Table 27: Global Policy Parameters->Phone Settings - Grandstream Networks UCM6200 User Manual

NTP Server
Time Zone
Default Call Settings
Dial Plan
Enable Call Features
Use # as Dial Key
Auto Answer by Call-
info
NAT Traversal
Use Random Port
LCD.
Configure the URL or IP address of the NTP server. The SIP end device may
obtain the date and time from the server.
Configure the time zone used on the SIP end device.
Table 27: Global Policy Parameters->Phone Settings
Configure the default dial plan rule. For syntax and examples, please refer to
user manual of the SIP devices to be provisioned for more details.
When enabled, "Do Not Disturb", "Call Forward" and other call features can be
used via the local feature code on the phone. Otherwise, the ITSP feature code
will be used.
If set to "Yes", pressing the number key "#" will immediately dial out the input
digits.
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 Call-
Info header sent from the server/proxy.
The default setting is enabled.
Configure which NAT traversal mechanism will be enabled on the endpoint
device.
If set to "STUN" and STUN server is configured, the phone system will
periodically send STUN message to the SUTN server to get the public IP
address of its NAT environment and keep the NAT port open. STUN will not
work if the NAT is symmetric type.
If set to "Keep-alive", the phone system will send the STUN packets to
maintain the connection that is first established during registration of the
phone. The "Keep-alive" packets will fool the NAT device into keeping the
connection open and this allows the host server to send SIP requests directly
to the registered phone.
If it needs to use OpenVPN to connect host server, it needs to set it to "VPN".
If the firewall and the SIP device behind the firewall are both able to use
UPNP, it can be set to "UPNP". The both parties will negotiate to use which
port to allow SIP through.
The default setting is "Keep-alive".
Configure whether to allow the endpoint device to use random ports for both
SIP and RTP messages. This is usually necessary when multiple phones are
behind the same full cone NAT. The default setting is "No". Note: This
parameter must be set to "No" for Direct IP Calling to work.
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