User`s manual
Table Of Contents
- Mediant 2000 & TP-1610 & TP-260/UNI SIP User’s Manual Version 5.0
- Table of Contents
- List of Figures
- List of Tables
- Notices
- 1. Overview
- 2. Physical Description
- 3. Installation
- 4. Getting Started
- 5. Web Management
- Computer Requirements
- Protection and Security Mechanisms
- Accessing the Embedded Web Server
- Getting Acquainted with the Web Interface
- Protocol Management
- Advanced Configuration
- Status & Diagnostic
- Software Update Menu
- Maintenance
- Logging Off the Embedded Web Server
- 6. Gateway's ini File Configuration
- Secured ini File
- Modifying an ini File
- The ini File Content
- The ini File Structure
- The ini File Example
- Networking Parameters
- System Parameters
- Web and Telnet Parameters
- Security Parameters
- RADIUS Parameters
- SNMP Parameters
- SIP Configuration Parameters
- Voice Mail Parameters
- ISDN and CAS Interworking-Related Parameters
- Number Manipulation and Routing Parameters
- E1/T1 Configuration Parameters
- Channel Parameters
- Configuration Files Parameters
- 7. Using BootP / DHCP
- 8. Telephony Capabilities
- Working with Supplementary Services
- Configuring the DTMF Transport Types
- Fax & Modem Transport Modes
- Event Notification using X-Detect Header
- ThroughPacket™
- Dynamic Jitter Buffer Operation
- Configuring the Gateway’s Alternative Routing (based on Conn
- Call Detail Report
- Supported RADIUS Attributes
- Trunk to Trunk Routing Example
- Proxy or Registrar Registration Example
- SIP Call Flow Example
- SIP Authentication Example
- 9. Networking Capabilities
- 10. Advanced PSTN Configuration
- 11. Advanced System Capabilities
- 12. Special Applications
- 13. Security
- 14. Diagnostics
- 15. SNMP-Based Management
- SNMP Standards and Objects
- Carrier Grade Alarm System
- Cold Start Trap
- Third-Party Performance Monitoring Measurements
- TrunkPack-VoP Series Supported MIBs
- Traps
- SNMP Interface Details
- SNMP Manager Backward Compatibility
- Dual Module Interface
- SNMP NAT Traversal
- SNMP Administrative State Control
- AudioCodes’ Element Management System
- 16. Configuration Files
- Appendix A. Selected Technical Specifications
- Appendix B. Supplied SIP Software Kit
- Appendix C. SIP Compliance Tables
- Appendix D. The BootP/TFTP Configuration Utility
- Appendix E. RTP/RTCP Payload Types and Port Allocation
- Appendix F. RTP Control Protocol Extended Reports (RTCP-XR)
- Appendix G. Accessory Programs and Tools
- Appendix H. Release Reason Mapping
- Appendix I. SNMP Traps
- Appendix J. Installation and Configuration of Apache HTTP Server
- Appendix K. Regulatory Information

SIP User's Manual 8. Telephony Capabilities
Version 5.0 207 October 2006
8 Telephony Capabilities
8.1 Working with Supplementary Services
The gateway gateway supports the following supplementary services:
Call Hold / Retrieve (refer to Section 8.1.1 on page 207)
Call Transfer (refer to Section 8.1.2 on page 207)
Call Forward (doesn't initiate call forward, only responds to call forward request)
Call Waiting
The gateway SIP users are only required to enable the Hold and Transfer features. The
call forward (supporting 30x redirecting responses) and call waiting (receive of 182
response) features are enabled by default. Note that all call participants must support the
specific used method.
Note: When working with application servers (such as BroadSoft’s BroadWorks)
in client server mode (the application server controls all supplementary
services and keypad features by itself), the gateway’s supplementary
services must be disabled.
8.1.1 Call Hold and Retrieve Features
The party that initiates the hold is called the holding party, the other party is called the
held party. The gateway can't initiate the hold, but it can respond to hold request, and
as such it is a held party.
After a successful hold, the held party should hear HELD_TONE, defined in the
gateway's Call Progress Tones file.
Retrieve can be performed only by the holding party while the call is held and active.
After a successful retrieve the voice should be connected again.
The hold and retrieve functionalities are implemented by Reinvite messages. The IP
address 0.0.0.0 as the connection IP address or the string ‘a=inactive’ in the received
Reinvite SDP cause the gateway to enter Hold state and to play held tone (configured
in the gateway) to the PBX/PSTN. If the string ‘a=recvonly’ is received in the SDP
message, the gateway stops sending RTP packets, but continues to listen to the
incoming RTP packets. Usually, the remote party plays, in this scenario, Music on
Hold (MOH) and the gateway forwards the MOH to the held party.
8.1.2 Call Transfer
There are two types of call transfers:
Consultation Transfer
Blind Transfer
The common way to perform a consultation transfer is as follows:
In the transfer scenario there are three parties:
Party A - transferring, Party B – transferred, Party C – transferred to.
A Calls B.
B answers.
A presses the hookflash and puts B on-hold (party B hears a hold tone).
A dials C.