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Template Module, Version 2.1, 11.08.2020 / Template Basiseinstellungen: Version 3.1 - 07.07.2020
Provider and PBX profiles
Gigaset N870 IP PRO / Admin IE-UK-International en / A31008-XXXXX-XXXX-X-XX / settings_provider.fm / 8/10/21
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Automatic negotiation of DTMF transmission
For each call, the phone attempts to set the appropriate DTMF signalling type for the codec
currently being negotiated: select Yes.
The system will use the transmission method matching best the received capabilities from
the peer based on the following priority order:
send via RFC2833, if the PT (Payload Type) for the telephone event is provided by the peer
send via SIP INFO application/dtmf-relay, if SIP INFO method is supported by the peer
send in-band audio
No automatic attempts to set DTMF transmission type: select No (DTMF transmission type is
Audio by default).
Send settings of DTMF transmission
Make the required settings for sending DTMF signals:
Connection ringtones
Different ringtones are possible for internal, external, group, door, emergency and optional calls.
Prerequisite: The provider/platform sends the correct information via the Alert-Info field in the
SIP header.
The user can select different ringtones for specific calls at the handset. You can here define which
different ringtones are allowed to be set by the user.
In the Name field enter the name for the menu entry that should be shown in the handset
menu.
Note: Internal calls cannot be changed.
The Alert-Info pattern field contains the Info-Alert definition that must be contained in the
SIP header to identify the appropriate call type.
Field is empty: The entry is not shown in the handset menu.
Settings for codecs
The voice quality of VoIP calls is mainly determined by the codec used for the transmission and
the available bandwidth of your network connection. A "better" codec (better voice quality)
means more data needs to be transferred, i.e. it requires a network connection with a larger
bandwidth. You can change the voice quality by selecting the voice codecs your phone is to use,
and specifying the order in which the codecs are to be suggested when a VoIP connection is
established. Default settings for the codecs used are stored in your phone; one setting optimised
for low bandwidths and one for high bandwidths.
Both parties involved in a phone connection (caller/sender and recipient) must use the same
voice codec. The voice codec is negotiated between the sender and the recipient when estab-
lishing a connection.
Audio or RFC 2833 DTMF signals are to be transmitted acoustically (in voice packets).
SIP Info DTMF signals are to be transmitted as code.
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