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Keep Alive Interval
User Agent
DTMF Type
Local Port
Ring Type
Enable Rport
Enable PRACK
Enable Long Contact
Convert URI
Dial Without
Registered
Ban Anonymous Call
Enable DNS SRV
Server Type
RFC Protocol Edition
Transport Protocol
Anonymous Call
Edition
Keep Authentication
Answer With A
Single Codec
Auto TCP
Enable Strict Proxy
Set examining interval of the server, default is 60
seconds.
Set the user agent if have, the default is VoIP Phone
1.0.
Select DTMF sending mode, there are three modes:
DTMF_RELAY
DTMF_RFC2833
DTMF_SIP_INFO
Different VoIP Service providers may provide
different modes.
Set sip port of each line.
Set ring type of each line.
Enable/Disable system to support RFC3581. rport is
special way to realize SIP NAT.
Enable or disable SIP PRACK function, suggest use
the default config.
Set more parameters in contact field; connection
with SEM server.
Convert # to %23 when send the URI.
Set call out by proxy without registration;
Set to ban Anonymous Call;
Support DNS looking up with _sip.udp mode.
Select the special type of server which is encrypted,
or has some unique requirements or call flows.
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.
Set transport protocols, TCP or UDP;
Set Anonymous call out safely; Support
RFC3323and RFC3325;
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.
Enable/Disable the function when call is incoming,
phone replies SIP message with just one codec
which phone supports.
Set to use automatically TCP protocol to guarantee
usability of transport as message is above 1300 byte
Support the special SIP server-when phone receives
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