User manual

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 sending mode, there are three modes:
 DTMF_RELAY
 DTMF_RFC2833
 DTMF_SIP_INFO
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 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;
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.
Answer With A Single
Codec
Enable/Disable the function when call is incoming,
phone replies SIP message with just one codec
which phone supports.