Kingtel KT52IP, User Manual

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Code will disallow the phone to make anonymous calls.
Ban Anonymous Off
Code Allow Anonymous Calling function described above. In other
words “Anonymous” will be transmitted for Caller ID.
Keep Alive Type 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.
Keep Alive Interval Set the NAT Keep Alive Interval. Default is 60 seconds
User Agent Set SIP User Agent value.
DTMF SIP INFO Mode You can chose Send 10/11 or Send */#
DTMF Type DTMF sending mode. There are four modes:
 In-band
 RFC2833
 SIP_INFO
 AUTO
Different VoIP Service providers may require different modes.
Local port SIP port. Default is 5060.
Ring type Set ring tone. There are 9 standard options and 3 user options.
Enable Rport Enable/Disable support for NAT traversal via RFC3581 (Rport).
Enable PRACK Enable or disable SIP PRACK function. Default is OFF. It is
suggested this be used.
Enable Long Contact Allow more parameters in contact field per RFC 3840
Convert URI Converts # to %23 when sending URI information.
Dial Without Registered Allow outgoing calls without registration.
Ban Anonymous Call Refuse Anonymous Calls
Enable DNS SRV Enables use of DNS SRV records
Enable Missed Call Log If enabled, the phone will save missed calls into the call history
record.
Server Type Configures phone for unique requirements of selected server.
RFC Protocol Edition Select SIP protocol version RFC3261 or RFC2543. Default is
RFC3261. Used for servers which only support RFC2543.
Transport Protocol Set transport protocol TCP, UDP or TLS.
Anonymous Call Edition Set privacy support RFC3323, RFC3325 or none
Keep Authentication 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.
Ans. With a Single Codec If enabled phone will respond to incoming calls with only one
codec.
Auto TCP Force the use of TCP protocol to guarantee usability of transport
for SIP messages above 1500 bytes
Enable Strict Proxy Enables the use of strict routing. When the phone receives
packets from the server
，
it will use the source IP address, not the