Voip Call Forward - Alcatel-Lucent 8001 User Manual

Play Hang-up Tone
Transfer Code
Conference Exit Result
Return Code When
Refuse
Return Code When DND
Flash Hook Time(<800ms) The time for the flash hook.
Called No Answer Time
Pound Send Method
RFC 2833 Play Load
P-Asserted-Identity
SIP Session Timer T1
SIP Session Timer T2
SIP Session Timer T4
Local SIP Port
RTP Port Range
Affiliated Port
Headset Mode
Ring Type On Seat Mode

6.1.3 VoIP Call Forward

ITEM
Always
If Busy
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Enable the tone for hang up when busy.
The code for transfer feature.
Choose what should happen when the Conference originator hangs up:
either keep the communication between the two other parties, or hang
up the whole conference.
Select the code to feedback to the server when you reject the call.
Select the code to feedback to the server when you a call is rejected
because of your DND function enabled.
When you receive an incoming call while this feature is enabled, and if
you don't answer the call, the caller will receive a time out after the
here-defined "Called No Answer Time" has elapsed.
Following RFC2396, the "#" will be interpreted as '#' or as %23 in the SIP
request method, depending on what you define here.
Default is 101, RTP Payload for DTMF Digits, Telephony Tones and
Telephony Signals
Enable/Disable the P-Asserted-Identity feature.
The SIP Session Timer T1 settings.
The SIP Session Timer T2 settings.
The SIP Session Timer T4 settings.
The port range settings for SIP, default is 5060.
The port range for RTP
Enable/Disable the affiliated port feature.
Select headset mode as normal or seat.
Select ring type mode with headset or speaker.
Always transfer the incoming calls (enter the destination number).
If the phone is busy, the call will be transfered (enter the destination
DECSRIPTION