IP SIP Phone v2
User’s Guide
Mar. 2005
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on Case 2: you have two calls at hand already, and you press
【
】
Conference on either
active channel:
a.
The green LED of
【
】
Conference will be on if
IP SIP Phone
can host a conference
among them.
b.
【
】
Otherwise, the red LED of
Conference
will flash for a while to indicate a
conference cannot be set up properly.
This may due to the fact that both connected
parties demand a low bit-rate compressed voice CODEC, such as G.723.1(A) or G.729
(A/B), and thus a conference cannot be held between them since the low bit-rate
compressed voice CODEC is shared between them (
IP SIP Phone
equips with only one
hardware low bit-rate G.723/G.729 CODEC).
c.
Note: When a conference is undergoing,
IP SIP Phone
will take or receive no more calls (just
like in busy state).
9.5.2.
Conference Tips
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Once the conference call initiator disconnects, the calls are terminated between all parties.
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To disconnect a specific party in a three-way local conference, you may ask the party to
whom you want to disconnect to hang up first. Alternatively, you may press
【
FLASH
】
to
tear down the conference and separate them into 2 independent calls first, then disconnect
each of them as your wish.
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To place a conference call on hold, press the
【
HOLD
】
key. The other parties can not talk
among themselves. To avoid disrupting the other callers, consider muting the call instead.
To mute the call, press the
【
MUTE
】
button.
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To place a conference call on speakerphone, press
【
SPK
】
.
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Press
【
MUTE
】
to mute the speakerphone. The conference parties cannot hear you but you
can hear them.
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The phone only allows three parties in a conference call.
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Call waiting calls cannot be conferenced in; the conferencing node must establish both legs
of the 3-Way conference.
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A holding call will be unhold automatically before adding into a conference. Besides, you
could put all parties on hold after setting up conference.
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Muting calls will be un-mute automatically before being conferenced in.
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You cannot perform transfer on conference calls.
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To conference in additional callers, the last person called with an
IP SIP Phone
can call an
additional person, that new person can then call someone else, and so on. This is called
“daisy-chaining.