with 9600-Series IP Deskphones

Feature Summary
®
Avaya Aura with 9600-Series IP Deskphones
May 2016 - Avaya Aura® Platform Release 7.0.1
Introduction
The Avaya Aura® solution is a rich, highly interoperable set of SIP components that takes enterprise
communications architecture to the next level. At the core of the solution is the Avaya Aura® Session
Manager providing SIP interoperability, dial plan generation, SIP normalization, SIP routing, and many
other SIP services to create a secure, centralized and easy-to-manage enterprise backbone network.
A chief advantage of the Avaya Aura® solution is the ability to deliver the right features to the right
users across an enterprise regardless of protocol or endpoint device. System architects and planners
have the advantage with the Avaya Aura® solution to choose the protocol and endpoint that satisfies
the user’s needs and deliver these services using the delivery mechanism and network of choice. In
addition, the Avaya Aura solution allows enterprises to mix and match not only TDM and IP, but SIP
and H.323 endpoints anywhere in the customer network.
Avaya Aura Protocol Support
The Avaya Aura® solution supports the largest variety of technologies in the industry including analog,
Digital (the Avaya “DCP” protocol), H.323 Internet Protocol (IP), and Session Initiation Protocol (SIP)
handsets and auxiliary equipment. Each technology is unique and offers differing levels of
functionality based on the technology supported with analog devices providing the lowest capabilities,
and H.323 and SIP the richest. These notes document the available services for the H.323 and SIP
protocols and are valid for Avaya Aura® Platform 7.0.1.
The information is this document is accurate as of the issue date and subject to change.
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Issued 13 May 2016
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Feature Summary
Avaya Aura® with 9600-Series IP Deskphones – Available Services
The following table shows each relevant Avaya Aura® service in alphabetical order with the support for
each configuration for comparison. The associated deskphones are the Avaya 9600-Series IP
Deskphones.

The “H.323” columns are based upon Avaya Aura® Communications Manager 7.0.1.
o The two columns distinguish between Avaya one-X® Deskphone H.323 3.2.5 software
which is used on the 9620L/9620C/9630G/9640/9640G/9650/9650C IP Deskphones
and the Avaya Deskphone H.323 6.5.0 software which is used on the
9608/9608G/9611G/9621G/9641G/9641GS IP Deskphones.

The “SIP – SM” columns are based upon Avaya Aura® Platform 7.0.1 (Avaya Aura®
Communications Manager 7.0.1 with Avaya Aura® Session Manager 7.0.1).
o The two columns distinguish between Avaya one-X® Deskphone SIP 2.6.14 software
which is used on the 9620L/9620C/9630G/9640/9640G/9650/9650C IP Deskphones
and the Avaya one-X® Deskphone SIP 7.0.1 software which is used on the
9601/9608/9608G/9611G/9621G/9641G/9641GS IP Deskphones.
Table entries which have been updated with this version of the document are identified in italic text.
The term “AST” is used to identify a set of SIP features that use standards IETF signaling to
accomplish, but are not contained in the “SIPPING-19” specifications. These items are “Advanced
SIP Telephony” and are identified in the following table by orange coloring.
H.323
(H.323 3.2.5)
H.323
(H.323 6.6.0)
SIP
(SIP 2.6.14)
SIP
(SIP 7.0.1)
Abandoned Call Logging
Supported
Supported
Supported
Supported
Abbreviated Dialing
Supported
Supported
Partially
Supported
(no button)
Partially
Supported
(no button)
Abort Transfer
Supported
Supported
Supported
Supported
Account Codes
Supported
Supported
Supported
Supported
Advice of Charge
Supported
Supported
Not
Supported
Not
Supported
Announcements
Supported
Supported
Supported
Supported
Service/Feature
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Feature Summary
H.323
(H.323 3.2.5)
H.323
(H.323 6.6.0)
SIP
(SIP 2.6.14)
SIP
(SIP 7.0.1)
Attendant Console
Supported
Supported
Not
Supported
Not
Supported
Authorization Codes
Supported
Supported
Supported
Supported
Automatic Answer Intercom
Supported
Supported
Supported
Supported
Automatic Callback
Supported
Supported
Supported
Supported
Automatic Dial Buttons
Supported
Supported
Supported
Supported
Automatic Exclusion
Supported
Supported
Supported
Supported
Automatic Hold
Supported
Supported
Supported
Supported
Automatic Route Selection
(ARS)
Supported
Supported
Supported
Supported
Aux-work for a Hunt Group
Supported
Supported
Not
Supported
Supported via
“Hunt Group
Busy Button”
Bridged Line (Call)
Appearances
Supported
Supported
Supported
Supported
Bridged Line Appearances Analog
Supported
Supported
Not
Supported
Not
Supported
Busy Line Indicator
Supported
Supported
Supported
Supported
Button Module (12,24)
Supported
(Model
specific)
Supported
(Model
specific)
Supported
(Model
specific)
Supported
(Model
specific)
Call Coverage (6 levels)
Supported
Supported
Supported
Supported
Caller ID (Name and number)
Supported
Supported
Supported
Supported
Call Forward (All, busy, don’t
answer, disable)
Supported
Supported
Supported
Supported
Call Hold, Resume
Supported
Supported
Supported
Supported
Service/Feature
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Feature Summary
H.323
(H.323 3.2.5)
H.323
(H.323 6.6.0)
SIP
(SIP 2.6.14)
SIP
(SIP 7.0.1)
Call Log
(Missed/Answered/Outgoing
calls, Call/Delete/Details)
Supported
(HTTP
backup)
Supported
(HTTP
backup)
Supported
(local)
Supported
(local)
Call Log (Busy)
Supported
Supported
Not
Supported
Supported
Call Log (Offline)
Supported
with CES
Supported
Not
Supported
Supported
Call Park, Answer Back
Supported
Supported
Supported
Supported
Calling Party Number Block,
Unblock
Supported
Supported
Supported
Supported
Calling Party Number Block,
Unblock of Internal Numbers
Supported
Supported
Partially
Supported
Partially
Supported
Call Pickup
Supported
Supported
Supported
Supported
Class of Service
Supported
Supported
Supported
Supported
Click to Conference
Supported
Supported
Not
Supported
Not
Supported
Code Calling
Supported
Supported
Supported
Supported
Conference (Ad-Hoc – 6
Party)
Supported
Supported
Supported
Supported
Contact Center (General)
Supported
Supported
Not
Supported
Supported
Contacts
(Add/Edit/Delete/Details)
Supported
Supported
Supported
Supported
Core Redundancy
Supported
Supported
Supported
Supported
Crisis Alert (Dial/View)
Supported
Supported
Partially
Supported
(Dial only)
Partially
Supported
(Dial only)
Directed Call Pickup
Supported
Supported
Supported
Supported
Service/Feature
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Feature Summary
H.323
(H.323 3.2.5)
H.323
(H.323 6.6.0)
SIP
(SIP 2.6.14)
SIP
(SIP 7.0.1)
Directory (Aura Integrated)
Supported
Supported
Not
Supported
Supported
Directory (LDAP)
Supported
Supported
Supported
Supported
Distinctive Ringing
Supported
Supported
Supported
Supported
Emergency Button (one touch
access)
Supported
Supported
Supported
Supported
Enhanced Call Forwarding
Supported
Supported
Supported
Supported
Enhanced Group Call Pickup
Alerting
Supported
Supported
Supported
Supported
Exclusion
Supported
Supported
Supported
Supported
Extended Call Pickup
Supported
Supported
Supported
Supported
Extension to Cellular (EC500)
Supported
Supported
Supported
Supported
Favorite Button
Not
Supported
Not
Supported
Supported
Supported
Feature Named Extensions
Supported
Supported
Supported
Supported
Flexible Language Displays
Supported
Supported
Supported
Supported
Forced Entry of Account
Codes
Supported
Supported
Supported
Supported
Guest Login
Supported
Supported
Not
Supported
Not
Supported
Group Paging
Supported
Supported
Partially
Supported
(initiate only)
Supported
Hold Recall
Supported
Supported
Supported
Supported
Hospitality (general)
Supported
Supported
Not
Supported
Not
Supported
Service/Feature
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Feature Summary
H.323
(H.323 3.2.5)
H.323
(H.323 6.6.0)
SIP
(SIP 2.6.14)
SIP
(SIP 7.0.1)
Hotline
Supported
Supported
Not
Supported
Supported
(settings file)
Hunt Groups
Supported
Supported
Supported
Supported
Hunt Group Busy Button
Not
Supported
Not
Supported
Not
Supported
Supported
Idle Line Appearance Select
Supported
Supported
Supported
Supported
Intercom - Automatic
Supported
Supported
Supported
Supported
Intercom - Dial
Supported
Supported
Supported
Supported
Last Number Dialed
Supported
Supported
Supported
Supported
Limit Number of Concurrent
Calls
Supported
Supported
Not
Supported
Supported
Local Survivability with
Survivable Remote
Supported
Supported
Supported
Supported
Local Survivability with Third
Party Gateways
Not
Supported
Not
Supported
Supported
Supported
Local Survivability with IP
Office Centralized
Not
Supported
Not
Supported
Supported
Supported
Loudspeaker Paging
Supported
Supported
Supported
Supported
Loudspeaker Paging - Deluxe
Supported
Supported
Supported
Supported
Malicious Call Trace (MCT) Activation
Supported
Supported
Supported
Supported
Malicious Call Trace (MCT) Controller
Supported
Supported
Not
Supported
Not
Supported
Manual Signaling
Supported
Supported
Not
Supported
Not
Supported
Meet-Me Conferencing - CM
Supported
Supported
Supported
Supported
Service/Feature
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Feature Summary
H.323
(H.323 3.2.5)
H.323
(H.323 6.6.0)
SIP
(SIP 2.6.14)
SIP
(SIP 7.0.1)
Message Retrieval (one
button)
Supported
Supported
Supported
Supported
Message Waiting Indication
(own number)
Supported
Supported
Supported
Supported
Message Waiting Indication
(third party number)
Supported
Supported
Supported
Supported
MLPP (Multiple Level
Precedence and Preemption) TDM Trunking
Supported
Supported
Supported
Supported
Multiple Call Handling,
Multiple Lines, Multiple Call
Appearances
Supported
Supported
Supported
Supported
Multiple Device Access
Not
Supported
Not
Supported
Partially
Supported
Supported
Multi-Language (Input and
Display)
Supported
Supported
Supported
Supported
Multi-Language Input
(Korean, Hebrew)
Not
Supported
Not
Supported
Supported
Not
Supported
Multi-Language Input (Arabic,
Japanese, Chinese)
Not
Supported
Not
Supported
Not
Supported
Not
Supported
Night Service
Supported
Supported
Not
Supported
Not
Supported
One Touch Recording for
Modular Messaging
Supported
Supported
Supported
Supported
one-X Communicator –
Deskphone Mode
Supported
Supported
Not
Supported
Supported
one-X Mobile Integration
Supported
Supported
Supported
Supported
one-X Portal Integration
Supported
Supported
Partially
Supported
Partially
Supported
Service/Feature
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Feature Summary
H.323
(H.323 3.2.5)
H.323
(H.323 6.6.0)
SIP
(SIP 2.6.14)
SIP
(SIP 7.0.1)
Pin Checking
Supported
Supported
Not
Supported
Not
Supported
Presence - Advertise
Supported
Supported
Supported
Supported
Presence - Display
Not
Supported
Not
Supported
Supported
Supported
Priority Calling
Supported
Supported
Supported
Supported
Pull Transfer
Supported
Supported
Not
Supported
Not
Supported
Remote Worker - SBC
Not
Supported
Not
Supported
Supported
Supported
Remote Worker – VPN
Integrated
Supported
Supported
Not
Supported
Not
Supported
Ring Tones – “Classic” or
“European”
Supported
Supported
Partially
Supported
(Classic only)
Supported
Supported
(9670G only)
Supported
Not
Supported
Supported
Ring Tones - Downloaded
Not
Supported
Not
Supported
Not
Supported
Supported
Ringing Control - Bridged
Line
Supported
Supported
Supported
Supported
Ringing Control – Per Contact
Not
Supported
Not
Supported
Not
Supported
Supported
Ringing - Abbreviated and
Delayed
Supported
Supported
Not
Supported
Not
Supported
Send All Calls
Supported
Supported
Supported
Supported
Service/Feature
Ring Tones - Rich
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Feature Summary
Service/Feature
H.323
(H.323 3.2.5)
H.323
(H.323 6.6.0)
SIP
(SIP 2.6.14)
SIP
(SIP 7.0.1)
Supported
Service Observing
Supported
Supported
Partially
Supported
(observed
only)
Simulated Bridged
Appearance
Supported
Supported
Supported
Supported
Station On-Hook Dialing
Supported
Supported
Supported
Supported
Team Button
Supported
Supported
Not
Supported
Supported
Temporary Bridged
Appearance
Supported
Supported
Supported
Supported
Time of Day Routing
Supported
Supported
Supported
Supported
Traffic Measurements
Supported
Supported
Supported
Supported
Transfer (Attended,
Unattended)
Supported
Supported
Supported
Supported
Transfer on Hang-up
Supported
Supported
Not
Supported
Supported
Transfer to Voicemail
(Answered)
Supported
Supported
Supported
Supported
Transfer to Voicemail
(Alerting)
Supported
Supported
Not
Supported
Not
Supported
VIP Calling
Supported
Supported
Supported
Supported
Voice Initiated Dialing
Supported
Not
Supported
Not
Supported
Not
Supported
Web Browser
Supported
Supported
Supported
Supported
Whisper Page Activate
Supported
Supported
Supported
Supported
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Feature Summary
Service Descriptions
The more detailed descriptions of the services listed in the previous table are described here in
alphabetical order. Differences between H.323 and SIP operation are highlighted.
Abandoned Call Logging
Calls which arrive, are not answered, and go to voicemail are added to the deskphone call log to
notify the user a call was received, but not answered.
Abbreviated Dialing
Abbreviated Dialing (AD) is used to reduce the number of digits needed to dial to place a call. Instead
of dialing the entire number, a short code is dialed to access the number. The system then dials the
stored number automatically. You can also assign abbreviated dialing buttons to H.323 deskphones,
so that you press a single button to dial frequently called numbers.
You cannot assign abbreviated dialing buttons to SIP deskphones. SIP deskphone users can
program the FAC and dial code number against a Contact or against a Speed Dial entry.
Abort Transfer
Abort Transfer is used to stop the transfer operation whenever a user presses a non-idle call
appearance button in the middle of the transfer operation, or when the user hangs up.
Account Codes
Account Code Dialing associates a call with an account number. A user enters a FAC for Account
Code Dialing before the user dials a deskphone number. You can specify the use of the FAC is
mandatory or optional for the user. When a user dials a deskphone number and the FAC, the system
records the:
●
Deskphone number
●
Account code
●
Trunk Access Code (TAC), or the Automatic Route Selection (ARS) access code
The system does not record the FAC for Account Code Dialing.
Advice of Charge
Users can view call charges on deskphone displays. From a display, a user can see the cost of an
outgoing call, both while the call is in progress and at the end of the call. If users are to control when
the display of the call charge information is displayed, a display button can be administered. The
system can also be administered so that the system displays call charges automatically whenever a
user places an outgoing call.
This feature is not supported on SIP deskphones.
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Feature Summary
Announcements
“Announcements” is a feature to play recordings for callers in the enterprise. For example, you can
inform callers that:
 The call cannot be completed as dialed
 The call is in a queue
 All lines are busy
Announcements are often used in conjunction with music. The source for announcements can be
either integrated or external.
Attendant Console
An attendant console is a terminal that has access to all the attendant features – Auto-Manual
Splitting, Attendant Conferencing, Attendant Trunk Group Access, Attendant Intrusion, Attendant
Overflow, etc. SIP deskphones can be “seen” (as extensions, busy line indicators, bridged
appearances, etc.) and connected with an H.323 or DCP attendant console. SIP deskphones cannot
function as full-featured attendant consoles.
Authorization Codes
Authorization Codes allows a deskphone user to input a personal identification code as a means for
extending the control of system users’ privileges and security for remote access callers. Authorization
codes may be used for any or all of the following reasons:
 allow a calling user to override the Facility Restriction Level (FRL) assigned to the originating
station or trunk
 restrict individual incoming tie trunks and remote access trunks from accessing an outgoing
trunk
 identify certain calls on call detail records for cost-allocation purposes
 and provide additional security control for the system.
Automatic Answer Intercom
Intercom calls to SIP deskphones when set for automatic answer will immediately go off hook on
speaker when an intercom call is received. Also, SIP deskphones may make intercom calls to other
deskphones set for auto answer for intercom calls. Incoming automatic answer intercom calls do not
alert audibly or visually, the receiving deskphones automatically answer the call, place the talkpath
automatically on speakerphone, and display the caller’s name on the active line appearance.
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Feature Summary
Automatic Callback
Automatic Callback (ACB) allows internal users who place calls to busy or unanswered internal
deskphones to be called back when the called deskphone becomes available. When a user activates
Automatic Callback, the system monitors the called deskphone and when the called deskphone
becomes available to receive a call, the system automatically originates the Automatic Callback call.
The originating party receives priority ringing. The calling party then lifts the handset, and the called
party receives the same ringing that the system provided on the original call.
SIP deskphones can make and receive ACB calls, but only when both deskphones are
controlled by the same Communication Manager.
Automatic Dial Buttons
Automatic dial buttons can be administered by the administrator or the user and appear on the button
list of the deskphone. SIP users can have the auto dial buttons administered with names and
numbers by the administrator or the individual users just like H.323.
Automatic Exclusion
Automatic exclusion allows a user of a deskphone to keep others with bridged line appearances of the
same extension from bridging onto an existing call. Automatic exclusion is administered on a per
station basis. Users may turn off automatic exclusion by pressing the exclusion button.
Automatic Hold
With automatic hold, a user can also press a second call appearance to put an active call on hold.
With deskphones, the user can choose with each call if hold or automatic hold is to be selected by
selecting the “hold and answer” or “drop and answer” options presented on the screen.
Automatic Route Selection (ARS)
Automatic Route Selection routes calls over the public network based on the preferred (normally the
least expensive) route available at the time the call is placed. ARS provides a choice of routes for any
given public network call.
Aux-work for a Hunt Group
If an agent is an ACD split and a hunt group member, the agent in the split usually has an AUX work
button that also activates or deactivates the Hunt Group Busy option. If an agent is the last available
member and pushes the AUX work button, the lamp on the button flashes until the queue is empty.
The flashing light means that the agent is still available. When the queue is empty, the lamp lights but
does not flash, and the system activates Hunt Group Busy. This capability is not supported on SIP
deskphones (refer to “Hunt Group Busy Button” for an alternative). As a workaround, the Hunt
Group Busy Activation/Deactivation FACs can be dialed from the SIP deskhone.
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Feature Summary
Bridged Line (Call) Appearances
The bridged line appearance feature is used to give multi-appearance deskphones an appearance of
another deskphone number. With a bridged line appearance, the user can originate, answer, and
bridge onto calls to or from the deskphone number of another user.
Bridged Line Appearances - Analog
Analog Bridged Line Appearances allow an analog phone to be bridged with a digital/IP phone.
This feature is not supported on SIP deskphones.
Busy Line Indicator
The busy line indicator provides deskphone users with a visual indicator of the busy or the idle status
of one of the following system resources:





An extension number
A trunk group
A terminating extension group (TEG)
A hunt group, either direct department calling (DDC) or uniform call distribution (UCD)
Any loudspeaker paging zone, including all zones
Button Module (12, 24)
The 12 button module can be used with the 9608/9608G/9611G/9641G IP Deskphone. The 24 button
module can be used on all models except the 9601/9620L/9620C/9621G/9670G. They connect to
either SIP or H.323 phones and provides up to 24 simultaneously visible and accessible buttons and
lamps to support bridged call appearances, busy line indicators and one-touch access to CM features
and visible feature status indications.
Call Coverage (6 levels)
Call Coverage provides automatic redirection of calls that meet specified criteria to alternate
answering positions in a call coverage path. Lead coverage paths can be administered to apply to all
calls at the same time, internal or external calls, or to apply to a specific day of the week or a specific
time of day. Different coverage paths are administered based on incoming call origination, type or
time. Building on the concept of call forwarding, personal call coverage programming redirects to a
defined path of answering deskphones and will default to the called party’s voice mailbox only as a
last resort. Calls will not be redirected to the forwarding position or voice mailbox of a station user
defined in the call coverage path; the originally called party’s coverage path overrides intermediary
station user call forwarding commands. The objective of personalized call coverage features is to
reduce dependency on voice mail systems because a human answering station rather than a noninteractive machine might be preferred by the caller. Call coverage is supported for 6 levels in CM,
and is available for both the deskphones covered for and as coverage points.
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Feature Summary
Caller ID (Name and number)
Caller ID is used to interpret calling party information that is signaled over ISDN, H.323 or SIP trunks,
and displaying the calling party number and calling party name on the deskphone. Caller ID is also
known as Incoming Call Line Identification (CLID). Name and number is also displayed for internal
callers as well.
Call Forward (All, busy, don’t answer, disable)
Call forwarding redirects any incoming calls to another destination. Deskphones can forward calls for
the following: all calls, calls when station is in use (busy), and calls that go unanswered. In addition,
users can enable or disable call forwarding from their deskphones.
The system forwards a call only once. For example, assume that extension A designates extension B
as its forwarded-to destination, and that extension B designates extension C as it’s forwarded-to
destination. When someone calls extension A, the system first attempts to ring the call at extension A.
If the system is unable to ring the call at extension A, the system attempts to ring the call at extension
B. If the system is unable to ring the call at extension B, the system redirects the call to the coverage
path of extension A, if a coverage path is available at extension A, and if the coverage criteria of
extension A are satisfied when applied at extension B. The system does not forward the call to
extension C under any circumstances.
Call Hold, Resume
Deskphone users use the “Hold” soft button to hold a call and the “Resume” soft button to un-hold the
call. The system holds the call at the call appearance that is used for the call. Deskphone users can
hold a call on each call appearance.
Call Log (Missed/Answered/Outgoing calls, Call/Delete/Details)
Call Log stores dialed station numbers and incoming identification numbers [internal CLID (Calling
Line Identification), ANI (Automatic Number Identification)]. The numbers that are stored are those of
the most recently dialed and incoming calls. Pressing a call log button brings up the display. Calls to
numbers appearing in the call log display field can be dialed automatically through menu control keys
and entries may also be deleted via the menu control keys.
Users can also add a call log entry to contacts/phonebook. The call log stores up to 100 calls. For
H.323 deskphones, the call log can be backed up to an HTTP server and loaded whenever a user
logs in, providing access to the same call history information from any H.323 deskphone in the
enterprise. For SIP deskphones, the call log is maintained locally.
Call Log (Busy)
Call logs for H.323 deskphones and SIP 6.5.x deskphones will show missed calls for calls which are
not presented to the deskphone due to all appearances being busy. Call logs for SIP 2.6.x
deskphones do not contain such entries.
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Feature Summary
Call Log (Offline)
Call logs for H.323 6.6.x and SIP 6.5.x deskphones which were off-line when a call was received will
appear in the log when the deskphone logs back in. Call logs for H.323 3.2.x deskphones which were
off-line when a call was received will appear in the log when the H.323 deskphone logs backs in if
CES is used. Call logs for SIP 2.6.x deskphones do not contain such entries.
Call Park, Answer Back
Call park and answer back are used to park a held call and retrieve it later from any other deskphone
within the system. For example, a user can answer a call at one extension, put the call on hold, and
then retrieve the call at another extension. Or the user can answer a call at any deskphone after an
attendant or another user pages the user. Deskphones can park a call either using the FAC or the
“call-park” feature button. Calls are un-parked using the answer back FAC, or the “call-park” button.
Calling Party Number Block, Unblock
Calls from to outside trunks can block the calling party number for outside calls using either a FAC or
feature buttons on the set. Deskphone users can invoke either a block or unblock operation before
each call made.
Calling Party Number Block, Unblock of Internal Numbers
A typical Communication Manager internal call generally provides calling/called party numbers and an
administered text name string which are displayed on the involved parties’ terminals. The internal
calling party block feature provides stations the capability (via proper COR administration) of masking
off the calling party name and number and replacing it with a “hard-coded,” system-wide text string,
"Info Restricted," which will be displayed on the called party‘s terminal. In addition, the feature also
provides a way of administering so the CPN/Name information not be masked-off when the call is
made. Deskphone users can either block or unblock the CPN and name information with either the
FAC codes or feature buttons on the set.
Call Pickup
Call Pickup allows users to answer calls for one another. The Call Pickup feature requires that users
be members of the same pickup group. With Call Pickup, the administrator creates one or more
pickup groups where a pickup group is a collection, or list, of individual deskphone extensions.
A user extension can belong to only one pickup group.
To pick up a call, the user enters the feature access code (FAC) of the call pickup feature and then
the call pickup group number of the ringing call. Alternatively, the user can press the “call-pkup” call
pickup button of the group that has been assigned to his/her deskphone to pickup the call.
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Feature Summary
Class of Service
COS is used to allow or deny user access to some system features for deskphones, such as:
 Automatic Callback
 Call Forwarding
 Data Privacy
 Trunk-to-Trunk Transfer Override
 QSIG Call Offer Originations
 Contact Closure Activation
 Console Permission
Click to Conference
Click to conference is only supported for H.323 deskphones. Conferencing is accomplished by
highlighting the appropriate contact in the contact list.
Code Calling
One type of loudspeaker paging is “code calling” or chime paging. If frequent voice pages are
undesirable, you can assign a unique series of chimes, or a chime code to each extension. The chime
code assigned to that extension plays over the speakers when that extension is paged.
Conference (Ad-Hoc – 6 Party)
Ad-hoc conferences of up to 6 parties can be created.
Contact Center (General)
Communication Manager has many features that are associated with contact centers or ACD
(automatic call distribution). Examples of these are: agent login logout, after call work, aux work, etc.
These features are not supported with SIP 2.6.x software.
Contacts (Add/Edit/Delete/Details)
Deskphones provide access to a maximum of 250 contacts (phonebook entries) with 3 (H.323) or 6
(SIP) numbers each. Changes to contacts are automatically backed up to a server (PPM for SIP,
HTTP for H.323 if configured) and loaded whenever a user logs in.
Core Redundancy
Deskphones can connect to multiple IP addresses in the core allowing a core element or network
connectivity failure without sacrificing service to the endpoint. H.323 deskphones do this by allowing
connections to multiple CLANs or through connecting directly with the Processor Ethernet (PE) server
interface. SIP deskphones allow for connection to two core Session Managers, and one Survivable
Remote.
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Feature Summary
Crisis Alert
The Crisis Alert capability notifies the attendant, and up to 10 other designated users when someone
dials an emergency number (like 911 in the US). When a user dials an emergency number, the
system sends both an audible alert and a visual alert to the attendant console and phones designated
for crisis alert displays. Both H.323 deskphones and SIP deskphones can make emergency calls that
are viewable using crisis alert. H.323 deskphones can be used to view crisis alert information. SIP
deskphones cannot be used to view crisis information.
Directed Call Pickup
With Directed Call Pickup, users can specify what other deskphone they want to answer. Call pickup
groups are not needed with Directed Call Pickup. With directed call pickup, a user can answer an
incoming call on another deskphone by entering the Directed Call Pickup feature access code (FAC)
followed by the extension of the ringing call. Alternatively, the user can use the “dir-pkup” directed call
pickup button on the deskphone followed by the extension to pickup a call ringing on another
extension.
Directory (Aura Integrated)
Avaya Aura® has an integrated directory which contains the names and extensions of all users which
are provisioned on the system.
H.323 deskphones access this directory via the “Directory” softkey. Deskphones with SIP 6.5.x
software access this directory via the “Contacts” button followed by “Search” softkey. This capability
is not supported on Deskphones with SIP 2.6.x software. Once the correct name is found, a call
can be made to the associated number or the name/extension can be added to the local contacts.
Directory (LDAP)
Deskphones can access the entire enterprise name and number database for calling using the LDAP
integration. Avaya distributes a script that can be installed on a standard web server to provide
access from the WML browser on the deskphones to an enterprise LDAP directory. This feature is
also available from the Utility Server Template as a “Directory Application”.
Distinctive Ringing
Distinctive Ringing provides different ringing patterns for different types of calls. Deskphones
configured to use either “Classic” or “European” ringtones will ring with different patterns.
Deskphones configured to use either “Rich” or downloaded ringtones will ring with the same pattern
irrespective of the type of call.
Emergency Button (one touch access)
The Emergency button on deskphones provides one-touch access to make an emergency call even
when the phone is logged out, registered but inactive, or locked.
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Feature Summary
Enhanced Call Forwarding
Enhanced Call Forwarding allows a deskphone to forward calls to different locations based upon the
status of the phone at the time that the call is received as well as the source (internal/external) of the
call. There are six types of Enhanced Call Forwarding:
 Unconditional – internal calls
 Unconditional – external calls
 Busy – internal calls
 Busy – external calls
 No Reply- internal calls
 No Reply – external calls
Deskphone users can activate or deactivate any of these types from their phone, and can specify
different destinations for each type of call. Deskphone users with SIP 6.5.x software can program all
six from a single screen.
Enhanced Group Call Pickup Alerting
The enhanced call pickup button on a deskphone can be administered with different alerting options
for the group members including, silent, single alert, continuous, delayed, and abbreviated.
This feature only works for the call pickup group button with enhanced call pickup alerting turned on
(in the system features form).
Exclusion
Exclusion or “manual” exclusion allows a user of deskphones to keep others with bridged line
appearances of the same extension from bridging onto an existing call. The user presses the
exclusion button either before the user places the call or when the user is active on the call to keep
others from bridging onto the call. If the user presses the exclusion button while others are bridged
onto the call, CM will drop the other users from the call. To turn off manual exclusion, the user
presses the exclusion button.
Extended Call Pickup
With Extended Call Pickup, users in one pickup group can answer calls that come in for users in
another pickup group. Extended Call Pickup allows the administrator to define one or more extended
pickup groups and calls are “picked-up” by entering the Extended Call Pickup feature access code
(FAC) and the 1-2 digit number to indicate the group of the ringing call to be picked up. There is no
feature button that can be administered as an “extended call pickup” button, but an abbreviated dial
button can be administered with the Extended Call Pickup FAC code for faster operation.
Extension to Cellular (EC500)
Extension to Cellular (EC500) allows a deskphone and associated cellular phone to ring
simultaneously when there is an incoming call. It also allows users to “extend” an active call on their
deskphone to their cellular phone.
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Feature Summary
Favorite Button
On SIP deskphones, buttons that appear on the “features” screen or other screens than the main
screen can be programmed at the phone to appear on the main screen. This is quite useful for
features that visibly alert or are used often like call pickup, EC500, extend call, etc. H.323
deskphones do not have this capability.
Feature Name Extensions (FNE)
Once Extension to Cellular is enabled, a cell phone user can activate certain Communication
Manager features by dialing a feature name extension (FNE). FNEs correspond to a direct inward
dialing (DID) number for each feature. The administrator creates the FNEs. The FNEs can be
administered on a per-CM basis.
Users with SIP deskphones can use Extension to Cellular FNEs to activate features on their SIP
deskphones like active appearance select, activate/deactivate SAC, etc.
Flexible Language Displays
Flexible language displays allows the deskphone to display any one of approximately 15
languages. For H.323, this also includes that ability (with a downloadable tool) to customize the
words and phrases displayed by the UI for additional languages.
Forced Entry of Account Codes
Forced Entry of Account Codes requires the entry of an account code for every call. The following
can be administered:
 All users enter an account code for all calls
 All users enter an account code for calls that are made on a specific trunk.
 All users enter an account code for calls that are made to a specific deskphone number
 A specific user enters an account for all calls that are made by that user
With Forced Entry of Account Codes (FEAC), the system rejects any call a user makes without an
account code FAC, if the call requires an account code. When the system rejects the call, the user
hears intercept tone.
Guest Login
Guest Login allows users to temporarily log into an already-logged-in deskphone. When they
subsequently log out, the deskphone reverts to the previously logged-in extension. Guest Login is
only supported on H.323 deskphones.
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Feature Summary
Group Paging
With the Group Paging feature, you can create a page group, and assign extensions as members of
the group. You assign an identifying extension to each page group, which users dial to page the
group. When a user dials the extension of the paging group, Avaya Communication Manager
activates the speakers on all the deskphones in the group and each user can hear what the “pager”
speaks. Group Paging is one-way communication: Group members hear the person place the page,
but cannot respond directly.
Deskphones with SIP 2.6.x software can initiate a page but cannot receive a page.
Hold Recall
Calls that are held by a user will return to the deskphone that held the call after a fixed period of time
(the long hold recall timer). When the call returns to the deskphone that held the call, the deskphone
alerts and the display indicates the call has exceeded the long hold recall timer.
Hospitality
A set of features used in the hotel or hospitality industry. Examples include “Automatic Wakeup” and
“Do not Disturb”, etc. These work for H.323 deskphones, but are not operational for SIP
deskphones.
Hotline
Hotline is the ability to place calls to pre-arranged extensions without entering the extension of the
destination with the keypad. H.323 deskphones support this feature. SIP deskphones with SIP 6.5.x
require the use of the 46xxsettings file to administer the deskphone for hotline. SIP deskphones with
SIP 2.6.x software do not support this feature.
Hunt Group Busy Button
The “Hunt Group Busy Button” allows the user of the deskphone to opt in/out of a hunt group and
includes a visual indication of their status in that hunt group. This button is supported with SIP
deskphones running SIP 7.0.1 or later software.
Intercom - Automatic
Automatic intercom is a feature where a button is administered that calls a predefined extension when
the button is pressed. An intercom call makes a unique alerting sound. If the deskphone has an
intercom button with a status lamp, the lamp also flashes.
To control which users can make intercom calls to each other, the user deskphones are placed in
groups called "intercom groups." Once you add a set of deskphones to the group, users can make
intercom calls by administering an automatic intercom button on their deskphone.
Intercom – Dial
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Feature Summary
Dial intercom allows one user to call another user in the same intercom group by pressing the dial
intercom and one or two other digits. An intercom call makes a unique alerting sound. If the
deskphone has an intercom button with a status lamp, the lamp also flashes.
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Feature Summary
Last Number Dialed (LND)
The Last Number Dialed feature redials the same number a user just dialed. LND can be set to 'off',
'redial last number' or 'redial from a list'. If it is set to 'redial last number', pressing the redial softkey
will cause the phone to redial the last number dialed; if it is set to 'redial from list', the phone will
display a list of the 3 or 6 (a full screen's worth) of the most recently dialed numbers so the user can
select one of them to call.
Limit Number of Concurrent Calls
Limit Number of Concurrent Calls (LNCC) is used to keep one Call Appearance idle such that it can
be used for outgoing calls. When the LNCC feature is enabled and all but one Call Appearances are
busy, subsequent incoming calls receive a busy signal (no coverage path) or follow the coverage path
if administered. LNCC is activated and deactivated by a feature button (limit-call), or by using two
new Feature Access Codes (Limit Number of Concurrent Calls Activation/Deactivation). The limit-call
button indicates visually whether the LNCC feature is active or not.
LNCC is not supported on SIP 2.6.x deskphones.
Local Survivability (Survivable Remote)
Local survivability is accomplished with Survivable Remote for full featured survivability for H.323 and
SIP deskphones.
Local Survivability with Third Party Gateways
Local survivability is accomplished with third party gateways for SIP deskphones. Avaya SIP
deskphones will connect and register alternatively with select third party gateways and receive basic
functionality when the branch is disconnected from the core Session Manager servers.
Local Survivability with IP Office Centralized
IP Office 9.0 introduced a Centralized configuration. SIP deskphones will register and receive “sunny
day” service from a core Avaya Aura® system, and connect and receive basic “rainy day” functionality
from the IP Office when connectivity is lost.
Loudspeaker Paging
Loudspeaker Paging provides deskphone users and/or attendants dial access to voice paging
systems. This is useful for paging purposes regardless of the deskphone user’s location within the
premises environment. It is often used with the call park feature. The system can provide as many
as 9-individual paging zones. In addition, 1-zone can be provided by the system to activate all zones
simultaneously. Deskphones users can activate paging by pressing Abbreviated Dialing buttons.
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Feature Summary
Loudspeaker Paging - Deluxe
Deluxe Paging is similar to Loudspeaker Paging, but Deluxe Paging adds the convenience of
automatically parking a call. When one user is away from their desk and receives a call, another user
can answer the call, and send out a page. To page and park an active call simultaneously, users
press Transfer, dial the trunk access code + an extension number where the call will be parked, make
the announcement, and press Transfer again. The called party dials the Answer-Back TAC to retrieve
the call. Deluxe Paging also provides Meet-Me Paging and Meet-Me Conferencing; a user pages
another party and adds the party onto a conference call when they call the paging party. Note that
without Deluxe Paging, paging and parking are two separate operations.
Malicious Call Trace (MCT) - Activation
A user can use either a feature button (mct-act) or a Feature Access Code (FAC) to activate MCT.
Either the recipient of the call, or another user or attendant, can activate MCT. When a user or an
attendant activates MCT, the system notifies potential MCT controllers. A potential MCT controller is
a user or an attendant who is assigned a feature button (mct-contr) that can control MCT. The
controller can direct the call to be recorded, and Communication Manager will output a record of the
malicious call.
Malicious Call Trace (MCT) - Controller
An MCT controller is a user or an attendant who is assigned a feature button (mct-contr) that can
control MCT. The first user or attendant who presses an mct-contr button becomes the MCT
controller for the call, and the system stops alerting other potential MCT controllers. The display of
the MCT controller shows the information for the traced MCT call.
This feature is not supported with SIP deskphones.
Manual Signaling
The manual signaling feature allows one user to signal another user. When a user presses the
manual signaling button, the other user hears a 2-second ring. The status lamp of the user who
presses the button lights for two seconds. If the deskphone of the intended recipient of the signal is
already alerting, the system:
 Does not generate the 2-second ring
 Causes the manual signaling button lamp of the user who presses the button to flicker
briefly
This feature is not supported with SIP deskphones.
Meet-Me Conferencing – CM
The Meet-me Conference feature is used to set up a dial-in conference of up to six parties. Users can
dial into the conference.
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Feature Summary
Message Retrieval (one button)
Pressing the "Message" button on a deskphone will connect the user to the enterprise voicemail
system to retrieve their voice mail.
MLPP TDM Trunking
The Multiple Level Precedence and Preemption (MLPP) feature allows users to request priority
processing of calls during critical situations and select or pre-empt TDM trunks.
Multiple Call Handling, Multiple Lines, Multiple Call Appearances
Deskphones can have multiple line appearances and handle multiple calls on these appearances
simultaneously.
Multiple Device Access
Multiple Device Access (MDA) allows up to ten SIP stations to be registered to the same username
and password. A deskphone with SIP 2.6.x software can be at least one of those stations. A
deskphone with SIP 6.5.x software can be at least one of those stations and additionally provides a
similar user experience if only one device is registered or if multiple devices are registered. For more
details on MDA and the limitations associated with that feature, please refer to the “Avaya Aura® Multi
Device Access White Paper”.
Multi-Language
The 9600-series IP Deskphones support an extended list of languages for display and text input. SIP
deskphones support a few more languages for text input than H.323 sets. The languages supported
for display are: Arabic, Simplified Chinese, Dutch, English, French (Canadian and Parisian), German,
Hebrew, Italian, Japanese, Korean, Portuguese, Russian, Spanish (Castilian and Latin American).
Arabic is not supported on the 9601/9608/9608G.
Night Service
Night Service is used to direct incoming calls to other answering points at night. Night service can be
provided for trunk groups, attendants, hunt groups, ACD groups, etc.
One Touch Recording for Modular Messaging (OTR)
OTR is used to record deskphone conversations by pressing a single button on an a deskphone. OTR
uses Modular Messaging to record a deskphone conversation. A user needs to press only one
feature button on the deskphone to activate OTR. OTR then stores the recorded conversation as a
message in the voice mailbox of the user. The system allows OTR only after a call is answered.
one-X Communicator – Deskphone Mode
one-X Communicator Deskphone Mode allows users of one-X Communicator to make or receive
phone calls on an associated IP Deskphone.
This feature is not supported on IP Deskphones with SIP 2.6.x software.
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Feature Summary
one-X Mobile Integration
one-X Mobile allows users of deskphones to add the capabilities of a mobile handset for a single,
enterprise extension.
one-X Portal Integration
one-X Portal is a browser-based tool for making and receiving calls remotely, when access to the
deskphone is either not possible or not required. one-X portal may be accessed by users of H.323
and SIP deskphones.
PIN Checking
PIN checking requires a deskphone user to enter a PIN before dialing. The redial log does not
contain the PIN so that redials will not work to keep users from dialing without knowing the PIN. This
feature works only for H.323 deskphones.
Presence - Advertise
SIP and H.323 deskphones provide their presence status to Avaya Aura® Presence Services.
Presence - Display
The display of the presence status of users in the local Contact list is supported on SIP deskphones
only. The presence status is shown in the Contact list, Call Log, Conference Roster, and Home
screen (9621G/9641G only).
Priority Calling
The Priority Calling feature is used by deskphones to provide a special type of call alerting between
internal users, including the attendant. The called party hears a distinctive ringing when the calling
party uses priority calling.
Pull Transfer
Pull Transfer is used to allow either the transferring party or the transferred-to party to press the
Transfer button to complete the transfer operation. To use Pull Transfer, calling parties and called
parties must be on the same Communication Manager.
Pull Transfer is not supported with SIP deskphones.
Remote Worker - SBC
SIP deskphones can be used in remote offices connecting to the main Avaya Aura® system over the
Internet by pairing it with an SBC Session Border Controller (either ACME or ASBCE). The SBC
allows only authorized users to connect and all signaling and media can be encrypted while it is
traversing the Internet.
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Feature Summary
Remote Worker – VPN Integrated
VPN Remote Access via IP-SEC is supported in H.323 deskphones providing customers the ability to
use those phones in the home or small office locations and leverage high speed internet to connect to
the main location.
Ring Tones – Classic, European, Rich, Downloaded
All deskphones users can choose between eight ringtones. Deskphones with H.323 6.6.x or SIP 6.5.x
software can optionally use either eight “Classic” or eight “European” ringtones. 9670G users or
Deskphones with SIP 6.5.x software can additionally choose another six “rich” ring tones. SIP 6.5.x
software also supports the ability to download up to 40 additional ringtones from the HTTP/HTTPS
provisioning server.
Ringing Control - Bridged Line
Administrators can individually set the ringer settings for each line bridged line appearance. Ringers
can be set to audibly alert or not.
Ringing Control – Per Contact
Users of deskphones with SIP 6.5.x software can program any of the available ringtones against each
of the six numbers associated with their Contacts.
Ringing - Abbreviated and Delayed
The Ringing - Abbreviated and Delayed feature is supported on H.323 deskphones only and has two
categories of ringing:
Ringing that alerts consistently and does not change:
- Ringing, in which the lamp flashes and audible ringing occurs
- Silent ringing, in which the lamp flashes and audible ringing does not occur
Ringing that transitions from one ringing state to another:
- Abbreviated ringing, in which ringing continues for the number of cycles that you
specify with the automatic abbreviated transition interval or the delayed transition
interval, and then changes to silent alerting
- Delayed ringing, in which visual alerting continues for the number of cycles that you
specify with the automatic abbreviated transition interval or the delayed transition
interval, and then changes to ringing
The Ringing - Abbreviated and Delayed feature is most useful in bridging situations in which some
users want to:
● Have a call audibly alert as soon as the call arrives
● Be audibly notified if the call is unanswered within a specified number of rings
● Stop the audible alerting
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Feature Summary
Send All Calls
Send All Calls allows users to temporarily divert all incoming calls to coverage regardless of the
assigned call coverage redirection criteria. The feature also allows covering users to temporarily
remove their voice terminals from the coverage path. The feature button on the phone will light when
the Send All Calls feature is active. A button may be created for the deskphone on which it is
administered (own station) or for another station (other station) so that an assistant can activate and
monitor Send All Calls for a boss for example.
Service Observing
With Service Observing, designated users who are usually supervisors can listen to other user calls.
H.323 and SIP deskphones’ calls can be observed. H.323 deskphones or deskphones with SIP 7.0.0
or later software can be used to observe other calls. Note that Call Center Elite 7.0.1 is also required
to allow the use of SIP Service Observe. Deskphones with SIP 2.6.x software cannot be used to
observe other calls.
Simulated Bridged Appearance
Calls that are redirected to call coverage maintain a simulated bridged appearance on the called
deskphone if a call appearance is available to handle the call. The called party can bridge onto the
call at any time. The system can be administered to allow a simulated bridged appearance of the call
to either remain at, or be removed from, the covering deskphone after the principal bridges onto the
call. If two parties are bridged together on an active call with a third party, all three parties hear the
bridging tone.
Station On-Hook Dialing
On-hook dialing is the ability to begin a call by merely pressing the dialed digits without going off hook
on the receiver or pressing the speaker phone button on the handset. The phone will automatically
activate the speakerphone with on-hook dialing. On-hook dialing is useful for users that require a
minimum number of button presses to initiate an outgoing call.
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Feature Summary
Team Button
The Team Button has two generic functions, a display function and an execution function. The display
function will allow any member of a team (monitoring station) to observe the station state of other
team members (monitored station). The indication is done either via the green and red button LED or
as an icon on the display, depending of the type of the station. In addition the ringing of a call on a
monitored station may be indicated with audible ringing on the monitoring station. Furthermore active
call forwarding from the monitored station to the monitoring station and active send all calls path at the
monitored station with the monitoring station as first destination in the coverage path will be displayed.
As an execution function, the Team Button can be used as Speed Dial Button or Pick-Up Button.
Depending on the state of the monitored station, when the Team Button on the monitoring station is
pushed, a call to the monitored station is established directly or a ringing call is picked from the
monitored station.
Team Button is not supported on SIP deskphones with 2.6.x software.
Temporary Bridged Appearance (TBA)
When a call is made to an individual that is part of a pickup group, one particular member of the group
can be the most qualified person to handle the given call. If this individual does not answer the call
originally, this individual can bridge onto the call using a temporary bridged appearance. The
answering party does not have to transfer the call. A call to an individual can be answered by a
member of a call pickup group and while the call is still connected, the called party can bridge onto the
call, and the answering party hangs up. TBAs can be administered system-wide to either on or off
using the system-parameters features form on CM.
Time of Day Routing
Time of Day Routing can be used to redirect calls to coverage paths according to the time of the day
and the day of the week. Calls can be routed based on the least expensive route according to the
time of day and the day of the week that the call is made. You can also deny outgoing long distance
calls after business hours to help prevent toll fraud. Time of Day Routing applies to all AAR or ARS
outgoing calls and trunks that the system uses for call forwarding to external numbers.
Traffic Measurements
Both H.323 and SIP deskphones can be covered by the traffic measurements used in SM and CM to
record traffic levels, usage on trunks, etc.
Transfer (Attended, Unattended)
The Transfer feature is used to allow deskphone users to transfer trunk calls or internal calls to other
deskphones or trunks without attendant assistance. Attended transfers are accomplished by making
connection with the final destination of the call before completing the transfer; unattended transfers
are completed before the final destination station answers the call.
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Feature Summary
Transfer on Hang-up
Use “Transfer Upon Hang-up” to transfer a call without the need to press the Transfer button a second
time. The user presses the Transfer button, dials the number to which the call is being transferred,
and then hangs up the receiver. Transfer upon hang-up is an optional capability administered at the
system level. Even when transfer upon hang-up is administered, users can still press the transfer
button a second time to transfer the call.
This feature is not supported with SIP deskphones with 2.6.x software.
Transfer to Voicemail
Transfer to voicemail allows a user to transfer a live (or alerting) call to a designated voice mail box
with the press of a deskphone button. This is useful for assistants who answer calls for others to
easily enable callers to leave voice messages for the called parties when they are not available.
H.323 deskphones allow the administration of the transfer to voicemail button, but SIP deskphones do
not. Also, SIP deskphones do not allow transfer to voicemail while a call is alerting, only after the call
is answered can a SIP deskphone use the transfer to voicemail button.
VIP Calling
Administration of the Class of Service on CM enables automatic priority calling when assigned to the
originator of the call. Thus, a “VIP” can be set so that every call he/she makes to non VIP
deskphones will automatically become a priority call.
Voice Initiated Dialing (VID)
A user can speak a name to search for and call any contact when voice dialing is enabled. A user can
optionally add a qualifier like "at home" or "mobile" with the name to get to a specific number for the
contact. The first two times a user uses voice dialing, a help screen displays to assist in using the
feature. VID is not supported on SIP deskphones. VID is not supported on H.323 deskphones
with 6.3.0 or later software.
Web Browser
Deskphones include a WML browser which includes support for the Avaya Web API.
Whisper Page Activation
By activating whisper page, only the person on the paged extension can hear the “page” (an assistant
“whispering” a message to the paged party). Other parties on the call cannot hear the page, and the
person who activates the page cannot hear anyone on the call. If the paged user has an H.323 or SIP
deskphone, the paged user can see who makes the whisper page. Both H.323 and SIP deskphones
can activate and receive whisper pages..
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Feature Summary
Notice
While reasonable efforts were made to ensure that the information in this document was complete
and accurate at the time of printing, Avaya Inc. can assume no liability for any errors. Changes and
corrections to the information in this document may be incorporated in future releases.
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