Matrix ETERNITY NE System Manual 151
22. To change the Transmit Gain of the Speakerphone MIC Volume, set Speaker Transmit Volume to the
desired level, from 0 to 7. Default: 4.
23. To change the Receive Gain of the Speakerphone MIC Volume, set Speaker Receive Volume to the
desired level, from 0 to 7. Default: 4.
24. To use a Headset with the IP phone, set Headset Connected? to Yes. Default: No.
Make sure that you connect a Headset to the IP phone, if you select Yes.
25. Select the Auto Answer check box to enable this feature on the IP phone. Default: Disabled.
When you set the “Auto Answer” feature on the IP phone, 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.
26. 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 IP phone should wait before going OFF-Hook to auto
answer a call. The range of this timer is 1 to 9 seconds. Default: 1 second.
27. 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.
28. Set the Back Light Off Timer (sec) to the desired value, if required, from 000 to 999 seconds. Default: 10
seconds.
29. Set the LCD Contrast Level to a level from 1 to 4 that is comfortable to you. Default: 3.
30. Define SIP/RTP Ports:
• SIP 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 SIP messages to the remote peer. The valid range for
this port is 1024-65534. Default: 5060.
• 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.
31. Set the SIP Quality of Service (QoS) for SIP signaling as:
• SIP DiffServe/ToS. Valid range is 00 to 63. Default: 26.
OR
• RTP DiffServe/ToS. Valid range is 00 to 63. Default: 46.
32. If the IP phone is connected behind a NAT router, configure NAT Keep Alive.
• Select the check box Enable NAT Keep Alive to send Keep Alive messages periodically to refresh the
binding in the NAT router. Default: Disabled.
• Define as Interval (sec), the time period, from 001 to 999 seconds, after which the phone should send
Keep Alive message. Default: 120 seconds.
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