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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