P31003C1020M1000276A9, 01/2018
44 OpenScape Cordless IP V2, Administrator Documentation
Provider and PBX profiles
Configuring telephony server profiles
The RFC3263 rules will be used to locate SIP servers and select them for load balancing and redundancy.
Outbound proxy port is a fixed number:
The usage of DNS SRV records according to RFC3263 is blocked.
SIP SUBSCRIBE for Net-AM MWI
When activated a subscription is established for the purpose of receiving notifications about new messages on the
network mailbox.
Enable/disable SIP subscription via the radio boxes next to SIP SUBSCRIBE for Net-AM MWI.
DTMF over VoIP Connections
DTMF signalling (Dual Tone Multi Frequency) is required, for example, for querying and controlling certain network
mailboxes via digit codes, for controlling of automatic directory enquiries or for remote operation of the local
answering machine.
To send DTMF signals via VoIP, you must define how key codes should be converted into and sent as DTMF sig-
nals: as audible information via the speech channel or as a "SIP Info" message.
Ask your VoIP provider which type of DTMF transmission it supports.
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 PT for 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
Specify the DTMF signalling type explicitly: select No Select the send settings for DTMF transmission.
Send settings of DTMF transmission
Make the required settings for sending DTMF signals:
Settings for codecs
The voice quality of VoIP calls is mainly determined by the codec used for the transmission and the available band-
width 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.
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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