Feature Description Guide
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DHCP setup
Voice-system
features
When a Cisco Unified IP Phone is connected to the Cisco Unified Communications 500 system, it automatically
queries for a Dynamic Host Configuration Protocol server (onboard or external to the application). The DHCP
server responds by assigning an IP address to the Cisco Unified IP Phone and providing the IP address of the
Trivial File Transfer Protocol (TFTP) server through DHCP option 150. Then the phone registers with the Cisco
Unified Communications 500 and attempts to get configuration and phone firmware files from the TFTP server.
Differentiated
Services Code
Point
Voice-system
features
DSCP packet marking specifies the class of service for each packet. Cisco Unified IP Phones get their DSCP
information from the configuration file that is downloaded to the device.
Directed call
pickup
Voice-system
features
Any local phone user can pick up a ringing call on another phone by pressing a softkey and then dialing the
extension. You do not need to belong to a pickup group to use this method. The softkey that you press, either
GPickUp or PickUp, depends on your configuration.
Directories
Voice-system
features
Local, called-name display, and directory search use directory services.
Called-name
display
Users,
phones, and
extensions
The called-name display feature can display either of the following types of names:
● Name for a directory number in a local directory
● Name associated with an overlay directory number
Calls to the first directory number in a set of overlay numbers display a caller ID. Calls to the remaining directory
numbers in the overlay set display the name associated with the directory number.
Display phone
header bar
Users,
phones, and
extensions
You can customize the content of an IP phone header bar, which is the top line of the IP phone display.
The IP phone header bar, or top line, of a Cisco Unified IP Phone normally replicates the text that appears next to
the first line button. The header bar can contain a user-definable message instead of the extension number. For
example, the header bar can be used to display a name or the full E.164 number of the phone. If no description is
specified, the header bar replicates the extension number that appears next to the first button on the phone.
Phone system-
message
display
Users,
phones, and
extensions
The system-message display feature allows you to specify a custom text or display messages to appear in the
lower part of the display window on display-capable IP phones. If you do not set a custom text or display
message, a default message is displayed.
Distinctive ring
Users,
phones, and
extensions
Distinctive ring is used to identify internal and external incoming calls. An internal call is defined as a call
originating from any Cisco Unified IP Phone that is registered in the Cisco Unified Communications 500 or is
routed through the local FXS port.
Do not disturb
(DND)
Users,
phones, and
extensions
The DND feature prevents incoming calls from audibly ringing a phone. When DND is enabled, the phone flashes
an alert to visually indicate an incoming call instead of ringing, and you can answer the call if desired.
DSP
Voice-system
features
A digital-signal-processor (DSP) chip provides analog FXS and FXO and digital BRI/PRI to IP connectivity, in
addition to conferencing features for audio calls.
Direct station
select (DSS)
Users,
phones, and
extensions
DSS allows a multibutton phone user to transfer calls to an idle monitored line by pressing the Transfer key and
the appropriate monitored line button. A monitored line is one that appears on two phones; one phone can use the
line to make and receive calls and the other phone simply monitors whether the line is in use. Consultative
transfers can occur during DSS transfers (transferring calls to idle monitored lines).
Dual-tone
multifrequency
(DTMF) relay
(SIP call
control only)
Voice-system
features
DTMF relay handles incoming and outgoing DTMF signals for SIP calls.
DTMF relay for
H.323
networks
Voice-system
features
For IP phones on H.323 networks, DTMF is relayed using the H.245 alphanumeric method, which is defined by
the ITU H.245 standard. This method separates DTMF digits from the voice stream and sends them as ASCII
characters in H.245 user input indication messages through the H.245 signaling channel instead of the Real-Time
Transport Protocol (RTP) channel.
DTMF relay for
SIP trunks
Voice-system
features
To use remote voicemail or interactive-voice-response (IVR) applications on SIP networks from Cisco Unified
Communications 500 phones, the DTMF digits used by these phones must be converted to the RFC 2833 in-band
DTMF relay mechanism used by SIP phones. The SIP DTMF relay method is needed in the following situations:
● When SIP is used to connect a Cisco Unified Communications 500 system to a remote SIP-based IVR or
voicemail applications
● When SIP is used to connect a Cisco Unified Communications 500 system to a remote SIP-PSTN voice
gateway that goes through the PSTN to a voicemail or IVR application; SIP phones natively support in-band
DTMF relay as specified in RFC 2833
Encrypting
stored
personal
identification
numbers
(PINs)
Voice-system
features
Voicemail PIN codes are stored in encrypted form for security reasons.
Ephone
Voice-system
features
Ephone is a term for Cisco Unified Communications 500 configuration for phones using SCCP or the single-
channel-per-carrier (SCPC) protocol.
Extension
mobility
Users,
phones, and
extensions
Extension mobility provides the benefit of phone mobility for end users. It offers a user login service that allows
phone users to temporarily access a physical phone other than their own phone and use their personal settings,
such as directory number, speed-dial lists, and services, as if the phone were their own desk phone. The phone