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Grandstream GXP1610 Administration Guide User Manual

Page 30

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GXP1610/GXP1620/GXP1625/GXP1628

Administration Guide

Page 29 of 49

order to keep the "ping hole" on the NAT router to open. The default setting is
20 seconds.

Use NAT IP

The NAT IP address used in SIP/SDP messages. This field is blank at the
default settings. It should ONLY be used if it's required by your ITSP.

STUN Server

The IP address or Domain name of the STUN server. STUN resolution results
are displayed in the STATUS page of the Web GUI. Only non-symmetric NAT
routers work with STUN.

Public Mode

Configures to turn on/off public mode for hot desking feature on the phone. If
set to "Yes", users would need fill in the SIP Server address for account 1 as
well. Then reboot the phone. When the phone boots up, users will require
entering SIP User ID and Password on the LCD to login and use the phone.

Note:
When the phone is in public mode login screen, press CONF button will have
the IP address of the phone displayed.

Settings -> Call Features

Off-hook Auto Dial

Configures a User ID/extension to dial automatically when the phone is
offhook. The phone will use the first account to dial out. The default setting is
"No".

Off-hook Timeout

If configured, when the phone is onhook, it will go offhook after the timeout (in
seconds). The default value is 30 seconds.

Intercom User ID

Configures the intercom extension number for account 1 to dial out. This User
ID is mapped to the INTERCOM button on the phone.

Disable Call Waiting

Disables the call waiting feature. The default setting is "No".

Disable Call Waiting
Tone

Disables the call waiting tone when call waiting is on. The default setting is
"No".

Disable Direct IP Call

Disables Direct IP Call. The default setting is "No".

Use Quick IP Call mode

When set to "Yes", users can dial an IP address under the same LAN/VPN
segment by entering the last octet in the IP address. To dial quick IP call,
offhook the phone and dial #XXX (X is 0-9 and XXX <=255), phone will make
direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP
address REGARDLESS of subnet mask. #XX or #X are also valid so leading
0 is not required (but OK). No SIP server is required to make quick IP call.
The default setting is "No".

Disable Conference

Disables the Conference function. The default setting is "No".

Disable in-call DTMF
Display

When it's set to "Yes", the DTMF digits entered during the call will not display.
The default setting is "No".

Enable Sending DTMF
via specific MPKs

Allows certain MPKs to send DTMF in-call. This option does not affect Dial
DTMF.

Mute Key Functions
While Idle

When set to “DND”, the DND will be enabled for future incoming call if
pressing MUTE key in idle state; If this feature is set to “Idle Mute”, MUTE key
will take effect in idle state and future incoming call will be answered with
mute; Otherwise, MUTE key will not take effect in idle state.

Disable Transfer

Disables the Transfer function. The default setting is "No".

In-call dial number on
pressing transfer key

Configures the number for the phone to dial as DTMF during the call using
TRAN button.

Auto-Attended Transfer

If set to "Yes", the phone will use attended transfer by default. The default
setting is "No".

Do Not Escape #
as %23 in SIP URI

Specifies whether to replace # by %23or not for some special situations. The
default setting is "No".