Grandstream Networks GXP1100 User Manual page 39

Local RTP Port
Use Random Port
Keep-alive Interval
Use NAT IP
STUN Server
Settings -> Call Features
Off-hook Auto Dial
Off-hook Timeout
Disable Call Waiting
Disable Call Waiting Tone
Disable Direct IP Call
Use Quick IP Call mode
Disable Conference
Enable sending DTMF via
Speed Dial
FIRMWARE VERSION 1.0.8.4
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 a User ID/extension to dial automatically when the phone is
off hook. The phone will use the first account to dial out. The default
setting is "No".
If configured, when the phone is on hook, it will go off hook after the
timeout (in seconds). The default value is 30 seconds.
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, off hook 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".
Enables Multi Purpose Key to send DTMF during the call. The default
setting is "No".
GXP1100/GXP1105 USER MANUAL Page 37 of 53