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 calls between IP endpoints and/or lines must go through the IP Office.
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If disabled, calls go through the IP Office and use its resources. RTP relay allows calls between devices using the same audio codec to not require a voice compression channel.
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If enabled, calls can take routes other than through the IP Office system. Both ends must support direct media and have matching VoIP settings. For example, both ends must use the same protocol (SIP or H.323), same addressing (IPv4 or IPv6), and so on. Otherwise, the call goes through the IP Office system.
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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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