Yealink SIP-T40P, User Manual

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User Guide for the SIP-T40P IP Phone
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3.
Click
Confirm
to accept the change.
Note
Receiving RTP Stream
You can configure the phone to receive a Real Time Transport Protocol (RTP) stream from the
pre-configured multicast address(es) and channel(s) without involving SIP signaling. You can
specify up to 31 multicast addresses and channels that the phone listens to on the network.
Note
How the phone handles incoming multicast paging calls depends on Paging Barge, Ignore DND
and Paging Priority Active parameters configured via web user interface.
Paging Barge
The paging barge parameter defines the priority of the voice call in progress. If the priority of an
incoming multicast paging call is lower than that of the active call, it will be ignored
automatically. Valid values in the Paging Barge field:

1 to 31
: Define the priority of the active call, 1 with the highest priority, 31 with the lowest.

Disabled
: The voice call in progress will take precedence over all incoming paging calls.
Ignore DND
The ignore DND parameter defines the lowest priority of multicast listening address from which
the phone can receive an RTP stream when DND is activated. If a priority is selected from the
pull-down list of Ignore DND, the phone will ignore incoming multicast paging calls with lower
priorities when DND is activated in phone mode. Valid values in the Ignore DND field:

1 to 31
: Define the lowest priority of the multicast listening address from which the phone
can receive an RTP stream, 1 with the highest priority, 31 with the lowest.

Disabled
: All the incoming multicast paging calls will be ignored when DND is activated in
phone mode.
The phone will automatically answer all incoming multicast paging calls when DND is activated
in custom mode.
If G722 codec is used for multicast paging, the LCD screen will display the icon to indicate
that it is providing high definition voice.
Default codec for multicast paging is configurable via web user interface only.
RTP stream is listened in the hands-free (speakerphone) mode by default. If you want to listen the
RTP stream using the engaged audio device (speakerphone, handset or headset), contact your
system administrator for more information.
Fixed volume to play RTP stream for specified paging group is configurable by your system
administrator.