Genesys 7.6 SIP Server Deployment Guide
Reklama
Reklama
Framework 7.6
SIP Server
Deployment Guide
The information contained herein is proprietary and confidential and cannot be disclosed or duplicated without the prior written consent of Genesys Telecommunications Laboratories, Inc.
Copyright © 2004–2009 Genesys Telecommunications Laboratories, Inc. All rights reserved.
About Genesys
Genesys Telecommunications Laboratories, Inc., a subsidiary of Alcatel-Lucent, is 100% focused on software for call centers. Genesys recognizes that better interactions drive better business and build company reputations. Customer service solutions from Genesys deliver on this promise for Global 2000 enterprises, government organizations, and telecommunications service providers across 80 countries, directing more than 100 million customer interactions every day. Sophisticated routing and reporting across voice, e-mail, and Web channels ensure that customers are quickly connected to the best available resource—the first time. Genesys offers solutions for customer service, help desks, order desks, collections, outbound telesales and service, and workforce management. Visit www.genesyslab.com
for more information.
Each product has its own documentation for online viewing at the Genesys Technical Support website or on the
Documentation Library DVD, which is available from Genesys upon request. For more information, contact your sales representative.
Notice
Although reasonable effort is made to ensure that the information in this document is complete and accurate at the time of release, Genesys Telecommunications Laboratories, Inc., cannot assume responsibility for any existing errors.
Changes and/or corrections to the information contained in this document may be incorporated in future versions.
Your Responsibility for Your System’s Security
You are responsible for the security of your system. Product administration to prevent unauthorized use is your responsibility. Your system administrator should read all documents provided with this product to fully understand the features available that reduce your risk of incurring charges for unlicensed use of Genesys products.
Trademarks
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Inc. All other trademarks and trade names referred to in this document are the property of other companies. The
Crystal monospace font is used by permission of Software Renovation Corporation, www.SoftwareRenovation.com
.
Technical Support from VARs
If you have purchased support from a value-added reseller (VAR), please contact the VAR for technical support.
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Region
North and Latin America
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Telephone
+888-369-5555 or +506-674-6767
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+81-3-6361-8950
Prior to contacting technical support, please refer to the
Genesys Technical Support Guide
for complete contact information and procedures.
Ordering and Licensing Information
Complete information on ordering and licensing Genesys products can be found in the
Genesys 7 Licensing Guide
.
Released by
Genesys Telecommunications Laboratories, Inc. www.genesyslab.com
Document Version: 76fr_dep-sip_03-2009_v7.6.003.00
Preface
Part 1
Chapter 1
Chapter 2
Table of Contents
Reading Prerequisites ........................................................................ 14
Document Conventions ........................................................................... 16
Making Comments on This Document .................................................... 19
Part One: SIP Server Deployment ............................. 21
SIP Server Fundamentals .................................................................... 25
SIP Server Deployment Modes .......................................................... 26
Media Server Deployment Architecture .............................................. 29
Redundant SIP Servers........................................................................... 30
SIP Server General Deployment.......................................................... 33
Software Requirements ...................................................................... 34
Hardware and Network Environment Requirements .......................... 34
Licensing Requirements ..................................................................... 35
About Configuration Options............................................................... 37
Network Considerations .......................................................................... 37
Bandwidth Requirements.................................................................... 39
Remote Agent Configuration .............................................................. 39
SIP Server—Deployment Guide 3
Table of Contents
Chapter 3
Chapter 4
Chapter 5
Deployment Sequence ............................................................................ 40
Wizard Deployment of SIP Server........................................................... 41
Wizard Configuration of SIP Server.................................................... 41
Wizard Installation of SIP Server ........................................................ 42
Manual Deployment of SIP Server .......................................................... 44
Manual Configuration of Telephony Objects ....................................... 44
Manual Configuration of SIP Server ................................................... 46
Manual Installation of SIP Server........................................................ 47
High-Availability Deployment.............................................................. 51
Warm Standby Redundancy Type ...................................................... 53
Hot Standby Redundancy Type .......................................................... 53
Synchronization Between Redundant SIP Servers ............................ 54
Warm Standby Configuration................................................................... 55
General Order of Deployment............................................................. 55
Manual Modification of SIP Servers for Warm Standby ...................... 56
Hot Standby Configuration....................................................................... 57
General Order of Deployment............................................................. 58
Manual Modification of SIP Servers for Hot Standby .......................... 58
Warm/Hot Standby Deployment on Solaris ............................................. 60
Warm/Hot Standby Deployment on Windows.......................................... 62
Network Load Balancing..................................................................... 62
Configuration Procedures ................................................................... 63
Starting and Stopping SIP Server ....................................................... 69
Command-Line Parameters .................................................................... 69
Starting and Stopping with the Management Layer ................................. 71
Verifying Successful Startup .................................................................... 74
Starting and Stopping with Windows Services Manager ......................... 75
SIP Devices Support ............................................................................ 77
4 Framework 7.6
Table of Contents
Chapter 6
Configuring Devices and Services........................................................... 78
Configuring ACD Queues ................................................................... 79
Configuring MCUs .............................................................................. 79
Configuring Endpoints......................................................................... 81
Configuring Gateways ........................................................................ 83
Configuring Music Servers.................................................................. 85
Configuring Routing Points ................................................................. 87
Configuring Softswitches .................................................................... 88
Configuring an Application Service..................................................... 89
Configuring a Recording Service ........................................................ 90
Configuring a Treatment Service ........................................................ 91
Configuring Out-of-Service Detection...................................................... 92
Load-Balancing Mode......................................................................... 92
Primary-Standby Mode ....................................................................... 92
Active Out-of-Service Detection.......................................................... 93
Configuring Agent Logins ........................................................................ 94
Configuring Stream Manager................................................................... 94
SIP Server Feature Support................................................................. 97
Associating an ACD Queue with a Routing Point.................................... 98
Asterisk Voice Mail Integration ................................................................ 98
Genesys Voice Mail Configuration Adapter for Asterisk ..................... 99
Message Waiting Indicator Functionality ............................................ 99
Feature Configuration ....................................................................... 100
Busy Lamp Field Feature ...................................................................... 101
Calls Outside the Premise ..................................................................... 101
Regular Call Recording..................................................................... 102
Feature Configuration ....................................................................... 103
Emergency (Manual) Call Recording................................................ 104
Feature Configuration ....................................................................... 107
Multi-Site Supervision ....................................................................... 112
Remote Supervision ......................................................................... 115
Call Transfer and Conference................................................................ 123
Conference Calls .............................................................................. 125
Consultation Transfers and Conferences ......................................... 125
Feature Limitations ........................................................................... 126
Outbound Dialing Rules.................................................................... 127
Ring-Through Rules.......................................................................... 134
Feature Configuration ....................................................................... 134
SIP Server—Deployment Guide 5
Table of Contents
6
DTMF Tones Generation ....................................................................... 137
Feature Configuration ....................................................................... 138
Business-Call Handling..................................................................... 138
Emulated Agents Support ................................................................. 139
Endpoint Service Monitoring.................................................................. 142
Instant Messaging Transcript............................................................ 143
Supported Call Operations ............................................................... 144
Feature Configuration ....................................................................... 147
Mapping SIP Headers and SDP Messages........................................... 149
From SIP Messages to T-Library Messages..................................... 149
From T-Library Messages to SIP Messages..................................... 151
SDP Message Mapping .................................................................... 154
Music and Announcements ................................................................... 154
Announcement Treatments on Routing Points ................................. 155
Music Treatments on Routing Points ................................................ 156
Other Treatments on Routing Points................................................. 157
No-Answer Supervision ......................................................................... 157
Business and Private Calls ............................................................... 157
Agent No-Answer Supervision.......................................................... 158
Extension No-Answer Supervision ................................................... 158
Position No-Answer Supervision ...................................................... 159
Device-Specific Overrides................................................................. 159
Extension Attributes for Overrides for Individual Calls...................... 159
Feature Limitations ........................................................................... 160
Feature Configuration ....................................................................... 160
Presence Subscription........................................................................... 161
Feature Configuration ....................................................................... 161
Updating Agent State........................................................................ 162
Providing Call Participant Info ............................................................... 164
Feature Configuration ....................................................................... 165
Remote Server Registration .................................................................. 166
Trunk Capacity Support ......................................................................... 166
Trunk Optimization for Multi-Site Transfers ........................................... 166
Feature Configuration ....................................................................... 170
Feature Limitation ............................................................................. 170
Framework 7.6
Table of Contents
Chapter 7
Chapter 8
Part 2
Chapter 9
Chapter 10
Other Supported Scenarios .............................................................. 172
Feature Configuration ....................................................................... 173
Genesys Voice Platform Integration ...................................................... 175
T-Library Functionality Support........................................................ 177
Using T-Library Functions...................................................................... 177
Using the Extensions Attribute .............................................................. 186
Third-Party Equipment—Known Limitations ..................................... 195
SIP Server Configuration Options .................................................... 197
Application-Level Options...................................................................... 197
T-Server Section............................................................................... 197
UPDATE, INVITE, INFO, and REFER Sections ............................... 227
Agent Login–Level and DN-Level Options ............................................ 228
GVP Integration Options........................................................................ 247
Changes from 7.5 to 7.6 ........................................................................ 249
New for All T-Servers in 7.6................................................................... 254
T-Server Fundamentals...................................................................... 255
Learning About T-Server ....................................................................... 255
Framework and Media Layer Architecture........................................ 255
T-Server Requests and Events ......................................................... 257
Advanced Disconnect Detection Protocol ............................................. 261
Redundant T-Servers ............................................................................ 262
Multi-Site Fundamentals........................................................................ 268
ISCC Call Data Transfer Service ........................................................... 269
ISCC Transaction Types ................................................................... 274
T-Server Transaction Type Support .................................................. 282
SIP Server—Deployment Guide 7
Table of Contents
Chapter 11
Chapter 12
Transfer Connect Service Feature.................................................... 286
Number Translation Feature.................................................................. 291
Number Translation Rules ................................................................ 292
Network Attended Transfer/Conference Feature................................... 299
Event Propagation Feature.................................................................... 301
User Data Propagation ..................................................................... 301
Party Events Propagation ................................................................. 303
Basic and Advanced Configuration................................................... 303
ISCC Transaction Monitoring Feature ................................................... 306
Configuring Multi-Site Support............................................................... 307
Switches and Access Codes ............................................................ 308
Configuration Examples.................................................................... 318
Common Configuration Options....................................................... 321
Setting Configuration Options................................................................ 321
Log Output Options........................................................................... 328
Debug Log Options........................................................................... 333
Log-Extended Section ........................................................................... 336
Changes from 7.5 to 7.6 ........................................................................ 340
T-Server Common Configuration Options ....................................... 343
Setting Configuration Options................................................................ 343
Agent-Reservation Section.................................................................... 351
Multi-Site Support Section ..................................................................... 352
ISCC Transaction Options ................................................................ 355
Transfer Connect Service Options.................................................... 358
ISCC/COF Options ........................................................................... 359
Event Propagation Option................................................................. 361
8 Framework 7.6
Table of Contents
Part 3
Chapter 13
Chapter 14
Appendix
Number Translation Option............................................................... 361
Translation Rules Section...................................................................... 362
Backup-Synchronization Section........................................................... 362
Timeout Value Format ........................................................................... 365
Changes from Release 7.5 to 7.6.......................................................... 366
Part Three: DMX Reference Information ................. 367
DMX Deployment ................................................................................ 369
DMX General Network Modes .......................................................... 369
Wizard Configuration ........................................................................ 375
Manual Configuration........................................................................ 375
DMX Configuration Options................................................................... 376
X-Config Section............................................................................... 377
Gatekeeper Section .......................................................................... 386
Changes from 7.5 to 7.6 ................................................................... 388
Configuring for Use with the Management Layer .................................. 388
VOIP Gateways ................................................................................ 389
Voice Terminal Endpoints.................................................................. 390
Installing using the Wizard ................................................................ 391
Installing Manually ............................................................................ 391
Starting and Stopping DMX ................................................................... 392
VOIP and Video-Over-IP Support.......................................................... 399
Session Initiation Protocol (SIP) Overview ...................................... 401
TCP/IP Model ................................................................................... 401
SIP Server—Deployment Guide 9
Table of Contents
Index
SIP Functionality............................................................................... 402
Proxy and Registrar Servers............................................................. 403
SIP Messaging.................................................................................. 403
ITU-T and Call Models...................................................................... 406
H.xxx Protocols................................................................................. 406
10 Framework 7.6
List of Procedures
Installing SIP Server on UNIX using the wizard . . . . . . . . . . . . . . . . . . . 42
Installing SIP Server on Windows using the wizard . . . . . . . . . . . . . . . . 43
Configuring SIP Server manually . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46
Installing SIP Server on UNIX manually . . . . . . . . . . . . . . . . . . . . . . . . . 47
Installing SIP Server on Windows manually . . . . . . . . . . . . . . . . . . . . . . 48
Verifying the manual installation of SIP Server . . . . . . . . . . . . . . . . . . . 49
Modifying the primary SIP Server configuration for warm standby. . . . . 56
Modifying the backup SIP Server configuration for warm standby . . . . . 57
Modifying the primary SIP Server configuration for hot standby . . . . . . 58
Modifying the backup SIP Server configuration for hot standby. . . . . . . 60
Configuring cluster hosts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
Configuring primary/backup SIP Servers . . . . . . . . . . . . . . . . . . . . . . . . 66
Configuring Alarm Conditions. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
Configuring SIP Server to start with the Management Layer . . . . . . . . . 71
Starting SIP Server on UNIX with a startup file . . . . . . . . . . . . . . . . . . . 72
Starting SIP Server on Windows with a startup file . . . . . . . . . . . . . . . . 73
Starting SIP Server on UNIX manually. . . . . . . . . . . . . . . . . . . . . . . . . . 74
Starting SIP Server on Windows manually. . . . . . . . . . . . . . . . . . . . . . . 74
Stopping SIP Server on UNIX manually . . . . . . . . . . . . . . . . . . . . . . . . . 75
Stopping SIP Server on Windows manually . . . . . . . . . . . . . . . . . . . . . . 75
Configuring an ACD Queue . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 79
Configuring a gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84
Configuring a Music Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86
Configuring Routing Points. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 87
Configuring softswitches. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 88
Configuring an application service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89
Configuring a recording service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90
SIP Server—Deployment Guide 11
List of Procedures
Configuring a treatment service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91
Configuring remote supervision . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 116
Configuring a COS DN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
Configuring a personal greeting . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 160
Enabling presence subscription . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161
Configuring a gcti::video device . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 174
Configuring a video service . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 174
Activating Transfer Connect Service . . . . . . . . . . . . . . . . . . . . . . . . . . 286
Configuring Number Translation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 298
Activating Event Propagation: basic configuration . . . . . . . . . . . . . . . . 304
Modifying Event Propagation: advanced configuration . . . . . . . . . . . . 305
Configuring T-Server Applications . . . . . . . . . . . . . . . . . . . . . . . . . . . . 307
Configuring Default Access Codes. . . . . . . . . . . . . . . . . . . . . . . . . . . . 309
Configuring Access Codes . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 310
Configuring access resources for the route transaction type . . . . . . . . 314
Configuring access resources for the dnis-pool transaction type . . . . . 315
Configuring access resources for direct-* transaction types . . . . . . . . 316
Configuring access resources for ISCC/COF. . . . . . . . . . . . . . . . . . . . 316
Configuring access resources for non-unique ANI . . . . . . . . . . . . . . . . 317
Modifying DNs for isolated switch partitioning . . . . . . . . . . . . . . . . . . . 317
Configuring DMX manually. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 375
Starting DMX using Solution Control Interface . . . . . . . . . . . . . . . . . . . 392
Stopping DMX using Solution Control Interface . . . . . . . . . . . . . . . . . . 392
12 Framework 7.6
Preface
Welcome to the Framework 7.6 SIP Server Deployment Guide. This document introduces you to the concepts, terminology, and procedures relevant to
T-Servers ® in general and provides detailed reference information about SIP
Server.
The information includes, but is not limited to, configuration options, limitations, and switch-specific functionality. You must configure the configuration objects and options described in this document in the Framework
Configuration Layer.
Use this document only after you have read through the Framework 7.6
Deployment Guide, and the Release Note for SIP Server.
This document is valid only for the 7.6 release of this product.
Note:
For releases of this document created for other releases of this product, please visit the Genesys Technical Support website, or request the
Documentation Library DVD, which you can order by e-mail from
Genesys Order Management at [email protected]
.
z z z z z
This preface provides an overview of this document, identifies the primary audience, introduces document conventions, and lists related reference information:
Making Comments on This Document, page 19
SIP Server is the Genesys software component that provides an interface between your telephony hardware and the rest of the Genesys software components in your enterprise. It translates and keeps track of events and requests that come from, and are sent to the telephony device. SIP Server is a
TCP/IP-based server that can also act as a messaging interface between SIP
Server clients. It is the critical point in allowing your Genesys solution to facilitate and track the contacts that flow through your enterprise.
SIP Server—Deployment Guide 13
Preface Intended Audience
Intended Audience
This guide is intended primarily for system administrators, both those who are new to SIP Server and those who are familiar with it.
•
If you are new to SIP Server, read the Framework 7.6 Deployment Guide and the Release Note mentioned earlier, and then read all of the sections of this document that apply to your software and its accompanying components. Refer back to the Framework 7.6 Deployment Guide as needed.
•
If you are an experienced SIP Server user—someone with computer expertise, who is used to installing, configuring, testing, or maintaining
Genesys software—you may find it more time efficient to go to the Index to see what is new or different in SIP Server release 7.6. If you take that approach, please also read Release Notes and refer to other related resources, such as the Genesys 7 Events and Models Reference Manual.
In general, this document assumes that you have a basic understanding of, and familiarity with:
•
Network design and operation.
•
Your own network configurations.
•
Your telephony hardware and software.
•
Genesys Framework architecture and functions.
•
Configuration Manager interface and object-managing operations.
Based on your specific contact center environment and your responsibilities in it, you may need to be familiar with a much wider range of issues as you deploy T-Server.
Reading Prerequisites
You must read the Framework 7.6 Deployment Guide before using this SIP
Server Deployment Guide. That book contains information about the Genesys software you must deploy before deploying SIP Server.
Chapter Summaries
This SIP Server Deployment Guide encompasses all information—including conceptual, procedural, and reference information—about Genesys T-Servers in general, and SIP Server in particular. Depending on the subject addressed in a particular section, the document style may move from narration, to instructions, to technical reference. To distinguish between SIP Server, general
T-Server sections, and those chapters intended for DMX, this document is divided into three main parts.
14 Framework 7.6
Preface Chapter Summaries
Part One—SIP Server Deployment
Part One of this SIP Server document, “SIP Server Deployment,” consists of
through 8 . These chapters contain deployment information specific
to SIP Server.
•
Chapter 1, “SIP Server Fundamentals,” on page 25 , provides information
about SIP Server architectures and deployment considerations.
•
Chapter 2, “SIP Server General Deployment,” on page 33 , presents
configuration and installation procedures for SIP Server.
•
Chapter 3, “High-Availability Deployment,” on page 51
, describes how to implement a high-availability (HA) configuration for SIP Server.
•
Chapter 4, “Starting and Stopping SIP Server,” on page 69 , describes how,
and in what order, to start up SIP Server among other Framework components. It also provides possible stopping commands.
•
Chapter 5, “SIP Devices Support,” on page 77
, describes compatibility and configuration information specific to SIP Server, including how to set the
DN properties and recommendations for the device configuration.
•
Chapter 6, “SIP Server Feature Support,” on page 97
, describes which features SIP Supports and how to configure them.
•
Chapter 7, “T-Library Functionality Support,” on page 177
, describes
T-Library functionality that SIP Server supports, known limitations, and error messages.
•
Chapter 8, “SIP Server Configuration Options,” on page 197
, describes configuration options specific to SIP Server.
Although you must refer to Part Two if you have never before configured or installed SIP Server. You might also use the chapters in Part Two, even if you are already familiar with SIP Server, to discover any changes to functionality, configuration, and installation since you last deployed this component.
Genesys recommends that you use wizards to deploy SIP Server. If you do, first read
to familiarize yourself with SIP Server general deployment, and then proceed with the deployment process using Framework wizards.
Part Two—T-Server Common Functions and
Procedures
Part Two of this SIP Server document, “T-Server Common Functions and
Procedures,” consists of
through 12 . These chapters contain
architectural, functional, and procedural information common to all T-Servers.
They also contain information about configuration options that are common to all T-Servers.
SIP Server—Deployment Guide 15
Preface Document Conventions
•
Chapter 9, “T-Server Fundamentals,” on page 255 , describes T-Server, its
place in the Framework 7 architecture, T-Server redundancy, and multi-site issues. It does not, however, provide configuration and installation information.
•
Chapter 10, “Multi-Site Support,” on page 267
, describes the variations available for T-Server implementations across geographical locations.
•
Chapter 11, “Common Configuration Options,” on page 321
, describes log configuration options common to all Genesys server applications.
•
Chapter 12, “T-Server Common Configuration Options,” on page 343
, describes configuration options common to all T-Server types including options for multi-site configuration.
Part Three—DMX Reference Information
Part Three of this SIP Server document, “DMX Reference Information,”
14 . These chapters contain reference information
specific to DMX.
•
Chapter 13, “DMX Deployment,” on page 369
, describes how to configure, install, and start DMX.
•
Chapter 14, “DMX Reference,” on page 393 , provides detailed information
about DMX.
Appendix
•
The appendix “Session Initiation Protocol (SIP) Overview” on page 401 ,
provides basic information about the SIP protocol.
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:
76fr_sip_06-2008_v7.6.001.00
You will need this number when you are talking with Genesys Technical
Support about this product.
16 Framework 7.6
Preface Document Conventions
Type Styles
Examples:
Italic
In this document, italic is used for emphasis, for documents’ titles, for definitions of (or first references to) unfamiliar terms, and for mathematical variables.
•
Please consult the Genesys Migration Guide for more information.
•
A customary and usual practice is one that is widely accepted and used within a particular industry or profession.
•
Do not use this value for this option.
•
The formula, x +1 = 7 where x stands for . . .
Examples:
Example:
Monospace Font
A monospace font, which looks like teletype or typewriter text
, is used for all programming identifiers and GUI elements.
This convention includes the names of directories, files, folders, configuration objects, paths, scripts, dialog boxes, options, fields, text and list boxes, operational modes, all buttons (including radio buttons), check boxes, commands, tabs, CTI events, and error messages; the values of options; logical arguments and command syntax; and code samples.
•
Select the
Show variables on screen
check box.
•
Click the
Summation
button.
•
In the
Properties
dialog box, enter the value for the host server in your environment.
•
In the
Operand
text box, enter your formula.
•
Click
OK
to exit the
Properties
dialog box.
•
The following table presents the complete set of error messages T-Server distributes in EventError events.
•
If you select true for the inbound-bsns-calls option, all established inbound calls on a local agent are considered business calls.
Monospace is also used for any text that users must manually enter during a configuration or installation procedure, or on a command line:
•
Enter exit
on the command line.
Screen Captures Used in This Document
Screen captures from the product GUI (graphical user interface), as used in this document, may sometimes contain a minor spelling, capitalization, or grammatical error. The text accompanying and explaining the screen captures corrects such errors except when such a correction prevents you from
SIP Server—Deployment Guide 17
Preface Related Resources installing, configuring, or successfully using the product. For example, if the name of an option contains a usage error, the name must be presented exactly as it appears in the product GUI; the error must not be corrected in any accompanying text.
Square Brackets
Square brackets indicate that a particular parameter or value is optional within a logical argument, a command, or some programming syntax. That is, the parameter’s or value’s presence is not required to resolve the argument, command, or block of code. The user decides whether to include this optional information. Here is a sample: smcp_server -host [/flags]
Angle Brackets
Angle brackets indicate a placeholder for a value that the user must specify.
This might be a DN or port number specific to your enterprise. Here is a sample: smcp_server -host
<confighost>
Related Resources
Consult these additional resources as necessary:
•
The Framework 7.6 Deployment Guide, which will help you configure, install, start, and stop Framework components.
•
The Framework 7.6 Stream Manager Deployment Guide, which will help you configure, install, start, and stop Stream Manager.
•
The Framework 7.6 Configuration Options Reference Manual, which will provide you with descriptions of configuration options for other
Framework components.
•
The Framework 7.6 Configuration Manager Help, which will help you use
Configuration Manager.
•
The Genesys Migration Guide, also on the Genesys Documentation
Library DVD, which contains a documented migration strategy from
Genesys product releases 5.x and later to all Genesys 7.x releases. Contact
Genesys Technical Support for additional information.
•
The Genesys 7 Events and Models Reference Manual, which contains the
T-Library API, information on
TEvents,
and an extensive collection of call models.
18 Framework 7.6
Preface Making Comments on This Document
•
The Genesys 7 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.
•
The Release Notes and Product Advisories for this product, which are available on the Genesys Technical Support website at http://genesyslab.com/support .
Information on supported hardware and third-party software is available on the
Genesys Technical Support website in the following documents:
•
Genesys Supported Operating Environment Reference Manual
•
Genesys Supported Media Interfaces Reference Manual
Genesys product documentation is available on the:
•
Genesys Technical Support website at http://genesyslab.com/support
.
•
Genesys Documentation Library DVD, which you can order by e-mail from Genesys Order Management at [email protected]
.
Making Comments on This Document
If you especially like or dislike anything about this document, please feel free to e-mail your comments to
.
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 information in this document only and to the way in which the information is presented. Speak to Genesys Technical
Support 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.
SIP Server—Deployment Guide 19
Preface Making Comments on This Document
20 Framework 7.6
Part
1
Part One:
SIP Server Deployment
Part One of this SIP Server Deployment Guide contains deployment information specific to your SIP Server. The information in Part One is divided into the following chapters:
•
Chapter 1, “SIP Server Fundamentals,” on page 25 , provides information
about SIP Server architectures and deployment considerations.
•
Chapter 2, “SIP Server General Deployment,” on page 33 , presents
configuration and installation procedures for SIP Server.
•
Chapter 3, “High-Availability Deployment,” on page 51
, describes how to implement a high-availability (HA) configuration for SIP Server.
•
Chapter 4, “Starting and Stopping SIP Server,” on page 69 , describes how,
and in what order, to start up T-Server among other Framework components. It also provides possible stopping commands.
•
Chapter 5, “SIP Devices Support,” on page 77
, describes compatibility and configuration information specific to SIP Server, including instructions for setting the DN properties and recommendations for the device configuration.
•
Chapter 6, “SIP Server Feature Support,” on page 97
, describes which features SIP Server supports and how to configure them.
•
Chapter 7, “T-Library Functionality Support,” on page 177
, describes the
T-Library functionality that SIP Server supports, known limitations, and error messages.
•
Chapter 8, “SIP Server Configuration Options,” on page 197
, describes configuration options specific to SIP Server.
SIP Server—Deployment Guide 21
Part One: SIP Server Deployment New in SIP Server for 7.6
New in SIP Server for 7.6
The following new features are now available in the initial 7.6 release of SIP
Server:
•
Silence Treatment in conference. SIP Server now provides a silent treatment for conference call participants when one of them places the call
on hold. See “Silence Treatment in Conference” on page 125
for details.
•
Trunk optimization for multi-site support. SIP Server now supports trunk optimization for multi-site transfers, to ensure required information
•
Providing call participant info. SIP Server can now distribute
information about call participants to logged-in agents. See “Providing
Call Participant Info” on page 164 for details.
•
Enhanced call transfer and conference functionality. SIP Server now supports additional call transfer and conference functionality, including: routing to external destination using
REFER
, consultation transfer for calls on a Routing Point, and the ability to alternate between main and
consultation calls. See “Call Transfer and Conference” on page 123
for details.
•
Enhanced instant messaging support. SIP Server now supports the following call supervision modes for instant messaging (IM) calls: Silent
Monitoring, Whisper Coaching, Open Supervisor Presence, and Intrusion.
SIP Server also supports multiple IM sessions and delivery of Instant
Messages transcripts. See “Instant Messaging” on page 143
for details.
•
Class of Service support. SIP Server now supports Class of Service. See
“Class of Service” on page 127 for details.
•
Remote Supervision support. The Remote Supervision feature enables supervisors to monitor agent calls from outside the contact center. See
“Remote Supervision” on page 115
for details.
•
Multi-Site Supervision support. The Multi-Site Supervision feature enables supervisors at a local site to monitor agents located at remote sites.
See “Multi-Site Supervision” on page 112 for details.
•
Enhanced SIP headers mapping support. SIP Server can now extract data from a
REFER
message, and map it to either the
Extension
or
UserData
attribute of T-Library event messages. See “Mapping SIP Headers and
SDP Messages” on page 149 for details
•
TSendDTMF request support. SIP Server now supports
TSendDTMF requests for devices that conform to RFC 2976. To support this functionality, a new DN-level configuration option, rfc-2976-dtmf,
has been added. This option applies to a DN of type
Trunk
or
Extension
.
22 Framework 7.6
Part One: SIP Server Deployment New in SIP Server for 7.6
•
Enhanced single-step transfer to a Routing Point support. SIP Server can now perform a single-step call transfer to a Routing Point DN when the call is silently monitored by a supervisor, or when the call is in emergency recording mode.
•
TAlternateCall request support. SIP Server now supports
TAlternateCall
requests for DNs configured with the dual-dialog-enabled configuration option set to either false
or true.
•
Blocking SIP headers support. SIP Server can now filter out specific headers during the
INVITE
message propagation. To support this functionality, a new application-level configuration option, sip-blockheaders,
has been added.
•
Trunk capacity support. The number of calls per trunk can now be controlled with new DN-level configuration options: capacity
and capacity group.
See the option descriptions on
.
•
Stream Manager overloading scenario support. SIP Server supports the load-sharing mechanism implemented in Stream Manager. In an environment with distributed Stream Managers, if a dialog is rejected solely because the rejecting Stream Manager is overloaded, SIP Server tries to re-send that dialog to another Stream Manager. See the Framework
7.6 Stream Manager Deployment Guide for details.
•
OPTIONS messages support. SIP Server now supports processing of
OPTIONS
messages.
•
DTMF tones generation support. SIP Server can now send a request to
Stream Manager to generate DTMF tones using the
TApplyTreatment request with
TreatmentType
set to
PlayApplication
.
•
Enhanced high-availability support. When operating in a high-availability environment, SIP Server can now synchronize calls that are in ringing
state.
Notes:
For a list of new features common to all T-Servers, see “New for All
of this document.
For a complete list of supported features and their configuration, see
Chapter 6, “SIP Server Feature Support,” on page 97
.
For a list of configuration option changes that apply to SIP Server, see
“Changes from 7.5 to 7.6” on page 249
.
SIP Server—Deployment Guide 23
Part One: SIP Server Deployment New in SIP Server for 7.6
24 Framework 7.6
Chapter
1
SIP Server Fundamentals
This chapter provides more in-depth information about SIP Server and contains the following sections:
SIP Server Architecture, page 26
z z z z
Redundant SIP Servers, page 30
z z
Overview
SIP Server has the same position in the Genesys Media Layer as all Genesys
T-Servers. SIP Server is a combined T-Server and a call-switching component, in which the call-switching element functions as a SIP (Session Initiation
Protocol) Back-to-Back User Agent (B2BUA). In concrete terms, this means that call switching and control is performed by Genesys—no third-party PBX or ACD system is required. A call’s audio signal and its associated data travel on a single network, which eliminates the problems associated with synchronizing separate voice and data networks. Because SIP Server supports the Internet Engineering Task Force (IETF) SIP RFC 3261 suite, it is compatible with the most popular SIP-compatible, off-the-shelf hardware or software.
SIP Server can operate with or without a third-party softswitch. Genesys SIP
Server gives the entire Genesys line of products access to SIP networks, offering a standards-based, platform-independent means of taking full advantage of the benefits of voice/data convergence.
SIP Server—Deployment Guide 25
Chapter 1: SIP Server Fundamentals SIP Server Architecture
SIP Server Architecture
Figure 1 presents a generalized architecture of the SIP Server network.
PSTN Gateway
PSTN
Customer
Endpoint
PBX Gateway
PBX
SIP Server
Genesys
Suite
IP Network
Softswitch
Customer
Endpoint
Agent
Endpoint
Figure 1: SIP Network Architecture
SIP Server provides all SIP signaling and T-Server functions. Stream Manager is an optional component that is used to play music-on-hold, music-in-queue and announcements, and to collect DTMF digits. A third-party music server is an optional component that is used as an external music source for music-on-hold. A Multipoint Conference Unit (MCU) is an optional component that is used for third-party call control (3pcc) conference calls. It is also used for silent voice monitoring, whisper coaching, intrusion monitoring, and agent-initiated call recording.
The SIP messages that SIP Server sends or receives are very similar in all configurations, but the destination to which SIP Server sends the SIP requests differs according to the deployment configuration. This mostly applies to the routing of
INVITE messages. Other messages follow the path established by
INVITE.
See Chapter 5, “SIP Devices Support,” on page 77
for full details on how to configure the elements of such a network.
SIP Server Deployment Modes
The following SIP Server deployment modes are currently supported:
•
Stand-alone mode
•
Application Server mode
26 Framework 7.6
Chapter 1: SIP Server Fundamentals SIP Server Architecture
•
Customer-side proxy mode
Stream Manager is used as an MCU in these scenarios. For more information, see the Framework 7.6 Stream Manager Deployment Guide.
Stand-alone Mode
Figure 2 illustrates SIP Server in the Stand-alone mode.
SIP
SIP
SIP Server
Agent
Endpoint
Customer
Endpoint
SIP
Agent
Endpoint
Figure 2: Stand-alone Mode
In this configuration, SIP Server sends all messages to the addresses of the customer and agent endpoints. The IP addresses in this scenario are determined by SIP Server from either of the following sources:
•
Configuration Manager.
When, for each agent DN, an IP address is configured on the DN
Annex
tab.
For example, for DN
1077
on the
Annex
tab in the
TServer
section, you set the contact
option to the
[email protected] value. This is useful for agent endpoints.
•
Lookup in the local registrar.
When the agent DN is defined with the registrar as [email protected], and its SIP endpoint has registered this SIP URI (Uniform Resource
Indicator) with the registrar as
the
INVITE
message is sent to the IP address
192.168.2.55
.
•
SIP Server resolves the name as it was dialed.
For example, if the dialed name is [email protected], the request is sent to the address somedomain.com.
Application Server Mode
In the configuration shown in
Figure 3 , SIP Server is deployed as an
Application Server behind a softswitch. This is the most common deployment of SIP Server.
SIP Server—Deployment Guide 27
Chapter 1: SIP Server Fundamentals
28
SIP Server Architecture
Customer
Endpoint
SIP
SIP
Agent
Endpoint
SIP
Softswitch
SIP
SIP Server
Agent
Endpoint
Figure 3: Application Server Mode
In the Application Server mode, SIP Server communicates with a single softswitch. SIP Server is configured to send all
INVITE
requests to the IP address of the softswitch.
In this configuration, the softswitch bypasses SIP Server for direct agent-to-agent calls. As a consequence, agent-to-agent calls are not visible to
SIP Server, and it cannot provide any control over these calls.
Customer-Side Proxy Mode
In the configuration shown in
Figure 4 , a softswitch is deployed between SIP
Server and agent endpoints, but customer endpoints communicate directly with
SIP Server.
SIP Server
SIP
Softswitch
SIP
SIP
Agent
Endpoint
Customer
Endpoint
Agent
Endpoint
Figure 4: Customer-Side Proxy Mode
All inbound calls (from customers to agents) are routed by SIP Server to a softswitch, the IP address of which is configured in Configuration Manager.
For outbound calls (from agents to customers), the IP addresses are determined from the
Request URI
message, or they are configured in Configuration
Manager as gateways (DNs of type
Trunk).
Alternatively, SIP Server can resolve the name as it was dialed. For example, if the dialed name is [email protected], the request is sent to the address somedomain.com.
Framework 7.6
Chapter 1: SIP Server Fundamentals SIP Server Architecture
In this configuration, the softswitch bypasses SIP Server for direct agent-to-agent calls. As a consequence, agent-to-agent calls are not visible to
SIP Server, and it cannot provide any control over these calls.
Media Server Deployment Architecture
Figure 5 illustrates one possible deployment architecture for a third-party
media server (such as a music server or MCU), or for Genesys Stream
Manager used in conjunction with SIP Server and Genesys business applications.
Media Server
SIP
Business
Application
SIP
T-Library
SIP
SIP Server
Customer
Endpoint
Agent
Endpoint
RTP
Figure 5: Media Server Deployment Architecture
Call Scenario
1. A call arrives and is established with a contact center agent. SIP Server operates as a SIP B2BUA and maintains two separate SIP dialogs, one for the customer and one for the agent. The RTP stream is negotiated between the customer and agent endpoints directly, using the codec. SIP Server provides flexibility with manipulation of SDP information.
2. The agent invokes a call-hold operation either by using a
THoldCall
request to SIP Server, or by pressing the
Hold
button on the endpoint.
3. SIP Server selects a media server in sequence from among all configured servers with the same priority, and then establishes a new SIP dialog to the music server. SIP Server then sends the
INVITE message to the customer session to connect the RTP stream to the music server, and then re-
INVITEs the agent session to stop RTP traffic from it.
4. When the agent invokes a call-retrieve operation either by using the
TRetrieveCall
request, or by pressing the
Retrieve
button on the endpoint,
SIP Server terminates the music dialog by sending a
BYE
message, and then re-
INVITEs
the customer session to connect the RTP stream with the agent endpoint.
SIP Server—Deployment Guide 29
Chapter 1: SIP Server Fundamentals Redundant SIP Servers
Redundant SIP Servers
SIP Servers can operate in a high-availability (HA) environment, providing you with redundant systems. One basic principle of redundant SIP Servers is the standby redundancy type, which dictates how quickly a backup SIP Server steps in when the primary SIP Server goes down. The Framework
Management Layer currently supports two types of redundant configurations: warm standby
and hot standby
.
SIP Server in an HA configuration differs from most Genesys T-Servers in the role it performs in the SIP network. It is not a switch, but it does have traditional switching capabilities.
Instructions for configuring SIP Server redundancy are available in Chapter 3,
“High-Availability Deployment,” on page 51 .
Load Balancing
Figure 6 illustrates a load-balancing architecture for situations in which the call
rate exceeds the capacity of a single SIP Server.
Customer
Endpoint
PSTN
Network SIP
Server
302 Moved
INVITE
INVITE
PSTN Gateway
SIP Server
SIP
Agent
Endpoint
SIP Server
SIP
Agent
Endpoint
SIP Server
SIP
Agent
Endpoint
Figure 6: SIP Server Sample Load-Balancing Configuration
In this configuration, all inbound calls first arrive at the Network SIP Server, which performs all initial routing. The routing on the Network SIP Server is either direct to agents, or to the second tier of Routing Points on the SIP
Server.
Routing inbound calls directly to agents assumes the following:
•
Multiple TDM-to-VoIP gateways (or incoming SIP firewalls for pure VoIP calls) are deployed to provide sufficient call capacity.
30 Framework 7.6
Chapter 1: SIP Server Fundamentals Multi-Site Support
•
Multiple SIP Servers are deployed. Each can serve up to 1000 agents.
•
One or more Network SIP Servers are deployed. Given the high throughput of a Network SIP Server, it is very likely that a single SIP
Server will be sufficient for most deployments.
•
Gateways are configured to send all calls to the Network SIP Servers. If multiple Network SIP Servers are required, you can partition gateways, so that they send calls to different SIP Servers; or you can configure each gateway to send calls to one of the Network SIP Servers, based, for example, on call origination or destination.
As a result, the following provides a generalized call flow:
1. Network SIP Servers communicate with Universal Routing Server (URS)
(not shown in Figure 6 on page 30
). URS selects an agent on one of the SIP
Servers by using real-time state information available from the SIP Servers via Stat Servers (not shown). URS responds to the Network SIP Server with the agent’s DN and the location name of the SIP Server.
2. Network SIP Server communicates the
ConnID
and attached data to the selected SIP Server. It then responds to the PSTN gateway with a
302 message containing the IP address and External Routing Point number on the destination SIP Server.
3. The gateway processes the
302
message, and then sends a new
INVITE message to the selected SIP Server.
4. SIP Server receives the
INVITE
message from the External Routing Point, matches the call, and reroutes it to the selected agent by using Inter Server
Call Control (ISCC).
Multi-Site Support
SIP Server, like any conventional T-Server, is built with the T-Server Common
Part that contains the ISCC component responsible for call data transfer between multiple sites. Currently, SIP Server supports the following ISCC transaction types: route, direct-notoken, direct-uui, pullback,
and reroute.
However,
direct-uui is supported only in a pure SIP environment.
For instructions on installing and configuring a multi-site environment,
including information on the ISCC features, please see Chapter 10, “Multi-Site
Next Steps
Now that you have gained a general understanding of the roles and features available in SIP Server, you are ready to learn how SIP Server is installed and configured. That information is presented in the next few chapters of this
Deployment Guide. So unless you are already familiar with SIP Server
SIP Server—Deployment Guide 31
Chapter 1: SIP Server Fundamentals Next Steps
deployment and operation procedures, continue with Chapter 2, “SIP Server
General Deployment,” on page 33 . Otherwise, you may want to proceed to
Chapter 6, “SIP Server Feature Support,” on page 97
, where you will find information about feature configurations that SIP Server supports.
32 Framework 7.6
Chapter
2
SIP Server General
Deployment
This chapter contains general information for the deployment, configuration, and installation of your SIP Server. You may have to complete additional configuration and installation steps specific to your SIP Server and devices.
You will find these steps in Part One of this document.
z z z z z z
This chapter contains the following sections:
Network Considerations, page 37
Wizard Deployment of SIP Server, page 41
Manual Deployment of SIP Server, page 44
Note:
You must read the Framework 7.6 Deployment Guide before proceeding with this SIP Server guide. That document contains information about the Genesys software you must deploy before deploying SIP Server.
Prerequisites
SIP Server has a number of prerequisites for deployment. Read through this section before deploying your SIP Server.
SIP Server—Deployment Guide 33
Chapter 2: SIP Server General Deployment Prerequisites
Software Requirements
Framework Components
You can only configure SIP Server after you have deployed the Configuration
Layer of Genesys Framework. This layer contains DB Server, Configuration
Server, Configuration Manager, and, at your option, Deployment Wizards. If you intend to monitor or control SIP Server through the Management Layer, you must also install and configure components of this Framework layer, such as Local Control Agent (LCA), Message Server, Solution Control Server
(SCS), and Solution Control Interface (SCI), before deploying SIP Server.
Refer to the Framework 7.6 Deployment Guide for information about, and deployment instructions for, these Framework components.
Media Layer and LCA
To monitor the status of components in the Media Layer through the
Management Layer, you must load an instance of LCA on every host running
Media Layer components. Without LCA, Management Layer cannot monitor the status of any of these components. If you do not use the Management
Layer, LCA is not required.
Supported Platforms
Refer to the Genesys Supported Operating Environment white paper for the list of operating systems and database systems supported in Genesys releases 7.x.
You can find this document on the Genesys Technical Support website at http://genesyslab.com/support/dl/retrieve/ default.asp?item=B6C52FB62DB42BB229B02755A3D92054&view=item
.
For UNIX-based (UNIX) operating systems, also review the list of patches
Genesys uses for software product builds, and upgrade your patch configuration if necessary. A description of patch configuration is linked to installation read_me.html
files for the Genesys applications that operate on
UNIX, and is available within the installation packages.
Hardware and Network Environment Requirements
Hosting
Genesys recommends that you or your IT specialist assign host computers to
Genesys software before you start Genesys installation. Remember the following restrictions:
•
Do not install all the Genesys server applications on the same host computer.
34 Framework 7.6
Chapter 2: SIP Server General Deployment Prerequisites
•
When installing a few server applications on the same host computer, prevent them (except for Configuration Server) from using the swap area.
Installation Privileges
During deployment, be sure to log in with an account that will permit you to perform administrative functions—that is, one that has root privileges.
Server Locations
Refer to the “Network Locations for Framework Components” chapter of the
Framework 7.6 Deployment Guide for recommendations on server locations.
Supported Platforms
Refer to the Genesys Supported Media Interfaces white paper for the list of supported switch and PABX versions. You can find this document on the
Genesys Technical Support website at http://genesyslab.com/support/dl/retrieve/ default.asp?item=A9CB309AF4DEB8127C5640A3C32445A7&view=item
.
Licensing Requirements
All Genesys software is licensed—that is, it is not shareware. Genesys products are protected through legal license conditions as part of your purchase contract. However, the level of technical license-control enforcement varies across different solutions and components.
Before you begin to install SIP Server, remember that, although you may not have had to use technical licenses for your software when you deployed the
Configuration and Management Layers in their basic configurations, this is not the case with the Media Layer.
SIP Server requires seat-related DN technical licenses to operate even in its most basic configuration. Without appropriate licenses, you cannot install and start SIP Server. If you have not already done so, Genesys recommends that you install License Manager and configure a license file at this point. For complete information on which products require what types of licenses, and on the installation procedure for License Manager, refer to the Genesys 7
Licensing Guide available on the Genesys Documentation Library DVD.
The sections that follow briefly describe the T-Server/SIP Server license types.
Note:
Starting with release 7.2, the licensing requirements for T-Server
(including SIP Server) have changed from previous releases. Please read this section carefully and refer to the Genesys 7 Licensing Guide for complete licensing information.
SIP Server—Deployment Guide 35
Chapter 2: SIP Server General Deployment Prerequisites
Licensing Basic Implementations
A stand-alone SIP Server serving a single site requires licenses to register all
DNs it monitors. DNs that agents use in day-to-day contact center operations, such as Extensions and ACD Positions, have to be registered using licenses that control agent seats.
Note:
Configure all seat DNs that agents use (Extensions and ACD Positions) in the Configuration Layer. This enables detailed call monitoring through Genesys reporting, and generally allows you to control access to individual DNs.
Licensing HA Implementations
SIP Servers operating with the hot standby
redundancy type require a special
CTI HA technical license, which allows for high-availability implementations, in addition to regular SIP Server licenses. Neither SIP Server in a redundant pair configured for hot standby
starts if this license is unavailable. Moreover, the primary and backup SIP Servers must use the same licenses to control the same pool of DNs. If your SIP Servers are configured with the hot standby redundancy type, order licenses for CTI HA support.
Licensing Multi-Site Implementations
SIP Servers performing multi-site operations require licenses that allow for such operations, in addition to regular SIP Server licenses. If some of your SIP
Servers are configured for multi-site routing while others are not, either order licenses for multi-site support for all SIP Servers or install an additional
License Manager to handle the SIP Servers involved in multi-site routing.
Note:
You do not need licenses for multi-site support if some SIP Server clients include the local location as the location
attribute value in their requests for routing within the same site.
Configuring License Files
You need a license to configure and install Media Layer components. Genesys recommends that, if you have not already done so, at this point you:
1. Install License Manager.
2. Configure license files.
36 Framework 7.6
Chapter 2: SIP Server General Deployment Network Considerations
Note:
If you use the
<port>@<server>
format when entering the name of the license server during installation, remember that some operating systems use
@
as a special character. In this case, the installation routine is unable to write license information for SIP Server to the
Configuration Layer or the run.sh
file. Therefore, when you use the
<port>@<server>
format, you must manually modify the command-line license parameter after installing SIP Server.
For information about which products require what types of licenses and for the installation procedure for License Manager, refer to the Genesys 7
Licensing Guide available on the Genesys Documentation Library DVD.
About Configuration Options
Configuring SIP Server is not a onetime operation. It is something you do at the time of installation and then in an ongoing way to ensure the continued optimal performance of your software. You must enter values for SIP Server configuration options on the relevant Wizard pages or on the
Options
tab of your SIP Server
Application
object in Configuration Manager. The instructions for configuring and installing SIP Server that you see here are only the most rudimentary parts of the process. You must refer extensively to the configuration options chapters located in Part One and Part Two of this book.
Pay particular attention to the configuration options specific to SIP Server.
Configuration options common to all T-Servers, independent of switch type,
are described in Chapter 12, “T-Server Common Configuration Options,” on page 343
. SIP Server-specific configuration options are described in Chapter 8,
“SIP Server Configuration Options,” on page 197 . SIP Server also supports
unified Genesys log options, as described in the Chapter 11, “Common
Configuration Options,” on page 321 .
Options that configure values for the TSCP (T-Server Common Part) software in your SIP Server are common to all T-Servers. Options based on the SIP custom features apply to your SIP Server only. Familiarize yourself with both types of options. You will want to adjust them to accommodate your production environment and the business rules that you want implemented there.
Network Considerations
This section is for system administrators, contact center operations heads, and developers who are planning to deploy Genesys SIP Server.
Deploying SIP Server is similar in many ways to deploying other components of the Genesys Framework, with the significant exception that the voice signal is carried over the data network. This has serious implications for network
SIP Server—Deployment Guide 37
Chapter 2: SIP Server General Deployment Network Considerations planning and server sizing. The primary purpose of this section is to highlight the major planning and resource concerns you face in rolling out SIP Server, and to explain how it overlaps with the underlying data network. However, this section is not intended to be an exhaustive guide to network planning. Refer to the Framework 7.6 Deployment Guide for further help with Framework rollout.
The performance of SIP Server is directly linked to that of the underlying data network. It is essential that you perform a proper network audit to ensure that the data network has been properly sized and tuned for real-time (voice) packet transport. This section discusses the factors that affect overall performance of an IP-based configuration, and provides some general rules to follow when deploying SIP Server.
Voice Quality
The following factors have a direct impact on voice quality:
•
Network latency—Overall network delay.
In a system in which packets are emitted at 20-millisecond (msec) intervals, some packets actually arrive at intervals ranging from 0 to 32 msec. To minimize network latency and ensure acceptable voice quality, you need to tune the network to prioritize real-time voice packets. There are various available schemes for prioritizing voice packets, depending on the IP router vendor.
•
Packet loss—Voice packets that are dropped for various reasons (physical media error, timeout due to network congestion, and so on).
Packet loss is a function result of several factors, including network bandwidth. As a general rule, the maximum sustained packet loss must not exceed 5 percent. Zero (0) percent packet loss is easily achievable on localarea networks (LANs) that use full-duplex Ethernet connections and this must be your goal for reliable high-quality voice operations.
•
Packet jitter—Variation in voice packet arrival times.
You can minimize packet jitter by using a jitter buffer at the endpoint device. As a general rule, you must set the buffer size to the maximum anticipated deviation from the typical interpacket emission time.
Other factors that influence voice quality include:
•
Packet misordering—Packets arrive in the wrong order (similar to packet loss).
•
Type of codec used—Codecs that do not compress the audio signal produce better voice quality but use greater bandwidth.
•
Silence suppression—Silence suppression can save bandwidth, but it can also impact voice quality.
38 Framework 7.6
Chapter 2: SIP Server General Deployment Network Considerations
Bandwidth Requirements
Determining the bandwidth requirements for the underlying data network is another critical step in achieving proper performance and voice quality.
Bandwidth requirements for a video connection are, of course, much higher.
Genesys recommends that you verify network performance and voice quality by conducting performance tests and measurements in a lab environment prior to production rollout.
For an IP/Ethernet network, two factors that affect bandwidth requirements are:
•
Codec used.
•
Protocol headers.
Genesys has found that a full-duplex voice conversation using the G.723.1 codec requires approximately 20 Kbps (kilobits per second). When estimating actual network bandwidth needs, you must also consider such factors as network efficiency and utilization.
Note:
Genesys recommends that LANs not have a throughput higher than 30 percent of the theoretical maximum (10 Mbps/100 Mbps/1 Gbps).
Exceeding the 30-percent level often leads to packet loss, because the
IP switches and routers are overloaded.
Remote Agent Configuration
SIP Server’s remote agent capabilities range from a single remote agent, to a group of remote agents in a branch office environment. The distributed nature of branch office and remote agent architectures adds to the complexity of network sizing and tuning.
Note:
Remote agent deployment behind media gateways on circuit-switching networks (such as PSTN) is not fully supported in the current release of
SIP Server. If inbound calls are answered by, or transferred to, such remote agents, the
Annex tab >
Tserver
section of the corresponding
DN
object in Configuration Manager must have the refer-enabled
option set to false.
This value limits call-transfer functionality to
Consult
Transfer
only. In addition, your media gateway must be configured as a DN of type
Trunk,
with the refer-enabled
option set to
false
.
Bandwidth and Network Tuning
Just as for local network deployment of a VOIP-based system, you must, if at all possible, allot proper bandwidth for voice communication and tune the underlying network for real-time media. Remote agents using a dial-up connection require greater bandwidth (at least 33 Kbps, with 56 Kbps
SIP Server—Deployment Guide 39
Chapter 2: SIP Server General Deployment Deployment Sequence recommended) because of the extra network overhead. This assumes use of the
G.723.1 codec, although some dial-up connections may accommodate G.729. A
Digital Subscriber Line (DSL) connection is a better alternative than a dial-up connection. The choice of remote access method is important—avoid sending voice communication over an unmanaged data network, such as the public
Internet, where voice quality cannot be guaranteed.
For a branch office environment, network bandwidth requirements depend on the number of agents. Again, wide-area network (WAN) connectivity to the corporate LAN must be tuned for real-time voice communications. You need to ensure that service-level agreements from your virtual private network
(VPN) provider give details of such requirements. End-to-end network latency must not exceed 250 msec.
Firewalls
This release of SIP Server provides no explicit support for Network Address
Translation (NAT). Genesys recommends using virtual private networks (such as PPTP) and ensuring that all agents are on the same network, without NAT translators between the agents and SIP Server.
Deployment Sequence
Genesys recommends deploying SIP Server by using the SIP Server
Configuration Wizard. However, if for some reason you must manually deploy
SIP Server, you will also find instructions for doing that in this chapter.
The recommended sequence to follow before deploying SIP Server is described below. Steps 1 through 3 apply for both Wizard-based and manual deployment. For Wizard deployment, Steps 4 and 5 take place within the
Wizard deployment process itself.
Wizard or Manual
Deployment
Manual
Deployment
1. Deploy Configuration Layer objects and ensure Configuration Manager is running (see the Framework 7.6 Deployment Guide).
2. Deploy Network objects (such as Host objects).
3. Deploy the Management Layer (see the Framework 7.6 Deployment
Guide).
When manually deploying SIP Server, you must continue with the next two steps. If you are deploying SIP Server with the Configuration Wizard, the next two steps take place within the Wizard deployment process itself, where you can create and configure all the necessary objects for SIP Server deployment.
4. Configure Telephony objects (see “Manual Configuration of Telephony
Switching Offices
Switches
40 Framework 7.6
Chapter 2: SIP Server General Deployment Wizard Deployment of SIP Server
Verifying Starting
Parameters
Agent Logins
DNs
5. Deploy the Media Layer:
SIP Server (beginning with “Manual Configuration of SIP Server” on page 46 ).
If, during the installation procedure for any of the Genesys applications, the script warns you that Configuration Server is unavailable and that the configuration cannot be updated, continue with the installation. Following the installation, you must complete the information on the
Start Info
tab to ensure that SIP Server will run.
When installation is complete, verify the information on the
Start Info
tab to
ensure that SIP Server will run. See “Verifying the manual installation of SIP
.
Wizard Deployment of SIP Server
Configuration wizards facilitate component deployment. SIP Server configuration and installation involves many steps, and Genesys strongly recommends that you set up SIP Server using the Wizard rather than manually.
The SIP Server Configuration Wizard guides you through a series of steps and options to customize your deployment of SIP Server.
Wizard Configuration of SIP Server
The first step to take for a Wizard-based configuration is to install and launch
Genesys Wizard Manager. (Refer to the Framework 7.6 Deployment Guide for instructions.) When you first launch Genesys Wizard Manager, it suggests that you set up the Management Layer and then the Framework. The Framework setup begins with configuring and creating the objects related to SIP Server, starting with the
Switch
and
Switching Office
objects, and the SIP Server’s
Application
object itself.
Note:
With the wizard, you create your SIP Server
Application
object in the course of creating your
Switch
object.
During creation of the
Switch
object, you also have an opportunity to run the
Log Wizard to set up SIP Server logging. Then, you can specify values for the most important SIP Server options. Finally, you can create contact center objects related to SIP Server, such as
DN s,
Agent Login s, and some others.
SIP Server—Deployment Guide 41
Chapter 2: SIP Server General Deployment Wizard Deployment of SIP Server
Note:
During configuration of a
Switch
object, the Wizard prompts you to copy a SIP Server installation package to an assigned computer. After that package is copied to the destination directory on the SIP Server host, complete the last steps of the SIP Server configuration. Then, install SIP Server on its host.
After you complete the Framework configuration, the Genesys Wizard
Manager screen no longer prompts you to set up the Framework. Instead, it suggests that you set up your solutions or add various contact center objects to the Framework configuration, including the
Switch, DNs
and
Places, Agent
Logins, Agent Groups, Place Groups
, and, in a multi-tenant environment, a
Tenant
. In each case, click the link for the object you wish to create. Again, you create a new SIP Server
Application
object in the course of creating a new
Switch
object.
Wizard Installation of SIP Server
After creating and configuring your SIP Server and its related components with the wizard, you can proceed to SIP Server installation. That installation process closely mimics that of previously installed components.
Note:
Certain Wizard-related procedures are not described in this document.
Refer to the Framework 7.6 Deployment Guide for general instructions.
Warning!
Genesys does not recommend installation of its components via a
Microsoft Remote Desktop connection. The installation should be performed locally.
Procedure:
Installing SIP Server on UNIX using the wizard
Start of procedure
1. In the directory to which the SIP Server installation package was copied during Wizard configuration, locate a shell script called install.sh.
2. Run this script from the command prompt by typing sh
and the file name.
For example: sh install.sh.
3. When prompted, confirm the host name of the computer on which SIP
Server is to be installed.
4. When prompted, confirm the application name of the SIP Server that is to be installed.
42 Framework 7.6
Chapter 2: SIP Server General Deployment Wizard Deployment of SIP Server
5. Specify the destination directory into which SIP Server is to be installed, with the full path to it.
6. If asked which version of the product to install, the 32-bit or the 64-bit, choose the one appropriate to your environment.
7. Specify the license information that SIP Server is to use.
8. As soon as the installation process is finished, a message appears announcing that installation was successful. The process places SIP Server in the directory with the name specified during the installation.
End of procedure
Next Steps
•
To test your configuration and installation, go to Chapter 4, “Starting and
Stopping SIP Server,” on page 69
, and try it out.
•
To configure and install redundant SIP Servers, see Chapter 3, “High-
Availability Deployment,” on page 51
.
•
To install T-Servers for a multi-site environment, proceed to Chapter 10,
“Multi-Site Support,” on page 267 .
Procedure:
Installing SIP Server on Windows using the wizard
Start of procedure
1. Open the directory to which the SIP Server installation package was copied during Wizard configuration.
2. Locate and double-click
Setup.exe
to start the installation. The
Welcome screen launches.
3. When prompted, specify the connection parameters to the Configuration
Server associated with this SIP Server.
4. Identify the SIP Server
Application
object in the Configuration Layer to be used by this SIP Server.
5. Specify the license information that SIP Server is to use.
6. Specify the destination directory into which SIP Server is to be installed.
7. Click
Install
to begin the installation.
8. Click
Finish
to complete the installation.
By default, SIP Server is installed as a Genesys service (Windows
Services
) with
Automatic
startup type.
End of procedure
SIP Server—Deployment Guide 43
Chapter 2: SIP Server General Deployment Manual Deployment of SIP Server
Next Steps
•
To test your configuration and installation, go to Chapter 4, “Starting and
Stopping SIP Server,” on page 69
, and try it out.
•
To configure and install redundant SIP Servers, see Chapter 3, “High-
Availability Deployment,” on page 51
.
•
To install SIP Servers for a multi-site environment, proceed to Chapter 10,
“Multi-Site Support,” on page 267 .
Manual Deployment of SIP Server
Deploying SIP Server manually requires that you configure a number of different objects in the Configuration Layer prior to setting up your SIP Server objects and then installing SIP Server. This section describes the manual deployment process.
Manual Configuration of Telephony Objects
This section describes how to manually configure SIP Server telephony objects if you are using Configuration Manager.
Recommendations
Genesys recommends registering (configuring) only those entities you plan to use in the current configuration. The more data there is in the Configuration
Database, the longer it takes for the CTI setup to start, and the longer it will take to process configuration data. Remember that adding configuration objects to the Genesys Configuration Database does not cause any interruption in contact center operation.
Depending on how much work is required to manually configure all applications and objects, consider registering more
Person
objects first, with a set of privileges that lets them perform configuration tasks.
Switching Offices
Your telephony network may contain many switching offices, but you should only configure those that are involved with customer interactions.
Using Configuration Manager, be sure to register a
Switching Office object of type
SIP Switch
that accommodates your
Switch
object under
Environment.
Until you have done this, you cannot register a
Switch
object under
Resources
(single-tenant environment) or a
Tenant
(multi-tenant environment).
Note:
The value for the switching office name must not have spaces in it.
44 Framework 7.6
Chapter 2: SIP Server General Deployment Manual Deployment of SIP Server
Switches
1. Configure a
Switch
object for each switch on your telephony network.
Assign each
Switch
object to the appropriate SIP Server
Application object.
2. If implementing the multi-site configuration, specify access codes for all switches on the network so that the call-processing applications can route and transfer calls between switches.
Two types of access codes exist in a Genesys configuration:
Default access codes that specify how to reach this switch from any other switch in the Genesys environment.
Switch-to-switch access codes that specify how to reach a particular switch from any other switch. Use this type when either a nondefault dial number or routing type is required between any two locations.
When a switch-to-switch access code is configured, its value has a higher priority than that of a default access code.
See Chapter 10, “Multi-Site Support,” on page 267 , for step-by-step
instructions.
Note:
When the numbering plan uses unique directory number (DN) assignment across sites and multi-site routing is not used, you do not have to configure access codes.
DNs and Agent Logins
For each SIP Server for which you are configuring DNs, you must configure all DNs that agents and their supervisors use in day-to-day contact center operation—so-called seat-related DNs—such as Extensions and ACD
Positions. Otherwise, SIP Server does not register such DNs.
1. To configure telephony objects within each switch, consult the switch documentation. For configuration information specific to your SIP devices,
see Chapter 5, “SIP Devices Support,” on page 77
.
2. Check the numbering plan for different types of DNs, to see if you can save time by registering Ranges of DNs. Usually, DNs of the same type have consecutive numbers, which will make an otherwise tedious configuration task easy.
Agent Login
objects almost always have consecutive numbers, which means you can register them through the
Range of Agent Logins feature as well.
3. If you plan to use Virtual Queues and Virtual Routing Points in the contact center operation, Genesys recommends registering them after you have outlined the call-processing algorithms and identified your reporting needs.
SIP Server—Deployment Guide 45
Chapter 2: SIP Server General Deployment Manual Deployment of SIP Server
Note:
Remember that CTI applications, not the switch, generate telephony events for DNs of these types.
Multi-Site Operations
See the section, “Configuring Multi-Site Support” on page 307 , for
information on setting up DNs for multi-site operations.
Manual Configuration of SIP Server
Note:
Use the Framework 7.6 Deployment Guide to prepare accurate configuration information. You may also want to consult Configuration
Manager Help, which contains detailed information about configuring objects.
Recommendations
Genesys recommends using an Application Template when you are configuring your SIP Server application. The Application Template for SIP
Server contains the most important configuration options set to the values recommended for the majority of environments. When modifying configuration options for your SIP Server application later in the process, you can change the values inherited from the template rather than create all the options by yourself.
Procedure:
Configuring SIP Server manually
Start of procedure
1. Follow the standard procedure for configuring all
Application
objects to begin configuring your SIP Server
Application
object. Refer to the
Framework 7.6 Deployment Guide for instructions.
2. In a multi-tenant environment, specify the
Tenant
to which this SIP Server belongs on the
General
tab of the
Properties
dialog box.
3. On the
Connections
tab, add all Genesys applications to which SIP Server must connect.
Note:
For multi-site deployments, you should also specify SIP Server connections on the
Connections tab for any SIP Servers that may transfer calls directly to each other.
46 Framework 7.6
Chapter 2: SIP Server General Deployment Manual Deployment of SIP Server
4. On the
Options
tab, specify values for configuration options as appropriate for your environment.
Note:
For SIP Server option descriptions, see Chapter 8, “SIP Server
Configuration Options,” on page 197 . The configuration options
common to all T-Servers are described in the Chapter 12,
“T-Server Common Configuration Options,” on page 343
chapter.
SIP Server also uses common Genesys log options, described in
the Chapter 11, “Common Configuration Options,” on page 321
.
5. In a multi-site environment, you must complete additional SIP Server
configuration steps to support multi-site operations; see Chapter 10,
“Multi-Site Support,” on page 267 .
End of procedure
Next Steps
•
See “Manual Installation of SIP Server” on page 47 .
Manual Installation of SIP Server
The following directories on the Genesys 7.6 SIP Server product CD contain
SIP Server installation packages:
•
SIP_Server/<component>/<platform>
for UNIX installations, where
<component>
is SIP Server or DMX, and
<platform>
is your operating system.
•
SIP_Server\<component>\windows
for Windows installations, where
<component>
is SIP Server or DMX.
Procedure:
Installing SIP Server on UNIX manually
Note:
During installation on UNIX, all files are copied into the directory you specify. No additional directories are created within this directory.
Therefore, do not install different products into the same directory.
Start of procedure
1. In the directory to which the SIP Server installation package was copied, locate a shell script called install.sh.
2. Run this script from the command prompt by typing sh
and the file name.
For example: sh install.sh.
SIP Server—Deployment Guide 47
Chapter 2: SIP Server General Deployment Manual Deployment of SIP Server
3. When prompted, confirm the host name of the computer on which SIP
Server is to be installed.
4. When prompted, specify the host and port of Configuration Server.
5. When prompted, enter the user name and password to access Configuration
Server.
6. When prompted, select the SIP Server application you configured in
“Configuring SIP Server manually” on page 46
from the list of applications.
7. Specify the destination directory into which SIP Server is to be installed, with the full path to it.
8. If asked which version of the product to install, the 32-bit or the 64-bit, choose the one appropriate to your environment.
9. Specify the license information that SIP Server is to use: either the full path to, and the name of, the license file, or the license server parameters.
10. As soon as the installation process is finished, a message appears announcing that installation was successful. The process places SIP Server in the directory with the name specified during the installation.
End of procedure
Next Steps
•
To verify manual installation, go to “Verifying the manual installation of
.
•
To test your configuration and installation, go to Chapter 4, “Starting and
Stopping SIP Server,” on page 69
, and try it out.
•
To configure and install redundant SIP Servers, see Chapter 3, “High-
Availability Deployment,” on page 51
.
•
To install SIP Servers for a multi-site environment, proceed to Chapter 10,
“Multi-Site Support,” on page 267 .
Procedure:
Installing SIP Server on Windows manually
Start of procedure
1. In the directory to which the SIP Server installation package was copied, locate and double-click
Setup.exe
to start the installation.
2. When prompted, specify the connection parameters to the Configuration
Server associated with this SIP Server.
48 Framework 7.6
Chapter 2: SIP Server General Deployment Manual Deployment of SIP Server
3. When prompted, select the SIP Server
Application
object you configured
in “Configuring SIP Server manually” on page 46 from the list of
applications.
4. Specify the license information that SIP Server is to use: either the full path to, and the name of, the license file, or the license server parameters.
5. Specify the destination directory into which SIP Server is to be installed.
6. Click
Install
to begin the installation.
7. Click
Finish
to complete the installation.
By default, SIP Server is installed as a Genesys service (Windows
Services
) with
Automatic
startup type.
End of procedure
Next Steps
•
To verify manual installation, go to “Verifying the manual installation of
.
•
To test your configuration and installation, go to Chapter 4, “Starting and
Stopping SIP Server,” on page 69
, and try it out.
•
To configure and install redundant T-Servers, see Chapter 3, “High-
Availability Deployment,” on page 51
.
•
To install SIP Servers for a multi-site environment, proceed to Chapter 10,
“Multi-Site Support,” on page 267 .
Procedure:
Verifying the manual installation of SIP Server
Purpose: To verify the completeness of the manual installation of SIP Server to ensure that SIP Server will run.
Prerequisites
•
Installing SIP Server on UNIX manually, page 47
•
Installing SIP Server on Windows manually, page 48
Start of procedure
1. Open the
Properties
dialog box for a corresponding
Application
object in
Configuration Manager.
2. Verify that the
State Enabled
check box on the
General
tab is selected.
3. Verify that the
Working Directory, command-line,
and
Command-Line
Arguments
are specified correctly on the
Start Info
tab.
SIP Server—Deployment Guide 49
Chapter 2: SIP Server General Deployment
4. Click
Apply
and
OK
to save any configuration updates.
End of procedure
Next Steps
Next Steps
At this point, you have either used the wizard to configure and install SIP
Server, or you have done it manually, using Configuration Manager. In either
case, if you want to test your configuration and installation, go to Chapter 4,
“Starting and Stopping SIP Server,” on page 69 , and try it out. Otherwise, if
you want to configure and install redundant SIP Servers, see Chapter 3, “High-
Availability Deployment,” on page 51
. If you want to install SIP Server for a
multi-site environment, proceed to Chapter 10, “Multi-Site Support,” on page 267
.
50 Framework 7.6
Chapter
3
High-Availability
Deployment
This chapter describes the general steps for setting up a high-availability (HA) environment for SIP Server. The high-availability architecture implies the existence of redundant applications, a primary and a backup. These are monitored by a management application so that, if one application fails, the other can take over its operations without any significant loss of contact center data. z z z z z z z
The Framework Management Layer currently supports two types of redundant configurations: warm standby
and hot standby
. This chapter describes the redundant architecture and how to configure SIP Server so that it operates with either type. This chapter contains the following sections:
Warm Standby Configuration, page 55
Hot Standby Configuration, page 57
Warm/Hot Standby Deployment on Solaris, page 60
Warm/Hot Standby Deployment on Windows, page 62
Overview
A SIP Server high-availability deployment utilizes the concept of a Virtual IP
address, hiding two hosts of primary and backup SIP Servers behind one
Virtual IP address. SIP endpoints and gateways are configured to send all SIP messages to SIP Server using this single Virtual IP address. The Virtual IP address is preserved during the switchover, and SIP messages are delivered only to the host with SIP Server running in primary mode. When the
Management Layer detects a SIP Server failure, it disables the SIP messages
SIP Server—Deployment Guide 51
Chapter 3: High-Availability Deployment Overview traffic to the backup SIP Server and enables it to the primary server. This method effectively hides the switchover (and switchback) from the endpoints and gateways.
Note:
The traffic switchover from one host to another may take a few seconds, during which some SIP messages could be lost and not delivered to the primary SIP Server.
The Management Layer and Configuration Layer components and T-Library clients must use a unique IP address for the host for communication to SIP
Server and Local Control Agent (LCA).
SIP Server supports two types of redundant configurations: warm standby
and hot standby using the Network Load Balancing (NLB) cluster (Windows) or the Virtual IP address (UNIX).
The configuration option sip-address
identifies the SIP interface address of
SIP Server: the Virtual IP address of the NLB cluster for Windows, and the
Virtual IP interface for UNIX. This option must be configured for both primary and backup servers, and the option setting must be the same.
Note:
If the NLB cluster (Windows) is configured to operate in unicast
mode
(the default), two network adapters are required on each host to enable communication between the cluster hosts. One adapter handles the network traffic for cluster operations using the Virtual IP address. The second adapter handles the traffic from the Management Layer and
Configuration Layer components, from T-Library clients, and from primary and backup SIP Servers communication using the unique host
IP address.
Figure 7 illustrates SIP Server HA deployment.
52 Framework 7.6
Chapter 3: High-Availability Deployment
T-Library
Host 1
IP Address
T-Library Client
T-Library
Host 2
IP Address
Primary
SIP Server
Backup
SIP Server
Overview
SIP
NLB Cluster or
Virtual IP Address
SIP
SIP
SIP Phone
SIP Phone
SIP Phone
Figure 7: SIP Server High-Availability Deployment
Warm Standby Redundancy Type
Genesys uses the expression warm standby to describe the redundancy type in which a backup server application remains initialized and ready to take over the operations of the primary server. The warm standby
redundancy type reduces to a minimum the inability to process interactions that may have originated during the time it took to detect the failure. It also eliminates the need to bring a standby server online, thereby increasing solution availability.
SIP Server supports warm standby
the same way as others T-Servers do. There is no propagation of information from a primary SIP Server about calls, devices, monitoring subscriptions, and agent states to a backup SIP Server.
Hot Standby Redundancy Type
Genesys uses the expression hot standby to describe the redundancy type in which a backup server application remains initialized, clients connect to both the primary and backup servers at startup, and the backup server data is synchronized from the primary server.
The hot standby
SIP Server deployment is similar to warm standby
. It is based on the architecture and design currently used in other Genesys T-Servers.
SIP Server—Deployment Guide 53
Chapter 3: High-Availability Deployment Prerequisites
Primary and backup SIP Servers must be deployed either using the NLB cluster (Windows) or the Virtual IP address (UNIX).
Data synchronization and existing client connections to the backup guarantee higher availability of a component. Data synchronization comprises information about calls, device states, monitoring subscriptions, and agent states.
SIP Server supports hot standby
mode for established calls, for calls in the ringing
state, and for calls that are parked on a Routing Point. Any telephony function can now be performed on all synchronized calls after a switchover.
Known Limitations
The following hot standby limitations exist in this implementation:
•
Client requests sent during the failure and switchover may be lost.
•
SIP requests sent by the endpoints during the failure and switchover may be lost.
•
SIP Server does not synchronize interactions that begin before it starts.
•
Some T-Library events might be duplicated or lost.
•
Reference IDs from client requests can be lost in events.
Prerequisites
This section presents basic requirements and recommendations for configuring and using redundant SIP Servers.
Requirements
You must install the Management Layer if you are installing redundant SIP
Server applications. In particular, install LCA on each computer that runs SIP
Server.
Warning!
Genesys strongly recommends that you install the backup and primary SIP Servers on different host computers.
Synchronization Between Redundant SIP Servers
When SIP Servers operate in a high-availability environment, the backup SIP
Server must be ready to take on the primary role when required. For this purpose, both SIP Servers must be running and must have the same information. When you configure redundant SIP Servers to operate with the hot standby
type, the primary SIP Server uses the connection to the backup to deliver synchronization updates. Genesys recommends that you enable the
54 Framework 7.6
Chapter 3: High-Availability Deployment Warm Standby Configuration
Advanced Disconnect Detection Protocol (ADDP) for this connection. Do so using the configuration options in the “Backup-Synchronization Section”
section. See Chapter 12, “T-Server Common Configuration Options,” on page 343
for option descriptions.
Note:
The option internal-registrar-persistent
true
to enable Configuration Server to propagate changes of the contact information to the backup SIP Server when an endpoint registers at SIP
Server.
Configuration Warnings
When configuring SIP Servers to support either the warm standby
or hot standby
redundancy type, remember:
•
When at least one of the two SIP Servers that operate in a redundant mode is running, do not change a redundancy type, host, or port in either SIP
Server configuration.
•
When both the primary and backup SIP Servers are running, do not remove the backup SIP Server
Application
object from the configuration.
You are responsible for the option synchronization in the configuration of the primary and backup SIP Servers; Configuration Server does not synchronize either options or their values in different SIP Server
Application
objects. That is, you must configure both SIP Servers to have the same options with the same values. If you change a value in one SIP Server configuration, you must manually change it in the other SIP Server configuration. However, the log options in the primary SIP Server can differ from those in the backup SIP
Server configuration.
Warm Standby Configuration
This section describes how to configure redundant SIP Servers to work with the warm standby
redundancy type, including details on their connections and settings.
General Order of Deployment
Wizard
Deployment
Manual
Deployment
The general guidelines for SIP Server warm standby
configuration are:
If you used wizards to configure SIP Servers and selected the warm standby redundancy type, no additional configuration is required for your SIP Servers.
If you did not use wizards to configure SIP Servers:
1. Manually configure two SIP Server
Application
objects as described in
“Manual Configuration of SIP Server” on page 46 .
SIP Server—Deployment Guide 55
Chapter 3: High-Availability Deployment Warm Standby Configuration
2. Make sure the
Switch
object is configured for the switch these SIP Servers
should serve, as described in “Manual Configuration of SIP Server” on page 46 .
3. Modify the configuration of the primary and backup SIP Servers as instructed in the following sections.
After completing the configuration steps, ensure that both SIP Servers are installed.
Manual Modification of SIP Servers for Warm Standby
Modify the configuration of both the primary and backup SIP Server
Application
objects as described in the following sections.
Note:
Starting with release 7.5, you can configure multiple ports for any application of type server
. When multiple ports are configured for a server in a warm standby
redundancy pair, the primary and backup servers must match with respect to the number of ports, their
Port IDs, and the
Listening Mode
settings.
Procedure:
Modifying the primary SIP Server configuration for warm standby
Start of procedure
1. Stop both the primary and backup SIP Servers if they are already running.
2. Open the Configuration Manager main window.
3. Open the
Properties
dialog box of the
Application
object for the SIP
Server that you want to configure as a primary server.
4. Click the
Switches
tab.
5. Ensure that it specifies the
Switch
object that this SIP Server
Application object should serve. If necessary, select the correct
Switch
object using the
Browse
button.
6. Click
Apply
to save the configuration changes.
7. Click the
Server Info
tab.
8. Specify the SIP Server
Application
object you want to use as the backup server. Use the
Browse
button next to the
Backup Server
text box to locate the backup SIP Server
Application
object.
9. Select
Warm Standby
as the
Redundancy Type
.
10. Click
Apply
to save the configuration changes.
56 Framework 7.6
Chapter 3: High-Availability Deployment Hot Standby Configuration
11. Click the
Start Info
tab.
12. Select
Auto-Restart
.
13. Click
Apply
and
OK
to save the configuration changes.
End of procedure
Next Steps
•
Modifying the backup SIP Server configuration for warm standby, page 57
Procedure:
Modifying the backup SIP Server configuration for warm standby
Start of procedure
1. Make sure the two SIP Server are not running.
2. Open the Configuration Manager main window.
3. Open the
Properties
dialog box of the
Application
object for the SIP
Server that you want to configure as a backup server.
4. Click the
Switches
tab.
5. Using the
Browse
button, select the same
Switch
object you associated with the primary SIP Server
Application
object.
6. Click
Apply
to save the configuration changes.
7. Click the
Start Info
tab.
8. Select
Auto-Restart.
9. Click
Apply
and
OK
to save the configuration changes.
End of procedure
Hot Standby Configuration
This section describes how to configure redundant SIP Servers to work with the hot standby
redundancy type, including details on their connections and settings.
SIP Server—Deployment Guide 57
Chapter 3: High-Availability Deployment Hot Standby Configuration
General Order of Deployment
Wizard
Deployment
Manual
Deployment
The general guidelines for SIP Server hot standby
configuration are:
•
If you used wizards to configure SIP Servers and selected the hot standby redundancy type, no additional configuration is required for your SIP
Servers.
•
If you did not use wizards to configure SIP Servers:
a. Manually configure two SIP Server
Applications
objects as described
in “Manual Configuration of Telephony Objects” on page 44 .
b. Make sure the
Switch
object is configured for the switch these SIP
Servers should serve, as described in “Manual Configuration of
Telephony Objects” on page 44 .
c. Modify the configuration of the primary and backup SIP Servers as instructed in the following sections.
After completing the configuration steps, ensure that both SIP Servers are installed.
Manual Modification of SIP Servers for Hot Standby
Modify the configuration of both the primary and backup SIP Server
Application
objects for hot standby
redundancy as described in the following sections.
Note:
Starting with release 7.5, you can configure multiple ports for any application of type server
. When multiple ports are configured for a server in a hot standby
redundancy pair, the primary and backup servers must match with respect to the number of ports, their
Port IDs, and the
Listening Mode
settings.
Procedure:
Modifying the primary SIP Server configuration for hot standby
Start of procedure
1. Stop both primary and backup SIP Servers if they are already running.
2. Open the Configuration Manager main window.
3. Open the
Properties
dialog box of the
Application
object for the SIP
Server that you want to configure as a primary server.
4. Click the
Switches
tab.
58 Framework 7.6
Chapter 3: High-Availability Deployment Hot Standby Configuration
5. Ensure that it specifies the
Switch
object that this SIP Server
Application object should serve. If necessary, select the correct
Switch
object using the
Browse
button.
6. Click
Apply
to save the configuration changes.
7. Click the
Server Info
tab.
8. In the
Ports
section, select the port to which the backup server will connect for HA data synchronization, and click
Edit Port
.
a. In the
Port Properties
dialog box, on the
Port Info
tab, select the
HA sync
check box.
b. Click
OK.
Note:
If the
HA sync
check box is not selected, the backup SIP Server will connect to the default port of the primary SIP Server.
9. Specify the SIP Server
Application
object you want to use as the backup server. Use the
Browse
button next to the
Backup Server
field to locate the backup SIP Server
Application
object.
10. Select
Hot Standby
as the
Redundancy Type
.
11. Click
Apply
to save the configuration changes.
12. Click the
Start Info
tab.
13. Select
Auto-Restart
.
14. Click
Apply
to save the configuration changes.
15. To enable ADDP between the primary and backup SIP Servers, click the
Options
tab. Open or create the backup-sync
section, and configure corresponding options.
Note:
For a list of options and valid values, see the “Backup-
Synchronization Section” section in Chapter 12, “T-Server
Common Configuration Options,” on page 343
.
16. Click
Apply
and
OK
to save the configuration changes.
End of procedure
Next Steps
•
Modifying the backup SIP Server configuration for hot standby, page 60
SIP Server—Deployment Guide 59
Chapter 3: High-Availability Deployment Warm/Hot Standby Deployment on Solaris
Procedure:
Modifying the backup SIP Server configuration for hot standby
Start of procedure
1. Make sure the two SIP Servers are not running.
2. Open the Configuration Manager main window.
3. Open the
Properties
dialog box of the
Application
object for the SIP
Server that you want to configure as a backup server.
4. Click the
Switches
tab.
5. Using the
Browse
button, select the same
Switch
object you associated with the primary SIP Server
Application object.
6. Click the
Start Info
tab.
7. Select
Auto-Restart
.
8. Click the
Options
tab.
9. Modify the values for all necessary configuration options. Genesys recommends that you set all configuration options for the backup SIP
Server to the same values as for the primary SIP Server; the only exceptions are the log options and the server-id
option.
10. Click
Apply
and
OK
to save the configuration changes.
End of procedure
Warm/Hot Standby Deployment on Solaris
The warm/hot standby
redundancy on Solaris is achieved by hiding two hosts, on which primary and backup servers are running, behind the same IP address.
This method effectively hides the switchover (and switchback) from the endpoints and gateways.
Unique and
Common
Interfaces
Two Sun hosts on the same network subnet (“host1” and “host2”) are used.
Each host has two logical IP interfaces assigned to a single Ethernet interface.
The first IP interface (called “unique” in this example) has the unique IP address of the host on the subnet. The second IP interface (called “common” in this example, though it is sometimes referred to as “virtual”) has an address on the same subnet and this address is shared between two hosts.
The unique interface should always be activated on each host. T-Library clients, and management and configuration software also use the unique interface of the hosts. The common interface should be activated only if SIP
Server is working in primary role on that host. Otherwise, the common
60 Framework 7.6
Chapter 3: High-Availability Deployment Warm/Hot Standby Deployment on Solaris
Alarm Reaction
Scripts
Alarm Conditions
interface should be deactivated. SIP User Agents (endpoints) are only aware of the common interface.
The address and hostname of the common interface, as well as addresses and hostnames of the unique interfaces, must be known to the DNS server.
On each host, the
/etc/hosts
file must contain the record about the common interface, in the form:
<IP_address> <common_name_of_the_host>
On each host, inside the
/etc directory, you must create the file: hostname.name_of_ethernet_interface:1
Where
name_of_ethernet_interface
is the actual name of the Ethernet interface on that machine—for example; hostname.dmfe0:1.
This file must contain the hostname of the common interface as it is known to the DNS server and is recorded inside the
/etc/hosts
file.
Management of the state of the common interface is provided by the Sun administrative command ifconfig
.
To bring up the common interface, you must issue the command: ifconfig name_of_ethernet_interface:1 up
To bring it down, you must issue the command: ifconfig name_of_ethernet_interface:1 down
This command should be wrapped by shell batch files. Each host must contain two such shell files, one to bring up the common interface, and one to bring it down.
The Configuration Layer contains four applications, which are referenced by these shell files as third-party servers.
Four corresponding Alarm Reaction Scripts, which are responsible for the start of each of the third-party servers mentioned above, should be created in order to execute these shell files:
1. Virtual IP Down on host1
2. Virtual IP Up on host1
3. Virtual IP Down on host2
4. Virtual IP Up on host2
You should also create four corresponding Alarm Conditions:
1. Primary for host1
For Warm Standby: Alarm Condition 1 is linked with Log Event
00-04562
Warm Standby (Primary) mode activated
for the server that is running on host1. It starts Alarm Reaction Scripts 2 and 3.
For Hot Standby: Alarm Condition 1 is linked with Log Event
00-04563
Hot Standby (Primary) mode activated
for the server that is running on host1. It starts Alarm Reaction Scripts 2 and 3.
2. Primary for host2
SIP Server—Deployment Guide 61
Chapter 3: High-Availability Deployment Warm/Hot Standby Deployment on Windows
For Warm Standby: Alarm Condition 2 is linked with Log Event
00-04562
Warm Standby (Primary) mode activated
for the server that is running on host2. It starts Alarm Reaction Scripts 1 and 4.
For Hot Standby: Alarm Condition 2 is linked with Log Event
00-04563
Hot Standby (Primary) mode activated
for the server that is running on host2. It starts Alarm Reaction Scripts 1 and 4.
3. Backup for host1
For Warm Standby: Alarm Condition 3 is linked with Log Event
00-04560
Warm Standby (backup) mode activated for the server that is running on host1. It starts Alarm Reaction Script 1.
For Hot Standby: Alarm Condition 3 is linked with Log Event
00-04561
Hot Standby (backup) mode activated
for the server that is running on host1. It starts Alarm Reaction Script 1.
4. Backup for host2
For Warm Standby: Alarm Condition 4 is linked with Log Event
00-04560
Warm Standby (backup) mode activated for the server that is running on host2. It starts Alarm Reaction Script 3.
For Hot Standby: Alarm Condition 4 is linked with Log Event
00-04561
Hot Standby (backup) mode activated
for the server that is running on host2. It starts Alarm Reaction Script 3.
Notes:
The
Cancel timeout parameter for all Alarm Conditions must be set to
1
in Configuration Manager.
If your host machines have more than one Ethernet interface, Genesys server clients should use the unique IP interface of the same Ethernet interface where the common IP interface is assigned.
Warm/Hot Standby Deployment on
Windows
The warm/hot standby
redundancy on Windows is achieved by using the
Network Load Balancing (NLB) technology included in Microsoft Windows
Server 2003. The unique and fully distributed architecture of NLB enables it to deliver failover protection without requiring any special hardware.
Network Load Balancing
A Network Load Balancing cluster uses the concept of a Virtual IP address.
The SIP switch and/or all endpoints are configured to send all SIP requests to
SIP Server using this single address. The cluster software delivers the requests
62 Framework 7.6
Chapter 3: High-Availability Deployment Warm/Hot Standby Deployment on Windows to only one of the servers in the cluster, and switches over to another SIP
Server if it detects a SIP Server failure.
The advantages of a Network Load Balancing solution are:
•
It requires no special configuration on the SIP switch or SIP endpoints— you need to configure only one IP address.
•
The switch or endpoints do not require any special actions during switchover.
•
Orderly primary/backup switchover can be invoked from the Genesys
Management Layer (for maintenance purposes).
•
No special hardware is required on the NLB cluster machines.
Configuration Procedures
This section provides detailed procedures for configuring the various elements required for configuring the warm/hot standby
redundancy on Windows.
Procedure:
Configuring cluster hosts
Start of procedure
1. On each cluster host, (see Figure 7 on page 53
), install Windows Server
2003 and enable the NLB feature.
Notes:
Genesys recommends that you install the Management and
Configuration Layers outside the NLB cluster. Genesys Universal
Router Servers can work outside or inside the cluster.
The NLB architecture uses the subnet switch to deliver incoming network traffic to all cluster hosts. As result the entire incoming network traffic for the Virtual IP address distributed by the subnet switch to all ports. Genesys recommends that you create a separate
Virtual LAN for all cluster adapters from the same cluster, or use a dedicated switch for cluster adapters.
2. On each cluster host, install SIP Servers and LCAs.
3. On each cluster host, configure the NLB parameters as follows:
•
Port Range
: Must include a port as configured in the
•
Protocols
:
UDP and
TCP,
if required.
•
Filtering mode
:
Multiple
(for multiple hosts).
•
Affinity
:
None
or
Single.
SIP Server—Deployment Guide 63
Chapter 3: High-Availability Deployment Warm/Hot Standby Deployment on Windows
•
Load weight
:
Equal.
A datagram from the SIP switch arrives at both hosts of the cluster, but the
NLB Driver lets traffic go to the Application Layer of only one cluster host.
For more information about how to administer Network Load Balancing technology, see the Windows 2003 Server documentation or online help.
Note:
You must configure the softswitch or endpoints to send datagrams to the virtual address of the NLB cluster—that is, to the port as it is configured in the
End of procedure
Next Steps
•
Procedure:
Creating a batch file
Prerequisites
•
Configuring cluster hosts, page 63
Start of procedure
◆
On the host where SIP Server is installed, create a batch file containing the
following commands (see “Recommended Sample Batch File” on page 65 ):
•
wlbs enable
XXXX
<cluster_name>:<ID_Primary>
•
wlbs disable
XXXX
<cluster_name>:<ID_Backup>
Where:
•
wlbs is name of the Windows utility used to control Network Load-
Balancing.
•
enable
is the command to enable traffic handling for the host on which the primary SIP Server is running.
•
XXXX
is the decimal value number of the port as it is configured in the sip-port
option.
•
cluster_name
is the Virtual IP address of the cluster.
•
ID_Primary
is the unique host ID of the host where the primary SIP
Server is currently running.
•
disable
is the command to disable traffic handling for the host on which the backup SIP Server is running.
64 Framework 7.6
Chapter 3: High-Availability Deployment Warm/Hot Standby Deployment on Windows
•
ID_Backup
is the unique host ID of the host where the backup SIP
Server is currently running.
If these commands are issued at the moment when SIP Server changes its roles, the primary SIP Server always handles the traffic.
End of procedure
Next Steps
•
Configuring primary/backup SIP Servers
Recommended Sample Batch File
This section contains a sample of the batch file you should use to monitor the state of the host as a member of the NLB cluster. This script should be configured as a third-party server on host1 of the cluster.
title Tserver and cluster monitor wlbs enable 5060 clustername:1 wlbs disable 5060 clustername:2
:alive
@sleep 1
@pulist |findstr /C:"sip_server.exe" >dump.log
@if errorlevel 1 goto ex
@wlbs query testcluster:2 |findstr /C:"Host 1 converg" >dump.log
@if not errorlevel 1 goto alive
@date /T>>monitor.log
@TIME /T>>monitor.log
@echo Host not in cluster anymore,t-server will be killed>> monitor.log
tskill sip_server* exit
:ex
@date /T>>monitor.log
@TIME /T>>monitor.log
@echo t-server is not running anymore>> monitor.log
exit
For host2 of the cluster, use the same script but, with host numbers reversed.
For a technical overview of Network Load Balancing and Windows Server
2003 clustering technologies, go to the Microsoft website.
SIP Server—Deployment Guide 65
Chapter 3: High-Availability Deployment Warm/Hot Standby Deployment on Windows
Procedure:
Configuring primary/backup SIP Servers
Prerequisites
•
Configuring cluster hosts, page 63
•
Creating a batch file, page 64
Start of procedure
1. In Configuration Manager, configure SIP Server
Application
objects in warm standby
mode (that is, in the primary/backup relationship).
2. On the
Options
tab in the
TServer
section, configure the following options:
•
Set the same value for both SIP Servers.
•
Set to the SIP interface address of SIP Server, which is the Virtual IP address of the NLB cluster for Windows. This option must be configured for both primary and backup servers, and the option setting must be the same.
3. Create a new
Application object that uses the
Third Party Server
Application Template.
4. On the
Server Info
tab of the new
Application
object, set the host to the name of the host on which SIP Server is installed. Set a valid value for the communication port. Use the default values for all other parameters on this tab.
5. On the
Start Info
tab, set the working directory to the location of the
batch file you created in “Creating a batch file” on page 64 . For the
command-line parameter, use the name of that batch file. Use the default values for all other parameters on this tab.
6. In the Configuration Database, for the host name (TCP/IP addresses) for each SIP Server, do not use the virtual TCP/IP address of the cluster, but use the actual TCP/IP address of the host on which SIP Server will be running.
End of procedure
Next Steps
•
66 Framework 7.6
Chapter 3: High-Availability Deployment Warm/Hot Standby Deployment on Windows
Procedure:
Configuring Alarm Conditions
Purpose: To create and set Alarm Conditions using the Solution Control
Interface (SCI) for both the primary and backup SIP Servers. For each SIP
Server two alarm conditions should be created.
Prerequisites
•
Configuring cluster hosts, page 63
•
Creating a batch file, page 64
•
Configuring primary/backup SIP Servers, page 66
Start of procedure
1. Create a new Alarm Condition.
2. For Warm Standby: Set the Alarm Detection to Detection Log Event
00-04562 (Warm-Standby Primary mode activated)
.
3. For Hot Standby: Set the Alarm Detection to Detection Log Event
00-04563 Hot Standby (Primary) mode activated.
4. Set the
Alarm Source
to the
Application
object corresponding to the relevant SIP Server.
5. On the
Assigning Alarm Reactions
tab, click
Add.
Right-click to create a new script.
6. On the
Alarm Reaction
tab, select
Start Specified Application.
7. For the application to start parameter, choose the Third Party Server
Application
object you created in “Configuring primary/backup SIP
8. Create a new Alarm Condition.
9. For Warm Standby: Set the Alarm Detection to Detection Log Event
00-04560 (Warm-Standby Backup mode activated).
10. For Hot Standby: Set the Alarm Detection to Detection Log Event
00-04561 Hot Standby (backup) mode activated.
11. Set the
Alarm Source
to the specific
Application
object corresponding to the relevant SIP Server.
12. On the
Assigning Alarm Reactions
tab, click
Add.
Right-click to create a new script.
13. On the
Alarm Reaction
tab, select
Stop Specified Application.
SIP Server—Deployment Guide 67
Chapter 3: High-Availability Deployment Next Steps
14. For the application to stop
parameter, choose the Third Party Server
Application
object you created in “Configuring primary/backup SIP
End of procedure
Next Steps
At this point, you have learned how to configure and install redundant SIP
Servers. Go to Chapter 4, “Starting and Stopping SIP Server,” on page 69
, to
test your configuration and installation, or continue with Chapter 10, “Multi-
Site Support,” on page 267 , for more possibilities.
68 Framework 7.6
Chapter
4
Starting and Stopping
SIP Server
z z z z z z z z
This chapter describes methods for stopping and starting SIP Server, focusing on manual startup for SIP Server. It contains the following sections:
Command-Line Parameters, page 69
Starting and Stopping with the Management Layer, page 71
Starting with Startup Files, page 72
Verifying Successful Startup, page 74
Starting and Stopping with Windows Services Manager, page 75
Command-Line Parameters
You can start and stop Framework components using the Management Layer, a startup file, a manual procedure, or the Windows Services Manager.
With all these methods, command-line parameters are usually required for a server application in addition to an executable file name.
Common command-line parameters are as follows:
-host
-port
-app
The name of the host on which Configuration Server is running.
The communication port that client applications must use to connect to Configuration Server.
The exact name of an Application object as configured in the Configuration Database.
SIP Server—Deployment Guide 69
Chapter 4: Starting and Stopping SIP Server Command-Line Parameters
-l
-V
-nco X/Y
-lmspath
The license address. Use for the server applications that check out technical licenses. Can be either of the following:
• The full path to, and the exact name of, the license file used by an application. For example,
-l /opt/mlink/ license/license.dat
.
• The host name and port of the license server, as specified in the
SERVER
line of the license file, in the port@host format. For example,
-l 7260@ctiserver.
Note: Specifying the License Manager’s host and port parameter eliminates the need to store a copy of a license file on all computers running licensed applications.
The version of a Framework component. Note that specifying this parameter does not start an application, but returns its version number instead. You can use either uppercase or lowercase.
The Nonstop Operation feature is activated;
X
exceptions occurring within
Y
seconds do not cause an application to exit. If the specified number of exceptions is exceeded within the specified number of seconds, the application exits or, if so configured, the Management Layer restarts the application. If the
-nco
parameter is not specified, the default value of
6
exceptions handled in
10
seconds applies. To disable the Nonstop Operation feature, specify
-nco 0
when starting the application.
The full path to log messages files (the common file named common.lms
and the application-specific file with the extension
*.lms
) that an application uses to generate log events. This parameter is used when the common and application-specific log message files are located in a directory other than the application’s working directory, such as when the application’s working directory differs from the directory to which the application is originally installed.
Note that if the full path to the executable file is specified in the startup command-line (for instance, c:\gcti\multiserver.exe
), the path specified for the executable file is used for locating the
*.lms
files, and the value of the lmspath
parameter is ignored.
Note:
In the command-line examples in this document, angle brackets indicate variables that must be replaced with appropriate values.
70 Framework 7.6
Chapter 4: Starting and Stopping SIP Server Starting and Stopping with the Management Layer
Starting and Stopping with the
Management Layer
Procedure:
Configuring SIP Server to start with the Management
Layer
Start of procedure
1. Open the SIP Server
Application’s Properties
dialog box.
2. Click the
Start Info
tab.
3. Specify the directory where the application is installed and/or is to run as the
Working Directory
.
4. Specify the name of the executable file as the command-line
.
5. Specify command-line parameters as the
Command
–
Line Arguments
.
The command-line parameters common to Framework server components are described on
6. When you are finished, click
Apply
.
7. Click
OK
to save your changes and exit the
Properties
dialog box.
End of procedure
Note:
Before starting an application with the Management Layer, make sure the startup parameters of the application are correctly specified in the application’s
Properties
dialog box in Configuration Manager.
After its command-line parameters are correctly specified in the
Properties dialog box, you can start and stop SIP Server from Solution Control Interface
(SCI), which is the graphical interface component of the Management Layer.
(The starting procedure for SCI is described in the Framework 7.6 Deployment
Guide.) Framework 7.6 Solution Control Interface Help provides complete instructions on starting and stopping applications.
You can also use the Management Layer to start a SIP Server that has failed.
To enable SIP Server’s auto-restart functionality, select the corresponding check box in the
Application’s Properties
dialog box.
Note that when you start (or restart) an application via the Management Layer, the application inherits environment variables from Local Control Agent
(LCA), which executes the startup command. Therefore, you must also set the environment variables required by the application for the account that runs LCA.
SIP Server—Deployment Guide 71
Chapter 4: Starting and Stopping SIP Server Starting with Startup Files
Warning!
Stopping an application via the Management Layer is not considered an application failure. Therefore, the Management
Layer does not restart applications that it has stopped unless an appropriate alarm condition and alarm reaction are configured for these applications.
Starting with Startup Files
Startup files are files with the extension run.sh
(on UNIX) or startServer.bat
(on Windows), which installation scripts create and place into the applications’ directories during the installations. These files are created for all Framework server applications except:
•
Configuration Server (primary or backup) running on Windows.
•
Backup Configuration Server running on UNIX.
•
DB Server running on Windows.
•
LCA running on either Windows or UNIX.
When using a startup file, verify that the startup parameters the installation script inserted in the startup file are correct. Use the following instructions for
UNIX and Windows to start those application for which startup files are
created. See the appropriate sections in “Starting Manually” on page 73 to
identify which applications should be running for a particular application to start.
Procedure:
Starting SIP Server on UNIX with a startup file
Start of procedure
1. Go to the directory where an application is installed.
2. Type the following command line: sh run.sh
End of procedure
72 Framework 7.6
Chapter 4: Starting and Stopping SIP Server Starting Manually
Procedure:
Starting SIP Server on Windows with a startup file
Start of procedure
To start SIP Server on Windows with a startup file, use either of these methods:
•
Go to the directory where an application is installed and double-click the startServer.bat
icon.
Or
•
From the
MS-DOS
window, go to the directory where the application is installed and type the following command-line: startServer.bat
End of procedure
Starting Manually
When starting an application manually, you must specify the startup parameters at the command prompt, whether you are starting on UNIX or
Windows. At the command prompt, command-line parameters must follow the name of the executable file. On the
Shortcut
tab of the Program
Properties dialog box, command-line parameters must also follow the name of the executable file.
The command-line parameters common to Framework server components are
If an
Application
object name, as configured in the Configuration Database, contains spaces (for example,
T-Server Nortel
), the
Application
name must be surrounded by quotation marks in the command-line:
-app “T-Server Nortel”
Before starting SIP Server, be sure that the following components are running:
•
DB Server that provides access to the Configuration Database
•
Configuration Server
•
License Manager
SIP Server—Deployment Guide 73
Chapter 4: Starting and Stopping SIP Server Verifying Successful Startup
Procedure:
Starting SIP Server on UNIX manually
Start of procedure
◆
Go to the directory where SIP Server is installed, and type the following command-line: sip_server -host <Configuration Server host>
-port <Configuration Server port> -app <SIP Server Application>
-l <license address> -nco [X]/[Y]
End of procedure
Procedure:
Starting SIP Server on Windows manually
Start of procedure
◆
Start SIP Server from either the
Start
menu or the
MS-DOS
window. If you use the
MS-DOS
window, go to the directory where SIP Server is installed, and type the following command-line parameters: sip_server.exe -host <Configuration Server host>
-port <Configuration Server port> -app <T-Server Application>
-l <license address> -nco [X]/[Y]
End of procedure
Verifying Successful Startup
After executing the startup command, you might want to check whether it was successful.
If you used the Management Layer to start SIP Server, check whether Solution
Control Interface displays
Started
or
Service Unavailable
status for the corresponding application. Refer to the “Troubleshooting” section of the
Framework 7.6 Management Layer User’s Guide if the startup command does not result in either
Started
or
Service Unavailable
status for some period of time.
If you start your SIP Server with startup files or manually, and if you have configured logging to console or a log file, check the log for messages similar to the following:
•
SIP Server log file:
Link connected
74 Framework 7.6
Chapter 4: Starting and Stopping SIP Server Stopping Manually
Stopping Manually
The following stopping procedures apply to Genesys server applications, such as DB Server, Configuration Server, Message Server, Local Control Agent,
Solution Control Server, HA Proxy, T-Server, SIP Server, and Stat Server.
Procedure:
Stopping SIP Server on UNIX manually
Start of procedure
To stop a server application from its console window on UNIX, use either of these commands:
•
Ctrl+C
•
kill <process number>
End of procedure
Procedure:
Stopping SIP Server on Windows manually
Start of procedure
To stop a server application on Windows, use either of these commands:
•
To stop a server application from its console window on Windows, use the
Ctrl+C
command.
•
To stop a server application on Windows, use the
End Task
button on the
Windows
Task Manager
.
End of procedure
Starting and Stopping with Windows
Services Manager
When starting an application installed as a Windows Service, make sure the startup parameters of the application are correctly specified in the
ImagePath
in the
Application
folder in the
Registry Editor
. The
ImagePath
must have the following value data:
<full path>\<executable file name> -service <Application Name as
Service> -host <Configuration Server host>
SIP Server—Deployment Guide 75
Chapter 4: Starting and Stopping SIP Server Next Steps
-port <Configuration Server port> -app <Application Name>
-l <license address> where the command-line parameters common to Framework server
components are described on page 69 and
-service
The name of the Application running as a Windows Service; typically, it matches the Application name specified in the
-app
command-line parameter.
Framework components installed as Windows Services with the autostart capability are automatically started each time a computer on which they are installed is rebooted.
You can start Framework components installed as Windows Services with the manual start capability with the
Start
button in
Services Manager.
Note:
Use the
Windows Services
window to change the startup mode from
Automatic
to
Manual
and vice versa.
Regardless of a component’s start capability, you can stop Framework components installed as Windows Services with the
Stop
button in
Services
Manager
.
Next Steps
This chapter concludes SIP Server general deployment. Refer to subsequent chapters in this guide for detailed reference information and any special procedural instructions that pertain to SIP Server.
76 Framework 7.6
Chapter
5
SIP Devices Support
This chapter presents reference information for configuring devices and the switch elements of SIP Server. It contains the following sections:
Configuring Devices and Services, page 78
z z z
Configuring Out-of-Service Detection, page 92
Configuring Agent Logins, page 94
Configuring Stream Manager, page 94
z z
Overview
SIP devices that represent SIP endpoints are configured in Configuration
Manager as the following types of DN:
•
Extension
(or
ACD Position)
—An agent’s endpoint (SIP Phone)
•
Trunk
—Any external number (for example, a gateway access number)
•
Voice over IP Service
—SIP services (Stream Manager, Music-On-Hold server, and so on)
•
Routing Point
—Used internally by SIP Server
•
ACD Queue
—Used internally by SIP Server
Note:
DNs of type
External Routing Point
are also supported by SIP Server.
They are not specific to SIP Server and are used by the T-Server
Common Part component of SIP Server in a multi-site environment.
Table 1 contains cross-reference information on SIP devices and Genesys DN
types. Use this information to configure SIP devices properly in the
Configuration Layer.
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Table 1: Device Type Cross Reference
SIP Device Type
Endpoints (SIP phones)
Routing Points
Genesys DN Type
Extension
(or
ACD Position)
Routing Point
Routing Queue
ACD Queue
Trunk
Gateway
SIP Proxy
SIP Server in a multi-site deployment
Voice Mail Service (Asterisk only)
MCU
Softswitch
Music servers
Treatment service
Recording service
Voice over IP Service, with service type
set to mcu
Voice over IP Service, with service-type set to softswitch
Voice over IP Service, with service-type
set to music
Voice over IP Service, with service-type
set to treatment
Voice over IP Service, with service-type
set to recorder
Application service
Voice over IP Service, with service-type
set to application
Configuring Devices and Services
This section describes how to configure the SIP device types for SIP Server environments. It contains the following sections:
•
“Configuring ACD Queues” on page 79
•
•
“Configuring Endpoints” on page 81
•
“Configuring Gateways” on page 83
•
“Configuring Music Servers” on page 85
•
“Configuring Routing Points” on page 87
•
“Configuring Softswitches” on page 88
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•
“Configuring an Application Service” on page 89
•
“Configuring a Recording Service” on page 90
•
“Configuring a Treatment Service” on page 91
Configuring ACD Queues
To configure ACD Queues, follow the provided procedure using Configuration
Manager.
Procedure:
Configuring an ACD Queue
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the number of the configured DN. This value must be a dialable number on the switch. You must not use the @ symbol or a computer name when configuring this property.
b.
Type:
Select
ACD Queue from the drop-down box.
3. When you are finished, click
Apply
.
End of procedure
Configuring MCUs
To configure a Multipoint Conference Unit (MCU), follow the provided procedure using Configuration Manager.
Procedure:
Configuring an MCU
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
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2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the MCU name. This name is used during SIP registration only if the MCU registers with the SIP registrar. If the
MCU does not register with the registrar, enter a short description of
MCU for this property
.
b.
Type:
Select
Voice over IP Service from the drop-down box.
3. Click the
Annex
tab.
4. Create a section named
TServer.
In the
TServer
section, create options as specified in
Table 2: Configuring an MCU
Option Name Option Value Description
SIP URI
0
0
A string
0
–
300
–
30
–
86400 seconds
Specifies the contact URI (Uniform
Resource Indicator) that SIP Server uses for communication with the MCU. See the URI format and option description on
(Optional) Specifies how often (in seconds)
SIP Server checks a device for out-of-service status.
See the option description on
(Optional) Specifies the time interval (in seconds) that SIP Server waits before placing a device that does not respond in out-of-service
state when the oos-check option is enabled.
See the option description on page 236 .
(Optional) Specifies the starting digits of the number that are used when sending calls to MCU.
See the option description on
(Optional) Specifies whether an MCU is taken out of service when an error is encountered, and for how long it is out of service.
See the option description on
mcu
Set this option to mcu.
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5. When you are finished, click
Apply
.
End of procedure
You can configure multiple MCUs. In this case, SIP Server distributes the load for all MCUs in a round-robin fashion.
Configuring Endpoints
To configure SIP endpoints, follow the provided procedure using
Configuration Manager.
Note:
In order to update the
DN
object, SIP Server must have
Full Control permission for it. By default, it does not have this permission. You must grant the
System
account
Full Control permission by changing the
Permissions
on the
DN s folder object in Configuration Manager.
Procedure:
Configuring endpoints
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the username
part of the endpoint’s Address of Record
(AOR) as an alphanumeric string. You must not use the @ symbol or a computer name when configuring this property.
b.
Type:
Select
Extension
(or
ACD Position
) from the drop-down box.
3. Click the
Annex
tab.
4. Create a section named
TServer.
In the
TServer
section, create options as specified in
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Table 3: Configuring Endpoints
Option Name Option Values Description
register, invite
Specifies whether incoming SIP requests are treated with an authentication procedure under the following conditions:
1. The name of the incoming SIP message exists in the list of the authenticate-requests
parameter.
2. The option password
is configured on the same
DN
object.
See the option description on page 228
.
SIP URI
true, false
(Optional, depends on the phone registration) Specifies the contact URI that
SIP Server uses for communication with the endpoint. See the URI format and
option description on page 230
.
Set the value to false
for endpoints that accept only one active SIP dialog, or cannot provide remote CTI control by the
NOTIFY
message to answer, hold, or retrieve call operations. Set the value to false for Siemens optiPoint phones that are used in re-
INVITE
mode for third-party call control (3pcc) operations.
See the option description on page 232
.
1, 2
A string
Specifies which SIP call flow will be used when a call is initiated by the
TMakeCall request. Only flow 1 and flow 2 from RFC
3725 are currently supported.
See the option description on page 233
.
Specifies the password for SIP endpoint registration with the local registrar. If it is present, registration attempts are challenged, and the password is verified. If it is not present, the registration is not challenged.
The realm for password authentication is configured globally; there is one realm per
SIP Server.
See the option description on page 238
.
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Table 3: Configuring Endpoints (Continued)
Option Name Option Values Description
true, false
true, false
Specifies whether the
REFER method is sent to an endpoint. The recommended setting is true
.
See the option description on page 240
.
(Optional, for Genesys SIP Endpoints only) Specifies whether the endpoint is placed on hold by re-inviting it with the hold SDP.
See the option description on page 240
.
SIP URI talk, hold
Specifies the value of the
Request-URI address to be used in the
INVITE
message, if that address is different from the address where the message will be sent.
See the option description on page 241
.
Specifies the behavior of the DN representing the SIP endpoint that supports the BroadSoft SIP Extension Event
Package.
See the option description on page 244
.
5. When you are finished, click
Apply
.
End of procedure
Configuring Gateways
To configure gateways, follow the provided procedure using Configuration
Manager.
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Chapter 5: SIP Devices Support Configuring Devices and Services
Procedure:
Configuring a gateway
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the gateway name. This name is used only during SIP registration when the gateway registers with the SIP registrar. If the gateway does not register with the registrar, enter a short description of the gateway for this property.
b.
Type:
Select
Trunk
from the drop-down box.
3. Click the
Annex
tab.
4. Create a section named
TServer.
In the
TServer
section, create options as specified in
Table 4: Configuring a Gateway
Option Name Option Value Description
SIP URI Specifies the contact URI that SIP Server uses for communication with the gateway.
See the URI format and option description
.
0
–
300
0
–
30
A string
(Optional) Specifies how often (in seconds) SIP Server checks a device for out-of-service status.
See the option description on page 236
.
(Optional) Specifies the time interval (in seconds) that SIP Server waits before placing a device that does not respond in out-of-service
state when the oos-check option is enabled.
See the option description on page 236
.
(Optional) Specifies the password for gateway registration with the local registrar. This is used for incoming
REGISTER
requests, not for outgoing
INVITE requests.
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Table 4: Configuring a Gateway (Continued)
Option Name Option Value Description
A string
Any nonnegative integer
(Optional) Contains the initial digits of the number that must match a particular gateway for that gateway to be selected. If multiple gateways match the prefix, the gateway with the longest prefix that matches is selected.
(Optional) Specifies a gateway priority when deciding a route—a smaller number designates higher priority. If more than one gateway with the same prefix is selected, the gateway with highest priority is normally selected. This option is used to control primary-backup gateway switchover during a failure, and to provide lowest-cost routing.
true, false
Specifies whether the
REFER method is sent to an endpoint. The recommended setting is true
.
See the option description on page 240
.
0
–
86400 seconds
A digit string
(Optional) Specifies whether a gateway is taken out of service when an error is encountered, and for how long it is out-ofservice.
See the option description on page 239
.
(Optional) Specifies the digits that are inserted in the DN instead of the prefix for the gateway. If this annex is empty or absent, the number is not modified.
See the option description on page 241
.
5. When you are finished, click
Apply
.
End of procedure
Configuring Music Servers
To configure external music servers, follow the provided procedure using
Configuration Manager.
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Chapter 5: SIP Devices Support Configuring Devices and Services
Procedure:
Configuring a Music Server
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the Music Server name. This name is used only during
SIP registration if the Music Server registers with the SIP registrar. If the music server does not register with the registrar, enter a short description of the Music Server for this property.
b.
Type:
Select
Voice over IP Service
from the drop-down box.
3. Click the
Annex
tab.
4. Create a section named
TServer.
In the
TServer
section, create options as specified in
Table 5: Configuring a Music Server
Option Name
Option Values Description
SIP URI Specifies the contact URI that SIP
Server uses for communication with the music server. See the URI format and option description on
0
–
300
0
–
30
(Optional) Specifies how often (in seconds) SIP Server checks a device for out-of-service status.
See the option description on page 236
.
(Optional) Specifies the time interval
(in seconds) that SIP Server waits before placing a device that does not respond in out-of-service
state when the oos-check
option is enabled.
See the option description on page 236
.
0
–
86400 seconds (Optional) Specifies whether a music server is taken out of service when an error is encountered, and for how long it is out of service.
See the option description on page 239
.
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Table 5: Configuring a Music Server (Continued)
Option Name Option Values Description
SIP URI
music or moh
5. When you are finished, click
Apply
.
Specifies the value of the
Request-URI address to be used in the
INVITE message, if that address is different from the address where the message will be sent.
See the option description on page 241
.
Set this option to music
or moh
.
End of procedure
Configuring Routing Points
To configure routing points, follow the provided procedure using
Configuration Manager.
Procedure:
Configuring Routing Points
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the numeric-only DN number that is easily dialed directly from a phone keypad. You must not use the @ symbol or a computer name when configuring this property.
b.
Type:
Select either
Routing Point
or
Routing Queue
from the drop-down box.
3. When you are finished, click
Apply
.
End of procedure
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Configuring Softswitches
If you deploy proxies or softswitches between SIP Server and any internal DNs or agent endpoints, configure the proxies or softswitches using the provided procedure in Configuration Manager.
Procedure:
Configuring softswitches
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the softswitch server name. This name is currently not used for any messaging, but it must still be unique. Enter a short description for this property.
b.
Type:
Select
Voice over IP Service
from the drop-down box.
3. Click the
Annex
tab.
4. Create a section named
TServer.
In the
TServer
section, create options as specified in
Table 6: Configuring Softswitches
Option Name Option Value Description
SIP URI Specifies the contact URI that SIP Server uses for communication with the softswitch.
On some softswitches this is the same as the public IP address used by endpoints to contact the softswitch. However, other softswitches require a separate port. See the
URI format and option description on
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Table 6: Configuring Softswitches (Continued)
Option Name Option Value Description
An IP address Contains the public host:port
pair for a softswitch. This is the public IP address of the softswitch. SIP Server uses this address to fill the destination (
Refer-To
) address in
REFER
requests.
On some switches, this is the same as the contact
address; if this is the case, you do not need to specify this parameter.
softswitch
Set this option to softswitch.
5. When you are finished, click
Apply
.
End of procedure
Additional Information
You can configure multiple softswitches in either an active load-balancing configuration or in a primary-standby configuration. For load-balancing, define services with the same priority to each service. For the primary-standby configuration, give higher priority to the primary service entry.
Configuring an Application Service
To configure an application service, follow the provided procedure using
Configuration Manager.
Procedure:
Configuring an application service
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the treatment server name.
b.
Type:
Select
Voice over IP Service
from the drop-down box.
3. Click the
Annex
tab.
4. Create a section named
TServer.
In the
TServer
section, create options as specified in
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Chapter 5: SIP Devices Support Configuring Devices and Services
Table 7: Configuring an Application Service
Option Name Option Values Description
SIP URI application
Specifies the contact URI that SIP Server uses for communication with the treatment server. See the URI format and option description on
Set this option to application.
5. When you are finished, click
Apply
.
End of procedure
Configuring a Recording Service
To configure a recording service, follow the provided procedure using
Configuration Manager.
Procedure:
Configuring a recording service
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the recorder server name.
b.
Type:
Select
Voice over IP Service
from the drop-down box.
3. Click the
Annex
tab.
4. Create a section named
TServer.
In the
TServer
section, create options as specified in
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Table 8: Configuring a Recording Service
Option Name Option Value Description
SIP URI
SIP URI
Specifies the contact URI that SIP Server uses for communication with the recorder server. See the URI format and option
.
Specifies the value of the
Request-URI address to be used in the
INVITE
message, if that address is different from the address where the message will be sent.
See the option description on
recorder
Set this option to recorder.
5. When you are finished, click
Apply
.
End of procedure
Additional Information
SIP Server can also record a file name when emergency recording is initiated
by an agent. See the emergency-recording-filename
configuration option
) for more information.
Configuring a Treatment Service
To configure a treatment service, follow the provided procedure using
Configuration Manager.
Procedure:
Configuring a treatment service
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the
File menu, select
New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, on the
General
tab, specify the following properties:
a.
Number:
Enter the treatment server name.
b.
Type:
Select
Voice over IP Service
from the drop-down box.
3. Click the
Annex
tab.
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Chapter 5: SIP Devices Support Configuring Out-of-Service Detection
4. Create a section named
TServer.
In the
TServer
section, create options as specified in
Table 9: Configuring a Treatment Service
Option Name Option Values Description
SIP URI treatment
Specifies the contact URI that SIP Server uses for communication with the treatment server. See the URI format and option description on
Set this option to treatment.
5. When you are finished, click
Apply
.
End of procedure
Configuring Out-of-Service Detection
The VOIP devices can be deployed in either of these modes:
•
Load-balancing mode (for the MCU, music server, treatment service, recording service, and application service)
•
Primary-standby mode
Load-Balancing Mode
When multiple devices are configured in this mode, SIP Server chooses a device in a round-robin fashion. When it encounters a failure of a device that has a non-zero recovery timeout, SIP Server will disable the device and only use the remaining ones.
SIP Server will re-enable a device when the following occurs:
•
A timeout configured by the recovery-timeout
•
The device entry in Configuration Manager is changed.
The recovery-timeout
configuration option enables an automatic periodic switchback attempt by SIP Server.
Primary-Standby Mode
When two or more devices of the same type are configured in this mode, SIP
Server chooses a device with a higher priority as configured by the
option to direct all traffic to it during normal operation. A failure of the primary device will move it to out-of-service status. The secondary device will then be selected on future requests.
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If active out-of-service detection is not configured, switchback to the primary device will occur when the recovery-timeout
option value has expired, despite the in-service or out-of-service status of the device. Therefore, the recovery-timeout
option can be enabled for an automatic periodic switchback attempt.
If active out-of-service detection is configured, this mechanism checks the device status after the recovery-timeout
option has expired, and will force the device back into service only when its recovery has been detected.
Active Out-of-Service Detection
SIP Server supports active out-of-service detection that can be enabled for the following types of DNs:
•
Voice over IP Service
(MCU, treatment, softswitches, and so on)
•
Trunk
To enable out-of-service detection, configure the following options in the
TServer
section of the
Annex
tab of the corresponding DN:
•
oos-check
•
oos-force
•
recovery-timeout
(see
)
The oos-check option specifies how often (in seconds) SIP Server checks a device for out-of-service status. When no response is received, and the oos-force
option is configured, SIP Server waits the specified oos-force timeout expires before placing a device that does not respond in out-of-service state. For
Voice over IP Service
DNs, SIP Server generates the following log message stating that the specified device is out of service based on active outof-service detection:
52000|STANDARD|GCTI_DEVICE_OUT_OF_SERVICE|Device [the name of the device] is out of service
The DN will be set back in service automatically when the recovery-timeout timer expires. For
Voice over IP Service
DNs, SIP Server generates the following log message stating that the specified device is back in service based on active out-of-service detection:
52001|STANDARD|GCTI_DEVICE_BACK_IN_SERVICE|Device [the name of the device] is back in service
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Notes:
Active out-of-service detection on an MCU does not lead to switching to another MCU when an out-of-service device is detected. When an out-of-service MCU is detected, it is marked as unavailable and will not be used for further conferencing until it is back in service.
All treatments are restarted on another treatment DN when SIP Server detects that the DN of type
Voice over IP Service with the service-type
option set to treatment
is out of service.
In a High-Availability Environment
When operating in a high-availability environment, the states of devices are synchronized to the backup SIP Server. As a result, in-service or out-of-service states are preserved during a switchover. After the switchover, the new primary
SIP Server will restart the oos-check, oos-force,
and recovery-timeout
timers for those devices that are out of service, and that have the corresponding options set in their configuration.
Configuring Agent Logins
SIP Server can work either with softswitches or in stand-alone mode, in which the SIP endpoint communicates directly with SIP Server. In both scenarios, you must configure the
Switch
object in the Configuration Layer. The manner in which you configure your SIP Server must reflect the properties of all the objects that your SIP Server monitors. If a client issues a
TRegisterAddress request for a DN that is not configured in Configuration Manager, SIP Server will generate an
EventError
message.
Because only SIP Server uses agent logins, they do not need to match user information on the softswitch. SIP Server manages the status of agents who use these logins, and allows these agents to log in to the SIP addresses.
Configuring Stream Manager
Stream Manager is a Genesys client application that streams media files in order to provide announcements and music to callers queued on Routing Points and ACD queues. It can also serve as a music server or as an MCU. SIP Server does not need Stream Manager for normal call handling.
You configure Stream Manager as a
DN
object of type
Voice over IP service, with the service-type
option set in the
TServer
section on the
Annex tab. The values are: music, mcu, treatment, conference, and recorder.
To use Stream Manager as a music server, it must be configured as described in
“Configuring Music Servers” on page 85
.
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Stream Manager must be able to access the audio files that SIP Server requests to play. These files are located in subdirectories of the installed Stream
Manager root directory. The files must be in the appropriate codec format, with the filename suffix corresponding to the codec type. All Stream Managers integrated with SIP Server must contain the same announcement/music files in the same directory structure. The treatment will fail if the file does not exist in the specified directory.
If you are using a gateway, you must determine the codec it uses before you determine which codec is specified in Genesys.
See the Framework 7.6 Stream Manager Deployment Guide for more information about Stream Manager.
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96 Framework 7.6
Chapter
6
SIP Server Feature Support
This chapter describes the advanced functionality that SIP Server supports. It contains the following sections:
Associating an ACD Queue with a Routing Point, page 98
Asterisk Voice Mail Integration, page 98
z z z z z z z z z z z z z z z z z z z z z z z z
Busy Lamp Field Feature, page 101
Calls Outside the Premise, page 101
Call Transfer and Conference, page 123
DTMF Tones Generation, page 137
Endpoint Service Monitoring, page 142
Mapping SIP Headers and SDP Messages, page 149
Music and Announcements, page 154
No-Answer Supervision, page 157
Presence Subscription, page 161
Preview Interactions, page 164
Providing Call Participant Info, page 164
Providing a Caller ID, page 165
Remote Server Registration, page 166
Trunk Capacity Support, page 166
Trunk Optimization for Multi-Site Transfers, page 166
Genesys Voice Platform Integration, page 175
z z
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Associating an ACD Queue with a Routing
Point
SIP Server is able to associate an ACD Queue with a Routing Point by specifying the
Routing Point
DN in the
Association
field in the
Properties dialog box of the
ACD Queue
DN object in Configuration Manager.
The call flow for this functionality is as follows:
•
Agents log into the ACD Queue.
•
An inbound call arrives at the ACD Queue and at the associated Routing
Point. The call is not auto-distributed to an agent in that ACD Queue.
•
A Universal Routing Server (URS) strategy on the Routing Point selects an available agent in the ACD Queue.
•
The call is routed to an agent's DN, which responds with a SIP Ringing message. As a result, an
EventDiverted
message is distributed against the
ACD Queue and
EventRouteUsed
and
EventDiverted
messages are distributed against the Routing Point.
•
The agent answers the call.
Notes:
The inbound call will be treated as a regular call to the ACD Queue if no URS application has registered for the Routing Point associated with the ACD Queue.
The inbound call will be treated as a regular call to the ACD Queue if a
URS application has registered for the Routing Point associated with the ACD Queue, but the routing timeout expires.
Asterisk Voice Mail Integration
SIP Server exchanges SIP messaging with an Asterisk-based Voice Mail system using a SIP trunk. voice mail boxes with Asterisk have the same names as their corresponding DNs, but you do not need to configure voice mail boxes as DNs in Configuration Manager. You can access these voice mail boxes by dialing a user-defined prefix and a DN number. It is also possible to dial a prefix and a special extension so that the call will be delivered to the Voice
Mail Main Menu.
SIP Server configuration options are used to organize the delivery of calls to the Voice Mail system when certain conditions are met. For example, when an incoming call is not answered within a specified timeout. Universal Routing
Server strategies are used to organize more flexible control over delivery of the call to a voice mail box.
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Chapter 6: SIP Server Feature Support Asterisk Voice Mail Integration
Note:
Voice mail for DNs is handled by an Asterisk softswitch when using
Asterisk as a softswitch with SIP Server. You do not need to configure voice mail and Message Waiting Indicator (MWI) in SIP Server for any DNs. However, voice mail is configured for agents and Agent
Groups in this scenario.
Genesys Voice Mail Configuration Adapter for Asterisk
The Genesys Voice Mail Configuration Adapter for Asterisk (GVMA) is a utility that allows you to synchronize the Asterisk Voice Mail configuration file ( voicemail.conf
) with data from the Configuration Server. By default,
GVMA creates a voice mail box in Asterisk for each extension DN configured in Configuration Manager. GVMA can also be configured to use DNs of other types for the same purpose.
GVMA obtains the necessary Asterisk Voice Mail configuration properties from the following objects in Configuration Manager:
•
DNs are used to configure voice mail boxes for extensions.
•
Agent Logins are used to configure voice mail boxes for agents.
•
Agent Groups are used to configure of voice mail boxes for agent groups.
When it has retrieved this information, GVMA performs the following tasks:
1. Connects to Configuration Server.
2. Makes a backup copy of the current Asterisk configuration.
3. Loads DNs from Configuration Server.
4. Updates the Asterisk Voice Mail configuration file based on the retrieved data.
5. Instructs Asterisk to reload configuration files.
GVMA is run manually or scheduled to run periodically using your operating system’s scheduling tools.
Message Waiting Indicator Functionality
SIP Server supports Message Waiting Indicator (MWI) notifications by registering with Asterisk and processing MWI notifications sent by Asterisk within
NOTIFY
messages. SIP Server sends to a DN an
EventUserEvent
message with detailed information about the message waiting status attached in the
UserData
fields.
MWI Processing for Extension Voice Mail Boxes
SIP Server registers (using the
REGISTER
request) all extensions as voice mail boxes with Asterisk when the option mwi-extension-enable
) has
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. Asterisk sends
NOTIFY
SIP messages to SIP Server that contain
MWI information. SIP Server stores MWI information received from Asterisk, and distributes an
EventUserEvent
message with the latest MWI information to each registered extension voice mail box.
MWI Processing for Agent Voice Mail Boxes
SIP Server registers each agent voice mail box with Asterisk immediately after the
EventAgentLogin
message is received and when the option mwi-agent-enable
) has a value of true
. SIP Server stores MWI information received from Asterisk for each agent voice mail box and distributes an
EventUserEvent
message with the latest MWI information to the
DN that the agent has logged into. SIP Server will terminate this registration immediately after an
EventAgentLogout
message is received.
MWI Processing for Agent Group Voice Mail Boxes
SIP Server registers the agent group voice mail box with Asterisk immediately after the first member’s
EventAgentLogin
message is received and when the option mwi-group-enable
(see
) has a value of
true
. SIP Server stores
MWI information received from Asterisk for each agent group voice mail box and distributes an
EventUserEvent
message with the latest MWI information to all DNs associated with agents that are logged in and configured to use this voice mail box. SIP Server will terminate this registration immediately after an
EventAgentLogout
message is received from the last agent associated with this agent group’s voice mail box.
Note:
SIP Server associates an agent with an agent group voice mail box by using the option gvm_group_mailbox
located in the
TServer
section on the
Annex
tab of the
Agent Login
configuration object. Any agent configuration object that is configured with the same voice mail box specified in this option is associated with this agent group’s voice mail box.
Feature Configuration
MWI functionality is configured on the Asterisk side and on the SIP Server side. The following SIP Server configuration options support MWI functionality:
•
mwi-host
•
mwi-port
•
mwi-domain
)
•
mwi-extension-enable
(
•
mwi-agent-enable
)
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•
mwi-group-enable
)
•
mwi-mode
(
page 215 ) (for backward compatibility)
For Asterisk configuration and integration with SIP Server, see the Framework
7.6 SIP Server Integration Reference Manual.
Busy Lamp Field Feature
This feature enables SIP Server to interoperate with the BroadWorks softswitch and to receive notifications about BroadWorks line usage.
See the Framework 7.6 SIP Server Integration Reference Manual for more information about Busy Lamp Feature (BLF) support in SIP Server.
Calls Outside the Premise
Participants within VoIP conversations are divided into two groups:
•
Internal parties that represent agents or supervisor SIP endpoints. They communicate directly with each other by signaling directly with SIP Server and by RTP streams. In Configuration Manager, they are configured as
DNs of type
Extension.
•
External parties that represent customers or agents at remote sites. They communicate with call center devices using a media gateway or other proxy services. External parties do not have a direct representation at the
Configuration Layer, but must be represented as a
Trunk
DN or a “media gateway” that is associated with a SIP Server
Switch
object.
You need to choose a gateway at the same premise where the agent SIP endpoint is located to minimize network load for RTP traffic and for VoIP media services, such as music on hold, central mixing conferencing, or voice recording. Devices in the same premise must be configured with the same
•
An internal party has the same option value as the corresponding
Extension
DN object when a call is established.
•
An external party has the same option value as the corresponding
Trunk
DN object when a call is established.
•
A media server has the same option value as the corresponding
Voice over
IP Service
DN object when a call is established.
SIP Server determines which gateway or trunk to choose for the outbound call,
based on the settings of the find-trunk-by-location
option.
To determine the gateway for an external party of an inbound call, the IP address from the
Via
header of the incoming
INVITE
message must match with the host address of the contact
option of the
Trunk
DN. If a match is successful, the geo-location
label for the matched trunk will be used as the
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label for the external party. In order to make this match work, the contact of the corresponding trunk must be the same because it is expected to be inside the
Via
header of the incoming
INVITE
message (most likely a decimal IP address).
For other services, such as music, treatment, recorder,
or mcu,
SIP Server searches for the same geo-location
label as the party requesting such service.
Call Recording
SIP Server supports both regular call recording and emergency call recording.
Regular Call Recording
Call recording is performed by passing an RTP stream through Stream
Manager. Stream Manager acts as a media stream proxy, recording all media packets into a file. Depending on the configuration, Stream Manager may perform media mixing, or it may save the RTP packets as is, thus improving call recording performance. (See the Framework 7.6 Stream Manager
Deployment Guide for details.) Call recording is always enabled on a single call leg, such as a leg with a gateway or a leg with a SIP phone.
Call recording starts after a call becomes established. It does not result in any changes to the call itself, to event processing, or to any other generated
TEvents.
When call recording starts, SIP Server creates two new SIP dialogs with
Stream Manager. SIP Server sends re-
INVITE
requests to all call participants with the SDP from Stream Manager. As a result, the RTP stream between call participants is passed via Stream Manager.
Call recording has the highest priority compared to other operations that can be performed on the call when it is established. That is, when the
EventEstablished
message is generated on the destination DN, the operations on the call are performed in the following order:
1. Call recording, if enabled
2. Personal greeting, if enabled
3. Supervisor monitoring, if enabled
Recording is not available for consultation calls. If configuration enables recording on a DN, and a consultation call is made to this DN, recording will not start.
Recording is only available for calls with audio media. If a call contains video or IM media, recording will not start.
Only one recording is allowed on a call. If configuration enables recording on both devices in the call, recording will only start on the device that first signals
EventEstablished
.
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Feature Configuration
provides an overview of the main steps required to configure the call recording functionality.
Table 10: Task Flow—Configuring Call Recording
Objective Related Procedures and Actions
1. Configure a DN.
To enable call recording on a particular DN, in the
TServer
section on the
Annex
tab of the
DN
object in Configuration Manager, set the configuration
true.
To record all inbound calls coming for a particular media gateway, set the record
option to true
on the
Trunk
DN that represents this gateway.
2. Configure a SIP Server
Application
object.
In the
TServer
section on the
Options
tab of the
SIP Server
Application
object in Configuration
Manager, set the configuration option
to the name of the recorded file—for example: call-$ANI$-$DNIS$-$DATE$-
$TIME$-$CONNID$-$UUID$-$AGENTDN$-$AGENTID$
3. Configure an
Extension
attribute.
Specify an
Extension
attribute with key record the
TRouteCall
request. See the key values in
Table 24, “Use of the Extensions Attribute,” on page 186
in
The routing strategy will determine whether call recording is needed.
4. Configure a Stream
Manager
Application object.
Set configuration options in the Stream Manager
Application
object for precise control of how recording is performed. See the Framework 7.6
Stream Manager Deployment Guide for more information.
It is recommended to use pcap
recording mode on
Stream Manager for best performance.
5. Configure a recording service.
Configure a DN of type
Voice over IP Service with the following configuration options:
• contact:
Set to the device’s IP address used for recording.
• request-uri:
Set to the SIP URI.
• service-type:
Set to recorder
.
See “Configuring a recording service” on page 90
for details.
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Emergency (Manual) Call Recording
SIP Server performs emergency call recording when processing a single-step conference call request that specifies
AttributeOtherDN
as a
Trunk
DN specifying the gcti::record
number. When this attribute is set, SIP Server recognizes this special request and initiates call recording as follows:
•
Selects one of the available call recording units that are configured in
Configuration Manager. See “Configuring a Recording Service” on page 90 for more information.
•
Performs a single-step conference call and adds the selected call recording unit to the call.
•
Creates the file name as configured in the emergency-recording-filename option that is described on
To stop emergency call recording, the agent must issue the
TDeleteFromConference
request using the gcti::record
number.
Note:
Refer to the SIP Server 7.5.0 Call Recording White Paper for more information about call recording. This document is available from
Genesys Technical Support or Genesys Professional Services.
Feature Limitations
Emergency call recording cannot be activated on a consultation call if it has already been activated from the same DN on the primary call. Emergency call recording can only be activated on both primary and consultation calls if initiated from different DNs.
Call Supervision
Call supervision functionality is designed to enable contact center managers to monitor agents, and it also enables agents to invite their supervisors to the call when dealing with a customer.
SIP Server supports the following call supervision scenarios:
•
Standard Call Supervision—Enables supervisors to monitor agents where supervisors and agents are located on the same site.
•
Multi-Site Supervision—Enables supervisors at a local site, from an endpoint controlled by a local SIP Server, to monitor remote agents, whose
endpoints are controlled by another SIP Server. See “Multi-Site
•
Remote Supervision feature—Enables supervisors to monitor agents from outside the contact center—for example, from an off-premise cell phone.
See “Remote Supervision” on page 115
.
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Overview
Call Supervision
There are two types of call supervision that SIP Server supports:
•
Subscription monitoring enables supervisors to subscribe and monitor one agent. If the subscription is active, SIP Server automatically invites the supervisor to all calls where the agent participates. SIP Server stops working in this mode when the subscription is cancelled.
•
Assistance monitoring is activated by an agent by issuing an assistance request sent to the supervisor. The agent can issue this while he or she is on a call with a customer.
Supervision Modes
Call supervision is performed in three different modes:
•
Silent monitoring hides the supervisor’s presence from all call participants, including the monitored agent who is the target of supervisor’s attention.
•
Whisper coaching hides the supervisor’s presence from all call participants but the monitored agent. Only the agent can hear the supervisor.
•
Open supervisor presence invites the supervisor to the call through subscription or assistance call supervision scenarios, but all call participants are aware of the supervisor’s presence and can hear him or her.
The supervisor can choose any of these three modes for the call supervision subscription, but the agent can only use the last two modes for an assistance request.
Supervision Scopes
The call supervision scope specifies the time frame when the supervisor must participate in the call. There are two scopes available in SIP Server:
•
Agent scope allows the supervisor to monitor the agent. The supervisor joins the call when the call is established on the monitored agent’s DN. The supervisor leaves the call immediately after the agent leaves the call.
•
Call scope allows the supervisor to control the customer’s experience. The supervisor joins the call when the call is established on the monitored agent’s DN, or when the supervisor receives the assistance request from the agent. SIP Server keeps the supervisor as part of the call as long as either a customer or monitored agent remains in the call.
The supervisor can choose either of these scopes for the monitoring subscription.
An assistance request issued by the agent does not specify the supervision scope, so the scope always contains the call
value. Therefore, if a supervisor is invited to a call through an assistance request, he or she will stay on the call until the call is finished.
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Supervision Types
The call supervision type specifies the number of calls to be monitored—either one call or all calls.
•
If one call is chosen for the subscription, the subscription is cancelled automatically when the supervisor finishes monitoring the first call on the monitored agent.
•
If all calls is chosen for the subscription, the supervisor must cancel the subscription manually when he or she want to stop monitoring the agent’s calls.
The call supervision type cannot be specified for an assistance request. The one call type is always used when call supervision is initiated through an assistance request. The type cannot be changed through the configuration settings.
Monitoring Session
A monitoring session is the process in which a supervisor listens to an agent-customer conversation. There are two types of monitoring sessions that are defined by the session creation scenario:
•
A subscription session is created by SIP Server automatically when a call is delivered to an agent’s DN, using the existing call supervision subscription.
•
An assistance session is created as a result of the assistance request sent by an agent to a supervisor.
A monitoring session of any type must be initialized with the following three parameters when it is created:
•
Supervision type
•
Supervision mode
•
Supervision scope
These parameters in the subscription session are initialized with the values of the corresponding parameters in the subscription from which this session was derived. An assistance session uses information passed in the assistance request and includes some configuration parameters for the initialization
purpose. See “Feature Configuration” on page 107 for more information.
A monitoring session begins when a supervisor joins a call, and it ends when the supervisor disconnects from the call.
One call can have multiple monitoring sessions of both types, and all are active at the same time. Each monitoring session is uniquely identified by the supervisor involved. As a result, the supervisor can participate in only one monitoring session at a time, but one agent can be part of multiple monitoring sessions.
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The following example demonstrates how multiple monitoring sessions are created in one call:
•
Agent1 answers an incoming call, and Supervisor1 is invited to the call based on the existing subscription.
•
Agent1 sends an assistance request to Supervisor2, who also joins the call.
This call has two monitoring sessions active at the same time: the first session has a subscription type, and the second session is an assistance session.
Intrusion
Intrusion occurs when a supervisor activates a new call supervision subscription to monitor an agent who is currently on a call. SIP Server creates the requested subscription and immediately invites the supervisor to join the existing call.
Feature Configuration
This section describe how to configure call supervision. It covers the following topics:
•
•
•
Supervisor Auto-release, page 109
•
Hiding Supervisor Presence, page 111
•
Configuration Options, page 111
Subscription
Call supervision subscription is controlled by two T-Library requests:
• TMonitorNextCall
• TCancelMonitoring
The supervisor’s desktop must be able to process these two requests in order to perform call supervision.
The first request creates a new subscription, and the second request cancels the existing subscription. These requests use
AttributeThisDN
to identify the supervisor and
AttributeOtherDN
to identify the monitored agent.
Subscription Creation
SIP Server creates a new subscription based on the
TMonitorNextCall
request from the supervisor. The request is either accepted or rejected.
SIP Server rejects the request in the following scenarios:
•
The supervisor or the monitored agent already has an active subscription.
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However, if the
TMonitorNextCall
request tries to activate a monitoring subscription that is already active (for example, the supervisor who submitted this request is already set up to monitor the agent), SIP Server responds with the standard
EventMonitoringNextCall
messages sent to the agent and supervisor DNs. This request is not rejected, because it does not create multiple subscriptions on one DN.
•
The supervisor or the agent DN is not configured in Configuration
Manager.
If the request is accepted, SIP Server creates a new subscription and initializes it with the type, mode, and scope information that was defined in the request.
This information is part of the request as the following attributes:
•
AttributeMonitorNextCallType, which defines the type of call supervision. Its possible values are
MonitorOneCall
and
MonitorAllCalls
.
•
AttributeExtensions/MonitorMode,
which defines the mode of call supervision. Its possible values are
normal, mute, coach,
and connect
.
•
AttributeExtensions/MonitorScope,
which defines the scope of call supervision. Its possible values are call
and agent
.
If one or both of the monitoring extensions are missing or incorrect, the following values are used:
•
for
MonitorScope
•
MonitorMode
SIP Server confirms the new subscription for both the supervisor and the agent by sending an
EventMonitoringNextCall
message to both destinations. This event always contains
AttributeExtensions
that include both monitoring extensions. These extensions represent the monitoring configuration for a new subscription.
See “Using the Extensions Attribute” on page 186 for more information.
Note:
SIP Server identifies the agent for the call supervision by the agent DN specified in the
OtherDN
attribute of the
TMonitorNextCall
request. The agent’s login ID is not used for this purpose. In particular, this means that SIP Server does not try to identify the agent who is logged in on the monitored DN, or to analyze the agent’s state to decide if supervision should be activated for a call. SIP Server monitors calls made to or from the specified DN, regardless of the person using this
DN, until supervision scope expires (see “Supervision Scopes” on page 105
).
Subscription Cancellation
SIP Server can cancel active subscriptions using the following methods:
•
Manual, where a supervisor submits a
TCancelMonitoring
request.
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•
Automatic, where SIP Server cancels the subscription when a
MonitorOneCall
-type monitoring session is terminated.
A supervisor can submit a
TCancelMonitoring
request at any time. SIP Server identifies a subscription by the pair of supervisor and agent DNs. If this subscription exists, then it will be cancelled. Otherwise, SIP Server returns an
EventError
message.
SIP Server generates
EventMonitoringCancelled
events for both the supervisor and the agent, to inform them that the subscription was cancelled.
Assistance Request
An assistance request is a
TSingleStepConference
request containing the
AssistMode
parameter in the extensions. SIP Server creates a new monitoring session based on the assistance request, but a monitoring subscription is not created.
The
AssistMode
extension is identical to the
MonitorMode
extension used in the
TMonitorNextCall
request. The difference is that
AssistMode
can contain only the connect
and coach
values.
There are no parameters to define the scope and type of the monitoring in an assistance request, so the following monitoring parameters are used:
•
MonitorScope
set to call
•
MonitorType
set to
MonitorOneCall
These two settings are hard-coded and cannot be changed.
Supervisor Auto-release
Depending on the type of monitoring scope and mode, SIP Server determines whether to release a supervisor from the call. If the monitoring scope is agent,
SIP Server releases the supervisor from the call at the same time that the monitored agent leaves the call. If the monitoring scope is call
and the other party of the call is aware of the supervisor’s presence on the call and can hear this supervisor, SIP Server does not release the supervisor from the call.
Example 1
Call Scenarios
This section presents two-party and three-party call scenarios to demonstrate how auto-release rules work.
Three-party call,
MonitorScope
= call
:
1. A call is established with three parties: a caller, a supervisor, and Agent 1
(a monitored target of the supervisor).
2. Agent 1 transfers the call to Agent 2 (who is not monitored by the supervisor).
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Example 2
Example 3
Example 4
3. The call now has the following parties: the caller, the supervisor, and
Agent 2.
The supervisor is not released in this step, because
MonitorScope
is set to call,
and the call is not finished yet (the monitor scope has not expired).
4. The caller hangs up. Now this call contains only two parties.
5. One of the following happens:
If
MonitorMode
is set to mute
or
coach,
SIP Server will release the supervisor and the call, because the supervisor is on the call with the agent (Agent 2) who is not the monitoring target of this supervisor, and the agent is not aware of the supervisor’s presence.
If
MonitorMode
is set to connect,
SIP Server will not release the supervisor, so Agent 2 can hear the supervisor.
Three-party call,
MonitorScope
= agent
:
1. A call is established with three parties: a caller, a supervisor, Agent 1 (a monitored target of the supervisor).
2. Agent 1 transfers a call to Agent 2 (who is not monitored by a supervisor).
3. SIP Server releases the supervisor from the call. The caller and Agent 2 remain on the call.
Three-party call with recording,
MonitorScope
= call, MonitorMode=mute
:
1. A call is established with three parties and a recorder: a caller, a supervisor,
Agent 1 (a monitored target of the supervisor), and the recorder.
2. The caller hangs up. Now this call contains three parties: Agent 1, the supervisor, and the recorder.
3. SIP Server releases the supervisor and the call, because
MonitorMode
is set to mute and the agent cannot talk to the supervisor.
Three-party call with recording,
MonitorScope
= call, MonitorMode
= connect
:
1. A call is established with three parties and a recorder: a caller, a supervisor,
Agent 1 (a monitored target of the supervisor), and the recorder.
2. Agent 1 transfers the call to Agent 2 (who is not monitored by the supervisor).
Now the call has the following parties: the caller, the supervisor, Agent 2, and the recorder. The supervisor is not released in this scenario, because
MonitorScope
is set to call,
and the call is not finished yet (the monitor scope is not expired).
3. The caller hangs up.
4. SIP Server does not auto-release the call, but will enable the supervisor to continue talking to Agent 2.
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Hiding Supervisor Presence
A supervisor who is performing silent monitoring or whisper coaching must be hidden from other call participants. If the scenario involves whisper coaching, only the monitored agent (who can hear the supervisor) must be aware of his or her presence on the call.
Call participants receive information about other participants joining or leaving the call from the corresponding T-Library events distributed by SIP Server.
The T-Library desktop applications used by call center employees must be able to process the T-Library events and indicate the recent changes in a call status.
For example, they can show that new participant has just joined or left the call.
Hiding a supervisor’s presence means filtering out any events that inform other participants about the supervisor's activity. SIP Server inserts specific information into the T-Library events that allow T-Library clients to decide if a particular event must be shown to the customer or it must be suppressed. SIP
Server make modifications to the events if at least one monitoring session is active on a call. The following attributes support this functionality:
•
AttributeCallState
•
AttributeOtherDNRole
•
AttributeThirdPartyDN
•
AttributeThirdPartyDNRole
The details on how those attributes are modified are found in the Genesys 7
Events and Models Reference Manual.
Configuration Options
The following SIP Server Application-level options support call supervision functionality:
•
cancel-monitor-on-disconnect (
•
default-monitor-mode (
•
default-monitor-scope ( page 204
)
•
intrusion-enabled (
•
monitor-internal-calls ( page 213
Feature Limitations
The following known limitations currently apply to call supervision:
•
SIP Server does not monitor consultation calls that are made either from or to a DN that is under call supervision. No monitoring is activated in this scenario, and the supervisor will not be invited to monitor the agent.
•
Call supervision functionality is disabled for video calls.
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•
A supervisor participating in a monitoring session cannot initiate a 1pcc or
3pcc call transfers or conference calls because it can change the supervisor’s status in the conference call.
•
If a supervisor is already engaged in a call when an agent DN that it is targeting joins a new call (which requires monitoring), SIP Server does not invite the supervisor to monitor the new agent conversation. Even if the supervisor disconnects from its current call, the monitoring session for the new agent conversation will not start. SIP Server will activate monitoring for the next call on the targeted DN.
•
Call supervision functionality is supported only when Stream Manager is used as an MCU. Stream Manager is required because SIP Server sends proprietary information in the SIP messages to set up a specific conference mode on the Stream Manager.
•
When two agents are monitored by two different supervisors, and one agent calls the other agent, SIP Server invites only one supervisor to the call.
Multi-Site Supervision
When SIP Servers operate in a multi-site environment, a supervisor at a local site, from an endpoint controlled by a local SIP Server, can monitor remote agents, whose endpoints are controlled by another SIP Server.
Feature Configuration
To enable this feature, both the supervisor’s SIP Server and the agent’s SIP
Server must be configured for mutual multi-site access with the ISCC transaction type route
or direct-uui
(see Chapter 10, “Multi-Site Support,” on page 267
). Additionally, a special
Routing Point
DN, dedicated for multi-site supervision, must be configured under the agent’s
Switch
object. The
Routing
Point
number must be specified in the observing-routing-point
agent’s SIP Server
Application
object. A special routing strategy must be loaded on the observing Routing Point to route the observing call leg to the
supervisor. (See “Routing Strategy Design Sample” on page 113
.)
A multi-site monitoring session can be initiated by a T-Library client, connected to the supervisor’s SIP Server, by issuing a
TMonitorNextCall request. The request must contain:
•
The
Location
parameter with the remote
value
•
(Optional) The
MonitorMode
parameter
•
(Optional) The
MonitorScope
parameter
If optional parameters are not specified in the
TMonitorNextCall
request, the
values will be taken from the default-monitor-mode
and
configuration options of the agent’s SIP Server
Application
object.
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The
TMonitorNextCall
request, issued by a T-Library client to the supervisor’s
SIP Server, is transmitted through the ISCC connection to the agent’s SIP
Server and registered on both servers.
After a call has been answered by an agent, the agent’s SIP Server initiates the observing service by creating a call leg to the Routing Point, specified by the observing-routing-point
option.
The
EventRouteRequest message generated by the agent’s SIP Server reports the supervisor’s switch name and the number in the
Location
and
Number extensions respectively. The routing strategy, loaded on the observing Routing
Point, must use this information to route the observing leg of the call to the supervisor’s endpoint.
When the supervisor answers, he or she will be connected to the call in the mode defined by the
MonitorMode
parameter of the
TMonitorNextCall
request.
During a multi-site supervision session, the supervisor’s connection to the monitored call can be changed between the initial
MonitorMode
and an open supervisor presence, with the
TSetMuteOff
and
TSetMuteOn
requests containing the supervisor’s DN in the dn
parameter. A supervision session can be canceled with the
TCancelMonitoring
request.
Routing Strategy Design Sample
This section provides a routing strategy design sample (see
should be loaded on the observing Routing Point at the agent’s SIP Server to support multi-site supervision.
Figure 8: A Routing Strategy Design Sample
The sample strategy uses a single the
Multi Function
routing object (see
Figure 9 ). The supervisor’s number and switch name are retrieved from the
EventRouteRequest
extensions by the
ExtensionData
function. These values are passed to the
TRoute
function in the
Destination
and
Location
parameters.
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Figure 9: The Multi Function Object
Call Supervision
114
Figure 10: The Function Properties
Feature Limitations
The following known limitation currently applies to multi-site supervision:
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•
SIP Server will report a correct
DNRole Observer (10) parameter in corresponding events for a supervisor’s DN only if SIP Server operates in a pure SIP environment.
Remote Supervision
The Remote Supervision feature enables supervisors to monitor agent calls from outside the contact center—for example, from an off-premise cell phone.
Call prompting when the supervisor first dials into the contact center is used to determine whether the supervisor is authorized to access the service, what target they want to monitor, and for how long.
The Remote Supervision feature includes the following functionality:
•
Credentials check—SIP Sever can check login and password credentials to verify that the supervisor is authorized to access the service.
•
Targeted monitoring—The supervisor can choose to monitor either an individual agent or calls distributed to agents from a particular Routing
Point or ACD Queue.
•
Session persistence—The session can continue after the first monitored call ends, and for all consecutive calls (for the selected target), until the supervisor decides to hang up. In between calls, the supervisor’s call is parked.
•
DN translation—If configured for it, SIP Server can translate the supervisor’s external DN to an internal DN, so that Reporting can monitor the call.
•
Standard Call Supervision supported—Remote Supervision also supports the following Call Supervision functions: Supervision Modes and
Supervision Scopes. For a description of these functions, see “Call
.
Feature Configuration
When using this feature, a remote supervisor dials from outside the contact center to a Routing Point with a special URS routing strategy. The strategy collects a caller’s login information. Additionally, the strategy may collect the following information from the caller (or otherwise specify):
•
A desired monitoring target number
•
A supervision type (
AllCalls
), mode, and scope
•
An associated internal DN, used for reporting purposes
•
A post-feature destination DN
The strategy places these parameters as
Extensions
attributes in the
TRouteCall request.
Monitoring session starts by routing a remote supervisor’s call to the special pre-defined DN with the number gcti::park.
This DN is used to park the
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While the call is parked, the supervisor hears silence or a music file, specified
Procedure:
Configuring remote supervision
Start of procedure
1. In Configuration Manager, select the SIP Server
Application
object and
TServer
section on the
Options
tab.
This option specifies the music file, which will be played to a remote party parked on the gcti::park
DN.
2. Plan the Remote Supervision routing strategy to meet your specific needs.
The sample strategy described below is a simplified prototype. You may design your own strategy to include any custom logic available in the URS, implement credential verification, based on accessing enterprise databases, and utilize custom prompts. As a result, the strategy should park a call on the gcti::park
device.
3. Prepare the recordings of the voice prompts, which are used in the routing strategy to collect the caller’s login information and optional feature selection. In this sample configuration, Stream Manager is used for announcement service and caller input collection.
a. Prepare the voice prompts in 8 KHz mono audio format, encoded with one of the codecs, supported by Stream Manager.
b. Assign a numeric ID, unique within the Stream Manager’s announcement
directory, to each announcement.
c. Name each recording file using the following format:
<ID>_<codec_suffix>.wav
Refer to the Stream Manager 7.6 Deployment Guide for the list of supported codecs and corresponding filename suffixes.
For efficiency, to avoid real-time trans-coding, you may also use the
SMzip utility, supplied with Stream Manager to trans-code your prompts off-line into zip files, containing the prompts encoded with all supported audio codecs.
To trans-code each prompt, use a command similar to the following:
>smzip -ac all 9000.zip 9000_pcmu.wav
d. Place the resulting audio files (either
.wav
or
.zip
) into the Stream
Manager’s announcement
directory. If you use multiple Stream
Managers, make sure that the files are replicated through all Stream
Managers’ announcement
directories.
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The following sample strategy collects the caller’s Agent Login and password, requests the monitoring target number, and then initiates multiple-call monitoring, using the collected caller’s input.
The strategy uses the following prompts.
Table 11: Strategy Prompts
Prompt text ID
Welcome to the Sample Remote Supervision. 9000
Please enter your Agent login.
9010
Please enter your password.
Please enter the monitoring target number.
An error occurred while processing your request.
9020
9030
9090
File Name
9000_pcmu.wav
9010_pcmu.wav
9020_pcmu.wav
9030_pcmu.wav
9090_pcmu.wav
4. In Interaction Routing Designer (IRD), design your routing strategy. See
“Routing Strategy Design Sample” on page 117 .
5. Save the complete strategy in IRD.
6. Load the strategy into a Routing Point by using the
Loading
tab in IRD.
7. Test your strategy by placing a call from an external phone to the Routing
Point number. Calls from internal DNs are not allowed.
End of procedure
Routing Strategy Design Sample
This section describes a strategy design sample (see Figure 11
), followed by the explanation of each block, called in the IRD terms routing object.
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118
Figure 11: A Routing Strategy Design Sample
Initial Greeting
The first
Play announcement
routing object is used to play back the initial
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Figure 12: Initial Greeting: Parameters Tab
The
LANGUAGE, MSGID and
MSGTXT
parameters are not used by the SIP Server implementation of the Announcement treatment. The
Wait for treatment end check box specifies to the URS that it should wait for the treatment to complete, before proceeding to the next strategy step.
On the
PROMPT
tab (see
), the
ID
specifies the prompt to be played. The
Interruptible
flag allows the caller to skip the greeting message by pressing any key on the phone keypad.
Figure 13: Initial Greeting: PROMPT Tab
Collecting Agent Login Code
Although your strategy may implement an arbitrary approach to perform caller verification, this sample strategy demonstrates the SIP Server’s built-in functionality. SIP Server verifies the login-id
and password
information provided in corresponding extensions parameters, against the Agent Login
Code
and
Password,
specified in the
Agent Login object in the Configuration
Layer. Therefore, for the purpose of this strategy, ensure that you have configured
Agent Login
objects with the numeric-only Agent Login
Codes
and
Passwords,
so that they could be entered through the phone keypad.
After the initial greeting, the strategy uses the
Play Announcement and collect digits
routing object to request the caller’s Agent Login (see Figure 14 ). It
plays the “
Please enter your Agent login
” prompt and retrieves the caller’s
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#
” key.
Figure 14: Collecting Digits: Parameters Tab
120
Figure 15: Collecting Digits: PROMPT Tab
Verifying Caller Input
The next
Generic segmentation
object verifies that the caller's input (returned by the
CED[]
function) is not empty (see Figure 16 ).
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Figure 16: The Generic Segmentation Object
If the caller did not enter any digits within 15 seconds (as specified by the
START_TIMEOUT
parameter in the preceding
Play Announcement and collect digits
object, the segmentation object will repeat the previous prompt.
Storing the Entered Agent Login Code in a Variable
The next
Assign routing object defines the
LoginId
internal strategy variable
(use
Variables
button to define the variable) and assigns the caller’s input
value to this variable (see Figure 17 ).
Figure 17: The Assign Object
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Collecting Password and Monitoring Target
Subsequent routing strategy blocks prompt the caller to enter a password and the desired monitoring target DN number, an a sequence, similar to the one used to collect the Agent Login information.
Collected input is placed into the
Password
and
AgentDN
internal variables.
Invoking Remote Monitoring
The key point of the strategy is the
Multi Function
object (see
).
122
Figure 18: The Multi Function Object
The
Multi Function
object attaches the extensions parameters to the
TRouteCall
request, as the request extensions, and routes the call to the gcti::park
DN.
SIP Server determines the desired feature—remote observing from the value of the feature
extension parameter—and verifies the user’s login information.
If the supplied information is correct, SIP Server starts Remote Monitoring session. The supervisor hears silence or a music file until a call comes to the specified target. After that, the caller connects to the monitored call.
It the parameters are incorrect (for example, wrong login, password, or monitoring target information), SIP Server responds with the
EventError message to the
TRouteCall
request. This will trigger the default branch of the
Multi Function object.
Verifying the Result
If an error occurs for any reason, the strategy uses the subsequent
Play announcement
object to notify the caller about the error and return the control flow to the point, where the user is requested to enter the Agent Login again.
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Use of the Extensions Attribute
The following
Extensions
attribute parameters can be used to configure the application that starts and defines the remote supervision session:
•
feature
—This key with the remote-observing
value triggers registration of the routed party as a supervisor with parameters specified in additional
Extensions
described in this section.
•
dn
—An optional DN number that can be used during the monitoring session, as a substitute for the external PSTN number that the supervisor used to dial in. If you do not include this parameter, no TEvents will be distributed for this DN.
•
login-id
and password
—These optional parameters are used to establish that the supervisor is authorized for remote access to the feature. The treatment can prompt for the login-id
alone, or for both the login-id
and the password
.
•
agent-dn
—The target that the supervisor wants to monitor. This can be a
Routing Point, ACD Queue, or an agent DN.
•
monitor-type (AllCalls)
—An optional parameter that enables the supervisor to monitor all consecutive calls for the selected target, until the supervisor decides to hang up and end the monitoring session. In between monitored calls, the supervisor’s call is parked.
•
post-feature-dn
—An optional parameter that specifies a Routing Point, to which the supervisor will be connected after the supervision session.
Feature Limitations
The following known limitations currently apply to remote supervision:
•
MonitorMode
of remote supervision session cannot be changed during active supervision.
•
One Call
supervision type is not supported.
Call Transfer and Conference
SIP Server supports the following call transfers:
•
First-party call control (1pcc) transfers: single-step and two-step transfers.
•
Third-party call control (3pcc) transfers: single-step and two-step transfers.
In these scenarios the
REFER
request method is used. If an endpoint does not support the
REFER
method, the re-
INVITE
method can be configured for use in two-step transfers.
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Note:
First-party call control calls are not supported when SIP Server is integrated with the Siemens HiPath 8000 switch.
Starting with release 7.5, SIP Server can send a
REFER
message to the transferred party when the following scenarios occur:
•
A
REFER
message was received from an endpoint.
•
A single-step call transfer was received from a client.
This removes SIP Server from the SIP signalling loop.
SIP Server analyzes the destination specified in either scenario and then determines if a different contact is specified in the outgoing
REFER
message based on the following criteria:
•
The destination is unknown to SIP Server (no regular DN and no
Trunk
DN contains the prefix that matches the specified destination).
•
The destination refers to a DN of type
Trunk
that contains the
option set to true
(see
).
If either of the scenarios is true, SIP Server prepares a
Contact
for the
Refer-To header of the outgoing
REFER
message, based on the following conditions:
•
If there is no DN, and no DN of type
Trunk
is specified as the destination, the
Contact
information from the caller DN or the
Trunk
DN will be specified in the
Refer-To
header of the outgoing
REFER
message. It is the responsibility of the caller to determine where to transfer the call.
•
If a
Trunk
DN is specified as the destination, and it contains the oosp-transfer-enabled
option set to true
, the contact information from this trunk will be specified in the
Refer-To
header of the outgoing
REFER message.
Note:
If the caller DN or the
Trunk
DN in either scenario contains the
option specified ( page 237 ), the value of this option
will be specified in the
Refer-To
header of the outgoing
REFER
message.
Routing to External Destination Using REFER
SIP Server supports the routing of an inbound call to an external destination by using the
REFER method. When the feature is activated, SIP Server places itself in the Out Of Signaling Path. This feature applies to the following scenarios:
•
An inbound call is routed from a Routing Point to an external destination.
•
An agent transfers an inbound call by using the single-step transfer to a
Routing Point, and then the call is routed to an external destination.
•
An agent transfers an inbound call by using the blind transfer to a Routing
Point, and then the call is routed to an external destination.
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Note:
This feature is not applicable for scenarios (the second and the third, above) where a conference (supervision or emergency recording) is involved.
To support this feature, configure the following options for a DN of type
Trunk
:
•
—Set the value for this option to true
•
—Set the value for this option to
true
(
Single-Step Transfer Using re-INVITE
Scenarios in which single-step transfers use the re-
INVITE
request method
require that the originating DN be configured with the refer-enabled
option set to false
).
Conference Calls
SIP Server supports third-party call control conferences with central mixing, using MCUs that support more than three participants.
Silence Treatment in Conference
SIP Server can provide a silent treatment for conference call participants when one of them places the call on hold. This will allow conference call participants to continue the conference without interruption (or hearing the music-on-hold treatment). This feature is applicable to conference calls where participants are located in single-site or multi-site environments.
For this feature to work, the music-in-conference-file
) must be set to the valid name of the silent audio file to be played in applicable conferences (more than two active participants).
Consultation Transfers and Conferences
SIP Server provides the ability for parties participating in a consultation call to initiate 3pcc (third-party call control) transfers or conferences. A typical supported scenario would be:
1. An inbound call is routed to Agent A.
2. Agent A originates a consultation call with Agent B.
3. Agent B originates a consultation call with Agent C.
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Consultation Transfers For Calls on a Routing Point
SIP Server allows an agent to complete a consultation transfer of a call that is located on a Routing Point. This transfer operation is supported in single-site and multi-site environments.
In a single-site environment, the call transfer can be completed both when a treatment is playing for the call on a Routing Point, or when the call is just parked on a Routing Point.
In a multi-site environment, when a consultation call is made to a Routing
Point located on another site, the call transfer can be completed only when a treatment is playing for the call on the Routing Point. If a call is just parked on a Routing Point, the complete transfer operation will not be successful and SIP
Server will generate an
EventError
(
Call in invalid state
) message.
Alternating Between Main and Consultation Calls
SIP Server enables agents to handle up to three 3pcc calls on their SIP endpoint. This functionality supports alternate call operation between the main call and an answered consultation call, as well as the main call and a consultation call that is queued on a Routing Point, as described in the following scenario:
1. A call is routed to Agent A.
2. Agent A places the call on hold and initiates a consultation call by dialing to a Routing Point.
3. Agent A is placed in a queue at the Routing Point, waiting for another agent to become available (a treatment is played).
4. Agent A places the consultation call on hold and retrieves the main call from hold.
For the alternate call operation to work transparently in a multi-site environment, a treatment must be applied to a call on a Routing Point at the earliest possible time. If a treatment is not applied, the alternate call operation will not be successful and SIP Server will generate an
EventError (Call in invalid state)
message.
Feature Limitations
The following known limitation currently applies to call conferences:
•
Three-way conference on the phone is not reported properly. Call participants can talk to each other, but such a call is not reported as conference.
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Class of Service
Class of Service (COS) is the functionality that defines telephony capabilities for a device or an agent. In SIP Server, COS telephony capabilities are defined by configuring the following:
•
Outbound dialing rules
•
Ring-through rules
Class of Service can be assigned to the device (a
DN
object in SIP Server
Switch configuration) or to the agent (an
Agent Login
object in SIP Server
Switch configuration).
The COS assigned to the agent takes precedence over the COS assigned to the device. That is, when different COSs are assigned to the device and to the agent, SIP Server will use the COS assigned to the agent.
Outbound Dialing Rules
Outbound dialing rules define whether an agent or device is able to make outbound calls, and they also translate dialed numbers according to the specified patterns.
An outbound dialing rule is specified as a pair of patterns. The first pattern, in-pattern,
is used to match the dialed number. The second pattern, out-pattern,
defines how to transform the dialed number, when matched, to the in-pattern
.
It is possible to specify several outbound dialing rules within a single COS definition.
When an outbound call is initiated either from a device (1pcc) or from a softphone (3pcc), SIP Server attempts to match the dialed number with one of the outbound dialing rules. If the matching rule is not found, SIP Server will proceed with the outbound call as it is dialed. If the matching rule is found
(that is, in-pattern
in one of the outbound dialing rules matched the dialed number), SIP Server will apply this rule to the call according to the following procedure:
1.
out-pattern
from the outbound dialing rule is applied to the dialed number to produce the resulting dialed number.
2. If the resulting dialed number is empty, SIP Server rejects an attempt to make the outbound call.
3. If the resulting dialed number is not empty, SIP Server attempts to place the outbound call to the resulting dialed number.
If there are multiple outbound dialing rules configured within a single COS, and in-pattern in more than one rule matches the dialed number, SIP Server will choose the outbound dialing rule with the most matched digits in it.
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Example
Here is an example of the outbound dialing rule: in-pattern=411;out-pattern=9411
Dialing Rule Format
Dialing rules must conform to the following syntax, represented using
Augmented Backus-Naur Form (ABNF) notation. dialing-plan-rule name in-pattern out-pattern digit-part symbol-part range sequence abstract-group
= [name] in-pattern [out-pattern]
= *( ALPHA / DIGIT / "-")
= 1*(digit-part / abstract-group)
= 1*(symbol-part / group-identifier) *param-part
= digits / range / sequence
= digits / symbols
= "[" digits "-" digits "]" group-identifier
= "[" 1*(digits [","] ) "]" group-identifier
= fixed-length-group / flexible-length-group flexible-length-group = "*" group-identifier param-part = ";" param-name "=" param-value param-name = "ext" / "phone-context" / "dn" symbols = 1*("-" / "+" / ")" / "(" / ".")
Common Syntax Notations
Syntax notations common to many of these rules include:
•
*
—Indicates that 0 to an infinite number of the item following this symbol are acceptable.
•
1*
—Indicates that one repetition is required. For T-Server, only one instance is acceptable.
•
/
—Indicates that any of the items mentioned, or a combination of those items, is acceptable.
Component Notations
Component notations include:
•
dialing-plan-rule = [name] in-pattern [out-pattern]
Where:
[name]
is the name of the rule option—for example, rule-01.
In ABNF notation, the brackets (
[)]
indicate that 0 or 1 instance of the component is required. However, for SIP Server, a name is required.
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in-pattern
is the part of the rule to which SIP Server looks when attempting to match the input number.
[out-pattern]
is the part of the rule that instructs SIP Server how to translate the input number into the required format. The brackets indicate that either 0 or 1 instance is required. You must create an outpattern
for number translation rules.
• name = *( ALPHA / DIGIT / "-")
Where:
ALPHA indicates that letters can be used in the name for the rule option.
DIGIT indicates that numbers can be used in the name for the rule option.
“-”
indicates that a dash (
-
) can also be used in the option name—for example, rule-01.
•
in-pattern = 1*(digit-part / abstract-group)
Where: digit-part
represents numbers. SIP Server uses this when selecting the most appropriate rule from the entire dialing plan. abstract-group
represents one or more letters with each letter representing one or more numbers. SIP Server uses this when transforming a dial string.
For example,
[1-9] is the digit-part
(representing a range of numbers) and
ABBB
is the abstract-group for in-pattern=[1-9]ABBB
.
•
out-pattern = 1*(symbol-part / group-identifier) *param-part
Where: symbol-part
represents digits, symbols, or a combination. Symbols are rarely used. They are not used in the United States.
group-identifier
are letters that represent groups of numbers. A letter in the out-pattern
represents one or more digits, based on the number of times the letter is used in the in-pattern.
*param-part
represents an additional parameter, such as phonecontext.
Remember that an asterisk (*) means that 0 to an infinite number of these are acceptable.
For example, in rule-04; in-pattern=1AAABBBCCC;out-pattern=91ABC, 91 is the symbol-part
;
A, B,
and
C
are group-identifiers in the out-pattern, each representing three digits, since there are three instances of each in the in-pattern
.
Note:
Prefix an out-pattern
value with a plus sign (+) for the inbound rule when the output must be in a global form (E.164 format).
• digit-part = digits / range / sequence
Where: digits
are numbers 0 through 9.
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range
is a series of digits—for example, 1–3.
sequence is a set of digits.
•
symbol-part = digits / symbols
Where: digits are numbers 0 through 9.
symbols include such characters as
+, -,
and so on.
•
range = "[" digits "-" digits "]" group-identifier
Where:
"[" digits "-" digits "]"
represents the numeric range—for example,
[1-2]
.
group-identifier
represents the group to which the number range is applied.
For example,
[1-2]
applies to group identifier
A
for in-pattern=[1-
2]ABBB.
When SIP Server evaluates the rule to determine if it matches the number, it examines whether the first digit of the number, identified as group-identifier A
, is
1
or
2.
•
sequence = "[" 1*(digits [","] ) "]" group-identifier
Where:
"
[" 1*(digits [","] ) "]"
represents a sequence of digits, separated by commas, and bracketed. SIP Server requires that each digit set have the same number of digits. For example, in
[415,650]
the sets have three digits.
group-identifier
represents the group to which the number sequence is applied.
For example, in in-pattern=1[415,650]A*B, [415,650]
applies to group-identifier A
. When SIP Server evaluates the rule to determine if it matches the number, it examines whether the three digits ( groupidentifier A
) following the
1
in the number are
415 or
650.
•
abstract-group = fixed-length-group / flexible-length-group
Where: fixed-length-group
specifies a group composed of a specific number of digits, and determined by the number of times the group identifier is included in the in-pattern.
For example, for in-pattern=1AAABBBCCCC, there are three digits in group
A
and
B,
but four in group
C.
When you create an out-pattern
, you include the group identifier only once, because the in-pattern
tells SIP Server how many digits belong in that group. For example, rule-04
) is in-pattern=1AAABBBCCCC; out-pattern=91ABC.
flexible-length-group
specifies a group composed of 0 or more digits in the group represented by the group-identifier.
For example, in in-pattern=1[415,650]A*B, *B
represents the flexible length group containing the remaining digits in the number.
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The component
abstract-group
is used only for the in-pattern
.
•
fixed-length-group = 1*group-identifier
See the earlier explanation under abstract-group.
•
flexible-length-group = "*" group-identifier
See the earlier explanation under abstract-group.
•
param-part = ";" param-name "=" param-value
Where:
“;”
is a required separator element.
param-name
is the name of the parameter.
“=”
is the next required element.
param-value
represents the value for param-name.
•
param-name = "ext" / "phone-context" / "dn"
Where:
“ext”
represents the extension.
“phone-context”
represents the value of the phone-context
option configured on the switch.
“dn”
represents the directory number.
•
param-value = 1*ANYSYMBOL
Where:
ANYSYMBOL
represents any number, letter, or symbol with no restrictions.
•
group-identifier = ALPHA
•
digits = 1*DIGIT
•
symbols = 1*("-" / "+" / ")" / "(" / ".")
Transformation
The transformation algorithm consists of two phases:
1. Input number parsing based on in-pattern.
2. Output number construction based on out-pattern
and the parsed input string.
The first phase consists of selecting all group elements from in-pattern
(the abstract-group, range,
and choice
parameters), and assigning them a corresponding group of digits from the input number. There are three kinds of such assignment:
•
Explicit—For ranges, sets, and fixed-length-groups where the length and position of the group is set explicitly in in-pattern.
•
Positional—For flexible-length-groups; the digits are selected exclusively based on their position with respect to other groups.
•
Analytical—For entities; the digits are selected based on some kind of analytical processing.
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Examples
These are examples of how SIP Server applies the rules configured above to various input numbers.
Example 1
Input Number
914159131472
in-pattern
9*A
out-pattern
+A
Output Number
+14159131472
011441581234567 011#CABBB*D +A-(B)-D
1472 [1-8]ABBB AB
911
14159131472
16503570622
AAA
*A9131472
1[415,650]A*B
9A
B
+44-(158)-1234567
1472
9911
80409131472 80409131472
3570622
rule-01 rule-02
Example 2
rule-01=in-pattern=[1-8]ABBB;out-pattern=AB rule-02=in-pattern=AAAA;out-pattern=A rule-03=in-pattern=1[415,650]A*B;out-pattern=B rule-04=in-pattern=1AAABBBCCCC;out-pattern=91ABC rule-05=in-pattern=*A913BBBB;out-pattern=80407913B rule-06=in-pattern=011#CA*B;out-pattern=9011AB
SIP Server receives input number
2326.
As a result of the rule selection process, SIP Server determines that the matching rule is rule-01: name=rule-01;in-pattern=[1-8]ABBB;out-pattern=AB
The matching count for this rule is
1,
because Group
A
matches the digit
2.
As a result of the parsing process, SIP Server detects two groups: Group
A = 2 and Group
B = 326.
SIP Server formats the output string as
2326.
SIP Server receives input number
9122.
As a result of the rule selection process, SIP Server determines that the matching rule is rule-02: name=rule-02;in-pattern=AAAA;out-pattern=A
The matching count for this rule is
0;
however, the overall length of the input number matches that of the in-pattern
configuration.
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rule-03 rule-04 rule-05 rule-06
As a result of the parsing process, SIP Server detects one group:
Group
A = 9122.
SIP Server formats the output string as
9122.
SIP Server receives input number
16503222332.
As a result of the rule selection process, SIP Server determines that the matching rule is rule-03: name=rule-03;in-pattern=1[415,650]A*B;out-pattern=B
The matching count for this rule is
4,
because the first digit matches and all three digits in Group
A
match.
As a result of the parsing process, SIP Server detects two groups:
Group
A = 650
and Group
B = 3222332.
SIP Server formats the output string as
3222332.
SIP Server receives input number
19253227676.
As a result of the rule selection process, SIP Server determines that the matching rule is rule-04: name=rule-04;in-pattern=1AAABBBCCCC;out-pattern=91ABC
The matching count for this rule is
1,
because the first digit matches.
As a result of parsing process, SIP Server detects three groups: Group
A = 925,
Group
B = 322,
and Group
C = 7676.
SIP Server formats the output string as
919253227676.
SIP Server receives input number
4089137676.
As a result of the rule selection process, SIP Server determines that the matching rule is rule-05: name=rule-05;in-pattern=*A913BBBB;out-pattern=80407913B
The matching count for this rule is
3,
because three digits match.
As a result of the parsing process, SIP Server detects two groups:
Group
A = 408
and Group
B = 7676.
SIP Server formats the output string as
804079137676.
SIP Server receives input number
011441112223333.
As a result of the rule selection process, SIP Server determines that the matching rule is rule-06: name=rule-06;in-pattern=011#CA*B;out-pattern=9011AB
The matching count for this rule is
3,
because three digits match.
As a result of the parsing process, SIP Server detects two groups: Group
A = 44 and Group
B = 1112223333.
SIP Server formats the output string as
9011441112223333.
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Call Rejection by Out-Dialing Rules
A call attempt can be rejected by the out-dialing rules. To indicate this condition, SIP Server generates an
EventError
response to the corresponding request, with the reason code
Invalid Destination DN (415).
Ring-Through Rules
The ring-through rules define whether a call is sent to an agent or a device. The following ring-through rules are supported by SIP Server:
•
Reject call when a device is already in a call
This rule is enforced by the
Switch
object-level configuration option reject-call-incall
(
) within COS.
•
Reject call when an agent is not ready on a device
This rule is enforced by the
Switch
object-level configuration option reject-call-notready
(
Call Rejection by COS Ring-Through Rules
A call attempt can be rejected by the COS ring-through rules. To indicate this condition, SIP Server generates an
EventError
response to the corresponding request, with the reason code
Invalid Destination DN (93)
.
Feature Configuration
provides an overview of the main steps required to configure Class of
Service.
Table 12: Task Flow—Configuring Class of Service
Objective
1. Configure a COS DN.
Related Procedures and Actions
Configure a DN of type
Voice over IP
Service to represent the COS entity itself. This
DN should have outbound dialing rules and ring-through rules specified in the options. See
“Configuring a COS DN” on page 135
.
2. Assign COS to a device.
Assign the COS DN to one or multiple DNs of type
Extension
or
ACD Position
within the same
Switch
configuration object. See
“Assigning COS to a Device” on page 136
3. Assign COS to an agent.
Assign the COS DN to one or multiple Agent
Logins. See “Assigning COS to an Agent” on page 136 .
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Procedure:
Configuring a COS DN
Start of procedure
1. Create a COS DN under the SIP Server
Switch
object with a type of
Voice
Over IP Service.
2. On the
Annex
tab of the COS
DN
object, in the
TServer section, set the configuration option service-type
to cos.
3. On the
Annex
tab of the COS
DN
object, in the
TServer section, specify the outbound dialing rules. Each rule is defined using the configuration option named out-rule-<n>
(
out-rule-1 = in-pattern=411;out-pattern=9411
Rules must be numbered sequentially—for example, out-rule-1, out-rule-2,
and so on.
4. On the
Annex
tab of the COS
DN
object, in the
TServer section, specify the ring-through rules—for example: reject-call-incall = true reject-call-notready = true
illustrates a sample configuration for the COS DN with outbound dialing rules and ring-through rules specified.
Figure 19: Configuring COS DN: Sample Configuration
5. When you finished, click
Apply
to save your configuration.
End of procedure
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Assigning COS to a Device
COS is assigned to a device by associating the COS DN with the device DN of type
Extension
or
ACD Position.
In the
TServer
section on the
Annex
tab, add the cos
configuration option, with the value set to the name of the COS DN.
illustrates a sample configuration of the device DN with COS assigned.
Figure 20: Assigning COS to a Device: Sample Configuration
COS may be assigned to multiple DNs by using the
Manage Options
command in Configuration Manager.
Assigning COS to an Agent
COS is assigned to an agent by associating the COS DN with the
Agent Login object. In the
TServer
section on the
Annex
tab of the
Agent Login
object, add the cos
configuration option, with the value set to the name of the COS DN.
illustrates a sample configuration of the
Agent Login
object with
COS assigned.
136
Figure 21: Assigning COS to an Agent Login: Sample Configuration
Framework 7.6
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Checking the Destination Availability
SIP Server uses COS to analyze the availability of the destination in the following order:
1. SIP Server checks if an agent is logged in on the extension.
2. If the agent is logged in, the Agent Login COS is applied.
3. If the agent is not logged in on the device, SIP Server checks if COS is configured for the device.
4. If COS is configured for the device, SIP Server applies this device’s COS.
5. If COS is not configured for the device, SIP Server checks if options reject-call-incall
and reject-call-notready
are specified for the device directly (without using COS).
6. If the reject-call-incall
and reject-call-notready
options are specified,
SIP Server uses these options.
7. If none of the preceding apply, SIP Server considers the destination available.
DTMF Tones Generation
SIP Server can send a request to Stream Manager to generate DTMF tones using the
TApplyTreatment
request with the
PlayApplication
treatment.
The following key-value pairs (see Table 13 ) are used in the attribute
parameters
for the
PlayApplication
treatment:
Table 13: Key-Value Pairs for TreatmentPlayApplication
Key
GSIP_APP_ID
Type
Integer
GSIP_DTMF_TO_DIAL String
GSIP_DTMF_DURATION Integer
Value
Specifies the application, which tells
SIP Server to send a request to Stream
Manager to generate DTMF tones.
The DTMF string to be generated.
The duration of the DTMF tone in msec. This parameter is optional with the default value of
100 msec.
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Feature Configuration
provides an overview of the main steps required to configure DTMF tones generation support.
Table 14: Task Flow—Configuring DTMF Tones Generation
Support
Objective Related Procedures and Actions
1. Configure a SIP Server
Application
object.
In the
TServer
section on the
Options
tab of the
SIP Server
Application
object in Configuration
Manager, set the configuration option sip-dtmfsend-rtp
true
.
2. Configure an application service.
Configure a DN of type
Voice over IP Service with the following configuration options:
• contact:
Set to the device’s IP address used for sending DTMF tones.
• service-type:
Set to application
.
See “Configuring an application service” on page 89 for details.
Emulated Agents
SIP Server fully emulates business-call–handling functions. It also performs agent emulation for any agent who logs in using a request, when the device provided in the
ThisQueue
attribute is defined as a Routing Point or ACD
Queue in the SIP Server configuration.
SIP Server provides a fully functional agent model that enables full agent support for SIP Server desktop applications as well as for other Genesys solutions.
Business-Call Handling
This section describes how SIP Server handles different types of calls.
SIP Server Call Classification
SIP Server automatically assigns every call to one of three categories—
business, work-related, or private. Based on this assignment, SIP Server applies the appropriate business-call handling after the call is released.
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Business Calls
SIP Server automatically categorizes as a business call any call distributed to an agent either from a Queue or from a Routing Point. Use the following configuration options to define what additional calls to or from an agent are classified as business calls:
•
inbound-bsns-calls
•
outbound-bsns-calls
)
•
inherit-bsns-type
(
•
internal-bsns-calls
)
•
unknown-bsns-calls
(
)
Work-Related Calls
SIP Server categorizes as a work-related call any non-business call that an agent makes while in After Call Work (ACW). SIP Server does not apply any automatic business-call handling after a work-related call.
Because emulated agents can make or receive a direct work-related call while in wrap-up time, SIP Server pauses the emulated wrap-up timer for the duration of such a call.
If an agent receives a direct work-related call during legal-guard time, SIP
Server cancels the legal-guard timer and reapplies it at the end of the workrelated call.
Private Calls
SIP Server categorizes as a private call any call that does not fall into the business or work-related categories. SIP Server does not apply any automatic business-call handling after a private call. If emulated agents receive a direct private call while in wrap-up or legal-guard time, the emulated wrap-up or legal-guard timer is not interrupted.
Emulated Agents Support
SIP Server provides a fully functional emulated-agent model that you can use either in addition to agent features available on the PBX, or in place of them where they are not available on the PBX.
When this feature is used, SIP Server emulates the following functionality:
•
Login and logout
•
Agent set Ready
•
Agent set Not Ready (using various work modes)
•
Automatic after-call work
•
After call work in idle
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Chapter 6: SIP Server Feature Support Emulated Agents
•
Automatic legal-guard time to provide a minimum break between business-related calls
Emulated Agent Login/Logout
You can configure SIP Server to perform emulated login either always, never, or on a per-request basis. Use the following SIP Server configuration options to configure emulated agent login:
•
emulate-login
)
•
emulated-login-state
(
•
agent-strict-id
)
Emulated Agent Ready/NotReady
Emulated agents can perform an emulated
Ready or
NotReady
request regardless of whether they are on a call, subject to the rules governing work modes.
SIP Server also reports any change in agent mode requested by the agent while remaining in a
NotReady
state (self-transition).
Note:
Note that the Genesys 7 Events and Models Reference Manual and the
Voice Platform SDK 7.6 .NET (or Java) API Reference define which agent state/agent mode transitions are permissible.
Timed and
Untimed ACW
Events and
Extensions
Emulated After-Call Work
SIP Server can apply emulated wrap-up (ACW) for agents after a business call is released, unless the agent is still involved in another business call (see
“Business Calls” on page 139 ).
SIP Server applies emulated ACW for an agent after any business call is released from an established state. SIP Server automatically returns the agent to the
Ready
state at the end of a timed ACW period. The agent must return to the
Ready
state manually when the ACW period is untimed.
SIP Server indicates the expected amount of ACW for an agent in
EventEstablished,
using the extension
WrapUpTime
. It is not indicated in
EventRinging, because the value may change between call ringing and call answer. Untimed ACW is indicated by the string value untimed;
otherwise, the value indicates the expected ACW period in seconds.
SIP Server reports ACW using
EventAgentNotReady
with workmode
=
3
(
AgentAfterCallWork
), and it indicates the amount of ACW it will apply using the extension
WrapUpTime.
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SIP Server sends
EventNotReady(ACW)
before
EventReleased at the end of the business call.
Emulated ACW Period
The amount of emulated ACW that SIP Server applies (when required) after a
business call is determined by the value in configuration option wrap-up-time .
Configuration option
integer value of wrap-up-time
indicates untimed ACW. To specify untimed
ACW in request extensions or user data, you should use the string untimed instead. All positive integer values are treated as indicating timed ACW (in seconds). For backward compatibility, the default value of untimed-wrapup-value
is
1000
.
Note:
Changing the value of untimed ACW should be done with care, because it may affect the interpretation of all integer values of the option wrap-up-time
in Configuration Manager. If lowered, it may change timed ACW to untimed ACW, or disable ACW altogether. If raised it may change untimed or disabled ACW to timed ACW. The use of the option (string) value
untimed
is encouraged where possible to minimize the impact of any future changes to the value of option untimed-wrap-up-value
.
ACW in Idle
An agent can activate wrap-up time on request when idle, by issuing a
TAgentNotReady request with workmode
=
3 (AgentAfterCallWork)
.
You can configure this feature using the following options:
•
timed-acw-in-idle
(
•
acw-in-idle-force-ready
(
Extending ACW
An agent can request an extension to the amount of emulated ACW for a call while in emulated ACW or in the legal-guard state.
The agent requests an extension to ACW by sending
RequestAgentNotReady with workmode
=
3 (AgentAfterCallWork)
. SIP Server determines the period of the extended ACW from the extension
WrapUpTime
, as follows:
•
Value =
0
—There is no change to the ACW period, but SIP Server reports how much ACW time remains.
•
Value greater than
0
—SIP Server adds the given number of seconds to the timed ACW period. Untimed ACW remains unaffected.
•
Value = untimed
—SIP Server applies untimed ACW.
SIP Server sends
EventAgentNotReady with workmode
=
3
(AgentAfterCallWork)
, reporting the newly extended amount of ACW using
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WrapUpTime
. If the agent was in the emulated legal-guard state,
SIP Server places the agent back into the emulated ACW state.
The agent may extend the period of ACW as many times as desired. At the end of the extended timed ACW period, SIP Server applies legal-guard time, if any is configured. No legal-guard time is applied if the emulated ACW was untimed.
Calls While in Emulated ACW
SIP Server’s handling of an agent making or receiving a call while in emulated
ACW is governed by the configuration option
(see
).
Emulated Legal-Guard Time
SIP Server applies emulated legal-guard time for agents before they are about to be automatically set
Ready
after any period of timed ACW, or after the last business call is released where there is no ACW to be applied. It is a regulatory requirement in many countries that agents have a break of a few seconds before the next call can arrive. No legal-guard time is applied if the ACW period was not timed, or if the agent is not being placed into the
Ready
state.
SIP Server reports legal-guard time using
EventAgentNotReady with workmode
=
2 (LegalGuard)
. If an agent requests to be logged out during emulated legal-guard time, SIP Server immediately logs the agent out.
If the agent requests to go to a
Not Ready
or
Ready state during legal-guard time, SIP Server terminates legal-guard time and transitions the agent to the requested state. If the agent requests to return to the ACW state, SIP Server reapplies legal-guard time at the end of ACW, provided that the agent still requires it according to the preceding criteria.
The period of legal-guard time is determined by the configuration option legal-guard-time
).
Endpoint Service Monitoring
When SIP Server starts up, it considers that all DNs configured in
Configuration Manager are in the
In Service
state.
Note:
Where SIP Server functions as an application server behind a softswitch, a DN is considered to be in
Out of Service
state if the softswitch responds with a
408 Request Timeout message to an
INVITE message during creation of a new call.
DNs are considered out of service in two scenarios:
•
The SIP endpoint fails to respond to the incoming
INVITE
message during the creation of a new call.
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•
The SIP endpoint fails to respond to the re-
INVITE
message during an already established telephone call.
In both scenarios—when the SIP endpoint fails to respond to the incoming
INVITE
message during the creation of a new call, or when it fails to respond to the re-
INVITE
message during an already established call—SIP Server generates an
EventDNOutOfService
message.
DNs are considered back in service in several scenarios:
•
When a SIP
REGISTER
message comes from the endpoint
•
When an endpoint initiates a call by sending an
INVITE
message
•
When an endpoint responds to an
INVITE
message with a
Ringing (OK) message
An
EventDNBackInService
message is generated in all scenarios.
Instant Messaging
SIP Server supports the Instant Messaging (IM) media type as follows:
•
A special SDP message for the IM media type is supported within the SIP
INVITE
dialog.
The IM SDP message must be in a form that is generated and accepted by
Microsoft Live Communication Server and Office Communicator. SIP
Server supports standard SIP call flows for IM, and the SIP
INVITE messages are assumed to be the same for the IM sessions. SIP Server produces the same TEvents for the IM sessions as it does for voice calls.
•
A SIP
MESSAGE
request method is supported within an established
INVITE dialog to exchange instant messages within a SIP session.
•
The content of an instant message can be distributed via a
EventUserEvent message to a DN of type
Communication DN
with the name gcti::im
.
Any application that need to receive the content of any instant message must register this DN. When a SIP Server
MESSAGE
request is received by the SIP Server IM session, SIP Server informs its clients by distributing an
EventUserEvent
message. The client applications can then perform additional tasks in response to the IM content.
Instant Messaging Transcript
SIP Server supports exchanging instant messages via T-Library within the established call context. This enables applications, such as Agent Desktop, to display instant messages that were sent or received during a conversation, including the chat transcript, and to send an instant message to other parties of the call.
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The instant message is delivered to a T-Library client via
EventPrivateInfo messages. A T-Library client can send an instant message using a T-Library request
TPrivateService.
SIP Server distributes the
EventPrivateInfo
message when one of the participants in the call sends an instant message. The
EventPrivateInfo message is sent to all other participants in the call.
AttributeExtensions
of the
EventPrivateInfo
message contains information about the instant message.
SIP Server distributes
EventPrivateInfo
messages with the Instant Messages
Transcript when a new participant is added to the call as a result of a transfer or conference operation. The Instant Messages Transcript contains all instant messages that were previously exchanged between the participants in the call.
EventPrivateInfo
with the transcript is sent to the new participant who was added to the call.
The following keys are supported:
•
im
—Contains the text of the instant message.
•
im-content-type
—Contains the value provided in the
Content-Type
header of the SIP message that delivered IM.
•
im-transcript
—Contains the transcript of the IM call. The value of this extension is a string containing an XML document that complies with the
Genesys Multimedia Chat Transcript Schema.
Note:
Transcript data delivery is only supported via T-Library.
Supported Call Operations
SIP Server supports the following call operations within an IM session.
Direct Calls
•
Direct 1pcc calls—SIP Server processes direct 1pcc calls with IM media the same way as voice calls. SIP Server will specify
AttributeMediaType =
5
(
TMediaChat
) in TEvents for such calls.
•
Direct 3pcc calls—SIP Server processes a
TMakeCall
request for a call with
IM media, when it is specified in the
Extensions
attribute with the chat
key containing a value of true
. If it is not specified, SIP Server will process the
TMakeCall
request in the normal fashion.
Hold
SIP Server supports the Hold operation in the same manner as for voice calls.
The
Hold
request has no effect on the SIP signaling level for IM calls. SIP
Server will not re-
INVITE
SIP endpoints with different SDP dialogs when
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Hold
operation, because no changes in the SDP dialog are necessary.
Note:
SIP Server does not support Music On Hold (MOH) for IM calls.
Transfer
SIP Server supports transfer in the same manner as for voice calls. However there are some exceptions:
•
SIP endpoints that are already part of a call are not considered for re-INVITE
requests.
•
The IM SDP dialog is used to
INVITE
the SIP endpoint that is the call transfer destination.
Single-step transfers are supported for non-conference calls. Single-step transfer of the conference is not supported.
Two-step transfers are supported for all calls. Agents can exchange IM text with each other during the consultation call. These IM text messages are not visible to the customer.
Conference
SIP Server supports conference for calls made within the IM session, and it is performed in the same manner as for voice calls. However, SIP endpoints that are already part of a call are not considered for re-
INVITE
requests. Also, unlike regular voice calls, SIP Server does not use Stream Manager to establish conference calls. SIP Server establishes an IM conference between the SIP endpoints itself, by issuing SIP
MESSAGE
requests to all conference participants.
Routing
SIP Server supports routing of IM calls using Universal Routing Server (URS) in the same manner as for voice calls. Multiple IM calls can be routed to the same agent. To achieve that, SIP Server distributes
AttributeMediaType set to
5
(
TMediaChat
) in TEvents for IM calls. URS and Stat Server can distinguish
IM calls by this attribute and, according to the configuration, route many calls with IM media to the same agent.
Treatments
SIP Server supports treatments for calls with IM media. However, unlike voice calls, SIP Server does not use Stream Manager to apply treatments. Instead,
SIP Server executes treatments for IM calls itself by using the SIP
MESSAGE request in accordance with the treatment request parameters.
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SIP Server supports the following treatments for the IM calls:
•
PlayAnnouncement
•
CollectDigits
•
PlayAnnouncementAndDigits
The
PlayAnnouncement
treatment is performed by SIP Server when it sends the
MESSAGE
request to the caller. The
MESSAGE
request is sent for each prompt specified in the
TreatmentPlayAnnouncement
request. The content of the
MESSAGE request is created using the
TEXT
parameter in the prompt specified by the
TreatmentPlayAnnouncement
request. When processing the
TreatmentPlayAnnouncement
request for IM calls, SIP Server supports the
TEXT parameter only.
The
CollectDigits
treatment is performed by SIP Server when it receives the
MESSAGE
request from the caller, and it then sends the complete content of the
MESSAGE
request as collected digits to URS. When processing the
TreatmentCollectDigits
request for IM calls, SIP Server supports the
TOTAL_TIMEOUT
parameter only. All other parameters are not supported.
The
PlayAnnouncementAndDigits
treatment is performed by SIP Server by performing
TreatmentPlayAnnouncement
and then
TreatmentCollectDigits
.
Note:
It is recommended that you start a routing strategy with
TreatmentCollectDigits
when applying treatments to an IM call because the strategy can collect the details of the initial IM and store it as
UserData
.
Supervision
SIP Server supports supervision functionality for IM calls, as described in
“Call Supervision” on page 104 . T-Library messaging for supervisor
monitoring scenarios for IM calls is the same as for voice calls, with the only exception that, for IM calls,
AttributeMediaType
is set to
5
(
TMediaChat
).
SIP Server supports the following call supervision modes for IM calls:
•
Silent monitoring
•
Whisper coaching
•
Open supervisor presence
SIP Server supports the following call supervision scopes for IM calls:
•
Agent
•
Call
SIP Server supports the following call supervision types for IM calls:
•
One call
•
All calls
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Silent Monitoring for IM Calls
When silent monitoring is applied to the IM call, SIP Server uses the following algorithm to distribute instant messages:
•
Instant messages sent by a caller or an agent are visible to all participants: the caller, the agent, and the supervisor.
•
Instant messages sent by a supervisor are not visible to the caller or the agent.
Whisper Coaching for IM Calls
When whisper coaching is applied to the IM call, SIP Server uses the following algorithm to distribute instant messages:
•
Instant messages sent by a caller or an agent are visible to all participants: the caller, the agent, and the supervisor.
•
Instant messages sent by a supervisor are visible to the agent only.
Open Supervisor Presence for IM Calls
When open supervisor presence is applied to the IM call, SIP Server uses the following algorithm to distribute instant messages:
•
Instant messages sent by caller or agent are visible to all participants: the caller, the agent, and the supervisor
•
Instant messages sent by supervisor are also visible to the all participants: the caller, the agent, and the supervisor
This mode is the same as the normal IM conference mode.
Multiple Instant Messaging Sessions
SIP Server supports the handling of several simultaneous IM sessions by one agent. The maximum number of simultaneous sessions for an agent is defined by a capacity rule. The agent can also handle one voice call and several IM sessions.
Feature Configuration
Processing UserData
The following options are used to configure how
UserData
is processed:
•
user-data-im-enabled
(see
•
user-data-im-format
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Configuring Microsoft Live Communication Server
Microsoft Office Communicator is the client of Microsoft Live
Communication Server (LCS). There is no direct communication link between
SIP Server and Office Communicator; LCS is the bridge in this scenario. As such, SIP Server is configured to register with LCS, or not to register with
LCS.
In either scenario, there is a common configuration:
•
SIP Server’s IP address must be configured as a trusted server with LCS.
•
The following options must be configured for each DN that is using Office
Communicator:
—The value of this option must be equal to the computer name from the SIP account of Office Communicator.
—The value of this option must be equal to the computer
name from the SIP account of Office Communicator.
—The value must specify the corresponding LCS IP address, the SIP port, and the TCP transport protocol. For example:
192.168.14.121:5060;transport=tcp
SIP Server Registers with LCS
In this scenario, LCS performs presence monitoring using a special account created in LCS. SIP Server registers with LCS using this account and subscribes for the Office Communicator presence statuses on behalf of this account. The
Trunk
DN configuration must contain the force-register
option
(see
).
SIP Server Does Not Register with LCS
The subscribe-presence-from
option is used to specify the SIP URI in the
From:
header of the
SUBSCRIBE
message sent to LCS from SIP Server.
If SIP Server does not register with LCS, this option contains a username in the
SIP URI that is not configured on LCS.
In order for non-LCS users to subscribe to the presence state of LCS users
(such as Office Communicator users), each LCS user must be configured to allow subscriptions to all users, or to a special user that is specified in the
option of the
Trunk
DN that represents LCS. You must add all users or a special user to the
Allow
list in the LCS user configuration.
Configuring the Instant Messaging Solution
There is a number of ways to implement the Instant Messaging solutions in
Genesys. This includes enabling DNs in your contact center to handle instant messages after they arrive at SIP Server. Deploying the Instant Messaging
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Mapping SIP Headers and SDP Messages
SIP Server can now extract data from some incoming SIP messages and map it to either an
Extensions
or
UserData
attribute in T-Library event messages. SIP
Server can map T-Library request attributes (passed in the
TRouteCall message) to SIP parameters in the outgoing
INVITE
message. SIP Server can also map the whole SDP message body, or any particular line in it as
Extensions
or
UserData attributes.
From SIP Messages to T-Library Messages
SIP Server processes SIP messages and can map related data to T-Library event attributes as described in this section. This information becomes further available to other Genesys Framework components. This functionality is supported for following SIP messages:
•
INVITE
•
INFO
•
UPDATE
•
REFER
INVITE Messages
SIP Server can extract data from an incoming
INVITE
message and send the data to Universal Routing Server if the call is made at a Routing Point. Data is retrieved as values from the headers and parameters of the
INVITE
message and then populated into the
Extensions
or
UserData
attributes in the
EventRouteRequest
message. You can configure which headers and parameters to extract data from by creating a section that corresponds to a SIP method name on the
Options
tab of the SIP Server
Application
object.
For example, the
INVITE
section lists the values of the headers and the header parameters that are extracted from the SIP message. The names of the options within this section contain a prefix, a dash (-), and a suffix, in the following format:
<prefix><-><suffix>
The prefix is the name of the TEvent attribute, and begins with either extensions
or
userdata
. The prefix extensions
instructs SIP Server to put the
SIP header or parameter into the
Extensions
attribute. The prefix userdata
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UserData attribute.
The suffix value is a number, and must be unique for all option names containing the same prefix.
The option value determines which header in the header parameter of the SIP message is processed.
The value of the header or header parameter is added as a key-value pair into the attribute, using the form
<header_name>=<header_value>
. Use the colon character (
:
) to address the parameter name of a header. The SIP method
(
INVITE
) must be used instead of the header name if you want to populate the parameter from the SIP Request-line parameter.
Mapping occurs only if both the following conditions are true:
•
The header or header parameter is contained within the incoming
INVITE message.
•
The header or header parameter is configured within the
INVITE
section.
As a result of mapping, the following key-value pair will be created within the
EventRouteRequest
message:
•
The key in the attribute will be equal to the value of the configuration option.
•
The value of the attribute will be equal to the value of the header or header parameter within the SIP
INVITE
message.
INFO and UPDATE Messages
SIP Server can generate an
EventAttachedDataChanged
message if it receives
SIP
INFO
or
UPDATE
messages. The
UserData
attribute in the event and the corresponding call can contain information from the SIP message.
The
INFO
and
UPDATE
configuration sections must be used to configure mapping from these SIP messages to the corresponding T-Library events. The rules for this configuration are the same as the rules for configuring the
INVITE
section.
Only mapping to the
UserData
attribute is supported for the
INFO
and
UPDATE messages. Therefore, the
INFO
and
UPDATE
sections can contain userdata-<n> options, but they may not contain extensions-<n>
options.
REFER Messages
The SIP
REFER
method provides single-step transfer functionality. SIP Server retrieves data from headers and parameters of the
REFER
message, and then populates it into the
Extensions
or
UserData
attributes in all events associated with a transfer transaction. This feature also enables Genesys Voice Platform
(GVP) to transfer a call to Genesys Framework with attached data.
When a
REFER
message arrives, SIP Server analyzes the user part of the
REFER
TO
URI to determine if the destination of the call is an internal DN or an
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REFER
section of SIP Server
Application
object. If such mappings exist, SIP Server extracts the values of the appropriate SIP headers into the attribute
Extensions
or
UserData,
according to the configuration. If the destination is a Routing Point, after this processing, SIP Server generates an
EventRouteRequest
message containing the necessary attribute values.
You can configure which headers and parameters to extract data from by creating the
REFER
section on the
Options
tab of the SIP Server
Application object. The rules for this configuration are the same as the rules for configuring the
INVITE
).
To map data from the
REFER message into the
Extensions
and/or
UserData attributes, configure userdata-<n>
options and extensions-<n>
option in the
REFER
section.
It is possible to configure SIP Server to extract data from custom headers added in the
REFER
message, and to process the data from the custom SIP headers.
Known Limitations
This feature is applicable only to scenarios where a call is made to a Routing
Point using the
REFER
method. To pass attached data in other scenarios, use the mapping configuration of the
INFO
and
UPDATE
messages instead.
From T-Library Messages to SIP Messages
SIP Server can map headers or header parameters passed in the
RequestRouteCall
message to the outgoing
INVITE
message that is sent as a result of the call routing process. There are two ways to specify the values of the headers or header parameters to be mapped:
•
Use the extensions-<n>
SIP Server configuration option. This method maps only headers.
•
Use the
SIP_HEADERS
and
SIP_REQUEST_PARAMETERS
extension of the
RequestRouteCall
message.
Both methods can work simultaneously; for example—you can create a mapping list using extensions-<n>
options in the SIP Server configuration, and also specify the names in the
SIP_HEADERS
extension of the
TRouteCall
request.
Using the extensions-<n> Option
To use extensions-<n>
to configure T-Library request attributes to the SIP mapping, you configure extensions-<n>
options in a SIP Server
Application object using Configuration Manager. Those options are specified in the section named after the SIP request used to route the call, which is
INVITE
.
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Configuration Example
This example demonstrates how to map a
TRouteCall
request extension called
InfoToSendInInvite
to the outgoing
INVITE
message.
The following steps show how to configure SIP Server:
1. Create the
INVITE
section on the
Options
tab of the SIP Server
Application object.
2. In the
INVITE
section, create an option named extensions-1,
and set the value to
InfoToSendInInvite
.
Note:
This configuration example is based on the assumption that this is the first extensions-<n>
option in the
INVITE
section.
SIP Server uses this configuration when it receives a
TRouteCall
request from the URS with either
InfoToSendInInvite
or
InfoToSendInRefer
extensions defined.
The following log excerpt provides the details: message RequestRouteCall
AttributeThisDN
AttributeConnID
AttributeOtherDN
AttributeExtensions
'InfoToSendInInvite’
AttributeDNIS
AttributeRouteType
'5000'
006e01886c3d7001
'21101'
[371] 00 0B 00 00..
‘INVITE from SIP Server'
'5000'
1 (RouteTypeDefault)
AttributeReferenceID 9
SIP Server adds a new header
InfoToSendInInvite
to the outgoing
INVITE message:
INVITE sip:21101@DestinationHost:21101 SIP/2.0
From: <sip:7102@SourceHost:7102>;tag=28B10B44
To: <sip:21101@ DestinationHost >
Call-ID: 931E620E-F3F9-4D72-A451-36B1BB259532-1
CSeq: 1 INVITE
Content-Length: 145
Content-Type: application/sdp
Contact: <sip: SourceHost:5060>
InfoToSendInInvite: INVITE from SIP Server
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: timer
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Using SIP_HEADERS and
SIP_REQUEST_PARAMETERS
This method does not require any changes to the configuration of the SIP
Server
Application
object. In this case, all mapping information is provided in the
TRouteCall
request that is created by the URS routing strategy and sent to
SIP Server.
The
TRouteCall
request should contain two specialized extensions to trigger
T-Library-to-SIP mapping in SIP Server:
•
SIP_HEADERS
—Contains a list of extension names of
TRouteCall
to be mapped to the outgoing SIP message as headers.
•
SIP_REQUEST_PARAMETERS
—Contains a list of extension names to be mapped to the outgoing SIP message as Request URI parameters.
The values of headers and header parameters must be also specified in the
TRouteCall
request.
Example
This example demonstrates how mapping works in SIP Server when it receives the following
TRouteCall
request: message RequestRouteCall
AttributeThisDN
AttributeConnID
AttributeOtherDN
AttributeExtensions
'SIP_HEADERS'
'SIP_REQUEST_PARAMETERS'
'hdr-host1'
'hdr-host2’
'prm-host1'
'prm-host2'
AttributeDNIS
AttributeRouteType
AttributeReferenceID
'5000'
006e01886c3d7001
'21101'
[371] 00 0B 00 00..
'hdr-host1,hdr-host2’
'prm-host1,prm-host2'
'host1'
'host2'
‘local1'
‘local2'
'5000'
1 (RouteTypeDefault)
9
This message contains both
SIP_HEADERS
and
SIP_REQUEST_PARAMETERS extensions, which means that both new headers and new Request URI parameters should be added to the outgoing SIP message:
INVITE sip:21101@ DestinationHost:21101;prm-host1=local1; prm-host2=local2 SIP/2.0
From: <sip:7102@ SourceHost:7102>;tag=28B10B44
To: <sip:21101@ DestinationHost >
Call-ID: 931E620E-F3F9
CSeq: 1 INVITE
Content-Length: 145
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Content-Type: application/sdp
Contact: <sip: SourceHost > hdr-host1: host1 hdr-host2: host2
Max-Forwards: 70
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: timer
SDP Message Mapping
SIP Server can map the whole SDP message body, or any particular line in it as
AttributeExtensions
or
AttributeUserData
. Configuring this type of mapping is similar to configuring mapping of SIP Server headers or parameters. The option names in the
INVITE
section must use the same rules described for SIP
Server message mapping; however, the option name must be
SDP,
and it must be followed by a colon and a letter that indicates the type of mapped SDP message. For example, the userdata-1=SDP:m
option is mapped using the following parameters from
AttributeUserData
: m=audio 18234 RTP/AVP 8 101
Note:
If the
AttributeUserData
parameter in the
EventRouteRequest
message contains a pair such as:
SDP:m audio 18234 RTP/AVP 8 101
You must use the
SDP
value to configure the mapping of the whole SDP message.
Music and Announcements
SIP Server is able to control the playing of announcements by using Stream
Manager. Stream Manager plays files using a codec negotiated with the SIP
Server switch. The list of possible codecs is configured in SIP Server with the audio-codecs
option
(
)
. For information about how to configure
Stream Manager refer to “Configuring Stream Manager” on page 94
.
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Announcement Treatments on Routing Points
The two announcement treatments,
PlayAnnouncement
and
PlayAnnouncementAndDigits,
include the parameters listed in Table 15
.
Table 15: Announcement Treatment Parameters
Parameter
LANGUAGE
MSGID
MSGTXT
PROMPT
INTERRUPTABLE
ID
DIGITS
USER_ID
USER_ANN_ID
TEXT
Description
Ignored.
Ignored.
Ignored.
Contains up to 10 subprompts. Each contains a music file, and these are played in order.
When this check box is selected, the caller can interrupt the announcement with a
DTMF
keystroke.
Contains an integer, that refers to the Stream Manager announcement/<integer>
file. For example, the value
1 refers to the file announcement/1_alaw.wav
, if the G.711
A-law codec is used.
Ignored.
Supported by Stream Manager using the users/<customer id>_<ann.id>
file.
Supported by Stream Manager using the users/<customer id>_<ann.id>
file.
Ignored.
Music-on-hold is not available for these treatment types. Stream Manager is the only source for the announcements.
If the treatment is terminated early because of a problem with Stream Manager or SIP Server, SIP Server sets the
Extension
data fields
ERR_CODE and
ERR_TEXT.
To determine whether these fields and their values exist, from a routing strategy, use the function
ExtensionData
. Place this function on a normal completion branch (not the error branch) after the treatment.
Refer to the Universal Routing 7.6 Reference Manual for more information on the use and configuration of strategies.
Note:
Leave the
Wait For Treatment End check box selected, to allow the treatment to play until completion.
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Music Treatments on Routing Points
describes the parameters for music treatments:
Table 16: Music Treatment Parameters
Parameter
MUSIC_DN
DURATION
Description
Specifies the music source that Stream Manager plays. The format is:
<directory>/<music file name>
Where
<directory>
is a sub-directory of the Stream Manager root directory, and
<music file name> refers to the name of the file— without the codec extension. For example, music/in_queue
refers to the file music/in_queue_alaw.wav if the G.711 A-law codec is used.
To specify the number of repetitions, the parameter repeat=<N>
must be used, where
<N>
is any positive integer. If no repetition is specified, the music file loops forever. The valid formats are:
•
<directory>/<music file name>
—The specified file loops endlessly.
•
<directory>/<music file name>;repeat=<N>
—The specified file is repeated
<N>
times.
option is used if the value of the
MUSIC_DN
parameter is not specified.
Specifies the duration of the music (in seconds).
Note: This parameter is ignored if
MUSIC_DN
is blank.
This treatment ends before music is played. To continue playing music after the treatment terminates, consider creating one of the following strategies:
• Execute the treatment inside a route-selection treatment block. In this case, the treatment continues until a route target is selected.
• Follow the treatment with the
SuspendForTreatmentEnd
function. In this case, the treatment plays music until terminated after the delay specified in option
DURATION
.
• Follow the treatment with the delay function. In this case, the treatment plays music for the period specified in option delay
. If
DURATION
is less than delay
, silence is played for the time difference.
Refer to the Universal Routing 7.6 Reference Manual for more information on use and configuration of strategies.
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Other Treatments on Routing Points
These treatments in
Table 17 continuously loop a pre-defined audio file to a
call. You must configure Stream Manager to use these treatments.
Music-onhold
cannot be the source of the audio file. The treatment types are as follows:
Table 17: Other Treatments on Routing Points
Treatment Type
Busy
Fast Busy
Silence
Ringback
CollectDigits
RecordUser-
Announcement
Description
Plays a busy tone. To define the busy tone audio file, configure the SIP Server
Plays a fast busy tone. To define the fast busy tone audio file, configure the SIP Server
option (
Plays no sound. To define the silence
audio file, configure the SIP Server
(
Plays ringback tone. To define the ringback
audio file, configure the SIP Server
Collects customer-entered digits.
Records a user’s announcement and saves into a users folder.
Refer to the Universal Routing 7.6 Reference Manual for more information about the use and configuration of strategies.
No-Answer Supervision
This section describes SIP Server’s No-Answer Supervision feature and its configuration.
Business and Private Calls
No-Answer Supervision can be applied to business and private calls.
Business Calls
SIP Server automatically categorizes as a business call any call distributed to an agent either from a Queue or from a Routing Point. Use the following configuration options to define what additional calls to or from an agent are classified as business calls:
•
inbound-bsns-calls
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•
outbound-bsns-calls
)
•
inherit-bsns-type
(
•
internal-bsns-calls
)
•
unknown-bsns-calls
(
)
Private Calls
SIP Server categorizes as a private call any call that does not fall into the business or work-related categories. SIP Server does not apply any automatic business-call handling after a private call. If an agent receives a direct private call while in wrap-up or legal-guard time, the emulated wrap-up or legal-guard timer is not interrupted.
You can apply No-Answer Supervision to private calls, using the configuration option nas-private
Agent No-Answer Supervision
This feature provides the following functionality:
•
If an agent does not answer a call within a specified timeout, SIP Server can divert the call to a sequence of overflow destinations. Alternatively, you can configure SIP Server to return calls automatically to the last distribution device.
•
If an agent fails to answer a call within a specified timeout, you can configure SIP Server to either log out the agent or set the agent to
NotReady to prevent further calls from arriving.
Configuration Options
SIP Server provides the following configuration options for defining the behavior of the Agent No-Answer Supervision feature:
•
agent-no-answer-action ( page 199
•
agent-no-answer-overflow ( page 199
)
•
agent-no-answer-timeout (
)
•
nas-private (
Extension No-Answer Supervision
The No-Answer Supervision feature includes devices of type
Extension.
If a call is not answered on an extension within a specified timeout, SIP Server can divert the call to a sequence of overflow destinations. Alternatively, you can configure SIP Server to return calls automatically to the last distribution device.
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Configuration Options
SIP Server provides the following configuration options for defining the behavior of No-Answer Supervision with devices of type
Extension
:
•
extn-no-answer-overflow (
)
•
extn-no-answer-timeout
Position No-Answer Supervision
The No-Answer Supervision feature includes devices of type
ACD Position.
If a call is not answered on a position within a specified timeout, SIP Server can divert the call to a sequence of overflow destinations. Alternatively, you can configure SIP Server to return calls automatically to the last distribution device.
Configuration Options
SIP Server provides two configuration options for defining the behavior of No-
Answer Supervision with devices of type
ACD Position
:
•
posn-no-answer-overflow (
)
•
posn-no-answer-timeout ( page 218
Device-Specific Overrides
SIP Server provides three configuration options with which you can configure device-specific overrides for individual devices. You set the values for these options in the
TServer
section on the
Annex
tab of the individual device. The options are:
•
no-answer-action
)
•
no-answer-overflow
(
)
•
no-answer-timeout
(
Extension Attributes for Overrides for Individual Calls
For all of the No-Answer Supervision options, you can specify the corresponding
Extension
attribute in the
TRouteCall
request, to override the configured value for individual calls. This method allows the no-answer behavior to be determined in a routing strategy. The three extensions are:
•
NO_ANSWER_TIMEOUT
•
NO_ANSWER_OVERFLOW
•
NO_ANSWER_ACTION
See “Using the Extensions Attribute” on page 186 for more information.
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Feature Limitations
No-Answer Supervision functionality will not be activated if the divert-onringing
configuration option is set to false
.
Personal Greeting
Personal greeting functionality enables Stream Manager to play a media file to a customer and an agent when the agent answers the call. It is possible to play the same file or different files to the customer and agent.
Feature Configuration
To configure personal greetings, follow the provided procedure using
Configuration Manager.
Procedure:
Configuring a personal greeting
Start of procedure
1. Under a configured
Switch object, select the
Agent Logins folder. From the
File
menu, select
New > Agent Login to create a new
Agent Login
object.
2. In the
New Agent Login Properties
dialog box, on the
General
tab, specify the Agent Login name in the
Code
text box.
3. Click the
Annex
tab.
4. Click or create a section named
TServer.
In the
TServer
section, create options as specified in
.
Table 18: Configuring a Personal Greeting
Option Name Option Value Description
A file name
A file name
Specifies the name of the media file that will be used as a greeting for the agent.
See the option description on page 228
.
Specifies the name of the media file that will be used as a greeting for the customer. The customer greeting plays continuously until the agent greeting finishes playing.
See the option description on page 230
.
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5. When you are finished, click
Apply
.
End of procedure
Additional Information
You can also attach these values from a routing strategy. The agent-greeting and customer-greeting keys enable personal greetings. These keys are processed by SIP Server using the
AttributeExtension
parameters of the
TRouteCall
request.
Notes:
The keys that are contained in attribute
Extension
have a higher priority than the options specified in the
Agent Login
object.
The customer greeting plays continuously until the agent greeting finishes playing.
Presence Subscription
Presence is an indicator of an agent’s status regarding possible communication.
An agent’s client application provides presence information (or state) to SIP
Server, which distributes this information to it’s clients. SIP Server supports
“accepting” presence subscriptions and “subscribing” presence subscriptions.
These subscriptions are used with DNs of type
Extension and DNs of type
Routing Point
.
When subscribing to DNs of type
Extension,
SIP Server notifies subscribers that the status of the endpoint is open
when that endpoint registers with SIP
Server. SIP Server will notify any subscribers that the status of the endpoint is closed when the SIP endpoint is not registered or when the registration has expired.
When subscribing to DNs of type
Routing Point,
SIP Server always notifies subscribers that the status of the endpoint is open
.
Feature Configuration
Procedure:
Enabling presence subscription
Purpose: To enable presence subscription for a particular DN. This procedure can also be used when subscribing to the presence of any endpoints that are located behind any presence agent or a softswitch supporting presence notifications.
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Start of procedure
1. Create a DN of type
Trunk in Configuration Manager. Parameters for all presence subscriptions from the SIP Server to a particular softswitch are configured in this
Trunk
DN.
2. Configure these options in the
TServer
section on the
Annex
tab of the
Trunk
DN:
•
contact
•
subscribe-presence-domain
(
)
•
subscribe-presence-from
(
)
•
subscribe-presence-expire
(
)
3. Create a DN of type
Extension in Configuration Manager.
4. Configure these options in the
TServer
section on the
Annex
tab of the
Extension
DN:
•
contact
•
request-uri
(
)
•
subscribe-presence
)
•
enable-agentlogin-presence
)
5. Create an
Agent Login
object in Configuration Manager for each DN that will have subscription enabled. The
Agent Login
name must be equal to the
DN
object name. Each
Agent Login
object must be associated with an agent.
End of procedure
Note:
Any internal calls that are made with this softswitch will not be monitored by SIP Server, and the agent state will be changed by SIP
Server to
Not Ready
.
Updating Agent State
SIP Server changes the agent state in response to any notifications about presence state changes by using the
PUBLISH
request method. SIP Server accepts the
PUBLISH
request and provides automatic agent state updates based on any presence updates received within the
PUBLISH
request. SIP Server distributes notifications about presence updates to all subscribers based on the presence update received within the
PUBLISH
request.
SIP Server accepts
PUBLISH
requests when they are received for a DN in an internal domain. SIP Server processes the presence update from the
PUBLISH request and distributes presence update notification to all subscribers for this
DN.
The
PUBLISH
request functionality is enabled at the DN level in Configuration
Manager by specifying the subscribe-presence
option ( page 246 ). The value
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to indicate that presence change notifications are issued from the
PUBLISH
request.
Note:
This functionality has been verified with the Eyebeam SIP endpoint when it is configured to work in
Presence Agent
mode. This mode enables
PUBLISH
request processing.
SIP Server updates the agent state when the agent login name matches the DN name. Agent updates are processed as follows:
•
When SIP Server receives a presence notification with an open
status, it performs the following steps:
Confirms if the agent is logged in. If the agent is not logged in, SIP
Server sends an
EventAgentLogin
message.
Confirms if any activity is indicated in the presence notification.
If there is no activity, and if the agent is in a
NotReady
state, SIP
Server sends an
EventAgentReady
message.
If there is activity, and if the agent is in a
Ready
state, SIP Server sends an
EventAgentNotReady
message and attaches the activity from the presence notification as the
ReasonCode
attribute.
•
When SIP Server receives a presence notification with a closed
status, it confirms that the agent is logged in. If the agent is logged in, SIP Server then sends an
EventAgentLogout
message.
•
All notifications about the changes of an agent state are ignored when the agent is in the
NotReady
(
AfterCallWork
) state. The requested agent state is applied when the ACW time is over. For example, if an agent completes the call, SIP Server transfers the agent into the
ACW
state, and the
PUBLISH request with an open
status comes from the agent’s SIP phone, then SIP
Server does not change the agent state immediately. It waits for the ACW time to expire, and then places this agent into the
Ready
state. If in the same scenario SIP Server receives the
PUBLISH
request with a busy
status from the agent’s SIP phone, SIP Server will not change the agent state until the
ACW timer is over, meaning that the agent remains in the
NotReady
state.
Notes:
When configuring presence subscription for Microsoft’s Live
Communication Server (LCS), you must also configure the
Trunk
DN
as specified in “Remote Server Registration” on page 166 .
When Microsoft’s Office Communicator is integrated with Microsoft
Outlook and the presence state is set to
In a Meeting
or
Vacation,
Microsoft’s Live Communications Server sends a presence notification with the
Busy
presence state to SIP Server. However, SIP Server is unable to provide
In a Meeting
or
Vacation
presence states to any subscriber. Instead, a notification with the
Busy
presence state is generated by SIP Server.
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Preview Interactions
Preview interactions allow agents to preview desktop interactions before receiving a call. SIP Server sends Preview Interaction messages to the desktop applications using the
EventPrivateInfo
message. The desktop application sends preview interaction messages using the
TPrivateService
request.
SIP Server sends a previewInteractionRequest
message to the desktop application when it receives a
RouteCall
request to a DN that is configured with the
option (
) set to true
.
The desktop application responds with a previewInteractionResponse
message to SIP Server. The previewInteractionResponse
message provides SIP Server with information regarding the agent’s ability to process the incoming interaction. The status
field contains an accepted
value or a rejected
value that specifies if the agent will accept the interaction.
SIP Server sends a previewInteractionAcknowledge
message to the desktop application after it receives the previewInteractionResponse
message from it.
This message informs the desktop application that the previewInteractionResponse
message was successfully processed by SIP
Server.
The previewInteractionCancel
message is sent by SIP Server to an application in the following scenarios if there was an unsuccessful completion of a preview interaction:
•
The preview timeout expired. SIP Server sends the previewInteractionCancel
message with the status
field set to expired
to an application when the previewInteractionRequest
message was issued but SIP Server did not receive a previewInteractionResponse
message
within the specified timeout value for the preview-expired
).
•
The call was abandoned. SIP Server sends the previewInteractionCancel message to an application with the status
field set to canceled
.
Providing Call Participant Info
SIP Server can distribute information about all call participants—except the trunks allocated for communication between SIP Servers, distribution devices
(such as Routing Points or ACD), and supervisors’ DNs—to logged-in agents by using the SIP
NOTIFY
method and
EventUserEvent
messages.
The information about the call participants is reported in the
Extensions attribute of the relevant event using the following key-value pairs:
•
LCTPartiesLength
—An integer that specifies how many parties are involved in a single call.
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Chapter 6: SIP Server Feature Support Providing a Caller ID
•
LCTParty<n>
—An integer that represents a party of the call, where n is an integer value starting from 0.
Feature Configuration
provides an overview of the steps required to configure call info for agents.
Table 19: Task Flow—Configuring Call Info for Agents
Objective
1. Configure the
Trunk
DNs.
2. Configure the SIP
Server
Application object.
Related Procedures and Actions
Set the
option to true for DNs of type
Trunk
that are allocated for direct signaling between SIP Servers. The
NOTIFY method will be sent only to sessions that are established through such trunks.
For more information, see “Trunk Optimization for Multi-Site Transfers” on page 166 .
option to true
.
Providing a Caller ID
SIP Server supports providing caller ID information that is displayed on a destination party’s phone, and replacing the caller ID with another number if necessary. This feature is supported using either of the following methods:
•
The
Extensions
attribute with the
CPNDigits
key in the following messages:
TMakeCall, TMakePredicitiveCall, TInitiateConference
, and
TInitiateTransfer
.
If the
CPNDigits
key is set in the
Extensions
attribute, the value of this key overrides the username provided in the URI in the
From
header of the
INVITE
message. This
Extension
is not applicable when performing a
TMakeCall
request using the
REFER
method.
•
The
Trunk
DN level. In this case, the caller ID information will be replaced by the SIP URI setting in this option for all outgoing calls through this
Trunk
DN.
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Chapter 6: SIP Server Feature Support Remote Server Registration
Remote Server Registration
SIP Server supports registering with a remote server under a specified account.
The remote server registration is enabled on a per-
Trunk
DN basis. SIP Server registers
Trunk
DNs at a remote server when the force-register
option
SIP Server also uses the values of the following options when registering with a remote server:
•
contact
, when determining where to send the
REGISTER
request.
•
password
, when the
REGISTER
request is challenged.
See “Configuring endpoints” on page 81 for more information about this
options.
Remote Talk
The Remote Talk feature enables the answering of an incoming call remotely by a T-Library client, by sending the
TAnswerCall
request to SIP Server. For
this feature to work, the sip-cti-control
talk
(see
)
.
The SIP endpoint must support the BroadSoft Application Server interface in order to use the Remote Talk feature for remote call control.
Trunk Capacity Support
SIP Server enables control of the number of calls to be handled by a specific
Voice over IP device represented in the SIP Server configuration as
Trunk.
The following DN-level configuration options support this feature:
. See the option descriptions and configuration examples on page 229
.
Trunk Optimization for Multi-Site Transfers
SIP Server supports trunk optimization for multi-site transfers. When the trunk optimization functionality is in use, the
OtherDN
attribute contains correct information and is reported properly in
EventPartyChanged
messages in the following scenarios:
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Customer
Endpoint
Trunk Optimization for Multi-Site Transfers
Scenario 1
Figures 22 and 23 show the state of the call before and after the multi-site
transfer.
SIP signaling 1
SIP Server 1
SIP signaling 2
SIP Server 2
Gateway
SIP signaling 3
SIP signaling
SIP signaling 3
Agent B
Endpoint
Figure 22: Call Before REFER with Replaces Transfer
Agent A
Endpoint
1. An inbound call is routed to Agent A at the SIP Server 2 site.
2. Agent A initiates a two-step transfer to Agent B at the SIP Server 1 site.
In this scenario, SIP Server uses a SIP
REFER
request with the
Replaces
header to report call data for Agent B.
SIP signaling 1
SIP Server 1
Customer
Endpoint
Gateway
SIP signaling
Agent B
Endpoint
Figure 23: Call After REFER with Replaces Transfer
After the transfer is completed, both the transferring agent (Agent 1) and secondary SIP Server (SIP Server 2) are released from the call.
Scenario 2
Figures 24 and 25 show the state of the call before and after the multi-site
transfer.
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Chapter 6: SIP Server Feature Support
Customer
Endpoint
Trunk Optimization for Multi-Site Transfers
Gateway
SIP signaling 1
SIP Server 1
SIP signaling 2
SIP signaling
SIP Server 2
SIP signaling 3
Agent A
Endpoint
Figure 24: Call Before INVITE with Replaces Transfer
Agent B
Endpoint
1. An inbound call is routed to Agent A at the SIP Server 1 site.
2. Agent A initiates a two-step transfer to Agent B at the SIP Server 2 site.
In this scenario, SIP Server uses a SIP
INVITE
request with the
Replaces
header to report call data for Agent B.
SIP signaling 1
SIP Server 1
SIP signaling 2
SIP Server 2
Customer
Endpoint
Gateway
SIP signaling
SIP signaling 3
Agent B
Endpoint
Figure 25: Call After INVITE with Replaces Transfer
In this case, the consultation call between the agents are merged on
SIP Server 1, with user data propagated to the destination SIP Server
(SIP Server 2). After the transfer is completed, SIP Server 1 remains in the signaling path—only the transferring agent (Agent A) is released from the call.
Scenario 3
Figures 26 and 27 show the state of the call before and after the multi-site
transfer.
168 Framework 7.6
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SIP Server 1
SIP signaling 1
SIP Server 2
SIP signaling 2
SIP Server 3
SIP signaling
SIP signaling
SIP signaling
Customer
Endpoint
Gateway
Agent B
Endpoint
Agent C
Endpoint
Figure 26: Call Before INVITE with Replaces Transfer
1. From a SIP Server 1 site, a call arrives to Agent B at the SIP Server 2 site.
2. Agent A initiates a two-step transfer to Agent C at the SIP Server 3 site.
In this scenario, SIP Server uses a SIP
INVITE
request with the
Replaces
header to report call data for Agent C.
SIP Server 1
SIP signaling
SIP Server 3
SIP signaling
SIP Server 2
SIP signaling
Customer
Endpoint
Gateway
Figure 27: Call After INVITE with Replaces Transfer
Agent C
Endpoint
After the transfer is completed, SIP Server 2 is removed from the signaling path. An
EventPartyChanged
message is generated for Agent C on SIP Server 3, based on information received in the
INVITE
request with the
Replaces
header.
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Chapter 6: SIP Server Feature Support Video Support
Feature Configuration
provides an overview of the main steps required to configure trunk optimization.
Table 20: Task Flow—Configuring Trunk Optimization
Objective
1. Create
Trunk
DNs.
2. Configure the SIP Server
Application
object.
Related Procedures and Actions
In each SIP Server configuration (origination and destination), in the corresponding SIP
Switch, configure a DN of type
Trunk
. These
Trunk
DNs will be used for direct signaling between SIP Servers.
For each
Trunk
DN, in the
TServer
section of the
Annex
tab, configure the following options:
•
—Set this option to true
(see
).
•
true
).
•
—Set this option to true
).
In multi-site routing, to avoid reporting an access resource as
AttributeOtherDN
in related events, in the extrouter
section of the
SIP Server
Application
option to an ISCC direct transaction type
(such as direct-uui
).
Feature Limitation
This functionality requires direct signaling (no media gateways or session border controllers) between any two SIP Server instances, with no alteration of the SIP attributes (
CALL-ID
, to
header, from
header) as these are used for unique call context matching.
Video Support
SIP Server supports the following scenarios related to Video Call functionality:
•
Push Video
•
Video Call on Hold
•
Video Call Transfer
•
Video Call Treatment
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•
Outbound Video Call
Push Video
Video Support
Push Video functionality enables a person to play a video file to another call participant during a call. SIP Server can support video streams using Stream
Manager and T-Library functions.
Start Video
To start playing a video file, SIP Server uses the
TSingleStepConference function to push video from an agent to a customer. This function must contain the following attributes:
•
OtherDN
—Represents a video source. It is always defined as the gcti::video
string.
•
Extensions
—Must contain the following key-value pairs:
VideoFile
—A string that contains the name of the video file that will be played for the customer. If this key-value pair is not specified, the default video file will be played. The default video file is configured in the SIP Server
Application
object, using the
configuration option (
AgentVideo
—A string that identifies the origin of the video stream played to the agent. The values are as follows: from-third-party
—The agent receives video from a third party—that is, the party that participated in the call before the operation started.
to-third-party
—The agent receives the same video stream as played to the third party—that is, video from the file specified by the
VideoFile
parameter. (The from-video-file value can be used as an alias.)
In either case, both the customer and agent hear each other and the audio that comes with the video file. The customer, the agent, and the audio source from the video file are three participants in the audio conference. When the pushed video ends, the customer and the agent continue a regular two-party conversation.
If the
AgentVideo
key is not specified, or if it is empty, the to-third-party
value will be used.
Stop Video
There are several ways to stop playing a video file:
•
By deleting a party from a conference
•
By releasing the gcti::video
device
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Chapter 6: SIP Server Feature Support Video Support
•
When the video file ends
Deleting a Conference Call
SIP Server uses the
TDeleteFromConference
function to stop a video stream from a conference call. In this scenario, the
OtherDN
attribute is always defined as the gcti::video
string.
Releasing the Device
SIP Server uses the
TReleaseCall
function to stop a video stream by releasing the gcti::video
device. In this scenario, the
ThisDN
attribute is always defined as the gcti::video
string.
When the File Ends
Stream Manager will end the SIP dialog for the gcti::video
device when the video file ends, but it will not end the other SIP dialogs that belong to the conference.
Other Supported Scenarios
Video Call on Hold
The video call can be put on hold by using the
THoldCall
function. Stream
Manager analyzes the endpoint capabilities submitted inside the endpoint’s
SDP message, and when supported, plays a video file.
Video Call Transfer
SIP Server supports a video call transfer by providing a regular offer/answer
SDP message exchange between an endpoint during 1pcc operation.
Video Call Treatment
SIP Server supports a video call treatment in which a video file can be played to a customer when their call is on a Routing Point. The treatment prompts can be defined in a URS strategy that points to video files. For the video calls,
Stream Manager plays both video and audio when a video prompt is specified.
Outbound Video Call
An agent can initiate an outbound video call using the
TMakeCall
function. In
the SIP Server configuration, the refer-enabled
true,
or the
option must be set to
1
. The
INVITE
message to an
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Chapter 6: SIP Server Feature Support Video Support external destination will contain SDP information with the agent’s endpoint video capabilities.
When the agent initiates an outbound video call, and a recipient accepts it, the video file starts playing. If the recipient’s endpoint does not have video capabilities or refuses the video connection, only an audio connection— without the video—is established.
Feature Configuration
provides an overview of the main steps required to configure video support.
Table 21: Task Flow—Configuring Video Support
Objective Related Procedures and Actions
1. Install a PC video camera.
Follow the instructions in the video camera documentation.
2. Configure a SIP endpoint to support video functionality.
Follow the instructions specific to the SIP endpoint you are using.
If using the Genesys SIP Endpoint, launch the
Audio and Video Tuning Wizard. Complete the wizard steps, and select the installed video camera on the corresponding wizard page.
3. Configure a SIP Server
Application
object.
4. Configure a device.
gcti::video
In the SIP Server
Application
object, in the
TServer
section on the
Options
tab, specify
option.This option contains the name of the video file that is played to the caller if a single-step conference to the gcti::video device does not contain a
VideoFile
key in the
Extensions
attribute.
For Push Video, complete the following procedure:
•
Configuring a gcti::video device, page 174
5. Configure a video service.
For Push Video, complete the following procedure:
•
Configuring a video service, page 174
Follow the provided procedures using Configuration Manager.
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Chapter 6: SIP Server Feature Support Video Support
Procedure:
Configuring a gcti::video device
Start of procedure
1. Under a configured
Switch
object, select the
DNs
folder. From the main menu, select
File > New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, click the
General
tab, and then specify the following properties:
a.
Number
: Enter gcti::video
.
b.
Type
: Select
Trunk
from the drop-down box.
3. When you are finished, click
Apply
.
End of procedure
Next Steps
•
Procedure:
Configuring a video service
Start of procedure
1. Under a configured
Switch object, select the
DNs
folder. From the main menu, select
File > New > DN
to create a new
DN
object.
2. In the
New DN Properties
dialog box, click the
General
tab, and then specify the following properties:
a.
Number
: Enter the DN name. This name is currently not used for any messaging, but it must still be unique.
b.
Type
: Select
Voice over IP Service
from the drop-down box.
3. Click the
Annex
tab, and create the
TServer
section. Under the
TServer section, set the options as specified in
.
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Table 22: Configuring a Video Service
Option Name Option Values Description
SIP URI
SIP URI video
Specifies the value using the Stream
Manager application settings in the following format:
IP address::SIP port
Specifies the value to be used as a template for the source of the video stream and as the value of the
Request-URI parameter in the
INVITE
message: annc@<stream_manager_hostport>;play=
<file>);repeat=<number>)
Set this option to video.
4. When you are finished, click
Apply
.
End of procedure
Feature Limitations
The Video Conference functionality is not supported.
Genesys Voice Platform Integration
For detailed information about SIP Server integration with the Genesys Voice
Platform (GVP), see the following documents:
•
The Genesys 7.5 GVP–SIP Server Integration Guide—This guide provides an overview of the GVP–SIP Server integration in its various modes—In-
Front, Behind, and Stand-Alone—as well as the relevant procedures for completing the integration. This document applies to the 7.5 release of SIP
Server, and the 7.5 and 7.6 releases of GVP.
•
The Genesys 8.0 Voice Platform Solution 8.0 Integration Guide—This guide provides an overview of the Voice Platform Solution (VPS), with the aim of integrating the various components that make up the solution. This document applies to the 7.6 release of SIP Server and the 8.0 release of
GVP.
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Chapter 6: SIP Server Feature Support Genesys Voice Platform Integration
176 Framework 7.6
Chapter
7
T-Library Functionality
Support
z z z z
This chapter describes the T-Library functionality that SIP Server supports. It contains the following sections:
Using T-Library Functions, page 177
Using the Extensions Attribute, page 186
Using T-Library Functions
presents the T-Library functionality supported in SIP Server. The table entries use these notations:
N—Not supported
Y—Supported
E—Event only is supported
I—Supported, but reserved for Genesys Engineering
In Table 23 , when a set of events is sent in response to a single request, the
events are listed in an arbitrary order. An asterisk (*) indicates the event that contains the same
Reference ID
as the request. For more information, refer to the Genesys 7 Events and Models Reference Manual and the Voice Platform
SDK 7.6 .NET (or Java) API Reference for complete information on the
T-Server events, call models, and requests.
reflects only the switch functionality that Genesys software supports and might not include the complete set of events that the switch offers.
Certain requests listed in
are reserved for Genesys Engineering and are listed here merely for completeness of information.
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Chapter 7: T-Library Functionality Support Using T-Library Functions
Table 23: Supported Functionality
Feature Request
TOpenServer
TOpenServerEx
TCloseServer
TSetInputMask
TDispatch
TScanServer
TScanServerEx
TRegisterAddress a
TUnregisterAddress a
TMakeCall b
TAnswerCall
TReleaseCall
TClearCall
THoldCall
TRetrieveCall
TRedirectCall
TMakePredictiveCall d
Request Subtype Corresponding Event(s) Supported
General Requests
EventServerConnected
EventServerConnected
EventServerDisconnected
EventACK
Y
Y
Y
Y
Y
Y
Y
Not Applicable
Not Applicable
Not Applicable
Registration Requests
EventRegistered
EventUnregistered
Call-Handling Requests
EventDialing Regular
DirectAgent
SupervisorAssist
Priority
DirectPriority
Y
Y
Y
N
N
N
N
Y c
Y
EventEstablished
EventReleased
EventReleased
EventHeld
EventRetrieved
EventReleased
EventDialing*,
EventQueued
N
Y
Y
Y
Y
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Table 23: Supported Functionality (Continued)
Feature Request
TInitiateTransfer b
TCompleteTransfer
TInitiateConference b
TCompleteConference
TDeleteFromConference
TReconnectCall
TAlternateCall
TMergeCalls
TMuteTransfer b
TSingleStepTransfer b
TSingleStepConference
Request Subtype Corresponding Event(s) Supported
Transfer/Conference Requests
EventHeld, EventDialing*
EventReleased*,
EventPartyChanged
EventHeld, EventDialing*
EventReleased*,
EventRetrieved,
EventPartyChanged,
EventPartyAdded
Y
Y
Y
Y
Y
ForTransfer
ForConference
EventPartyDeleted*,
EventReleased
EventReleased,
EventRetrieved*
EventHeld*, EventRetrieved
EventHeld, EventReleased*,
EventRetrieved
EventPartyChanged
EventHeld, EventReleased*,
EventRetrieved,
EventPartyChanged,
EventPartyAdded
EventHeld, EventDialing*,
EventReleased,
EventPartyChanged
EventReleased*,
EventPartyChanged
EventRinging*,
EventEstablished
Y
Y
N
N
N
Y
Y
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Table 23: Supported Functionality (Continued)
Feature Request
TRouteCall b
TApplyTreatment
Request Subtype
Unknown
Default
Call-Routing Requests
EventRouteUsed
Label
OverwriteDNIS
DDD
IDDD
Direct
Reject
Announcement
PostFeature
DirectAgent
Priority
DirectPriority
AgentID
Corresponding Event(s) Supported
Y
Y
N
N
N
N
N
Y
N
N
N
N
N
N
N CallDisconnect
Call-Treatment Requests
Unknown
IVR
(EventTreatmentApplied +
EventTreatmentEnd)/
EventTreatmentNotApplied
Music
RingBack
N
N
Y
Y
Silence
Busy
CollectDigits
PlayAnnouncement
Y
Y
Y
Y
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Table 23: Supported Functionality (Continued)
Feature Request Request Subtype
PlayAnnouncementAnd-
Digits
PlayApplication
VerifyDigits
RecordUserAnnouncement
DeleteUserAnnouncement
CancelCall
SetDefaultRoute
TextToSpeech
TextToSpeechAndDigits
FastBusy
Corresponding Event(s) Supported
TGiveMusicTreatment
TGiveRingBackTreatment
EventTreatmentApplied
EventTreatmentApplied
TGiveSilenceTreatment EventTreatmentApplied
DTMF (Dual-Tone MultiFrequency) Requests
TCollectDigits
TSendDTMF
EventDigitsCollected
EventDTMFSent
Voice-Mail Requests
TOpenVoiceFile
TCloseVoiceFile
TLoginMailBox
TLogoutMailBox
TPlayVoice
TAgentLogin
TAgentLogout
EventVoiceFileOpened
EventVoiceFileClosed
EventMailBoxLogin
EventMailBoxLogout
EventVoiceFileEndPlay
Agent & DN Feature Requests
EventAgentLogin
EventAgentLogout
Y
N
Y
N
N
N
N
N
Y
Y
N
N
N
N
Y
N
Y
N
N
N
Y
N
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Table 23: Supported Functionality (Continued)
Feature Request
TAgentSetIdleReason
TAgentSetReady
TAgentSetNotReady
TMonitorNextCall
TCancelMonitoring
TCallSetForward
TCallCancelForward
TSetMuteOff
TSetMuteOn
TListenDisconnect
TListenReconnect
TSetDNDOn
TSetDNDOff
TSetMessageWaitingOn
TSetMessageWaitingOff
TQuerySwitch a
Request Subtype
OneCall
AllCalls
None
Unconditional
OnBusy
OnNoAnswer
OnBusyAndNoAnswer
SendAllCalls
DateTime
ClassifierStat
Query Requests
Corresponding Event(s) Supported
EventAgentIdleReasonSet N
EventAgentReady
EventAgentNotReady
EventMonitoringNextCall
Y
Y
Y
Y
EventMonitoringCanceled
EventForwardSet
N
N
N
N
N
N
Y
N
EventForwardCancel
EventMuteOff
EventMuteOn
EventListenDisconnected
EventListenReconnected
EventDNDOn
EventDNDOff
EventMessageWaitingOn
EventMessageWaitingOff
EventSwitchInfo
N
N
Y
Y
N
N
Y
Y
N
N
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Table 23: Supported Functionality (Continued)
Feature Request
TQueryCall a
TQueryAddress a
Request Subtype
PartiesQuery
StatusQuery
AddressStatus
MessageWaitingStatus
AssociationStatus
CallForwardingStatus
AgentStatus
NumberOfAgentsInQueue
NumberOfAvailableAgents-
InQueue
NumberOfCallsInQueue
AddressType
CallsQuery
SendAllCallsStatus
QueueLoginAudit
NumberOfIdleClassifiers
NumberOfClassifiersInUse
NumberOfIdleTrunks
NumberOfTrunksInUse
DatabaseValue
DNStatus
QueueStatus
Corresponding Event(s) Supported
EventPartyInfo N
EventAddressInfo
N
N
Y
Y
Y
Y
N
Y
N
N
N
Y
Y
Y
Y
N
Y
N
N
Y
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Table 23: Supported Functionality (Continued)
Feature Request
TQueryLocation a
Request Subtype
AllLocations
LocationData
MonitorLocation
CancelMonitorLocation
MonitorAllLocations
Corresponding Event(s) Supported
EventLocationInfo I
I
I
I
I
I
I
I
CancelMonitorAllLocations
LocationMonitorCanceled
AllLocationsMonitor-
Canceled
TQueryServer a
EventServerInfo
User-Data Requests
TAttachUserData EventAttachedDataChanged
TUpdateUserData EventAttachedDataChanged
TDeleteUserData
TDeleteAllUserData
EventAttachedDataChanged
EventAttachedDataChanged
ISCC (Inter Server Call Control) Requests
TGetAccessNumber b
TCancelRegGetAccess-
Number
EventAnswerAccessNumber
EventReqGetAccess-
NumberCanceled
Special Requests
Y
Y
Y
Y
Y
I
I
TReserveAgent
TSendEvent
TSendEventEx
TSetCallAttributes
TSendUserEvent
TPrivateService
EventAgentReserved
EventACK
EventACK
EventCallInfoChanged
EventACK
EventPrivateInfo
I
Y
Y
I
I
I
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Table 23: Supported Functionality (Continued)
Feature Request Request Subtype Corresponding Event(s) Supported
TNetworkConsult
TNetworkAlternate
TNetworkTransfer
TNetworkMerge
TNetworkReconnect
TNetworkSingleStep-
Transfer
TNetworkPrivateService
Network Attended Transfer/Conference Requests e
EventNetworkCallStatus
EventNetworkCallStatus
EventNetworkCallStatus
EventNetworkCallStatus
EventNetworkCallStatus
EventNetworkCallStatus
TTransactionMonitoring
EventNetworkPrivateInfo
ISCC Transaction Monitoring Requests
EventACK
EventTransactionStatus
N
Y
E
N
N
N
N
N
N a. Only the requestor receives a notification of the event associated with this request. b. This feature request may be made across locations in a multi-site environment. However, if the location attribute of the request contains a value relating to any location other than the local site—except when the response to this request is
EventError
—there will be a second event response that contains the same reference
ID
as the first event. This second event will be either
EventRemoteConnectionSuccess
or
EventRemoteConnectionFailed.
c. Supported for SIP endpoints that have the Remote Talk feature activated.
d. SIP Server does not use the extensions
parameter. Any data in this parameter is ignored.
e. All T-Servers support NAT/C requests with
AttributeHomeLocation
provided that this attribute identifies a network location that is capable of processing such requests. Refer to the Network T-Server Deployment
Guides to determine whether a specific Network T-Server can process these requests.
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Chapter 7: T-Library Functionality Support Using the Extensions Attribute
Using the Extensions Attribute
SIP Server supports the use of the
Extensions
attribute as documented in the
Genesys 7 Events and Models Reference Manual and the Voice Platform SDK
7.6 .NET (or Java) API Reference. See those documents for complete information on the T-Server events, call models, and requests. Additionally, the
Extensions
are also supported.
Table 24: Use of the Extensions Attribute
Extension Used In
Key Type
record String
Call Recording
TRouteCall
Description
• When set to destination
, call recording is initiated on the routing destination DN (agent), and will continue until the agent leaves the call.
• When set to source
, call recording is initiated on the DN that sent a call to the Routing
Point (customer), and will continue until the customer leaves the call.
See “Call Recording” on page 102
for details.
MonitorMode String
Call Supervision
TMonitorNextCall Specifies the monitor mode:
• mute, normal
—A mute connection.
• connect
—A three-party conference call.
• coach
: Only the agent can hear the supervisor (whisper coaching).
See “Call Supervision” on page 104
for details.
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Table 24: Use of the Extensions Attribute (Continued)
Extension Used In
Key
MonitorScope
Type
String TMonitorNextCall
Description
AssistMode String TSingleStep-
Conference
Specifies the required intrusion/observation scope:
• agent
—The monitoring is initiated for a specific agent.
The supervisor is disconnected when the call is transferred or released, but will be connected to the next call that is routed to the same agent.
• call
—The monitoring is initiated to track an entire customer call. If the call is transferred to another agent, queue, or VRU, the monitoring function continues with the call until the customer disconnects the call.
Specifies the required assistance mode:
• connect
—This is the default value - a three-party conference call.
• coach
—Only the agent can hear the supervisor (whisper coaching).
NO_ANSWER_TIMEOUT String
No Answer Supervision
TRouteCall If set, the value of this Extension overrides any value set in any of the following configuration options for the current call:
• no-answer-timeout
• agent-no-answer-timeout
• extn-no-answer-timeout
• posn-no-answer-timeout
See “No-Answer Supervision” on page 157
for details.
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Table 24: Use of the Extensions Attribute (Continued)
Extension Used In Description
Key
NO_ANSWER_ACTION
Type
String TRouteCall If set, the value of this Extension overrides any value set in any of the following configuration options for the current call:
• no-answer-action
• agent-no-answer-action
NO_ANSWER_OVERFLOW Comma-
LCTPartiesLength separated list
TRouteCall If set, the value of this Extension overrides any value set in any of the following configuration options for the current call:
• no-answer-overflow
• agent-no-answer-overflow
• extn-no-answer-overflow
• posn-no-answer-overflow
Providing Call Participant Info
Integer EventUserEvent If set, the value of this Extension specifies the how many parties are involved in a single call.
See “Providing Call Participant
Info” on page 164 for details.
LCTParty<n> Integer EventUserEvent If set, the value of this Extension represents a party of the call, where n is an integer value starting from
0
.
See “Providing Call Participant
Info” on page 164 for details.
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Table 24: Use of the Extensions Attribute (Continued)
Key
CPNDigits feature dn agent-dn login-id
Extension
Type
String TMakeCall,
TMakePredictiveCall,
TInitiateConference,
TInitiateTransfer
If set, the value of this Extension overrides the username provided in the URI in the
From
header of the
INVITE
message.
This Extension is not applicable when performing a
TMakeCall request using the
REFER
method.
See “Providing a Caller ID” on page 165
for details.
Remote Supervision
String
“remoteobserving”
TRouteCall
String
Providing Caller ID
TRouteCall
This extension triggers registration of a routed party as a supervisor with parameters specified in additional
Extensions
described in this subsection.
See “Remote Supervision” on page 115
for details.
An optional DN number that can be used during the monitoring session, as a substitute for the external PSTN number that the supervisor used to dial in.
String
String
Used In
TRouteCall
TRouteCall
Description
The target that the supervisor wants to monitor. This parameter can be a Routing Point, ACD
Queue, or an agent DN.
(Optional) A Login ID, which will be used for remote client authorization.
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Table 24: Use of the Extensions Attribute (Continued)
Extension Used In
Key
monitor-type
Type
AllCalls TRouteCall password post-feature-dn
String
String
TRouteCall
TRouteCall
Description
(Optional) The supervisor will monitor all consecutive calls for the selected target, until the supervisor decides to hang up and end the monitoring session. In between monitored calls, the supervisor’s call is parked.
(Optional) A password, which will be used for remote client authorization for a specified
Login ID.
(Optional) A Routing Point, to which the remote supervisor will be connected after the supervision session.
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Table 24: Use of the Extensions Attribute (Continued)
Extension Used In
Key Type
VideoFile String
Video Support
TSingleStep-
Conference
AgentVideo String TSingleStep-
Conference
Description
A string that contains the name of the video file that will be played for the customer. If this key-value pair is not specified, the default video file will be played. The default video file is configured in the SIP Server
Application
using the
configuration option.
See “Video Support” on page 170
for details.
A string that identifies the origin of the video stream played to the agent. The values are as follows:
• from-third-party
—The agent receives video from a third party—that is, the party that participated in the call before the operation started.
• to-third-party
—The agent receives the same video stream played to the third party—that is, video from the file specified by the
VideoFile
parameter.
See “Video Support” on page 170
for details.
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Error Messages
presents the complete set of error messages SIP Server distributes in
EventError,
which SIP Server generates when it cannot execute a request because of an error condition
.
Table 25: Error Messages for SIP Server
Code Symbolic Name
40 TERR_NOMORE_LICENSE
54
55
56
57
51
52
53
41
42
43
50
TERR_RESOURCE_SEIZED
TERR_IN_SAME_STATE
TERR_UNKNOWN_ERROR
TERR_UNSUP_OPER
TERR_INTERNAL
TERR_INVALID_ATTR
TERR_NO_SWITCH
TERR_PROTO_VERS
TERR_INV_CONNID
TERR_TIMEOUT
Description
No more licenses are available.
Resource is already seized.
Object is already in requested state.
Unknown error code. Request cannot be processed.
Operation is not supported.
Internal error.
Attribute in request operation is invalid.
No connection to the switch.
Incorrect protocol version.
Connection ID
in request is invalid.
Switch or T-Server did not respond in time.
58
59
122
TERR_OUT_OF_SERVICE
TERR_NOT_CONFIGURED
TERR_CANT_REG_DNS
Switch or T-Server is out of service.
DN is not configured in the Configuration
Database.
DN
in request is invalid.
61
93
96
TERR_INV_CALL_DN
TERR_DEST_INV_STATE Invalid destination state.
TERR_CANT_COMPLETE_CONF
119 TERR_BAD_PASSWD
Call cannot add new conference party.
Password
was invalid. May occur when SIP
Server receives a
404 Not Found message from the Media Gateway after an unsuccessful attempt to route a call
Cannot register DNs on the switch.
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Table 25: Error Messages for SIP Server (Continued)
Code Symbolic Name
128 TERR_BAD_DN_TYPE
Description
Invalid DN type for DN registration.
166 TERR_RES_UNAVAIL (JTAPI object) resource is not available.
168 TERR_INV_ORIG_ADDR Originating
177 TERR_TARG_DN_INV DN target (in route call) was invalid.
195 TERR_CFW_DN_INV Call
243 TERR_CLNT_NOT_MON
259 TERR_INV_PASSWD
302 TERR_INV_DTMF_STRING
410 TERR_INAPPR_TRTM
Internal error—client corrupted in T-Server.
Invalid credentials (login_id or password).
DTMF string invalid.
Invalid treatment type.
415 TERR_INV_DEST_DN The invalid.
470 TERR_PARTY_NOT_ON_CALL
496 TERR_INV_CALL_STATE
506 TERR_RECVD_INV_STATE
700 TERR_INV_LOGIN_REQ
Party in request is not involved in a call.
Party in request is in the call
state.
Call/Party is in invalid state for this time
Agent cannot log in at this time.
701 TERR_INV_LOGOUT_REQUEST
702 TERR_INV_READY_REQ
1605 TERR_INVALIDPARTY Party in request was invalid on switch.
1183 TERR_CSTA_SUBRES_OUTST_LIMIT_EXC Rejects the second consecutive call party control request if it comes in less than one second after the first one.
3002 TERR_PRIVVIOLATION
Agent cannot logout.
Agent cannot go to ready
state.
3005 TERR_UNSUCC_ROUTECALL
User doesn’t have security privilege on the switch.
Routecall
request was unsuccessful.
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Chapter 7: T-Library Functionality Support Known Limitations
Known Limitations
Several known limitations result from the current SIP Server and softswitches/gateways interface:
1. The Stuck Calls Cleanup functionality of T-Server Common Part is not supported.
2. Due to the specifics of gateway behavior in performing SIP
REFER
methods, support for remote agents has some limitations. In order to use remote agents, you must perform one of the two following steps:
Provision customers and remote agents to use physically separate gateways (otherwise, calls from agents to customers take shortcuts within gateways, which means that SIP Server loses track of the call and therefore cannot perform call control). Even in this configuration, direct calls between two remote agents on the same gateway are not visible to SIP Server.
Or,
Disable the SIP
REFER
method for the gateways where the remote agents are located. This enables SIP Server to see agent-to-customer and agent-to-agent calls.
3. You must configure the Outbound Contact Solution to use a single-step transfer when you use
Transfer
Mode. SIP Server does not support consultation calls when it is working with the Outbound Contact Solution.
4. A single-step conference from an existing call or an existing conference
(including monitored calls) is supported to an internal DN destination only.
A single-step conference from an existing call or an existing conference to a Routing Point, an ACD Queue, or an external destination is not supported.
5. A single-step transfer cannot be established from a DN in a
Ringing
state.
6. An
EventReleased (switch::)
message is issued when the last internal party leaves a call.
7. A Network Attended Transfer and Conference is not supported when SIP
Server is the originating server.
8. SIP Server allows an agent to complete a consultation transfer of a call only if the call is located on a Routing Point. Third-party call control
(3pcc) blind conference calls are not supported.
9. The
TDeleteFromConference
request is not supported for first-party call control (1pcc) conference calls.
10. SIP Server does not report a first-party call control (1pcc) conference with mixing on an endpoint.
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Third-Party Equipment—Known Limitations
The known limitations when SIP Server is operating with a third-party equipment are as follows:
1. The Siemens HiPath 8000 switch is supported with the following limitations:
1pcc (first-party call control) calls are not supported.
The remote answer feature (
TAnswerCall
) is not available with version
2.0. However, the remote answer feature is supported with version 2.2.
Call scenarios containing the
REFER
message are not supported with this switch. SIP Server must be configured to use
re-INVITE
-based call control methods.
Genesys recommends setting the dual-dialog-enabled
configuration option to false
if Siemens optiPoint phones are used in re-INVITE mode for third-party call control (3pcc) operations.
Sometimes SIP Server cannot retrieve a call within a mixed phone environment. To avoid this problem, set the sip-hold-rfc3264
option with a proper value on the
Annex
tab of the DN.
2. The following media gateways support re-INVITE
-based call transfers only:
Alcatel 7515
Cisco A5350
Cisco A5400
Asterisk
Sonus
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196 Framework 7.6
Chapter
8
SIP Server Configuration
Options
z z z z z
This chapter describes the configuration options that are unique to SIP Server and contains the following sections:
Application-Level Options, page 197
Agent Login–Level and DN-Level Options, page 228
GVP Integration Options, page 247
Changes from 7.5 to 7.6, page 249
SIP Server also supports common log options described in Chapter 11 on page 321
and options common to all T-Servers described in Chapter 12 on page 343
. Please refer to the Framework 7.6 Stream Manager Deployment
Guide if you need to configure Stream Manager for use with SIP Server.
Application-Level Options
Configuration options specific to SIP Server functionality are set in
Configuration Manager, in the corresponding sections on the
Options tab of the SIP Server
Application
object.
T-Server Section
This section must be called
TServer.
For ease of reference, the options have been arranged in alphabetical order.
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acw-in-idle-force-ready
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Emulated Agents” on page 138
Specifies whether, after timed manual wrap-up (when option
is set to true
), SIP Server forces the agent to the
Ready
state. With value false
,
SIP Server returns the agent to the state he or she was in prior to requesting manual wrap-up.
Note:
For compatibility with the previous SIP Server releases, you can use the name cwk-in-idle-force-ready for this option as an alias.
after-routing-timeout
Default Value:
10
Valid Values: Any integer
Changes Take Effect: Immediately
Specifies the length of time (in seconds) that SIP Server waits before diverting the call from the Routing Point DN to the destination DN after
RequestRouteCall
was processed. When the call is not diverted before the specified value, the
EventError
message is issued. It will contain the
Reference
ID
of the
TRouteCall
request.
Notes:
Set the value of the after-routing-timeout
option less than the value of the rq-expire-tmout option.
The after-routing-timeout option is also dependent on the
• When the divert-on-ringing
option is set to
true,
the call is considered as “diverted” when the
180 Ringing
message arrives from the destination DN.
• When the divert-on-ringing
option is set to
false, the call is considered as “diverted” when the
200 OK
message arrives from the destination DN.
agent-group
Default Value: None
Valid Value: Any agent group value
Changes Take Effect: At the next agent login session
Specifies a value for an agent group that will be used for SIP Server reporting.
SIP Server obtains the value for this option in the following order of precedence:
1. In the
TServer
section of the
Annex
tab of the
DN
object.
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2.
In the
TServer
section of the
Options
tab of the SIP Server
Application object.
agent-no-answer-action
Default Value:
none
Valid Values: none notready logout
SIP Server takes no action on agents when calls are not answered.
SIP Server sets agents to
NotReady
when calls are not answered.
SIP Server automatically logs out agents when calls are not answered.
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Defines SIP Server’s default action if a logged-in agent fails to answer a call within the time defined in the agent-no-answer-timeout
option. See also the
NO_ANSWER_ACTION
extension in section “Using the Extensions Attribute” on page 186
for more information about how this option is used
.
SIP Server obtains the value for this option in the following order of precedence:
1. In the
TServer
section of the
Annex
tab of the
Agent Login
object.
2.
In the
TServer
section of the
Options
tab of the SIP Server
Application object.
agent-no-answer-overflow
Default Value: No default value
Valid Values: none recall release
Any valid overflow destination
SIP Server does not attempt to overflow a call on an agent desktop when the agent-no-answer-timeout
option expires.
SIP Server returns the call to the last distribution device (the device reported in the
ThisQueue
attribute of the call) when the agent-no-answer-timeout
option expires.
SIP Server releases the call.
SIP Server returns the call to the specified destination when the value set when the agent-no-answer-timeout option expires.
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Specifies a sequence of overflow destinations (separated by comma) that SIP
Server attempts to overflow to when the time specified in the agent-no-answer-timeout option expires. SIP Server attempts to overflow in the order specified in the list.
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•
When all overflow attempts fail, SIP Server abandons overflow. See also the
NO_ANSWER_OVERFLOW
extension
in section “Using the Extensions
for more information about how this option is used .
•
When the list of overflow destinations contains the recall
value and the call was not distributed, SIP Server skips to the next destination in the list.
SIP Server obtains the value for this option in the following order of precedence:
1. In the
TServer
section of the
Annex
tab of the
Agent Login
object.
2.
In the
TServer
section of the
Options
tab of the SIP Server
Application object.
agent-no-answer-timeout
Default Value:
15
Valid Value: Any integer from
0
–
600
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Defines the default time (in seconds) that SIP Server allows for a logged-in agent to answer a call before executing the actions defined in the agent-no-answer-overflow
and agent-no-answer-action
options
.
When set to
0
, the Agent No-Answer Supervision feature is disabled. See the
NO_ANSWER_TIMEOUT
extension in section “Using the Extensions Attribute” on page 186
for more information about how this option is used.
SIP Server obtains the value for this option in the following order of precedence:
1. In the
TServer
section of the
Annex
tab of the
Agent Login
object.
2.
In the
TServer
section of the
Options
tab of the SIP Server
Application object.
agent-strict-id
Default Value: false
Valid Values: true , false
Changes Take Effect: Immediately
Related Feature: “Emulated Agents” on page 138
Specifies whether, for agents, SIP Server enables any
AgentID
to be used during login (value false
), or only those configured in the Configuration Layer
(value
true
).
am-detected
Default Value: drop
Valid Values: drop connect
The call is released.
The connected call stays connected.
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Changes Take Effect: Immediately
Specifies the behavior of SIP Server where CPD is operating and an answering machine is detected on an Outbound call. SIP Server provides the CPD result in
UserData
attached to the call as a key-value pair with key
AnswerClass containing the value
AM
. This
UserData
in
EventRouteRequest
provides extra information to the strategy, so that the strategy can decide to drop the
AM
call if required.
audio-codecs
Default Value: telephone-event, PCMU, PCMA, G723, G729, GSM
Valid Values: Any from the list of telephone-event, PCMU, PCMA, G723,
G729, and
GSM
words, delimited by commas. Unrecognized words are ignored.
Changes Take Effect: Immediately
Specifies the list of audio codec files that Stream Manager release 7.0.2 (or earlier) uses to play treatments.
Note:
This option is obsolete. It can be used for backward compatibility with the previous SIP Server releases.
Calls While in
Emulated ACW
backwds-compat-acw-behavior
Default Value: false
Valid Value: true, false
Changes Take Effect: Immediately
Related Feature: “Emulated Agents” on page 138
Specifies whether pre-7.5 behavior after-call work is enabled (value = true
) or disabled (value = false
), for backward compatibility.
With value true,
if an agent receives or makes a business call while in emulated ACW, SIP Server does the following:
1. Stops the ACW timer.
2.
Forces the agent to the
Ready
state.
3.
Restarts ACW (and the legal-guard timer) after the new business call is released.
If an agent makes or receives a work-related call while in ACW, SIP Server does the following:
1. Suspends the ACW, but leaves the agent in the ACW state.
2.
Resumes the ACW timer once the work-related call is released.
SIP Server categorizes as a work-related call any call that an agent makes while in the
NotReady
state with workmode
set to
AfterCallWork
or
AuxWork
.
After the ACW and any configured legal-guard time have been completed, the agent is forced to the
Ready state. If an agent makes or receives a private call during ACW, no action is taken and the ACW timer keeps running.
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Business Call
While Not Ready
With value false,
if an agent receives or makes a business call while in emulated ACW, SIP Server does the following:
1. Stops the ACW timer and adds the remaining amount of ACW to the ACW period for the new call. If either of the ACW periods is untimed, the resulting ACW will also be untimed.
2.
Forces the agent to the
NotReady (ManualIn) state.
3.
Restarts ACW after the business call is released.
If an agent makes or receives a work-related or private call while in ACW, SIP
Server does the following:
1. Suspends the ACW timer.
2.
Forces the agent to the
NotReady (ManualIn) state.
3.
Returns the agent to the ACW state, and resumes the ACW timer once the call is released.
After the ACW and any configured legal-guard time have been completed, the agent is forced to the
Ready state.
With value true,
if an agent receives a business call while in an emulated
NotReady
state, except for ACW or legal-guard time, SIP Server sets the agent state to
Ready
for the duration of the business call.
With value false,
if an agent receives a business call while in emulated
NotReady
state, except for ACW or legal-guard time, SIP Server will maintain the current agent state for the duration of the business call. After the call and any associated wrap-up are completed, SIP Server will return to his or her previous
NotReady
state. Note that no legal-guard time is applied, because the agent does not go into the
Ready
state.
busy-tone
Default Value: music/busy_5sec
Valid Values: Name and path of any valid audio file
Changes Take Effect: Immediately for all new calls
Specifies the audio file to be played for the
Busy
treatment.
call-rq-gap
Default Value:
0
Valid Value: Any integer from
0
–
1000
Changes Take Effect: Immediately
Specifies (in milliseconds) the length of delay applied to a request issued against a busy call (a call that has another request working on it already). This prevents race conditions on the different call legs.
Note:
Genesys recommends that you set this option to
0.
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cancel-monitor-on-disconnect
Default Value: true
Valid Values: true false
Call supervision subscription is canceled.
Call supervision subscription is not canceled.
Changes Take Effect: Immediately
Related Feature: “Call Supervision” on page 104
Specifies whether call supervision subscription is canceled when the client that requested it disconnects from SIP Server.
collect-tone
Default Value: music/collect
Valid Values: Name and path of any valid audio file
Changes Take Effect: Immediately for all new calls
Specifies the audio file that SIP Server uses to produce the noncompletion tone played during DTMF digit collection.
cpd-info-timeout
Default Value:
3
Valid Values:
0-30
Changes Take Effect: Immediately
Specifies the time interval, in seconds, during which SIP Server waits for SIP
INFO
messages with the results of call progress detection from a gateway. This timeout starts right after the
200 OK
messages is received from the gateway.
default-dn
Default Value:
NULL
Valid Value: Any valid DN
Changes Take Effect: Immediately
Specifies the DN to which calls are sent when URS is nonoperational, or when the timeout specified in the
Note:
You can also use this option for emergency ACD routing.
This option can be set as the SIP Server Application level and at the
Switch/DN level. The setting at the Switch/DN level takes precedence over the Application level setting.
default-monitor-mode
Default Value: mute
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Valid Values: mute
(or normal
)
Silent monitoring is used (supervisor connection is mute) coach connect
Whisper coaching is used (only the monitored agent can hear the supervisor)
The open supervisor presence is used
Changes Take Effect: Immediately
Related Feature: “Call Supervision” on page 104
Initializes a new call supervision subscription monitor mode if the
MonitorMode extension is not provided (or if its value is specified incorrectly) in the
TMonitorNextCall
request.
default-monitor-scope
Default Value: call
Valid Values: call agent
The supervisor remains on the call until it is finished.
SIP Server disconnects the supervisor from the call automatically when the monitored agent leaves the call.
Changes Take Effect: Immediately
Related Feature: “Call Supervision” on page 104
Initializes a new call supervision subscription monitor scope if the
MonitorScope
extension is not provided (or its value is specified incorrectly) in the
TMonitorNextCall
request.
default-music
Default Value: music/on_hold
Valid Value: Name and path of any valid audio file
Changes Take Effect: Immediately for all new calls
Specifies the name of the file that is played for the music treatment, if none is specified in
TApplyTreatment,
or if the specified file is missing.
default-video-file
Default Value:
NULL
Valid Values: Any valid video file codec and path
Changes Take Effect: Immediately
Related Feature: “Video Support” on page 170
Contains the name of the video file that will be played to the caller if a single-step conference to the gcti::video
device does not contain a
VideoFile key in the
Extensions
attribute.
delay-between-refresh-on-switchover
Default Value:
0
Valid Values: A standard timeout value format
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Changes Take Effect: Immediately
Specifies the delay timeout between dialogs re-invites that SIP Server performs after the delay-to-start-refresh-on-switchover
option setting expires. When this option is set to
0
(the default), the dialogs are not updated.
Recommended values between
10
and
50 msec.
delay-to-start-refresh-on-switchover
Default Value:
10000 (10 seconds)
Valid Values: A standard timeout value format
Changes Take Effect: Immediately
Specifies the time interval that SIP Server waits after a switchover before re-inviting previously connected dialogs. The timeout must be long enough to account for possible network switching delays.
divert-on-ringing
Default Value: true
Valid Values: true false
SIP Server generates
EventRouteUsed
and
EventDiverted messages when the SIP
180 Ringing
response arrives for the
INVITE
request at the routing destination.
SIP Server postpones
EventRouteUsed
and
EventDiverted messages until the call is answered by the routing destination with a SIP
200 OK
message. If the call is not answered within the value specified by the rq-expire-tmout
option, the destination
SIP dialog is canceled and an
EventError
message is generated.
Changes Take Effect: Immediately
Determines SIP Server behavior when routing calls.
dtmf-payload
Default Value:
101
Valid Values: Any integer from
0
–
128
Changes Take Effect: Immediately
Specifies the value to be used in negotiations for the DTMF payload type, when Stream Manager release 7.0.2 (or earlier) treatments are used.
Note:
This option is obsolete. It can be used for backward compatibility with the previous SIP Server releases.
emergency-recording-cleanup-enabled
Default value: false
Valid values: true false
SIP Server automatically terminates emergency recording.
SIP Server does not terminate emergency recording.
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Changes Take Effect: Immediately
Specifies whether SIP Server automatically terminates emergency recording when no internal parties remain on a call.
emergency-recording-filename
Default Value:
NULL
Valid Values: Any valid file name using the variables specified below
Changes Take Effect: When the next emergency call recording is initiated
Specifies the recorded file name when emergency call recording is initiated by an agent. When this option contains a value, the generated emergency call recording file name is added as
UserData
to the call with the
GSIP_EMRGREC_FN key. When this option does not contain a value, the recorded file name will be the
UUID
of the call.
The following variables are used when creating the file:
$ANI$
: The
$DNIS$
: The
$DATE$
$TIME$
:
: The current date (GMT) in the Y-M-D format.
The current time (GMT) in the H-M-S format.
$CONNID$
: The of the call.
$UUID$
: The
$AGENTID$
: The ID, if the agent is logged in on the device where the emergency call recording is initiated.
$AGENTDN$
where the emergency call recording is initiated.
emulated-login-state
Default Value: ready
Valid Values: ready not-ready
SIP Server distributes
EventAgentReady
after
EventAgentLogin
.
SIP Server distributes
EventAgentNotReady after
EventAgentLogin
.
Changes Take Effect: Immediately
Related Feature: “Emulated Agents” on page 138
When SIP Server performs an emulated agent login and the client specifies an agent work mode other than
ManualIn
or
AutoIn,
SIP Server uses this option to determine which event to distribute.
When the client specifies the agent work mode
ManualIn,
SIP Server distributes
EventAgentNotReady
after
EventAgentLogin,
and places the agent in the
NotReady
state.
When the client specifies the agent work mode
AutoIn,
SIP Server distributes
EventAgentReady
after
EventAgentLogin,
and places the agent in the
Ready
state.
This option can be set in a number of places, and SIP Server processes it in the following order of precedence, highest first. If the value is not present at the higher level, SIP Server checks the next level, and so on.
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1. In the
Agent Login
object, on the
Annex tab in the
TServer
section.
2.
In the
DN
object which represents the device, on the
Annex tab in the
TServer
section.
3.
In the
DN
object which represents the Agent Group, such as an ACD Queue, on the
Annex
tab in the
TServer
section.
4.
In the SIP Server
Application
object, on the
Options tab in the
TServer section.
enforce-external-domains
Default Value:
NULL
Valid Values: A list of computer names or IP addresses that are external to SIP
Server. The list can be separated by semicolons (;).
Changes Take Effect: Immediately
When a value is configured, SIP Server checks the list of computer names or
IP addresses against the computer names or IP addresses specified in the URI of the
From
header. If there is a match, then the DN is considered external.
When a value is not configured, SIP Server uses the user part of the URI only to find the device.
event-ringing-on-100trying
Default Value:
false
Valid Values: true false
SIP Server generates
EventRinging
.
SIP Server does not generate
EventRinging
.
Changes Take Effect: Immediately
Specifies whether SIP Server generates an
EventRinging
message for a DN when it receives a
100 Trying
SIP message. Normally, the
EventRinging message is generated on
180 Ringing
SIP message, but this option allows for
GVP integration when the IVR Server is configured in Behind-the-Switch mode.
Note:
This option must be set at both the
Application
and at the
Switch/DN level because it is used for proper synchronization with the I-Server
Application
.
external-registrar
Default Value:
NULL
Valid Values: String conforming to the SIP-URI syntax of RFC 3261, defined as: sip:[userinfo]hostport uri-parameters[headers]
Changes Take Effect: Immediately
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Specifies the location of an external registrar service. SIP Server implements very limited registrar functionality in order to support clients that can only register dynamically (for example, Microsoft Messenger 4.7–5.1). Such clients must be configured in the Configuration Layer as DNs. Depending on the state of the internal SIP Server registrar, registration subscriptions from either all, or only unconfigured, clients are forwarded to the external registrar.
For example: sip:192.168.8.100:5090;transport=tcp
If no external registrar is specified, a
503 Service Unavailable
error is returned for the
REGISTER method.
extn-no-answer-overflow
Default Value: None
Valid Values: none recall release
Any valid overflow destination
SIP Server does not attempt to overflow a call on an extension when extn-no-answer-timeout
expires.
SIP Server returns the call to the last distribution device (the device reported in the
ThisQueue
attribute of the call) when extn-no-answer-timeout
expires.
SIP Server releases the call.
SIP Server returns the call to the specified destination when the value set for the extn-no-answer-timeout
option expires.
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Specifies a sequence of overflow destinations (separated by comma) that SIP
Server attempts to overflow to when the time specified in option extn-noanswer-timeout has expired. SIP Server attempts to overflow in the order specified in the list.
•
When all overflow attempts fail, SIP Server abandons overflow. See also the
NO_ANSWER_OVERFLOW
extension
in section “Using the Extensions
for more information about how this option is used
.
•
When the list of overflow destinations contains the recall
value and the call was not distributed, T-Server skips to the next destination in the list.
SIP Server obtains the value for this option in the following order of precedence:
1. In the
TServer
section of the
Annex
tab of the
DN
object of type
Extension
.
2.
In the
TServer
section of the
Options
tab of the SIP Server
Application object.
extn-no-answer-timeout
Default Value:
15
Valid Value: Any integer from
0
–
600
Changes Take Effect: Immediately
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Related Feature: “No-Answer Supervision” on page 157
Defines the default no-answer timeout (in seconds) that SIP Server applies to any device of type extension.
When the timeout ends, SIP Server executes the actions defined in option extn-no-answer-overflow.
When set to
0
, the No Answer Supervision feature for
DN s of type
Extension
is disabled. See the
NO_ANSWER_TIMEOUT
extension in section “Using the
Extensions Attribute” on page 186 for more information about how this option
is used
.
SIP Server obtains the value for this option in the following order of precedence:
1. In the
TServer
section of the
Annex
tab of the
DN
object of type
Extension
.
2.
In the
TServer
section of the
Options
tab of the SIP Server
Application object.
fast-busy-tone
Default Value: music/atb_5sec
Valid Values: Name and path of any valid audio file
Changes Take Effect: Immediately for all new calls
Specifies the audio file to be played for the
FastBusy
treatment.
fax-detected
Default Value: drop
Valid Values: drop connect
The call is released.
The connected call stays connected.
Changes Take Effect: Immediately
Specifies the behavior of SIP Server where CPD is operating and a fax machine is detected on an outbound call. SIP Server provides the CPD result in
UserData
attached to the call as a key-value pair with key
AnswerClass containing the value
Fax
. This
UserData
in
EventRouteRequest
provides extra information to the strategy, so that the strategy can decide to drop the
Fax
call if required.
find-trunk-by-location
Default Value: false
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Valid Values: true false
SIP Server chooses a gateway or trunk for the outbound call by matching the value of the
option of the DN with the value of the geo-location
option for the
Trunk
device.
SIP Server chooses a gateway or trunk from the pool of all configured trunks which are in service based on the prefix match. (The value of the geo-location
option will be ignored.)
If there is more than one trunk in the pool, SIP Server chooses the trunk in a round-robin algorithm that provides equal gateway load. However, if an external party is transferred to an outbound destination, the same gateway that connected the external party to the call is used for the outbound transfer.
Changes Take Effect: Immediately
Determines SIP Server behavior for choosing a gateway or trunk for the outbound call.
forced-notready
Default Value: true
Valid Values: true false
The desktop is forced into the
Not Ready
state.
The desktop is not forced into the
Not Ready
state.
Changes Take Effect: Immediately for all future calls
Determines whether the desktop is forced into a
Not Ready
state when it does not respond after a
Preview Interaction
dialog box has been displayed on the desktop.
Note:
This option works with the
options to determine what action to take when a desktop does not respond to a preview interaction before the time expires.
inbound-bsns-calls
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Business-Call Handling” on page 138
Specifies whether SIP Server considers all established inbound calls on an agent as business calls.
inherit-bsns-type
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Business-Call Handling” on page 138
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Determines whether a consultation call that is made from a business primary call contains the business call
attribute.
internal-bsns-calls
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Business-Call Handling” on page 138
Determines whether SIP Server considers internal calls made to any agent as business calls.
internal-registrar-domains
Default Value:
NULL
Valid Values: Any valid computer names, separated by a semi-colon (;)
Changes Take Effect: Immediately. Existing subscriptions remain valid.
Specifies the list of logical computer names, registration subscriptions from the endpoints of which are handled by the internal registrar. For example, if DN
4813
is configured in Configuration Manager, DN
4814
is not configured, and the internal registrar is enabled, then:
•
REGISTER
from
4813@world
is accepted.
•
REGISTER
from
4813@galaxy is forwarded to the external registrar.
•
REGISTER
from
4814@world
is rejected with
404 Not Found.
internal-registrar-enabled
Default Value: true
Valid Values: true false
SIP Server’s internal registrar is enabled.
All registration subscriptions are proxied to external registrar
Changes Take Effect: Immediately. Existing subscriptions remain valid.
Specifies whether the internal registrar is enabled. When this option is set to false, a
503 Service Unavailable
error is returned for the
REGISTER method.
internal-registrar-persistent
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Enables SIP Server to update the DN attribute contact
in the configuration database. When an endpoint registers, SIP Server takes the contact information from the
REGISTER
request and updates or creates a key called contact
in the
Annex
tab of the corresponding DN.
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In hot standby
configuration, set the internal-registrar-persistent
option to true
to enable Configuration Server to propagate changes of the contact information to the backup SIP Server.
Note:
SIP Server must have
Full Control
permission for the
DN
objects in order to update a configuration object. By default, it does not have this permission. You need to grant
Full Control
permission for the
System account for the all DNs on the corresponding switch. It is done for all
DNs at once by changing the permissions for the
System
account on the
DN
folder in the
Switch
object. Or, you can start SIP Server under another account that has
Change
permission on the necessary DNs.
intrusion-enabled
Default Value: true
Valid Values: true false
SIP Server invites the supervisor to the current call.
SIP Server does not invite the supervisor to the current call.
Instead, SIP Server will wait for the next call on the monitored agent’s DN to invite the supervisor.
Changes Take Effect: Immediately
Related Feature: “Call Supervision” on page 104
Determines SIP Server behavior when a
TMonitorNextCall
request is submitted while the monitored agent is on a call.
legal-guard-time
Default Value:
0
Valid Value: Any integer from
0
–
30
Changes Take Effect: Immediately
Related Feature: “Emulated Agents” on page 138
Specifies a legal-guard time (in seconds) for agents to postpone the transition to the
Ready
state after a business call or after timed ACW. SIP Server always considers a routed call as a business call. The default value of
0
(zero) disables the functionality of this option.
logout-on-disconnect
Default Value: true
Valid Values: true false
The
EventLogout
message is distributed as soon as the client that requested the login disconnects from SIP Server or unregisters the DN in question. The
EventLogout
message is distributed when SIP Server distributes
EventOutOfService
.
The
EventLogout
message is not distributed.
Changes Take Effect: Immediately
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Specifies how the
EventLogout
message is distributed.
make-call-alert-info
Default Value:
NULL
Valid Value: Any string
Changes Take Effect: Immediately
The contents of this field are passed in the
Alert-Info
header of the
INVITE message sent to the origination party in response to a
TMakeCall
request. This is used to enable a distinctive ringtone or auto-answer on the originating party’s endpoint.
For example, setting this field to
<file://Bellcore-dr3>
turns on a triple ring on Cisco 7940 endpoints.
max-legs-per-sm
Default Value:
0
Valid Values: Any integer from
0
–
65000
Changes Take Effect: Immediately for new legs. Previously created legs are not dropped.
Specifies the maximum number of legs that are created on each connected
Stream Manager. If all connected Stream Managers already have the specified number of legs, no new leg is created.
monitor-internal-calls
Default Value: true
Valid Values: true false
SIP Server starts monitoring sessions for all calls on the DN where call supervision subscription is active.
SIP Server starts monitoring sessions only if external parties participate in the call.
Changes Take Effect: Immediately
Related Feature: “Call Supervision” on page 104
Specifies SIP Server behavior to start monitoring sessions.
music-in-conference-file
Default Value: The value is taken from the default-music
Valid Values: A string containing the valid name of the music file
Changes Take Effect: Available for next 3pcc or 1pcc hold operation
Related Feature: “Silence Treatment in Conference” on page 125
Specifies the silent audio file to be played in applicable conferences (more than two active participants). If the conference has only two active participants, then the music file defined in the default-music
option will be played for the other party when the call is placed on hold. For conferences with more than two active participants, the music-in-conference-file
option is used for a silent MOH treatment instead. Recommended value is music/silence
.
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For example, if a supervisor in silent monitoring mode listens in on a call between a customer and an agent, the supervisor is not considered an active participant. If the agent or customer places the call on hold, the remaining participants will hear the default-music MOH treatment. However if the supervisor places the call on hold, the music/silence file is played instead, so that the customer and agent can continue their conversation undisturbed.
music-in-queue-file
Default Value: None
Valid Values:
<default_music_directory>/<file_name>
Changes Take Effect: Immediately for all new calls
Specifies the file name of the music file to be played when a call is queued on a particular ACD Queue.
Notes:
This option is set at the SIP Server
Application
level and at the
Switch/DN
level (DN of type
ACD Queue
). The setting at the
Switch/DN level takes precedence over the
Application
level setting.
If there is no value specified for this option, the value of the Stream
Manager DN request-uri
option is used instead. For more information about this parameter, see the Framework 7.6 Stream Manager
Deployment Guide.
mwi-agent-enable
Default Value: false
Valid Values: true false
MWI for the agent’s voice mail box is enabled.
MWI for the agent’s voice mail box is disabled.
Changes Take Effect: Immediately
Related Feature: “Message Waiting Indicator Functionality” on page 99
Enables or disables MWI for the agent’s voice mail box.
mwi-domain
Default Value: None
Valid Values: Any computer name
Changes Take Effect: During the next attempt to register for MWI
Related Feature: “Message Waiting Indicator Functionality” on page 99
Specifies the computer name in the URI of the
REGISTER
request. SIP Server sends this information to Asterisk in order to initiate MWI. The value of this option must be a computer name that is recognized by Asterisk.
mwi-extension-enable
Default Value: false
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Valid Values: true false
MWI for the extension’s voice mail box is enabled.
MWI for the extension’s voice mail box is disabled.
Changes Take Effect: Immediately
Related Feature: “Message Waiting Indicator Functionality” on page 99
Enables or disables MWI for the extension’s voice mail box.
mwi-group-enable
Default Value: false
Valid Values: true false
MWI for the agent groups’s voice mail box is enabled
MWI for the agent group’s voice mail box is disabled.
Changes Take Effect: Immediately
Related Feature: “Message Waiting Indicator Functionality” on page 99
Enables or disables MWI for the agent groups’s voice mail box.
mwi-host
Default Value: None
Valid Values: Any host name or IP address
Changes Take Effect: During the next attempt to register for MWI
Related Feature: “Message Waiting Indicator Functionality” on page 99
Specifies the host name of the Voice Mail system to get MWI notification from the host where Asterisk is running. SIP Server will send a
REGISTER
request to mwi-host:mwi-port
to initiate MWI.
mwi-mode
Default Value:
REGISTER
Valid Values:
REGISTER, SUBSCRIBE
Changes Take Effect: After SIP Server restart
Related Feature: “Message Waiting Indicator Functionality” on page 99
When this option is set to
SUBSCRIBE,
SIP Server activates SIP subscriptions for all voice mail box owners as configured by other mwi-<>
options. When set to
REGISTER, the MWI functionality is enabled using the
REGISTER
SIP request method. For backward compatibility with the previous SIP Server releases, set this option to a value of
REGISTER
.
mwi-port
Default Value: None
Valid Values: Any available port
Changes Take Effect: During the next attempt to register for MWI
Related Feature: “Message Waiting Indicator Functionality” on page 99
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Specifies the port of the Voice Mail system to get MWI notification from the port where Asterisk is running. SIP Server will send a
REGISTER
request to mwi-host:mwi-port
to initiate MWI.
nas-private
Default Value: false
Valid Values:
true, false
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Specifies whether No-Answer Supervision is enabled for private calls.
You can set this option at the
Application
and
Switch/Agent Login
or
Switch/DN level (DN of type
Extension
). When set at the
Application
level, the option value is applied globally to all private calls. When set at the
Switch level, the option value is applied to a particular DN or Agent Login.
Note:
The option setting at the
Switch
level takes precedence over the
Application
level setting.
observing-routing-point
Default Value: None
Valid Values: A
Routing Point
DN
Changes Take Effect: For the next call
Related Feature: “Remote Supervision” on page 115
Specifies the service observing Routing Point used for Multi-Site Supervision of the agents, whose endpoints are controlled by this SIP Server. This option must contain a number of a valid
Routing Point
DN in order for the Multi-Site
Supervision feature to work. No routing strategy is required to be loaded on the observing Routing Point.
outbound-bsns-calls
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Business-Call Handling” on page 138
Specifies whether SIP Server considers all established outbound calls on an agent as business calls.
override-to-on-divert
Default Value: false
Valid Values: true false
The username is equal to the destination DN.
The username is equal to the Routing Point or ACD Queue number.
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Changes Take Effect: Immediately
Controls the username part of the
To
header URI for outgoing
INVITE
messages when a call is diverted from a Routing Point or an ACD Queue.
parking-music
Default value: music/silence
Valid Values:
Changes Takes Effect: For the next parked call
Related Feature: “Remote Supervision” on page 115
Specifies the music file, which is played to the remote party parked on the gcti::park
DN.
posn-no-answer-overflow
Default Value: None.
Valid Values: none recall release
Any valid overflow destination
SIP Server does not attempt to overflow a call on a position when posn-no-answer-timeout
expires.
SIP Server returns the call to the last distribution device (the device reported in the ThisQueue attribute of the call) when posn-no-answer-timeout
expires.
SIP Server releases the call.
SIP Server returns the call to the specified destination when the value set for the posn-no-answer-timeout
option expires.
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Specifies a sequence of overflow destinations (separated by comma) that SIP
Server attempts to overflow to when the time specified in option posn-no-answer-timeout expires. SIP Server attempts to overflow in the order specified in the list.
•
When all overflow attempts fail, SIP Server abandons overflow. See also extension
NO_ANSWER_OVERFLOW
.
•
When the list of overflow destinations contains the value recall
and the call was not distributed, SIP Server skips to the next destination in the list.
SIP Server obtains the value for this option in the following order of precedence:
1. In the
TServer
section of the
Annex
tab of the
DN
object of type
ACD
Position
.
2.
In the
TServer
section of the
Options
tab of the SIP Server
Application object.
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posn-no-answer-timeout
Default Value:
15
Valid Value: Any integer from
0
–
600
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Defines the default no-answer timeout (in seconds) that SIP Server applies to any device of type position.
When the timeout ends, SIP Server executes the actions defined in option posn-no-answer-overflow.
When set to
0, the No Answer Supervision feature for
DN s of type
ACD
Position
is disabled. See the
NO_ANSWER_TIMEOUT
extension in section “Using the Extensions Attribute” on page 186
for more information about how this option is used.
SIP Server obtains the value for this option in the following order of precedence:
1. In the
TServer
section of the
Annex
tab of the
DN
object of type
Position
.
2.
In the
TServer
section of the
Options
tab of the SIP Server
Application object.
predictive-call-router-timeout
Default Value:
20
Valid Value: Any non-negative integer
Changes Take Effect: After SIP Server restart
Specifies the maximum time (in seconds) that an answered predictive call can wait on a
Routing Point
DN for a Universal Routing Server (URS) request. If there is no request during this time, the call is dropped. This is primarily a clean-up mechanism for scenarios when URS is non-operational.
preview-expired
Default Value:
90
Valid Values: Any positive integer
Changes Take Effect: Immediately for future calls
Related Feature: “Preview Interactions” on page 164
Specifies the time (in seconds) that the
Preview Interaction
dialog box remains open on a desktop. After the time expires, the dialog box closes and the desktop changes to a Not Ready
state.
Note:
The preview-expired
option works with the
and
options to determine what action to take when a desktop does not respond to a preview interaction before the time expires.
recording-filename
Default Value:
NULL
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Valid Values: Any valid file name using the variables specified below
Changes Take Effect: When the next call recording is initiated
Related Feature: “Call Recording” on page 102
Specifies the file name for call recording when call recording is initiated automatically, according to the SIP Server configuration. When this option contains a value, the generated file name is added as
UserData
to the call with the
GSIP_REC_FN
key. When this option does not contain a value, the file name is the
UUID
of the call.
The following variables are used when creating the file:
$ANI$
: The
$DNIS$
: The
$DATE$
$TIME$
:
:
The current date (GMT) in the Y-M-D format.
The current time (GMT) in the H-M-S format.
$CONNID$
: The of the call.
$UUID$
: The
$AGENTID$
: The ID, if the agent is logged in on the device where the call recording is initiated.
$AGENTDN$
: The
registrar-default-timeout
Default Value:
1800
Valid Values:
1 - 3600
Changes Take Effect: Immediately
Specifies the expiration timeout for a
REGISTER
request as a value (in seconds) in the
200 OK
response that is sent by SIP Server to the SIP endpoint.
ring-tone
Default Value: music/ring_back
Valid Values: Name and path of any valid audio file
Changes Take Effect: Immediately for all new calls
Specifies the audio file to be played for the
RingBack
treatment.
ringing-on-route-point
Default Value: true
Valid Values: true false
SIP Server responds with
a 180 Ringing message.
SIP Server does not respond with a
180 Ringing
message.
Changes Take Effect: Immediately
Specifies whether SIP Server responds with a
180 Ringing message when a call arrives at a Routing Point. It enables transfers for calls waiting at Routing
Points. The disadvantages are:
•
Possible undesirable ringback tone.
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•
Multiple ringing messages delivered for the same call.
router-timeout
Default Value:
10
Valid Value: Any non-negative integer
Changes Take Effect: Immediately
Specifies the maximum time (in seconds) that a call remains on a Routing
Point before a timeout is triggered and the call is sent to the DN specified in default-dn.
session-refresh-interval
Default Value:
1800
Valid Values:
0, 90
–
1800
Changes Take Effect: Immediately
Specifies (in seconds) how often active calls are checked to see if they are still active. A
0
(zero) value disables this feature (the session refresh mechanism is turned off). Values between
1
and
89 ( inclusive) are treated as value
90
.
This option is used to remove stuck calls that must accumulate if endpoints terminate calls without sending the appropriate SIP message.
set-notready-on-busy
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
With this option set to true
, when a call is distributed to a ready agent (that is, the agent is not previously engaged in a call), and the agent endpoint responds to the
INVITE
with a 4xx, 5xx, or 6xx message, SIP Server places the agent in the
Not Ready
state (an
EventAgentNotReady
message is distributed). In addition, a
ReasonCode
key with a value equal to a returned error will be reported in the
Extensions
attribute in the
EventAgentNotReady
message. If a call is distributed to an agent via an ACD queue, the agent is placed in the
Not
Ready
state and the call is diverted to the same ACD queue (at the end of the queue).
silence-tone
Default Value: music/silence
Valid Values: Name and path of any valid audio file
Changes Take Effect: Immediately for all new calls
Specifies the audio file to be played for the
Silence
treatment.
sip-address
Default Value:
NULL
Valid Values: Any valid IP address or host name
Changes Take Effect: After SIP Server restart
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Specifies an IP address of the SIP Server interface. This option must be set when deploying SIP Server on a host with multiple network interfaces. SIP
Server uses this value to build the
Via
and the
Contact
headers in SIP messages. When this option is not set, SIP Server attempts to detect the IP address automatically.
sip-block-headers
Default Value: An empty string
Valid Values: A comma-separated list of the headers to be filtered out during
INVITE
message propagation
Changes Take Effect: Immediately
Specifies a way to filter out headers during
INVITE
message propagation. With an empty string, no headers will be filtered out.
sip-call-retain-timeout
Default Value:
1
Valid Values:
0
–
3600
Changes Take Effect: Immediately
Defines how long (in seconds) a T-Library call is kept in SIP Server’s memory after all call parties (members of the call) are released. This option does not set any delay in generating an
EventReleased
message for a particular DN when the
BYE
message is received (or sent).
sip-dtmf-send-rtp
Default Value: false
Valid Values: true false
SIP Server instructs Stream Manager to send DTMF tones to all call participants using one or both of the following DTMF generation methods: RTP packets with Named Telephone Event
(NTE) payload as specified by RFC 2833, and in-band audio tones according to ITU-T Recommendation Q.23.
The feature is disabled.
Changes Take Effect: Immediately
Related Feature: “DTMF Tones Generation” on page 137
Specifies whether SIP Server instructs Stream Manager to send DTMF tones when a T-Library client issues a
TSendDTMF
request.
sip-enable-call-info
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Providing Call Participant Info” on page 164
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When set to true
, SIP Server distributes the information about call participants to logged-in agents by using the SIP
NOTIFY
method and
EventUserEvent messages.
sip-enable-100rel
Default Value: true
Valid Values: true false
SIP Server advertises support for
100rel
, and requires it whenever the other side indicates support.
SIP Server does not negotiate support for the reliability of provisional responses.
Changes Take Effect: Immediately
Enables processing of
100rel
parameters and messages.
Note:
SIP Server does not process reliable responses in the following scenarios:
• In third-party call control
(TMakeCall)
operations
• When re-
INVITE
methods are used
• When call routing is performed to any destination
sip-enable-moh
Default Value: false
Valid Value: true false
Music-on-hold is enabled.
Music-on-hold is disabled.
Changes Take Effect: At the next
Hold/THoldCall
operation
Enables or disables music-on-hold.
sip-enable-sdp-codec-filter
Default Value: false
Valid Values: true false
SIP Server modifies the SDP message body during SIP renegotiation.
SIP Server does not modify the SDP message body.
Changes Take Effect: Immediately
Specifies whether SIP Server modifies the SDP message body during SIP renegotiation. All codecs that are not in the list of values for the
option are deleted from the SDP. As a result, all call center audio traffic is
established based on the codecs listed in the audio-codecs
sip-hold-rfc3264
Default Value: false
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Valid Value: true false
RFC3264-compliant implementation.
RFC2543-compliant implementation.
Changes Take Effect: Immediately
Specifies which implementation of hold media SDP is used by SIP Server for third-party call control (3pcc) hold operations.
Note:
When this option is set at the
DN
level in the
Annex
tab >
TServer section, it will overrides the
Application
level value.
sip-invite-timeout
Default Value:
0
Valid Values:
0
–
34
Changes Take Effect: Immediately
Specifies the number of seconds that SIP Server waits for a response to the
INVITE
message. The call times out if no response is received. When set to
0
, or if a value is not specified, then the default SIP call timeout of
32
seconds is used.
sip-invite-treatment-timeout
Default Value:
0
Valid Values:
0
–
34
Changes Take Effect: Immediately
Specifies the number of seconds that SIP Server waits for a response to the
INVITE
message for a treatment (such as an announcement or music-on-hold).
The call times out if no response is received. When set to
0
, or if a value is not specified, then the default SIP call timeout of
32
seconds is used.
sip-port
Default Value:
5060
Valid Value: Any valid TCP/IP port
Changes Take Effect: After SIP Server restart
Specifies the port on which SIP Server listens for incoming SIP requests. The same port number is used for both TCP and UDP transports.
sip-refer-to-sst-enabled
Default Value: true
Valid Values: true false
The re-
INVITE method is used instead of the
REFER method for single-step transfer functionality.
The
REFER method from the endpoint is not converted to a re-
INVITE method
and is also rejected by SIP Server if the endpoint has the
option set to
false
.
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Changes Take Effect: Immediately
Specifies whether a re-
INVITE
method is used instead of a
REFER
method for single-step transfer functionality. Use this option when the switch does not support the
REFER
method when performing a single-step transfer.
sip-retry-timeout
Default Value:
30
Valid Values:
1-3600
Changes Take Effect: Immediately
Specifies the time interval, in seconds, after which SIP Server initiates a new subscription if the previous
SUBSCRIBE
dialog is terminated.
sip-ring-tone-mode
Default Value:
0
Valid Values:
0
,
1
Changes Take Effect: Immediately
With the option set to
0
, SIP Server connects Stream Manager to a call to play an audio ring tone. With the option set to
1
, SIP Server waits for a response from the called device, and connects Stream Manager to a call to play an audio ring tone, only when the returned response cannot be used as the offer to a calling device.
Note:
This option can be set at both the SIP Server Application level and at the Switch/DN level. The setting at the Switch/DN level takes precedence over the Application-level setting.
sip-treatments-continuous
Default Value:
false
Valid Values: true false
A routing strategy treatment is played continuously played until the routing destination has answered the call.
A routing strategy treatment is not played continuously.
Changes Take Effect: Immediately for all new calls
Enables or disables a routing strategy treatment to be continuously played until the routing destination has answered the call.
subscription-timeout
Default Value:
180
Valid Values:
1
–
3600
Changes Take Effect: Immediately
Specifies the time interval (in seconds) in the
Expire
header of the
200 OK response message to a subscriber.
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timed-acw-in-idle
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Emulated Agents” on page 138
Specifies whether SIP Server applies the automatic wrap-up timer (using the wrap-up-time parameter) when an agent sends the
TAgentNotReady request while in idle state.
When set to false,
SIP Server does not automatically end manual wrap-up— the agent must return manually from ACW.
Note:
For compatibility with the previous SIP Server releases, you can use the name timed-cwk-in-idle for this option as an alias.
unknown-bsns-calls
Default Value:
false
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Business-Call Handling” on page 138
Determines whether SIP Server considers calls of unknown call type made from or to any agent as business calls.
untimed-wrap-up-value
Default Value:
1000
Valid Value: Any nonzero positive integer
Changes Take Effect: Immediately
Related Feature: “Emulated Agents” on page 138
Specifies the threshold (in seconds) at which the timing of ACW stops and manual intervention is required (untimed ACW).
user-data-im-enabled
Default Value:
NULL
Valid Values: page session
The
UserData
content in the IM is sent in page mode. The
MESSAGE
request(s) are exchanged in the endpoints without dialog. You must use this value for EyeBeam SIP endpoints
When the first SIP dialog containing an IM SDP is created, the
MESSAGE
request(s) are exchanged only in this dialog. You must use this value for Microsoft Office Communicator endpoints.
Changes Take Effect: Immediately
Related Feature: “Instant Messaging” on page 143
Enables the
UserData
content in the Instant Messaging (IM) for a DN.
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user-data-im-format
Default Value:
NULL
Valid Values: text html
The
UserData
content in the IM is encoded in text (text/plain) format. You must use this value for Microsoft Office
Communicator endpoints
The
UserData
content in the IM is encoded in html (text/html) format. You must use this value for Eyebeam SIP endpoints.
Changes Take Effect: Immediately
Related Feature: “Instant Messaging” on page 143
Specifies the format of the
UserData
IM content when different SIP endpoints support different IM formats.
userdata-map-trans-prefix
Default Value: None
Valid Values: A string
Changes Take Effect: Immediately
Related Feature: “Mapping SIP Headers and SDP Messages” on page 149
Contains a transport prefix to indicate what headers in the SIP message carry the mapped
UserData
. SIP Server adds this prefix to all data mapped to the outgoing
INVITE
message. SIP Server scans incoming
INVITE
or
REFER messages used to place a call on the Routing Point for headers that start with this prefix, in addition to performing the normal mapping procedure.
If this option is not specified, no prefix is added to the transmitted data.
wrap-up-time
Default Value:
0
Valid Value: Any positive integer, untimed
0
Value greater than 0 but less
Value equal to untimed-wrapup-value
Value greater than untimedwrap-up-value
ACW is disabled.
The number of seconds of timed ACW, after which SIP Server returns the agent to the
Ready state.
ACW is untimed and the agent must manually return to the
Ready
state.
ACW is disabled.
untimed
ACW is untimed and the agent must manually return to the
Ready
state.
Changes Take Effect: Immediately
Related Feature: “Emulated Agents” on page 138
Specifies the amount of ACW wrap-up time allocated to emulated agents at the end of a business call.
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This option can be set in a number of places, and SIP Server processes it in the following order of precedence, highest first. If the value is not present at the higher level, SIP Server checks the next level, and so on.
SIP Server option priority processing:
1. In the call, in user data
WrapUpTime
(limited to ISCC scenarios).
2.
In a
DN
configuration object of type
Routing Point
, on the
Annex
tab in the
TServer
section.
3.
In a
DN
configuration object of type
ACD Queue
, on the
Annex
tab in the
TServer
section.
4.
In the
TAgentLogin
request, in attribute extension
WrapUpTime
(applies to this agent only).
5.
In an
Agent Login configuration object, on the
Annex
tab in the
TServer section.
6.
In a
DN configuration object of type
Extension
, on the
Annex
tab in the
TServer
section.
7.
In a
DN
configuration object of type
ACD Queue
or
Routing Point that represents logged-in agents (
Agent Group
), on the
Annex
tab in the
TServer section.
8.
In the SIP Server
Application object, on the
Options
tab in the
TServer section.
9.
While in ACW, in the
TAgentNotReady
request with
WorkMode=ACW
(Extending ACW), in attribute extension
WrapUpTime
(applies to this agent only).
UPDATE, INVITE, INFO, and REFER Sections
The option names in this section are a combination of the
TEvent
attribute name ( extensions
or userdata
), a dash, and then a numeric value.
extensions-<n>
Default Value: None
Valid Values: See description
Changes Take Effect: Immediately
Related Feature: “Mapping SIP Headers and SDP Messages” on page 149
The extension
prefix instructs SIP Server to put the SIP header (parameter) into the
Extensions attribute.
The value determines which header within the header parameter of the SIP message is processed. The value of the header/header parameter is added as a key-value pair into the attribute as follows:
<header_name>=<header_value>
You can use the colon character to address the parameter name of a header. For example, the extensions-8=From:tag
option puts the
From:tag=979E46B1-0FDF-
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418F-BA3F-C03F95A4D6E0-2
key-value pair into the
Extensions attribute of the
EventRouteRequest
message.
userdata-<n>
Default Value: None
Valid Values: See description
Changes Take Effect: Immediately
Related Feature: “Mapping SIP Headers and SDP Messages” on page 149
The userdata
prefix instructs SIP Server to put the SIP header (parameter) into the
UserData attribute.
Agent Login–Level and DN-Level Options
You set configuration options described in this section in the
TServer
section on the
Annex
tab of the relevant
Agent Login
or
DN
object in Configuration
Manager.
agent-greeting
Default Value:
NULL
Valid Values: Any file name that will be played to the agent
Changes Take Effect: Immediately
Related Feature: “Personal Greeting” on page 160
Specifies the media file name that will be used as a greeting for the agent.
authenticate-requests
Default Value: None
Valid Values: register
, invite
Changes Take Effect: Immediately
Determines if incoming SIP requests (
REGISTER
or
INVITE
) are treated with an authentication procedure when the following conditions are true:
•
The name of the incoming SIP message exits in the list of the authenticate-requests
parameter.
•
The option password
is configured on the same
DN
object.
If the authenticate-requests
option is not configured on the DN, the
REGISTER request still will be treated with an authentication procedure when the password configuration option is configured on the DN. If neither authenticate-requests
nor password
configuration options are configured on the DN, no requests will be authenticated.
auto-redirect-enabled
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
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Enables automatic processing of the redirect (
3XX
) response from the Sonus
PSX gateway.
The following events occur when this option is set to true
:
•
SIP Server sends an
INVITE
request to the Sonus PSX gateway.
•
A
3XX
response containing a new Uniform Resource Identifier (URI) target is received.
•
SIP Server sends the same
INVITE
request to the new target URI (the Sonus
GSX gateway).
This option must be set in the
TServer
section of the
Annex
tab on the
Trunk
DN that represents the Sonus PSX gateway.
capacity
Default Value:
0
Valid Values: Any positive integer
Changes Take Effect: Immediately
Specifies how many calls can be handled by a specific Voice over IP device represented in the SIP Server configuration as
Trunk.
capacity-group
Default Value:
<DN name>
Valid Values: Any non-empty string
Changes Take Effect: Immediately
Specifies the name of the
DN
object of type
Trunk,
as configured in the
Configuration Layer, that represents a physical Voice over IP device. All DNs configured with the same capacity-group
share the device capacity defined in the
option. See the following option configuration examples.
Note:
The value of the capacity
option must be defined in only one
Trunk
DN.
Example 1
Example 2
DN of type
Trunk
with the Number
Cisco-8340
[TServer] capacity=200 prefix=8340
With these settings, the number of calls to the
Cisco-8340
trunk will be limited to 200.
•
DN of type
Trunk
with the Number
Cisco-8340
[TServer] capacity=200 capacity-group=Cisco-GW prefix=8340
•
DN of type
Trunk
with the Number
Cisco-8341
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[TServer] capacity-group=Cisco-GW prefix=8340
With these settings, the number of calls to the
Cisco-8340
and
Cisco-8341 trunks will be limited to 200.
contact
Default Value: None
Valid Values: Any alphanumerical string
Changes Take Effect: Immediately
Contains the contact URI, specifying the device’s IP address, if this address is fixed. This option is necessary only for stand-alone configurations, and only if the configured device does not register itself in the SIP Server registrar. It is part of the persistent registrar feature.
For example, if the SIP device sends a
REGISTER
request to a SIP Server, and this request is accepted, SIP Server uses the contact information from the
REGISTER
request, and updates (or creates) in Configuration Manager the option contact
in the
TServer
section of the
Annex
tab of the corresponding
DN
object.
The URI format is:
[sip:][number@]hostport[
; transport={tcp/udp}]
Where:
•
sip:
is an optional prefix.
•
number is the DN number. This is currently ignored.
•
hostport
is a host:port
pair, where host
is either a dotted IP address or a
DNS-resolvable hostname for the endpoint.
•
transport=tcp
or transport=udp
is used to select the network transport.
The default value is udp
.
customer-greeting
Default Value:
NULL
Valid Values: Any file name that will be played to the customer
Changes Take Effect: Immediately
Related Feature: “Personal Greeting” on page 160
Specifies the media file name that will be used as a greeting for the customer.
The customer greeting plays continuously until the agent greeting finishes playing. The
agent-greeting
and customer-greeting option values are used as follows:
•
When both options contain different file name values, each file will be played to the customer and the agent as specified.
•
When only one option contains a value, the same file will be played to both the customer and the agent.
230 Framework 7.6
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•
When neither option contains a value, no greeting will be played to either the customer or the agent.
cpn
Default Value: None
Valid Values: The SIP URI format according to the Augmented Backus-Naur
Form (ABNF) in RFC 3261
Changes Take Effect: Immediately
When the value is specified for a DN of type
Trunk
, it will be used as the user part of the SIP URI in the
From
header of the
INVITE
message sent by SIP
Server through this trunk. This option must not be configured on trunks that are allocated for direct signaling between SIP Servers.
Note:
If the
CPNDigits
parameter is specified in the
Extensions
attribute in
TMakeCall
,
TMakePredictiveCall
,
TInitiateConference
, or
TInitiateTransfer
requests, it takes precedence over the cpn
option setting.
default-dn
Default Value:
NULL
Valid Values: Any valid DN
Changes Take Effect: Immediately
This option can be configured only on DNs of type
Routing Point
.
Specifies the DN to which calls are sent when URS is nonoperational, or when the timeout specified in the
router-timeout
option expires. This option does not apply to calls that are delivered to an ACD Queue associated with the Routing Point.
Note:
This option can be set at the SIP Server Application level and at the
Switch/DN level. The setting at the Switch/DN level takes precedence over the Application level setting.
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dual-dialog-enabled
Default Value: true
Valid Values: true false
Set the option to true
for endpoints that accept more than one
SIP dialog and provide remote CTI control by the
NOTIFY message.
Set the option to false
for endpoints that can only accept one active SIP dialog, or cannot provide remote CTI control by the
NOTIFY
message to answer, hold, or retrieve call operations.
Changes Take Effect: Immediately
Enables the SIP dialog functionality for making consultation calls, according to the endpoint type.
enable-agentlogin-presence
Default Value:
false
Valid Values: true false
This value must be used in deployments where agent desktops are not used and all information about agent states is determined by presence subscription. In this environment, SIP
Server controls the agent state based on SIP-level information.
EventAgentLogin
and
EventAgentReady
messages are generated if an endpoint registers with SIP Server using the
REGISTER
request, or if the endpoint submits the
PUBLISH request with the presence content indicating an open
status. If the endpoint terminates the SIP registration or submits the
PUBLISH
request indicating a closed
status, then SIP Server generates
EventAgentLogout.
All TEvents are generated on behalf of the agent with the Agent ID set to the same value as the DN name for which all SIP messages are received.
This functionality is disabled.
Changes Take Effect: Immediately
Enables an agent login using presence notification. See “Presence
Subscription” on page 161 for more information.
Note:
You must enable the subscribe-presence
option before enabling this option.
force-register
Default Value:
NULL
Valid Values: Any SIP endpoint address
Changes Take Effect: Immediately
Enables trunk registration and used as the
From
header in the
REGISTER
request.
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geo-location
Default Value: None
Valid Values: Any alphanumeric string
Changes Take Effect: Immediately
describes the possible DNs that can use this option:
Table 26: DN Configuration Objects
Device Type
Agent SIP endpoint
Media Gateway
Music-on-hold or Music-inqueue server (such as Stream
Manager)
Voice Treatment Server (such as Stream Manager)
Voice Recorder (such as
Stream Manager)
Multipoint Conference Unit
(such as Stream Manager)
Genesys DN Type
Extension
Trunk
VoIP Service
with service-type=music
VoIP Service
with service-type=treatment
VoIP Service
with service-type=recorder
VoIP Service
with service-type=mcu
Note:
Virtual resources such as Routing Points or ACD Queues must not use this option.
make-call-rfc3725-flow
Default Value:
2
Valid Values:
1, 2
Changes Take Effect: Immediately
Controls which SIP call flow to choose when a call is initiated by a
TMakeCall request. The specified value is equal to the call flow number as described in
RFC 3725. Only flow 1 and flow 2 from RFC 3725 are currently supported.
Note:
This option is enabled only when the option
is set to false
for that DN.
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no-answer-action
Default Value: none
Valid Values: none notready logout
SIP Server takes no action on agents when business calls are not answered.
SIP Server sets agents to
NotReady
when business calls are not answered.
SIP Server automatically logs out agents when business calls are not answered.
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Determines the action taken on an agent if the agent receives a SIP Server business call but fails to answer the call within the time defined in option
.
This option is defined on any
Agent Login object.
When set, the value overrides the global agent-no-answer-action
SIP Server configuration option for that agent.
Note:
If a call is abandoned before either agent-no-answer-action or
supervised-route-timeout expires (depending on which timer is applicable), SIP Server performs no action on this agent.
no-answer-overflow
Default Value: None
Valid Values: none recall release default
Any valid overflow destination
SIP Server does not attempt to overflow a call on an agent desktop when agent-no-answer-timeout
expires.
SIP Server returns the call to the last distribution device (the device reported in the
ThisQueue
attribute of the call) when agent-no-answer-timeout
expires.
SIP Server releases the call.
SIP Server stops execution of the current overflow sequence and continues with the SIP Server default overflow sequence, as defined by the relevant overflow option in the main SIP Server section.
SIP Server returns the call to the specified destination when the value set for the agent-no-answer-timeout option expires.
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Defines a sequence of overflow destinations (separated by comma) in the order listed:
1. When the first overflow destination fails, SIP Server attempts the next one in the list.
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2.
When all overflow destinations in the list fail, SIP Server abandons overflow. When the list of overflow destinations contains the value recall and the call is not distributed, SIP Server skips to the next destination in the list.
This option is defined in the
Switches
folder on any of the following objects:
•
Agent Login
•
DN
of type
Extension
•
DN
of type
ACD Position
When set, this option overrides any of the following global SIP Server configuration options for the object where it has been set (depending on configuration object type):
•
agent-no-answer-overflow if defined for an
Agent Login
object.
•
extn-no-answer-overflow
if defined for a
DN
of type
Extension
object.
•
posn-no-answer-overflow if defined for a
DN
of type
ACD Position object.
no-answer-timeout
Default Value: Same as value in corresponding global option
Valid Value: Any integer from
0
–
600
Changes Take Effect: Immediately
Related Feature: “No-Answer Supervision” on page 157
Defines the time (in seconds) that SIP Server waits for a call that is ringing on the device in question to be answered.
When the timer expires, SIP Server applies the appropriate overflow, and, in the case of agents, the appropriate
Logout
or
Not Ready action.
This option is defined in the
Switches
folder on any of the following objects:
•
DN
of type
Extension
•
DN
of type
ACD Position
•
Agent Login
object
When set to
0,
the NoAnswer Supervision feature for this device is disabled.
When set, this option overrides any of the following global SIP Server configuration options for the object where it has been set (depending on configuration object type):
•
agent-no-answer-timeout
if defined for an
Agent Login object.
•
extn-no-answer-timeout
if defined for a
DN
of type
Extension
object.
•
posn-no-answer-timeout
if defined for a
DN
of type
ACD Position
object.
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oos-check
Default Value:
0
Valid Values:
0
–
300
Changes Take Effect: Immediately
Related Feature: “Active Out-of-Service Detection” on page 93
Specifies how often (in seconds) SIP Server checks a device for out-of-service
status. This option can be used in conjunction with the oos-force
and
options, as follows:
•
When no response is received, and the
oos-force
option is configured, SIP
Server will mark a device as out of service when the oos-force
timeout expires.
•
When the recovery-timeout
option setting is less than the oos-check timeout, SIP Server will wait the amount of time specified as the recovery-timeout
value before checking the DN that was previously detected as out of service.
•
When the oos-check
option is set to
0,
the feature is disabled.
Note:
This option is only supported on the following DN types:
•
Voice over IP Service
•
Trunk
The oos-check
option is not applicable on internal DNs (DNs of type
Extension
or
ACD Position
).
oos-force
Default Value:
0
Valid Values:
0
–
30
Changes Take Effect: Immediately
Related Feature: “Active Out-of-Service Detection” on page 93
Specifies the time interval (in seconds) that SIP Server waits before placing a device that does not respond in out-of-service state when the
is enabled.
oosp-transfer-enabled
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When set to true,
SIP Server puts itself in the Out Of Signaling Path (OOSP) after the single-step transfer or routing to the external destination has been completed.
Note:
This option is configured for
Trunk
DNs only, and the caller DN or the
Trunk
DN must support the
REFER
method.
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out-rule-<n>
Default Value: No default value
Valid Value: Any valid string in the following format: in-pattern=<input pattern value>;out-pattern=<output pattern value>
Changes Take Effect: Immediately
Related Feature: “Class of Service” on page 127
Defines a rule to be applied to an inbound number. The two parts of the option value describe the input and output patterns in the rule. When configuring the
pattern values, follow the syntax defined in “Dialing Rule Format” on page 128
. See “Examples” on page 132 for examples of these rules as well as
detailed instructions for creating rules for your installation. For example, a value for this configuration option might look like this: out-rule-01 = in-pattern=0111#CABBB*ccD;out-pattern=ABD
override-domain
Default Value:
NULL
Valid Values: Any computer name
Changes Take Effect: Immediately
Enables an override of the specified computer name in the SIP
To:
header for a
DN. It is used to contact a particular DN in a domain in the
To:
header that is different than the SIP Server internal registrar computer name.
Note:
This option must be specified for the DN that represents Microsoft
Office Communicator behind LCS.
override-domain-from
Default Value:
NULL
Valid Values: Any computer name string
Changes Take Effect: Immediately
When set, SIP Server substitutes the computer name in the
URI
of the
From headers with the value of this option when it sends the initial
INVITE
message to a DN or
Trunk
DN.
override-call-type
Default Value:
0
Valid Values:
0 CallTypeUnknown
1 CallTypeInternal
2 CallTypeInbound
3 CallTypeOutbound
Changes Take Effect: Immediately
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Determines the value SIP Server will use as the
CallType
attribute for internal calls made directly to a DN of type
Routing Point
. When set to
0,
SIP Server specifies the
CallType
attribute as
Internal.
password
Default Value: None
Valid Values: Any alphanumerical string
Changes Take Effect: Immediately
In the endpoint configuration: Specifies the password for the SIP endpoint registration with the local registrar. If it is present, registration attempts are challenged and the password is verified. If it is not present, the registration is not challenged. The realm for password authentication is configured globally; there is one realm per SIP Server.
In the gateway configuration: Contains the password for gateway registration with the local registrar. This is used for incoming
REGISTER
requests, not for outgoing
INVITE
requests.
prefix
Default Value: None
Valid Values: Any alphanumerical string
Changes Take Effect: Immediately
In the MCU configuration:
Specifies the starting digits of the number that are used when sending calls to MCU. The full number is built as:
<prefix><connid>@<ipaddr>:port
. Typically, MCU servers require a prefix consisting of digits in order to identify a type of conference (for example, voice only, voice and video, and so on). Set the value as conf=
if Stream
Manager is used as the MCU.
In the gateway configuration: Contains the initial digits of the number that must match a particular gateway for that gateway to be selected. If multiple gateways match a number, the gateway with the longest prefix is selected.
preview-interaction
Default Value: false
Valid Values: true false
The protocol is enabled.
The protocol is disabled.
Changes Take Effect: Immediately for all future calls
Related Feature: “Preview Interactions” on page 164
Determines if the
Preview Interaction
protocol is enabled when incoming calls are diverted from a Routing Point.
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Note:
This option works with the
and
options to determine what action to take when a desktop does not respond to a preview interaction before the time expires.
priority
Default Value:
0
Valid Values: Any non-negative integer
Changes Take Effect: Immediately
Specifies the device priority for the device selection algorithm. A smaller value designates a higher priority. SIP Server will choose a device in round-robin fashion across all devices if more than one device with the same priority is configured. This option is used to control the device switchover during a failure, and to provide lowest-cost routing.
public-contact
Default Value: None
Valid Values: Any alphanumerical string
Changes Take Effect: Immediately
Contains the public host:port
pair for a softswitch. This is the public IP address of the softswitch. SIP Server uses this address to fill the destination
(
Refer-To
) address in
REFER
requests. On some switches, this is the same as the contact
address; if this is the case, you do not need to specify this parameter.
Note:
The public-contact
option is only applicable to Alcatel 5020.
record
Default Value: false
Valid Values: true, false
Changes Take Effect: When the next call is established on the DN
Related Feature: “Call Recording” on page 102
When set to true,
call recording begins automatically when the call is established on the DN. Call recording stops when the DN leaves the call.
recovery-timeout
Default Value:
0
Valid Values:
0
–
86400 seconds
Changes Take Effect: Immediately
Controls whether a device is taken out of service when an error is encountered, and for how long it is out-of-service. When set to
0, automatic out-of-service-on-error
functionality is disabled.
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rfc-2976-dtmf
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When this option is set to true
in a particular DN (type of
Trunk
or
Extension
) configuration, SIP Server will send DTMF tones in the RFC 2976 format to that device using the
INFO
request method when an agent issues a
TSendDTMF request.
Notes:
If a
TSendDTMF
request contains a string with multiple digits (for example,
12345#
), SIP Server issues multiple
INFO
requests (one per digit).
If a
TSendDTMF
request contains a string with multiple digits, and there are unsupported DTMF tones in this string (for example,
123a67
), SIP
Server still attempts to send the
INFO
request for each digit contained in the string, ignoring possible error responses from a gateway, and continuing to send subsequent digits.
refer-enabled
Default Value: true
Valid Values:
true, false
Changes Take Effect: With the next new call on this DN
Specifies whether the
REFER
method is sent to an endpoint. When set to
true, the
REFER
method is sent to:
•
The call party that originates a
TMakeCall
request.
•
The call party that initiates a consultation call.
•
The call party that is transferred to another destination during a single-step transfer.
When set to false,
SIP Server uses the re-
INVITE
method instead.
reinvite-requires-hold
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When set to true,
SIP Server enables the endpoint to be placed on hold by re-inviting it with a hold SDP.
Note:
This option prevents an audio delay during 3pcc (third-party call control) conferencing with an RTC-based endpoint.
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reject-call-incall
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When set to true,
a call attempt to a DN that is already on a call will be rejected, and the
Invalid Destination State (93) error message will be generated.
reject-call-notready
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When set to true, a call attempt to a DN at which an agent is in a
Logout
,
NotReady, or
AfterCallWork
state will be rejected, and the
Invalid
Destination State (93) error message will be generated.
Note:
The reject-call-incall
and reject-call-notready
options are applicable to the following T-Library requests:
•
TMakeCall
•
TInitiateTransfer
•
TInitiateConference
•
TSingleStepTransfer
•
TSingleStepConference
replace-prefix
Default Value:
0
Valid Values: Any non-negative integer
Changes Take Effect: Immediately
Contains the digits that are inserted in the DN instead of the prefix for the gateway. If this option value is absent, the number is not modified.
request-uri
Default Value: None
Valid Values: Any SIP URI
Changes Take Effect: Immediately
Specifies the value of the
Request-URI
address inside the
INVITE
message that is different from the address where the message will be sent. Any DN used by
SIP Server must be configured with this option if SIP Server will be using it to place a call to a DMX application.
When used with DMX, set the contact
field in the option configured for the
DN as the IP address of the DMX and the request-uri
option as actual address of the H.323 endpoint.
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In video support configuration: Creates a template for specifying the source of the video stream as the value of the
Request-URI
parameter in the
INVITE message: annc@<stream_manager_hostport>;play=<file)
reuse-sdp-on-reinvite
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When a call is routed to an endpoint, and this option is set to true in the destination endpoint configuration, SIP Server generates an offer by sending a re-
INVITE
message to the origination party (or to the MCU). When the origination party answers the offer, SIP Server sends the
INVITE
message with
SDP information to the destination.
Note:
The value must be set to true
when using EyeBeam version 1.1.
ring-tone-on-make-call
Default Value: true
Valid Values: true, false
Changes Take Effect: Immediately
Affects the
TMakeCall
request when using the re-
INVITE
procedure.
When set to true,
SIP Server connects the caller with an audio ringtone from
Stream Manager when the destination endpoint responds with a
180 Ringing message. In addition, the following options must also be configured for these scenarios:
•
The calling DN initiates a
TMakeCall
request must be configured with the following options:
refer-enabled
set to false
make-call-rfc3725-flow
set to
1
(see
)
•
The calling DN initiates a consultation call must be configured with the following option:
dual-dialog-enabled
set to false
(see
When the ring-tone-on-make-call
option is set to false,
there is no ring tone.
service-type
Default Value: None
Valid Values: Any string
Changes Take Effect: Immediately
Specifies the configured SIP device type or service (see
). See
Chapter 5, “SIP Devices Support,” on page 77 for more information on using
this option.
242 Framework 7.6
Chapter 8: SIP Server Configuration Options Agent Login–Level and DN-Level Options
Table 27: Service-Type Settings for SIP Devices
SIP Device Type or
Service
Genesys DN Type
Conference Server / MCU Voice over IP Service
Softswitch
Music-on-Hold servers
Treatment service
Recording service
Application service
Voice over IP Service
Voice over IP Service
Voice over IP Service
Voice over IP Service
Voice over IP Service
Service-Type
Setting
mcu softswitch music or moh treatment recorder application
sip-add-primary-call-id
Default Value: false
Valid Values: false, true
Changes Take Effect: Immediately
When set to true
, any
TMakeCall
request that is initiated for a DN already on call will start with an
INVITE
message that has a proprietary
P-gcti-primarycall-id header. The value of this header is the
CallID
SIP attribute from the primary SIP call dialog for that DN.
sip-busy-type
Default Value:
0
Valid Values:
0, 1, 2
Changes Take Effect: Immediately
When this option is set to
0
(the default), a busy tone is always played. When this option is set to
1
, a busy tone is played for a calling party only if a treatment is previously applied to a call or a call is originated by a 3pcc make
call operation, and the refer-enabled
option is set to false
. Otherwise, the rejected response is sent back to the calling party. When this option is set to
2
, a busy tone is not applied, and if SIP Server does not accept an
INVITE
session from a calling party, the rejected response is sent back to the calling party.
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sip-cti-control
Default Value: None
Valid Values:
talk, hold talk hold
The
TAnswerCall
request is issued against the DN, which means that the call is answered remotely by a T-Library client. The SIP method
NOTIFY (event talk)
is used. Otherwise, the
TAnswerCall
request is not supported.
The
THoldCall
request is processed by a
NOTIFY
( event hold
) message. The
TRetrieveCall
request is processed by a
NOTIFY
(event talk) message.
Changes Take Effect: Immediately
Specifies the behavior of DN which represents a SIP endpoint which supports the BroadSoft SIP Extension Event Package.
Note:
Both values are used simultaneously as a list of comma-separated values.
sip-ring-tone-mode
Default Value:
0
Valid Values:
0
,
1
Changes Take Effect: Immediately
With the option set to
0
, SIP Server connects Stream Manager to a call to play an audio ring tone. With the option set to
1
, SIP Server waits for a response from the called device, and connects Stream Manager to a call to play an audio ring tone, only when the returned response cannot be used as the offer to a calling device.
Note:
This option can be set at both the SIP Server Application level and at the Switch/DN level. The setting at the Switch/DN level takes precedence over the Application-level setting.
sip-server-inter-trunk
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Related Feature: “Trunk Optimization for Multi-Site Transfers” on page 166
When set to true
, depending on the scenario, SIP Server determines whether to complete the transfer operation using the
REFER
or
INVITE
request with the
Replaces
header.
244 Framework 7.6
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straight-forward
Default Value: false
Valid Values: true false
SIP Server sends a new
INVITE
message directly to the location specified in the
Contact
header.
SIP Server resolves the username
parameter for the
Contact header using information from the DN configuration.
Changes Take Effect: Immediately
Describes how SIP Server responds to the
300
or
302
SIP messages received from an endpoint or a server.
Note:
This option is required for GVP integration in Stand-Alone mode.
Trunk
DNs configured for use with Resource Manager and IP
Communication Server must contain the value true
.
subscription-id
Default Value:
NULL
Valid Values: Any valid string
Changes Take Effect: Immediately
Enables the subscription for multiple DNs with one
Subscribe
message. The value must be same as the user part of the
Subscribe URI
.
Note:
For integration with GVP, this value must be set to
GVP
.
subscribe-presence-domain
Default Value:
NULL
Valid Values: Any valid computer name on the softswitch
Changes Take Effect: Immediately
Related Feature: “Presence Subscription” on page 161
Specifies the subscription domain information for the
Trunk
DN. This option value will be used with the DN name to form the
SUBSCRIBE
request
URI
and the
To:
header.
subscribe-presence-from
Default Value:
NULL
Valid Values: Any valid SIP URI
Changes Take Effect: Immediately
Related Feature: “Presence Subscription” on page 161
Specifies the subscription endpoint information. This option value will be used to form the
From:
header in the
SUBSCRIBE
request to the softswitch.
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Note:
For softswitches such as Microsoft LCS and Asterisk, the username part of this SIP URI must not be configured in the softswitch.
subscribe-presence-expire
Default Value:
NULL
Valid Values: Any valid positive integer
Changes Take Effect: Immediately
Related Feature: “Presence Subscription” on page 161
Specifies the subscription renewal interval (in seconds).
subscribe-presence
Default Value:
NULL
Valid Values: publish
, or the name of the
Trunk
DN representing the softswitch
Changes Take Effect: Immediately
Related Feature: “Presence Subscription” on page 161
Enables presence subscription and mapping of a presence state to an agent state.
•
When set to publish,
SIP Server uses presence updates from a
PUBLISH
SIP request sent by a SIP Endpoint, and maps the presence state from the
PUBLISH
request to the agent state.
•
When set to the name of a
Trunk
DN that contains the subscription parameters is specified, the enable-agentlogin-presence
option (see
) must also be configured for the same
Trunk
DN.
userdata-map-filter
Default Value: None
Valid Values:
*:
A list of prefixes
All data is mapped.
A comma-separated list of prefixes used to identify the
UserData key-value pair to be mapped.
Changes Take Effect: Immediately
Related Feature: “Mapping SIP Headers and SDP Messages” on page 149
Specifies the names of the key-value pairs to be mapped. If this option is not specified, no data will be mapped.
use-register-for-service-state
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
Determines the
Extension
DN service state after it has been unregistered with
SIP Server.
246 Framework 7.6
Chapter 8: SIP Server Configuration Options GVP Integration Options
•
When set to true
, the DN is set to the
Out of Service
state in the following scenarios:
The SIP
REGISTER
request contains an
Expires
header value equal to
0
.
The SIP registration timer has expired.
An
EventDNOutOfService
message is generated to indicate that the DN is currently out of service.
•
When set to false
(or not configured), the DN service state is not set to the
Out of Service
state when it has been unregistered with SIP Server.
GVP Integration Options
This section describes a configuration option specific to the Genesys Voice
Platform (GVP) functionality with SIP Server. Configure this option in the extrouter
section on the
Options
tab for the SIP Server
Application
object in
Configuration Manager.
handle-vsp
Default Value: no
Valid Values: requests events
The ISCC component of SIP Server will attempt to translate requests related to this DN before submitting them to the service provider.
The ISCC component of SIP Server will attempt to process events received from the service provider before distributing them to SIP Server clients.
all no
The ISCC component of SIP Server will handle both the events and requests.
No processing will take place.
Changes Take Effect: Immediately
Specifies the way SIP Server will handle events from, and requests to, an external service provider registered for a DN using the
AddressType
attribute set to
VSP
.
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Chapter 8: SIP Server Configuration Options Reserved Options
Reserved Options
Warning!
The options documented in this section are reserved for Genesys
Engineering and their values cannot be changed.
Table 28: Reserved Configuration Options
Option Name
accept-dn-type
Option Section
Application level >
TServer
section backup-mode call-max-outstanding
Application level >
Application level >
TServer
TServer
section
section clid-withheld-name Application level >
TServer
section default-dn-type dn-del-mode emulate-login expire-call-tout
Application level >
TServer
section
Application level >
TServer
section
Application level >
TServer
section
Application level >
TServer
section
248 max-pred-req-delay override-switch-acw
Application level >
TServer
section
Application level >
TServer
section
Framework 7.6
Chapter 8: SIP Server Configuration Options Changes from 7.5 to 7.6
Table 28: Reserved Configuration Options (Continued)
Option Name Option Section
prd-dist-call-ans-time Application level >
TServer
section quiet-cleanup Application level >
TServer
section recall-no-answer-timeout reg-delay reg-silent rq-expire-tmout sync-emu-agent unknown-xfer-merge-udata
Application level >
TServer
section
Application level > link-control
section
Application level > link-control
section
Application level >
TServer
section level section
Application level >
TServer
section
Application level >
TServer
section
Changes from 7.5 to 7.6
lists the configuration options that:
•
Are new or changed in the 7.6 release of SIP Server
•
Have been added or changed since the most recent 7.5 release of this document
If a configuration option has been replaced with another that enables the same functionality, the new option name and its location in this chapter are noted.
Table 29: Option Changes from Release 7.5 to 7.6
Option Name Option
Values
Type of
Change
Details
alternate-call-enabled
Application Level > Options Tab > TServer Section
true, false New in 7.5,
Obsolete in
7.6
Set to true
by default.
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Chapter 8: SIP Server Configuration Options Changes from 7.5 to 7.6
Table 29: Option Changes from Release 7.5 to 7.6 (Continued)
Option Name
cpd-info-timeout delay-between-refresh-onswitchover delay-to-start-refresh-onswitchover enforce-external-domains mwi-mode music-in-conference file
Option
Values
0-30
Any timeout value
Any timeout value
A list of computer names or IP addresses
REGISTER,
SUBSCRIBE
A path to a music file
Type of
Change
Details
New in 7.5
See the option description on
New in 7.5
See the option description on
New in 7.5
See the option description on
New in 7.5
See the option description on
New in 7.6
See the option description on
New in 7.6
See the option description on
observing-routing-point parking-music sip-block-headers sip-dtmf-send-rtp sip-enable-call-info sip-enforce-sdp-origin-rules sip-server-inter-trunk sip-sync-local-contact sip-sync-peer-contact userdata-map-trans-prefix
7.6
Any valid DN New in 7.6 See the option description on
music/silence New in 7.6
See the option description on
A commaseparated string
New in 7.5
See the option description on
true, false true, false
New in 7.5
New in 7.6
Obsolete in
7.6
See the option description on
See the option description on
true, false
A string
New in 7.6
See the option description on
Obsolete in
7.6
Obsolete in
7.6
New in 7.6
See the option description on
250 Framework 7.6
Chapter 8: SIP Server Configuration Options Changes from 7.5 to 7.6
Table 29: Option Changes from Release 7.5 to 7.6 (Continued)
Option Name Option
Values
Type of
Change
Details
default-network-call-idmatching
Application Level > Options Tab > extrouter Section
sip New in 7.6
See the option description on
handle-vsp request, events, all, no
New in 7.6
See the option description on
Application Level > Options Tab > UPDATE, INVITE, INFO, REFER Sections
extensions-<n> userdata-<n> capacity capacity-group header/header parameter header/header parameter
New in 7.6
New in 7.6
See the option description on
See the option description on
DN Level > Annex Tab > TServer Section
New in 7.6
See the option description on
Any positive integer
A non-empty string
New in 7.6
See the option description on
cpn default-dn out-rule-<n> override-call-type rfc-2976-dtmf reuse-sdp-on-reinvite
SIP URI format New in 7.6
See the option description on
Any valid DN New in 7.6
See the option description on
A string
0-4
New in 7.6
New in 7.5
See the option description on
See the option description on
sip-busy-type userdata-map-filter true, false true, false
New in 7.5
See the option description on
Modified in
7.6
The option functionality modified in
7.6. See the option description on
.
0, 1, 2
*:
or a commaseparated string
New in 7.6
See the option description on
New in 7.6
See the option description on
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252 Framework 7.6
Part
2
Part Two: T-Server
Common Functions and Procedures
Part Two of this SIP Server Deployment Guide familiarizes the reader with
T-Server in general. It addresses architectural, functional, and procedural information common to all T-Servers.
The information in Part Two is divided into the following chapters:
•
Chapter 9, “T-Server Fundamentals,” on page 255 , describes T-Server, its
place in the Framework 7 architecture, T-Server redundancy, and multi-site issues. It does not, however, provide configuration and installation information.
•
Chapter 10, “Multi-Site Support,” on page 267
, describes the variations available for T-Server implementations across geographical locations.
•
Chapter 11, “Common Configuration Options,” on page 321
, describes log configuration options common to all Genesys server applications.
•
Chapter 12, “T-Server Common Configuration Options,” on page 343
, describes configuration options common to all T-Server types including options for multi-site configuration.
SIP Server—Deployment Guide 253
Part Two: T-Server Common Functions and Procedures New for All T-Servers in 7.6
New for All T-Servers in 7.6
Before looking at T-Server’s place in Genesys solutions and in the architecture of the Genesys Framework, note the following general changes that have been implemented in the 7.6 release of T-Server:
•
ISCC Transaction Monitoring support. This release of T-Server supports the ISCC Transaction Monitoring that allows T-Server clients to monitor ISCC transactions of the call data transfer between T-Servers in a
multi-site environment. See “ISCC Transaction Monitoring Feature” on page 306
for details.
•
ANI information distribution control. This release introduces a new configuration option that controls the distribution of the ANI information in
TEvent
messages. See “ani-distribution” on page 344 for details.
•
Enhancement of use-data-from configuration option. This option now includes the new valid value active-data-original-call.
See “use-datafrom” on page 354
for details.
•
Enhanced agent session ID reporting. T-Server now generates and reports a session ID associated with each new agent login (key
AgentSessionID in
AttributeExtensions)
in agent-state events
(
EventAgentLogin, EventAgentLogout, EventAgentReady,
and
EventAgentNotReady)
, and also in the
EventRegistered
and
EventAddressInfo
messages for resynchronization. The agent session IDs are not synchronized with a backup T-Server and new agent session IDs will be assigned to existing agent sessions after a T-Server switchover. See the T-Server client’s documentation for agent session ID reporting. Refer to the Genesys 7 Events and Models Reference Manual and/or Voice
Platform SDK 7.6 .NET (or Java) API Reference for details on the key
AgentSessionID in
AttributeExtensions
.
•
Client-side port definition support. This release of T-Server supports a new security feature that allows a client application to define its connection parameters before connecting to the server application. Refer to the
Genesys 7.6 Security Deployment Guide for details.
Notes:
• Configuration option changes common to all T-Servers are
described in “Changes from Release 7.5 to 7.6” on page 366
.
• For information about the new features that are available in your
T-Server in the initial 7.6 release, see Part Two of this document.
254 Framework 7.6
Chapter
9
T-Server Fundamentals
This chapter presents overview information that applies to all T-Servers (and
Network T-Servers) and their deployment. This chapter is divided into the following sections:
Learning About T-Server, page 255
z z z z z
Advanced Disconnect Detection Protocol, page 261
z
Learning About T-Server
The Framework 7.6 Deployment Guide provides you with a high-level introduction to the role that T-Server plays in the Genesys Framework. If you have already looked through that guide, you may recall that T-Server is the most important component of the Framework Media Layer (the other two components are Load Distribution Server (LDS) and HA Proxy). The Media
Layer enables Genesys solutions to communicate with various media, including traditional telephony systems, voice over IP (VoIP), e-mail, and the
Web. This layer also provides the mechanism for distributing interaction-related business data, also referred to as attached data, within and across solutions.
Framework and Media Layer Architecture
illustrates the position Framework holds in a Genesys solution.
SIP Server—Deployment Guide 255
Chapter 9: T-Server Fundamentals Learning About T-Server
Agent
Desktop
IVR
Genesys
Framework
Genesys
Applications
Switch
Figure 28: Framework in a Genesys Solution
Moving a bit deeper,
presents the various layers of the Framework architecture.
Framework
Services Layer
Management
Layer
Configuration
Layer
Media Layer
(including
T-Server)
Routing
Services
Reporting
Services
Figure 29: The Media Layer in the Framework Architecture
T-Server is the heart of the Media Layer—translating the information of the media-device realm into information that Genesys solutions can use. It enables your contact center to handle the computer-based form of the interactions that arrive and it translates the information surrounding a customer contact into reportable and actionable data.
presents the generalized architecture of the Media Layer.
256 Framework 7.6
Chapter 9: T-Server Fundamentals Learning About T-Server
Internet
Media
Interaction
Server
Traditional
Telephony
T-Server
VoIP
Telephony
T-Servers for
IP Solutions
SOLUTIONS
Figure 30: Media Layer Architecture
In addition to being the most important component of the Media Layer,
T-Server plays the most significant role in making information about telephony traffic and its data available to Framework as a whole.
One or more components in practically every solution are T-Server clients.
Solutions comprise a number of different Genesys software packages, from collections of components for various types of routing to those that allow for outbound dialing to still others. Framework in general, and T-Server in particular, enable these solutions to function in your enterprise.
T-Server has several typical clients: Stat Server, Call Concentrator, Universal
Routing Server, and agent desktop applications. T-Server gets the information it needs about the enterprise from Configuration Server. Additionally, if you use the Management Layer, T-Server provides its ongoing status and various other log messages to server components of the Management Layer (for instance, allowing you to set alarms).
T-Server Requests and Events
This section outlines the roles that T-Server plays in a contact center. While it is possible to describe roles for all T-Servers, at a detailed level, T-Server’s functionality depends on the hardware to which it is connected. (For example, when connected to a traditional switch, it performs CTI functions, but when connected to a VOIP-based telephony device, it controls IP traffic.) The CTI connection is only for the switch.
Details of T-Server Functionality
T-Server is a TCP/IP server that enables intelligent communication between media-specific protocols (such as the various CTI protocols, including CSTA and ASAI) and TCP/IP-based clients of T-Server. Applications that are clients
SIP Server—Deployment Guide 257
Chapter 9: T-Server Fundamentals Learning About T-Server to T-Server use the T-Library format to transmit requests to T-Server through a
TCP/IP socket. T-Server can then either translate those requests to CTI protocol for switch use or relay them directly to other TCP/IP clients.
T-Server performs three general functions in the contact center: Bridging,
Messaging, and Interaction Tracking.
Bridging
T-Server acts as a platform-independent interface between media devices and business applications. In the case of a telephony device, for instance, it receives messages from and sends commands to the telephony equipment using either CTI links provided by the switch manufacturer or interface protocols provided by telephony network vendors.
On the client-application end, T-Server offers three models (call model, agent model, and device model) unified for all switches. The core functionality (such as processing an inbound call, an agent login, or a call-forwarding request) translates into a unified application programming interface (API) called
T-Library, so that applications do not need to know what specific switch model they are dealing with. On the other hand, T-Library accommodates many functions that are unique to a specific switch, so that client applications are able to derive the maximum functionality offered by a particular switch.
Refer to the Genesys 7 Events and Models Reference Manual for complete information on all T-Server events and call models and to the
TServer.Requests
portion of the Voice Platform SDK 7.6 .NET (or Java) API
Reference for technical details of T-Library functions.
Messaging
In addition to translating requests and events for the client application involved in an interaction, T-Server:
•
Provides a subscription mechanism that applications can use to receive notifications about interaction-related and non-interaction-related events within the contact center.
•
Broadcasts messages of major importance (such as a notification that the link is down) to all clients.
•
B
roadcasts messages originated by
a T-Server
client to other T-Server clients.
The subscription mechanism consists of two parts, the DN subscription and event-type masking. Applications must register for a DN or a set of DNs to receive notifications about all events that occur in association with each registered DN. For example, when two softphone applications are registered for the same DN, and the first application initiates a call from the DN,
T-Server notifies both applications that the call is initiated from the DN.
Client applications can also specify one or more types of events, and T-Server will filter out events of the non-specified types and only send events of the
258 Framework 7.6
Chapter 9: T-Server Fundamentals Learning About T-Server requested types. For example, if agent supervisors are interested in receiving agent-related events, such as
AgentLogin
and
AgentLogout,
they have to mask
EventAgentLogin
and
EventAgentLogout,
provided that a particular T-Server supports these events.
The combination of each client’s subscription for DNs and masking of event types defines what messages T-Server distributes to what client.
Interaction Tracking
T-Server maintains call information for the life of the call (or other
T-Server-supported media type) and enables client applications to attach user data to the call. Call information includes:
•
A unique identifier, connection ID, that T-Server assigns when creating the call.
•
Automatic Number Identification (
ANI
) and Dialed Number Identification
Service (
DNIS
), if reported by the CTI link.
•
User data that a client application (such as an Interactive Voice Response unit or Genesys Universal Routing Server) provides.
Difference and Likeness Across T-Servers
Although Figure 30 on page 257
(and other figures) depicts T-Server that works with telephony systems as a single product, this is a simplification.
Because almost every traditional telephony device has its own characteristics and communication protocols, Genesys makes different T-Servers for different telephony systems. (That means T-Server you have will not work with another switch.) Thus, all T-Servers play a common role in the architecture, but their specific features differ from implementation to implementation, based on the media device in use.
Despite their switch-based differences, T-Servers for telephony systems are similar to one another in at least one important respect: they are all built with a certain amount of shared software code. This shared code is rolled into a single unit and is called T-Server Common Part (TSCP). TSCP is the central, common component for all T-Servers and has its own Release Note, which is accessible via a hyperlink from your T-Server’s Release Note.
Note:
This document separates common-code features based on TSCP into separate sections and chapters, such as the “T-Server Common
Configuration Options” chapter. These are the options for all T-Servers that TSCP makes available for configuration.
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Switch
Learning About T-Server
T-Server Functional Steps During a Sample Call
The following example, Figure 31 , outlines some basic steps that T-Server
might take when a call arrives from outside the contact center. In this scenario,
T-Server starts tracking the call even before it is delivered to the agent.
T-Server then informs the selected agent that a call has arrived. When the switch delivers the call to the agent’s extension, T-Server presents account information, collected at an Interactive Voice Response (IVR) unit, to the agent at the agent desktop application.
1. Switch notifies
T-Server of call arrival. T-Server assigns ConnID.
2. Switch delivers call to IVR for automated service
T-Server
6. T-Server notifies agent application of call.
7. T-Server delivers call data to the desktop.
Agent Desktop
Application
3. IVR asks T-Server to attach account number to call.
4. IVR sends call to ACD queue.
IVR
5. ACD diverts call to agent’s DN.
Agent DN
Figure 31: Functional T-Server Steps
Step 1
When the call arrives at the switch, T-Server creates a call in its internal structure. T-Server assigns the call a unique identifier, connection ID.
Step 2
The switch delivers the call to an Interactive Voice Response (IVR) unit, which begins automated interactions with the caller.
Step 3
IVR acquires user information from the caller through prompts and requests
T-Server to attach that information to the call. T-Server updates the call with the user information.
Step 4
IVR sends the call to an ACD (Automated Call Distribution) queue.
260 Framework 7.6
Chapter 9: T-Server Fundamentals Advanced Disconnect Detection Protocol
Step 5
The ACD unit distributes the call to an available agent logged in to a particular
DN (directory number).
Step 6
T-Server notifies the agent desktop application that the call is ringing on the agent’s DN. The notification event contains call data including
ANI, DNIS,
and account information that the IVR has collected.
Step 7
The agent desktop application presents the account information, including the name of the person whose account this is, on the agent’s screen, so that the agent answering the call has all the relevant information.
These seven steps illustrate just a small part of T-Server’s bridging, messaging, and interaction-processing capabilities.
Advanced Disconnect Detection Protocol
Since the 6.0 release of T-Server, the Advanced Disconnect Detection Protocol
(ADDP) has replaced the Keep-Alive Protocol (KPL) as the method to detect failures for certain T-Server connections, including connections between two
T-Servers and between a T-Server and its clients.
Notes:
• Starting with release 7.5, the KPL backward-compatibility feature is no longer supported.
• ADDP applies only to connections between Genesys software components.
With ADDP, protocol activation and initialization is made on the client’s side and you can change these parameters. No additional messages are sent when there is existing activity over the connection. T-Server client applications and the remote T-Server (if any) must be listening to the socket and respond promptly to the polling signal for the connection to be preserved.
If you are going to enable ADDP, you must do it using the
,
and
configuration options.
When configuring a timeout, consider the following issues:
•
The configured timeout must be at least twice as long as the maximum network latency.
•
There may be an interval when T-Server does not check for network activity.
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Chapter 9: T-Server Fundamentals Redundant T-Servers
•
If the link connection fails but the client is not notified (for example, because the host is turned off, or because a network cable is unplugged), the maximum reaction time to a link-connection failure is equal to double the configured timeout plus the established network latency.
Also keep in mind that the T-Server receiving the polling signal may not respond immediately, and that a delay occurs between the polling signal and the response to travel from one T-Server to another. If you don’t account for these contingencies when configuring a timeout, the connection that ADDP is monitoring will be dropped periodically.
Redundant T-Servers
T-Servers can operate in a high-availability (HA) configuration, providing you with redundant systems. The basics of each T-Server’s redundant capabilities differ from T-Server to T-Server. One basic principle of redundant T-Servers is the standby redundancy type, which dictates how quickly a backup T-Server steps in when the primary T-Server goes down.
The Framework Management Layer currently supports two types of redundant configurations: warm standby
and hot standby
. All T-Servers offer the warm standby
redundancy type and, starting with release 7.1, the hot standby redundancy type is implemented in T-Servers for most types of switches. (See
.)
Instructions for configuring T-Server redundancy are available in Chapter 3,
“High-Availability Configuration and Installation.” Specifics on your
T-Server’s HA capabilities are outlined in Part Two of this document.
Notes:
• Network T-Servers use a load-sharing redundancy schema instead of warm
or hot standby
. Specifics on your T-Server’s HA capabilities are discussed in Part Two of this document.
• IVR Server does not support simultaneous configuration of both
Load Balancing functionality and warm standby
. Only one of these is supported at a time.
Support for Hot Standby Redundancy in Various T-Servers
Use
to determine whether your T-Server supports the hot standby redundancy type. The table also indicates whether HA Proxy components are required for this support, and, if so, how many are required per pair of redundant T-Servers (or per link if so noted).
262 Framework 7.6
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only summarizes hot standby
redundancy support in various
T-Servers. For detailed, up-to-date information on the subject, see the Genesys
Supported Media Interfaces white paper located on the Technical Support website at http://genesyslab.com/support/dl/retrieve/ default.asp?item=A9CB309AF4DEB8127C5640A3C32445A7&view=item
.
Table 30: T-Server Support of the Hot Standby Redundancy Type
T-Server Type
Alcatel A4200/OXO
Alcatel A4400/OXE
Aspect ACD
Avaya Communication Manager
Avaya INDeX
Cisco CallManager
DataVoice Dharma
Digitro AXS/20
EADS Intecom
M6880
EADS Telecom M6500 eOn eQueue
Ericsson MD110
Fujitsu F9600
Huawei C&C08
Mitel SX-2000/MN-3300
NEC NEAX/APEX
Nortel Communication Server 2000/2100
Nortel Communication Server 1000 with
SCCS/MLS
Philips Sopho iS3000
Radvision iContact
Rockwell Spectrum
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Hot Standby
Supported
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
No
Yes
No
No
No
No
No
No
No
No
Yes b
, No c
No
HA Proxy
Required
No
No
No
No a
No
No
No
No
No d
—
No
Number of HA
Proxy Components
—
—
—
—
—
—
—
—
—
—
—
—
—
—
—
—
1 per link
—
1
—
—
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Table 30: T-Server Support of the Hot Standby Redundancy Type (Continued)
T-Server Type
Samsung IP-PCX IAP
Siemens Hicom 300/HiPath 4000 CSTA I
Siemens HiPath 3000
Siemens HiPath 4000 CSTA III
Siemens HiPath DX
SIP Server
Tadiran Coral
Teltronics 20-20
Tenovis Integral 33/55
AT&T
Concert
CRSP
DTAG
GenSpec
ISCP
IVR Server, using network configuration
KPN
Hot Standby
Supported
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Network T-Servers e
No
No
No
No
No
No
No
No
HA Proxy
Required
No
No
No
No
No
No
No
Yes
No
—
—
—
—
—
—
—
—
MCI
NGSN
Network SIP Server
Sprint
SR3511
Stentor
No
No
No
No
No
No
—
—
—
—
—
—
Number of HA
Proxy Components
—
—
—
—
—
—
—
1
—
—
—
—
—
—
—
—
—
—
—
—
—
—
—
264 Framework 7.6
Chapter 9: T-Server Fundamentals Multi-Site Support a. With release 7.1, T-Server for Avaya Communication Manager no longer uses HA Proxy for its support of hot standby
. Earlier releases of this T-Server require two HA Proxies (for which there is a Configuration
Wizard) to support hot standby
.
b. For T-Server for Nortel Communication Server 2000/2100 in high-availability ( hot standby
) configuration,
Genesys recommends that you use link version SCAI14 or above with call-progress and noncontroller-released messages enabled. See the switch-specific information in Part 2 of this Deployment Guide for additional information on HA configurations. c. Starting with release 7.5, T-Server for Nortel Communication Server 2000/2100 supports HA without HA
Proxy when operating in Dual CTI Links mode. See the switch-specific information in Part 2 of this De-
ployment Guide for additional information on HA configurations.
d.
Starting with release 6.5.3, T-Server for Philips Sopho iS3000
Proxy.
supports HA both with and without HA e. Although they do not support high availability per se, Network T-Servers do support a load-sharing schema.
Multi-Site Support
Multi-site configuration implies the existence of two or more switches that belong to the same enterprise or service provider, and that share the Genesys
Configuration Database. (In some cases this may include isolated partitions on a given switch served by different T-Servers.) The main goal of T-Server support for multi-site operations is to maintain critical information about a call as it travels from one switch to another.
For instructions on installing and configuring a multi-site environment, including information on the Inter Server Call Control (ISCC) features, please
see Chapter 10, “Multi-Site Support,” on page 267 .
Agent Reservation
T-Server provides support for clients to invoke the agent reservation function,
TReserveAgent().
This function allows a server application that is a client of
T-Server to reserve a DN along with an agent, a
Place,
or both, so that no other T-Server client can route calls to it during a specified reservation
interval. Alternatively, when clients use the ISCC feature (see “ISCC Call Data
), they can use an agent reservation embedded in an ISCC request. (To do so, clients have to specify a certain
Extensions
attribute in an ISCC request when initiating an ISCC transaction. See page 274
for the list of ISCC requests.)
The reservation does not currently prevent the reserved objects from receiving direct calls or calls distributed from ACD Queues; agent reservation is intended as a way of synchronizing the operation of several clients. See
RequestReserveAgent
in the Voice Platform SDK 7.6 .NET (or Java) API
Reference for more details on this function from the client’s point of view.
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In addition to invoking the TReserveAgent function, you can customize the
Agent Reservation feature by configuring options in the T-Server
Application
object. See“Agent-Reservation Section” on page 351
in the “T-Server
Common Configuration Options” chapter in Part Two for more details.
Client Connections
The number of connections T-Server can accept from its clients depend on the operating system that T-Server runs.
illustrates the number of client connections that T-Server support.
Table 31: Number of T-Server’s Client Connections
Operating System
AIX 32-bit and 64-bit modes
(versions 5.1, 5.2, 5.3)
HP-UX 32-bit and 64-bit modes
(versions 11.0, 11.11, 11i v2)
Linux 32-bit mode
(versions RHEL 3.0, RHEL 4.0)
Solaris 32-bit mode
(versions 2.7, 8, 9)
Solaris 64-bit mode
(versions 2.7, 8, 9, 10)
Tru64 UNIX
(versions 4.0F, 5.1, 5.1B)
Windows Server 2003
Number of Connections
32767
2048
32768
4096
65536
4096
4096
266 Framework 7.6
Chapter
10
Multi-Site Support
This chapter contains general information about multi-site environments, as well as information on deploying a multi-site environment for your T-Server. z z z z z z z z z
This chapter is divided into the following sections:
Multi-Site Fundamentals, page 268
ISCC Call Data Transfer Service, page 269
Number Translation Feature, page 291
Network Attended Transfer/Conference Feature, page 299
Event Propagation Feature, page 301
ISCC Transaction Monitoring Feature, page 306
Configuring Multi-Site Support, page 307
Note:
Each switch/T-Server combination offers different multi-site options.
For details describing your specific switch/T-Server environment, refer
to Chapter 12, “T-Server Common Configuration Options,” on page 343
.
The following instructions apply to both local and remote switches and
T-Servers. Because different vendor switches can be installed at the local and remote locations, this chapter covers several, but not all, possible configurations. To help determine which sections of this chapter apply to your
situation, refer to Table 32 on page 283
For more information on your specific switch/T-Server environment, refer to the multi-site topics in Part Two of this guide.
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Chapter 10: Multi-Site Support Multi-Site Fundamentals
Multi-Site Fundamentals
A multi-site configuration has two or more switches that belong to the same enterprise or service provider and that share the Genesys Configuration
Database. (In some cases, this may include isolated partitions on a given switch served by different T-Servers.) The main goal of T-Server support for multi-site operations is to maintain critical information about a call as it travels from one switch to another.
T-Server supports multi-site operations using its Inter Server Call Control
(ISCC; formerly called External Routing), which supports the following functions:
•
Call matching—To link instances of a call distributed across multiple sites and to re-attach essential data associated with the call (
ConnID
,
UserData
, call history). The following T-Server features support this capability:
ISCC Call Data Transfer Service (active external routing)—when requested by a T-Server client by specifying the desired destination in the location
parameter, and also with various ISCC strategies performed by direct dial or by using the Transfer Connect Service. See
“ISCC Transaction Types” on page 274
.
Inter Server Call Control/Call Overflow (ISCC/COF) feature (passive external routing)—applicable when calls are overflowed to another site
either directly or manually (see page 287
).
Number Translation feature (see
).
Network Attended Transfer/Conference (NAT/C) feature (see
).
Note:
When ISCC detects call instance reappearance on a given site, the call is assigned a unique
ConnID
and the user data is synchronized with the previous call instances. This ensures that
ConnID s assigned to different instances of the same call on a given site are unique.
•
Call data synchronization between associated call instances (ISCC
Event Propagation)—To provide the most current data to call instances residing on remote T-Servers. The following T-Server features support this capability:
User Data propagation (see page 301 )
Party Events propagation (see page 303 )
Note:
ISCC automatically detects topology loops and prevents continuous updates.
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Note:
In distributed networks, Genesys recommends using call flows that prevent multiple reappearances of the same call instance, and call topology loops. This approach ensures that all T-Servers involved with the call report the same
ConnID
, and also optimizes telephony trunk allocation (that is, it prevents trunk tromboning).
The T-Server configuration contains information about other T-Servers with which it will communicate. T-Server uses this information to connect with the other T-Servers. During this “handshake” process, T-Servers exchange information about the following parameters:
•
Protocol type
•
Switch type
•
Server name
•
Location name (switch name)
•
T-Server role (primary or backup)
To complete the handshake process, T-Servers exchange messages about the current condition of the links to their switches. After the handshake process is complete, T-Server is ready to support a multi-site operation.
ISCC Call Data Transfer Service
Because ISCC supports active external routing, T-Servers that serve different switches (usually on different sites) can exchange call data when a call is passed from one switch to another. With this functionality, T-Server provides its clients with the following additional information about each call received from another switch:
•
The
ConnID
of the call
•
Updates to user data attached to the call at the previous site
•
Call history
Note:
Load-sharing IVR Servers and Network T-Servers cannot be designated as the destination location for ISCC.
shows the steps that occur during a typical external routing (ISCC) transaction. Note that the location where a call is initially processed is called the origination location, and the location to which the call is passed is called the destination location.
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Origination
Location
Origination Switch
6. Call
Passing
5. Telephony
Request to Move
Call to Destination
Location
1. T-Library
Request for ISCC routing
T-Server w/
ISCC feature
10. EventRemote-
Connection-
Success/Failure
T-Server Client
2. Routing Service
Availability Request
4. Routing Service
Availability Response
Destination Switch
9. Processing
Result Notification
7. Alerting Event
Processing
Destination
Location
3. Routing Service
Availability Check
T-Server w/
ISCC feature
8. Call Arrival
Processing
Figure 32: Steps in the ISCC Process
ISCC Call Flow
The following section identifies the steps (shown in
) that occur during an ISCC transfer of a call.
Step 1
A client connected to the T-Server at the origination location requests this
T-Server to pass a call with call data to another location. For this purpose, the client must specify the location
parameter (Attribute
Location
) when calling a corresponding T-Library function. ISCC processes the following T-Library requests:
•
TInitiateConference
•
TInitiateTransfer
•
TMakeCall
•
TMuteTransfer
•
TRouteCall
•
TSingleStepTransfer
Step 2
Upon receiving a client’s request, the origination T-Server checks that the:
1. Connection to the destination T-Server is configured in the origination
T-Server
Properties
dialog box.
2. The connection to the destination T-Server is active.
3. The destination T-Server is connected to its link.
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4. The origination T-Server is connected to its link.
If these four conditions are met, the origination T-Server determines the transaction type that will be used for passing call data to another location in this transaction. The following possibilities exist:
•
The client can request what ISCC transaction type (or simply transaction
type) to use by specifying an appropriate key-value pair in the
Extensions attribute of the request. The key-value pair must have a key equal to isccxaction-type
and either an integer value as specified in the
TXRouteType enumeration (see the Voice Platform SDK 7.6 .NET (or Java) API
Reference) or a string value equal to one of the following: default
, route
, direct
(or direct-callid
), direct-network-callid
, direct-notoken
, direct-ani
, direct-uui
, direct-digits
, reroute
, dnis-pool
, pullback
, or route-uui
.
•
If the client does not specify the transaction type in the request or specifies the default
transaction type, T-Server checks the Switch configuration for the transaction type configured in the
Access Code
(or
Default Access
Code
) properties:
If the
Route Type
property of the
Access Code
is set to any value other than default
, T-Server uses the specified value as the transaction type.
If the
Route Type
property of the
Access Code
is set to the default
value, T-Server uses the first value from the list specified in the casttype
configuration option configured for the destination T-Server. If no value has been specified for the
option, the default value of route
is used as the transaction type.
Note:
See “Switches and Access Codes” on page 308 for more
information on Access Codes and Default Access Codes.
After the origination T-Server determines the requested transaction type, it determines if the destination T-Server supports this transaction type.
You must list the transaction types T-Server supports in the
configuration option.
The origination T-Server issues a request for routing service availability and sends it to the destination T-Server. The T-Server request contains data that should be passed along with the call to the destination location. This data includes the transaction type,
ConnID
,
UserData,
and
CallHistory.
The timer specified by the
configuration option is set when the
origination T-Server sends the request. If either the specified timeout expires or the call is abandoned before the origination T-Server receives a response from the destination T-Server, the operation is considered failed. In this scenario, the origination T-Server:
1. Generates a request to the destination T-Server to cancel the request for routing service.
2. Sends
EventError
to the client that requested the service.
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3. Deletes information about the request.
Step 3
The destination T-Server receives the request for routing service availability and checks the requested type of routing. Depending on the ISCC transaction type, it stores the request information and, when appropriate, allocates access resources for the coming call. For example, an External Routing Point is allocated when the transaction type is route
, and a DNIS number is allocated when the transaction type is dnis-pool
.
Note:
The
and
configuration options determine how resources are allocated. Refer to
Chapter 12, “T-Server Common Configuration Options,” on page 343
for option descriptions.
If resources are unavailable, the request is queued at the destination location until a resource is free or the origination T-Server cancels the request. If the request is canceled, the destination T-Server deletes all information about the request.
If resources are unavailable because of incorrect configuration, the destination
T-Server returns an error event to the origination T-Server.
Step 4
If resources are available, the destination T-Server generates a positive response and the timer is started for the interval specified by the
configuration option of the destination T-Server.
Step 5
If the origination T-Server receives a negative response, it sends an
EventError message to the client and clears all data about the request.
If the origination T-Server receives the confirmation about routing service availability, it processes the client’s request and sends a corresponding message to the switch. The timer on the origination T-Server is also started for
the interval specified by the timeout
configuration option of the destination
T-Server.
Step 6
The origination switch processes the T-Server request and passes the call to the destination switch.
Step 7
If the call arrives at the destination switch, the switch generates an alerting event.
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The destination T-Server waits for the call no longer than the interval specified by the timeout configured on the destination T-Server. If the call is not received at the destination location within this interval, the destination
T-Server issues a failure notification to the origination T-Server, deletes all data about the request, and, when appropriate, frees the resources previously allocated for the request.
If either the specified timeout expires or the call is abandoned before the origination T-Server receives a response from the destination T-Server, the operation is considered failed. In this case, the origination T-Server:
1. Generates a request to the destination T-Server to cancel the request for routing service.
2. Responds to the client that requested the service in one of the following ways:
If the origination T-Server has already sent a response to the request the client sent in Step 1, the origination T-Server supplements its response with
EventRemoteConnectionFailed
.
If the origination T-Server has not yet sent a response to the client, the origination T-Server sends
EventError
.
3. Deletes information about the request.
Step 8
If the destination T-Server matches the arrived call, it updates the
ConnID
,
UserData,
and
CallHistory
attributes with the data received in the request for routing service availability. The connection ID is updated as follows:
The arrived call is assigned the
ConnID
that is specified in the request for routing service availability, but only if this
ConnID
does not coincide with the
ConnID
of a call that has existed at the destination site. If two such
ConnID s are identical, the arrived call is assigned a new unique
ConnID
.
For direct-*
transaction types (where the asterisk stands for a callid
, uui
, ani
, or digits
extension), the call reaches the destination DN directly.
For the transaction types route and
route-uui
, the call first arrives at an
External Routing Point from which it is routed to the destination DN. The call info is updated when the call reaches the External Routing Point. An External
Routing Point is considered free when the first alerting event (
EventQueued
or
EventRouteRequest
) is distributed.
Please keep the following issues in mind when using the ISCC feature:
•
If routing from a dedicated External Routing Point to the destination DN fails, T-Server considers the transaction failed. However, the
ConnID,
UserData,
and
CallHistory
attributes are updated. Then, T-Server attempts to route the call to one of the Default DNs configured for this External
Routing Point.
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Chapter 10: Multi-Site Support ISCC Call Data Transfer Service
•
If the destination T-Server did not receive a request for routing service availability, but a call arrives at an External Routing Point, T-Server considers the call to be unexpected and routes the call to the DN specified by the
configuration option. When no alternative
targets are defined, the call remains at the External Routing Point until diverted by the switch or abandoned by the caller.
For reroute
and pullback
transaction types, the call returns to the network location. For the dnis-pool
transaction type, the call reaches the destination
DN directly.
Step 9
If, in Step 8, the call does not arrive within the configured timeout, or the transaction fails, the destination T-Server sends a notification of failure to the origination T-Server.
Otherwise, the destination T-Server notifies the origination T-Server that the routing service was successful and deletes all information about the request.
Step 10
The origination T-Server notifies the client that the routing service was successful (or failed) and deletes all information about the request.
ISCC Transaction Types
As switches of different types provide calls with different sets of information parameters, a single mechanism for passing call data between the switches is not feasible in some cases. Therefore, the ISCC feature supports a number of mechanisms for passing call data along with calls between locations. This section describes ISCC transaction type principles, identifies which transaction types are supported for each T-Server, and defines each transaction type
(beginning with “direct-ani” on page 276
).
It is important to distinguish the two roles that T-Servers play in an external routing (ISCC) transaction—namely origination T-Server and destination
T-Server.
•
The origination T-Server initiates an ISCC transaction. It prepares to send the call to another T-Server and coordinates the process.
•
The destination T-Server receives call data from an origination T-Server and matches this data to a call that will arrive at some time in the future.
The distinction between these roles is important because the range of telephony-hardware functionality often requires T-Servers to support two entirely different sets of ISCC transactions based on which of the two roles they play. For instance, it is very common for a particular T-Server to support many types of ISCC transactions when it takes on the origination role, but fewer when it takes on the role of a destination T-Server.
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The ISCC transaction type reroute
is a good example. Most T-Servers support
Reroute
as origination T-Servers, but very few support
Reroute
as destination
T-Servers.
Determining and Configuring Transaction Type
Support
You can find descriptions of these transaction types starting on page 276
. Use
Table 32 on page 283 to identify the transaction types your destination
T-Server supports. A blank table cell indicates that T-Server does not support a certain transaction type.
You can configure the transaction types specific to your T-Server as values of the
configuration option specified in the ISCC configuration section
extrouter
ISCC Transaction Type General Principles
Generally, since most of the ISCC implementation is done at the T-Server
Common Part (TSCP) code level, all T-Servers support certain ISCC transaction types. Any T-Server can act as the origination T-Server for the following transaction types:
•
direct-ani
,
•
direct-notoken
•
dnis-pool
•
pullback
•
reroute
,
•
route
(aliased as route-notoken
), the default transaction type,
The following transaction types are unevenly supported for both the origination and destination T-Server roles:
•
direct-callid
(aliased as direct
),
•
direct-digits
(reserved for Genesys Engineering)
•
direct-network-callid
,
•
direct-uui
,
•
route-uui
,
The
reroute
and pullback
transaction types are supported only for selected
T-Servers in the destination role. However, if you implement this support, other transaction types require additional configuration and testing—even those that would normally be supported by default.
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direct-ani
With the transaction type direct-ani
, the
ANI
network attribute is taken as the parameter for call matching. Properly configured switches and trunks can keep the
ANI
attribute when a call is transferred over the network. T-Server is capable of using this network feature for call matching.
Warnings!
• Depending on the switch platform, it is possible to inherit the
ANI attribute after routing a call to a remote destination, and after performing a Single-Step Transfer and other telephone actions.
However, ISCC only works properly in scenarios where the
ANI attribute on the destination T-Server is represented by exactly the same digit string as on the origination T-Server.
• Typically, the
ANI
attribute represents the original call identifier
(customer phone number), which guarantees that the attribute remains unique. However, you can use the non-unique-ani resource type to block ISCC from matching calls based on an ANI
Notes:
• Some switches, such as Nortel Communication Server 2000/2100
(formerly DMS-100) and Avaya Communication Manager
(formerly DEFINITY ECS (MV) , may omit the
ANI
attribute for internal calls—that is, for calls whose origination and destination
DNs belong to the same switch. If this is the case, do not use the direct-ani
transaction type when making, routing, or transferring internal calls with the ISCC feature.
• When the direct-ani transaction type is in use, the Number
Translation feature becomes active. See “Number Translation
for more information on the feature configuration.
• With respect to the direct
transaction types, Network T-Servers and load-sharing IVR Servers are not meant to play the role of destination T-Servers for call routing. Using Network T-Server with these transaction types requires special architecture.
direct-callid
With the transaction type direct-callid
, the call reaches the destination DN directly from another location, and the
CallID
of the call is taken as the attribute for call matching. When a call arrives at the final destination, the
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CallID
, and updates the call info if the
CallID matches.
Use this transaction type when the destination switch has the capability to assign to an incoming call the same network-wide unique
CallID
that the origination switch has already assigned to that call.
Notes:
• The direct-callid
transaction type is used only in conjunction with the
TRouteCall
and
TSingleStepTransfer
function calls. They are applied only to the call that is in progress, and do not apply to functions that involve in the creation of a new call (for example,
TMakeCall
.)
• For T-Server for Nortel Communication Server 2000/2100, the direct-callid
transaction type is also applied to the
TMuteTransfer function.
direct-network-callid
With the transaction type direct-network-callid
, the call reaches the destination DN directly from another location, and the
NetworkCallID
of the call is taken as the attribute for call matching. When a call arrives at the final destination, the destination T-Server identifies its
NetworkCallID
, and updates the call info if the
NetworkCallID
matches.
Use this transaction type when the destination switch has the capability to assign to an incoming call the same network-wide unique
NetworkCallID
that the origination switch has already assigned to that call.
Note:
To support this transaction type, you must configure
Target Type
and
ISCC Protocol Parameters
fields of the corresponding
Switch Access
Code
in the Configuration Layer. Refer to Part Two of this document for information about settings specific for your T-Server type.
direct-uui
With the transaction type direct-uui
, so-called user-to-user information (UUI) is taken as the attribute for call matching. Some switches make it possible to send a small data packet along with a call. T-Server can use this data to recognize a call passed from one switch to another. The destination T-Server generates a local unique value for
UUI
, and then notifies the origination
T-Server .
The origination T-Server uses a provided value to mark the call coming from the origination location. The destination T-Server receives a call and checks whether it is marked with an exact
UUI
value. If so, the call is considered as matched.
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On the Avaya Communication Manager and the Aspect ACD, UUI is referred to as “user-to-user information.” On the Siemens Hicom 300 switch with
CallBridge, UUI is referred to as “Private User Data.” On the Alcatel A4400/
OXE switch, UUI is referred to as “correlator data.”
Note:
To support this transaction type, you must configure your switches to pass the UUI provided by your T-Server. Moreover, the trunks involved must not drop this data.
direct-notoken
With the transaction type direct-notoken
, T-Server expects a call to arrive from another location to the destination DN specified in the request for routing service availability. When a call reaches the specified DN, T-Server processes the call as the expected externally routed call.
Notes:
• This matching criterion is weak because any call that reaches the specified DN is considered to be the expected call. Genesys recommends that you use this transaction type only in a contact center subdivision that can be reached from within the contact center only (for example, the second line of support, which customers cannot contact directly).
• With respect to the direct
transaction types, Network T-Servers and load-sharing IVR Servers are not meant to play the role of destination T-Servers for call routing. Using Network T-Server with these transaction types requires special architecture.
dnis-pool
With the dnis-pool
transaction type, T-Server reserves one of its DNIS access resources and waits for the call that has the same
DNIS
attribute as the name of the reserved DNIS access resource.
If the arrived call is matched successfully, the destination T-Server may update the value of the
DNIS
attribute of the call (along with
ConnID
,
UserData
, and
CallHistory
) with the value of the
DNIS
attribute of the original call. This occurs when the value of the
DNIS
attribute of the original call is specified as a value of the key-value pair
_ISCC_TRACKING_NUMBER_
in the
Extensions
attribute of the original client request.
The DNIS matching can be based on any number of digits out of all the digits that comprise the
DNIS
attribute. The number of digits that T-Server should use for DNIS matching is specified for the destination switch as the
ISCC Protocol
Parameters
property of the Switch Access Code. The value syntax should be as follows:
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For example, if this property is set to the dnis-tail=7
value, ISCC matches only the last seven digits of a
DNIS.
You must configure DNIS access resources in the switch; otherwise, ISCC fails to use this transaction type and sends
EventError
in response to the client application request.
Note:
The dnis-pool
transaction type is typically used for networks employing a “behind the SCP” architecture—network IVR. Network
T-Server for GenSpec and IServer are two examples of this, but other
Network T-Servers might also be used in this architecture.
In Load-Balancing Mode
When T-Server uses load balancing for call routing with the dnis-pool transaction type, the following processes occur:
1. A client of the origination T-Server sends a request to pass a call to the location with a DNIS access resource specified in the key-value pair iscc-selected-dnis.
2. The origination T-Server distributes the request for a routing service to all destination T-Servers.
3. The destination T-Servers receive the request and check that the specified
DNIS is not being used by another routing service request.
4. The origination T-Server expects to receive a positive response from each destination T-Server. If the origination T-Server receives a negative response from at least one T-Server, it sends an
EventError
to the client and clears all data about the request. If the origination T-Server receives the confirmation about routing service availability from all destination
T-Servers, it processes the client’s request and sends a corresponding message to the switch.
5. The origination switch processes the T-Server request and passes the call to the destination switch.
6. The call arrives at the destination switch, which generates an alerting event to one of the corresponding load-balanced destination T-Servers.
7. That destination T-Server processes the call and notifies the origination
T-Server that the routing service was successful and deletes all information about the request.
8. The origination T-Server sends a routing service request cancellation to all other destination T-Servers.
9. The origination T-Server notifies the client that the routing service has been successful and deletes all information about the request.
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pullback
Pullback
is used in the following scenario, for those T-Servers that support it:
1. A call arrives at Site A served by a Network T-Server.
2. At Site A, a Network T-Server client requests to pass the call by means of
ISCC routing to Site B served by a premise T-Server. Any transaction type except reroute
or pullback
can be specified in this request.
3. The call arrives at Site B and is either answered by an agent or delivered to a routing point.
4. A client of the premise T-Server at Site B sends a
TRouteCall
, or
TSingleStepTransfer
request to transfer the call to the network.
5. The Site B premise T-Server notifies the Network T-Server about this request.
6. The network T-Server receives the notification and issues an
EventRouteRequest
to obtain a new destination.
7. After receiving the new destination information, the Network T-Server disconnects the call from its current premise location at Site B and attempts to route the call to the new destination.
8. The Site B premise T-Server stops tracking the call, which has disconnected from the premise’s agent DN or routing point and is delivered to the network.
9. The network T-Server completes routing the call to its new destination.
Note:
The transaction type pullback
can be used only to return a call from a premise T-Server to the Network T-Server that serves the site from which the call was previously transferred.
reroute
Reroute
is used in the following scenario, for those T-Servers that support it:
1. A call arrives at Site A served by a Network T-Server.
2. At Site A, a Network T-Server client requests to pass the call by means of
ISCC to Site B served by a premise T-Server. Any transaction type except reroute
or pullback
can be specified in this request.
3. An agent at Site B answers the call.
4. A client of the premise T-Server at Site B sends a
TSingleStepTransfer
or
TRouteCall
request to transfer the call elsewhere (to a PSTN, to an agent, or to a routing point).
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5. The Site B premise T-Server notifies the Network T-Server about this request and releases the call leg that resides at the agent’s phone (using
TReleaseCall
) or at the Routing Point (using
TRouteCall
with the parameter
RouteTypeCallDisconnect
).
6. The Network T-Server receives the notification and reroutes the call to the requested destination—that is, it sends
EventRouteRequest
and attaches the call’s user data.
Notes:
• The transaction type reroute
can be used only to return a call from a premise T-Server to the Network T-Server that serves the site from which the call was previously transferred.
• To perform multi-site operations that are initiated with
TRouteCall and for which the reroute
transaction type is requested, the origination T-Server must support the
RouteTypeCallDisconnect subtype of
TRouteCall
.
route
With the transaction type route
(aliased as route-notoken), a call from the origination location reaches a dedicated External Routing Point, and from there, it is routed to a destination DN.
To control configured External Routing Points, T-Server must register these
DNs with the switch. Failure to register implies that the External Routing Point is not available for ISCC purposes. Client applications can register External
Routing Points via T-Server for monitoring purposes only.
Point-to-Point (One-to-One)
In the Point-to-Point access mode, only one trunk line is used to access an
External Routing Point (for example, VDN, CDN) at the destination site. See
.
Trunk Line 1
Trunk Line 2
Trunk Line 3
External Routing Point 1
External Routing Point 2
External Routing Point 3
Figure 33: Point-to-Point Trunk Configuration
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Note:
Dedicated DNs of the
External Routing Point
type must be configured
in a switch. See “Configuring Multi-Site Support” on page 307
.
Multiple-to-Point (Multiple-to-One)
In the Multiple-to-Point access mode, trunk lines are assigned to the destination switch’s trunk group, from which calls are routed to the final
Trunk Line 1
Trunk Line 2
External Routing Point
Trunk Line 3
Figure 34: Multiple-to-Point Trunk Configuration
With this configuration, all calls reach the same External Routing Point. The
DNIS
attribute of a specific call differs from that of other calls and uniquely identifies the trunk from which the call arrived.
Note:
To switch to this operating mode, you must configure the route-dn
configuration option for T-Server.
route-uui
The route-uui transaction type employs the dedicated External Routing Point feature of the route
transaction type (
page 281 ) and the UUI matching feature
of the direct-uui
transaction type ( page 277 ). This transaction type
accommodates those switches that require a designated External Routing Point even though they use UUI for tracking.
Note:
To support this transaction type, you must configure your switches to pass the UUI provided by your T-Server. Moreover, the trunks involved must not drop this data.
T-Server Transaction Type Support
shows which transaction types are supported by a specific T-Server.
Use this table to determine the transaction types that are available for use with
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you specify in the configuration options for your T-Server, and to any client-designated routetype requests specified for transfers of calls. A blank table cell indicates that
T-Server does not support a certain transaction type.
Table 32: T-Server Support of Transaction Types
T-Server
Type
Alcatel
A4200/OXO
Alcatel
A4400/OXE
Aspect ACD
Yes
Yes
Avaya
Communication Manager
Yes
Avaya INDeX Yes
Yes Cisco
CallManager
DataVoice
Dharma
Yes
Yes
Yes a,b,c
Yes
Yes
Yes
Transaction Type route one-toone multipleto-one reroute
Yes
directcallid directuui / routeuui directnotoken directani directdigits directnetworkcallid dnispool pullback
Yes Yes Yes
Yes
Yes d
Yes
Yes
Yes
Yes
Yes
Yes f
Yes
Yes
Yes
Yes
Yes e
Yes Yes Digitro
AXS/20
EADS
Intecom
M6880
EADS
Telecom
M6500
Yes
Yes
Yes eOn eQueue
Ericsson
MD110
Yes
Yes
Fujitsu F9600 Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
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Table 32: T-Server Support of Transaction Types (Continued)
T-Server
Type
Huawei
C&C08
Mitel SX-
2000/MN3300
route one-toone multipleto-one reroute directcallid directuui / routeuui directnotoken directani directdigits directnetworkcallid dnispool pullback
Yes Yes
Transaction Type
Yes Yes Yes Yes
Yes Yes Yes Yes NEC NEAX/
APEX
Nortel
Communication Server
2000/2100
Nortel
Communication Server
1000 with
SCCS/MLS
Philips Sopho iS3000
Radvision iContact
Rockwell
Spectrum
Samsung
IP-PCX IAP
Siemens
Hicom 300/
HiPath 4000
CSTA I
Siemens
HiPath 3000
Siemens
HiPath 4000
CSTA III
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
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Table 32: T-Server Support of Transaction Types (Continued)
AT&T
Concert
CRSP
DTAG
GenSpec
IVR Server, using network configuration
KPN
ISCP
MCI
NGSN
Network SIP
Server
Sprint
T-Server
Type
Siemens
HiPath DX
Yes
SIP Server Yes
Tadiran Coral Yes
Yes Teltronics
20-20
Tenovis
Integral 33/55
Yes
Transaction Type route one-toone multipleto-one reroute directcallid directuui / routeuui directnotoken directani directdigits directnetworkcallid dnispool pullback
Yes Yes Yes Yes
Yes
Yes
Yes
Yes
Yes g
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Network T-Servers
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes
Yes Yes
Yes
Yes
Yes
Yes
Yes
Yes
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Table 32: T-Server Support of Transaction Types (Continued)
T-Server
Type
Transaction Type route one-toone multipleto-one reroute directcallid directuui / routeuui directnotoken directani directdigits directnetworkcallid dnispool pullback
SR-3511
Stentor a. Not supported in the case of function
TRequestRouteCall
on a virtual routing point: a routing point can be simulated using a hunt group with calls being deflected or transferred from the hunt-group member when routing. When a two-step (typically mute) transfer is used on such a hunt-group member,
CallID
and
ANI
usually change; thus, the direct-callid
and direct-ani
types do not work.
b. Not supported in the case of function
TSingleStepTransfer
when the T-Server service is simulated using a two-step transfer to the switch. In this case,
CallID
and
ANI
change; thus, the direct-callid
and direct-ani
types do not work.
c. Not supported if two T-Servers are connected to different nodes.
d. There are some switch-specific limitations when assigning CSTA correlator data UUI to a call.
e. Supported only on ABCF trunks (Alcatel internal network).
f. To use this transaction type, you must select the
Use Override
check box on the
Advanced
tab of the DN
Properties
dialog box.
g. SIP Server supports the direct-uui
type.
Transfer Connect Service Feature
The Transfer Connect Service (TCS) feature supports transfer connect services available on some telephony networks. When this feature is enabled, ISCC passes user data to remote locations to which calls are transferred or conferenced using transfer connect services.
Procedure:
Activating Transfer Connect Service
Start of procedure
1. Open the T-Server
Application’s Properties
dialog box.
2. Click the
Options
tab.
always.
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configuration option to the number of a DN on the origination switch.
ISCC uses this DN as an intermediate step when sending calls to the remote location. The DN that is configured as tcs-queue
receives attached data indicating the Feature Access Code (FAC) needed to reach the remote site. After a call is directed to the DN with data, a monitoring application takes the data and generates the required DTMF (dual-tone multifrequency) tones to redirect the call through the network to the remote location.
5. When you are finished, click
Apply
.
6. Click
OK
to save your changes and exit the
Properties
dialog box.
End of procedure
Note:
With T-Server for Avaya Communication Manager, you can use
RequestRouteCall
with
RouteTypeOverwriteDNIS
to initiate the playing of DTMF tones. This is done through the use of another intermediate
DN (typically, an announcement port configured to give the silence treatment), to which the call is routed. When the call is established on this DN, T-Server requests that the digits sent in the
DNIS
field of the
TRequestRouteCall
be played via the
ASAI-send-DTMF-single procedure.
ISCC/COF Feature
The Inter Server Call Control/Call Overflow (ISCC/COF) feature of T-Server, that supports passive external routing, is specifically designed to handle calls delivered between sites by means other than ISCC. Such scenarios include contact center overflows and manual call transfers.
An overflow situation occurs when a call comes into a contact center where all agents are currently busy. In this situation, the switch can transfer (overflow) the incoming call to another site where there is an available agent.
T-Server uses two methods to handle call overflow and manual transfer scenarios. The first method is based on
NetworkCallID
matching and the second method is based on
ANI/OtherDN
matching.
When connected to each other via switch-specific networks, switches of some types can pass additional information along with transferred calls. This information may contain the
NetworkCallID
of a call, which is a networkwide unique identifier of the call.
When connected via a regular PSTN, switches of all types can send the
ANI and/or
OtherDN
attributes to the destination switch during any call transfer operation.
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While all T-Servers support the ISCC/COF feature using the
ANI
and/or
OtherDN
attributes, only a few support this feature using the
NetworkCallID attribute.
shows the T-Server types that provide the
NetworkCallID
of a call.
Table 33: T-Server Support of NetworkCallID for ISCC/COF
Feature
T
-
Server Type
Alcatel A4400/OXE
Aspect ACD
Avaya Communication Manager
Nortel Communication Server 2000/2100
Nortel Communication Server 1000 with
SCCS/MLS
Rockwell Spectrum
SIP Server
Supported NetworkCallID
Attribute
Yes
Yes
Yes
Yes
Yes
Yes
Yes
The ISCC/COF feature can use any of the three attributes (
NetworkCallID
,
ANI
, or
OtherDN
) as criteria for matching the arriving call with an existing call at another location. Consequently, the attribute that is used determines what
ConnID
,
UserData
, and
CallHistory
are received for the matched call from the call’s previous location.
Warning!
Depending on the switch platform, it is possible to inherit the ANI attribute after routing a call to a remote destination, and after performing a Single-Step Transfer and other telephone actions.
However, ISCC/COF works properly only in scenarios where the
ANI
attribute on the destination T-Server is represented by exactly the same unique digit string as on the origination T-Server.
Typically the
ANI
attribute represents the original call identifier
(customer phone number), which guarantees that the attribute remains unique.
Note:
When the ISCC/COF feature is in use, the Number Translation feature
becomes active. See “Number Translation Feature” on page 291 for
more information on the feature configuration.
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Site A
Agent
1. Call Made/
Transferred to
Another
Location
ISCC/COF Call Flow
shows the sequence of steps that occur in an ISCC/COF scenario when a call is made or transferred by an agent at Site A to a DN at Site B, or when a call is overflowed from Site A to Site B.
Origination Switch
Destination Switch
Site B
2. Call
Passing
T-Server w/
ISCC feature
4. Call Data
Request
6. Call Data
Passing
3. Arrived Call
Notification
T-Server w/
ISCC feature
8. Event
Distribution
5. Call Data Waiting
7. Call Data Update
9. Info Clearing
T-Server
Client
Figure 35: Steps in the ISCC/COF Process
Step 1
An agent makes or transfers a call manually to another location or a call is overflowed from Site A (origination location) to Site B (destination location).
Step 2
Switch A (the origination switch) passes the call to Switch B (the destination switch).
Step 3
The call reaches the destination switch, which notifies the destination T-Server about the arrived call.
Step 4
The destination T-Server verifies with remote locations whether the call was overflowed from any of them.
To determine which calls to check as possibly overflowed, T-Server relies on the
Switch
object configuration:
•
If no COF DNs (that is, DNs of the
Access Resources
type with the
Resource Type
set to cof-in
or cof-not-in
) are configured for the destination switch, the ISCC/COF feature of the destination T-Server checks all arriving calls.
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•
If a number of COF DNs are configured for the destination switch, one of three scenarios occurs:
If the COF DNs with the cof-in
setting for the
Resource Type
property are configured, the ISCC/COF checks for overflow only those calls that arrive to those cof-in
DNs that are
Enabled
.
If no DNs with the cof-in
setting for the
Resource Type
property are configured, but some DNs have the cof-not-in
setting for the
Resource
Type
property, the ISCC/COF checks for overflow only those calls that arrive to those cof-not-in
DNs that are
Disabled
.
If no DNs with the cof-in
setting for the
Resource Type
property are configured, some DNs have the cof-not-in
setting for the
Resource
Type
property, and some other DNs do not have any setting for the
Resource Type
property, the ISCC/COF checks for overflow only those calls that arrive to the DNs without any setting for the
Resource Type property.
•
In all other cases, no calls are checked for overflow.
To determine which location the call arrived from, T-Server checks the call type and checks whether the call has the
NetworkCallID
,
ANI
, or
OtherDN attribute:
•
If the call is not an inbound call, the request for call data is sent to all remote locations except those whose Switch Access Code has the ISCC
Call Overflow
Parameters
property set to inbound-only=true
.
•
If the call of any type has the
NetworkCallID
attribute, the destination
T-Server sends a request for call data to the remote locations of the same switch type as the destination location if their Switch Access Codes have the ISCC Call Overflow
Parameters
property set to match-callid
.
•
If the call of any type has the
ANI
or
OtherDN
attribute, the request for call data is sent to remote locations whose Switch Access Code has the ISCC
Call Overflow
Parameters
property set to match-ani
.
Step 5
The destination T-Server waits (suspending events related to that call) for the call data from the remote T-Server for the time interval specified in the
configuration option. Within this interval, T-Server holds any events related to the call. In addition, the
option on the origination T-Server establishes the time interval only after which that
T-Server deletes the call information. And the cof-ci-wait-all
,
if set to true, forces the origination T-Server to wait for responses related to possible call overflow situations before updating call data.
Step 6
The T-Server at the location from which the call was transferred or overflowed sends call data to the requesting T-Server.
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Step 7
If a positive response to the call-data request is received, T-Server updates
ConnID
,
UserData
, and
CallHistory
, distributes all suspended events related to that call and deletes all information regarding the transaction (Step 9).
Step 8
If the timeout set by
expires, T-Server distributes all
suspended events, and starts the timeout specified by the cof-rci-tout
If a positive response is received within the timeout set by cof-rci-tout ,
T-Server updates the
ConnID, UserData, and
CallHistory
and notifies client applications by distributing
EventPartyChanged.
Step 9
T-Server deletes all information regarding the transaction when one of these results occurs:
•
The first positive response to the call-data request is received.
•
Negative responses from all queried locations are received.
•
The timeout specified by the cof-rci-tout
Number Translation Feature
The Number Translation feature of T-Server extends the ISCC/COF and direct-ani
transaction type functions to provide more flexibility for handling calls distributed across multiple sites. T-Server translates the input string (ANI string) into a number defined by the translation rules. This processing is called number translation. T-Servers participating in handling calls at multiple sites exchange the translated numbers in order to match the call instances.
The translation process involves two algorithms, one for rule selection and the other for the actual translation. Through the first algorithm, T-Server selects a rule that will be used for number translation. Through the second algorithm,
T-Server translates the number according to the selected rule definition. See
“Number Translation Rules” on page 292
for more information on configuring rules for your environment.
Number translation occurs as follows:
1. The switch reports a number, typically via
AttributeANI
.
2. T-Server evaluates all configured inbound rules to determine which one is the best fit for the received number. The best fit is determined by comparing the length of, and the specific digits in, the input number with
the inbound pattern of each configured rule. See “Rule Examples” on page 297
for specific examples.
3. T-Server translates the number according to the selected rule.
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To enable T-Server to translate numbers, you must perform specific
configuration tasks that are associated with translation. See “Configuring
Number Translation” on page 298 .
Number Translation Rules
T-Server uses the number translation rules that you define in the T-Server configuration object in two ways:
•
Rule selection—To determine which rule should be used for number translation
•
Number translation—To transform the number according to the selected rule
Using ABNF for Rules
The number translation rules must conform to the following syntax, represented using Augmented Backus-Naur Form (ABNF) notation. For more information about ABNF, see RFC 2234, “Augmented BNF for Syntax
Specifications: ABNF.”
Note:
The notations are explained starting at the highest level, with the name of a component notation and a basic definition of each component that comprises it. Some components require more detailed definitions, which are included later in this section.
Common Syntax Notations
Syntax notations common to many of these rules include:
•
*
—Indicates that 0 to an infinite number of the item following this symbol are acceptable.
•
1*
—Indicates that one repetition is required. For T-Server, only one instance is acceptable.
•
/
—Indicates that any of the items mentioned, or a combination of those items, is acceptable.
Component Notations
Component notations include:
•
dialing-plan = *dialing-plan-rule
where: dialing-plan-rule
represents the name of the rule. Each rule must have a unique name. There are no other naming restrictions, and you do not need to model your names according to the examples in this chapter.
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The rules are represented as separate options in the configuration. Also, fields from a rule are represented as parameters in a single option string.
•
rule = [name] in-pattern [out-pattern]
where:
[name]
is the name for the rule option, for example, rule-01.
In ABNF notation, the brackets
[]
indicate that 0 or 1 instance of the component is required. However, for T-Server, a name is required.
in-pattern
is the part of the rule to which T-Server looks when attempting to match the input number.
[out-pattern]
is the part of the rule that instructs T-Server on how to translate the input number into the required format. The brackets indicate that either 0 or 1 instance is required. You must create an outpattern
for number translation rules.
•
name = *( ALPHA / DIGIT / "-")
where:
ALPHA indicates that letters can be used in the name for the rule option.
DIGIT indicates that numbers can be used in the name for the rule option.
“-”
indicates that a dash (
-
) can also be used in the option name, for example, rule-01.
•
in-pattern = 1*(digit-part / abstract-group)
where: digit-part
represents numbers. T-Server uses this when selecting the most appropriate rule from the entire dialing plan. abstract-group
represents one or more letters with each letter representing one or more numbers. T-Server uses this when transforming a dial string.
For example,
[1-9] is the digit-part
(representing a range of numbers) and
ABBB
is the abstract-group for in-pattern=[1-9]ABBB
.
•
out-pattern = 1*(symbol-part / group-identifier) *param-part
where: symbol-part
represents digits, symbols, or a combination. Symbols are rarely used. They are not used in the United States.
group-identifier
are letters that represent groups of numbers. A letter in the out-pattern
represents one or more digits, based on the number of times the letter is used in the in-pattern.
*param-part
represents an additional parameter, such as phonecontext.
Reminder: an asterisk means that 0 to an infinite number of these are acceptable.
For example, in rule-04; in-pattern=1AAABBBCCC;out-pattern=91ABC, 91 is the symbol-part
;
A, B,
and
C
are group-identifiers in the out-pattern
,
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.
Note:
Prefix an out-pattern
value with a plus sign (+) for the inbound rule when the output must be in a global form (E.164 format).
•
digit-part = digits / range / sequence
where: digits
are numbers 0 through 9.
range
is a series of digits, for example, 1-3.
sequence is a set of digits.
•
symbol-part = digits / symbols
where: digits are numbers 0 through 9.
symbols include such characters as
+, -,
and so on.
•
range = "[" digits "-" digits "]" group-identifier
where:
"[" digits "-" digits "]"
represents the numeric range, for example,
[1-2]
.
group-identifier
represents the group to which the number range is applied.
For example,
[1-2]
applies to group identifier
A
for in-pattern=[1-
2]ABBB.
When T-Server evaluates the rule to determine if it matches the number, it examines whether the first digit of the number, identified as group-identifier A
, is
1
or
2.
•
sequence = "[" 1*(digits [","] ) "]" group-identifier
where:
"
[" 1*(digits [","] ) "]"
represents a sequence of digits, separated by commas, and bracketed. T-Server requires that each digit set have the same number of digits. For example, in
[415,650]
the sets have three digits.
group-identifier
represents the group to which the number sequence is applied.
For example, in in-pattern=1[415,650]A*B, [415,650]
applies to group-identifier A
. When T-Server evaluates the rule to determine if it matches the number, it examines whether the three digits ( groupidentifier A
) following the 1 in the number are
415 or
650.
•
abstract-group = fixed-length-group / flexible-length-group / entity where:
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fixed-length-group
specifies a group composed of a specific number of digits and determined by how many times the group identifier is included in the in-pattern.
For example, for in-pattern=1AAABBBCCCC, there are three digits in group
A
and
B
but four in group
C.
When you create an out-pattern
, you include the group identifier only once because the in-pattern
tells T-Server how many digits belong in that group. For example, rule-04
(see
) is in-pattern=1AAABBBCCCC; out-pattern=91ABC.
flexible-length-group
specifies a group composed of 0 or more digits in the group represented by the group-identifier.
For example, in in-pattern=1[415,650]A*B, *B
represents the flexible length group containing the remaining digits in the number.
entity
represents digits defined for a specific purpose, for example, country code.
The component
abstract-group
is used only for the in-pattern
.
•
fixed-length-group = 1*group-identifier
See the earlier explanation under abstract-group.
•
flexible-length-group = "*" group-identifier
See the earlier explanation under abstract-group.
•
entity = "#" entity-identifier group-identifier
where:
“#”
indicates the start of a Country Code entity-identifier
.
entity-identifier
must be the letter
C
which represents Country Code when preceded by a pound symbol (
#
)
.
Any other letter following the # causes an error.
group-identifier
represents the Country Code group when preceded by
#C.
The entity
component is a special group that assumes some kind of predefined processing, such as the Country Code detection.
•
param-part = ";" param-name "=" param-value
where:
“;”
is a required separator element.
param-name
is the name of the parameter.
“=”
is the next required element.
param-value
represents the value for param-name.
•
param-name = "ext" / "phone-context" / "dn"
where:
“ext”
refers to extension.
“phone-context”
represents the value of the phone-context
option configured on the switch.
“dn”
represents the directory number.
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•
param-value = 1*ANYSYMBOL
where:
ANYSYMBOL
represents any number, letter, or symbol with no restrictions.
•
group-identifier = ALPHA
•
entity-identifier = ALPHA
•
digits = 1*DIGIT
•
symbols = 1*("-" / "+" / ")" / "(" / ".")
Recommendations for Rule Configuration
The configuration of rules for inbound numbers usually depends on the settings in the corresponding PBX. These settings often define the form in which the PBX notifies its client applications about the number from which an inbound call is coming.
As a general guideline, configure rules that define how to process calls from:
•
Internal numbers.
•
External numbers within the same local dialing area.
•
External numbers within the same country.
•
International numbers.
Rules for inbound numbers, typically for North American locations, might look like this:
1. Two rules to transform internal numbers (extensions): name=rule-01;in-pattern=[1-9]ABBB;out-pattern=AB name=rule-02;in-pattern=[1-9]ABBBB;out-pattern=AB
2. A rule to transform local area code numbers (in
333-1234
format in this example): name=rule-03;in-pattern=[1-9]ABBBBBB;out-pattern=+1222AB
3. A rule to transform U.S. numbers (in
+1(222)333-4444
format): name=rule-04;in-pattern=1AAAAAAAAAA;out-pattern=+1A
4. A rule to transform U.S. numbers without the
+1
prefix (in
(222)333-4444 format): name=rule-05;in-pattern=[2-9]ABBBBBBBBB;out-pattern=+1AB
5. A rule to transform U.S. numbers with an outside prefix (in
9 +1(222)333-
4444
format): name=rule-06;in-pattern=91AAAAAAAAAA;out-pattern=+1A
6. A rule to transform international numbers with an IDD (international dialing digits) prefix (in
011 +44(111)222-3333
format): name=rule-07;in-pattern=011*A;out-pattern=+A
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7. A rule to transform international numbers without an IDD prefix (in
+44(111)222-3333
format): name=rule-08;in-pattern=[2-9]A*B;out-pattern=+AB
Rule Examples
This section provides examples of six rules that are configured as options in the Genesys Configuration Database. It also provides examples of how
T-Server applies rules to various input numbers.
rule-01 rule-02 rule-03 rule-04 rule-05 rule-06
Rules
in-pattern=[1-8]ABBB;out-pattern=AB in-pattern=AAAA;out-pattern=A in-pattern=1[415,650]A*B;out-pattern=B in-pattern=1AAABBBCCCC;out-pattern=91ABC in-pattern=*A913BBBB;out-pattern=80407913B in-pattern=011#CA*B;out-pattern=9011AB
Example 1
Example 2
Example 3
Examples
Here are examples of how T-Server applies configured above rules to various input numbers.
T-Server receives input number
2326.
As a result of the rule selection process, T-Server determines that the matching rule is rule-01: name=rule-01;in-pattern=[1-8]ABBB;out-pattern=AB
The matching count for this rule is
1,
because Group
A
matches the digit
2.
As a result of the parsing process, T-Server detects two groups: Group
A = 2 and Group
B = 326.
T-Server formats the output string as
2326.
T-Server receives input number
9122.
As a result of the rule selection process, T-Server determines that the matching rule is rule-02: name=rule-02;in-pattern=AAAA;out-pattern=A
The matching count for this rule is
0;
however, the overall length of the input number matches that of the in-pattern
configuration.
As a result of the parsing process, T-Server detects one group: Group
A =
9122.
T-Server formats the output string as
9122.
T-Server receives input number
16503222332.
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Example 4
Example 5
Example 6
As a result of the rule selection process, T-Server determines that the matching rule is rule-03: name=rule-03;in-pattern=1[415,650]A*B;out-pattern=B
The matching count for this rule is
4,
because the first digit matches and all three digits in Group
A
match.
As a result of the parsing process, T-Server detects two groups: Group
A = 650 and Group
B = 3222332.
T-Server formats the output string as
3222332.
T-Server receives input number
19253227676.
As a result of the rule selection process, T-Server determines that the matching rule is rule-04: name=rule-04;in-pattern=1AAABBBCCCC;out-pattern=91ABC
The matching count for this rule is
1,
because the first digit matches.
As a result of parsing process, T-Server detects three groups: Group
A = 925,
Group
B = 322,
and Group
C = 7676.
T-Server formats the output string as
919253227676.
T-Server receives input number
4089137676.
As a result of rule selection process, T-Server determines that the matching rule is rule-05: name=rule-05;in-pattern=*A913BBBB;out-pattern=80407913B
The matching count for this rule is
3,
because three digits match.
As a result of the parsing process, T-Server detects two groups: Group
A = 408 and Group
B = 7676.
T-Server formats the output string as
804079137676.
T-Server receives input number
011441112223333.
As a result of the rule selection process, T-Server determines that the matching rule is rule-06: name=rule-06;in-pattern=011#CA*B;out-pattern=9011AB
The matching count for this rule is
3,
because three digits match.
As a result of the parsing process, T-Server detects two groups: Group
A = 44 and Group
B = 1112223333.
T-Server formats the output string as
9011441112223333.
Procedure:
Configuring Number Translation
Purpose: To configure the Number Translation feature in T-Server to provide more flexibility for handling calls distributed across multiple sites.
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Overview
•
The Number Translation feature becomes active when the ISCC/COF feature and/or the direct-ani
transaction type are used.
•
This configuration procedure must be completed within the T-Server
Application
object corresponding to your T-Server.
Start of procedure
1. Open the T-Server
Application’s Properties
dialog box.
2. Click the
Options
tab.
3. Create a new section called extrouter
or open an existing section with this name.
4. Create a new option called
This option points to another section that describes the translation rules for inbound numbers.
5. In this section, create one configuration option for each rule. Specify the rule name as the option name. The values of these options are the rules for the number translation.
For the option description and its valid values, see Chapter 12, “T-Server
Common Configuration Options,” on page 343
.
6. When you are finished, click
Apply
.
7. Click
OK
to save your changes and exit the
Properties
dialog box.
End of procedure
Network Attended Transfer/
Conference Feature
The Network Attended Transfer/Conference (NAT/C) feature is designed to enable agents working in multi-site contact centers to consult with each other before making call transfers or conferences, regardless of whether both agents work at the same or different sites. It also enables the agent who requests a consultation to maintain his or her conversation with the customer while the system is looking for an available agent and setting up the consultation call.
The NAT/C feature does not rely on the call transfer capabilities of the local switch.
There are two modes in which the network attended transfer/conference can be performed: direct and URS-controlled.
shows the sequence of steps that occur in URS-controlled mode, when Agent A, who is handling a customer call, requests a consultation with another agent, and URS (Universal Routing
Server) selects Agent B, who is working at another site. The direct mode is
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Site A
1. Network
Consult
Request
Agent A
SCP URS
5. Consultation
Request and
Confirmation
2. EventRoute-
Request
Network T-Server w/
NAT/C feature
3. Route Call to
Agent B
4. Call Data
Passing
T-Server A
Switch A
T-Server B
Switch B
Site B
6. Consultation with Agent B
Agent B
Figure 36: Steps in the NAT/C Process in URS-Controlled Mode
Step 1
Agent A makes a request for a consultation with another agent. A
TNetworkConsult
request is relayed to the Network T-Server. Depending on the parameter settings of the
TNetworkConsult
request, the NAT/C feature will operate in either direct or URS-controlled mode. For more information, see the
Voice Platform SDK 7.6 .NET (or Java) API Reference.
Step 2
(URS-controlled mode only.) The Network T-Server sends
EventRouteRequest to URS.
Step 3
(URS-controlled mode only.) URS locates an available agent at Site B and instructs the Network T-Server to route the call to Agent B. The Network
T-Server confirms the initiation of the network transfer by sending
EventNetworkCallStatus
to T-Server A, which then relays it to Agent A.
Step 4
The Network T-Server proceeds to obtain the access number from T-Server B,
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Step 5
The Network T-Server instructs the Service Control Point (SCP) to initiate a new voice path with Agent B. Once the connection is confirmed, the Network
T-Server distributes
EventNetworkCallStatus
to both T-Server A and T-Server
B, which then relay it to Agent A and Agent B respectively, to indicate that the consultation call is being established.
The Network T-Server also distributes
EventRouteUsed
to URS to confirm successful routing of the call to the selected agent.
Step 6
At this point, the customer is on hold, and Agent A is consulting with Agent B.
Agent A can do one of the following:
•
End the consultation and retrieve the original customer call
•
Alternate between Agent B and the customer
•
Set up a conference call with Agent B and the customer
•
Transfer the customer call to Agent B
Note:
All T-Servers support NAT/C requests with
AttributeHomeLocation provided that this attribute identifies a network location that is capable of processing such requests. Refer to the Network T-Server
Deployment Guides to determine whether a specific Network T-Server can process these requests.
Event Propagation Feature
The Event Propagation feature complements the ISCC and ISCC/COF features by distributing updated user data and party-related events to remote T-Servers.
This feature is used when a call is being made, transferred, or conferenced to another location, and when, as a result, one or more instances of the call reside at one location while other call instances reside at another location. In this scenario, when a client at one location makes changes to user data, updated user data is passed (propagated) to T-Servers at other locations.
The Event Propagation feature consists of User Data update propagation and
Party Events propagation.
User Data Propagation
User data propagation takes place when a client at one location makes changes to user data associated with a call that was made, transferred, conferenced, or routed to other locations. The remote clients involved with the call are notified about the changes with
EventAttachedDataChanged
.
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When T-Server receives a local update to user data (that is, when a client of this T-Server has changed the call’s user data), T-Server determines if parties at remote locations are involved with the call and, if so, sends (propagates) the updated user data to the T-Servers at remote locations.
When T-Server receives a remote update to user data (that is, when a client of a remote T-Server has changed the call’s user data and the remote T-Server has used the Event Propagation feature to send the updated user data), T-Server:
1. Updates the user data of the corresponding local call.
2. Determines if parties at other remote locations are involved with the call and, if so, propagates the updated user data to T-Servers at other remote locations.
The locations to which user data is propagated are selected based on a call distribution topology. That is, the updated user data is passed directly to the location to which a call was sent and to the location from which the call was received, excluding the location from which the update was received.
For example, consider a call made from location A to location B, and then conferenced from location B to location C. The three instances of the call reside at different locations: the first instance is at location A, the second instance is at location B, and the third instance is at location C. The Event
Propagation feature is employed in the following scenarios:
•
When T-Server at location A receives a local update to user data, it notifies
T-Server at location B (to which it sent the call) about changes to the call’s user data. Thus, T-Server at location B receives a remote update to user data and, in turn, notifies T-Server at location C (to which it sent the call) about these changes.
Although T-Server at location C receives a remote update to user data, it does not pass the notification to any other T-Servers, because it did not send the call to any other locations. As mentioned earlier, T-Servers at locations B and C update the user data of the corresponding local calls and notify their clients about the changes with
EventAttachedDataChanged
.
•
When T-Server at location B receives a local update to user data, it notifies
T-Server at location C (to which it sent the call) and T-Server at location A
(from which it received the call) about changes to the call’s user data.
Thus, T-Servers at locations C and A receive a remote update to user data.
Because T-Server at location C did not send the call to any other locations, and T-Server at location A originated the call, neither of these T-Servers passes the notification to any other T-Servers. T-Servers at locations C and
A update the user data of the corresponding local calls and notify their clients about the changes with
EventAttachedDataChanged
.
•
When T-Server at location C receives a local update to user data, it notifies
T-Server at location B (from which it received the call) about changes to the call’s user data. Thus, T-Server at location B receives a remote update to user data and, in turn, notifies T-Server at location A (from which it received the call) about these changes.
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Although T-Server at location A receives a remote update to user data, it does not pass the notification to any other T-Servers, because it originated the call. T-Servers at locations B and A update the user data of the corresponding local calls and notify their clients about the changes with
EventAttachedDataChanged
.
When a call is distributed between location A and location C using location B, and is then deleted on location B, propagation between locations A and C still occurs through the transit node at location B.
Party Events Propagation
Party events propagation takes place when a transfer or a conference is completed for a call that was made to or from one or more remote locations, or when a conference party is removed from the conference.
In these cases, the Event Propagation feature distributes party events, such as
EventPartyChanged
,
EventPartyAdded
, and
EventPartyDeleted
, to remote locations involved with the call, according to appropriate call model scenarios.
For example, consider a call made from DN 1 to DN 2 on location A. A
TInitiateConference
request is then issued for DN 2 to transfer the call to external DN 3 on location B. That transfer is made by means of ISCC routing.
When this conference is completed on location A, the Event Propagation feature sends
EventPartyChanged
to location B and distributes this event to involved client applications that are connected to location B and registered for
DN 3. After that, if a party of the conference is removed from the conference
(for example, a party on DN 2), the Event Propagation feature sends
EventPartyDeleted
to location B and distributes this event to client applications registered for DN 3.
Warnings!
• The
OtherDN
and
ThirdPartyDN
attributes might not be present in the events distributed via the Event Propagation feature.
• The Event Propagation feature will not work properly with installations that use switch partitioning.
If a call involved in the propagation has no local parties but has two or more remote parties, the party events propagation is processed in the same manner as the propagation of user data updates.
For a complete event flow in such scenarios, refer to the Genesys 7 Events and
Models Reference Manual.
Basic and Advanced Configuration
The basic Event Propagation feature configuration includes the setting of specific configuration options at the T-Server
Application
level. The advanced feature configuration allows you to customize the feature at the
Switch
level.
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When determining whether to notify other T-Servers of changes to user data, or to distribute party events, T-Server checks:
1. Call topology (what location a call came from and to what location the call was then transferred or conferenced).
2. Outbound parameters of the
Switch
this T-Server relates to (whether propagation parameters are configured for the access codes this switch uses to reach the switch at the location a call came from and the switch at the location to which the call was then transferred or conferenced).
Warning!
The direction of user-data or party-events propagation does not necessarily match the direction of call distribution. Therefore, the access code used to deliver the call can differ from the access code used for the purpose of Event Propagation.
If one of the T-Servers along the call distribution path has the Event
Propagation feature disabled, that T-Server does not distribute events to remote locations.
Procedure:
Activating Event Propagation: basic configuration
Purpose: To activate the Event Propagation feature for User Data updates and call-party–associated events (Party Events) distribution.
Start of procedure
1. Open the T-Server
Application’s Properties
dialog box.
2. Click the
Options
tab.
3. Open the extrouter
section.
option to the list
value.
This setting enables User Data propagation. If you need to enable Party
Events propagation, perform Step 5.
current
value.
This setting enables Party Events propagation.
For the option description and its valid values, see Chapter 12, “T-Server
Common Configuration Options,” on page 343
.
6. When you are finished, click
Apply
.
7. Click
OK
to save your changes and exit the
Properties
dialog box.
End of procedure
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Next Steps
•
For advanced feature configuration, do the following procedure:
Modifying Event Propagation: advanced configuration, page 305
Procedure:
Modifying Event Propagation: advanced configuration
Purpose: To modify access codes for advanced Event Propagation configuration.
Prerequisites
•
Activating Event Propagation: basic configuration, page 304
Overview
You can set Event Propagation parameters using:
•
The
Default Access Code
properties of the
Switch
that receives an ISCCrouted call (the destination switch).
•
The
Access Code
properties of the
Switch
that passes an ISCC-routed call
(the origination switch).
If you do not set up Event Propagation parameters for a given
Access Code,
T-Server uses corresponding settings configured for the
Default Access Code of the destination switch.
The procedures for modifying
Default Access Codes
and
Access Codes
are very similar to each other.
Start of procedure
1. Among configured
Switches,
select the
Switch
that the configured
T-Server relates to.
2. Open the
Switch’s Properties
dialog box and click either the
Default
Access Codes
tab or the
Access Codes
tab.
3. Select a configured
Default Access Code
or configured
Access Code
and click
Edit.
Note:
If no
Default Access Code
is configured, see
instructions. If no
Access Codes
are configured, see page 310 for
instructions.
4. In the
Switch Access Code Properties
dialog box that opens, specify a value for the
ISCC Protocol Parameters
field as follows:
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To enable distribution of both user data associated with the call and call-party–associated events
1
, type: propagate=yes which is the default value.
To enable distribution of user data associated with the call and disable distribution of call-party–associated events, type: propagate=udata
To disable distribution of user data associated with the call and enable distribution of call-party–associated events, type: propagate=party
To disable distribution of both user data associated with the call and call-party–associated events, type: propagate=no
5. Click
OK
to save configuration updates and close the
Switch Access Code
Properties
dialog box.
6. Click
Apply
and
OK
to save configuration updates and close the
Switch
Properties
dialog box.
End of procedure
ISCC Transaction Monitoring Feature
This feature allows T-Server clients to monitor ISCC transactions that occur during the call data transfer between T-Servers in a multi-site environment.
In order to be able to monitor ISCC messaging, a T-Server client must subscribe to the ISCC Transaction Monitoring. Once a subscription request is confirmed, a client will receive updates about all multi-site operations of this
T-Server.
The
TTransactionMonitoring
request is used to instruct T-Server to start, stop, or modify a client’s subscription to Transaction Monitoring feature notifications by setting the
TSubscriptionOperationType
parameter to
SubscriptionStart, SubscriptionStop,
or
SubscriptionModify respectively.
The transaction status is reported in
EventTransactionStatus
messages to the subscribed clients.
To determine whether the Transaction Monitoring feature is supported by a specific T-Server, a T-Server client may query T-Server’s capabilities. See
Genesys 7 Events and Models Reference Manual and Voice Platform SDK 7.6
.NET (or Java) API Reference for more information about support of this feature.
1. The following are call-party–associated events:
EventPartyChanged
,
EventPartyDeleted
, and
EventPartyAdded.
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Configuring Multi-Site Support
Prior to configuring T-Server to support multi-site operation, you must read the
“Licensing Requirements” on page 35 , as well as previous sections of this
chapter on multi-site deployment. In particular, Table 32 on page 283
shows
which transaction types are supported by a specific T-Server, while Table 33 on page 288
shows whether your T-Server supports the
NetworkCallID
attribute for the ISCC/COF feature. Use this information as you follow the instructions in this chapter.
Note:
Before attempting to configure a multi-site environment, Genesys recommends that you plan the changes you want to make to your existing contact centers. You should then gather the configuration information you will need (such as the names of each T-Server application, port assignments, switch names, and so on), and use
Configuration Manager to create and partially configure each T-Server
object. Review multi-site option values in the “Multi-Site Support
and determine what these values need to be, based on your network topology.
For T-Server to support multi-site operation, you must create and configure three types of objects in the Configuration Layer:
1.
Applications
2.
Switches,
including
Access Codes
3.
DNs
You must configure these objects for origination and destination locations.
Multi-site support features activate automatically at T-Server startup. See
for details.
Applications
Ensure that T-Server
Application
objects, and their corresponding
Host objects, exist and are configured for origination and destination locations.
Once you’ve done that, use Configuration Manager to add this configuration to a T-Server
Application.
Procedure:
Configuring T-Server Applications
Purpose: To configure T-Server
Application
objects for multi-site operation support.
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Start of procedure
1. Open the T-Server
Application’s Properties
dialog box.
2. Click the
Connections
tab and click
Add
to add a connection to the appropriate T-Server. The
Connection Info Properties
dialog box displays.
3. Use the
Browse
button to search for the T-Server you want to connect to, and fill in the following values:
Port ID
Connection Protocol
Local Timeout
Remote Timeout
Trace Mode
4. Click the
Options
tab. Create a new section called extrouter
or open an existing section with this name.
Note:
If you do not create the extrouter
section, T-Server works according to the default values of the corresponding configuration options.
5. Open the extrouter
section. Configure the options used for multi-site support.
Note:
For a list of options and valid values, see “Multi-Site Support
, in the “T-Server Common Configuration
Options” chapter in Part Two of this document.
6. When you are finished, click
Apply
.
7. Repeat this procedure for all T-Servers for origination and destination locations that are used for multi-site operations.
End of procedure
Next Steps
•
See
Switches and Access Codes
Ensure that
Switching Office
and
Switch
objects are configured for both origination and destination locations.
You configure
Access Codes
to a destination switch in the origination
Switch’s
Properties
dialog box. The only exception is the
Default Access Code,
which is configured at the destination
Switch’s Properties
dialog box.
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You can configure two types of switch
Access Codes
in the
Switch’s
Properties
dialog box:
•
A
Default Access Code
(for inbound calls)—Specifies the access code that other switches can use to access this switch when they originate a multisite transaction.
•
An
Access Code
(for outbound calls)—Specifies the access code that this switch can use when it originates a multi-site transaction to access another switch.
When the origination T-Server processes a multi-site transaction, it looks for an access code to the destination switch. First, T-Server checks the
Access
Code
of the origination
Switch
:
•
If an access code to the destination switch is configured with the target type
Target ISCC
and with any transaction type except
Forbidden
, T-Server uses this access code to dial the destination switch.
•
If the access code to the destination switch is not configured on the
Access
Code
tab of the origination switch, the origination T-Server checks the
Default Access Code
tab of the destination switch. If an access code is configured there with the target type
Target ISCC
and with any transaction type except
Forbidden
, T-Server uses this access code to dial the destination switch.
•
If no access code with the required properties is found, T-Server rejects the transaction.
Note:
When migrating from previous releases of T-Servers to 7.6, or when using T-Servers of different releases (including 7.6) in the same
environment, see “Compatibility Notes” on page 313.
Procedure:
Configuring Default Access Codes
Purpose: To configure the
Default Access Codes
(one per
Switch
object) to be used by other switches to access this switch when they originate a multi-site transaction.
Prerequisites
•
Ensure that
Switching Office
and
Switch
objects are configured for both origination and destination locations.
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Start of procedure
1. Among configured
Switches
, select the
Switch
that the configured T-Server relates to.
2. Open the
Switch Properties
dialog box and click the
Default Access Codes tab.
3. Click
Add
to open the
Access Code Properties
dialog box.
4. In the
Code
field, specify the access code used by remote switches to reach a DN at this switch. An access code is used as a prefix to the remote switch numbers.
Note:
If no prefix is needed to dial to the configured switch, you can leave the
Code
field blank.
5. In the
Target Type
field, select
Target ISCC
.
6. In the
Route Type
field, select a value corresponding to the transaction type you want to use (given that it is supported for your switch type).
7. When you are finished, click
Apply
.
End of procedure
Next Steps
•
See
Procedure:
Configuring Access Codes
Purpose: To configure the
Access Codes
(one or more per
Switch
object) that this switch can use when it originates a multi-site transaction to access another switch.
Prerequisites
•
Ensure that
Switching Office
and
Switch
objects are configured for both origination and destination locations.
Start of procedure
1. Among configured
Switches
, select the
Switch
that the configured T-Server relates to.
2. Open the
Switch Properties
dialog box and click the
Access Codes
tab.
3. Click
Add
to open the
Access Code Properties
dialog box.
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4. In the
Switch
field, specify the switch that this switch can reach using this access code. Use the
Browse
button to locate the remote switch.
5. In the
Code
field, specify the access code used to reach a DN at the remote switch from this switch. An access code is used as a prefix to the remote switch numbers.
Note:
If no prefix is needed to dial from one switch to another, you can leave the
Code
field blank.
6. In the
Target Type
field, select
Target ISCC
.
When you select
Target ISCC as your target type, the
Properties
dialog box changes its lower pane to the
Sources
pane. It is here that you enter the extended parameters for your access codes, by specifying the
ISCC
Protocol
and
ISCC Call Overflow Parameters.
To set these parameters, locate the two drop-down boxes that appear below the
Target Type field in the
Sources
pane of that
Properties
dialog box.
a. In the
ISCC Protocol Parameters
drop-down box, enter the appropriate
ISCC Protocol parameter, as a comma-separated list of one or more of the following items shown in
Table 34: Target Type: ISCC Protocol Parameters
ISCC Protocol Parameters Description
dnis-tail=<number-of-digits
> Where
<number-of-digits
> is the number of significant DNIS digits (last digits) used for call matching
0
(zero) matches all digits.
propagate=<yes
, udata
, party
, no>
Default is
yes
. For more information,
see “Modifying Event Propagation: advanced configuration” on page 305 .
direct-network-callid=<>
For configuration information, see Part
Two of this document. (Use Table 32 on page 283 to determine if your T-Server
supports the direct-network-callid transaction type.)
b. In the
ISCC Call Overflow Parameters
drop-down box, enter call overflow parameters, as a comma-separated list of one or more of the following items shown in
:
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Table 35: Target Type: ISCC Call Overflow Parameters
ISCC Call Overflow Parameters Description
match-callid
Matches calls using network
CallID
.
match-ani inbound-only=<boolean>
Matches calls using ANI.
Default is
true.
Setting
inbound-only to true disables COF on consultation and outbound calls.
7. In the
Route Type
field, select a value corresponding to the transaction type you want to use (given that it is supported for your switch type).
contains cross-reference information on transaction types that the
Configuration Layer and T-Server use.
Table 36: Route Type and ISCC Transaction Type Cross-Reference
Route Type Field Value
Default
Direct
Direct ANI
Direct Digits
Direct DNIS and ANI
Direct Network Call ID
Direct No Token
Direct UUI
DNIS Pooling
Forbidden
ISCC defined protocol
PullBack
ISCC Transaction Type
The first value from the list of values specified in the cast-type
option for the
T-Server at the destination site direct-callid direct-ani direct-digits
Reserved direct-network-callid direct-notoken direct-uui dnis-pooling
External routing to this destination is not allowed
Reserved pullback
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Table 36: Route Type and ISCC Transaction Type Cross-Reference
(Continued)
Route Type Field Value
Re-Route
Route
8. When you are finished, click
Apply
.
ISCC Transaction Type
reroute route
End of procedure
Next Steps
•
After configuring a switch for multi-site support, proceed with the configuration of DNs assigned to this switch.
Compatibility Notes
When migrating from previous releases of T-Servers to 7.6, or when using
T-Servers of different releases (including 7.6) in the same environment, keep in mind the following compatibility issues:
•
The
Target External Routing Point
value of the
Target Type
field is obsolete and provided only for backward compatibility with T-Servers of releases 5.1 and 6.0. When two access codes for the same switch are configured, one with the
Target ISCC
target type and the other with the
Target External Routing Point
target type, T-Servers of releases 7.x, 6.5, and 6.1:
Use the
Target ISCC
access code for transactions with T-Servers of releases 7.x, 6.5, and 6.1.
Use the
Target External Routing Point
access code for transactions with T-Servers of releases 5.1 and 6.0.
When the only access code configured for a switch has the
Target External
Routing Point
target type, T-Server uses this access code for all transactions.
•
When the
Target External Routing Point
value of the
Target Type
field is configured, you must set the
Route Type
field to one of the following:
Default
to enable the route
transaction type
Label
to enable the direct-ani
transaction type
Direct
to enable the direct
transaction type
Note:
The direct
transaction type in releases 5.1 and 6.0 corresponds to the direct-callid
transaction type in releases 6.1, 6.5, and 7.x.
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UseExtProtocol
to enable the direct-uui
transaction type
PostFeature
to enable the reroute
transaction type
These values are fully compatible with the transaction types supported in
T-Server release 5.1.
•
For successful multi-site operations between any two locations served by release 5.1 T-Servers, identical
Route Type
values must be set in the
Switch’s Access Code Properties
dialog boxes for both the origination and destination switches.
DNs
Use the procedures from this section to configure access resources for various transaction types.
Procedure:
Configuring access resources for the route transaction type
Purpose: To configure dedicated DNs required for the route transaction type.
Prerequisites
•
Ensure that
Switching Office
and
Switch
objects are configured for both origination and destination locations.
Start of procedure
1. Under a configured
Switch
, select the
DNs
folder. From the main menu, select
File > New > DN
to create a new
DN
object.
2. On the
General
tab of the DN’s
Properties
dialog box, specify the number of the configured DN as the value of the
Number
field. This value must correspond to the Routing Point number on the switch.
3. Select
External Routing Point
as the value of the
Type
field.
4. If a dialable number for that Routing Point is different from its DN name, specify the number in the
Association
field.
5. Click the
Access Numbers
tab. Click
Add
and specify these access number parameters:
Origination switch.
Access number that must be dialed to reach this DN from the origination switch.
In determining an access number for the Routing Point, T-Server composes it of the values of the following properties (in the order listed):
a. Access number (if specified).
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b. Switch access code from the switch of the origination party to the switch to which the Routing Point belongs, concatenated with its
Association
(if the
Association
value is specified).
c. Switch access code from the switch of the origination party to the switch to which the Routing Point belongs, concatenated with the number for the DN.
d. Default access code of the switch to which the Routing Point belongs, concatenated with its
Association
(if the
Association
value is specified).
e. Default access code of the switch to which the Routing Point belongs, concatenated with the number for the DN.
Note:
If option
is set to true,
the access number composed of switch access code and DN can be used for external transfers of calls originating at switches for which an access number is not specified.
6. When you are finished, click
Apply
.
End of procedure
Procedure:
Configuring access resources for the dnis-pool transaction type
Purpose: To configure dedicated DNs required for the dnis-pool transaction type.
Start of procedure
1. Under a configured
Switch
, select the
DNs
folder. From the main menu, select
File > New > DN
to create a new
DN
object.
2. On the
General
tab of the DN’s
Properties
dialog box, specify the number of the configured DN as the value of the
Number
field. This value must be a dialable number on the switch.
3. Select
Access Resource
as the
Type
field and type dnis
as the value of the
Resource Type
field on the
Advanced
tab.
4. Click the
Access Numbers
tab. Click
Add
and specify these
Access Number parameters:
Origination switch.
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Access number that must be dialed to reach this DN from the origination switch.
An access number for the access resource is determined in the same manner as for the route
access resource.
5. When you are finished, click
Apply
.
End of procedure
Procedure:
Configuring access resources for direct-* transaction types
Overview
You can use any configured DN as an access resource for the direct-* transaction types. (The
*
symbol stands for any of the following: callid
, uui
, notoken
, ani
, or digits.
)
You can select the
Use Override
check box on the
Advanced
tab to indicate whether the override value should be used instead of the number value to dial to the DN. You must specify this value if the DN has a different DN name and dialable number. In fact, this value is required for T-Servers for some switch types—for example, Aspect ACD, Nortel Communication Server 2000/2100, and Spectrum.
Procedure:
Configuring access resources for ISCC/COF
Purpose: To configure dedicated DNs required for the ISCC/COF feature.
Start of procedure
Note:
Use Table 33 on page 288 to determine if your T-Server supports the
ISCC/COF feature.
1. Under a configured
Switch
, select the
DNs
folder. From the main menu, select
File > New > DN
to create a new
DN
object.
Note:
The number of the access resource must match the name of a DN configured on the switch (usually, an ACD Queue) so that
T-Server can determine if the calls arriving to this DN are overflowed calls.
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2. On the
General
tab of the DN
Properties
dialog box, specify the number of the configured DN as the value for the
Number
field.
3. Select
Access Resource
as the value for the
Type
field.
4. On the
Advanced
tab, type cof-in or cof-not-in
as the value for the
Resource Type
field.
Note:
Calls coming to DNs with the cof-not-in
value for the
Resource
Type
are never considered to be overflowed.
5. When you are finished, click
Apply
.
End of procedure
Procedure:
Configuring access resources for non-unique ANI
Purpose: To configure dedicated DNs required for the non-unique-ani resource type.
The non-unique-ani
resource type is used to block direct-ani
and
COF/ani from relaying on ANI when it matches configured/enabled resource digits.
Using non-unique-ani,
T-Server checks every ANI against a list of non-unique-ani
resources.
Start of procedure
1. Under a configured
Switch
, select the
DNs
folder. From the main menu, select
File > New > DN
to create a new
DN
object.
2. On the
General
tab of the DN
Properties
dialog box, specify the ANI digits that need to be excluded from normal processing.
3. Select
Access Resource
as the value for the
Type
field.
4. On the
Advanced
tab, specify the
Resource Type field as non-unique-ani
.
5. When you are finished, click
Apply
.
End of procedure
Procedure:
Modifying DNs for isolated switch partitioning
Purpose: To modify DNs that belong to a particular partition where switch partitioning is used.
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This configuration instructs T-Server to select an External Routing Point that has the same partition as the requested destination DN.
Note:
When a target DN is not configured or has no configured partition name, T-Server allocates a DN of the
External Routing Point
type that belongs to any partition.
Start of procedure
1. Under a
Switch
object, select the
DNs
folder.
2. Open the
Properties
dialog box of a particular
DN
.
3. Click the
Annex
tab.
4. Create a new section named
TServer
.
5. Within that section, create a new option named epn.
Set the option value to the partition name to which the DN belongs.
6. Repeat Steps 1–5 for all DNs, including DNs of the
External Routing
Point
type, that belong to the same switch partition.
7. When you are finished, click
Apply
.
End of procedure
Configuration Examples
This section provides two configuration examples and describes how the configuration settings affect T-Server’s behavior.
Multiple Transaction Types
This example demonstrates the difference in how ISCC directs a call when you specify two different transaction types ( route
and direct-ani
).
In this example, you configure an origination and a destination switch for as
described in “Switches and Access Codes” on page 308
.
1. Among configured
Switches
, select the origination
Switch
.
2. Open the
Switch Properties
dialog box and click the
Default Access Codes tab.
3. Click
Add
to open the
Access Code Properties
dialog box.
4. Set the
Access Code
field to
9.
5. When you are finished, click
Apply
.
6. Among configured
Switches
, select the destination
Switch
.
7. Under the destination
Switch, configure a DN as described in
“Configuring access resources for the route transaction type” on page 314 .
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8. Set the DN
Number
field to
5001234567.
9. Click the
Advanced
tab of this DN’s
Properties
dialog box.
10. Select the
Use Override
check box and enter
1234567
in the
Use Override field.
11. When you are finished, click
Apply
or
Save
.
12. Use a T-Server client application to register for this new DN with the destination T-Server and, therefore, with the switch.
13. Request to route a call from any DN at the origination switch to the destination DN you have just configured:
If you are using the route
ISCC transaction type, the client requests that T-Server deliver a call to a destination location using the DN number
5001234567.
ISCC requests that the switch dial one of the external routing points at the destination location, using the value either of the
Access Number
field or of the
Access Code
field, which is
9, concatenated with the external routing point at the destination location.
The call is routed to the DN number
5001234567
.
If you are using the direct-ani
ISCC transaction type, the client requests that T-Server deliver a call to a destination location using the
DN number
1234567
, which is the
Use Override
value. ISCC requests that the switch dial
91234567,
which is a combination of the
Switch
Access Code
value and the
Use Override
value. The destination T-Server is waiting for the call to directly arrive at DN number
5001234567.
Call Overflow Methods
This section demonstrates how to indicate which overflow methods a switch supports.
In this example, for T-Server to use ANI/OtherDN matching in call overflow and manual transfer scenarios, set the
ISCC Call Overflow Parameters
to: match-ani
, inbound-only=true when configuring
Switch Access Codes
as described on
.
With this setting, the switch’s location is queried for call data each time the destination T-Server receives an inbound call with the
ANI
or
OtherDN
attribute.
For T-Server to use
NetworkCallID
matching in call overflow and manual transfer scenarios, set the
ISCC Call Overflow Parameters
to (for example): match-callid
, inbound-only=false when configuring
Switch Access Codes
With this setting, the switch’s location is queried for call data each time the destination T-Server receives a call of any type (including inbound) with the
NetworkCallID
attribute.
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Next Steps
Continue with Chapter 4, “Starting and Stopping SIP Server,” on page 69
to test your configuration and installation.
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Chapter
11
Common Configuration
Options
z z z z z z z z
Unless otherwise noted, the common configuration options that this chapter describes are common to all Genesys server applications and applicable to any
Framework server component. This chapter includes the following sections:
Setting Configuration Options, page 321
Log-Extended Section, page 336
Log-Filter-Data Section, page 339
Changes from 7.5 to 7.6, page 340
Note:
Some server applications also support log options that are unique to them. For descriptions of a particular application’s unique log options, refer to the chapter/document about that application.
Setting Configuration Options
Unless it is otherwise specified in this document or in the documentation for your application, you set common configuration options in Configuration
Manager in the corresponding sections on the
Options tab of the
Application object.
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Warning!
Configuration section names, configuration option names, and predefined option values are case-sensitive. Type them in the
Configuration Manager interface exactly as they are documented in this chapter.
Mandatory Options
You do not have to configure any common options to start Server applications.
Log Section
This section must be called log.
verbose
Default Value: all
Valid Values: all debug trace interaction standard none
All log events (that is, log events of the Standard, Trace,
Interaction, and Debug levels) are generated.
The same as all.
Log events of the Trace level and higher (that is, log events of the Standard, Interaction, and Trace levels) are generated, but log events of the Debug level are not generated.
Log events of the Interaction level and higher (that is, log events of the Standard and Interaction levels) are generated, but log events of the Trace and Debug levels are not generated.
Log events of the Standard level are generated, but log events of the Interaction, Trace, and Debug levels are not generated.
No output is produced.
Changes Take Effect: Immediately
Determines whether a log output is created. If it is, specifies the minimum level of log events generated. The log events levels, starting with the highest
priority level, are Standard, Interaction, Trace, and Debug. See also “Log
Note:
For definitions of the Standard, Interaction, Trace, and Debug log levels, refer to the Framework 7.6 Deployment Guide or to Framework
7.6 Solution Control Interface Help.
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buffering
Default Value: true
Valid Values: true false
Enables buffering.
Disables buffering.
Changes Take Effect: Immediately
Turns on/off operating system file buffering. The option is applicable only to the stderr
and stdout
output (see
page 328 ). Setting this option to
true increases the output performance.
Note:
When buffering is enabled, there might be a delay before log messages appear at the console.
segment
Default Value: false
Valid Values: false
<number> KB or
<number>
<number> MB
<number> hr
No segmentation is allowed.
Sets the maximum segment size, in kilobytes. The minimum segment size is
100 KB
.
Sets the maximum segment size, in megabytes.
Sets the number of hours for the segment to stay open. The minimum number is
1
hour.
Changes Take Effect: Immediately
Specifies whether there is a segmentation limit for a log file. If there is, sets the 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. This option is ignored if log output is not configured to be sent to a log file.
expire
Default Value: false
Valid Values: false
<number> file or
<number>
<number> day
No expiration; all generated segments are stored.
Sets the maximum number of log files to store. Specify a number from
1
–
100.
Sets the maximum number of days before log files are deleted. Specify a number from
1
–
100.
Changes Take Effect: Immediately
Determines whether log files expire. If they do, sets the measurement for determining when they expire, along with the maximum number of files
(segments) or days before the files are removed. This option is ignored if log output is not configured to be sent to a log file.
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Note:
If an option’s value is set incorrectly—out of the range of valid values— it will be automatically reset to
10.
keep-startup-file
Default Value: false
Valid Values: false true
<number> KB
<number> MB
No startup segment of the log is kept.
A startup segment of the log is kept. The size of the segment equals the value of the segment
option.
Sets the maximum size, in kilobytes, for a startup segment of the log.
Sets the maximum size, in megabytes, for a startup segment of the log.
Changes Take Effect: After restart
Specifies whether a startup segment of the log, containing the initial T-Server configuration, is to be kept. If it is, this option can be set to true
or to a specific size. If set to true
, the size of the initial segment will be equal to the size of the regular log segment defined by the segment
option. The value of this option will be ignored if segmentation is turned off (that is, if the segment
option set to false
).
Note:
This option applies only to T-Servers.
messagefile
Default Value: As specified by a particular application
Valid Values:
<string>.lms
(message file name)
Changes Take Effect: Immediately, if an application cannot find its
*.lms
file at startup
Specifies the file name for application-specific log events. The name must be valid for the operating 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.
Warning!
An application that does not find its
*.lms
file at startup cannot generate application-specific log events and send them to Message
Server.
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message_format
Default Value: short
Valid Values: short full
An application uses compressed headers when writing log records in its log file.
An application uses complete headers when writing log records in its log file.
Changes Take Effect: Immediately
Specifies the format of log record headers that an application uses when writing logs in the log file. Using compressed log record headers improves application performance and reduces the log file’s size.
With the value set to short
:
•
A header of the log file or the log file segment contains information about the application (such as the application name, application type, host type, and time zone), whereas single log records within the file or segment omit this information.
•
A log message priority is abbreviated to
Std
,
Int
,
Trc
, or
Dbg
, for Standard,
Interaction, Trace, or Debug messages, respectively.
•
The message ID does not contain the prefix
GCTI
or the application type
ID.
A log record in the full format looks like this:
2002-05-07T18:11:38.196 Standard localhost cfg_dbserver GCTI-00-05060
Application started
A log record in the short format looks like this:
2002-05-07T18:15:33.952 Std 05060 Application started
Note:
Whether the full or short format is used, time is printed in the format
option.
time_convert
Default Value:
Local
Valid Values: local utc
The time of log record generation is expressed as a local time, based on the time zone and any seasonal adjustments. Time zone information of the application’s host computer is used.
The time of log record generation is expressed as Coordinated
Universal Time (UTC).
Changes Take Effect: Immediately
Specifies the system in which an application calculates the log record time when generating a log file. The time is converted from the time in seconds since the Epoch (00:00:00 UTC, January 1, 1970).
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time_format
Default Value: time
Valid Values: time locale
ISO8601
The time string is formatted according to the
HH:MM:SS.sss
(hours, minutes, seconds, and milliseconds) format.
The time string is formatted according to the system’s locale.
The date in the time string is formatted according to the ISO 8601 format. Fractional seconds are given in milliseconds.
Changes Take Effect: Immediately
Specifies how to represent, in a log file, the time when an application generates log records.
A log record’s time field in the ISO 8601 format looks like this:
2001-07-24T04:58:10.123
print-attributes
Default Value: false
Valid Values: true false
Attaches extended attributes, if any exist, to a log event sent to log output.
Does not attach extended attributes to a log event sent to log output.
Changes Take Effect: Immediately
Specifies whether the application attaches extended attributes, if any exist, to a log event that it sends to log output. Typically, log events of the Interaction log level and Audit-related log events contain extended attributes. Setting this option to true
enables audit capabilities, but negatively affects performance.
Genesys recommends enabling this option for Solution Control Server and
Configuration Server when using audit tracking. For other applications, refer to Genesys 7.6 Combined Log Events Help to find out whether an application generates Interaction-level and Audit-related log events; if it does, enable the option only when testing new interaction scenarios.
check-point
Default Value:
1
Valid Values:
0
–
24
Changes Take Effect: Immediately
Specifies, in hours, how often the application generates a check point
log event, to divide the log into 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.
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memory
Default Value: No default value
Valid Values:
<string>
(memory file name)
Changes Take Effect: Immediately
Specifies the name of the file to which the application regularly prints a
snapshot of the memory output, if it is configured to do this (see “Log Output
). The new snapshot overwrites the previously written data. If the application terminates abnormally, this file will contain the latest log messages. Memory output is not recommended for processors with a CPU frequency lower than 600 MHz.
Note:
If the file specified as the memory
file is located on a network drive, an application does not create a snapshot file (with the extension
*.memory.log
).
memory-storage-size
Default Value:
2 MB
Valid Values:
<number> KB or
<number>
The size of the memory output, in kilobytes.
The minimum value is
128 KB.
<number> MB
The size of the memory output, in megabytes.
The maximum value is
64 MB.
Changes Take Effect: When memory output is created
Specifies the buffer size for log output to the memory, if configured. See also
“Log Output Options” on page 328
.
spool
Default Value: The application’s working directory
Valid Values:
<path>
(the folder, with the full path to it)
Changes Take Effect: Immediately
Specifies the folder, including full path to it, in which an application creates temporary files related to network log output. If you change the option value while the application is running, the change does not affect the currently open network output.
compatible-output-priority
Default Value: false
Valid Values: true false
The log of the level specified by
“Log Output Options” is sent to the
specified output.
The log of the level specified by
and higher levels is sent to the specified output.
Changes Take Effect: Immediately
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Chapter 11: Common Configuration Options Log Section
Specifies whether the application uses 6.x output logic. For example, you configure the following options in the log
section for a 6.x application and for a 7.x application:
[log] verbose = all debug = file1 standard = file2
The log file content of a 6.x application is as follows:
•
file1
contains Debug messages only.
•
file2
contains Standard messages only.
The log file content of a 7.x application is as follows:
•
file1
contains Debug, Trace, Interaction, and Standard messages.
•
file2
contains Standard messages only.
If you set compatible-output-priority
to true
in the 7.x application, its log file content will be the same as for the 6.x application.
Warning!
Genesys does not recommend changing the default value of the
option unless you have specific reasons to use the 6.x log output logic—that is, to mimic the output priority as implemented in releases 6.x. Setting this option to true affects log consistency.
Log Output Options
To configure log outputs, set log level options ( all
,
,
) to the desired types of log output ( stdout
, stderr
, network
, memory
, and/or
[filename]
, for log file output).
You can use:
•
One log level option to specify different log outputs.
•
One log output type for different log levels.
•
Several log output types simultaneously, to log events of the same or different log levels.
You must separate the log output types by a comma when you are configuring
more than one output for the same log level. See “Examples” on page 332 .
Note:
The log output options are activated according to the setting of the
configuration option.
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Warnings!
• If you direct log output to a file on the network drive, an application does not create a snapshot log file (with the extension
*.snapshot.log
) in case it terminates abnormally.
• Directing log output to the console (by using the stdout
or stderr settings) can affect application performance. Avoid using these log output settings in a production environment.
all
Default Value: No default value
Valid Values (log output types): stdout stderr network
Log events are sent to the Standard output ( stdout
).
Log events are sent to the Standard error output ( stderr
).
Log events are sent to Message Server, which can reside anywhere on the network. Message Server stores the log events in the Log Database.
memory
[filename]
Setting the all
log level option to the network
output enables an application to send log events of the
Standard, Interaction, and
Trace
levels to Message Server.
Debug
-level log events are neither sent to Message Server nor stored in the Log Database.
Log events are sent to the memory output on the local disk. This is the safest output in terms of the application performance.
Log events are stored in a file with the specified name. If a path is not specified, the file is created in the application’s working directory.
Changes Take Effect: Immediately
Specifies the outputs to which an application sends all log events. The log output types must be separated by a comma when more than one output is configured. For example: all = stdout, logfile
Note:
To ease the troubleshooting process, consider using unique names for log files that different applications generate.
standard
Default Value: No default value
Valid Values (log output types): stdout stderr network
Log events are sent to the Standard output ( stdout
).
Log events are sent to the Standard error output ( stderr
).
Log events are sent to Message Server, which can reside anywhere on the network. Message Server stores the log events in the Log Database.
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Log events are sent to the memory output on the local disk. This is the safest output in terms of the application performance.
[filename]
Log events are stored in a file with the specified name. If a path is not specified, the file is created in the application’s working directory.
Changes Take Effect: Immediately
Specifies the outputs to which an application sends the log events of the
Standard
level. The log output types must be separated by a comma when more than one output is configured. For example: standard = stderr, network
interaction
Default Value: No default value
Valid Values (log output types): stdout stderr network memory
[filename]
Log events are sent to the Standard output ( stdout
).
Log events are sent to the Standard error output ( stderr
).
Log events are sent to Message Server, which can reside anywhere on the network. Message Server stores the log events in the Log Database.
Log events are sent to the memory output on the local disk. This is the safest output in terms of the application performance.
Log events are stored in a file with the specified name. If a path is not specified, the file is created in the application’s working directory.
Changes Take Effect: Immediately
Specifies the outputs to which an application sends the log events of the
Interaction
level and higher (that is, log events of the
Standard
and
Interaction
levels). The log outputs must be separated by a comma when more than one output is configured. For example: interaction = stderr, network
trace
Default Value: No default value
Valid Values (log output types): stdout stderr network memory
[filename]
Log events are sent to the Standard output ( stdout
).
Log events are sent to the Standard error output ( stderr
).
Log events are sent to Message Server, which can reside anywhere on the network. Message Server stores the log events in the Log Database.
Log events are sent to the memory output on the local disk. This is the safest output in terms of the application performance.
Log events are stored in a file with the specified name. If a path is not specified, the file is created in the application’s working directory.
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Changes Take Effect: Immediately
Specifies the outputs to which an application sends the log events of the
Trace level and higher (that is, log events of the
Standard
,
Interaction
, and
Trace levels). The log outputs must be separated by a comma when more than one output is configured. For example: trace = stderr, network
debug
Default Value: No default value
Valid Values (log output types): stdout stderr memory
[filename]
Log events are sent to the Standard output ( stdout
).
Log events are sent to the Standard error output ( stderr
).
Log events are sent to the memory output on the local disk. This is the safest output in terms of the application performance.
Log events are stored in a file with the specified name. If a path is not specified, the file is created in the application’s working directory.
Changes Take Effect: Immediately
Specifies the outputs to which an application sends the log events of the
Debug level and higher (that is, log events of the
Standard
,
Interaction
,
Trace
, and
Debug
levels). The log output types must be separated by a comma when more than one output is configured—for example: debug = stderr, /usr/local/genesys/logfile
Note:
Debug
-level log events are never sent to Message Server or stored in the
Log Database.
Log File Extensions
You can use the following file extensions to identify log files that an application creates for various types of output:
•
*.log
—Assigned to log files when you configure output to a log file. For example, if you set standard = confservlog
for Configuration Server, it prints log messages into a text file called confservlog.<time_stamp>.log.
•
*.qsp
—Assigned to temporary (spool) files when you configure output to the network but the network is temporarily unavailable. For example, if you set standard = network
for Configuration Server, it prints log messages into a file called confserv.<time_stamp>.qsp
during the time the network is not available.
•
*.snapshot.log
—Assigned to files that contain the output snapshot when you configure output to a log file. The file contains the last log messages that an application generates before it terminates abnormally. For example,
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Chapter 11: Common Configuration Options Log Section if you set standard = confservlog
for Configuration Server, it prints the last log message into a file called confserv.<time_stamp>.snapshot.log
in case of failure.
Note:
Provide
*.snapshot.log
files to Genesys Technical Support when reporting a problem.
•
*.memory.log
—Assigned to log files that contain the memory output snapshot when you configure output to memory and redirect the most recent memory output to a file. For example, if you set standard = memory and memory = confserv
for Configuration Server, it prints the latest memory output to a file called confserv.<time_stamp>.memory.log
.
Examples
This section presents examples of a log
section that you might configure for an application when that application is operating in production mode and in two lab modes, debugging and troubleshooting.
Production Mode Log Section
[log] verbose = standard standard = network, logfile
With this configuration, an application only generates the log events of the
Standard
level and sends them to Message Server, and to a file named logfile
, which the application creates in its working directory. Genesys recommends that you use this or a similar configuration in a production environment.
Warning!
Directing log output to the console (by using the stdout
or stderr settings) can affect application performance. Avoid using these log output settings in a production environment.
Lab Mode Log Section
[log] verbose = all all = stdout, /usr/local/genesys/logfile trace = network
With this configuration, an application generates log events of the
Standard
,
Interaction
,
Trace
, and
Debug
levels, and sends them to the standard output and to a file named logfile
, which the application creates in the
/usr/local/ genesys/
directory. In addition, the application sends log events of the
Standard
,
Interaction
, and
Trace
levels to Message Server. Use this configuration to test new interaction scenarios in a lab environment.
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Chapter 11: Common Configuration Options Log Section
Failure-Troubleshooting Log Section
[log] verbose = all standard = network all = memory memory = logfile memory-storage-size = 32 MB
With this configuration, an application generates log events of the
Standard level and sends them to Message Server. It also generates log events of the
Standard
,
Interaction
,
Trace
, and
Debug
levels, and sends them to the memory output. The most current log is stored to a file named logfile
, which the application creates in its working directory. Increased memory storage allows an application to save more of the log information generated before a failure.
Use this configuration when trying to reproduce an application’s failure. The memory log file will contain a snapshot of the application’s log at the moment of failure; this should help you and Genesys Technical Support identify the reason for the failure.
Note:
If you are running an application on UNIX, and you do not specify any files in which to store the memory output snapshot, a core file that the application produces before terminating contains the most current application log. Provide the application’s core file to Genesys
Technical Support when reporting a problem.
Debug Log Options
The following options enable you to generate
Debug logs containing information about specific operations of an application.
x-conn-debug-open
Default Value:
0
Valid Values:
0
1
Log records are not generated.
Log records are generated.
Changes Take Effect: After restart
Generates
Debug log records about “open connection” operations of the application.
Warning!
Use this option only when requested by Genesys Technical
Support.
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Chapter 11: Common Configuration Options Log Section
x-conn-debug-select
Default Value:
0
Valid Values:
0
1
Log records are not generated.
Log records are generated.
Changes Take Effect: After restart
Generates
Debug log records about “socket select” operations of the application.
Warning!
Use this option only when requested by Genesys Technical
Support.
x-conn-debug-timers
Default Value:
0
Valid Values:
0
1
Log records are not generated.
Log records are generated.
Changes Take Effect: After restart
Generates
Debug log records about the timer creation and deletion operations of the application.
Warning!
Use this option only when requested by Genesys Technical
Support.
x-conn-debug-write
Default Value:
0
Valid Values:
0
1
Log records are not generated.
Log records are generated.
Changes Take Effect: After restart
Generates
Debug log records about “write” operations of the application.
Warning!
Use this option only when requested by Genesys Technical
Support.
x-conn-debug-security
Default Value:
0
Valid Values:
0
1
Log records are not generated.
Log records are generated.
Changes Take Effect: After restart
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Chapter 11: Common Configuration Options Log Section
Generates
Debug log records about security-related operations, such as
Transport Layer Security and security certificates.
Warning!
Use this option only when requested by Genesys Technical
Support.
x-conn-debug-api
Default Value:
0
Valid Values:
0
1
Log records are not generated.
Log records are generated.
Changes Take Effect: After restart
Generates
Debug log records about connection library function calls.
Warning!
Use this option only when requested by Genesys Technical
Support.
x-conn-debug-dns
Default Value:
0
Valid Values:
0
1
Log records are not generated.
Log records are generated.
Changes Take Effect: After restart
Generates
Debug log records about DNS operations.
Warning!
Use this option only when requested by Genesys Technical
Support.
x-conn-debug-all
Default Value:
0
Valid Values:
0
1
Log records are not generated.
Log records are generated.
Changes Take Effect: After restart
Generates
Debug log records about open connection, socket select, timer creation and deletion, write, security-related, and DNS operations, and connection library function calls. This option is the same as enabling or disabling all of the previous x-conn-debug-<op type>
options.
Warning!
Use this option only when requested by Genesys Technical
Support.
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Log-Extended Section
This section must be called log-extended.
level-reassign-<eventID>
Default Value: Default value of log event
<eventID>
Valid Values: alarm standard interaction trace debug none
The log level of log event
<eventID>
is set to
Alarm.
The log level of log event
<eventID>
is set to
Standard.
The log level of log event
<eventID>
is set to
Interaction.
The log level of log event
<eventID>
is set to
Trace.
The log level of log event
<eventID>
is set to
Debug.
Log event
<eventID>
is not recorded in a log.
Changes Take Effect: Immediately
Specifies a log level for log event
<eventID>
that is different than its default level, or disables log event
<eventID>
completely. If no value is specified, the log event retains its default level. This option is useful when you want to customize the log level for selected log events.
These options can be deactivated with the option level-reassign-disable
(see
).
Warning!
Use caution when making these changes in a production environment.
Depending on the log configuration, changing the log level to a higher priority may cause the log event to be logged more often or to a greater number of outputs. This could affect system performance.
Likewise, changing the log level to a lower priority may cause the log event to be not logged at all, or to be not logged to specific outputs, thereby losing important information. The same applies to any alarms associated with that log event.
In addition to the preceding warning, take note of the following:
•
Logs can be customized only by release 7.6 or later applications.
•
When the log level of a log event is changed to any level except none
, it is subject to the other settings in the
[log]
section at its new level. If set to none
, it is not logged and is therefore not subject to any log configuration.
•
Using this feature to change the log level of a log changes only its priority; it does not change how that log is treated by the system. For example, increasing the priority of a log to
Alarm
level does not mean that an alarm will be associated with it.
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Chapter 11: Common Configuration Options Log-Extended Section
•
Each application in a High Availability (HA) pair can define its own unique set of log customizations, but the two sets are not synchronized with each other. This can result in different log behavior depending on which application is currently in primary mode.
•
This feature is not the same as a similar feature in Universal Routing
Server (URS) release 7.2 or later. In this Framework feature, the priority of log events are customized. In the URS feature, the priority of debug messages only are customized. Refer to the URS 7.6 Reference Manual for more information about the URS feature.
•
You cannot customize any log event that is not in the unified log record format. Log events of the
Alarm, Standard, Interaction,
and
Trace
levels feature the same unified log record format.
Example
This is an example of using customized log level settings, subject to the following log configuration:
[log] verbose=interaction all=stderr interaction=log_file standard=network
Before the log levels of the log are changed:
•
Log event 1020, with default level standard, is output to stderr
and log_file,
and sent to Message Server.
•
Log event 2020, with default level standard, is output to stderr
and log_file,
and sent to Message Server.
•
Log event 3020, with default level trace, is output to stderr.
•
Log event 4020, with default level debug, is output to stderr.
Extended log configuration section:
[log-extended] level-reassign-1020=none level-reassign-2020=interaction level-reassign-3020=interaction level-reassign-4020=standard
After the log levels are changed:
•
Log event 1020 is disabled and not logged.
•
Log event 2020 is output to stderr
and log_file.
•
Log event 3020 is output to stderr
and log_file.
•
Log event 4020 is output to stderr
and log_file,
and sent to Message
Server.
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level-reassign-disable
Default Value: false
Valid Values: true, false
Changes Take Effect: Immediately
When this option is set to true
, the original (default) log level of all log events in the
[log-extended]
section are restored. This option is useful when you want to use the default levels, but not delete the customization statements.
Log-Filter Section
This section must be called log-filter.
default-filter-type
Default Value: copy
Valid Values: copy hide skip
The keys and values of the KVList pairs are copied to the log.
The keys of the KVList pairs are copied to the log; the values are replaced with strings of asterisks.
The KVList pairs are not copied to the log.
Changes Take Effect: Immediately
Specifies the default way of presenting KVList information (including
UserData
,
Extensions
, and
Reasons
) in the log. The selected option will be applied to the attributes of all KVList pairs except the ones that are explicitly defined in the log-filter-data
section.
Example
[log-filter] default-filter-type
=
copy
Here is an example of a log using the default log filter settings: message RequestSetCallInfo
AttributeConsultType 3
AttributeOriginalConnID 008b012ece62c8be
AttributeUpdateRevision 2752651
AttributeUserData [111] 00 27 01 00
‘DNIS' '8410'
‘PASSWORD' '111111111'
'RECORD_ID' '8313427'
AttributeConnID 008b012ece62c922
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Log-Filter-Data Section
This section must be called log-filter-data.
<key name>
Default Value: copy
Valid Values: copy hide skip
The key and value of the given KVList pair are copied to the log.
The key of the given KVList pair is copied to the log; the value is replaced with a string of asterisks.
The KVList pair is not copied to the log.
Changes Take Effect: Immediately
Specifies the way of presenting the KVList pair defined by the key name in the log. Specification of this option supersedes the default way of KVList presentation as defined in the
log-filter
section for the given KVList pair.
Note:
If the T-Server common configuration option log-trace-flag
is set to
-udata,
it will disable writing of user data to the log regardless of settings of any options in the log-filter-data
section.
Example
[log-filter-data]
PASSWORD=hide
Here is an example of the log with option
PASSWORD
set to hide: message RequestSetCallInfo
AttributeConsultType 3
AttributeOriginalConnID 008b012ece62c8be
AttributeUpdateRevision 2752651
AttributeUserData [111] 00 27 01 00
'DNIS' '8410'
‘PASSWORD' '****'
'RECORD_ID' '8313427'
AttributeConnID 008b012ece62c922
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Chapter 11: Common Configuration Options Common Section
Common Section
This section must be called common.
enable-async-dns
Default Value: off
Valid Values: off on
Disables asynchronous processing of DNS requests.
Enables asynchronous processing of DNS requests.
Changes Take Effect: Immediately
Enables the asynchronous processing of DNS requests such as, for example, host-name resolution.
Warnings!
Use this option only when requested by Genesys Technical
Support.
Use this option only with T-Servers.
rebind-delay
Default Value:
10
Valid Values:
0
–
600
Changes Take Effect: After restart
Specifies the delay, in seconds, between socket-bind operations that are being executed by the server. Use this option if the server has not been able to successfully occupy a configured port.
Warning!
Use this option only when requested by Genesys Technical
Support.
Changes from 7.5 to 7.6
provides all the changes to common configuration options between release 7.5 and the latest 7.6 release.
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Chapter 11: Common Configuration Options Changes from 7.5 to 7.6
Table 37: Common Log Option Changes from 7.5 to 7.6
Option Name Option Values
Log Section
Type of
Change
Details
Use the following options only when requested by Genesys Technical Support.
x-conn-debug-open 0, 1 New See the description on
x-conn-debug-select 0, 1 New x-conn-debug-timers 0, 1 New
See the description on
See the description on
x-conn-debug-write 0, 1 New x-conn-debug-security x-conn-debug-api x-conn-debug-dns
0, 1
0, 1
0, 1
New
New
New
See the description on
See the description on
See the description on
See the description on
x-conn-debug-all 0, 1 New See the description on
Extended Log Section (New Section)
level-reassign-<eventID> alarm, standard, interaction, trace, debug, none level-reassign-disable true, false
New See the description on
New See the description on
Common Section (New Section)
Use the following options only when requested by Genesys Technical Support.
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Table 37: Common Log Option Changes from 7.5 to 7.6 (Continued)
Option Name
enable-async-dns
Option Values
off, on
Type of
Change
New
Details
rebind-delay 10–600 New
Use only with T-Servers.
See the description on
See the description on
342 Framework 7.6
Chapter
12
T-Server Common
Configuration Options
z z z z z z z z z z z z
This chapter describes the configuration options that are generally common to all T-Server types except where noted. It contains the following sections:
Setting Configuration Options, page 343
Agent-Reservation Section, page 351
Multi-Site Support Section, page 352
Translation Rules Section, page 362
Backup-Synchronization Section, page 362
Call-Cleanup Section, page 363
Timeout Value Format, page 365
Changes from Release 7.5 to 7.6, page 366
T-Server also supports common log options described in Chapter 11,
“Common Configuration Options,” on page 321 .
Setting Configuration Options
Unless it is specified otherwise, you set configuration options in Configuration
Manager in the corresponding sections on the
Options
tab for the T-Server
Application
object.
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Mandatory Options
Except as noted for certain environments, the configuration of common options is not required for basic T-Server operation.
T-Server Section
The T-Server section contains the configuration options that are used to support the core features common to all T-Servers.
TServer
This section must be called
TServer.
ani-distribution
Default Value: inbound-calls-only
Valid Values: inbound-calls-only, all-calls, suppressed
Changes Take Effect: Immediately
Controls the distribution of the ANI information in
TEvent
messages. When this option is set to all-calls,
the
ANI
attribute will be reported for all calls for which it is available. When this option is set to suppressed,
the
ANI
attribute will not be reported for any calls. When this option is set to inbound-calls-only,
the
ANI
attribute will be reported for inbound calls only.
background-processing
Default Value:
false
Valid Values:
true
, false
Changes Take Effect: Immediately
When set to true
, T-Server processes all client requests in the background, giving higher priority to the rest of the messages. This ensures that it processes these messages without any significant delay.
With Background Processing functionality enabled, T-Server processes all switch messages immediately and waits until there are no switch messages before processing the message queue associated with T-Server client requests.
T-Server reads all connection sockets immediately and places client requests in the input buffer, which prevents T-Server clients from disconnecting because of configured timeouts.
When T-Server processes client requests from the message queue, requests are processed in the order in which T-Server received them.
When set to false
, T-Server processes multiple requests from one T-Server client before proceeding to the requests from another T-Server client, and so on.
Note:
Use of this option can negatively impact T-Server processing speed.
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background-timeout
Default Value:
60 msec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: Immediately
Specifies the time interval that T-Server waits before processing client requests in background mode. You must set the
background-processing option to
true in order for this option to take effect.
check-tenant-profile
Default Value:
false
Valid Values:
true
, false
Changes Take Effect: For the next connected client
When set to
true
, T-Server checks whether a client provides the correct name and password of a tenant. If it does, T-Server allows that client to register DNs that are included in the switch configuration in the Configuration Database, but it does not allow the client to register DNs that are not included in the switch configuration.
Note:
To make T-Server compatible with 3.x and 5.x clients, set the
checktenant-profile option to
false.
compatibility-port
Default Value:
0
Valid Values:
0 or any valid TCP/IP port
Changes Take Effect: After T-Server has reconnected to the link
Specifies the TCP/IP port that 3.x clients use to establish connections with
T-Server. Connections to this port are accepted only if T-Server has a connection with the switch. If set to
0
(zero), this port is not used.
Note:
Starting with release 7.5, 3.x clients are no longer supported. You can use this option for backward compatibility with the previous T-Server releases.
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consult-user-data
Default Value:
separate
Valid Values: separate inherited joint
Stores user data for original and consultation calls in separate structures. The data attached to the original call is available for review or changes only to the parties of that call. The data attached to the consultation call is available only to the parties of the consultation call.
Copies user data from an original call to a consultation call when the consultation call is created; thereafter, stores user data separately for the original and the consultation call. Changes to the original call’s user data are not available to the parties of the consultation call, and vice versa.
Stores user data for an original call and a consultation call in one structure. The user data structure is associated with the original call, but the parties of both the original and consultation calls can see and make changes to the common user data.
Changes Take Effect: For the next consultation call created
Specifies the method for handling user data in a consultation call.
Note:
A T-Server client can also specify the
consult-user-data mode in the
Extensions attribute
ConsultUserData key for a conference or transfer request. If it is specified, the method of handling user data is based on the value of the
ConsultUserData key-value pair of the request and takes precedence over the T-Server
consult-user-data option. If it is not specified in the client request, the value specified in the consult-user-data option applies.
customer-id
Default Value: No default value. (A value must be specified for a multi-tenant environment.)
Valid Values: Any character string
Changes Take Effect: Immediately
Identifies the T-Server customer. You must set this option to the name of the tenant that is using this T-Server. You must specify a value for this option if you are working in a multi-tenant environment.
Note:
Do not configure the
customer-id option for single-tenant environments.
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log-trace-flags
Default Value:
+iscc, +cfg$dn, -cfgserv, +passwd, +udata, -devlink, -sw,
-req, -callops, -conn, -client
Valid Values (in any combination):
+/-iscc
+/-cfg$dn
+/-cfgserv
+/-passwd
+/-udata
+/-devlink
+/-sw
+/-req
+/-callops
+/-conn
+/-client
Turns on/off the writing of information about Inter Server Call
Control (ISCC) transactions.
Turns on/off the writing of information about DN configuration.
Turns on/off the writing of messages from Configuration
Server.
Turns on/off the writing of information about passwords.
Turns on/off the writing of attached data.
Turns on/off the writing of information about the link used to send CTI messages to the switch (for multilink environments).
Reserved by Genesys Engineering.
Reserved by Genesys Engineering.
Reserved by Genesys Engineering.
Reserved by Genesys Engineering.
Turns on/off the writing of additional information about the client’s connection.
Changes Take Effect: Immediately
Specifies—using a space-, comma- or semicolon-separated list—the types of information that are written to the log files.
management-port
Default Value:
0
Valid Values:
0 or any valid TCP/IP port
Changes Take Effect: After T-Server is restarted
Specifies the TCP/IP port that management agents use to communicate with
T-Server. If set to
0
(zero), this port is not used.
merged-user-data
Default Value:
main-only
Valid Values: main-only merged-only merged-over-main main-over-merged
T-Server attaches user data from the remaining call only.
T-Server attaches user data from the merging call.
T-Server attaches user data from the remaining and the merging call. In the event of equal keys, T-Server uses data from the merging call.
T-Server attaches data from the remaining and the merging call. In the event of equal keys, T-Server uses data from the remaining call.
Changes Take Effect: Immediately
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Specifies the data that is attached to the resulting call after a call transfer, conference, or merge completion.
Note:
The option setting does not affect the resulting data for merging calls if the
consult-user-data
option is set to
joint.
(See “consult-user-data” on page 346 .)
server-id
Default Value: An integer equal to the
ApplicationDBID
as reported by
Configuration Server
Valid Values: Any integer from
0
–
16383
Changes Take Effect: Immediately
Specifies the Server ID that T-Server uses to generate Connection IDs and other unique identifiers. In a multi-site environment, you must assign each
T-Server a unique Server ID, in order to avoid confusion in reporting applications and T-Server behavior.
Configuration of this option is necessary for Framework environments in which there are two or more instances of the Configuration Database.
Note:
If you do not specify a value for this option, T-Server populates it with the
ApplicationDBID as reported by Configuration Server. Each data object in the Configuration Database is assigned a separate DBID that maintains a unique Server ID for each T-Server configured in the database.
Warning!
Genesys does not recommend using multiple instances of the
Configuration Database.
user-data-limit
Default Value:
16000
Valid Values:
0
–
65535
Changes Take Effect: Immediately
Specifies the maximum size (in bytes) of user data in a packed format.
Note:
When T-Server works in mixed 7.x/6.x environment, the value of this option must not exceed the default value of
16000
bytes; otherwise, 6.x
T-Server clients might fail.
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License Section
The License section contains the configuration options that are used to configure T-Server licenses. They set the upper limit of the seat-related DN licenses ( tserver_sdn
) that T-Server tries to check out from a license file. See
“License Checkout” on page 350
.
license
This section must be called
license
.
Notes:
T-Server also supports the
license-file option described in the
Genesys 7 Licensing Guide.
The
License
section is not applicable to Network T-Server for DTAG.
If you use two or more T-Servers, and they share licenses, you must configure the following options in the license
section of the T-Servers.
num-of-licenses
Default Value:
0 or
max
(all available licenses)
Valid Values:
0
or string
max
Changes Take Effect: Immediately
Specifies how many DN licenses T-Server checks out. T-Server treats a value of
0
(zero) the same as it treats
max
—that is, it checks out all available licenses.
The sum of all
num-of-licenses values for all concurrently deployed
T-Servers must not exceed the number of seat-related DN licenses
( tserver_sdn
) in the corresponding license file. The primary and backup
T-Servers share the same licenses, and therefore they need to be counted only once. T-Server checks out the number of licenses indicated by the value for this option, regardless of the number actually in use.
num-sdn-licenses
Default Value:
0 or
max
(All DN licenses are seat-related)
Valid Values: String
max
(equal to the value of num-of-licenses
), or any integer from
0
–
9999
Changes Take Effect: Immediately
Specifies how many seat-related licenses T-Server checks out. A value of
0
(zero) means that T-Server does not grant control of seat-related DNs to any client, and it does not look for seat-related DN licenses at all.
The sum of all
num-sdn-licenses values for all concurrently deployed
T-Servers must not exceed the number of seat-related DN licenses
( tserver_sdn
) in the corresponding license file. The primary and backup
T-Servers share the same licenses, and therefore they need to be counted only once. T-Server checks out the number of licenses indicated by the value for this option, regardless of the number actually in use.
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Notes:
For Network T-Servers, Genesys recommends setting this option to
0
.
Be sure to configure in the Configuration Database all the DNs that agents use (Extensions and ACD Positions) and that T-Server should control.
License Checkout
shows how to determine the number of seat-related DN licenses that
T-Server attempts to check out. See the examples on page 351 .
Table 38: License Checkout Rules
Options Settings a num-of-licenses num-sdn-licenses
max (or 0) max (or 0) max (or 0) x x x max x
0 max y
0
License Checkout b
Seat-related DN licenses
all available x
0 x min (y, x)
0 a. In this table, the following conventions are used: x
and y
- are positive integers; max
is the maximum number of licenses that T-Server can check out; min (y, x) is the lesser of the two values defined by y
and x
, respectively.
b. The License Checkout column shows the number of licenses that T-Server attempts to check out. The actual number of licenses will depend on the licenses’ availability at the time of checkout, and it is limited to 9999.
Examples
This section presents examples of option settings in the license
section.
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Chapter 12: T-Server Common Configuration Options
Example 1
Agent-Reservation Section
If...
Options Settings
num-of-licenses = max num-sdn-licenses = max
Example 2
If...
Options Settings
num-of-licenses = 1000 num-sdn-licenses = max
Then...
License File Settings License Checkout
tserver_sdn = 500 500 seat-related DNs
Then...
License File Settings License Checkout
tserver_sdn = 500 500 seat-related DNs
Example 3
If...
Options Settings
num-of-licenses = 1000 num-sdn-licenses = 400
Then...
License File Settings License Checkout
tserver_sdn = 600 400 seat-related DNs
Example 4
If...
Options Settings
num-of-licenses = max num-sdn-licenses = 1000
Then...
License File Settings License Checkout
tserver_sdn = 5000 1000 seat-related DNs
Agent-Reservation Section
The Agent-Reservation section contains the configuration options that are used
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agent-reservation
This section must be called
agent-reservation.
Note:
The Agent Reservation functionality is currently a software-only feature that is used to coordinate multiple client applications. This feature does not apply to multiple direct or ACD-distributed calls.
reject-subsequent-request
Default Value:
true
Valid Values: true false
T-Server rejects subsequent requests.
A subsequent request prolongs the current reservation made by the same client application for the same agent.
Changes Take Effect: Immediately
Specifies whether T-Server rejects subsequent requests from the same client application, for an agent reservation for the same
Agent
object that is currently reserved.
Note:
Genesys does not recommend setting this option to
false in a multi-site environment in which remote locations use the
Agent-Reservation feature.
request-collection-time
Default Value:
100 msec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: Immediately
Specifies the interval that agent reservation requests are collected before a reservation is granted. During this interval, agent reservation requests are delayed, in order to balance successful reservations between client applications
(for example, Universal Routing Servers).
reservation-time
Default Value:
10000 msec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: Immediately
Specifies the default interval that an AgentDN is reserved to receive a routed call from a remote T-Server. During this interval, the agent cannot be reserved again.
Multi-Site Support Section
The Multi-Site Support section contains the configuration options that are used to support multi-site environments with the Inter Server Call Control (ISCC)
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extrouter
feature. The configuration options in this section are grouped with related options that support the same functionality (such as those for Transfer Connect
Service or the ISCC/Call Overflow feature).
This section must be called
extrouter
.
For a description of the ways in which T-Server supports multi-site configurations and for an explanation of the configuration possibilities for a
multi-site operation, see the “Multi-Site Support”
chapter.
Note:
In a multi-site environment, you must configure the
,
and
options with the same value for both the primary and backup T-Servers. If you do not do this, the value specified for the backup T-Server overrides the value specified for the primary T-Server.
match-call-once
Default Value:
true
Valid Values: true false
ISCC does not process (match) an inbound call that has already been processed (matched).
ISCC processes (attempts to match) a call as many times as it arrives at an ISCC resource or multi-site-transfer target.
Changes Take Effect: Immediately
Specifies how many times ISCC processes an inbound call when it arrives at an ISCC resource. When set to false,
ISCC processes (attempts to match) the call even if it has already been processed.
Note:
Genesys does not recommend changing the default value of the matchcall-once
option to false
unless you have specific reasons. Setting this option to false
may lead to excessive or inconsistent call data updates.
reconnect-tout
Default Value:
5 sec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: At the next reconnection attempt
Specifies the time interval after which a remote T-Server attempts to connect to this T-Server after an unsuccessful attempt or a lost connection. The number of attempts is unlimited. At startup, T-Server immediately attempts the first connection, without this timeout.
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report-connid-changes
Default Value:
false
Valid Values: true false
EventPartyChanged is generated.
EventPartyChanged is not generated.
Changes Take Effect: Immediately
Specifies whether the destination T-Server generates
EventPartyChanged for the incoming call when the resulting
ConnID attribute is different from the
ConnID attribute of an instance of the same call at the origination location.
use-data-from
Default Value: active
Valid Values: active original active-dataoriginal-call current
The values of
UserData
and
ConnID
attributes are taken from the consultation call.
The values of
UserData
and
ConnID
attributes are taken from the original call.
The value of the
UserData
attribute is taken from the consultation call and the value of
ConnID
attribute is taken from the original call.
If the value of current
is specified, the following occurs:
• Before the transfer or conference is completed, the
UserData
and
ConnID
attributes are taken from the consultation call.
• After the transfer or conference is completed,
EventPartyChanged
is generated, and the
UserData
and
ConnID
are taken from the original call.
Changes Take Effect: Immediately
Specifies the call from which the values for the
UserData and
ConnID attributes are taken for a consultation call that is routed or transferred to a remote location.
Note:
For compatibility with the previous T-Server releases, you can use the values
consult
, main
, and
consult-user-data for this option. These are aliases for
active
, original
, and
current
, respectively.
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ISCC Transaction Options
cast-type
Default Value: route, route-uui, reroute, direct-callid, direct-uui, direct-network-callid, direct-notoken, direct-digits, direct-ani, dnis-pool, pullback
Valid Values: route, route-uui, reroute, direct-callid, direct-uui, direct-network-callid, direct-notoken, direct-digits, direct-ani, dnis-pool, pullback
Changes Take Effect: For the next request for the remote service
Specifies—using a space-, comma- or semicolon-separated list—the routing types that can be performed for this T-Server.
The valid values provide for a range of mechanisms that the ISCC feature can support with various T-Servers, in order to pass call data along with calls between locations.
Because switches of different types provide calls with different sets of information parameters, some values might not work with your T-Server. See
Table 32 on page 283 for information about supported transaction types by a
specific T-Server. The
chapter also provides detailed descriptions of all transaction types.
Notes:
For compatibility with the previous T-Server releases, you can use the direct
value for this option. This is an alias for direct-callid.
An alias, route-notoken,
has been added to the route
value.
default-dn
Default Value: No default value
Valid Values: Any DN
Changes Take Effect: For the next request for the remote service
Specifies the DN to which a call is routed when a Destination DN
(
AttributeOtherDN
) is not specified in the client’s request for routing. If neither this option nor the client’s request contains the destination DN, the client receives
EventError
.
Note:
This option is used only for requests with route types
route, routeuui, direct-callid, direct-network-callid, direct-uui, directnotoken, direct-digits,
and
direct-ani.
direct-digits-key
Default Value:
CDT_Track_Num
Valid Values: Any valid key name of a key-value pair from the
UserData attribute
Changes Take Effect: For the next request for the remote service
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Specifies the name of a key from the
UserData attribute that contains a string of digits that are used as matching criteria for remote service requests with the direct-digits routing type.
Note:
For compatibility with the previous T-Server releases, this configuration option has an alias value of cdt-udata-key.
dn-for-unexpected-calls
Default Value: No default value
Valid Values: Any DN
Changes Take Effect: Immediately
Specifies a default DN for unexpected calls arriving on an External Routing
Point.
network-request-timeout
Default Value:
20 sec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: For the next network request
For a premise T-Server, this option specifies the time interval that the premise
T-Server waits for a response, after relaying a
TNetwork<...>
request to the
Network T-Server. For a Network T-Server, this option specifies the time interval that the Network T-Server waits for a response from an SCP (Service
Control Point), after initiating the processing of the request by the SCP.
When the allowed time expires, the T-Server cancels further processing of the request and generates
EventError
.
register-attempts
Default Value:
5
Valid Values: Any positive integer
Changes Take Effect: For the next registration
Specifies the number of attempts that T-Server makes to register a dedicated
External Routing Point.
register-tout
Default Value:
2 sec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: For the next registration
Specifies the time interval after which T-Server attempts to register a dedicated
External Routing Point. Counting starts when the attempt to register a Routing
Point fails.
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request-tout
Default Value:
20 sec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: For the next request for remote service
Specifies the time interval that a T-Server at the origination location waits for a notification of routing service availability from the destination location.
Counting starts when the T-Server sends a request for remote service to the destination site.
resource-allocation-mode
Default Value:
circular
Valid Values: home circular
T-Server takes an alphabetized (or numerically sequential) list of configured DNs and reserves the first available DN from the top of the list for each new request. For example, if the first DN is not available, the second DN is allocated for a new request. If the first
DN is freed by the time the next request comes, the first DN is allocated for this next request.
T-Server takes the same list of configured DNs, but reserves a subsequent DN for each subsequent request. For example, when the first request comes, T-Server allocates the first DN; when the second request comes, T-Server allocates the second DN; and so on.
T-Server does not reuse the first DN until reaching the end of the
DN list.
Changes Take Effect: Immediately
Specifies the manner in which T-Server allocates resources (that is, DNs of the
External Routing Point type and Access Resources with
Resource Type dnis
) for multi-site transaction requests.
resource-load-maximum
Default Value:
0
Valid Values: Any positive integer
Changes Take Effect: Immediately
Specifies the maximum number of ISCC routing transactions that can be concurrently processed at a single DN of the
External Routing Point route type. After a number of outstanding transactions at a particular DN of the
External Routing Point type reaches the specified number, T-Server considers the DN not available. Any subsequent request for this DN is queued until the number of outstanding transactions decreases. A value of
0
(zero) means that no limitation is set to the number of concurrent transactions at a single External Routing Point. In addition, the
0 value enables T-Server to perform load balancing of all incoming requests among all available External
Routing Points, in order to minimize the load on each DN.
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route-dn
Default Value: No default value
Valid Values: Any DN
Changes Take Effect: Immediately
Specifies the DN that serves as a Routing Point for the route transaction type in the multiple-to-one access mode.
timeout
Default Value:
60 sec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: For the next request for remote service
Specifies the time interval that the destination T-Server waits for a call routed from the origination location. Counting starts when this T-Server notifies the requesting T-Server about routing service availability. The timeout must be long enough to account for possible network delays in call arrival.
use-implicit-access-numbers
Default Value:
false
Valid Values:
true
, false
Changes Take Effect: After T-Server is restarted
Determines whether an External Routing Point in which at least one access number is specified is eligible for use as a resource for calls coming from switches for which an access number is not specified in the External Routing
Point. If this option is set to false,
the External Routing Point is not eligible for use as a resource for calls coming from such switches. If this option is set to true,
an implicit access number for the External Routing Point, composed of the switch access code and the DN number of the External Routing Point, will be used.
Note:
If an External Routing Point does not have an access number specified, this option will not affect its use.
Transfer Connect Service Options
tcs-queue
Default Value: No default value
Valid Values: Any valid DN number
Changes Take Effect: For the next request for the remote service
Specifies the TCS DN number to which a call, processed by the TCS feature, is dialed after the originating external router obtains an access number. This option applies only if the
option is activated.
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tcs-use
Default Value:
never
Valid Values: never always app-defined
The TCS feature is not used.
The TCS feature is used for every call.
In order to use the TCS feature for a multi-site call transfer request, a client application must add a key-value pair with a
TC-type key and a nonempty string value to the
UserData attribute of the request.
Changes Take Effect: Immediately
Specifies whether the Transfer Connect Service (TCS) feature is used.
Note:
For compatibility with the previous T-Server releases, you can use the value up-app-depended for this option. This is an alias for app-defined
.
ISCC/COF Options
cof-ci-defer-create
Default Value:
0
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: Immediately
Specifies the time interval that T-Server waits for call data from the switch before generating a negative response for a call data request from a remote
T-Server. If T-Server detects the matching call before this timeout expires, it sends the requested data. This option applies only if the
option is set to true
.
cof-ci-defer-delete
Default Value:
0
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: Immediately
Specifies the time interval that T-Server waits before deleting call data that might be overflowed. If set to
0,
deletion deferring is disabled. This option applies only if the
option is set to true
.
cof-ci-req-tout
Default Value:
500 msec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: For the next COF operation
Specifies the time interval during which T-Server will wait for call data requested with respect to a call originated at another site. After T-Server sends the call data request to remote T-Servers, all events related to this call will be suspended until either the requested call data is received or the specified
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option is set to true
.
cof-ci-wait-all
Default Value:
false
Valid Values: true false
T-Server waits for responses from all T-Servers that might have the requested call data before updating the call data with the latest information.
T-Server updates the call data with the information received from the first positive response.
Changes Take Effect: Immediately
Specifies whether T-Server, after sending a request for matching call data, waits for responses from other T-Servers before updating the call data (such as
CallHistory
,
ConnID
, and
UserData
) for a potentially overflowed call. The waiting period is specified by the
and
options.
This option applies only if the
true
.
cof-feature
Default Value:
false
Valid Values:
true, false
Changes Take Effect: Immediately
Enables or disables the Inter Server Call Control/Call Overflow (ISCC/COF) feature.
cof-rci-tout
Default Value:
10 sec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: For the next COF operation
Specifies the time interval that T-Server waits for call data from other
T-Servers’ transactions. Counting starts when
expires. This option applies only if the
true
.
local-node-id
Default Value:
0
Valid Values:
0 or any positive integer
Changes Take Effect: Immediately
This option, if enabled, checks all networked calls against the specified
NetworkNodeID
(the identity of the switch to which the call initially arrived). If the
NetworkNodeID is the same as the value of this option, the request for call information is not sent. The default value of
0 disables the functionality of this option. To establish an appropriate
NetworkNodeID,
specify a value other than the default. This option applies only if the
option is set to true
.
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Chapter 12: T-Server Common Configuration Options Multi-Site Support Section
Note:
This option applies only to T-Server for Nortel Communication Server
2000/2100 (formerly DMS-100).
default-network-call-id-matching
Default Value: No default value
Valid Values: sip
Changes Take Effect: Immediately
When this option is set to sip,
SIP Server will use the content of the
X-ISCC-CofId
header for the ISCC/COF call matching.
To activate this feature, the
extrouter
section of the
SIP Server
Application
object must also be set to true
.
Note:
This option applies only to SIP Server.
Event Propagation Option
event-propagation
Default Value:
list
Valid Values: list off
Changes in user data and party events are propagated to remote locations through call distribution topology.
The feature is disabled. Changes in user data and party events are not propagated to remote locations.
Changes Take Effect: Immediately
Specifies whether the Event Propagation feature is enabled.
Number Translation Option
inbound-translator-<n>
Default Value: No default value.
Valid Value: Any valid name
Changes Take Effect: Immediately
Specifies the name of another configuration section as the value for the inbound-translator
option. For example, inbound-translator-1 = ani-translator where ani-translator
is the name of the configuration that describes the translation rules for inbound numbers.
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Translation Rules Section
The section name is specified by the inbound-translator-< n
>
option. It contains options that define translation rules for inbound numbers.
You can choose any name for this section, provided that it matches the value of the section. Every option in this section corresponds to a rule and must conform to the format described below. You can configure as many rules as necessary to accommodate your business needs.
rule-<n>
Default Value: No default value
Valid Value: Any valid string in the following format: in-pattern=<input pattern value>;out-pattern=<output pattern value>
Changes Take Effect: Immediately
Defines a rule to be applied to an inbound number. The two parts of the option value describe the input and output patterns in the rule. When configuring the
pattern values, follow the syntax defined in “Using ABNF for Rules” on page 292
. See “Configuring Number Translation” on page 298
for examples of these rules as well as detailed instructions for creating rules for your installation. For example, a value for this configuration option might look like this: rule-01 = in-pattern=0111#CABBB*ccD;out-pattern=ABD
Backup-Synchronization Section
The Backup-Synchronization section contains the configuration options that are used to support a high-availability ( hot standby redundancy type) configuration.
backup-sync
This section must be called
backup-sync.
Note:
These options apply only to T-Servers that support the
hot standby redundancy type.
addp-remote-timeout
Default Value:
0
Valid Values: Any integer from
0
–
3600
Changes Take Effect: Immediately
Specifies the time interval that the redundant T-Server waits for a response from this T-Server after sending a polling signal. The default value of
0
(zero) disables the functionality of this option. To establish an appropriate timeout, specify a value other than the default. This option applies only if the
option is set to addp.
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addp-timeout
Default Value:
0
Valid Values: Any integer from
0
–
3600
Changes Take Effect: Immediately
Specifies the time interval that this T-Server waits for a response from another
T-Server after sending a polling signal. The default value of
0
(zero) disables the functionality of this option. To establish an appropriate timeout, specify a value other than the default. This option applies only if the
option is set to
addp.
addp-trace
Default Value:
off
Valid Values: off
, false
, no full
, both
No trace (default).
local
, on
, true
,
yes
Trace on this T-Server side only.
remote
Trace on the redundant T-Server side only.
Full trace (on both sides).
Changes Take Effect: Immediately
Specifies whether the option is active, and to what level the trace is performed.
This option applies only if the
addp
.
protocol
Default Value:
default
Valid Values: default addp
The feature is not active.
Activates the Advanced Disconnect Detection Protocol.
Changes Take Effect: When the next connection is established
Specifies the name of the method used to detect connection failures. If you specify the
addp
value, you must also specify a value for the addp-timeout ,
and
sync-reconnect-tout
Default Value:
20 sec
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: Immediately
Specifies the time interval after which the backup T-Server attempts to reconnect to the primary server (for a synchronized link).
Call-Cleanup Section
The Call-Cleanup section contains the configuration options that are used to control detection and cleanup of stuck calls in T-Server. For more information
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Chapter 12: T-Server Common Configuration Options Call-Cleanup Section
call-cleanup
on stuck call handling, refer to the “Stuck Call Management” chapter in the
Framework 7.6 Management Layer User’s Guide.
This section must be called
call-cleanup.
cleanup-idle-tout
Default Value:
0
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: Immediately
Specifies the time interval that T-Server waits for a call to be updated from its last update. After this time elapses, if no new events about the call are received, T-Server clears this call as a stuck call, either by querying the switch
(if a CTI link provides such capabilities) or by deleting the call information from memory unconditionally. The default value of
0 disables the stuck calls cleanup.
notify-idle-tout
Default Value:
0
Valid Values: See “Timeout Value Format” on page 365
Changes Take Effect: Immediately
Specifies the time interval that T-Server waits for a call to be updated from its last update. After this time elapses, if no new events about the call are received, T-Server reports this call as a stuck call. The default value of
0 disables the stuck calls notification.
periodic-check-tout
Default Value:
10 min
Valid Values: See “Timeout Value Format” on page 365 .
Changes Take Effect: Immediately
Specifies the time interval for periodic checks for stuck calls. These checks affect both notification and cleanup functionality, and are made by checking the T-Server’s own call information with call information available in the switch. For performance reasons, T-Server does not verify whether the notify-idle-tout or
cleanup-idle-tout option has expired before performing this checking.
Note:
Setting this option to a value of less than a few seconds can affect
T-Server performance.
Examples
This section presents examples of option settings in the call-cleanup
section.
Example 1
cleanup-idle-tout = 0 notify-idle-tout = 0
364 Framework 7.6
Chapter 12: T-Server Common Configuration Options Security Section periodic-check-tout = 10
With these settings, T-Server will not perform any checks for stuck calls.
Example 2
cleanup-idle-tout = 0 notify-idle-tout = 5 min periodic-check-tout = 10 min
With these settings, T-Server performs checks every 10 minutes and sends notifications about all calls that have been idle for at least 5 minutes.
Example 3
cleanup-idle-tout = 20 min notify-idle-tout = 5 min periodic-check-tout = 10 min
With these settings, T-Server performs checks every 10 minutes, sends notifications about all calls that have been idle for at least 5 minutes, and attempts to clean up all calls that have been idle for more than 20 minutes.
Security Section
The
Security
section contains the configuration options that are used to configure secure data exchange between T-Servers and other Genesys components. Refer to the Genesys 7.6 Security Deployment Guide for complete information on the security configuration.
Timeout Value Format
This section of the document describes the values to use for those T-Server common options that set various timeouts. The current format allows you to use fractional values and various time units for timeout settings.
For timeout-related options, you can specify any value that represents a time interval, provided that it is specified in one of the following formats:
[[[ hours
:] minutes
:] seconds
][ milliseconds
] or
[ hours hr][ minutes min][ seconds sec][ milliseconds msec]
Where a time unit name in italic (such as hours) is to be replaced by an integer value for this time unit.
Integer values with no measuring units are still supported, for compatibility with previous releases of T-Server. When you do not specify any measuring units, the units of the default value apply. For example, if the default value equals
60 sec
, specifying the value of
30 sets the option to 30 seconds.
Example 1
The following settings result in a value of 1 second, 250 milliseconds:
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sync-reconnect-tout = 1.25
sync-reconnect-tout = 1 sec 250 msec
Example 2
The following settings result in a value of 1 minute, 30 seconds: timeout = 1:30 timeout = 1 min 30 sec
Changes from Release 7.5 to 7.6
lists the configuration options that:
•
Are new or changed in the 7.6 release of T-Server
•
Have been added or changed since the most recent 7.5 release of this document
If a configuration option has been replaced with another that enables the same functionality, the new option name and its location in this chapter are noted.
Table 39: Option Changes from Release 7.5 to 7.6
Option Name Option Values Type of
Change
ani-distribution inbound-calls-only, all-calls, suppressed
TServer Section
New compatibility-port 0 or any valid TCP/IP port Obsolete
Details
See the option description on
.
See the option description on
.
use-data-from default-networkcall-id-matching
extrouter Section
active, original, current, active-data-original-call
New value sip New
New option value, active-dataoriginal-call.
See the option
.
See the option description on
.
network-providedaddress true, false
backup-sync Section
Obsolete
366 Framework 7.6
Part
3
Part Three: DMX Reference
Information
Part Three of this SIP Server Deployment Guide contains reference information specific to the Distributed Media eXchange (DMX) application.
The information is divided among these chapters:
•
Chapter 13, “DMX Deployment,” on page 369
, presents configuration and installation procedures for DMX.
•
Chapter 14, “DMX Reference,” on page 393 , presents reference
information for DMX.
New in DMX for 7.6
There are no new features available in the initial 7.6 release of DMX.
SIP Server—Deployment Guide 367
Part Three: DMX Reference Information New in DMX for 7.6
368 Framework 7.6
Chapter
13
DMX Deployment
This chapter describes the deployment of
Distributed Media eXchange
(DMX)
. It contains the following sections:
z z z z
DMX Configuration Options, page 376
Configuring for Use with the Management Layer, page 388
Starting and Stopping DMX, page 392
z z
Overview
Distributed Media eXchange (DMX) negotiates codecs and converts protocols between SIP and endpoints using H.323 (H.225/245). It also enables the exchange of audio and video media types across many networks, such as
PSTN, PBX, LAN, and WAN.
Note:
Microsoft NetMeeting and Windows Messenger are not supported with
DMX beginning with release 7.2.
DMX General Network Modes
DMX can perform in Traditional and Load Balancing modes.
Traditional Mode
In this mode, DMX is deployed with or without SIP Server (or any third-party
SIP Service). Several combinations are possible, and in all combinations DMX
SIP Server—Deployment Guide 369
Chapter 13: DMX Deployment Overview works as a proxy between H.323 and SIP networks. DMX can send requests to different destinations depending on the scenario used:
Scenario One
In the first scenario ( Figure 37 ), DMX is deployed between the gateway/PSTN
and SIP Server and functions as a proxy between SIP Server and the H.323 gateway.
PSTN
PSTN Gateway
DMX
SIP
SIP Server
SIP
Agent Endpoint
SIP
Agent Endpoint
Figure 37: Scenario One: DMX Traditional Mode
For each call DMX would have to find the proper destination to send messages:
•
If the destination is SIP Server, DMX obtains its IP address using the link from the
Connections
tab from Configuration Manager. If there are several servers listed, DMX gives preference to the first SIP Server.
•
If the destination is H.323, DMX resolves the IP address of the destination gateway from the
TO
header within the incoming
INVITE
message from SIP
Server. For example, if the
TO
header of the
INVITE
message is defined as
[email protected], then the H.323
SETUP
message would be sent to IP address 192.168.83.6.
•
If SIP Server modifies the Request-URI so that IP address in the Request-
URI differs from
TO
header, DMX processes the IP address from the
Request URI header. For example, if the
TO
header is defined as
[email protected], but the Request-URI is defined as
[email protected], then H.323
SETUP
message would be sent to IP address 192.168.83.76.
Scenario Two
In the second scenario ( Figure 38 ), DMX is deployed as a proxy between SIP
Server and H.323 caller endpoints.
Customer
Endpoint
DMX
SIP
SIP Server
SIP
Agent Endpoint
SIP
Agent Endpoint
Figure 38: Scenario Two: DMX Traditional Mode
370 Framework 7.6
Chapter 13: DMX Deployment Overview
For each call, DMX needs to find the proper destination to send messages:
•
For inbound calls from H.323 agents, DMX obtains its IP address using the link from the
Connections
tab from Configuration Manager. If there are several servers listed, DMX gives preference to the first SIP Server.
•
For inbound calls from the SIP Server, DMX resolves the IP address from the number/alias established by the URI if the destination URI of the incoming
INVITE
message does not have a direct IP address. To do this, the
H.323 client should have been previously registered by RAS. For example, if a H.323 client has registered with DMX as number 5650, with an IP address of 192.168.83.6, the destination URI of the incoming
INVITE message is defined as 5650 and the H.323
SETUP
message is sent to IP
192.168.83.6.
•
For inbound calls from a SIP Server or SIP Client that includes an
INVITE message with a direct IP address of the H.323 client, the IP addresses could be retrieved using these methods:
If the IP address from the
TO
header is similar to the DMX IP address,
DMX resolves the IP address from the number/alias established by the
URI if the destination URI of the incoming
INVITE
message does not have a direct IP address. To do this, the H.323 client should have been previously registered by RAS. For example, if an H.323 client has registered with DMX as number 5650, with an IP address of
192.168.83.6, the destination URI of the incoming
INVITE
message is defined as 5650 and the H.323
SETUP
message is sent to IP address
192.168.83.6.
If the IP address from the
TO
header is not equal to the DMX IP address, DMX uses the IP address from the URI path within the incoming
INVITE
message. For example, if the destination URI of the
INVITE
message is defined as [email protected] and the DMX was started at host identified by IP address 192.168.83.7, the H.323
SETUP message would be sent to IP 192.168.83.76.
If the
TO
header field specifies the desired “logical” recipient of the request, SIP Server can affect the Request-URI of the message.
Therefore the IP in the Request-URI field can differ from the IP in
TO header. In this scenario, the Request-URI field has higher priority and
DMX processes it only. For example, if the
TO
header is defined as
[email protected], but the Request-URI is defined as
[email protected], then an H.323
SETUP
message would be sent to
IP address 192.168.83.76.
Scenario Three
In the third scenario ( Figure 39 ), DMX is deployed only as a proxy between
the H.323 agents and the SIP agents.
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Chapter 13: DMX Deployment Overview
H.323
Customer Endpoint
H.323
Customer Endpoint
DMX
SIP
Agent Endpoint
Figure 39: Scenario Three: DMX Traditional Mode
The H.323 agent must register with DMX before any calls are made. DMX would have to find the proper destination to send messages for each call:
•
For inbound calls from H.323 agents, DMX obtains its IP address by using the link from the
Connections
tab from Configuration Manager. If there are several servers listed, DMX gives preference to the first SIP Server.
•
For inbound calls from SIP agents, DMX resolves the IP address from the number/alias established by the URI if the destination URI of the incoming
INVITE
message does not have a direct IP address. To do this, the H.323 client should have been previously registered by RAS. For example, if an
H.323 client has registered with DMX as number 5650, with an IP address of 192.168.83.6, the destination URI of the incoming
INVITE
message is defined as 5650 and the H.323
SETUP
message is sent to IP address
192.168.83.6.
•
For inbound calls from a SIP Server or SIP Client that includes an
INVITE message with a direct IP address of the H.323 client, the IP addresses could be retrieved using these methods:
If the IP address from the
TO
header is similar to the DMX IP address,
DMX resolves the IP address from the number/alias established by the
URI if the destination URI of the incoming
INVITE
message does not have a direct IP address. To do this, the H.323 client should have been previously registered by RAS. For example, if a H.323 client has registered with DMX as number 5650, with an IP address of
192.168.83.6, the destination URI of the incoming
INVITE
message is defined as 5650 and the H.323
SETUP
message is sent to IP address
192.168.83.6.
If the IP address from the
TO
header is not equal to the DMX IP address, DMX uses the IP address from the URI path within the incoming
INVITE
message. For example, if the destination URI of the
INVITE
message is defined as [email protected] and the DMX was started at a host identified by IP address 192.168.83.7, the H.323
SETUP message would be sent to IP address 192.168.83.76.
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Load Balancing Mode
In addition to Traditional mode, DMX can work as a gatekeeper with a load balancing mechanism. In this scenario, DMX can distribute H.323 calls between DMX clients which reduces the call load among several DMXs. All
DMX clients must be registered with DMX gatekeeper.
Customer
Endpoint
DMX
Gatekeeper
DMX Client 1
DMX Client 2
SIP Server
Agent Endpoint
SIP
Agent Endpoint
Figure 40: DMX Load Balancing Mode
If the H.323 agent sends a RAS Admission Request to the DMX gatekeeper,
DMX transfers the call to the DMX client with its IP address in the RAS
Admission Confirm response. If the H.323 agent sends a
SETUP
message to the
DMX gatekeeper, DMX processes the call. Therefore, DMX can work either as a DMX gatekeeper or as a DMX client simultaneously, depending on the message received: if DMX receives a
SETUP
message, it processes it and performs in Traditional mode; otherwise, DMX works as a gatekeeper.
DMX gatekeeper uses a comma-separated list in the services
option of the
Gatekeeper
section as the determination for call distribution among client
DMXs. Ordinarily, calls are distributed in round-robin fashion among all
DMXs registered with a gatekeeper, but if a client DMX has a service number specified, all calls matching that number are sent only to that client. The option access-registrations
in the properties of the DMX gatekeeper should be set to true
.
DMX client determines gatekeeper registration from the primary-gk-host
and primary-gk-port
options in the
Gatekeeper
section of the
Options
tab in
Configuration Manager.
DMX Configuration
You need to ensure your system has the following components installed and configured in order to properly operate DMX:
•
Genesys Framework components including DB Server, Configuration
Server, and Configuration Manager. Genesys Framework includes SIP
Server, which is the most important part of the Media Layer.
This layer enables Genesys solutions to communicate with various media, including traditional telephony systems, voice over IP (VOIP), e-mail, and the Web.
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Chapter 13: DMX Deployment DMX Configuration
This layer also provides the mechanism for distributing interactionrelated business data, also referred to as attached data, within and across solutions.
You can expand the basic setup by using multiple DMXs. Remember that by default DMX uses the preassigned TCP/UDP ports (5060 for SIP and 1720 for
H.323 communications), and so it cannot be run on the same host as any other
SIP/H.323 application (such as a VoIP endpoint, or a second DMX). However, if the DMX configuration uses only one VoIP protocol (either SIP or H.323), you can change the port used by DMX by using the h225-port
option, or the sip-port
option. The same port should also be specified in the DMX option callport
.
Templates
DMX 7.6 requires that you use the appropriate template for your SIP Server release. DMX is supplied with four templates matching different modes of
DMX operation as shown in Table 40 .
Table 40: DMX 7.6 Templates
Template Name
dmx_client_76.apd
dmx_dualmode_76.apd
dmx_gk_76.apd
dmx_standalone_mode_76.apd
Description
Requires the configuration of a DMX gatekeeper. This DMX client will receive and process the calls distributed by DMX gatekeeper.
Functions as both a gatekeeper and a client to a gatekeeper, allowing it to share the work of processing calls.
Acts to regulate other DMX modules. Deploying DMX in a gatekeeper topology will result in an enhanced availability of the solution, as well as the ability to use the component within a load balancing and/or multi-tenant application. By choosing the gatekeeper or client modes of operation, you must configure at least 2 DMX modules. One acts as a gatekeeper and one as a client to handle calls. The gatekeeper DMX monitors client DMXs and is able to reroute calls if a client fails. Because the gatekeeper distributes calls in round-robin fashion, the gatekeeper/client configuration enables load balancing among multiple DMXs.
Operates in the stand-alone mode with no gatekeeper required.
You have the option of deploying the DMX server as a stand-alone
H.323/SIP call control device. The stand-alone configuration results in the simplest deployment where only one DMX is required.
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Note:
Since DMX gatekeeper mode is implemented only in H.323, gatekeeper and client templates can be used only in an H.323 environment. Stand-alone and dual-mode templates can be used in SIP,
H.323, and mixed SIP-H.323 environments.
Wizard Configuration
To use the Wizards to install DMX, install the Wizard from the CD, run the
Genesys Wizard Manager, and follow the instructions.
Manual Configuration
This section describes setting up a simple configuration for DMX, in a single-tenant environment. For each configuration field, the tables in this section provide the value used in this example configuration, plus a description of what should go in the field so that users can tailor the example to their own environment. If a field is not mentioned, you should leave it unchanged.
Procedure:
Configuring DMX manually
Start of procedure
1. Import application templates:
a. Open Configuration Manager and select the
Environment\Application
Templates folder.
b. Select
File > Import Application Template
and browse to the
Templates directory.
c. Import the template for DMX.
2. Create a DMX
Application
object:
a. Select the
Applications
folder, and then select
File > New >
Application.
b. Select from one of the templates and enter field values in the resulting
Properties
dialog box, as shown in
.
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Chapter 13: DMX Deployment DMX Configuration Options
Table 41: DMX Properties
Tab
General
Server Info
Start Info
Connections
Field
Name
Description
Any descriptive name
Host The host on which DMX runs
Communication
Port
Not used since release 7.2; the value 0 is a placeholder to satisfy Configuration Manager minimum information conditions
Working
Directory
Pathname to the directory or folder holding the DMX application
Command Line Name of the DMX executable file
Command-line
Arguments
Host name and port number of Configuration
Server, name of DMX application
Timeout
(Startup,
Shutdown)
The time interval, in seconds, during which the application is expected to start/shut down
Any SIP Server configured in Configuration
Manager. If there is more than one SIP Server application specified, DMX will choose the first one.
3. When you are finished, click
Apply to save configuration changes
.
End of procedure
DMX Configuration Options
DMX configuration options are grouped into call, link, x-config,
and gatekeeper
sections, as follows.
Call Section
This section must be named call
. The name is case-sensitive.
call-address
Default Value:
$HOST
Valid Values:
$HOST
, any IP address
Changes Take Effect: Immediately
Specifies the IP address of the DMX.
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Chapter 13: DMX Deployment DMX Configuration Options
call-port
Default Value: None (protocol dependent)
Valid Values:
1-65535
Changes Take Effect: Immediately
This option represents a non-standard value of the TCP/IP port to call DMX.
call-protocol
Default Value: smcp
Valid Value: smcp
Changes Take Effect: On restart
Warning!
Do not alter this option from its Default Value.
Specifies the type of the application containing this call section.
Link Section
This section must be called link
. The name is case-sensitive.
protocol
Default Value: tcp
Valid Value: tcp
Changes Take Effect: Immediately
Warning!
Do not alter this option from its Default Value.
Describes low-level protocol.
verbose
Default Value:
1
Valid Values:
0, 1
Changes Take Effect: Immediately
Warning!
Do not alter this option from its Default Value.
Sets verbosity level for this link.
X-Config Section
DMX connects directly to Configuration Server.
Every x-config
section must include an option called x-type,
with values dmx or sm
specifying to which application it applies. The x-config
section can also include an option x-name
specifying the application name; if x-name
is absent,
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Chapter 13: DMX Deployment DMX Configuration Options the application name is taken from the object (application) name in the configuration system.
The following list includes the equivalent command-line parameter for each option.
This section must be called x-config
. The name is case-sensitive.
amd-mode
Default Value: false
Valid Values: true, false, 1, 0
Changes Take Effect: Immediately
Enables or disables Answering Machine Detection in outgoing H.323 calls according to the AudioCodes specification.
Note:
This option should be set to false
if DMX is used with a non-AudioCodes gateway or if Call Progress Detection is not used.
audio-codec
Default Value:
1,3
Valid Values: Comma-separated list of any of the following:
1
G.711 mu-Law
2
3
4
8
G.711 A-Law
G.723
G.729.A
MS-GSM and GSM Full Rate
Command-line Equivalent:
-audio_codec
Changes Take Effect: Immediately
Specifies the audio codec(s) to be used by this DMX. G.711 operates at a higher bit rate than G.723 and G.729.A, providing better quality but consuming more network resources. MS-GSM/GSM FR is intermediate, providing moderate quality and low network resource consumption. For G.711, mu-Law is used in North America and Japan and A-Law is used elsewhere, including international routes.
bit-rate
Default Value:
6217
Valid Value:
1-19200
Command-line Equivalent:
-bit_rate
Changes Take Effect: Immediately
Specifies the maximum bit rate for video codec(s). Applies only if the video-codec
option ( page 385 ) has a value other than
0
(zero).
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cif-mpi
Default Value:
0
Valid Value:
0-32
Command-line Equivalent:
-cif_mpi
Changes Take Effect: Immediately
Specifies the minimum picture interval, in units of 1/29.97, for the encoding and decoding of Common Intermediate Format (CIF) pictures for both H.261 and H.263 codecs. A value of
0
(zero) indicates no capability for CIF pictures.
Applies only if the video-codec
option ( page 385 ) has a value other than
0
(zero). The maximum value is
4
when using the H.261 codec. The maximum value is
32
for the H.263 codec.
debug-level
Default Value:
0
Valid Values:
0-3
Command-line Equivalent:
-debug_level
Changes Take Effect: Immediately
Applies only if the Log option verbose
is set to all
. It sets the amount of detail in log events of the Debug level. A higher number produces more detail.
delay-own-tcs
Default Value:
2000
Valid Values:
0-2000
Command-line Equivalent: None
Changes Take Effect: Immediately
Resolves the DMX and Cisco CallManager interoperability issue. The value is specified in milliseconds.
dtmf-payload-type
Default Value:
101
Valid Values:
96
–
127
Command-line Equivalent: None
Changes Take Effect: Immediately
Specifies the RTP Payload for DTMF digits carried in the RTP packets according to RFC2833 standards.
Note that in some cases, this value should be identical in both the gateway and in DMX. For example, when working with a Cisco Gateway, the gateway value and the DMX value for this option should be the same.
dtmf-sip
Default Value: rtp-nte
Valid Values: no
, rtp-nte
Command-line Equivalent:
-dtmf_sip
Changes Take Effect: Immediately
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Defines the way DTMF signals are processed in the SIP protocol. If the value is no
, no DTMF signal is processed. The rtp-nte
value allows DTMF relay using NTE RTP packets. DTMF tones are encoded in the NTE format and transported in the same RTP channel as the voice.
dtmf-source
Default Value: auto
Valid Values: auto h245ui h245sig
No rtp-nte
Chooses source based on Terminal Capabilities Set of the remote endpoint.
H.245 user input signals will take place.
H.245 signal will be applied.
No DTMF signals will be processed.
Allows you to encode DTMF signals into the RTP packets and transport them on the same channel as voice.
Command-line Equivalent:
-dtmf
Changes Take Effect: Immediately
Specifies the source for DTMF generation. Use the following formats: dtmf-source=value1,value2
or dtmf-source-value1
Where:
•
value1
or value2
can be rtp-nte.
•
h245ui, h245sign
cannot be assigned simultaneously.
•
The no
value has higher priority and, therefore, can silently discard other values.
h225-port
Default Value:
1720
Valid Value: Any positive integer (any valid port)
Command-line Equivalent:
-h225_port
Changes Take Effect: Upon restart
Specifies the listening port for the H.225 protocol. Use this option to override the default port setting for DMX if you want to run DMX on the same machine as another H.323 application.
h264-level
Default Value:
71
Valid Values: See
Changes Take Effect: Immediately
This is a mandatory setting. It is supported only when using the H.264 protocol with a vPoint client. This option defines parameters which are used during the
H.264 codec negotiation procedure. Use this option to improve video quality by setting the level or the degree of the video’s capabilities (for example, its
380 Framework 7.6
Chapter 13: DMX Deployment DMX Configuration Options pixel resolution, speed of decoding, number of Macroblocks per second, and so on). The video quality also depends on the endpoint’s encode/decode stream capability.
Table 42: Valid Values for the h.264-level Option
Level Parameter Value H.264 Level Number
15 1
22
29
1.1
1.2
36
43
50
64
1.3
2
2.1
3
99
106
113
71
78
85
92
3.1
3.2
4
4.1
4.2
5
5.1
h264-profile
Default Value:
64
Valid Values:
16, 32, 64 (
)
Changes Take Effect: Immediately
This is a mandatory setting. It is supported only when using the H.264 protocol with a vPoint client. This option defines parameters which are used during the
H.264 codec negotiation procedure. This option allows you to define the profile, which improves video quality. Each profile that you define represents a set of algorithmic features.
describes the Codes and Profiles used with this option, which are based on the H.264 standard. The video quality also depends on the endpoint’s encode/decode stream capability.
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Table 43: Profile Values and Descriptions
Valid Profile
Value
64
Profile Description
32
16
Baseline Profile. The basic goal of H.264 was to provide a royalty-free baseline profile to encourage early application of the H.264 protocol. The baseline profile consists of most of the major features of the H.264 codec, with the exception of: B slices and weighted prediction; CABAC encoding; field coding; and SP & SI slices. Thus, the Baseline Profile is appropriate for many progressive scan applications such as video conferencing and video-over-IP, but not for interlaced television or multiple stream applications.
Main Profile. The Main Profile contains all of the features as in
Baseline, except for flexible macro block ordering (FMO), arbitrary slice order (ASO) and redundant slices. However, it adds field coding, B slices and weighted prediction, and
CABAC entropy coding. This profile is appropriate for efficient coding of interlaced television applications where bit or packet error is not excessive, and where low latency is not a requirement.
Extended Profile. This profile contains all features from the
Baseline profile and Main Profile, except that CABAC is not supported. In addition, the Extended Profile adds SP and SI for stream switching, and up to 8 slice groups. This profile is appropriate for server-based streaming applications where bitrate scalability and error rate are very important; for example,
Mobile video services.
Note:
The vPoint HD conferencing system supports the Baseline Profile.
packet-size
Default Value:
20
Valid Values: Comma-separated list of any of:
1, 2, 3, 4, 20, 30
(see
)
Command-line Equivalent:
-pstr
Changes Take Effect: Immediately
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Specifies maximum packet size, as follows:
Table 44: Values of Packet-Size Option
Codec
G.711 mu-Law or A-Law
G.723, G.729.A
MS-GSM
GSM Full Rate
Default
Value
20
Valid
Values
20, 30
Unit
1 1, 2, 3, 4 milliseconds per packet frames per packet
Not applicable
Not applicable.
Note: This option does not apply to the MS-GSM and GSM Full Rate codecs.
If an
option is specified for more than one codec, the values of packet-size correspond one-to-one, in the same order. For example, suppose an audio-codec has values
1, 2, 4
and its packet-size has values
20, 30, 2.
That means that this DMX uses codecs G.711 mu-Law at 20 milliseconds per packet, G.711 A-Law at 30 milliseconds per packet, and G.729.A at 2 frames per packet. You do not have to list the audio-codec values in ascending order
(so audio-codec =
4, 2, 1
and packet-size =
2, 30, 20
would have the same effect as the example just described), although it is probably easier to do so. If a packet-size value is invalid for the codec it corresponds to, it is ignored and that codec receives the default packet size. This option does not apply to the
MS-GSM codec.
progress-inband
Default Value: false
Valid Values: true, false, 0, 1
Changes Take Effect: Immediately
When a caller makes a call from an IP network to a PSTN via DMX and Cisco gateway, an in-band tone from the Cisco gateway is received. The Cisco gateway should be configured with the following options for the Inband tones feature to work properly:
•
Dial-peer POTS (for example, dial-peer voice 522 pots
)
•
Contain the following configuration options:
progress_ind alert enable 8 progress_ind progress enable 8
In some IOSes, these options may be present but hidden (that is, no help or autocomplete options are available, but explicitly-issued commands will work).
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Note:
All dial peers (POTS type) should be configured separately. That is, there is no global configuration command that will preconfigure all
POTS dial peers. For additional information regarding Cisco GW configuration, please refer to http://www.cisco.com.
qcif-mpi
Default Value:
1
Valid values:
0-32
Command-line Equivalent:
-qcif_mpi
Changes Take Effect: Immediately
Specifies the minimum picture interval in units of 1/29.97 for the encoding and/or decoding of Quarter Common Intermediate Format (QCIF) pictures for both H.261 and H.263 codecs. A value of
0
(zero) indicates no capability for
QCIF pictures. This option applies only if the
option has a value other than
0
(zero). The maximum value is
4
when using the H.261 codec. The maximum value is
32
for the H.263 codec.
silence-supression
Default Value:
NULL
Valid Values: true, false, 1, 0
Command-line Equivalent: None
Changes Take Effect: Immediately
If this option is either ( a
) not present, or ( b
) set to anything other than a valid value (including
NULL
), the endpoints of a call can negotiate whether or not to apply silence suppression. If this option has one of the valid values, silence suppression is ( true
or
1
) or is not ( false
or
0
) applied, without regard to anything the endpoints may do.
Note:
For audio-codec G.723 (value
3
), this option must be set to false
.
sip-port
Default Value:
5060
Valid Values: Any integer greater than
1024
Command-line Equivalent:
-sip_port
Changes Take Effect: After DMX is started
Defines the port where DMX waits for incoming connections from SIP endpoints.
sip-transport
Default Value: udp
Valid Values: tcp
or udp
Command-line Equivalent:
-sip_transport
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Changes Take Effect: Immediately
Defines the transport protocol (TCP or UDP) used to initiate outgoing sip calls.
Note:
Preferably, you should set this option identically for all DMXs in a multi-DMX configuration.
still-img-trans
Default Value: false
Valid Values: true, false
Command-line Equivalent:
-still_img_trans
Changes Take Effect: Immediately
Indicates capability for still images as specified in Annex D of the H.261 codec. This option applies only if the option
0
(zero).
tcp-port-range
Default Value: None
Valid Values: x - y
where x
is any valid port number greater than
1024
and y
is any valid port number greater than x
Command-line Equivalent:
-tcp_port_range
Changes Take Effect: Immediately
Defines a range of ports for H.245 communication with DMX (may be necessary when operating with a firewall).
trade-off-cap
Default Value: true
Valid Values: true, false
Command-line Equivalent:
-trade_off_cap
Changes Take Effect: Immediately
Applies to both H.261 and H.263 codecs. If true,
the encoder is able to vary its tradeoff between temporal and spatial resolution as commanded by the
remote terminal. This option only applies if the video-codec
than
0
(zero).
video-codec
Default Value:
0
Valid Values:
2
3
0
1
No video
H.261
H.263
H.264
Command-line Equivalent:
-video_codec
Changes Take Effect: Immediately
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Specifies the preferred video codec.
x-type
Default Value: dmx
Valid Value: dmx
Command-line Equivalent: None
Changes Take Effect: Immediately
Specifies to which component the x-config
options apply.
Gatekeeper Section
RAS Server
Options
This section must be called gatekeeper
. The name is case sensitive
.
accept-registrations
Default Value: false
Valid Values: true, false, on, off
Changes Take Effect: Immediately
Enables/disables RAS server (gatekeeper) functionality. In the off/false position, no communication sockets are opened, no RAS requests are accepted, and none of the other RAS server options have any effect.
accept-calls
Default Values: false
Valid Values: on, off, true, false
Changes Take Effect: Immediately
Setting this option to true/on
allows a gatekeeper DMX to also process calls; in other words, the server functions as one of its own clients. The advantage of this is that the gatekeeper can include itself in call distribution for load balancing.
If this option is set to off/false,
a gatekeeper DMX will still process calls if
(1) it has no client DMX registered, and (2) it is connected to a SIP Server.
discovery-port
Default Values:
0
Valid Values:
0
(zero) or
1718
only
Changes Take Effect: Immediately
Defines the port on which the gatekeeper DMX listens for autodiscovery messages. A value of
0
(zero) disables the auto-discovery function.
Autodiscovery handles the situation in which an endpoint wants to register as a
RAS client but is not configured with any gatekeeper's address. The endpoint broadcasts a message to all gatekeepers on the network. Gatekeepers that have autodiscovery enabled then respond to the message, informing the endpoint of their addresses.
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RAS Client
Options
main-port
Default Value:
1719
Valid Values: integer between
1024
and
32767
Changes Take Effect: Immediately
Defines the port on which the gatekeeper DMX listens for RAS requests.
backup-gk-host
Default Value: None
Valid Values: Any host name or IP address
Changes Take Effect: Immediately, if primary-gk-host
and primary-gk-port are specified
Specifies a backup gatekeeper host.
backup-gk-port
Default Value: none
Valid Values: Any integer greater than
1024
Changes Take Effect: Immediately, if primary-gk-host
and primary-gk-port are specified.
Specifies the port of the backup gatekeeper.
primary-gk-host
Default Value: None
Valid Values: Any host name or IP address
Changes Take Effect: Immediately, if primary-gk-host
and primary-gk-port are specified.
Specifies the gatekeeper with which this DMX registers.
primary-gk-port
Default Value:
1719
Valid Values: Any integer greater than
1024
Changes Take Effect: Immediately, if primary-gk-host
and primary-gk-port are specified.
Specifies the gatekeeper port to which this DMX connects.
register
Default Value: true
Valid Values: on/off, true/false
Changes Take Effect: Immediately, if primary-gk-host
and primary-gk-port are specified.
This is the main option that enables/disables RAS client functionality. In the off/false
position, the executable does not try to register and communicate with a gatekeeper, and none of the other RAS client options have any effect.
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Server/Client
Option
services
Default Value: None
Valid Values: Any comma-separated series of positive integers
Changes Take Effect: Immediately, if primary-gk-host
and primary-gk-port are specified.
A comma-separated list of E.164 (phone) numbers, to be used as a destination pattern during call distribution among client DMXs. Ordinarily, calls are distributed in round-robin fashion among all DMXs registered with a gatekeeper. But if a client DMX has a Service Number specified, all calls matching that number are sent to it only. Consider for example a configuration with one gatekeeper DMX and clients DMX-A, DMX-B, and DMX-C. DMX-
A has the services
option set to 3, while DMX-B and DMX-C have no services specified. Calls whose dialed number begins with a number other than
3 are distributed among all three client DMXs in a round-robin fashion. Calls whose dialed number begins with 3 are sent to DMX-A only.
registration-timeout
Default Value:
60
Valid Values: Any positive integer
Changes Take Effect: Immediately
Specifies an interval (in seconds) within which the client DMX must repeat a registration request to the gatekeeper DMX. If three of these timeout intervals pass without any registration request, the gatekeeper assumes that the client has failed. You must set this option for both the client and the gatekeeper.
Changes from 7.5 to 7.6
There are no option changes between release 7.5 and release 7.6.
Configuring for Use with the Management
Layer
The Management Layer, part of Genesys Framework 7.6, centralizes many functions (such as starting and stopping, status monitoring, alarm processing, and processing of log records) for all components of the Genesys Solution. It is required for configuring redundant systems (as described in the next section).
Detailed information about the Management Layer is available in the
Framework 7.6 documentation. To operate DMX along with the Management
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Chapter 13: DMX Deployment Configuring for Use with the Management Layer
Layer, configure DMX applications as previously described with the exceptions described in
.
Table 45: Configuring DMX for the Management Layer
Object
Host
DMX
Tab
General
Field
LCA Port
Values to Enter
Number of the port on which LCA (Local Control
Agent) is running
Connections Server: Add Add the name of the Message Server
Add the name of the T-Server
Start Info Working Directory Path name to the directory or folder holding the
DMX application
Command Line dmx.exe
Command-line
Arguments
-host <hostname> -port <portnumber> -app
<appname>, where <hostname> and
<portnumber> are the host and port of
Configuration Server and <appname> is the name of the DMX application
Timeout (Startup,
Shutdown)
120 recommended
Redundancy Type Unspecified
Auto-Restart Put a check mark in the box if you want the
Management Layer to automatically restart the application.
VOIP Gateways
For every gateway that you plan to make a part of your interaction management solution, you need to have the following information available:
•
Type of gateway. It usually corresponds to the gateway vendor, brand name, and model number.
Note:
A gateway must support either SIP RFC 3261 or H.323 (v1 or v2).
•
Version of the gateway software.
•
Type of interface to the PBX/PSTN; for example T1 (ISDN PRI), FXO.
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Voice Terminal Endpoints
Voice terminal endpoints enable the agent to receive voice calls. Three types of voice terminals can be used.
shows the information you should have available for each type.
Table 46: Voice Terminals and Required Information
Voice Terminal Type Required Information
Multimedia PC with sound card, headphones, microphone
Traditional analog or digital phone connected to a VoIP gateway (via a traditional PBX for digital phone)
IP address and host name of PC
• Extension numbers corresponding to
PBX and/or gateway dialing plan
• IP address of the gateway or RAS
IP endpoint (SIP and H.323 compliant) IP address of endpoint
SIP Endpoints
H.323 Endpoints
Gateways
Cisco ATA 186 (See note below)
Cisco IP Phone CP-7960G
PingTel
Xten X-Lite
Polycom IP500
Open Phone
Cisco ATA 186 (See note below)
AudioCodes IPM260 (PCI-based)
Cisco 5300
Cisco 3725
RadVision VoIP Gateway
LG-1100
LG-2100
Vega-50
Vega-1000
SIP IP Phone
Polycom IP500
Installing DMX
Configuration Wizards facilitate component deployment. The DMX configuration and installation involves many steps, and Genesys strongly recommends that you set up DMX using the wizard rather than manually. The
DMX Wizard guides you through a series of steps and options to customize your deployment of DMX.
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Chapter 13: DMX Deployment Installing DMX
Note:
You should not attempt to run DMX on the same host as SIP Server or any other SIP clients.
Installing using the Wizard
To use the Wizard to install DMX, install the Wizard from the CD, run the
Genesys Wizard Manager, and follow the instructions.
Installing Manually
Consider the following when installing DMX:
•
Only one DMX can be active on a given physical server.
•
No other SIP or H.323 applications/components can be located on the same machine as DMX.
Note:
To override this default behavior, see the description of the DMX h225port
and
SIP-port
options.
On the product CD, the media_layer directory includes a subdirectory containing installation files for the DMX component. It contains subdirectories named windows
and solaris,
which contain the appropriate installation files for those operating systems.
UNIX
For each component’s directory, open the aix, linux,
or
solaris subdirectory, then open and read the file
README_FIRST.
It contains complete instructions on expanding the compressed file, running the shell script
<COMPONENT>-INSTALL.SH,
and completing the installation.
Note: Do not include spaces in the destination directory name when using a
UNIX operating system.
You are now ready to start DMX. See “Starting and Stopping DMX” on page 392
.
Windows 2000/2003
Open the subdirectory windows,
then double-click
Setup.exe.
Follow the
InstallShield instructions.
InstallShield creates a batch file in the folder of the component it installs. After installation, open the batch file and ensure that it includes the parameter
-app
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<appname>,
where
<appname>
is the name of the application object you created in Configuration Manager.
Note:
DMX is installed as a Windows 2000/2003 Service by default.
You are now ready to start DMX. See “Starting and Stopping DMX” on page 392
.
Starting and Stopping DMX
Be sure that DB Server, Configuration Server, and License Manager are running before starting DMX.
Procedure:
Starting DMX using Solution Control Interface
Purpose: To start DMX using Solution Control Interface (SCI).
Start of procedure
1. Start SCI either by clicking
Sci.exe
or selecting
Solution Control
Interface
from the
Start
menu.
2. Go to the
Applications
view.
3. Select the DMX application in the
List
pane.
4. Click
Start
or select
Start
from the shortcut menu.
The application’s status changes from
Stopped
to
Started
.
End of procedure
Procedure:
Stopping DMX using Solution Control Interface
Purpose: To stop DMX using SCI.
Start of procedure
1. Select the DMX component in the
Applications
folder.
2. Select
Stop
from the
Action
menu.
The application’s status changes from
Started
to
Stopped
.
End of procedure
392 Framework 7.6
Chapter
14
DMX Reference
This chapter provides reference information for DMX. It includes:
VOIP and Video-Over-IP Support, page 399
z z
DMX Log Events
DMX displays some log events from the common.lms
file. However, many other log events have their text hard coded within DMX rather than coming from the common.lms
file.
Table 47 lists the hard-coded events. In the
Message Text column,
<number>
and
<text>
stand for attributes that are filled in when the actual message is generated. The
Attributes
column lists the names of the attributes (if any) in the order that they appear in the message text. Thus, for example,
ID#40009
could also be represented as:
•
Option
<section name>:<option name>
was set to
<option value>
from the command line.
Table 47: DMX Log Events
ID
40004
40009
40011
40012
Level
STANDARD
Message Text
Total number of legs: <number> (v. <string>)
Attributes
Number of legs, DMX version
Section, name, value STANDARD Option <text>:<text> was set to <text> from the command line
STANDARD (p:<number>) LCA port was set from the command line
STANDARD DMX general options: <text>
Port
String of options
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Table 47: DMX Log Events (Continued)
ID
40015
40016
40018
40019
40021
40023
40039
40040
40041
40044
40045
40048
40049
40050
40052
40054
40055
40057
40060
Level Message Text
STANDARD DMX audio settings: <text>
STANDARD (l:<number>) (c:<number>) Received unsupported message (H225)
STANDARD Wrong protocol decriminator <hexadecimal number> (e:) (Q931)
STANDARD Wrong CRV length <number>, message length
<number> (e:) (Q931)
STANDARD ASN1 decode error (e:<text>), decode
<number>, PDU <number> (H225)
Attributes
String of options
STANDARD
STANDARD
STANDARD
STANDARD
DMX video settings: <text>
Version: <text>
Compilation Date: <text> <text>
Command line: <text>
STANDARD (c:<number>) Connection failed (e:) (H225)
STANDARD (l:<number>) (c:<number>) Function <text> failed (e:<number>) (H245)
STANDARD (c:<number>) Connection failed (e:) (H245)
STANDARD (l:<number>) Leg not found by ID (e:)
STANDARD Leg creation error (e:)
String of options
Version number
Date, time
Command-line string
Hostname STANDARD Cannot resolve IP address by host name <text>
(e:)
STANDARD (p:<number>) Port open for client H225s
(H225)<text>
Port number
STANDARD (p:<number>) Port closed for client H225s
(H225)<text>
Port number
STANDARD (p:<number>) Cannot open port for clients H225
(H225) (e:)
Port number
STANDARD (l:<number>) (c:<number>) Function <text> failed (e:<number>) (H225)
Leg ID, socket
Socket
Socket
Socket
Leg ID
Socket
Protocol discriminator
Length
Error message, decode,
PDU
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Table 47: DMX Log Events (Continued)
ID
40061
40064
40065
40083
40084
40085
40087
40088
40089
40100
40102
40105
40108
40109
40110
40111
40112
40113
Level Message Text
STANDARD ASN1 encode error (e:<text>), encode
<number> (H225)
STANDARD ASN1 decode error (e:<text>), decode
<number>, PDU <number> (H225)
STANDARD ASN1 encode error (e:<text>), encode
<number> (H225)
STANDARD (CSI:) Can’t connect to Configuration Server on
<text>:<text> <text> (e:)
STANDARD (CSI:) CfgLib error: <text> (e:)
Attributes
Error message, decode
Error message, decode,
PDU
Error message, decode
Host, port, error
Error description
STANDARD (CSI:) No Message Server for application <text>
(e:)
Application name
STANDARD (CSR:) Error on reconnection <text> (e:)
STANDARD (CSR:) DMX-server name <<text>> on reconnection is not as specified <<text>>(e:)
Explanation
Old name, new name
STANDARD (CSR:) DMX-server type <number> is not as should be <number> (e:)
STANDARD (CSU:) Message Server <text> on <text>:<text> added to DMX
STANDARD (CSE:) Delta object=NULL (e:)
Old type, new type
Name, host, port
STANDARD (CSU:) Message Server <text> removed
STANDARD (CSS:) Primary Configuration Server on
<text>:<text>
STANDARD (CSS:) Backup Configuration Server on
<text>:<text>
STANDARD (CSS:) No backup Configuration Server
Host, port
Host, port
STANDARD (CSS:) Message Server <text> Host's name was changed to <text>
STANDARD (CSS:) Configuration Server Disconnected
STANDARD (CSE:) Error: Can’t find <text> with dbid
<number> (e:)
Name, new host name
Type of object, database ID
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Table 47: DMX Log Events (Continued)
ID
40114
40115
40118
40128
40129
40132
40133
40134
40135
40137
40139
40140
40141
40142
40144
40145
40146
40147
Level Message Text
STANDARD (CSE:lib) List is not empty but head is null (e:)
STANDARD (CSU:) Can’t reconnect because do not have host/port info (e:)
STANDARD (RSO:) ‘services’ for client <text> was set to
<text>
Attributes
Client name, services
STANDARD (e:)(RSD:) Selected object not found
STANDARD RAS Discovery port closed
STANDARD Previous try was rejected, try in <number>
STANDARD (RSS:) RAS ASN1 decode: Error decode =
<number>,pdu=<number> (e:)
STANDARD (RSS:) Registration of <text> confirmed, aliases
<text> created
STANDARD (RSS:) Registration Rejected due to duplicated aliases <text>
STANDARD (RSS:) Registration confirmed, Reregistration without aliases, aliases cleared
STANDARD (RSC:) Impossible to prepare Registration message (e:)
STANDARD (RSC:) Impossible to prepare UnRegistration message (e:)
STANDARD (RSC:) Opened Listening Socket <number> for
RAS client on Port: <number>
STANDARD (RSS:) Opened Listening Socket for RAS
Discovery on Port: <number>
STANDARD (RSS:) Opened Listening Socket for RAS
Request and Status on Port: <number>
STANDARD (RSS:) Listening Socket for RAS Request and
Status Closed
STANDARD (RSC:) Stop working as Client
STANDARD (CSR:) Configuration Server reconnected on
<text>:<text>
Seconds
Decode, PDU
New client, aliases
Aliases
Socket, port
Port
Port
Host, port
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Table 47: DMX Log Events (Continued)
ID
40148
40149
40150
40151
40152
40153
40154
40155
40156
40157
40158
40159
40160
40161
40162
40163
Level Message Text
STANDARD (CSR:)Error,kvlist empty (e:)
Attributes
STANDARD (CSR:) Try to reconnect as <text> to
<text>:<text>
STANDARD (l:<number>) LCSE Error: Timer T103 fired (e:)
(H245)
Leg ID
STANDARD (l:<number>) LCSE Error: Improper state
(e:<number>) (H245)
Leg ID, error code
STANDARD (l:<number>) CESE Error: Timer T101 fired (e:)
(H245)
Leg ID
STANDARD (l:<number>) LCSE Error: Timer T106 fired (e:)
(H245)
Leg ID
STANDARD (l:<number>) LCSE Error: Improper state
(e:<number>) (H245)
Leg ID, error code
STANDARD (l: <number>) DMX attempts to call itself (e:)
STANDARD (CSI:) LcaPort <text> LCA port
Leg ID
Port number
STANDARD (CSI:) DMX application has empty set of options
STANDARD (CSI:)DMX application <text> defined as third party application configured incorrectly:<text>
(e:)
DMX name, description of problem
Error type, object type, object property
STANDARD (CSR:)Bad result of
ConfGetConfServerDBID(..) error_type
<number>, object-type
<number>,object_property <number> (e:)
STANDARD (CSR:)Bad result of
ConfGetConfServerBackupDBID(..) error_type
<number>, object-type
<number>,object_property <number> (e:)
Error type, object type, object property
STANDARD (CSU:)We have more then one Message Server attached, <text> and <text> (e:)
STANDARD (CSU:) Empty set of connections of DMX,
Message Server not searched
STANDARD (CSU:) Message Server <text> changed
SIP Server—Deployment Guide 397
Chapter 14: DMX Reference DMX Log Events
Table 47: DMX Log Events (Continued)
ID
40164
40165
40166
40167
40168
40169
40170
40175
40175
40179
40182
40183
40184
40190
40195
40197
40201
50001
Level Message Text
STANDARD (CSU:)ConfGetErrorInfo can’t get details of error (e:)
STANDARD (RSO:) ‘gatekeeper’ Option <text> was set to
<text> Option name, value
STANDARD (RSD:) Add new client <text>, services: <text>; current number of clients <number>
Attributes
Name of client, services list, number of clients
STANDARD (RSS:) RAS ReRegistration Confirmed, aliases set was empty, added new one(s)
STANDARD (RSS:)RAS Registration, Null call address,
EndPoint ID unknown (e:)
STANDARD (RSS:) RAS Registration, response from client
<text> not received, unregistered by force
STANDARD (RSS:) Agent <text> unregistered
STANDARD (RSC:)<text>, File:<text>,Line:<number> (e:)
Name of client
Name of client
Description of situation, file, line
STANDARD (RSC:) Send Unregistration message
STANDARD (ts:<text>) TS_EXCEPTION occurred (e:) (nco)
STANDARD List of local ip addresses: <text> Addresses
STANDARD (l:<number>) No more available ports. Increase tcpport-range value (e:)
STANDARD (l:<number>) No common audio codec found
(e:)
Leg ID
Leg ID
STANDARD Option <text>:<text> is wrong. Using default value <text>
Option name, option value, default option value
Port number STANDARD (p:<number>) Port closed for SIP clients
STANDARD (l:<number>) Sent SIP ‘<text’
STANDARD (l:<number>) Received SIP ‘<text>’
TRACE
LegID, SIP message
LegID, SIP message
(l:<number>) Leg created, CRV = <hexadecimal number>
Socket on H225, leg ID
398 Framework 7.6
Chapter 14: DMX Reference
Table 47: DMX Log Events (Continued)
ID
50002
50034
50038
50039
50040
50054
50056
Level
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
TRACE
Message Text
(l:<number>) Leg destroyed
(RSD:) No Registered Clients
(RSD:) Total num of clients =<number>
(RSD:) Add service <text> for server itself
(RSD:) Stop server as client for itself
(RSD:)<<text>>: Alias <text> was resolved
<text>:<number>
(RSD:) Admission request from <text>
Confirmed: <text>
VOIP and Video-Over-IP Support
Attributes
Leg ID
Number of clients
Service name
Sender, destination
VOIP and Video-Over-IP Support
The DMX specifications in Table 48 include information for working with
VOIP and/or Video over IP.
Table 48: DMX Specifications:
VOIP and Video-over-IP
Supported Features
VoIP protocols supported:
Possible Values
H.323 set (H.225.0, H.245)
SIP rfc3261 (TCP and UDP)
SIP SDP rfc2327
Codecs supported: voice: video:
Silence suppression:
DTMF type support:
G.711 (mLaw, ALaw), G.723.1,
G.729.A, GSM, MS-GSM
H.261, H.263, H.264; CIF and QCIF
Supported in G.711, G.729.A and GSM
H.245 user input, H.245 signal, rfc 2833
(RTP-NTE)
SIP Server—Deployment Guide 399
Chapter 14: DMX Reference VOIP and Video-Over-IP Support
Table 48: DMX Specifications:
VOIP and Video-over-IP
(Continued)
Supported Features
Additional protocol support:
Video still image support:
Firewall configuration support:
Possible Values
H.225.0 Facility call Forward (server)
H.450.2 FACILITY Single-step Transfer
SIP REFER (Single-step Transfer)
H.225.0 RAS Gatekeeper (server/client)
Supported in H.261
Supported in modes: standalone firewall; using DMX as proxy in DMZ
400 Framework 7.6
Appendix
Session Initiation Protocol
(SIP) Overview
z z
This chapter provides basic information about the SIP and H.323 protocols and contains the following sections:
SIP Overview
Session Initiation Protocol (SIP) was originally designed in 1996 as an
Application Layer control (signalling) protocol for creating, modifying, and
terminating media sessions. See Figure 41 on page 402
for more information about layers in the TCP/IP model.
IETF
The Internet Engineering Task Force (IETF) is a volunteer group that works with the World-wide Web Consortium (W3C) to develop and promote Internet standards such as Transmission Control Protocol/Internet Protocol (TCP/IP).
They also created documents known as “Request for Comments” (RFC) that discuss new research, innovations, and methodologies applicable to Internet technologies. The IETF is organized into a large number of working groups, one of which is SIP. There are a large number of SIP-related RFCs.
TCP/IP Model
SIP can work over any component in the Transport layer such as TCP or UDP.
SIP works with several other protocols and is only involved in the signalling portion of a session. SIP contains Session Description Protocol (SDP) information that describes the media content of the session, and is a packet in
Realtime Transport Protocol (RTP). RTP is the carrier for the actual voice or video content itself. Normally, SIP clients use port 5060 over TCP or UDP protocols to connect to SIP Servers and other SIP endpoints.
SIP Server—Deployment Guide 401
Appendix: Session Initiation Protocol (SIP) Overview SIP Overview
Note:
Hardware SIP endpoints share characteristics with traditional telephones, but use SIP and RTP to communicate.
Software SIP endpoints are also very common in many Instant
Messaging applications.
contains information about layers in the TCP/IP model.
Layer 5: Application - SIP
Layer 4: Transport – TCP, UDP
Layer 3: Network – IPv4, IPv6
Layer 2: Data Link – Ethernet, 802.11
Layer 1: Physical – Ethernet network card, ISDN, Modems
Figure 41: Five-layer TCP/IP Model
SIP Functionality
SIP provides signalling and call setup protocols for IP communication that are similar to those found in the Public Switched Telephone Network (PSTN). SIP also enables features such as Proxy Servers and User Agents. These features are similar to normal telephone operations. Although SIP and PSTN networks are different, the agent does not notice a difference. SIP can also enable many of the advanced call processing features that are found in a complex system such as Signalling System 7 (SS7). However, SIP is a peer-to-peer protocol, and requires only a core network with hardware or software endpoints. With
SIP, the advanced call processing features are implemented in the endpoints instead of being implemented in a SS7 network.
402 Framework 7.6
Appendix: Session Initiation Protocol (SIP) Overview SIP Overview
Proxy and Registrar Servers
SIP requires proxy and registrar network components to work. Although two
SIP endpoints can communicate without a SIP Server (peer-to-peer), this is not very practical so traditional switch hardware or software switch functionality is needed to act as registrar and proxy.
A registrar is a server that accepts
REGISTER
requests from endpoints and allows agents to specify their current location for the proxy servers.
A proxy server performs for the following functionalities:
•
Route media requests to an agent's location
•
Authenticate and authorize agents for media services
•
Implement call-routing strategies
•
Provide media features to agents
SIP Messaging
SIP is similar to HTTP and provides request-response text-based messaging, so it is easy to understand because it shares many HTTP status codes.
SIP is able to process the following messages:
Request Methods
Table 49: SIP Request Methods
Request Method Description
INVITE
ACK
BYE
CANCEL
OPTIONS
Indicates a client is being invited to participate in a call session.
Confirms that the client has received a final response to an
INVITE
request.
Terminates a call and is sent by either the sender or the receiver.
Cancels any pending searches but does not terminate a call that has already been accepted.
Queries the capabilities of servers.
REGISTER
INFO
Registers the address listed in the To header field with a
SIP server.
Sends mid-session information that does not modify the session state.
SIP Server—Deployment Guide 403
Appendix: Session Initiation Protocol (SIP) Overview SIP Overview
Table 49: SIP Request Methods (Continued)
Request Method Description
REFER
Ask recipient to issue SIP request, such as a call transfer.
MESSAGE
UPDATE
Transports instant messages using SIP.
Modifies the state of a session without changing the state of the dialog.
Provisional Responses
PRACK
Provisional acknowledgement
Notifications
SUBSCRIBE
Subscribes for an Event of Notification from the Notifier.
NOTIFY
Notifies the subscriber of a new Event.
Publishing Event Changes
PUBLISH
Publishes an event to the server.
Responses
1xx Informational Responses
100 Trying
180 Ringing
181 Call Is Being Forwarded
182 Queued
183 Session Progress
2xx Successful Responses
200 OK
202 accepted
3xx Redirection Responses
300 Multiple Choices
301 Moved Permanently
302 Moved Temporarily
305 Use Proxy
380 Alternative Service
404 Framework 7.6
Appendix: Session Initiation Protocol (SIP) Overview
4xx Client Failure Responses
400 Bad Request
401 Unauthorized
402 Payment Required (Reserved for future use)
403 Forbidden
404 Not Found: User not found
405 Method Not Allowed
406 Not Acceptable
407 Proxy Authentication Required
408 Request Timeout
410 Gone
413 Request Entity Too Large
414 Request-URI Too Long
415 Unsupported Media Type
416 Unsupported URI Scheme
420 Bad Extension
421 Extension Required
423 Interval Too Brief
479 Regretfully, we were not able to process the URI
480 Temporarily Unavailable
481 Call/Transaction Does Not Exist
482 Loop Detected
483 Too Many Hops
484 Address Incomplete
485 Ambiguous
486 Busy Here
487 Request Terminated
488 Not Acceptable Here
489 Bad Event
491 Request Pending
493 Undecipherable
494 Security Agreement Required
5xx Server Failure Responses
500 Server Internal Error
501 Not Implemented
SIP Server—Deployment Guide
SIP Overview
405
Appendix: Session Initiation Protocol (SIP) Overview
502 Bad Gateway
503 Service Unavailable
504 Server Time-out
505 Version Not Supported
513 Message Too Large
6xx Global Failure Responses
600 Busy Everywhere
603 Decline
604 Does Not Exist Anywhere
606 Not Acceptable
H.323 Overview
H.323 Overview
H.323 is a part of a group of protocols that provides for communications using
Integrated Services Digital Network (ISDN), Public Switched Telephone
Network (PSTN) or Signaling System 7 (SS7). H.323 was developed to transport media over Local Area Networks (LANs) but it has since become a means for implementing Voice Over Internet Protocol (VOIP) networks.
Note:
In Genesys, the DMX component provides H.323 protocol conversion for SIP Server.
ITU-T and Call Models
H.323 is a protocol that the ITU Telecommunication Standardization Sector
(ITU-T) developed to provide media communication sessions for a packet network. H.323 was the first VoIP protocol to set standards for the basic call model and additional services that were needed to complement existing PSTN service expectations. The H.323 protocol is integrated easily into existing
ISDN-based telephone switches. As a result, an IP switch is a H.323
Gatekeeper as well as a provider of additional calling services.
H.xxx Protocols
The H.323 protocol also uses other ITU-T protocols such as:
H.225: Describes call signalling, media, stream packets, media stream synchronization, and control messaging formats.
H.235: Describes forms of security.
H.239: Describes how to use two media streams when having a video conference call.
406 Framework 7.6
Appendix: Session Initiation Protocol (SIP) Overview H.323 Overview
H.245: Describes the messages and procedures used for manipulating media channels, capability exchange, control and indications.
H.261, H.263, H.264: Describe video encoding formats.
H.450: Describes any additional calling services.
H.460: Describes firewall implementation.
SIP Server—Deployment Guide 407
Appendix: Session Initiation Protocol (SIP) Overview H.323 Overview
408 Framework 7.6
Index
Numerics
1pcc transfer
. . . . . . . . . . . . . . . . . 123
3pcc transfer
. . . . . . . . . . . . . . . . . 123
3rd-party known limitations
A accept-calls gatekeeper section
accept-dn-type configuration option
accept-registrations gatekeeper section
Access Code configuration
. . . . . . . . . . . . . . . . 310
defined
. . . . . . . . . . . . . . . . . 45
acw-in-idle-force-ready configuration option
ADDP
. . . . . . . . . . . . . . . . . . . . . 59
addp-remote-timeout common configuration option
addp-timeout common configuration option
addp-trace common configuration option
Advanced Disconnect Detection Protocol
after-routing-timeout configuration option
Agent agent no-answer supervision
agent-greeting
. . . . . . . . . . . . . . . 228
agent-no-answer-action
agent-no-answer-overflow
agent-no-answer-timeout
agent-strict-id configuration option
Agent Login objects
agent reservation defined
. . . . . . . . . . . . . . . . . . . 265
agent-greeting configuration option
agent-group configuration option
emulated agent options
agent-no-answer-action configuration option
agent-no-answer-overflow configuration option
agent-no-answer-timeout configuration option
Agent-Reservation section all common configuration options
–
common log option
am-detected configuration option
amd-mode
. . . . . . . . . . . . . . . . . . 378
ANI
. . . . . . . . . . . . . . . . . . . . . . 276
ani-distribution common configuration option
app command line parameter
Application objects multi-site operation
application server mode
AssistMode
. . . . . . . . . . . . . . . . . 187
associating ACD Queue with Routing Point
Asterisk-based Voice Mail agent group registration
agent login registration
extension registration
feature configuration
Genesys voice mail configuration adapter
message waiting indicator
overview
. . . . . . . . . . . . . . . . . . 98
audio-codec x-config section
. . . . . . . . . . . . . . 378
audio-codecs configuration option
SIP Server—Deployment Guide 409
Index authenticate-requests configuration option
auto-redirect-enabled configuration option
B background-processing common configuration option
background-timeout common configuration option
Back-to-Back User Agent
backup servers
. . . . . . . . . . . . . . . . 51
backup-gk-host gatekeeper section
backup-gk-port gatekeeper section
backup-mode configuration option
backup-sync configuration section
Backup-Synchronization section common configuration options
–
backwds-compat-acw-behavior configuration option
bandwidth requirements
bit-rate x-config section
. . . . . . . . . . . . . . 378
buffering common log option
Busy Lamp Field overview
. . . . . . . . . . . . . . . . . . 101
busy-tone configuration option
C
Call Recording emergency
. . . . . . . . . . . . . . . . . 104
feature configuration
overview
. . . . . . . . . . . . . . . . . . 102
Call Supervision assistance request
configuration options
feature configuration
feature limitations
hiding supervisor presence
intrusion
. . . . . . . . . . . . . . . . . . 107
monitoring session
multi-site supervision
scopes
. . . . . . . . . . . . . . . . . . . 105
supervisor auto-release
types
. . . . . . . . . . . . . . . . . . . . 106
call-address call section
. . . . . . . . . . . . . . . . . 376
Call-Cleanup section common configuration options
–
call-max-outstanding configuration option
call-port call section
. . . . . . . . . . . . . . . . 377
call-protocol call section
. . . . . . . . . . . . . . . . 377
call-rq-gap configuration option
Calls Outside the Premise find-trunk-by-location
Calls outside the premise overview
. . . . . . . . . . . . . . . . . 101
cancel-monitor-on-disconnect configuration option
capacity configuration option
capacity-group configuration option
cast-type common configuration option
CDN
. . . . . . . . . . . . . . . . . . . . . 281
changes
SIP configuration options
changes from 7.5 to 7.6
common configuration options
configuration options
chapter summaries defining
. . . . . . . . . . . . . . . . . . . 14
check-point common log option
check-tenant-profile common configuration option
cif-mpi x-config section
. . . . . . . . . . . . . . 379
Class of Service
–
dialing rule format
feature configuration
outbound dialing rules
ring-through rules
transformation
. . . . . . . . . . . . . . . 131
cleanup-idle-tout common configuration option
clid-withheld-name configuration option
Code property
. . . . . . . . . . . . . .310
cof-ci-defer-create common configuration option
cof-ci-defer-delete common configuration option
cof-ci-req-tout common configuration option
cof-ci-wait-all common configuration option
410 Framework 7.6
Index cof-feature common configuration option
cof-rci-tout common configuration option
collect-tone configuration option
command line parameters
app
. . . . . . . . . . . . . . . . . . . . . 69
host
. . . . . . . . . . . . . . . . . . . . . 69
l
. . . . . . . . . . . . . . . . . . . . . . . 70
lmspath
. . . . . . . . . . . . . . . . . . . 70
nco X/Y
. . . . . . . . . . . . . . . . . . . 70
port
. . . . . . . . . . . . . . . . . . . . . 69
V
. . . . . . . . . . . . . . . . . . . . . . . 70
commenting on this document
common configuration options
–
addp-remote-timeout
addp-timeout
. . . . . . . . . . . . . . . 363
addp-trace
. . . . . . . . . . . . . . . . . 363
Agent-Reservation section
–
ani-distribution
. . . . . . . . . . . . . . . 344
background-processing
background-timeout
Backup-Synchronization section
–
Call-Cleanup section
–
cast-type
. . . . . . . . . . . . . . . . . . 355
changes from 7.5 to 7.6
check-tenant-profile
cleanup-idle-tout
. . . . . . . . . . . . . . 364
cof-ci-defer-create
cof-ci-defer-delete
cof-ci-req-tout
. . . . . . . . . . . . . . . 359
cof-ci-wait-all
. . . . . . . . . . . . . . . 360
cof-feature
. . . . . . . . . . . . . . . . . 360
cof-rci-tout
. . . . . . . . . . . . . . . . . 360
common section
. . . . . . . . . . . . . . 340
compatibility-port
consult-user-data
customer-id
. . . . . . . . . . . . . . . . 346
default-dn
. . . . . . . . . . . . . . . . . 355
direct-digits-key
. . . . . . . . . . . . . . 355
dn-for-unexpected-calls
enable-async-dns
event-propagation
inbound-translator-<n>
License section
–
local-node-id
. . . . . . . . . . . . . . . . 360
log section
. . . . . . . . . . . . . . .322
–
log-extended section
–
log-filter section
. . . . . . . . . . . . . . 338
log-filter-data section
log-trace-flags
. . . . . . . . . . . . . . . 347
management-port
mandatory
. . . . . . . . . . . . . . . . . 322
match-call-once
. . . . . . . . . . . . . . 353
merged-user-data
Multi-Site Support section
–
network-request-timeout
notify-idle-tout
. . . . . . . . . . . . . . . 364
num-of-licenses
. . . . . . . . . . . . . . 349
num-sdn-licenses
periodic-check-tout
protocol
. . . . . . . . . . . . . . . . . . 363
rebind-delay
. . . . . . . . . . . . . . . . 340
reconnect-tout
. . . . . . . . . . . . . . . 353
register-attempts
register-tout
. . . . . . . . . . . . . . . . 356
reject-subsequent-request
report-connid-changes
request-collection-time
request-tout
. . . . . . . . . . . . . . . . 357
reservation-time
. . . . . . . . . . . . . . 352
resource-allocation-mode
resource-load-maximum
route-dn
. . . . . . . . . . . . . . . . . . 358
rule-<n>
. . . . . . . . . . . . . . . . . . 362
Security section
. . . . . . . . . . . . . . 365
server-id
. . . . . . . . . . . . . . . . . . 348
setting
. . . . . . . . . . . . . . . . . 321
sync-reconnect-tout
tcs-queue
. . . . . . . . . . . . . . . . . 358
tcs-use
. . . . . . . . . . . . . . . . . . 359
timeout
. . . . . . . . . . . . . . . . . . 358
timeout value format
–
Translation Rules section
T-Server section
–
use-data-from
. . . . . . . . . . . . . . . 354
use-implicit-access-numbers
user-data-limit
. . . . . . . . . . . . . . . 348
common log options
–
all
. . . . . . . . . . . . . . . . . . . . . 329
buffering
. . . . . . . . . . . . . . . . . . 323
check-point
. . . . . . . . . . . . . . . . 326
compatible-output-priority
debug
. . . . . . . . . . . . . . . . . . . 331
default-filter-type
expire
. . . . . . . . . . . . . . . . . . . 323
interaction
. . . . . . . . . . . . . . . . . 330
keep-startup-file
. . . . . . . . . . . . . . 324
<key name>
. . . . . . . . . . . . . . . . 339
level-reassign-<eventID>
level-reassign-disable
log section
. . . . . . . . . . . . . . 322
–
log-extended section
–
log-filter section
. . . . . . . . . . . . . . 338
log-filter-data section
mandatory options
memory
. . . . . . . . . . . . . . . . . . 327
memory-storage-size
message_format
messagefile
. . . . . . . . . . . . . . . . 324
print-attributes
. . . . . . . . . . . . . . . 326
SIP Server—Deployment Guide 411
Index segment
. . . . . . . . . . . . . . . . . . 323
setting
. . . . . . . . . . . . . . . . . . . 321
spool
. . . . . . . . . . . . . . . . . . . . 327
standard
. . . . . . . . . . . . . . . . . . 329
time_convert
. . . . . . . . . . . . . . . . 325
time_format
. . . . . . . . . . . . . . . . 326
trace
. . . . . . . . . . . . . . . . . . . . 330
verbose
. . . . . . . . . . . . . . . . . . 322
x-conn-debug-all
. . . . . . . . . . . . . . 335
x-conn-debug-api
x-conn-debug-dns
x-conn-debug-open
x-conn-debug-security
x-conn-debug-select
x-conn-debug-timers
x-conn-debug-write
common options common log options
–
common section
. . . . . . . . . . . . . . 340
mandatory options
common section common options
. . . . . . . . . . . . . . 340
compatibility-port common configuration option
compatible-output-priority common log option
Configuration Manager configuring T-Server
configuration options accept-dn-type
. . . . . . . . . . . . . . . 248
acw-in-idle-force-ready
after-routing-timeout
agent-greeting
. . . . . . . . . . . . . . . 228
agent-group
. . . . . . . . . . . . . . . . 198
agent-no-answer-action
agent-no-answer-overflow
agent-no-answer-timeout
agent-strict-id
. . . . . . . . . . . . . . . 200
am-detected
. . . . . . . . . . . . . . . . 200
audio-codecs
. . . . . . . . . . . . . . . 201
authenticate-requests
auto-redirect-enabled
backup-mode
. . . . . . . . . . . . . . . 248
backwds-compat-acw-behavior
busy-tone
. . . . . . . . . . . . . . . . . 202
call-max-outstanding
call-rq-gap
. . . . . . . . . . . . . . . . . 202
cancel-monitor-on-disconnect
capacity
. . . . . . . . . . . . . . . . . . 229
capacity-group
. . . . . . . . . . . . . . . 229
changes from 7.5 to 7.6
clid-withheld-name
collect-tone
. . . . . . . . . . . . . . . . 203
common log options
–
common options
–
contact
. . . . . . . . . . . . . . . . . . . 230
412 correct-rqid
. . . . . . . . . . . . . . . . 248
cpd-info-timeout
. . . . . . . . . . . . . . 203
cpn
. . . . . . . . . . . . . . . . . . . . 231
customer-greeting
default-dn (application level)
default-dn (switch/DN level)
default-dn-type
. . . . . . . . . . . . . . 248
default-monitor-mode
default-monitor-scope
default-music
. . . . . . . . . . . . . . . 204
default-network-call-id-matching
default-video-file
delay-between-refresh-on-switchover
delay-to-start-refresh-on-switchover
divert-on-ringing
. . . . . . . . . . . . . . 205
dn-del-mode
. . . . . . . . . . . . . . . 248
dtmf-payload
. . . . . . . . . . . . . . . 205
dual-dialog-enabled
emergency-recording-cleanup-enabled
emergency-recording-filename
emulated-login-state
emulate-login
. . . . . . . . . . . . . . . 248
enable-agentlogin-presence
enable-ims
. . . . . . . . . . . . . . . . 248
enforce-external-domains
event-ringing-on-100trying
expire-call-tout
. . . . . . . . . . . . . . 248
extensions-<n>
external-registrar
extn-no-answer-overflow
ext-no-answer-action
fast-busy-tone
. . . . . . . . . . . . . . . 209
fax-detected
. . . . . . . . . . . . . . . . 209
find-trunk-by-location
forced-notready
. . . . . . . . . . . . . . 210
force-register
. . . . . . . . . . . . . . . 232
geo-location
. . . . . . . . . . . . . . . . 233
handle-vsp
. . . . . . . . . . . . . . . . 247
ims-default-icid-prefix
ims-default-icid-suffix
ims-default-orig-ioi
inbound-bsns-calls
inherit-bsns-type
init-dnis-by-ruri
. . . . . . . . . . . . . . 248
internal-bsns-calls
internal-registrar-domains
internal-registrar-enabled
internal-registrar-persistent
intrusion-enabled
kpl-interval
. . . . . . . . . . . . . . . . 248
kpl-loss-rate
. . . . . . . . . . . . . . . . 248
kpl-tolerance
. . . . . . . . . . . . . . . 248
legal-guard-time
. . . . . . . . . . . . . . 212
logout-on-disconnect
make-call-alert-info
make-call-rfc3725-flow
Framework 7.6
Index mandatory common
. . . . . . . . . . . . . . . . 322
max-legs-per-sm
max-pred-req-delay
monitor-internal-calls
music-in-conference-file
music-in-queue-file
mwi-agent-enable
mwi-domain
. . . . . . . . . . . . . . . . 214
mwi-extension-enable
mwi-group-enable
mwi-host
. . . . . . . . . . . . . . . . . . 215
mwi-mode
. . . . . . . . . . . . . . . . . 215
mwi-port
. . . . . . . . . . . . . . . . . . 215
nas-inidication
. . . . . . . . . . . . . . . 248
nas-private
. . . . . . . . . . . . . . . . . 216
no-answer-action
no-answer-overflow
no-answer-timeout
observing-routing-point
oos-check
. . . . . . . . . . . . . . . . . 236
oos-force
. . . . . . . . . . . . . . . . . 236
oosp-transfer-enabled
outbound-bsns-calls
out-rule-<n>
. . . . . . . . . . . . . . . . 237
override-call-type
override-domain
. . . . . . . . . . . . . . 237
override-domain-from
override-switch-acw
override-to-on-divert
parking-music
. . . . . . . . . . . . . . . 217
password
. . . . . . . . . . . . . . . . . 238
posn-no-answer-overflow
posn-no-answer-timeout
prd-dist-call-ans-time
predictive-call-router-timeout
prefix
. . . . . . . . . . . . . . . . . . . . 238
preview-expired
. . . . . . . . . . . . . . 218
preview-interaction
priority
. . . . . . . . . . . . . . . . . . . 239
public-contact
. . . . . . . . . . . . . . . 239
quiet-cleanup
. . . . . . . . . . . . . . . 249
quiet-startup
. . . . . . . . . . . . . . . . 249
recall-no-answer-timeout
record
. . . . . . . . . . . . . . . . . . . 239
recording-filename
recovery-timeout
. . . . . . . . . . . . . . 239
refer-enabled
. . . . . . . . . . . . . . . 240
reg-delay
. . . . . . . . . . . . . . . . . 249
reg-interval
. . . . . . . . . . . . . . . . 249
registrar-default-timeout
reg-silent
. . . . . . . . . . . . . . . . . . 249
reinvite-requires-hold
reject-call-incall
. . . . . . . . . . . . . . 241
reject-call-notready
replace-prefix
. . . . . . . . . . . . . . . 241
SIP Server—Deployment Guide request-uri
. . . . . . . . . . . . . . . . 241
restricted
. . . . . . . . . . . . . . . . . 248
reuse-sdp-on-reinvite
rfc-2976-dtmf
. . . . . . . . . . . . . . . 240
ringing-on-route-point
ring-tone
. . . . . . . . . . . . . . . . . 219
ring-tone-on-make-call
router-timeout
. . . . . . . . . . . . . . . 220
rq-expire-tmout
. . . . . . . . . . . . . . 249
service-type
. . . . . . . . . . . . . . . . 242
session-refresh-interval
set-notready-on-busy
setting common
. . . . . . . . . . . . . . . . 321
silence-tone
. . . . . . . . . . . . . . . . 220
sip-add-primary-call-id
sip-address
. . . . . . . . . . . . . . . . 220
sip-block-headers
sip-busy-type
. . . . . . . . . . . . . . . 243
sip-call-retain-timeout
sip-cti-control
. . . . . . . . . . . . . . . 244
sip-dtmf-send-rtp
sip-enable-100rel
sip-enable-call-info
sip-enable-moh
. . . . . . . . . . . . . . 222
sip-enable-sdp-codec-filter
sip-hold-rfc3264
. . . . . . . . . . . . . . 222
sip-invite-timeout
sip-invite-treatment-timeout
sip-port
. . . . . . . . . . . . . . . . . . 223
sip-proxy-headers-enabled
sip-refer-to-sst-enabled
sip-retry-timeout
. . . . . . . . . . . . . . 224
sip-ring-tone-mode (application level)
sip-ring-tone-mode (switch/DN level)
sip-server-inter-trunk
sip-treatments-continuous
straight-forward
. . . . . . . . . . . . . . 245
subscribe-presence
subscribe-presence-domain
subscribe-presence-expire
subscribe-presence-from
subscription-id
. . . . . . . . . . . . . . 245
subscription-timeout
sync-emu-agent
. . . . . . . . . . . . . . 249
timed-acw-in-idle
unknown-bsns-calls
unknown-xfer-merge-udata
untimed-wrap-up-value
use-display-name
userdata-<n>
. . . . . . . . . . . . . . . 228
user-data-im-enabled
user-data-im-format
userdata-map-filter
userdata-map-trans-prefix
use-register-for-service-state
413
Index wrap-up-threshold
wrap-up-time
. . . . . . . . . . . . . . . 226
configure
ACD Queue
. . . . . . . . . . . . . . . . . 79
application service
. . . . . . . . . . . . . . 89
device types
. . . . . . . . . . . . . . . . . 77
emergency recording
endpoint
. . . . . . . . . . . . . . . . . . . 81
external music servers
gateway
. . . . . . . . . . . . . . . . . . . 83
MCU
. . . . . . . . . . . . . . . . . . . . . 79
proxies
. . . . . . . . . . . . . . . . . . . . 88
Recorder servers
. . . . . . . . . . . . . . 90
Routing Point
. . . . . . . . . . . . . . . . 87
soft switches
. . . . . . . . . . . . . . . . . 88
Stream Manager
. . . . . . . . . . . . . . . 94
switch objects
. . . . . . . . . . . . . . . . 78
treatment service
. . . . . . . . . . . . . . 91
configuring
Agent Login options
DN options
. . . . . . . . . . . . . . . . . 228
high availability
T-Server
. . . . . . . . . . . . . . .58
INFO
. . . . . . . . . . . . . . . . . . . . 227
INVITE
. . . . . . . . . . . . . . . . . . . 227
multi-site operation
–
steps
. . . . . . . . . . . . . . . . . . 307
T-Server
. . . . . . . . . . . . . . . . . . . 46
TServer section
. . . . . . . . . . . . . . 197
UPDATE
. . . . . . . . . . . . . . . . . . 227
consult-user-data common configuration option
contact configuration option
correct-rqid configuration option
cpd-info-timeout configuration option
cpn configuration option
CPNDigits
. . . . . . . . . . . . . . . . . . 189
customer-greeting configuration option
customer-id common configuration option
customer-side proxy mode
D debug common log option
debug-level x-config section
. . . . . . . . . . . . . . 379
Default Access Code configuration
. . . . . . . . . . . . . . . . 309
defined
. . . . . . . . . . . . . . . . . . 309
default-dn common configuration option
configuration option (application level)
configuration option (switch/DN level)
default-dn-type configuration option
default-filter-type common log option
default-monitor-mode configuration option
default-monitor-scope configuration option
default-music configuration option
default-network-call-id-matching configuration option
default-video-file configuration option
delay-between-refresh-on-switchover configuration option
delay-own-tcs x-config section
. . . . . . . . . . . . . . 379
delay-to-start-refresh-on-switchover configuration option
destination location
destination T-Server
direct-ani
ISCC transaction type
direct-callid
ISCC transaction type
direct-digits transaction type
. . . . . . . . . . . . . . 283
direct-digits-key common configuration option
direct-network-callid
ISCC transaction type
direct-notoken
ISCC transaction type
direct-uui
ISCC transaction type
discovery-port gatekeeper section
divert-on-ringing configuration option
DMX load balancing mode
Log Events
. . . . . . . . . . . . . . . . 393
traditional mode
. . . . . . . . . . . . . . 369
DMX options accept-calls
. . . . . . . . . . . . . . . . 386
accept-registrations
audio-codec
. . . . . . . . . . . . . . . . 378
backup-gk-host
. . . . . . . . . . . . . . 387
backup-gk-port
. . . . . . . . . . . . . . 387
bit-rate
. . . . . . . . . . . . . . . . . . 378
414 Framework 7.6
Index call-address
. . . . . . . . . . . . . . . . 376
call-port
. . . . . . . . . . . . . . . . . . 377
call-protocol
. . . . . . . . . . . . . . . . 377
cif-mpi
. . . . . . . . . . . . . . . . . . . 379
debug-level
. . . . . . . . . . . . . . . . 379
delay-own-tcs
. . . . . . . . . . . . . . . 379
discovery-port
. . . . . . . . . . . . . . . 386
dtmf-payload-type
dtmf-sip
. . . . . . . . . . . . . . . . . . 379
dtmf-source
. . . . . . . . . . . . . . . . 380
h225-port
. . . . . . . . . . . . . . . . . 380
h264-level
. . . . . . . . . . . . . . . . . 380
h264-profile
. . . . . . . . . . . . . . . . 381
main-port
. . . . . . . . . . . . . . . . . 387
packet-size
. . . . . . . . . . . . . . . . 382
primary-gk-host
. . . . . . . . . . . . . . 387
primary-gk-port
. . . . . . . . . . . . . . 387
progress-inband
. . . . . . . . . . . . . . 383
protocol
. . . . . . . . . . . . . . . . . . 377
qcif-mpi
. . . . . . . . . . . . . . . . . . 384
register
. . . . . . . . . . . . . . . . . . 387
registration-timeout
services
. . . . . . . . . . . . . . . . . . 388
silence-supression
sip-port
. . . . . . . . . . . . . . . . . . 384
sip-transport
. . . . . . . . . . . . . . . . 384
still-img-trans
. . . . . . . . . . . . . . . 385
tcp-port-range
. . . . . . . . . . . . . . . 385
trade-off-cap
. . . . . . . . . . . . . . . . 385
verbose
. . . . . . . . . . . . . . . . . . 377
video-codec
. . . . . . . . . . . . . . . . 385
x-type
. . . . . . . . . . . . . . . . . . . 386
DMX parameters
-pstr
. . . . . . . . . . . . . . . . . . . . 382
DN objects
. . . . . . . . . . . . . . . . . . 45
dn-del-mode configuration option
dn-for-unexpected-calls common configuration option
dnis-pool in load-balancing mode
ISCC transaction type
DNs configuring for multi-sites
document conventions
. . . . . . . . . . . . . . . . . 16
errors, commenting on
version number
. . . . . . . . . . . . . . . 16
dtmf-payload configuration option
dtmf-payload-type x-config section
. . . . . . . . . . . . . . 379
dtmf-sip x-config section
. . . . . . . . . . . . . . 379
dtmf-source x-config section
. . . . . . . . . . . . . . 380
dual-dialog-enabled configuration option
E emergency-recording-cleanup-enabled configuration option
emergency-recording-filename configuration option
emulated agent options agent-group
. . . . . . . . . . . . . . . . 198
untimed-wrap-up-value
Emulated Agents
–
emulated agents timed-acw-in-idle
emulated-login-state configuration option
emulate-login configuration option
enable-agentlogin-presence configuration option
enable-async-dns common configuration option
enable-ims configuration option
endpoint monitoring
enforce-external-domains configuration option
error messages
. . . . . . . . . . . . . . . 192
Event Propagation defined
. . . . . . . . . . . . . . . . . . 301
EventAttachedDataChanged
event-propagation common configuration option
event-ringing-on-100trying configuration option
expire common log option
expire-call-tout configuration option
Extension
AssistMode
. . . . . . . . . . . . . . . . 187
attribute
. . . . . . . . . . . . . . . . . . 186
CPNDigits
. . . . . . . . . . . . . . . . . 189
MonitorMode
. . . . . . . . . . . . . . . 186
MonitorScope
. . . . . . . . . . . . . . . 187
NO_ANSWER_ACTION
NO_ANSWER_OVERFLOW
NO_ANSWER_TIMEOUT
no-answer supervision
record
. . . . . . . . . . . . . . . . . . . 186
extension-<n> configuration option
external-registrar configuration option
SIP Server—Deployment Guide 415
Index extn-no-answer-overflow configuration option
ext-no-answer-action configuration option
extrouter configuration section
F fast-busy-tone configuration option
fax-detected configuration option
Features Summary
. . . . . . . . . . . . . . 30
figures
Multiple-to-Point mode
Point-to-Point mode
steps in ISCC/Call Overflow
find-trunk-by-location configuration option
first-party call control transfer
forced-notready configuration option
force-register configuration option
G gateways
. . . . . . . . . . . . . . . . . . . 389
geo-location configuration option
gvm_group_mailbox
GVP integration
. . . . . . . . . . . . . . . 175
option
. . . . . . . . . . . . . . . . . . . 247
H
H.323
. . . . . . . . . . . . . . . . . . . . . 391
H.323 introduction
. . . . . . . . . . . . . . 406
call model
. . . . . . . . . . . . . . . . . 406
ITU-T
. . . . . . . . . . . . . . . . . . . 406
protocols
. . . . . . . . . . . . . . . . . . 406
H.323 port, with DMX
h225-port x-config section
. . . . . . . . . . . . . . 380
h264-level x-config section
. . . . . . . . . . . . . . 380
h264-profile x-config section
. . . . . . . . . . . . . . 381
HA
HA configuration
handle-vsp configuration option
high-availability configuration
host command line parameter
hot standby
defined
. . . . . . . . . . . . . . . . . . 262
T-Server configuration
I ims-default-icid-prefix configuration option
ims-default-icid-suffix configuration option
ims-default-orig-ioi configuration option
inbound-bsns-calls configuration option
inbound-translator-<n> common configuration option
inherit-bsns-type configuration option
init-dnis-by-ruri configuration option
Instant Messaging calling features
. . . . . . . . . . . . . . 144
feature configuration
LCS
. . . . . . . . . . . . . . . . . . . . 148
overview
. . . . . . . . . . . . . . . . . 143
supported call operations
transcript
. . . . . . . . . . . . . . . . . 143
treatments
. . . . . . . . . . . . . . . . . 145
Inter Server Call Control
–
Inter Server Call Control/Call Overflow
–
interaction common log option
internal-bsns-calls configuration option
internal-registrar-domains configuration option
internal-registrar-enabled configuration option
internal-registrar-persistent configuration option
intrusion-enabled configuration option
ISCC
. . . . . . . . . . . . . . . . . . . . . .31
destination T-Server
origination T-Server
ISCC transaction types
direct-ani
. . . . . . . . . . . . . . . 276
direct-callid
. . . . . . . . . . . . . . 276
direct-digits
. . . . . . . . . . . . . . . . 283
direct-network-callid
direct-notoken
416 Framework 7.6
Index direct-uui
. . . . . . . . . . . . . . . .277
dnis-pool
. . . . . . . . . . . . . . . .278
in load-balancing mode
pullback
. . . . . . . . . . . . . . . .280
reroute
. . . . . . . . . . . . . . . . .280
route
. . . . . . . . . . . . . . . . . .281
route-uui
. . . . . . . . . . . . . . . . . . 282
supported
. . . . . . . . . . . . . . . . . 283
ISCC/COF supported
. . . . . . . . . . . . . . . . . 288
iscc-xaction-type
. . . . . . . . . . . . . . . 271
K keep-startup-file common log option
<key name> common log option
known limitations
. . . . . . . . . . . . . . . 194
kpl-interval configuration option
kpl-loss-rate configuration option
kpl-tolerance configuration option
L l command line parameter
legal-guard-time configuration option
level-reassign-<eventID> common log option
level-reassign-disable common log option
License section common configuration options
–
lmspath command line parameter
Load Balancing
. . . . . . . . . . . . . . . . 30
local-node-id common configuration option
location parameter
. . . . . . . . . . . . . . 270
log configuration options
–
log section common log options
–
log-extended section common log options
–
log-filter section common log options
log-filter-data section common log options
logout-on-disconnect configuration option
SIP Server—Deployment Guide log-trace-flags common configuration option
M main-port gatekeeper section
make-call-alert-info configuration option
make-call-rfc3725-flow configuration option
Management Layer
management-port common configuration option
mandatory options common configuration options
mapping extract data from INVITE message
line in SDP message body
SIP headers
. . . . . . . . . . . . . . . . 149
whole SDP message body
match-call-once common configuration option
max-legs-per-sm configuration option
max-pred-req-delay configuration option
Media Layer
. . . . . . . . . . . . . . . . . .41
memory common log option
memory-storage-size common log option
merged-user-data common configuration option
Message Waiting Indicator mwi-agent-enable
mwi-domain
. . . . . . . . . . . . . . . . 214
mwi-extension-enable
mwi-group-enable
mwi-host
. . . . . . . . . . . . . . . . . 215
mwi-mode
. . . . . . . . . . . . . . . . . 215
mwi-port
. . . . . . . . . . . . . . . . . . 215
message_format common log option
messagefile common log option
Microsoft Live Communication Server (LCS)
mode application server
. . . . . . . . . . . . . . 28
customer-side proxy
stand-alone
. . . . . . . . . . . . . . . . . 27
Monitor
MonitorMode
. . . . . . . . . . . . . . . 186
MonitorScope
. . . . . . . . . . . . . . . 187
monitor-internal-calls configuration option
417
Index
Multiple-to-One mode
Multiple-to-Point mode
multi-site supervision
Multi-Site Support section common configuration options
–
Music and Announcements announcement treatments
music treatments
music-in-queue-file
other treatments
. . . . . . . . . . . . . . 157
overview
. . . . . . . . . . . . . . . . . . 154
music-in-conference-file configuration option
music-in-queue-file configuration option
mwi-agent-enable configuration option
mwi-domain configuration option
mwi-extension-enable configuration option
mwi-group-enable configuration option
mwi-host configuration option
mwi-mode configuration option
mwi-port configuration option
N nas-inidication configuration option
nas-private configuration option
NAT/C feature
. . . . . . . . . . . . . . . . 299
nco X/Y command line parameter
Network architecture
. . . . . . . . . . . . . . . . . 26
network attended transfer/conference
Network Considerations
bandwidth requirements
firewalls
. . . . . . . . . . . . . . . . . . . 40
remote agents
. . . . . . . . . . . . . . . . 39
tuning
. . . . . . . . . . . . . . . . . . . . 39
voice quality
. . . . . . . . . . . . . . . . . 38
network objects
. . . . . . . . . . . . . . . . 40
network-request-timeout common configuration option
NO_ANSWER
NO_ANSWER_ACTION
NO_ANSWER_OVERFLOW
NO_ANSWER_TIMEOUT
No-Answer Supervision agent configuration
agents
. . . . . . . . . . . . . . . . . . . 158
extension configuration
extensions
. . . . . . . . . . . . . . . . 158
position
. . . . . . . . . . . . . . . . . . 159
position configuration
no-answer-action configuration option
no-answer-overflow configuration option
no-answer-timeout configuration option
notify-idle-tout common configuration option
Number Translation feature
–
number translation rules
num-of-licenses common configuration option
num-sdn-licenses common configuration option
O objects
Agent Logins
. . . . . . . . . . . . . . . . 45
DNs
. . . . . . . . . . . . . . . . . . . . . 45
network
. . . . . . . . . . . . . . . . . . . 40
Switches
. . . . . . . . . . . . . . . . . . 45
Switching Offices
. . . . . . . . . . . . . . 44
telephony
. . . . . . . . . . . . . . . . . . 40
observing-routing-point configuration option
One-to-One mode
. . . . . . . . . . . . . . 281
oos-check configuration option
oos-force configuration option
oosp-transfer-enabled configuration option
origination location
. . . . . . . . . . . . . . 269
origination T-Server
Out Of Signaling Path oosp-transfer-enabled
outbound-bsns-calls configuration option
Out-of-Service oos-check
. . . . . . . . . . . . . . . . . 236
oos-force
. . . . . . . . . . . . . . . . . 236
out-rule-<n> configuration option
override-call-type configuration option
override-domain configuration option
override-domain-from
418 Framework 7.6
Index configuration option
override-switch-acw configuration option
override-to-on-divert configuration option
P packet-size x-config section
. . . . . . . . . . . . . . 382
parking-music configuration option
password configuration option
periodic-check-tout common configuration option
Personal greeting feature configuration
overview
. . . . . . . . . . . . . . . . . . 160
Point-to-Point mode
port command line parameter
position no-answer supervision
posn-no-answer-overflow configuration option
posn-no-answer-timeout configuration option
prd-dist-call-ans-time configuration option
predictive-call-router-timeout configuration option
prefix configuration option
Presence Subscription feature configuration
LCS
. . . . . . . . . . . . . . . . . . . . 163
overview
. . . . . . . . . . . . . . . . . . 161
processing userdata
PUBLISH request
subscribe-presence
subscribe-presence-domain
subscribe-presence-expire
subscribe-presence-from
subscription-id
. . . . . . . . . . . . . . . 245
subscription-timeout
updating agent state
Preview interactions
preview-expired configuration option
preview-interaction configuration option
primary servers
. . . . . . . . . . . . . . . . 51
primary-gk-host gatekeeper section
primary-gk-port gatekeeper section
SIP Server—Deployment Guide print-attributes common log option
priority configuration option
progress-inband x-config section
. . . . . . . . . . . . . . 383
protocol common configuration option
link section
. . . . . . . . . . . . . . . . 377
Providing a caller ID
public-contact configuration option
pullback
ISCC transaction type
Q qcif-mpi x-config section
. . . . . . . . . . . . . . 384
quiet-cleanup configuration option
quiet-startup configuration option
R rebind-delay common configuration option
recall-no-answer-timeout configuration option
reconnect-tout common configuration option
record
. . . . . . . . . . . . . . . . . . . . 186
configuration option
recording-filename configuration option
recovery-timeout configuration option
redundancy hot standby
warm standby
redundancy types
,
hot standby
. . . . . . . . . . . . . . . . 262
refer-enabled configuration option
reg-delay configuration option
reg-interval configuration option
register gatekeeper section
register-attempts common configuration option
register-tout common configuration option
419
Index registrar-default-timeout configuration option
registration-timeout gatekeeper section
reg-silent configuration option
reinvite-requires-hold configuration option
reject-call-incall configuration option
reject-call-notready configuration option
reject-subsequent-request common configuration option
remote agents
. . . . . . . . . . . . . . . . 39
Remote Server Registration
Remote Supervision feature configuration
feature limitations
overview
. . . . . . . . . . . . . . . . . . 115
use of Extensions attribute
Remote Talk
. . . . . . . . . . . . . . . . . 166
replace-prefix configuration option
report-connid-changes common configuration option
request-collection-time common configuration option
request-tout common configuration option
ISCC configuration option
request-uri configuration option
reroute
ISCC transaction type
reservation-time common configuration option
resource-allocation-mode common configuration option
resource-load-maximum common configuration option
restricted options configuration options
reuse-sdp-on-reinvite configuration option
rfc-2976-dtmf configuration option
ringing-on-route-point configuration option
ring-tone configuration option
ring-tone-on-make-call configuration option
route
ISCC transaction type
route-dn common configuration option
router-timeout configuration option
route-uui
ISCC transaction type
routing
Inter Server Call Control
–
rq-expire-tmout configuration option
rule-<n> common configuration option
run.bat
. . . . . . . . . . . . . . . . . . . . .73
run.sh
. . . . . . . . . . . . . . . . . . . . .72
S
Security section common configuration options
segment common log option
server-id common configuration option
services gatekeeper section
service-type configuration option
session-refresh-interval configuration option
set-notready-on-busy configuration option
setting common configuration options
setting configuration options common
. . . . . . . . . . . . . . . . . . 321
silence-supression x-config section
. . . . . . . . . . . . . . 384
silence-tone configuration option
SIP introduction
. . . . . . . . . . . . . . . 401
end points
. . . . . . . . . . . . . . . . . 402
IETF
. . . . . . . . . . . . . . . . . . . . 401
messages
. . . . . . . . . . . . . . . . . 403
proxy and registrar
RTP
. . . . . . . . . . . . . . . . . . . . 401
signalling and call setup
TCP/IP model
. . . . . . . . . . . . . . . 401
sip-add-primary-call-id configuration option
sip-address configuration option
sip-block-headers configuration option
sip-busy-type configuration option
420 Framework 7.6
Index sip-call-retain-timeout configuration option
sip-cti-control configuration option
sip-dtmf-send-rtp configuration option
sip-enable-100rel configuration option
sip-enable-call-info configuration option
sip-enable-moh configuration option
sip-enable-sdp-codec-filter configuration option
sip-hold-rfc3264 configuration option
sip-invite-timeout configuration option
sip-invite-treatment-timeout configuration option
sip-port configuration option
x-config section
. . . . . . . . . . . . . . 384
sip-proxy-headers-enabled configuration option
sip-refer-to-sst-enabled configuration option
sip-retry-timeout configuration option
sip-ring-tone-mode configuration option (application level)
configuration option (switch/DN level)
sip-server-inter-trunk configuration option
sip-transport x-config section
. . . . . . . . . . . . . . 384
sip-treatments-continuous configuration option
spool common log option
stand-alone mode
. . . . . . . . . . . . . . 27
standard common log option
Start Info
. . . . . . . . . . . . . . . . . . . 389
still-img-trans x-config section
. . . . . . . . . . . . . . 385
straight-forward configuration option
Stream Manager appropriate codec format
audio files
. . . . . . . . . . . . . . . . . . 95
defined
. . . . . . . . . . . . . . . . . . . . 94
Multipoint Conference Unit
Music-on-Hold (MoH) server
subscribe-presence configuration option
SIP Server—Deployment Guide subscribe-presence-domain configuration option
subscribe-presence-expire configuration option
subscribe-presence-from configuration option
subscription-id configuration option
subscription-timeout configuration option
Supported Functionality table
–
Switch objects
. . . . . . . . . . . . . . . . .45
multi-site operation
Switching Office objects
multi-site operation
Switch-Specific Configuration
sync-emu-agent configuration option
sync-reconnect-tout common configuration option
T
Target ISCC
Access Code configuration
Default Access Code configuration
tcp-port-range x-config section
. . . . . . . . . . . . . . 385
tcs-queue common configuration option
tcs-use common configuration option
telephony objects
. . . . . . . . . . . . . . .40
third-party known limitations
Music-on-Hold server architecture
third-party call control transfer
time_convert common log option
time_format common log option
timed-acw-in-idle configuration options
emulated agents
timeout common configuration option
ISCC configuration option
timeout value format common configuration options
–
TInitiateConference
TInitiateTransfer
. . . . . . . . . . . . . . . 270
T-Library Functionality
TMakeCall
. . . . . . . . . . . . . . . . . . 270
TMuteTransfer
. . . . . . . . . . . . . . . . 270
trace common log option
421
Index trade-off-cap x-config section
. . . . . . . . . . . . . . 385
transaction types (ISCC)
supported
. . . . . . . . . . . . . . . . . 283
transfer connect service
Translation Rules section common configuration options
treatments
Instant Messaging
TRouteCall
. . . . . . . . . . . . . . . . . . 270
keys
. . . . . . . . . . . . . . . . . . . . 187
trunk lines
. . . . . . . . . . . . . . . 281
T-Server configuring Application objects
for multi-sites
. . . . . . . . . . . . . . 307
configuring redundancy
HA
. . . . . . . . . . . . . . . . . . . . . . 57
high availability
. . . . . . . . . . . . . . . 57
hot standby
. . . . . . . . . . . . . . . . . 57
multi-site operation
–
redundancy
,
starting
. . . . . . . . . . . . . . . . . . . . 74
using Configuration Manager
warm standby
. . . . . . . . . . . . . . . . 55
TServer Section configuration options
T-Server section common configuration options
–
TSingleStepTransfer
TXRouteType
. . . . . . . . . . . . . . . . . 271
typical SIP Server network
typographical styles
U
UNIX installing T-Server
starting applications
starting T-Server
. . . . . . . . . . . . . . . 74
starting with run.sh
unknown-bsns-calls configuration option
unknown-xfer-merge-udata configuration option
untimed-wrap-up-value configuration option
emulated agent options
use-data-from common configuration option
use-display-name configuration option
use-implicit-access-numbers common configuration option
user data propagation
userdata-<n> configuration option
user-data-im-enabled configuration option
user-data-im-format configuration option
user-data-limit common configuration option
userdata-map-filter configuration option
userdata-map-trans-prefix configuration option
use-register-for-service-state configuration option
V
V command line parameters
VDN
. . . . . . . . . . . . . . . . . . . . . 281
verbose common log option
link section
. . . . . . . . . . . . . . . . 377
version numbering document
. . . . . . . . . . . . . . . . . . 16
Video feature configuration
Push Video
. . . . . . . . . . . . . . . . 171
start
. . . . . . . . . . . . . . . . . . . . 171
stop
. . . . . . . . . . . . . . . . . . . . 171
video-codec
. . . . . . . . . . . . . . . . . 385
voice quality
. . . . . . . . . . . . . . . . . .38
W warm standby
T-Server configuration
Windows installing T-Server
starting applications
starting T-Server
. . . . . . . . . . . . . . 74
starting with run.bat
wrap-up-threshold configuration option
wrap-up-time configuration option
X x-conn-debug-all common log option
x-conn-debug-api common log option
422 Framework 7.6
Index x-conn-debug-dns common log option
x-conn-debug-open common log option
x-conn-debug-security common log option
x-conn-debug-select common log option
x-conn-debug-timers common log option
x-conn-debug-write common log option
x-type x-config section
. . . . . . . . . . . . . . 386
SIP Server—Deployment Guide 423
Index
424 Framework 7.6
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