Grandstream HT503 User Manual - Manual (Page 58)

FIRMWARE VERSION 1.0.14.1 HT503 USER MANUAL Page 56 of 64
The user should know the frequency values and cadences of these tones.
Here is an example for the syntax for a busy tone in the U.S.A:
(Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3;)
(Note: freq: 0 - 4000Hz; vol: -30 - 0dBm)
(Default: Busy Tone - f1=480@-24,f2=620@-24,c=500/500;)
Note : Maximum supported cadences is 3
AC Termination Model You can select the AC termination by Country or by Impedance.
Country-Based 15 Countries are selectable in this version of the F/W.
Impedance-Based Select the Impedance used by the PSTN service provider.
Number of Rings Default is 4. This setting specifies number of phone rings (on the phone connected to
the FXS port) before a PSTN incoming call is bridged to VoIP
Note: The number of rings feature serves as a PSTN answer delay, and should be set
to a larger value to allow enough time for the HT503 to decode the Caller ID signal set
by the central office.
PSTN Ring Thru FXS If Yes, the phone connected to the FXS port will ring a configured amount of times (see
above). If not, the phone connected to the FXS port will not ring.
PSTN Ring Thru Delay
(sec)
If the PSTN Ring Thru Delay is set to Yes, all incoming PSTN calls through FXO will
ring the phone connected to the FXS port, after this delay or after caller id is detected
(whichever comes first).
PSTN Ring Timeout (sec) Range is 2-10 seconds. Default is 6 seconds. Option is used to detect PSTN
hang up when FXO port is not answered.
DTMF Digit Length (ms) Digit length and Dial Pause are port digit dialing configurations; FXO needs to dial out
digits for VOIP to PSTN 1 stage calls, and unconditional call forward to PSTN, and
route to PSTN. Digit Length is the play time for each digit.
Note: In order to receive the caller ID information, the delay should be set to a value
larger than the delay required to complete the PSTN caller ID delivery.
DTMF Dial Pause (ms) Dial pause is the time between 2 digits for the same scenario as explained above.
First Digit Timeout (sec) Used for PSTN to VoIP calls. PSTN users need to enter the FIRST digit within the first
digit timeout period. Otherwise the call will be dropped.
Inter Digit Timeout When dialing from the PSTN to VoIP, subsequent digits have to be input within the
period of inter-digit timeout. Otherwise the dial plan thinks it is the end of the digit input.
Wait for Dial Tone Wait for Dial tone is used for one stage VoIP to PSTN calls. If set to Yes, the device will
first obtain a PSTN line and a dial tone from a central office. After obtaining the dial
tone, the digits dialed will be sent to the central office.
Stage Method (1/2) This configuration is applicable for VoIP to PSTN calls and indicates one or two stage
dialing methods.