Genesys Framework 8.0
Manuals and User Guides for Genesys Framework 8.0. We found 3 manuals for free downloads Deployment Guide, Reference manual
Below you will find brief information for SIP Server Framework 8.0. SIP Server is a TCP/IP-based server that can also act as a messaging interface between SIP Server clients. It is the Genesys software component that provides an interface between your telephony hardware and the rest of the Genesys software components in your enterprise.
Key Features
Provides an interface between telephony hardware and Genesys software
Translates and keeps track of events and requests
Acts as a messaging interface between SIP Server clients
Supports redundant servers for high availability
Provides multi-site support for distributed deployments
Pages: 514 Genesys Framework 8.0 Deployment Guide
Brand: Genesys Category: Networking Size: 2 MB
Languages: English
Table of contents
- 1 Deployment Guide
- 3 Table of Contents
- 13 List of Procedures
- 15 Preface
- 15 About SIP Server
- 16 Intended Audience
- 16 Reading Prerequisites
- 16 Making Comments on This Document
- 17 Contacting Genesys Technical Support
- 19 SIP Server Deployment
- 19 New Features in Release 8.0 of SIP Server
- 25 SIP Server Fundamentals
- 25 Overview
- 26 SIP Server Architecture
- 26 SIP Server Deployment Modes
- 29 Media Server Deployment Architecture
- 30 Redundant SIP Servers (High Availability)
- 30 Load Balancing
- 31 Multi-Threaded Architecture
- 32 Multi-Site Support
- 32 Next Steps
- 33 SIP Server General Deployment
- 33 Prerequisites
- 34 Software Requirements
- 34 Hardware and Network Environment Requirements
- 35 Licensing Requirements
- 37 About Configuration Options
- 38 Network Considerations
- 38 Voice Quality
- 39 Bandwidth Requirements
- 39 Remote Agent Configuration
- 40 Deployment Sequence
- 41 Wizard Deployment of SIP Server
- 41 Wizard Configuration of SIP Server
- 42 Wizard Installation of SIP Server
- 44 Manual Deployment of SIP Server
- 44 Manual Configuration of Telephony Objects
- 46 Manual Configuration of SIP Server
- 47 Manual Installation of SIP Server
- 50 Next Steps
- 51 Starting and Stopping SIP Server
- 51 Command-Line Parameters
- 53 Starting and Stopping with the Management Layer
- 54 Starting with Startup Files
- 55 Starting Manually
- 56 Verifying Successful Startup
- 57 Stopping Manually
- 57 Starting and Stopping with Windows Services Manager
- 58 Next Steps
- 59 SIP Devices Support
- 59 Overview
- 61 About Trunk and Trunk Group DNs
- 62 Configuring Devices and Services
- 63 Configuring ACD Queues
- 63 Configuring MCUs
- 65 Configuring Endpoints
- 68 Configuring Gateways
- 70 Configuring Music Servers
- 72 Configuring Routing Points
- 72 Configuring Softswitches
- 74 Configuring an Application Service
- 75 Configuring a Recording Service
- 77 Configuring a Treatment Service
- 77 Configuring an MSML Service
- 80 Configuring Agent Logins
- 80 Configuring Stream Manager
- 81 Location of Media Files
- 81 Load Balancing of Media Services
- 81 Determining Codecs
- 81 Configuring Genesys Media Server
- 82 About Genesys Media Server
- 82 SIP Server and Media Server Integration
- 82 Genesys Media Server Integration
- 85 SIP Server Feature Support
- 86 Alternate Routing
- 86 Alternate Routing for Stranded Calls
- 89 Alternate Routing for Unresponsive DNs
- 90 Alternate Routing for Unresponsive URS
- 91 Alternate Routing for Calls to an External Destination
- 92 Application Failure Detection
- 92 How Failure Detection Works
- 93 Feature Configuration
- 93 Feature Limitation
- 93 Associating an ACD Queue with a Routing Point
- 94 Asterisk Voice Mail Integration
- 95 Genesys Voice Mail Configuration Adapter for Asterisk
- 95 Message Waiting Indicator Functionality
- 96 Feature Configuration
- 97 Automatic Agent Logout
- 97 Feature Configuration
- 97 Call Recording
- 97 Regular Call Recording
- 99 Feature Configuration
- 100 Emergency (Manual) Call Recording
- 101 Call Supervision
- 101 Overview
- 104 Feature Configuration
- 109 Multi-Site Supervision
- 112 Remote Supervision
- 120 Call Transfer and Conference
- 123 Conference Calls
- 124 Consultation Transfers and Conferences
- 126 Feature Limitations
- 126 Class of Service
- 127 Outbound Dialing Rules
- 134 Ring-Through Rules
- 134 Feature Configuration
- 137 Contact Header Handling Options
- 138 Sample Call Flow Scenario
- 138 Feature Configuration
- 139 Control of SIP Response Code from within a Routing Strategy
- 139 Feature Configuration
- 141 Dial Plan
- 141 Dial Plan Configuration Overview
- 142 Dial Plan Call Flow
- 142 The Dial-Plan Rule
- 147 About Privilege Levels
- 148 Feature Configuration
- 156 DTMF Tones Generation
- 156 DTMF Parameters for Play Application Treatments
- 157 Feature Configuration
- 157 Dummy SDP
- 158 How It Works
- 160 Feature Configuration
- 160 Early Media for Inbound Calls
- 161 Feature Configuration
- 162 Emulated Agents
- 162 Business-Call Handling
- 164 Emulated Agents Support
- 167 Endpoint Service Monitoring
- 167 Passive Out-of-Service Detection
- 168 Active Out-of-Service Detection
- 169 IMS Integration
- 169 Genesys Contact Center in the IMS Network
- 170 Feature Configuration
- 171 Instant Messaging
- 171 Instant Messaging Transcript
- 172 Supported Call Operations
- 176 Feature Configuration
- 177 Mapping SIP Headers and SDP Messages
- 177 From SIP Messages to T-Library Messages
- 180 From T-Library Messages to SIP Messages
- 182 SDP Message Mapping
- 183 Dynamic DN Replacement
- 183 Media Server Reliability
- 184 How SIP Server Detects a Media Server Failure
- 185 Reliability for Conference Calls
- 185 Reliability for Supervisor Features
- 185 Reliability for Voice Call Recording
- 185 Feature Configuration
- 186 Feature Limitation
- 186 Music and Announcements
- 186 Announcement Treatments on Routing Points
- 188 Music Treatments on Routing Points
- 189 Other Treatments on Routing Points
- 189 Feature Limitation
- 189 Nailed-Up Connection
- 190 How SIP Server Establishes the Nailed-Up Connection
- 191 Processing Contact Center Calls for the Nailed-Up Connection
- 191 Feature Configuration
- 192 Feature Limitations
- 192 Network Asserted Identity
- 192 How the Mechanism Works
- 193 How SIP Server Supports the Mechanism
- 194 Inbound Calls
- 196 Outbound Calls
- 197 Internal Call (CLIR)
- 198 Feature Configuration
- 200 No-Answer Supervision
- 200 Business and Private Calls
- 201 Agent No-Answer Supervision
- 201 Extension No-Answer Supervision
- 202 Position No-Answer Supervision
- 202 Device-Specific Overrides
- 202 Extension Attributes for Overrides for Individual Calls
- 202 Feature Limitations
- 203 Outbound IP Solution Integration
- 203 SIP Server Features for Outbound IP Solution
- 206 Feature Configuration
- 207 Feature Limitation
- 207 Overload Control
- 208 How Overload Control Works
- 210 Feature Configuration
- 211 Personal Greeting
- 212 Feature Configuration
- 214 Playing Multiple Media Files
- 216 Presence from Switches and Endpoints
- 217 Subscription to SIP Server
- 217 SIP Server Subscription to Endpoints Behind a Switch
- 218 Endpoint Sends PUBLISH Requests to SIP Server
- 219 Mapping Polycom Agent-Status Updates to T-Library Events
- 220 Presence Integration with Microsoft Office Communications Server 2007
- 226 Preview Interactions
- 226 Preview Interaction for IM
- 227 Private Headers
- 227 Providing Call Participant Info
- 227 Feature Configuration
- 228 Providing a Caller ID
- 228 Quality of Service
- 229 Remote Media on Genesys SIP Endpoint SDK 8.0
- 229 Feature Configuration
- 230 Remote Server Registration
- 230 Remote Talk
- 231 SIP Traffic Monitoring
- 231 How it Works
- 231 Feature Configuration
- 232 Feature Limitations
- 232 Transport Layer Security for SIP Traffic
- 232 About TLS
- 233 Feature Configuration
- 235 Trunk Capacity Support
- 235 Tromboning Control
- 236 Duplicated DN Names
- 237 Bounced Calls Between T-Servers
- 237 Feature Configuration
- 238 Trunk Optimization for Multi-Site Transfers
- 241 Feature Configuration
- 242 Feature Limitation
- 242 Video Support
- 242 Push Video
- 244 Other Supported Scenarios
- 244 Feature Configuration
- 247 Working with Multiple Devices
- 247 Device Selection Procedure
- 249 Selection Based on Geo-Location
- 249 Feature Configuration
- 253 Genesys Voice Platform Integration
- 253 Genesys Media Server
- 255 T-Library Functionality Support
- 255 Using T-Library Functions
- 264 Using the Extensions Attribute
- 272 Error Messages
- 274 Known Limitations
- 275 Third-Party Equipment—Known Limitations
- 277 SIP Server Configuration Options
- 277 Application-Level Options
- 277 T-Server Section
- 327 UPDATE, INVITE, INFO, and REFER Sections
- 328 Log Section
- 328 Multi-Site Support Section
- 328 SIP Error Map Section
- 330 Agent Login–Level and DN-Level Options
- 368 GVP Integration Options
- 369 Reserved Options
- 370 Changes from Release 7.6 to Release 8.0
- 375 T-Server Common Functions and Procedures
- 375 New for All T-Servers in 8.0
- 377 T-Server Fundamentals
- 377 Learning About T-Server
- 377 Framework and Media Layer Architecture
- 379 T-Server Requests and Events
- 383 Advanced Disconnect Detection Protocol
- 384 Redundant T-Servers
- 388 Multi-Site Support
- 388 Agent Reservation
- 389 Client Connections
- 391 Multi-Site Support
- 391 Multi-Site Fundamentals
- 393 ISCC Call Data Transfer Service
- 394 ISCC Call Flows
- 400 ISCC Transaction Types
- 408 T-Server Transaction Type Support
- 412 Transfer Connect Service Feature
- 413 ISCC/Call Overflow Feature
- 417 Number Translation Feature
- 418 Number Translation Rules
- 425 Network Attended Transfer/Conference Feature
- 427 Event Propagation Feature
- 428 User Data Propagation
- 429 Party Events Propagation
- 430 Switch Partitioning
- 431 Event Propagation Configuration
- 434 ISCC Transaction Monitoring Feature
- 434 Configuring Multi-Site Support
- 435 Applications
- 436 Switches and Access Codes
- 442 DNs
- 447 Configuration Examples
- 449 Common Configuration Options
- 450 Setting Configuration Options
- 450 Mandatory Options
- 450 Log Section
- 455 Log Output Options
- 459 Examples
- 460 Debug Log Options
- 463 Log-Extended Section
- 465 Log-Filter Section
- 465 Log-Filter-Data Section
- 465 Security Section
- 466 SML Section
- 467 Common Section
- 467 Changes from 7.6 to 8.0
- 469 T-Server Common Configuration Options
- 469 Setting Configuration Options
- 470 Mandatory Options
- 470 T-Server Section
- 475 License Section
- 478 Agent-Reservation Section
- 479 Multi-Site Support Section
- 481 ISCC Transaction Options
- 485 Transfer Connect Service Options
- 486 ISCC/COF Options
- 488 Event Propagation Options
- 489 Number Translation Option
- 489 Translation Rules Section
- 490 Backup-Synchronization Section
- 491 Call-Cleanup Section
- 493 Security Section
- 493 Timeout Value Format
- 494 Changes from Release 7.6 to 8.0
- 495 Related Documentation Resources
- 497 Document Conventions
- 499 Index
Other manuals and documents
Frequently Answers and Questions
What is SIP Server?
SIP Server is a Genesys software component that provides an interface between your telephony hardware and the rest of the Genesys software components in your enterprise.
What is the purpose of SIP Server?
SIP Server translates and keeps track of events and requests that come from, and are sent to, your telephony devices. It can also act as a messaging interface between SIP Server clients.
How does SIP Server support redundancy?
SIP Server supports redundant servers for high availability, ensuring that call handling continues uninterrupted even if one server fails.
What are the deployment modes of SIP Server?
SIP Server can be deployed in several modes, including single server, redundant servers (high availability), and load balanced servers.
What are the benefits of using SIP Server?
SIP Server allows for seamless integration of telephony devices with Genesys software, provides robust call handling capabilities, and supports various features like call recording, call transfer, and more.
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