This form is used to configure the VoIP setting applied to calls on a Legacy SIP DECT Line
These settings are not mergeable. Changes to these settings requires a reboot of the system.
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Field
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Description
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IP Address
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Default = Blank.
The IP address of the SIP DECT extension.
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Codec Selection
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Default = Custom
This field defines the codec or codecs offered during call setup. The codecs available to be used are set through .
The Codec Selection option allows specific configuration of the codec preferences to be different from the system Default Selection list. When Custom is selected, the list can be used to select which codecs are in the Unused list and in the Selected list and to change the order of the selected codecs. The D100 Base Station supports only G711 codecs.
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Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP connection. This field is not shown on Linux based platforms.
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Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM interface. This field is not shown on Linux based platforms.
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DTMF Support
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Default =RFC2833
The D100 Base Station supports only RFC2833.
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VoIP Silence Suppression
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Default = Off
When selected, this option will detect periods of silence on any call over the line and will not send any data during those silent periods. This feature is not used on IP lines using G.711 between systems. On trunk's between networked systems, the same setting should be set at both ends.
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Local Hold Music
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Default = Off
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Allow Direct Media Path
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Default = On
This settings controls whether IP calls must be routed via the system or can be routed alternately if possible within the network structure.
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If enabled, IP calls can take routes other than through the system. This removes the need for a voice compression channel. Both ends of the calls must support Direct Media and be using the same protocol (H.323 or SIP). Enabling this option may cause some vendors problems with changing the media path mid call.
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If disabled or not supported at on one end of the call, the call is routed via the system. RTP relay support allows calls between devices using the same audio codec to not require a voice compression channel.
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Reinvite Supported
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Default = Off.
When enabled, Re-Invite can be used during a session to change the characteristics of the session. For example when the target of an incoming call or a transfer does not support the codec originally negotiated on the trunk. Requires the ITSP to also support Re-Invite.
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