will disallow the phone to make anonymous calls.
Allow Anonymous Calling function described above. In other
words “Anonymous” will be transmitted for Caller ID.
Specifies the NAT keep alive type. If SIP Option is selected, the
phone will send SIP Option sip messages to the server every NAT
Keep Alive Period. The server will then respond with 200 OK.
If UDP is selected, the phone will send a UDP message to the
server every NAT Keep Alive Period.
Set the NAT Keep Alive Interval. Default is 60 seconds
Set SIP User Agent value.
You can chose Send 10/11 or Send */#
DTMF sending mode. There are four modes:
Different VoIP Service providers may require different modes.
SIP port. Default is 5060.
Set ring tone. There are 9 standard options and 3 user options.
Enable/Disable support for NAT traversal via RFC3581 (Rport).
Enable or disable SIP PRACK function. Default is OFF. It is
suggested this be used.
Allow more parameters in contact field per RFC 3840
Converts # to %23 when sending URI information.
Allow outgoing calls without registration.
Enables use of DNS SRV records
If enabled, the phone will save missed calls into the call history
record.
Configures phone for unique requirements of selected server.
Select SIP protocol version RFC3261 or RFC2543. Default is
RFC3261. Used for servers which only support RFC2543.
Set transport protocol TCP, UDP or TLS.
Set privacy support RFC3323, RFC3325 or none
Enable /disable registration with authentication. It will use the
last authentication field which passed authentication by server.
This will decrease the load on the server if enabled.
If enabled phone will respond to incoming calls with only one
codec.
Force the use of TCP protocol to guarantee usability of transport
for SIP messages above 1500 bytes
Enables the use of strict routing. When the phone receives
packets from the server，it will use the source IP address, not the
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