DPH-80 Phone User Manual
Using
the SIP Phone
•
Mute (Red): Steady on to indicate Mute status; off means normal
Tones
D-Link SIP phone plays the following tones depending on phone current state. D-Link
SIP supports different type of tones for different country selected through configuration.
•
Dial tone
•
Call progress tone
•
Ring back tone
•
Busy tone
•
Call alert (ringing) tone
•
Error tone
•
DTMF tones for all numeric keys
SIP phone Features
•
MUTE: When pressed the SIP phone turns off the microphone signal from the
handset but still play voice from other party.
•
HOLD: When pressed, the SIP phone disconnects both microphone and
speaker while connection is kept alive. No voice packets are transmitted from
the D-Link SIP phone. Hold LED is on. User may press again to release the
call. This feature requires support from the remote phone for proper
functioning.
•
REDIAL: When pressed, redials last dialed number.
•
TRANSFER: Toggle the hook-switch quickly to flash the call. The SIP phone
plays dial tone. Then dial the new party’s number to transfer the call. The SIP
phone transfers the call and plays busy tone. Flashing the hook twice before
dialing the number will restore the call to normal state (to call active state).
•
SPEAKER: One-touch dialing key. When pressed, speaker LED is on and
speaker itself is on while on-hook. If the user off-hooks after dialing or presses
the button again, one-touch operation is terminated, LED and speakers turn
off.
Algorithms
Codecs:
D-Link SIP phones supports G.711 U/A law, G.723.1, and G.729AB. The
browser configuration allows selecting the codec and their priority.
Voice activity detection, silence suppression, and comfort noise generation.
The VAD can
be disabled in the configuration irrespective of the codec being used.
Adjustable Jitter Buffer:
D-Link SIP phones uses a robust adaptive jitter buffer
algorithm. It can be disabled and a fixed size jitter buffer can be used instead through
configuration.
Other features
Remote software upgrade:
A predefined key sequence will download the SIP phone
software and restart the phone. The SIP phone should have been configured with right
TFTP server IP address.
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