 no cache url string . . . . . . . . . . . 159, 216
filter (IVR Profile) . . . . . . . . . . . . . 110 no cache url substring . . . . . . . . . . . . 267
VAR . . . . . . . . . . . . . . . . . . . . 450 noresource-response-code . . . . . . . 91, 474
metricsfilter configuration option . . . . . . . 110
mf sink log filter. . . . . . . . . . . . . . . . 199
mf sink metrics filter . . . . . . . . . . . . . 199 O
MIBs . . . . . . . . . . . . . . . . . . . . . 20
min db connection pool size . . . . . . . . . 249 offhook delay. . . . . . . . . . . . . . . . . 260
open-session timeout . . . . . . . . . . . . 207
minimum dashboard refreshing interval . . . 32
minimum download size for play . . . . . . . 259 openssl . . . . . . . . . . . . . . . . . . . . 46
minumum calls for service quality . . . . . . 283 OpenSSL Toolkit . . . . . . . . . . . . . . . . 46
or batch size . . . . . . . . . . . . . . . . . 200
mixed audio/video
file formats. . . . . . . . . . . . . . . . . 438 or reporting interval . . . . . . . . . . . . . 200
file formats, record. . . . . . . . . . . . . 441 out.<SIP request>.headers . . . . . . . . . 178
outbound
module IDs
listed . . . . . . . . . . . . . . . . . . . . 387 route set. . . . . . . . . . . . . . . . . . . 39
monitor configuration section. . . . . . . .85, 87 outbound call allowed . . . . . . . . . . . . 111
outbound call with native cpa ignore call connect
monitoring
events . . . . . . . . . . . . . . . . 158
reporting . . . . . . . . . . .293, 311, 329, 367
outbound calls with native cpa initial state . . 158
voice platform
outbound dialing
dashboard . 315, 316, 318, 319, 321, 322, 325
enabling . . . . . . . . . . . . . . . . . . 149
monitoring method . . . . . . . . . . . . . . 93
outgoing interface . . . . . . . . . . . 159, 216
monospace font . . . . . . . . . . . . . . . 570
output for level all . . . . . . . . . . . 100, 201
mpc configuration section . . . . . . . 154, 161
output for level debug . . . . . . . . . 101, 202
mrcp connection timeout . . . . . . . . . . . 205
mrcp proxy contact address . . . . . . . . . 206 output for level interaction . . . . . . . 101, 201
output for level standard . . . . . . . . 100, 201
MRCP server
assigning . . . . . . . . . . . . . . . . . 149 output for level trace . . . . . . . . . . 101, 202
MRCPv1
application templates . . . . . . . . . . . 147
speech servers in GVP . . . . . . . . . . 146
P
MRCPv2 pagecollector configuration section . . . . . 250
application templates . . . . . . . . . . . 147 parameters
speech servers in GVP . . . . . . . . . . 146 gvp.config . . . . . . . . . . . . . . . . . 443
msml . . . . . . . . . . . . . . . . . . . . . 153 gvp.rm.tenant-id. . . . . . . . . . . . . . . 24
msml allowed. . . . . . . . . . . . . . . . . 111 gvp-tenant-id . . . . . . . . . . . . . . . . 24
msml configuration section . . . . . . . 154, 168 vendor-specific, for TTS . . . . . . . . . 148
msml info allowed content types . . . . . . . 169 See also configuration options
mtinternal configuration section . . . . . . . 154 p-asserted-identity header . . . . . . . . . . 177
mtmpc configuration section . . . . . . . . . 154 paths
my member id . . . . . . . . . . . . . . . . 86 default SSL private key and certificate . . . 47
pause timeout . . . . . . . . . . . . . . . . 207
pay load factor . . . . . . . . . . . . . . . . 250
N p-called-party-id header . . . . . . . . . . . 177
persistence configuration section . . . 277, 282
native dtmf grammar maxage . . . . . . . . 188
persistent db file for cdr data. . . . . . . . . 199
native dtmf grammar maxstale . . . . . . . . 188
persistent db file for or data . . . . . . . . . 199
NETANN . . . . . . . . . . . . . . . . . . . 154
play
netann configuration section . . . . . . 154, 169
audio file formats . . . . . . . . . . . . . 435
network type . . . . . . . . . . . . . . . . . 262
audio/video file formats . . . . . . . . . . 438
new call confirmation . . . . . . . . . . . . . 262
video file formats . . . . . . . . . . . . . 438
new mrcp connection per session . . . . . . 191
policies
next retry interval . . . . . . . . . . . . . . . 250
database retention . . . . . . . . . . . . 274
NGI
postconnect . . . . . . . . . . . . . . . . . 457
configuration options . . . . . . . . . . . 155
postconnect priority . . . . . . . . . . . . . 260
module IDs . . . . . . . . . . . . . . . . 408
PRD#379932 . . . . . . . . . . . . . . . . 351
specifier IDs . . . . . . . . . . . . . . . . 408
User’s Guide 583

PRD#379956 . . . . . . . . . . . . . . . . . 351 burst limit . . . . . . . . . . . . . . 111
PRD#394700 . . . . . . . . . . . . . . . . . 162 range of directory numbers . . . . . . . . . 261
PRD#399267 . . . . . . . . . . . . . . . . . 112 read-only mode . . . . . . . . . . . . . . . 282
preconnect . . . . . . . . . . . . . . . . . . 457 Reason header . . . . . . . . . . . . . . . 474
preconnect priority . . . . . . . . . . . . . . 260 reclaim code . . . . . . . . . . . . . . . . . 126
prediction factor . . . . . . . . . . . . . . . 118 recognition-start-timers timeout . . . . . . . 207
preferred ip version used in basic http recognize timeout . . . . . . . . . . . . . . 207
access uri . . . . . . . . . . . . . . 215 record
preferred ip version used in create audio file formats . . . . . . . . . . . . . 439
session uri . . . . . . . . . . . . . . 215 audio/video file formats . . . . . . . . . . 441
preferred ip version used in SIP . . . . . . . 186 video file formats . . . . . . . . . . . . . 441
preferred ip version used in sip . . . . . . . 223 refer transfer hold . . . . . . . . . . . . . . 179
primary tserver listening port . . . . . . . . . 259 refer transfer retry refer
private key on the outbound leg . . . . . . . . . 180
creating . . . . . . . . . . . . . . . . . 46, 47 refresh-pattern rules . . . . . . . . . . . . . 269
default path . . . . . . . . . . . . . . . . . 47 registrar configuration section . . . . . . . . . 85
Procedure registration . . . . . . . . . . . . . . . 180, 181
Viewing or modifying GVP release asr engines on transfer . . . . . . . 189
configuration parameters, on page 32 . 256 remdial . . . . . . . . . . . . . . . . . . . . 149
profile types . . . . . . . . . . . . . . . . . 461 remdial configuration section . . . . . 154, 171
prompts remdial max calls . . . . . . . . . . . . . . 171
audio file formats . . . . . . . . . . . . . 435 remdial max client sockets. . . . . . . . . . 171
audio/video file formats . . . . . . . . . . 438 remdial port . . . . . . . . . . . . . . . . . 171
video file formats . . . . . . . . . . . . . 438 remdial telnet mode . . . . . . . . . . . . . 171
properties, device profile . . . . . . . . . . . 484 report filters . . . . . . . . . . . . . . . . . 306
protocols filter by granularity level . . . . . . . . . . 306
preferred SIP . . . . . . . . . . . . . . . . 38 filter by ivr profile . . . . . . . . . . . . . 307
provisioning reporting
ASR resources . . . . . . . . . . . . . . 146 active call list . . . . . . . . . . . . . . . 329
device profiles . . . . . . . . . . . . .484, 490 call completion summary . . . . . . . . . 377
GVP . . . . . . . . . . . . . . . . . . . . . 33 component call summary . . . . . . . . . 340
resources . . . . . . . . . . . . . . . . . . 89 component peaks . . . . . . . . . . . . . 347
TTS resources. . . . . . . . . . . . . . . 146 controlling access . . . . . . . . . . . . . 283
proxy configuration section. . . . . . . . .85, 88 database retention periods . . . . . . . . 274
ps service hostname . . . . . . . . . . . . . 102 database retention policies . . . . . . . . 274
ps service ip address. . . . . . . . . . . . . 102 enabling in GVP. . . . . . . . . . . . . . . 34
ps service port . . . . . . . . . . . . . . . . 102 granularity . . . . . . . . . . . . . . . . . 273
ps service protocol . . . . . . . . . . . . . . 102 historical call browser . . . . . . . . . . . 358
PSTN Connector historical reports . . . . . . . . . . . . . 337
client side connections . . . . . . . . . . . 73 HTTP Basic Authorization . . . . . . . . 283
functions . . . . . . . . . . . . . . . . . . . 19 ivr action list. . . . . . . . . . . . . . . . 363
PSTN Connector configuring. . . . . . . . . 256 ivr action summary . . . . . . . . . . . . 380
pstn connector sip port . . . . . . . . . . . . 257 ivr profile peaks . . . . . . . . . . . . . . 344
pstnc dashboard . . . . . . . . . . . . . . . 325 last ivr action used . . . . . . . . . . . . 382
ptime . . . . . . . . . . . . . . . . . . 162, 163 overview . . . . . . . . . . . . . . . . . 293
real-time reports. . . . . . . . . . . . . . 329
report filters . . . . . . . . . . . . . . . . 306
Q running a report . . . . . . . 294, 378, 380, 382
service quality reports. . . . . . . . . . . 367
queue low watermark . . . . . . . . . . . . 248
voice application reports . . . . . . . . . 377
voice platform dashboard . . . . . . . . . 311
R reporting configurationection. . . . . . . . . 281
reporting configuration
raise alarm for dialing rule section. . . . . . . . 102, 277, 279, 282
based rejection . . . . . . . . . . . 112 Reporting Server
raise alarm for exceeding client side connections . . . . . . . . . . . 71

configuring. . . . . . . . . . . . . . . . . 271 save ccxml files . . . . . . . . . . . . . . . 215
functions . . . . . . . . . . . . . . . . . . . 19 save script files . . . . . . . . . . . . . . . 215
web server and HTTPS . . . . . . . . . . . 44 schedule configuration section. . . . . 277, 283
reporting server http timeout . . . . . . . . . 32 schema
request acceptance time-out on/resource dn http xml . . . . . . . . . . . . . . . . . . 542
registration failure . . . . . . . . . . 250 script id key name . . . . . . . . . . . . . . 232
request acceptance time-out on sdp local host . . . . . . . . . . . . . . . . 220
sips connection failure . . . . . . . . 250 sdp local host ipv6 . . . . . . . . . . . . . . 220
request batch size . . . . . . . . . . . . . . 248 sdp orign name map . . . . . . . . . . . . . 166
resource dn registration failure sections
recovery interval . . . . . . . . . . . 250 See configuration sections
resource groups secure communications
configuring. . . . . . . . . . . . . . 89, 90, 94 enabling . . . . . . . . . . . . . . . . . . . 43
deleting . . . . . . . . . . . . . . . . . . . 91 secure protocol version . . . . . . . . . . . 247
Resource Manager secure random algorithm . . . . . . . . . . . 99
client side connections . . . . . . . . . . . 71 security provider . . . . . . . . . . . . . . . . 99
configuring. . . . . . . . . . . . . . . . . . 83 segment configuration option . . . . . . . . . 66
default SIP transports . . . . . . . . . . . . 40 self-signed SSL certificate . . . . . . . . . . . 47
functions . . . . . . . . . . . . . . . . . . . 17 send alert . . . . . . . . . . . . . . . . . . 181
managing resources. . . . . . . . . . . . . 90 send dtmf relay sip info messages. . . . . . 170
module IDs . . . . . . . . . . . . . . . . 417 send sdp in invite for media redirect . . . . . 187
session management . . . . . . . . . . . . 76 send sip progressing. . . . . . . . . . . . . 214
session timers . . . . . . . . . . . . 76, 78, 79 service quality reports . . . . . . . . . . . . 367
specifier IDs . . . . . . . . . . . . . . . . 417 sq call failures . . . . . . . . . . . . . . . 367
resources sq failure summary . . . . . . . . . . . . 370
managing . . . . . . . . . . . . . . . . . . 90 sq latency summary. . . . . . . . . . . . 372
provisioning . . . . . . . . . . . . . . . . . 89 service type . . . . . . . . . . . . . . . . . 109
provisioning ASR and TTS . . . . . . . . 146 services
resume timeout. . . . . . . . . . . . . . . . 208 capabilities . . . . . . . . . . . . . . . . 117
ringback filename . . . . . . . . . . . . . . 259 conference, configuring . . . . . . . . . . . 59
rm configuration section . . . . . . . . . .85, 87 session
root directory for cpd recording. . . . . . . . 169 identifiers . . . . . . . . . . . . . . . . . . 22
root directory for play media . . . . . . . . . 169 timer configuration options . . . . . . . . . 77
root directory for prompt media . . . . . . . 169 timers . . . . . . . . . . . . . . . . . . . . 76
root directory for record media . . . . . 169, 170 session clean interval . . . . . . . . . . . . 206
round robin load balancing . . . . . . . . . . 92 session configuration section . . . . . . . . 213
route description . . . . . . . . . . . . . . . 260 session max idle timeout. . . . . . . . . . . 207
route set . . . . . . . . . . . . . . . . . . . 223 session timer interval . . . . . . . . . . . . 257
configuring. . . . . . . . . . . . . . . . . . 39 Session ID . . . . . . . . . . . . . . . . . . . 22
route type. . . . . . . . . . . . . . . . . . . 264 Session-Expires header . . . . . . . . . . . . 76
routeset configuration options . . . . . . . . 39 sessmgr configuration section . . . .43, 154, 187
routing set-params timeout . . . . . . . . . . . . . 208
configuring. . . . . . . . . . . . . . . . . . 39 Settings tab . . . . . . . . . . . . . . . . . . 30
rs db maintenance process . . . . . . . . . 283 changing the display . . . . . . . . . . . . 31
rs.query.limit.<granularity> . . . . . . . . . . 273 show local time . . . . . . . . . . . . . . . . 33
rtp de-jitter delay . . . . . . . . . . . . . . . 166 signaling type . . . . . . . . . . . . . . . . 263
rtp de-jitter timeout . . . . . . . . . . . . . . 166 signal-to-noise ratio . . . . . . . . . . . . . 461
RTP media path . . . . . . . . . . . . . . . 154 SIP . . . . . . . . . . . . . . . . . . . . . . . 39
rtp send mode . . . . . . . . . . . . . . . . 165 default transports . . . . . . . . . . . . . . 40
rtsp port range for mrcpv1client . . . . . . . 205 INFO messages. . . . . . . . . . . . . . 174
rule-<n> configuration option. . . . . . 119, 120 transports . . . . . . . . . . . . . . . . . . 38
running a report . . . . . . . . . . . . . . . 294 sip configuration
section. 43, 155, 171, 213, 220, 230, 233
sip destination ip address . . . . . . . . . . 257
S sip destination port number . . . . . . . . . 257
sip header for dnis . . . . . . . . . . . . . . . 87
safe ports, Squid . . . . . . . . . . . . . . . 270
User’s Guide 585

SIP headers Squid
used by Media Control Platform . . . . . . 445 configuring . . . . . . . . . . . . . . . . 268
X-Genesys-CallUUID . . . . . . . . . . . . 22 refresh-pattern rules . . . . . . . . . . . 269
X-Genesys-gsw-ivr-profile-id . . . . . . . . 24 safe ports . . . . . . . . . . . . . . . . . 270
X-Genesys-gsw-session-dbid . . . . . . . . 24 SSL ports . . . . . . . . . . . . . . . . . 270
X-Genesys-GVP-Session-ID . . . . . . . . 22 srm default response timeout . . . . . . . . 188
X-Genesys-RM-Application-dbid . . . . . . 24 srm ping frequency . . . . . . . . . . . . . 188
sip proxy . . . . . . . . . . . . . . . . . . . 220 srm ping timeout . . . . . . . . . . . . . . . 188
sip resource options interval . . . . . . . . . 87 SRTP
SIP responses . . . . . . . . . . . . . . . . 449 enabling . . . . . . . . . . . . . . . . . 43, 44
100 Trying . . . . . . . . . . . . . . . . . 466 srtp mode . . . . . . . . . . . . . . . . . . 164
180 Ringing . . . . . . . . . . . . . . . . 466 ssg configuration section . . . . . . . 246, 248
183 Session Progress . . . . . . . . . . . 466 ssg dashboard . . . . . . . . . . . . . . . . 322
202 Accepted . . . . . . . . . . . . . . . 467 SSL
302 Moved Temporarily . . . . . . . . . . 467 certificate, creating . . . . . . . . . . . . . 46
3xx. . . . . . . . . . . . . . . . . . . . . 467 default certificate path. . . . . . . . . . . . 47
400 Bad Request . . . . . . . . . . . . . 468 default private key path . . . . . . . . . . . 47
403 Forbidden . . . . . . . . . . . . . . . 468 Fetching Module configuration options . . 267
404 Not Found. . . . . . . . . . . . . . . 469 ports, Squid . . . . . . . . . . . . . . . . 270
405 Method Not Allowed . . . . . . . . . 469 private key, creating. . . . . . . . . . . 46, 47
408 Request Timeout . . . . . . . . . . . 469 self-signed certificate, creating . . . . . . . 47
420 Bad Extension . . . . . . . . . . . . 469 ssl ca info . . . . . . . . . . . . . . . . . . 160
423 Interval Too Brief . . . . . . . . . . . 470 ssl ca path . . . . . . . . . . . . . . . 160, 217
480 Temporarily Unavailable . . . . . . . 470 ssl certificate . . . . . . . . . . . . . . 159, 216
481 Call Does Not Exist . . . . . . . . . . 471 ssl certificate algorithm . . . . . . . . . . . . 99
487 Request Terminated . . . . . . . . . 471 ssl certificate type . . . . . . . . . . . 160, 217
488 Not Acceptable Here . . . . . . . . . 471 ssl cipher list . . . . . . . . . . . . . . 161, 218
500 Server Internal Error . . . . . . . . . 472 ssl key . . . . . . . . . . . . . . . . . 160, 217
503 Service Unavailable. . . . . . . . . . 472 ssl key password. . . . . . . . . . 100, 160, 217
customizing . . . . . . . . . . . . . . . . . 74 ssl key type . . . . . . . . . . . . . . 160, 217
gateway failures . . . . . . . . . . . . . . . 91 ssl keystore password . . . . . . . . . . . . . 98
in GVP . . . . . . . . . . . . . . . . . . . 465 ssl keystore path . . . . . . . . . . . . . . . . 98
sip session timer interval . . . . . . . . 110, 137 ssl keystore type . . . . . . . . . . . . . . . . 99
sip static route list . . . . . . . . . . . . . . 223 ssl random file seed . . . . . . . . . . 160, 217
sip unavailable resource options ssl verify host . . . . . . . . . . . . . 161, 218
interval . . . . . . . . . . . . . . . . 88 ssl version . . . . . . . . . . . . . . . 160, 217
sip.sessiontimer configuration option . . . . 77 ssl_* configuration options . . . . . . . . . . 267
sip.transport.<x> configuration options. . . . 38 stack configuration section. . . . . 155, 198, 205
SIPS standard configuration option . . . . . . . . . 64
enabling . . . . . . . . . . . . . . . . . . . 43 stop timeout . . . . . . . . . . . . . . . . . 208
supported . . . . . . . . . . . . . . . . . . 42 strict grammar mode . . . . . . . . . . . . . 189
sips connection failure recovery subscription configuration section . . . . . . . 85
interval . . . . . . . . . . . . . . . . 250 summarization buffer time . . . . . . . . . . 282
snmp configuration section. . . . . . . 198, 247 Supplementary Services Gateway
snmp configuration client side connections . . . . . . . . . . . 72
section . . . . . . . . . . . 198, 205, 230 configuring . . . . . . . . . . . . . . . . 246
snmp task timeout . . . . . . . . . . . . 68, 205 functions . . . . . . . . . . . . . . . . . . 19
speak timeout . . . . . . . . . . . . . . . . 208 http interface . . . . . . . . . . . . . . . 515
specifiers module IDs . . . . . . . . . . . . . . . . 426
IDs . . . . . . . . . . . . . . . . . . . . . 387 Supplemetary Services Gateway
speech resource uri . . . . . . . . . . . . . 191 specifier IDs . . . . . . . . . . . . . . . . 426
sq call failures . . . . . . . . . . . . . . . . 367 support
sq failure summary . . . . . . . . . . . . . . 370 IPv6 . . . . . . . . . . . . . . . . . . . . . 53
sq latency dashboard . . . . . . . . . . . . 319 SIPS . . . . . . . . . . . . . . . . . . . . 42
sq latency summary . . . . . . . . . . . . . 372 supported gateway cpa events . . . . . . . 158
square brackets . . . . . . . . . . . . . . . 570 supported local codec type . . . . . . . . . 257

T configuration options . . . . . . . . . . . 191
enabling . . . . . . . . . . . . . . . . . . 146
t1-rb anidnis delimieter . . . . . . . . . . . . 263 provisioning resources . . . . . . . . . . 146
t1-rb anidnis order . . . . . . . . . . . . . . 263 vendor-specific parameters . . . . . . . . 148
t1-rb protocol file . . . . . . . . . . . . . . . 263 tts configuration section . . . . . . . . 155, 187
t1-rb remove anidnis . . . . . . . . . . . . . 263 tts default engine. . . . . . . . . . . . . . . 121
task summary tts engine default. . . . . . . . . . . . . . . 187
configuring GVP . . . . . . . . . . . . . . . 33 tts gender . . . . . . . . . . . . . . . . . . 127
provisioning GVP . . . . . . . . . . . . . . 33 tts vendor . . . . . . . . . . . . . . . . . . 127
tcp reconnect interval . . . . . . . . . . . . 208 tuning . . . . . . . . . . . . . . . 459, 462, 463
tenant. . . . . . . . . . . . . . . . . . 137, 139 two channel transfer type . . . . . . . . . . 264
assigning default IVR Profile . . . . . . . 137 type styles
IVR Profile settings . . . . . . . . . . . . 136 conventions . . . . . . . . . . . . . . . . 569
tenant, Environment italic . . . . . . . . . . . . . . . . . . . . 569
session timers . . . . . . . . . . . . . . . . 77 monospace . . . . . . . . . . . . . . . . 570
tenant1 configuration section. . . . . . 234, 246 typographical styles . . . . . . . . . . . . . 569
threshold criteria for latency . . . . . . . . . 279
time format for log messages . . . . . 102, 203
time generation for log messages . . . . . . 203 U
time to live limit . . . . . . . . . . . . . . . . 249
timeouts . . . . . . . . . . . . . . . . . .76, 79 ULAW . . . . . . . . . . . . . . . . . . . . 163
timers. . . . . . . . . . . . . . . . . . . . . 79 universals grammar uri . . . . . . . . . . . 189
inactivity . . . . . . . . . . . . . . . . . . . 76 unknown headers allowed
session . . . . . . . . . . . . . . . . . . . 76 or a sip message . . . . . . . . . . 220
session expiry . . . . . . . . . . . . . . . . 76 URL
timezone offset . . . . . . . . . . . . . . . . 33 for Genesys Administrator . . . . . . . . . 21
TLS usage limit exceeded response code . . . . 114
and CCXML applications . . . . . . . . . . 43 usage limits . . . . . . . . . . . . . . . . . 112
SIP transport . . . . . . . . . . . . . . . . 43 use original gateway in
tls certificate for reporting client . . . . . . . 201 outbound call . . . . . . . . . . . . 179
TLSv1 . . . . . . . . . . . . . . . . . . . . 53 use same gateway . . . . . . . . . . . . . . 118
toll free number. . . . . . . . . . . . . . . . 109 use tserver to make calls . . . . . . . . . . 259
tone definition . . . . . . . . . . . . . . . . 456 userdata prefix . . . . . . . . . . . . . . . . 190
trace configuration option . . . . . . . . . . 64 using Genesys Administrator . . . . . . . . . 30
transaction configuration section . . . . . . . 277
transcoders. . . . . . . . . . . . . . . . . . 167
transfer allowed . . . . . . . . . . . . 111, 190
V
transfer connect . . . . . . . . . . . . . . . 126 VAR
transfer connect script . . . . . . . . . . . . 126 metrics . . . . . . . . . . . . . . . . . . 450
transfer connect url. . . . . . . . . . . . . . 126 VAR API . . . . . . . . . . . . . . . . . . . 495
transfer copy headers . . . . . . . . . . . . 183 VAR call detail records . . . . . . . . . . 495
transfer methods . . . . . . . . . . . . . . . 182 VAR call summary records . . . . . . . . 496
transfer option . . . . . . . . . . . . . . . . 126 VAR ivr action summary records . . . . . 496
transfer type . . . . . . . . . . . . . . . . . 126 voicexml <log> extension . . . . . . . 497, 498
transport instance 0 . . . . . . . . 184, 185, 186 vendor name . . . . . . . . . . . . . . . . . 191
transport instance 1 . . . . . . . . . . . . . 186 vendor-specific parameters, for TTS. . . . . 148
transport instance 2 . . . . . . . . . . . . . 186 verbose level. . . . . . . . . . . . .67, 100, 201
transports verify peer certificate. . . . . . . . . . 160, 217
default . . . . . . . . . . . . . . . . . . . . 40 version numbering, document . . . . . . . . 569
transports, SIP video
for Resource Manager. . . . . . . . . . . . 38 file formats, play. . . . . . . . . . . . . . 438
preferred protocol . . . . . . . . . . . . . . 38 file formats, record . . . . . . . . . . . . 441
trap hook . . . . . . . . . . . . . . . . . . . 127 video support . . . . . . . . 455, 503, 555, 559
trunk group id. . . . . . . . . . . . . . . . . 231 advanced features . . . . . . . . . . . . 506
tserver reconnect timeout . . . . . . . . . . 260 features . . . . . . . . . . . . . . . . . . 504
TTS protocols . . . . . . . . . . . . . . . . . 503
User’s Guide 587

specifications . . . . . . . . . . . . . . . 503
virtual ip address . . . . . . . . . . . . . . . 86
voice application reporter
API. . . . . . . . . . . . . . . . . . . . . 495
voice application reports . . . . . . . . . . . 377
voice platform dashboard . . . . . . . . . . 311
component utilization . . . . . . . . .316, 318
fetch dashboard . . . . . . . . . . . . . . 321
ivr profile utilization . . . . . . . . . . . . 315
pstnc dashboard. . . . . . . . . . . . . . 325
sq latency dashboard . . . . . . . . . . . 319
ssg dashboard. . . . . . . . . . . . . . . 322
Voice Platform Solution (VPS) . . . . . . . . 29
VoiceXML applications
and IVR Profiles . . . . . . . . . . . . . . 103
identifiers . . . . . . . . . . . . . . . . . . 23
receiving events . . . . . . . . . . . . . . 174
sending events . . . . . . . . . . . . . . 174
triggering . . . . . . . . . . . . . . . . . . 29
voicexml dialog allowed . . . . . . . . . . . 111
voicexml level2 usage limit . . . . . . . . . . 112
voicexml level3 usage limit . . . . . . . . . . 112
voicexml url invite . . . . . . . . . . . . . . 183
voicexml usage limit . . . . . . . . . . . . . 113
vrm configuration section. . . . . . . . 155, 188
vrmproxy configuration section. . . . . 198, 205
vxmli configuration section . . . . . . . 155, 189
W
wait for offhook confirmation . . . . . . . . . 260
warning headers . . . . . . . . . . . . . . . 183
web server
Reporting Server, and HTTPS. . . . . . . . 44
X
X-Genesys-CallUUID header . . . . . . . . 22
X-Genesys-gsw-ivr-profile-id header . . . . . 24
X-Genesys-gsw-session-dbid header . . . . 24
X-Genesys-GVP-Session-ID header
defined. . . . . . . . . . . . . . . . . . . . 22
X-Genesys-RM-Application-dbid header . . . 24
X-Lite (device profile). . . . . . . . . . . . . 493

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Genesys Voice Platform 8.

Your Responsibility for Your System’s Security

Customer Care from VARs

Customer Care from Genesys

Ordering and Licensing Information

Part 1 Provisioning GVP........................................................ 27

Chapter 2 Configuration and Provisioning Overview......................................... 29

Chapter 3 Configuring Common Features........................................................... 37

Chapter 4 Configuring the Resource Manager.................................................... 83

Chapter 5 Configuring Policy Server ................................................................... 97

Chapter 6 Provisioning IVR Profiles................................................................... 103

Chapter 7 Configuring the Media Control Platform .......................................... 141

4 Genesys Voice Platform 8.5

Chapter 9 Configuring the Call Control Platform.............................................. 209

Chapter 10 Configuring the CTI Connector ......................................................... 225

Chapter 11 Configuring the Supplementary Services Gateway ........................ 245

Chapter 12 Configuring the PSTN Connector ..................................................... 255

Chapter 13 Configuring the Fetching Module and Squid Proxy........................ 265

Chapter 15 Configuring GVP in Multi-Site Environments .................................. 287

Part 2 Monitoring GVP ......................................................... 291

Chapter 16 Reporting Overview............................................................................ 293

Chapter 17 Voice Platform Dashboards............................................................... 311

6 Genesys Voice Platform 8.5

Chapter 19 Historical Reports............................................................................... 337

Chapter 20 Service Quality Reports ..................................................................... 367

Chapter 21 Voice Application Reports................................................................. 377

Part 3 Appendixes................................................................ 385

Appendix A Module and Specifier IDs................................................................... 387

Appendix B Media Control Platform Reference Information............................... 435

Appendix C Tuning Call Progress Detection ........................................................ 455

Appendix D SIP Response Codes.......................................................................... 465

8 Genesys Voice Platform 8.5

Appendix F VAR API ............................................................................................... 495

Appendix G Video Support ..................................................................................... 503

Appendix H Custom Log Sinks .............................................................................. 509

Appendix I SSG HTTP Interface............................................................................ 515

Appendix J Network Partitioning Configuration Options ................................... 555

Appendix K SIP Customizable Headers and Parameters .................................... 559

Supplements Related Documentation Resources ................................................... 565

Document Conventions ...................................................................... 569

Index ............................................................................................................... 571

10 Genesys Voice Platform 8.5

This preface contains the following sections:

About Genesys Voice Platform

Routing Solution (ERS), Network Routing Solution (NRS), and

Note: GVP is a scalable solution with flexible configuration and deployment

Making Comments on This Document

Contacting Genesys Customer Care

14 Genesys Voice Platform 8.5

Document Change History

16 Genesys Voice Platform 8.5

18 Genesys Voice Platform 8.5

20 Genesys Voice Platform 8.5

Figure 1: Genesys Administrator

To access Genesys Administrator in your Genesys deployment, go to the

GVP Identifiers and SIP Headers

Table 1: GVP Session Identifiers and SIP Headers

Session ID Description SIP Header

Genesys CallUUID The Universally Unique Identifier (UUID) X-Genesys-CallUUID