user manual
Table Of Contents
- Voice Gateways System Manual
- About This Manual
- Contents
- Chapter 1 - System Description
- Chapter 2 - Installation
- Chapter 3 - Using the Web Configuration Server
- 3.1 Introduction to the Web Configuration Server
- 3.2 Accessing the Web Configuration Server
- 3.3 Using the Web Configuration Server
- 3.4 Home Menu - Product Info Page
- 3.5 WAN Menu
- 3.6 VLAN Tagging Menu
- 3.7 Telephone Menu
- 3.8 BW Reservation - DRAP Configuration Page
- 3.9 System Menu
- 3.10 Upgrade Page
- 3.11 Restart Page
- 3.12 Logout Page
- 3.13 Parameters Summary
- Appendix A - Internal Class 5 Services
- Appendix B - Default Telephony Parameters
- Appendix C - New Features
- Glossary

Telephone Menu
Voice Gateways System Manual 41
Table 3-6: SIP Configuration/H323 Telephone Page Parameters
Parameter Description
Dialplan The Dialplan parameter defines how the Voice Gateway decides
that a complete number has been dialed. See more details in
Section
3.7.7.
The default value is xx.T|xx.#, which means that each of the
following schemes can be used:
xx.T: Dial timeout. Any number of digits may be dialed.
Following T seconds in which no new digit is dialed, a decision
is reached that dialing was completed and the unit will send the
dialing sequence received up to this time as a complete
telephone number. This is necessary since the whole telephone
number is sent at once and not digit by digit.
xx.#: Any number of digits may be dialed. A decision that dialing
was completed will be reached once # is pressed.
The combination of both schemes means that dialing is completed
either after a timeout of T seconds or after pressing #.
Dial timeout The timeout in seconds for the dial timeout dialplan. The number of
seconds that the unit waits before it sends a complete telephone
number. This is necessary since the whole telephone number is
sent at once and not digit by digit.
The range is 1 to 60 seconds
Default value is 4 seconds.
Use # Use # as a quick dial function. To send the # along with the number
to the server, uncheck the box.
The default is enabled.
RTP Port Range
(SIP model only)
The start and end UDP port-range for RTP protocol.
Recommended values for Start and End ports are in the range
1030-65535.
The default Start port is 8000. The default End port is 8015.
Telephone line Switch the telephone line On or Off. The default is Off.