Saving The Configuration Changes - Grandstream Networks HT503 User Manual

Current Disconnect
Threshold (ms)
Enable PSTN Disconnect
Tone Detection
PSTN Disconnect Tone
AC Termination Model
Country-Based
Impedance-Based
Number of Rings
PSTN Ring Thru FXS
PSTN Ring Thru Delay
(sec)
DTMF Digit Length (ms)
DTMF Dial Pause (ms)
First Digit Timeout (sec)
Inter Digit Timeout
Wait for Dial Tone
Stage Method (1/2)

Saving the Configuration Changes

After user makes a change to the configuration, press the "Update" button in the Configuration Menu. The
web browser will then display a message window to confirm saved changes.
Grandstream recommends reboot or power cycle the IP phone after saving changes.
Grandstream Networks, Inc.
This is a preconfigured value of duration for a line power drop used by specific service
providers. For example, for a configured value of 500ms the device will ignore any
random voltage drops on the line if duration of such drop is less than 500ms and the
call will NOT be considered as terminated. This is useful to prevent unnecessary call
drops in some low quality PSTN lines.
If set to Yes, arrived Busy Tone is used as the disconnect signal.
In certain countries, the central office will send a special busy tone to indicate when a
call is disconnected from the remote side. User can pre-configure this tone on the
ATA. 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;)
You can select the AC termination by Country or by Impedance.
15 Countries are selectable in this version of the F/W.
Select the Impedance used by the PSTN service provider.
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.
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.
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).
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.
Dial pause is the time between 2 digits for the same scenario as explained above.
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.
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 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.
This configuration is applicable for VoIP to PSTN calls and indicates one or two stage
dialing methods.
HT503 User Manual
Firmware 1.0.1.57
Page 33 of 37
Last Updated: 3/2010