Help > Using the Web Interface > Identities > Options for experts
Options for experts
Fallback for
No fallback
No fallback is possible for the current identity.
Configured/Unconfigured identity
Sets the main identity for which the fallback identity applies.
Configured identity: displays Display name
Configured identity: displays Identity + seq. no.
*Notices:In the case of a main identity, the field is greyed out and displays the entered fallback identity. A fallback identity can also be created for identities that have not yet been configured.
*Further help under Fallback identity
CLIR type
(Number suppression)
Area in the From header in which the VoIP provider expects to receive number presentation suppression. Selection as set in the PBX/at the provider.
Anonymous
The sent display text in the From header is "anonymous".
User anonymous
Both the display text and the User name section in the From header is "anonymous".
*Further help under Anonymous call
Voicemail number
Enter the voicemail number assigned by the VoIP provider or the voicemail number entered in the PBX.
*Further help under Voicemail number
Pick-up code
This is required to perform a call pick-up. Enter the character string stored on the PBX/at the provider, e.g.##06 for Auerswald PBXs.
*Further help under Pick-Up
Music on hold
If a connection or call is on hold, the "music on hold" is played.
*Further help under Music on hold
IP version
IPv4
Sets IPv4 for the registrar.
IPv6
Sets IPv6 for the registrar.
Auto
Automatically sets the protocol used by the registrar.
*Further help under Identities
SRTP
Transport protocol for encrypted connections.
Mandatory
This setting forces voice encryption to be on. If the voice partner (VoIP provider, PBX, external VoIP subscriber) does not support SRTP, the connection is not established.
Preferred
Switches on negotiation for the encryption of call data via SRTP. When a call is made, the call partner will be asked if encryption is possible. If selected, voice data is transmitted in encrypted form. If not selected, it is not encrypted.
Disabled
This setting forces voice encryption to be off. If the voice partner (VoIP provider, PBX, external VoIP subscriber) does not support encryption, the connection is not established.
*Further help under SRTP
SIPS
Activates the transmission of encrypted SIP messages over TLS for connections with this identity.
The destination in the invite package header is contacted with an encrypted transmission.
*Note: To create a successful, secure connection, a certificate must be provided for the provided host, if necessary.
*Further help under SIPS
*Further help under Certificate
Peer-to-peer TLS
Forces the encryption of SIP messages over the entire route to the destination.
*Note: The call is not made if TLS is not available on the entire connection.
*Further help under Identities
Certificate
If the SIPS function is activated, the system checks whether the certificate belongs to the domain/IP.
*Further help under Certificate
Session timer
Switches on the check after a connection for a call that is still in existence.
*Note: When the SIP session timer is switched on, this may result in the call being interrupted more frequently after the specified interval, if a VoIP provider has not implemented session renewal properly. In this case, set a different session timeout or disabled the session timer.
*Further help under SIP
Session timeout (in min.)
2 … 255 minutes, default: 15 minutes
Specifies the number of minutes after which the SIP session timer is to check a call's connection.
*Further help under SIP
Protocol type
UDP
(User Datagram Protocol). Used to send data packets over connectionless non-secure communication lines.
*Note: If very large data packets are present, TCP is used instead of UDP. The maximum size of a data packet can vary according to the network. (RFC 3261 > TCP)
TCP
(Transmission Control Protocol). Segments data into packets, from a specified size, and sends these individual data packets to the recipient address until receipt has been confirmed.
*Further help under SIP
*Important: If encryption using SIPS is enabled, the TCP transport protocol is used. Manual settings are overwritten.
Subscription interval (min)
This sets the frequency at which the status of potential changes are queried on the PBX. Default: 45 minutes
The value you enter here should be a compromise between a short interval (which generates traffic) and rapid updates.
*Further help under SIP
Repeat subscription
Sets the interval at which attempts are made to configure a subscription on the PBX/provider, if an error occurs.
•Never
The device sends exactly one subscription to the PBX/provider. No other subscriptions are sent.
•Fixed interval
Attempts to subscribe on the PBX/provider are made at the specified interval.
•Redouble interval
The set number of seconds doubles after each attempt to subscribe on the PBX/provider.
*Further help under SIP
Subscription: Interval in seconds
Sets the time gap between two subscription attempts.
Minimum: 10 sec
Default: 180 sec
*Further help under SIP
Repeat register
Sets the interval at which attempts are made to register on the PBX/provider, if an error occurs.
•Never
The device makes exactly one attempt to register on the PBX/provider. There are no more registration attempts.
•Fixed interval
Attempts to register on the PBX/provider are made at the specified interval.
•Redouble interval
The set number of seconds doubles after each attempt to register on the PBX/provider.
*Further help under SIP
Register: Interval in seconds
Sets the time gap between two registration attempts.
Minimum: 1 sec
Default: 10 sec