Yealink SIP-T42S User Manual

User Guide for the SIP-T42S 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 high definition voice is provided.
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.

Yealink SIP-T42S Specifications

General

Manual	Yes
Cables included	LAN (RJ-45)
Number of handles included	1 pc(s)
Lines quantity	12 lines
Phonebook capacity	1000 entries
Number of VoIP accounts	12
Mounting	Desk/Wall
Handset type	Wired handset
Product type	IP Phone
Product color	Black
Voice codecs	CNG, G.711Mu, G.711a, G.723.1, G.726, G.729ab, VAD, iLBC
Operating temperature (T-T)	-10 - 50 °C
Operating relative humidity (H-H)	10 - 95 %
Display diagonal	2.7 \
Display resolution	192 x 64 pixels
AC adapter input voltage	100 - 240 V
AC adapter output current	1.2 A
AC adapter output voltage	5 V
Dual-Tone Multi-Frequency (DTMF) modes	In-band, Out-of band, SIP info
Volume control	Digital
Number of programmable keys	6
USB 2.0 ports quantity	1
Ethernet LAN (RJ-45) ports	2
Networking standards	IEEE 802.3af
Ethernet LAN data rates	10, 100 Mbit/s
Ethernet LAN interface type	Fast Ethernet
Supported network protocols	SIP v1, SIP v2, LLDP/CDP/DHCP, OpenVPN, IEEE802.1X, IPv6, FTP/TFTP/HTTP/HTTPS

Weight and Dimensions

Depth	189 mm
Width	212 mm
Height	175 mm
Package depth	223 mm
Package width	246 mm
Package height	120 mm