Grandstream GXP21xx Series User Manual - Manual (Page 59)

FIRMWARE VERSION 1.0.8.4 GXP2120/GXP2110/GXP2100/GXP14xx USER MANUAL Page 57 of 85
Ring Timeout
Defines the timeout (in seconds) for the rings on no answer. The default setting
is 60 seconds.
Send Anonymous
If set to "Yes", the "From" header in outgoing INVITE messages will be set to
anonymous, essentially blocking the Caller ID to be displayed.
Anonymous Call
Rejection
If set to "Yes", anonymous calls will be rejected. The default setting is "No".
Auto Answer
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep.
Allow Auto Answer by
Call-Info
If set to "Yes", the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep, based on the SIP info
header sent from the server/proxy. The default setting is "No".
Refer-To Use Target
Contact If set to "Yes", the "Refer-To" header uses the transferred target's Contact
header information for attended transfer. The default setting is "No".
Transfer on Conference
Hangup Defines whether or not the call is transferred to the other party if the initiator of
the conference hangs up. The default setting is "No".
No Key Entry Timeout (s)
Defines the timeout (in seconds) for no key entry. If no key is pressed after the
timeout, the digits will be sent out. The default value is 4 seconds.
Use # as Dial Key
Allows users to configure the "#" key as the "Send" key. If set to "Yes", the "#"
key will immediately dial out the input digits. In this case, this key is essentially
equivalent to the "Send" key. If set to "No", the "#" key is included as part of the
dialing string.
Hold Method
Specifies Hold method to Auto or RFC3264. Auto is to satisfy both rfc3264 and
rfc2xxx for the maximum capability.
DND Call Feature On Configures feature code to turn on DND
DND Call Feature Off Configures feature code to turn off DND
SETTINGS PAGE DEFINITIONS
Settings -> General Settings
Local RTP Port
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.
Use Random Port
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).