Display Name  – set the name to be displayed as CLIP on the called party's phone.

Phone Number (ID) – set your device phone number (or another unique ID composed of characters and digits). Together with the domain, this number uniquely identifies the device in calls and registration.

Domain – set the domain name of the service with which the device is registered. Typically, it is equivalent to the SIP Proxy or Registrar address.

Test Call – display a dialog box enabling you to make a test call to a selected phone number, see below.

Authentication ID – set the alternative user ID for device authentication.

Password – set the device authentication password. If your PBX requires no authentication, the parameter will not be applied.

Proxy Address – set the SIP Proxy IP address or domain name.

Proxy Port – set the SIP Proxy port (typically 5060).

Backup Proxy Address – set the backup SIP Proxy IP address or domain name. The address is used where the main proxy fails to respond to requests.

Backup Proxy Port – set the backup SIP Proxy port (typically 5060).

Registration Enabled – enable device registration with the set SIP Registrar.

Registrar address – set the SIP Registrar IP address or domain name.

Registrar Port – set the SIP Registrar port (typically 5060).

Backup Registrar Address – set the backup SIP Registrar IP address or domain name. The address is used where the main Registrar fails to respond to requests.

Backup Registrar Port – set the backup SIP registrar port (typically 5060).

Registration Expires – set the registration expiry, which affects the network and SIP Registrar load by periodically sent registration requests. The SIP Registrar can alter the value without letting you know.

Registration State – display the current registration state (Unregistered, Registering..., Registered, Unregistering...).

Failure Reason – display the reason for the last registration attempt failure: the registrar’s last error reply, e.g. 404 Not Found.

SIP Transport Protocol – set the SIP communication protocol: UDP (default), TCP or TLS.

Lowest Allowed TLS Version – set the lowest TLS version to be accepted for device connection.

Verify Server Certificate – verify the SIP server public certificate against the CA certificates uploaded in the device.

Client Certificate – specify the client certificate and private key used for verifying the intercom’s authority to communicate with the SIP server.

Local SIP Port – set the local port for the device for SIP signaling. A change of this parameter will not be applied until the device is restarted. The default value is 5060.

PRACK Enabled – enable the PRACK method for reliable confirmation of SIP messages with codes 101–199.

REFER Enabled – enable call forwarding via the REFER method.

Send Keep Alive Packets – set that the device shall inquire periodically about the state of the called station via SIP OPTIONS requests during the call (used for station failure detection during the call).

IP Address Filter Enabled – enable the blocking of SIP packet receiving from addresses other than SIP Proxy and SIP Registrar. The primary purpose of the function is to enhance communication security and eliminate unauthorized phone calls.

Receive Encrypted Calls Only (SRTP) – set that SRTP encrypted calls shall only be received on this account. Unencrypted calls will be rejected. At the same time, TLS is recommended as the SIP transport protocol for higher security.

Encrypted Outgoing Calls (SRTP) – set that outgoing calls shall be SRTP encrypted on this account. At the same time, TLS is recommended as the SIP transport protocol for higher security.

Use MKI in SRTP Packets – enable the use of MKI (Master Key Identifier) if required by the counterparty for master key identification when multiple keys rotate in the SRTP packets.

Do Not Play Incoming Early Media – disable playing of the incoming audio stream before the call sent by some PBXs or other devices is picked up (early media). A standard local ringtone is played instead.

QoS DSCP Value – set the SIP packet priority in the network. The set value is sent in the TOS (Type of Service) field in the IP packet header. Enter the value as a decimal number. A change of this parameter will not be applied until the device is restarted.

STUN Enabled – enable STUN functionality for the SIP account. Address and ports acquired from the configured STUN server will be used in SIP headers and SDP media negotiation.

STUN server address – set the IP address of the STUN server that will be used for this SIP account.

STUN server port – set the port of the STUN server that will be used for this SIP account.

External IP Address – set the public IP address or router name to which the device is connected. If the device IP address is public, leave this parameter empty.

Starting RTP Port – set the initial local RTP port in the range of 64 ports used for audio and video transmission. The default value is 4900 (i.e. the range is 4900–4963). The parameter is only set for SIP 1 but applies to both the SIP accounts.

RTP Timeout – set the audio stream RTP packet receiving timeout during a call. If this limit is exceeded (RTP packets are not delivered), the call will be terminated by the device. Enter 0 to disable this parameter. The parameter is only set for account 1 but applies to both the SIP accounts.

Broadsoft Compatibility Mode – set the Broadsoft PBX compatibility mode. Having received re-invite from a PBX in this mode, the intercom replies by repeating the last sent SDP with currently used codecs instead of sending a complete offer.

Rotate SRV Records – allow SRV record rotation for SIP Proxy and Registrar. This is an alternative method of transition to backup servers in the event of main server failure or unavailability.