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time_on and time_off are system specified values (respectively to be 300 and 1023). Other signaling
tone must be configured four parameters. If this switch is single frequency, then the high frequency will
take the invalid value 2001.
After completing configuring and exiting from the configure mode, the current configuration just takes
effect. All DSPs and ports on a same slot will take the same configuration. Therefore we suggest you
use ports on different slots when connecting different switchs except that the parameters of the two
switch signaling tone are consistent.
After completing configuring and saving them, current configuration will be always taken. Use
command default cptone slot_num to resume the default configuration of this slot signaling tone.
When configuring cptone, we suggest you first use the command default cptone slot_num to reset
corresponding of a slot into default value before entering into cptone configure mode, or else after
entering into configure mode the ext signaling tone configured will be added behind former ext signal,
but pbx signaling tone will overlay the former configuration. If the cptone of this slot has no ext signaling
tone, before configuring you needn't use command default to reset.
12.4 Configure Voice over IP
This chapter shows you how to configure Voice over IP (VoIP) on the D-LinkIP telephone equipments.
VoIP is a protocol that carry voice traffic over an IP network. Voice over IP is primarily a software feature;
V100 has the fixed FXS voice port, to use this feature on D-Link DI-1750 and DI-3660 router, you must
install a voice network module (VNM) or a voice interface card, each of interface card corresponding a
particular signaling type associated with a voice port.
Voice over IP offers the following benefits:
♦ Toll bypass
♦ Remote PBX presence over WANs
♦ Unified voice/data trunking
♦ POTS-Internet telephony gateways
12.4.1 How Voice over IP Processes a Telephone Call
Before configuring Voice over IP, it helps to understand what happens at an application level when you
place a call using Voice over IP. The general flow of a two-party voice call using Voice over IP is as
follows (FXS port):
(1)
The user picks up the handset; this signals an off-hook condition to the signaling application part
of Voice over IP.
(2)
The session application part of Voice over IP issues a dial tone and waits for the user to dial a
telephone number.
(3)
The user dials the telephone number; those numbers are accumulated and stored by the
session application.
(4)
After enough digits are accumulated to match a configured destination pattern, the telephone
number is mapped to an IP host via the dial plan mapper. The IP host has a direct connection to
either the destination telephone number or a PBX.
(5)
The session application then runs the H.323 session protocol to establish a transmission and a
reception channel for each direction over the IP network. If the call is being handled by a PBX,
the PBX forwards the call to the destination telephone. If RSVP has been configured, the RSVP
reservations are put into effect to achieve the desired quality of service over the IP network.
(6)
The CODECs are enabled for both ends of the connection and the conversation proceeds using
RTP/UDP/IP as the protocol stack.
(7)
When either end of the call hangs up, the RSVP reservations are torn down (if RSVP is used)
and the session ends. Each end becomes idle, waiting for the next off-hook condition to trigger
another call setup.
Model Name
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