194 Matrix ETERNITY NE System Manual
• To change the Transmit Gain of the Speakerphone MIC Volume, set Speaker Transmit Volume Level to
the desired level, from 0 to 7. Default: 4.
• To change the Receive Gain of the Speakerphone MIC Volume, set Speaker Receive Volume Level to
the desired level, from 0 to 7. Default: 4.
• To use a Headset with the IP phone, set Headset Connected? to Yes. Default: No.
Make sure that you connect a Headset to the SPARSH VP510, if you enable this option.
• Select the Auto Answer check box to enable this feature on the SPARSH VP510. Default: Disabled.
When you set the “Auto Answer” feature on the SPARSH VP510, the phone goes OFF-Hook automatically
after a preset period of time, without the extension user having to pick up the handset or press the speaker
or headset key. When you enable Auto Answer, you must configure the Auto Answer Timer.
• If you enabled Auto Answer on the phone, set the Auto Answer Timer (sec) to the desired value.
This timer defines the time in seconds that the SPARSH VP510 should wait before going OFF-Hook to
auto answer a call. The range of this timer is 1 to 9 seconds. Default: 1 second.
• Adjust the Backlight brightness of the phone’s LCD display, by setting the LCD Backlight Level to the
desired value, from 1 to 4. Default: 3.
• Set the Back Light Off Timer (sec) to the desired value, if required, from 000 to 9999 seconds. Default:
10 seconds.
• Set the LCD Contrast Level to a level from 1 to 4 that is comfortable to you. Default: 3.
• Select Transport Mode and enable SRTP.
• Select the protocol to be used to transport the SIP messages. You can select the Transport Mode as
TCP or TLS.
• If you select TCP, make sure the SIP Over TCP is selected in VoIP Port Parameters.
• If you select TLS, make sure the SIP Over TLS is selected in VoIP Port Parameters.
• For secure conversations over SIP, select the Enable SRTP? check box. The SIP messages will be
transported over SRTP only.
If you select this check box, make sure you have selected SRTP Mode as Forced or Optional in the
General Parameters under SIP Extension Settings.
• Define RTP Port.
• RTP Listening Port: This is the port on which the IP phone listens for SIP messages over TCP. This
port is also used as the source port for sending RTP packets. This port is also used as the source port
for sending RTP packets to the remote peer. The valid range for this port is 1025-65278. Default: 8000.
• Set the Quality of Service (QoS) for SIP signaling as:
• SIP DiffServe/ToS. Valid range is 00 to 63. Default: 26.
OR
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