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
Mediant 2000 & TP-1610 & TP-260
SIP User's Manual 174 Document #: LTRT-68805
Table 6-9: ISDN and CAS Interworking-Related Parameters (continues on pages 172 to 179)
ini File Field Name
Web Parameter Name
Valid Range and Description
PlayRBTone2IP
[Play Ringback Tone to IP]
0 = Ringback tone isn’t played (default).
1 = Ringback tone is played (to IP) after SIP 183+SDP or 180+SDP response is
sent.
If configured to 1 ('Play'), and if EnableEarlyMedia = 1, for IP-to-Tel calls the
gateway may play a ringback tone to IP, according to the following:
• For CAS interfaces, the gateway opens a voice channel, sends a 183+SDP
response and plays a Ringback tone to IP.
• For ISDN interfaces, if a Progress or an Alert message with PI (1 or 8) is
received from the ISDN, the gateway opens a voice channel, sends a
183+SDP or 180+SDP response, but it doesn’t play a Ringback tone to IP. If
PI (1 or 8) is received from the ISDN, the gateway assumes that Ringback
tone is played by the ISDN Switch. Otherwise, the fateway plays a Ringback
tone to IP after receiving an Alert message from the ISDN. It sends a
180+SDP response, signaling to the originating party to open a voice channel
to hear the played Ringback tone.
Note 1: To enable the gateway to send a 183/180+SDP responses, set
EnableEarlyMedia to 1.
Note 2: If EnableDigitDelivery = 1, the gateway doesn’t play a Ringback tone to IP
and doesn’t send 183 or 180+SDP responses.
DefaultCauseMapISDN2I
P
[Default Cause Mapping
From ISDN to IP]
Defines a single default ISDN Release Cause that is used (in ISDN to IP calls)
instead of all received release causes except when the following Q.931 cause
values are received: Normal Call Clearing (16), User Busy (17), No User
Responding (18) or No Answer from User (19).
The range is valid Q.931 release causes (0 to 127). The default value is 0
(indicates that the parameter is not configured - static mapping is used).
CauseMapSIP2ISDN_ID
[Release Cause Mapping
from SIP to ISDN]
Defines a flexible mapping of SIP Responses and Q.850 Release Causes.
CauseMapSIP2ISDN_<ID> = <SIP Response>,<Q.850 Release Cause>
When a SIP response is received (from the IP side), the gateway searches this
mapping table for a match. If the SIP response is found, the Release Cause
assigned to it is sent to the PSTN. If no match is found, the default static mapping
is used.
For example:
CauseMapSIP2ISDN=404,3
Note: This parameter can appear up to 12 times.
CauseMapISDN2SIP_ID
[Release Cause Mapping
from ISDN to SIP]
Defines a flexible mapping of Q.850 Release Causes and SIP Responses.
CauseMapISDN2SIP_<ID> = <Q.850 Release Cause>,<SIP Response>
When a Release Cause is received (from the PSTN side), the gateway searches
this mapping table for a match. If the Q.850 Release Cause is found, the SIP
response assigned to it is sent to the IP side. If no match is found, the default
static mapping is used.
For example:
CauseMapISDN2SIP=6,406
Note: This parameter can appear up to 12 times.
RemoveCLIWhenRestrict
ed
[Remove CLI when
Restricted]
Determines (for IP to Tel calls) whether the Calling Number IE and Calling Name
IE are removed from the outgoing ISDN Setup message if the presentation is set
to Restricted.
0 = IE aren’t removed (default).
1 = IE are removed.