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Anonymous Off Code
Set the Anonymous Off Code, When you choose to disable the
anonymous call function on your IP phone, it will send
information to the server, and the server will disable the
anonymous call function for your IP phone automatically.
Keep Alive Type
Specify the keep alive type, if the type is option, the
phone will send option sip message to server every NAT Keep
Alive Period(s), then the server responses with 200 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 Mode
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 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.
Long 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 RFC 2543, else phone may
not cancel call normally. System uses RFC3261 as default.
Transport Protocol
Set transport protocols, TCP or UDP;
Anonymous Call 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.
Ans. With a Single Codec
Enable/Disable the function when call is incoming, phone replies
SIP message with just one codec which phone supports.
Auto TCP
Set to use automatically TCP protocol to guarantee usability of
transport as message is above 1300 byte
Enable Strict Proxy
Support the special SIP server-when phone receives the packets
sent from server
，
phone will use the source IP address, not the
address in via field.
Enable GRUU
Set to support GRUU
Enable Display name
Quote
Set to make quotation mark to display name as the phone sends
out signal, in order to be compatible with server.
Enable user=phone
Enable user=phone by selecting it, it is contained in the invite sip
message, in order to be compatible with server
Enable Missed Call Log
Enable the missed call log by it, the phone will save the missed
call log into the call history record and display the missed calls
on the idle screen, or won’t save the missed call log into the call
history record and display the missed calls on the idle screen.
