to keep alive. If the type is UDP, the phone will send
UDP message to server to keep alive every NAT Keep
Alive Period(s).
Keep Alive Interval
Set examining interval of the server, default is 60
seconds
User Agent
Set the user agent if have, the default is VoIP Phone 1.0
DTMF Type
Select DTMF sent mode, there are three modes:
DTMF_RELAY
DTMF_RFC2833
DTMF_SIP_INFO
DTMF_AUTO
Different VoIP Service providers may provide different
modes.
Local Port
Set sip port of each line
Ring Type
Set ring type of each line
Enable Via Rport
Enable/Disable system to support RFC3581. Via rport is
special way to realize SIP NAT.
Enable PRACK
Enable or disable SIP PRACK function, suggest use the
default config.
Enable Long Contact
Set more parameters in contact field; connection with
SEM server
Convert URI
Convert # to %23 when send the URI.
Dial Without Registered Set call out by proxy without registration;
Ban Anonymous Call
Set to ban Anonymous incoming Call;
Enable DNS SRV
Support DNS looking up with _sip.udp mode
Server Type
Select the special type of server which is encrypted, or
has some unique requirements or call flows.
RFC Protocol Edition
Select SIP protocol version to adapt for the SIP server
which uses the same version as you select. For example,
if the server is CISCO5300, you need to change to
RFC2543; else phone may not cancel call normally.
System uses RFC3261 as default.
Transport Protocol
Set transport protocols, TCP or UDP or TLS;
RFC Protocol Edition
Set Anonymous call out safely; Support RFC3323and
RFC3325;
Keep Authentication
Enable/Disable Keep Authentication System will take
the last authentication field which is passed the
authentication by server to the request packet. It will
decrease the server’s repeat authorization work, if it is
enable.
