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Enable PRACK
Enable or disable SIP PRACK function, suggest use the default
config.
Long Contact
Set more parameters in contact field; connection with SEM
server
Enable URI Convert
Convert # to %23 when send the URI.
Dial Without Register
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
Forward Type
Select call forward mode, the default is Off
z
Off
:
Close down calling forward
z
Busy
:
If the phone is busy, incoming calls will be forwarded
to the appointed phone.
z
No answer
:
If there is no answer, incoming calls will be
forwarded to the appointed phone.
z
Always
:
Incoming calls will be forwarded to the appoint
phone directly.
The phone will Prompt the incoming while doing forward.
Forward Phone Number
Appoint your forward phone number.
Server Type
Select the special type of server which is encrypted, or has some
unique requirements or call flows.
DTMF Mode
Select DTMF sending mode, there are three modes:
z
DTMF_RELAY
z
DTMF_RFC2833
z
DTMF_SIP_INFO
Different VoIP Service providers may provide different modes.
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 Privacy Edition
Set Anonymous call out safely; Support RFC3323and RFC3325;
Transfer Expire Time
For the phone supports the transfer of certain special features
server, set interval time between sending “bye” and hanging up
after the phone transfers a call.
Click to Talk
Set click to Talk ( need practical software support).
Signal Encode
Enable/Disable Signal Encrypt.
RTP Encode
Enable/Disable RTP Encrypt.
Enable Session Timer
Set Enable/Disable Session Timer, whether support RFC4028.It
will refresh the SIP sessions.
Answer With 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 recieves the
Summary of Contents for FV6030
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