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– enter the opponent IP address (operator, PBX) to which
SIP Proxy IP*
2N SIP
®
should connect for call direction.
Mic
– set the SIP Proxy listening port (typically 5060).
Port
– allow registration with the opponent and specify the phone
Use registration
number for registration (Caller ID). If the device is not registered, no call setup
requests can be set to it.
– choose UDP or TCP for SIP communication.
Transport protocol
– set the opponent IP address (operator, PBX) to
Registration server IP address
which
should connect for registration requests.
2N SIP Mic
®
– set the SIP Registrar server listening port (typically 5060).
Port
– set the user name for
authentication.
Name
2N SIP Mic
®
– set the password for
authentication. If your IP PBX does
Password
2N SIP Mic
®
not request authentication, the parameter is not applied.
– set the registration expiry to control the registration
Registration expiry
request load of your network and SIP Registrar. The SIP Registrar can modify the
expiry without letting you know.
– set the timeout for the
to
Timeout after wrong SIP registration
2N SIP Mic
®
register after an unsuccessful registration attempt.
– STUN server helps NAT clients (i.e. the PCs behind the firewall) set up
STUN/NAT IP
phone calls to the VoIP hosted outside the LAN.
Use local IP –
use the local device IP address.
STUN
– enter the STUN server address (IP or domain name). The
STUN server
address is used if STUN IP is selected in the RTP interface configuration.
– set the port to be used for STUN. The default value is
.
Port
port 3478
STUN, keep alive
– enter the STUN server address (IP or domain name). The
STUN server
address is used if STUN IP is selected in the RTP interface configuration..
– set the port to be used for STUN. The default value is
.
Port
port 3478
– define the keep-alive packet sending period. The default
Refresh rate
