AES 256 Encryption Support for Phones
|
Enhances security by supporting TLS 1.2 and new ciphers.
|
Alphanumeric Dialing
|
Allows users to place a call with alphanumeric characters. You can use these characters for alphanumeric dialing: a-z, A-Z,
0-9, -, _, ., and +.
|
Any Call Pickup
|
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.
|
Assisted Directed Call Park
|
Enables users to park a call by pressing only one button using
the Direct Park feature. Administrators must configure a Busy Lamp Field (BLF)
Assisted Directed Call Park button. When users press an idle BLF Assisted
Directed Call Park button for an active call, the active call is parked at the
Direct Park slot associated with the Assisted Directed Call Park button.
|
Audio Settings
|
Configures audio settings for the phone speaker, the handset, and the headsets that are connected to the phone.
|
Auto Answer
|
Connects incoming calls automatically after a ring or two.
Auto Answer works with either the speakerphone or the headset.
|
Blind Transfer
|
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.
|
Busy Lamp Field (BLF)
|
Allows user to monitor call state of a directory number.
|
Busy
Lamp Field (BLF) Pickup
|
Allows user to pick up incoming calls to the directory number monitored through BLF.
|
Call Back
|
Provides users with an audio and visual alert on the phone when
a busy or unavailable party becomes available.
|
Call Display Restrictions
|
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.
|
Call Forward
|
Allows users to redirect incoming calls to another number. Call Forward services include Call Forward All, Call Forward Busy,
Call Forward No Answer.
|
Call Forward Destination Override
|
Allows you to override Call Forward All (CFA) in cases where the CFA target places a call to the CFA initiator. This feature
allows the CFA target to reach the CFA initiator for important calls. The override works whether the CFA target phone number
is internal or external.
|
Call Forward Notification
|
Allows
you to configure the information that the user sees when receiving a forwarded
call.
|
Call History for Shared Line
|
Allows you to view shared line activity in the phone Call History. This feature:
|
Call Park
|
Allows
users to park (temporarily store) a call and then retrieve the call by using
another phone.
|
Call Pickup
|
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.
|
Call Waiting
|
Indicates (and allows users to answer) an incoming call that
rings while on another call. Incoming call information appears on the phone
display.
|
Caller ID
|
Caller
identification such as a phone number, name, or other descriptive text appear
on the phone display.
|
Caller ID Blocking
|
Allows
a user to block their phone number or name from phones that have caller
identification enabled.
|
Calling Party Normalization
|
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.
|
Cisco Extension Mobility
|
Allows
users to temporarily access their Cisco IP Phone configuration such as line
appearances, services, and speed dials from shared Cisco IP Phone by logging
into the Cisco Extension Mobility service on that phone when they log into the
Cisco Extension Mobility service on that phone.
Cisco
Extension Mobility can be useful if users work from a variety of locations
within your company or if they share a workspace with coworkers.
|
Cisco Extension Mobility Cross Cluster (EMCC)
|
Enables a user configured in one cluster to log into a Cisco IP
Phone in another cluster. Users from a home cluster log into a Cisco IP Phone
at a visiting cluster.
Note
|
Configure Cisco Extension Mobility on Cisco IP Phones before you
configure EMCC.
|
|
Cisco Unified Video Advantage (CUVA)
|
Allows users to make video calls by using a Cisco IP Phone, a personal computer, and an external video camera.
Note
|
Configure the Video Capabilities parameter in the Product Specific Configuration Layout section in Phone Configuration.
|
See the Cisco Unified Video Advantage documentation.
|
Cisco WebDialer
|
Allows
users to make calls from web and desktop applications.
|
Classic Ringtone
|
Supports narrowband and wideband ringtones. The feature makes
the available ringtones common with other Cisco IP Phones.
|
Client Matter Code (CMC)
|
Enables a user to specify that a call relates to a specific
client matter.
|
Conference
|
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
|
Be
sure to inform your users whether these features are activated.
|
|
Configurable RTP/sRTP Port Range
|
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
|
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.
|
You configure the RTP and sRTP port range in the SIP Profile.
|
Contacts Management of the BroadSoft Personal Directory on the Phone
|
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.
|
CTI Applications
|
A
computer telephony integration (CTI) route point can designate a virtual device
to receive multiple, simultaneous calls for application-controlled redirection.
|
Device Invoked Recording
|
Provides end users with the ability to record their telephone
calls via a softkey.
In
addition administrators may continue to record telephone calls via the CTI User
Interface.
|
Directed Call Park
|
Allows
a user to transfer an active call to an available directed call park number
that the user dials or speed dials. A Call Park BLF button indicates whether a
directed call park number is occupied and provides speed-dial access to the
directed call park number.
Note
|
If
you implement Directed Call Park, avoid configuring the Park softkey. This
prevents users from confusing the two Call Park features.
|
|
Directed Call Pickup
|
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.
|
Divert
|
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.
|
Do Not Disturb (DND)
|
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.
|
DND and Call Forward Indication on Non-selected Line Key
|
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.
|
Emergency Calls
|
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.
|
EnergyWise
|
Enables an IP Phone to sleep (power down) and wake (power up) at
predetermined times, to promote energy savings.
|
Enhanced Secure Extension Mobility Cross Cluster (EMCC)
|
Improves the Secure Extension Mobility Cross Cluster (EMCC)
feature by preserving the network and security configurations on the login
phone. By so doing, security policies are maintained, network bandwidth is
preserved and network failure is avoided within the visiting cluster (VC).
|
Extension Mobility Size Safe and Feature Safe
|
With
Feature Safe, your phone can use any phone button template that has the same
number of line buttons that the phone model supports.
Size
Safe allows your phone to use any phone button template that is configured on
the system.
|
Executive-Assistant
|
Indicates shared call control for executives and their assistants.
|
Executive-Assistant Setting Enhancements
|
Allows you to show or hide the Call filter menu item on the phone for the users of the assistant role.
Enables the executive to set the call filtering criteria and view the associated assistants.
Enables the assistant to view the associated executives and choose to opt in to or opt out of the executive's assistants pool.
Allows the assistant to activate or deactivate call diversion and call filtering.
|
Forced
Authorization Code (FAC)
|
Controls the types of calls that certain users can place.
|
Feature Activation Code
|
Allows a user to enable, disable, or configure the Call Forward All service.
|
Headset Sidetone Control
|
Allows an administrator to set the sidetone level of a wired headset.
|
Group Call Pickup
|
Allows
a user to answer a call that is ringing on a directory number in another group.
|
Hold Status
|
Enables phones with a shared line to distinguish between the
local and remote lines that placed a call on hold.
|
Hold/Resume
|
Allows
the user to move a connected call from an active state to a held state.
|
HTTP Download
|
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.
|
HTTP Proxy
|
Allows you to set up a proxy server for the phone.
|
HTTPS for Phone Services
|
Increases security by requiring communication using HTTPS.
Note
|
When the web is in HTTPS mode, the phone is an HTTPS server.
|
|
Improve Caller Name and Number Display
|
Improves the display of caller names and numbers. If the Caller Name is known, then the Caller Number is displayed instead
of Unknown.
|
IPv6 Support
|
Provides support for expanded IP addressing on Cisco IP Phones. IPv6 support is
provided in standalone or in dual-stack configurations. In dual-stack mode, the
phone is able to communicate using IPv4 and IPv6 simultaneously, independent of
the content.
|
Jitter Buffer
|
The
Jitter Buffer feature handles jitter from 10 milliseconds (ms) to 1000 ms for
both audio and video streams.
|
Join
Across Lines
|
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.
|
Join
|
Allows
users to combine two calls that are on one line to create a conference call and
remain on thecall.
|
Line
Display Enhancement
|
Improves Call Display by removing the central dividing line when
it is not required.
This feature applies to the Cisco IP Phone 7841 only.
|
Log
out of hunt groups
|
Allows
users to log out of a hunt group and temporarily block calls from ringing their
phone when they are not available to take calls. Logging out of hunt groups
does not prevent nonhunt group calls from ringing their phone.
|
Malicious Caller Identification (MCID)
|
Allows
users to notify the system administrator about suspicious calls that are
received.
|
Meet
Me Conference
|
Allows
a user to host a Meet Me conference in which other participants call a
predetermined number at a scheduled time.
|
Message Waiting
|
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.
|
Message Waiting Indicator
|
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.
|
Minimum Ring Volume
|
Sets a
minimum ringer volume level for an IP phone.
|
Missed
Call Logging
|
Allows
a user to specify whether missed calls will be logged in the missed calls
directory for a given line appearance.
|
Mobile
Connect
|
Enables users to manage business calls using a single phone
number and pick up in-progress calls on the desk phone and a remote device such
as a mobile phone. Users can restrict the group of callers according to phone
number and time of day.
|
Mobile
Voice Access
|
Extends Mobile Connect capabilities by allowing users to access
an interactive voice response (IVR) system to originate a call from a remote
device such as a cellular phone.
|
Monitoring and Recording
|
Allows
a supervisor to silently monitor an active call. The supervisor cannot be heard
by either party on the call. The user might hear a monitoring audible alert
tone during a call when it is being monitored.
When a
call is secured, the security status of the call is displayed as a lock icon on
Cisco IP Phones. The connected parties might also hear an audible alert tone
that indicates the call is secured and is being monitored.
Note
|
When
an active call is being monitored or recorded, the use can receive or place
intercom calls; however, if the user place an intercom call, the active call
will be put on hold, which causes the recording session to terminate and the
monitoring session to suspend. To resume the monitoring session, the party
whose call is being monitored must resume the call.
|
|
Multicasting Paging
|
Enables users to page some or all phones. If the phone is on an active call while a group page starts, the incoming page is
ignored.
|
Multiple Calls Per Line Appearance
|
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.
|
Music
On Hold
|
Plays music while callers are on hold.
|
Mute
|
Mutes the handset or headset microphone.
|
No
Alert Name
|
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.
|
Noise Removal
|
Allows a user to filter out background noises (such as, keyboard typing, dog barking, clapping, and so on) in a call or meeting.
|
Onhook Dialing
|
Allows a user to dial a number without going off hook. The user can then either pick up the handset or press Dial.
|
Other
Group Pickup
|
Allows
a user to answer a call ringing on a phone in another group that is associated
with the user's group.
|
Pause
in Speed Dial
|
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.
|
Peer Firmware Sharing (PFS)
|
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:
-
Limits congestion on TFTP transfers to centralized remote TFTP servers
-
Eliminates the need to manually control firmware upgrades
-
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.
|
PLK
Support for Queue Statistics
|
The
PLK Support for Queue Statistics feature enables the users to query the call
queue statistics for hunt pilots and the information appears on phone screen.
|
Plus
Dialing
|
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.
|
Power
Negotiation over LLDP
|
Allows
the phone to negotiate power using Link Level Endpoint Discovery Protocol
(LLDP) and Cisco Discovery Protocol (CDP).
|
Problem Reporting Tool
|
Submits phone logs or reports problems to an administrator.
|
Programmable Feature Buttons
|
You can assign features, such as New Call, Call Back, and Call Forward All to line buttons.
|
Quality Reporting Tool (QRT)
|
Allows
users to submit information about problem phone calls by pressing a button. QRT
can be configured for either of two user modes, depending upon the amount of
user interaction desired with QRT.
|
Redial
|
Allows
users to call the most recently dialed phone number by pressing a button or the
Redial softkey.
|
Remote Customization (RC)
|
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.
|
Ringtone Setting
|
Identifies ring type used for a line when a phone has another
active call.
|
Reverse Name Lookup
|
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.
|
RTCP
Hold For SIP
|
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.
|
Secure
Conference
|
Allows
secure phones to place conference calls using a secured conference bridge. As
new participants are added by using Confrn, Join, cBarge softkeys or MeetMe
conferencing, the secure call icon displays as long as all participants use
secure phones.
The
Conference List displays the security level of each conference participant.
Initiators can remove nonsecure participants from the Conference List.
Noninitiators can add or remove conference participants if the Advanced Adhoc
Conference Enabled parameter is set.
|
Serviceability for SIP Endpoints
|
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.
|
Shared
Line
|
Allows
a user with multiple phones to share the same phone number or allows a user to
share a phone number with a coworker.
|
Show Caller Name and Caller Number
|
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.
|
Show Product Configuration Version
|
Allows you to customize the product configuration version that shows on the phone screen Product information.
|
Show
Duration for Call History
|
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.
|
Silence Incoming Call
|
Allows you to silence an incoming call by pressing Ignore softkey or by pressing the volume button down.
|
SIP Transport Auto-Selection
|
Configures the phone to select the appropriate SIP transport protocol automatically, based on the NAPTR records on the DNS
server.
See Configure the SIP Transport.
|
Speed
Dial
|
Dials
a specified number that has been previously stored.
|
Support Executive and Assistant Roles for a User
|
Allows you to set the preference for the executive-assistant role. The phone can select the role when it retrieves both roles
from the BroadWorks server.
|
Synchronization of Call Waiting and Anonymous Call Rejection
|
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.
|
Time
Zone Update
|
Updates the Cisco IP Phone with time zone changes.
|
Transfer
|
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.
|
Voice/Video data priorities
|
Enables you to prioritise voice or video data in limited bandwidth conditions, by specifying different ToS field values for
voice and video packets.
|
Voice Message System
|
Enables callers to leave messages if calls are unanswered.
|
VPN Connection
|
Allows you to set up a VPN connection for the phone.
|
Web Access Enable by Default
|
Web services are enabled by default.
|
XSI call logs display
|
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.
|