Help > Using the Web Interface > Identities > Options for experts
Options for experts
Fallback for
Determines whether the identity has a fallback identity or is itself a fallback identity.
No fallback
There is no fallback provided for the identity.
The identity is not a fallback identity.
Identity
The identity is fallback identity for the selected identity.
Display only:Identity has a fallback
For the identity, the displayed identity is intended as a fallback.
The identity itself can no longer be selected as a fallback identity.
*Further help under Identities
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.
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Pickup code
This is required to perform a call pickup. Enter the character string stored on the PBX/at the provider, e.g.##06 for Auerswald PBXs.
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Music on hold
If a connection or call is on hold, the "music on hold" is played.
*Note: If "Music On Hold" has already been enabled on the PBX/at the provider, this function can remain switched off.
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IP version
IPv4
Sets IPv4 for the registrar.
IPv6
Sets IPv6 for the registrar.
Auto
Automatically sets the protocol used by the registrar.
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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.
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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.
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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.
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Certificate
If the SIPS function is activated, the system checks whether the certificate belongs to the domain/IP.
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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.
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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.
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Protocol type
UDP
(the User Datagram Protocol) is 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) sends data packets individually, segmented from a certain size and until receipt has been confirmed.
*Important: If encryption using SIPS is enabled, the TCP transport protocol is used. Manual settings are overwritten.
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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.
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Retry subscriptions
Sets the interval at which attempts are made to configure a subscription on the PBX/provider, if an error occurs.
•1x
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.
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Retry subscriptions: Interval in seconds
Sets the time gap between two subscription attempts.
Minimum: 10 sec
Default: 180 sec
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Retry register
Sets the interval at which attempts are made to register on the PBX/provider, if an error occurs.
•1x
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.
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Retry register: Interval in seconds
Sets the time gap between two subscription attempts.
Minimum: 1 sec
Default: 10 sec
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Ringer Tone
Assigns a ringtone to the identity.
*Further help under Sound
Name sources
Active
The sequence in which the name sources are displayed in the list sets the sequence in which they are applied. The first source that contains a name is used for the display.
Inactive
You can select inactive name sources and drag and drop them into the Active list to arrange them, or remove them from the list.
Reset to default
Returns the list to its default state.
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DTMF method
Specifies the DTMF method used to transmit signals.
•RTP event
Transmission of event packets in the RTP stream
•Inband
Transmission of coded sound signals, directly in the RTP stream
•SIP info
Transmission of SIP info messages
*Further help under DTMF