Grandstream Networks GXP1610 Administration Manual page 31

Settings -> General Settings
Local RTP Port
Use Random Port
Keep-alive Interval
Use NAT IP
STUN Server
Public Mode
Settings -> Call Features
Off-hook Auto Dial
Off-hook Timeout
Intercom User ID
Bypass Dial Plan
through Call History
and Directories
Disable Call Waiting
Disable Call Waiting
Tone
Disable Direct IP Call
Use Quick IP Call mode
Disable Conference
Disable in-call DTMF
Display
Enable Sending DTMF
via specific MPKs
This parameter defines the local RTP port used to listen and transmit. It is the
base RTP port for channel 0. When configured, channel 0 will use this port
_value for RTP; channel 1 will use port_value+2 for RTP. Local RTP port
ranges from 1024 to 65400 and must be even. The default value is 5004.
When set to "Yes", this parameter will force random generation of both the local
SIP and RTP ports. This is usually necessary when multiple phones are behind
the same full cone NAT. The default setting is "Yes" (This parameter must be
set to "No" for Direct IP Calling to work).
Specifies how often the phone sends a blank UDP packet to the SIP server in
order to keep the "ping hole" on the NAT router to open. The default setting is
20 seconds.
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.
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.
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.
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".
If configured, when the phone is onhook, it will go offhook after the timeout (in
seconds). The default value is 30 seconds.
Configures the intercom extension number for account 1 to dial out. This User
ID is mapped to the INTERCOM button on the phone.
Enable/disable dial plan check while dialing through the call history and any
phonebook directories.
Disables the call waiting feature. The default setting is "No".
Disables the call waiting tone when call waiting is on. The default setting is "No".
Disables Direct IP Call. The default setting is "No".
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".
Disables the Conference function. The default setting is "No".
When it's set to "Yes", the DTMF digits entered during the call will not display.
The default setting is "No".
Allows certain MPKs to send DTMF in-call. This option does not affect Dial
DTMF.
GXP1610/GXP1620/GXP1625/GXP1628/
GXP1630 Administration Guide
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