AES 256 Encryption Support for Phones
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Enhances security by supporting TLS 1.2 and new ciphers.
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Any Call Pickup
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Allows
users to pick up a call on any line in their call pickup group, regardless of
how the call was routed to the phone.
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Audio Settings
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Configures audio settings for the phone speaker, the handset, and the headsets that are connected to the phone.
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Auto Answer
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Connects incoming calls automatically after a ring or two.
Auto Answer works with either the speakerphone or the headset.
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Blind Transfer
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Blind Transfer: This transfer joins two established calls (call is in hold or in connected state) into one call and drops
the feature initiator from the call. Blind Transfer does not initiate a consultation call and does not put the active call
on hold.
Some JTAPI/TAPI applications are not compatible with the Join and Blind Transfer feature implementation on the Cisco IP Phone
and you may need to configure the Join and Direct Transfer Policy to disable join and direct transfer on the same line or
possibly across lines.
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Call Back
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Provides users with an audio and visual alert on the phone when
a busy or unavailable party becomes available.
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Call Display Restrictions
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Determines the information that will display for calling or
connected lines, depending on the parties who are involved in the call.
RPID and PAID caller id handling are supported.
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Call Forward
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Allows users to redirect incoming calls to another number. Call Forward services include Call Forward All, Call Forward Busy,
Call Forward No Answer.
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Call Forward Notification
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Allows
you to configure the information that the user sees when receiving a forwarded
call.
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Call History for Shared Line
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Allows you to view shared line activity in the phone Call History. This feature:
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Call Park
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Allows
users to park (temporarily store) a call and then retrieve the call by using
another phone.
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Call Pickup
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Allows
users to redirect a call that is ringing on another phone within their pickup
group to their phone.
You
can configure an audio and visual alert for the primary line on the phone. This
alert notifies the users that a call is ringing in their pickup group.
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Call Waiting
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Indicates (and allows users to answer) an incoming call that
rings while on another call. Incoming call information appears on the phone
display.
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Caller ID
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Caller
identification such as a phone number, name, or other descriptive text appear
on the phone display.
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Caller ID Blocking
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Allows
a user to block their phone number or name from phones that have caller
identification enabled.
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Calling Party Normalization
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Calling party normalization presents phone calls to the user
with a dialable phone number. Any escape codes are added to the number so that
the user can easily connect to the caller again. The dialable number is saved
in the call history and can be saved in the Personal Address Book.
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Conference
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Allows
a user to talk simultaneously with multiple parties by calling each participant
individually.
Allows a noninitiator in a standard (adhoc) conference to add or remove participants; also allows any conference participant
to join together two standard conferences on the same line.
Note
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Be
sure to inform your users whether these features are activated.
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Configurable RTP/sRTP Port Range
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Provides a configurable port range (Port Min to Port Max) for Real-Time Transport Protocol (RTP) and secure Real-Time Transport
Protocol (sRTP).
The value range for the Port Min and Port Max is 2048 to 49151.
The default RTP and sRTP port range is 16384 to 16482.
Note
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If the value range (Port Max - Port Min) is less than 16 or you use an incorrect port range, the port range (16382 to 32766)
is used instead.
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You configure the RTP and sRTP port range in the SIP Profile.
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Contacts Management of the BroadSoft Personal Directory on the Phone
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Provides the user with the ability to add, edit, and delete in the BroadSoft Personal directory. Allows the user to add contacts
from recent calls or any types of directories (if enabled).
In addition administrator can set the BroadSoft Personal directory as the target directory to store new contacts.
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Directed Call Pickup
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Allows a user to pick up a ringing call on a DN directly by pressing the GPickUp softkey and entering the directory number
of the device that is ringing.
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Divert
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Allows
a user to transfer a ringing, connected, or held call directly to a
voice-messaging system. When a call is diverted, the line becomes available to
make or receive new calls.
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Do Not Disturb (DND)
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When
DND is turned on, either no audible rings occur during the ringing-in state of
a call, or no audible or visual notifications of any type occur.
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DND and Call Forward Indication on Non-selected Line Key
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Displays the DND and call forward icons next the to the line key label. The line key should be enabled with feature key sync.
The line key should also be enabled with DND or call forward.
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Emergency Calls
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Enables users to make emergency calls. The emergency services receive the phone's location and a call-back number, to use
when the emergency call unexpectedly disconnects.
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Feature Activation Code
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Allows a user to enable, disable, or configure the Call Forward All service.
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Headset Sidetone Control
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Allows an administrator to set the sidetone level of a wired headset.
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Group Call Pickup
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Allows
a user to answer a call that is ringing on a directory number in another group.
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Hold Status
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Enables phones with a shared line to distinguish between the
local and remote lines that placed a call on hold.
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Hold/Resume
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Allows
the user to move a connected call from an active state to a held state.
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HTTP Download
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Enhances the file download process to the phone to use HTTP by
default. If the HTTP download fails, the phone reverts to using the TFTP
download.
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HTTP Proxy
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Allows you to set up a proxy server for the phone.
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HTTPS for Phone Services
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Increases security by requiring communication using HTTPS.
Note
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When the web is in HTTPS mode, the phone is an HTTPS server.
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Improve Caller Name and Number Display
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Improves the display of caller names and numbers. If the Caller Name is known, then the Caller Number is displayed instead
of Unknown.
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Jitter Buffer
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The
Jitter Buffer feature handles jitter from 10 milliseconds (ms) to 1000 ms for
both audio and video streams.
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Join
Across Lines
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Allows
users to combine calls that are on multiple phone lines to create a conference
call.
Some
JTAPI/TAPI applications are not compatible with the Join and Direct Transfer
feature implementation on the Cisco IP Phone and you may need to configure the
Join and Direct Transfer Policy to disable join and direct transfer on the same
line or possibly across lines.
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Join
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Allows
users to combine two calls that are on one line to create a conference call and
remain on thecall.
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Message Waiting
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Defines directory numbers for message waiting on and off
indicators. A directly-connected voice-message system uses the specified
directory number to set or to clear a message waiting indication for a
particular Cisco IP Phone.
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Message Waiting Indicator
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A light on the handset that indicates that a user has one or more new voice messages.
A line key LED or a KEM key LED that indicates that a monitored voicemail user or group has one or more new voice messages.
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Minimum Ring Volume
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Sets a
minimum ringer volume level for an IP phone.
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Missed
Call Logging
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Allows
a user to specify whether missed calls will be logged in the missed calls
directory for a given line appearance.
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Multiple Calls Per Line Appearance
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Each
line can support multiple calls. By default, the phone supports two active
calls per line, and a maximum of ten active calls per line. Only one call can
be connected at any time; other calls are automatically placed on hold.
The
system allows you to configure maximum calls/busy trigger not more than 10/6.
Any configuration more than 10/6 is not officially supported.
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Music
On Hold
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Plays music while callers are on hold.
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Mute
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Mutes the handset or headset microphone.
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No
Alert Name
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Makes
it easier for end users to identify transferred calls by displaying the
original caller’s phone number. The call appears as an Alert Call followed by
the caller’s telephone number.
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Pause
in Speed Dial
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Users can set up the speed-dial feature to reach destinations that require Forced Authorization Code (FAC) or Client Matter
Code (CMC), dialing pauses, and additional digits (such as a user extension, a meeting access code, or a voicemail PIN) without
manual intervention. When the user presses the speed dial, the phone establishes the call to the specified DN and sends the
specified FAC, CMC, and DTMF digits to the destination and inserts the necessary dialing pauses.
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Peer Firmware Sharing (PFS)
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Allows IP Phones located at remote sites to share the firmware files amongst them, which saves bandwidth when the upgrade
process takes place. This feature uses Cisco Peer-to-Peer-Distribution Protocol (CPPDP) which is a Cisco proprietary protocol
used to form a peer-to-peer hierarchy of devices. CPPDP is also used to copy firmware or other files from peer devices to
the neighbouring devices.
PFS aids in firmware upgrades in branch/remote office deployment scenarios that run over bandwidth-limited WAN links.
Provides the following advantages over the traditional upgrade method:
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Limits congestion on TFTP transfers to centralized remote TFTP servers
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Eliminates the need to manually control firmware upgrades
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Reduces phone downtime during upgrades when large numbers of devices are reset simultaneously
The more the number of IP phones, the better it's performance compared to the traditional firmware upgrade method.
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Plus
Dialing
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Allows
the user to dial E.164 numbers prefixed with a plus (+) sign.
To
dial the + sign, the user needs to press and hold the star (*) key for at least
1 second. This applies to dialing the first digit for an on-hook (including
edit mode) or off-hook call.
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Power
Negotiation over LLDP
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Allows
the phone to negotiate power using Link Level Endpoint Discovery Protocol
(LLDP) and Cisco Discovery Protocol (CDP).
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Problem Reporting Tool
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Submits phone logs or reports problems to an administrator.
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Programmable Feature Buttons
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You can assign features, such as New Call, Call Back, and Call Forward All to line buttons.
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Redial
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Allows
users to call the most recently dialed phone number by pressing a button or the
Redial softkey.
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Remote Customization (RC)
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Allows a service provider to customize the phone remotely. There is no need for either the service provider to physically
touch the phone or a user to configure the phone. The service provider can work with a sales engineer at the time of ordering
to set this up.
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Ringtone Setting
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Identifies ring type used for a line when a phone has another
active call.
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Reverse Name Lookup
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Identifies the caller name using the incoming or outgoing call number. You must configure either the LDAP Directory or the
XML directory. You can enable or disable the reverse name lookup using the phone administration web page.
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RTCP
Hold For SIP
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Ensures that held calls are not dropped by the gateway. The
gateway checks the status of the RTCP port to determine if a call is active or
not. By keeping the phone port open, the gateway will not end held calls.
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Serviceability for SIP Endpoints
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Enables administrators to quickly and easily gather debug information from
phones.
This
feature uses SSH to remotely access each IP phone. SSH must be enabled on each
phone for this feature to function.
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Shared
Line
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Allows
a user with multiple phones to share the same phone number or allows a user to
share a phone number with a coworker.
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Show Caller Name and Caller Number
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The phones can display both the caller name and caller number for incoming calls. The phone screen size limits the length
of the caller name and the caller number that display.
If boxes are displayed in the caller name, follow the procedure in Display Caller Number Instead of Unresolved Caller Name.
This feature applies to the incoming call alert only and doesn’t change the Call Forward and Hunt Group features.
See “Caller ID” in this table.
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Show Product Configuration Version
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Allows you to customize the product configuration version that shows on the phone screen Product information.
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Show
Duration for Call History
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Displays the time duration of placed and received calls in the Call History
details.
If the
duration is greater than or equal to one hour, the time is displayed in the
Hour, Minute, Second (HH:MM:SS) format.
If the
duration is less than one hour, the time is displayed in the Minute, Second
(MM:SS) format.
If the
duration is less than one minute, the time is displayed in the Second (SS)
format.
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Silence Incoming Call
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Allows you to silence an incoming call by pressing Ignore softkey or by pressing the volume button down.
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Speed
Dial
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Dials
a specified number that has been previously stored.
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Synchronization of Call Waiting and Anonymous Call Rejection
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Allows you to enable or disable synchronization of the Call Waiting and Anonymous Call Rejection functions between a specific
line and a BroadSoft XSI server.
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Time
Zone Update
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Updates the Cisco IP Phone with time zone changes.
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Transfer
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Allows
users to redirect connected calls from their phones to another number.
Some
JTAPI/TAPI applications are not compatible with the Join and Direct Transfer
feature implementation on the Cisco IP Phone and you may need to configure the
Join and Direct Transfer Policy to disable join and direct transfer on the same
line or possibly across lines.
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Voice Message System
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Enables callers to leave messages if calls are unanswered.
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VPN Connection
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Allows you to set up a VPN connection for the phone.
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Web Access Enable by Default
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Web services are enabled by default.
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XSI call logs display
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Allows you to configure a phone to display recent call logs from either the BroadWorks server or the local phone. After you
enable the feature, the Recents screen has a Display recents from menu and the user can choose the XSI call logs or the local call logs.
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