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Phone Number
Input the phone number assigned by your VoIP service provider.
Phone will not register if there is no phone number configured.
Display Name
Set the display name.
Proxy Server Address
Set proxy server IP address
(
Usually, Register SIP Server
configuration is the same as Proxy SIP Server. But if your VoIP
service provider give different configurations between Register
SIP Server and Proxy SIP Server, you need make different
settings.
)
Proxy Server Port
Set your Proxy SIP server port.
Proxy Username
Input your Proxy SIP server account.
Proxy Password
Input your Proxy SIP server password.
Domain Realm
Set the sip domain if needed, otherwise this VoIP phone will use
the Register server address as sip domain automatically. (Usually
it is same with registered server and proxy server IP address).
Enable Register
Start to register or not by selecting it or not.
Register Expire Time
Set expire time of SIP server register, default is 60 seconds. If the
register time of the server requested is longer or shorter than the
expire time set, the phone will change automatically the time into
the time recommended by the server, and register again.
NAT 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
Signal Key
Set the key for signal encryption
Media Key
Set the key for RTP encryption
Local port
Set sip port of each line
Ring type
Set ring type of each line
Subscribe Expire Time
Set the interval of Subscribe.
Conference Number
Set the server conference number to join the room
Enable DNS SRV
Support DNS looking up with _sip.udp mode
Enable Subscribe
Enable Subscribe.
Enable Keep
Authentication
Enable/Disable Keep Authentication.
NAT Keep Alive
Enable/Disable keeps NAT of SIP alive.
If some server refuse to register with too short interval time, and
has no packets sending to device in private network to keep NAT
alive, user could set this function ON. It need set the keep alive
interval time less than the NAT server’s.
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.
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;
Forward Type
Select call forward mode, the default is Off
Off
:
Close down calling forward
Busy
:
If the phone is busy, incoming calls will be forwarded
to the appointed phone.
No answer
:
If there is no answer, incoming calls will be
forwarded to the appointed phone.
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.