Administration of Avaya Communication Manager
Administration of Avaya Communication Manager
Administration of Avaya Communication Manager
Communication Manager
Release 8.1.x
Issue 10
March 2021
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Contents
Troubleshooting TTI....................................................................................................... 122
Removing telephones.......................................................................................................... 124
Adding a fax or a modem..................................................................................................... 125
Enabling transmission over IP networks for modem, TTY, and fax calls.................................... 126
IP Softphones..................................................................................................................... 127
Enabling the system to use IP softphone......................................................................... 128
Road Warrior Mode....................................................................................................... 128
Adding a softphone in telecommuter mode...................................................................... 130
Troubleshooting IP softphones....................................................................................... 131
IP Telephones..................................................................................................................... 131
Adding an IP telephone.................................................................................................. 132
Changing from dual-connect to single-connect IP telephones............................................ 132
Setting up emergency calls on IP telephones................................................................... 134
Remote office setup............................................................................................................. 135
Adding Remote Office to Communication Manager.......................................................... 135
Setting up a trunk group................................................................................................. 137
Setting up a signaling group........................................................................................... 137
Setting up Remote Office on network regions.................................................................. 138
Adding telephones to Remote Office............................................................................... 139
Downloading firmware to multiple stations....................................................................... 139
Displaying firmware download status.............................................................................. 140
Disabling firmware downloads........................................................................................ 141
Native Support of Avaya 1408 and 1416 digital telephones............................................... 141
Native support for 96x1 H.323 and SIP deskphones......................................................... 142
Native support for Avaya J100 Series IP Phones............................................................. 142
Native support of Avaya 9404 and 9408 digital telephones............................................... 143
Administer location per station.............................................................................................. 144
Preparing to administer location number on Station screen............................................... 144
Setting up location number on Station screen.................................................................. 144
Chapter 7: Telephone Features........................................................................................... 145
Adding feature buttons......................................................................................................... 145
Increasing Text Fields for Feature Buttons............................................................................. 146
Enabling extended text fields for feature buttons.................................................................... 146
Restricting customization of feature button types.................................................................... 147
Telephone feature buttons table............................................................................................ 148
Abbreviated Dialing Lists...................................................................................................... 167
Setting up a station to access a new group list................................................................. 167
Adding Abbreviated Dialing Lists.................................................................................... 168
Troubleshooting abbreviated dialing lists......................................................................... 169
Bridged Call Appearances.................................................................................................... 170
Setting Up Bridged Call Appearances............................................................................. 171
Enabling Enhanced Bridged Call Appearance.................................................................. 172
When to use Bridged Call Appearances.......................................................................... 173
Extension to Cellular............................................................................................................ 173
Extension to Cellular Setup Table................................................................................... 173
Setting Up Extension To Cellular Feature Access Button.................................................. 175
Terminal Self-Administration................................................................................................. 176
Setting Up Terminal Self-Administration........................................................................... 177
Fixing Problems in Terminal Self-Administration............................................................... 178
Enterprise Mobility User....................................................................................................... 178
System Requirements — EMU....................................................................................... 179
Configuring your System for the Enterprise Mobility User.................................................. 180
Setting EMU options for stations..................................................................................... 180
Defining options for calling party identification.................................................................. 181
Activating EMU............................................................................................................. 181
Deactivating EMU.......................................................................................................... 182
Chapter 8: Managing Attendant Consoles......................................................................... 183
Attendant Consoles............................................................................................................. 183
302A/B Console............................................................................................................ 185
302C Console............................................................................................................... 186
302D Console............................................................................................................... 187
Adding an Attendant Console............................................................................................... 188
Attendant Console Feature Buttons...................................................................................... 189
Setting Console Parameters................................................................................................. 194
Removing an Attendant Console.......................................................................................... 195
Providing Backup for an Attendant........................................................................................ 196
Chapter 9: Managing Telephone Displays.......................................................................... 198
Displaying administration..................................................................................................... 198
Displaying ANI Calling Party Information............................................................................... 198
Displaying ICLID Information................................................................................................ 199
Setting the Display Language............................................................................................... 199
Administering Unicode Display....................................................................................... 200
Unicode Native Name support........................................................................................ 202
Fixing Problems............................................................................................................ 206
Related Topics.............................................................................................................. 206
Setting the Directory Buttons................................................................................................ 206
Chapter 10: Handling Incoming Calls................................................................................. 208
Basic Call Coverage............................................................................................................ 208
Administering system-wide call coverage characteristics.................................................. 208
Advanced call coverage....................................................................................................... 211
Covering calls redirected to an off-site location................................................................. 211
Defining coverage for calls redirected to external numbers............................................... 212
Defining time-of-day coverage........................................................................................ 213
Creating coverage answer groups.................................................................................. 214
Call Forwarding................................................................................................................... 214
Determining extensions having call forwarding activated................................................... 215
Adding a media-server................................................................................................... 377
Verifying that the media-server is in-service..................................................................... 378
Removing a media server.............................................................................................. 379
®
Managing Avaya Aura Media Server related documents....................................................... 379
Chapter 18: Telephone announcements............................................................................. 381
VAL or Gateway Virtual VAL resources.................................................................................. 381
Chapter 19: Managing Group Communications................................................................ 384
Voice Paging Over Loudspeakers setup................................................................................ 384
Preparing to set up Voice Paging Over Loudspeakers...................................................... 384
Setting Up Voice Paging Over Loudspeakers example..................................................... 384
Loudspeaker Paging troubleshooting.............................................................................. 385
User considerations for Voice Paging Over Loudspeakers................................................ 386
Chime Paging Over Loudspeakers setup............................................................................... 386
Preparing to set up Chime Paging Over Loudspeakers..................................................... 387
Setting up Chime Paging Over Loudspeakers example.................................................... 387
Assigning chime codes example..................................................................................... 388
Chime Paging Over Loudspeakers troubleshooting.......................................................... 388
User considerations for Chime Paging Over Loudspeakers............................................... 388
Speakerphone paging setup................................................................................................. 389
Preparing to set up speakerphone paging....................................................................... 389
Setting up speakerphone paging example....................................................................... 389
Speakerphone paging troubleshooting............................................................................ 390
Speakerphone paging capacities.................................................................................... 390
Whisper Paging users who are on active calls....................................................................... 390
Preparing to set up Whisper Paging................................................................................ 390
Whisper Paging setup.................................................................................................... 391
Telephones as Intercoms administration................................................................................ 391
Administering intercom feature buttons example.............................................................. 392
Administering an intercom group example....................................................................... 393
Automatic Answer Intercom Calls setup................................................................................ 393
Administering Auto Answer ICOM example..................................................................... 393
Service Observing Calls....................................................................................................... 394
Preparing to set up Service Observing............................................................................ 394
Setting up Service Observing example............................................................................ 395
Best practices for service observing................................................................................ 396
Chapter 20: Managing Data Calls........................................................................................ 397
Types of Data Connections.................................................................................................. 397
Data Call Setup................................................................................................................... 397
Data Call Setup Administration....................................................................................... 398
DCP data modules........................................................................................................ 400
ISDN-BRI data modules................................................................................................. 402
Analog modems............................................................................................................ 403
Considerations for Data Call Setup................................................................................. 403
External Numbers.......................................................................................................... 457
Chapter 24: Communication Manager objects................................................................... 459
Chapter 25: Endpoints.......................................................................................................... 460
Chapter 26: Templates.......................................................................................................... 461
Chapter 27: Overview of Inventory Management............................................................... 462
Chapter 28: Messaging......................................................................................................... 463
Subscriber Management...................................................................................................... 463
Adding a subscriber............................................................................................................. 463
Editing a subscriber............................................................................................................. 464
Viewing a subscriber............................................................................................................ 464
Deleting a subscriber........................................................................................................... 464
Subscriber list..................................................................................................................... 465
Filtering subscribers............................................................................................................ 465
®
Subscribers (Avaya Aura Messaging) field descriptions......................................................... 466
Chapter 29: Administering LDAP Directory Application................................................... 472
LDAP Directory Application overview.................................................................................... 472
Configuring Directory Application.......................................................................................... 472
96xx and 96x1 telephones URL configuration........................................................................ 473
Chapter 30: Administering IP DECT.................................................................................... 474
IP DECT............................................................................................................................. 474
Enabling multiple locations for IP DECT.......................................................................... 474
Verifying system capacities............................................................................................ 474
Assigning the codec...................................................................................................... 475
Configuring the network region....................................................................................... 476
Configuring the trunk group............................................................................................ 476
Configuring the signaling group...................................................................................... 477
Configuring the station................................................................................................... 478
Chapter 31: Administering SIP trunk optimization............................................................ 480
SIP trunk optimization.......................................................................................................... 480
Adding Session Managers to a cluster............................................................................ 482
Administering the number of members on a trunk group................................................... 483
Chapter 32: Certificate Management................................................................................... 484
Identity Certificates.............................................................................................................. 485
Displaying a certificate................................................................................................... 485
Addition of an identity certificate..................................................................................... 485
Adding an identity certificate for Simplex server............................................................... 486
Adding an identity certificate for Duplex server................................................................. 487
Removing an identity certificate...................................................................................... 488
Copying an identity certificate......................................................................................... 488
Trust certificates.................................................................................................................. 489
Displaying a certificate................................................................................................... 489
Adding a trusted certificate for Simplex server................................................................. 489
Purpose
This book describes the procedures and screens used in administering Communication Manager
that runs on any of the following:
• Avaya servers, HP ProLiant DL360p G8, HP ProLiant DL360 G9, and Dell™ PowerEdge™
R620, Dell™ PowerEdge™ R630.
• Avaya servers configured as a Survivable remote server, S8300E, Avaya Solutions Platform
(ASP) 120 and 130.
• Customer provided VMware Environment.
• Avaya branch gateways including G430 Branch Gateway, and G450 Branch Gateway.
Newer releases of Communication Manager contain the features of the previous releases.
This document is intended for people who perform the product or solution system administration
tasks.
Change history
Supported browsers
The following table lists the minimum supported versions of the browsers for Avaya Aura®
applications.
System login
You must log in before you can administer your system. If you are performing remote
administration, you must establish a remote administration link. You can also assign the remote
administration extension to a hunt group before you log in.
For information about setting up remote administration, do the following:
• Go to the Avaya Support website at http://support.avaya.com.
• See Administering Avaya Aura® System Manager.
When you are not using the system, log off for security reasons.
Out-of-Band management
For information about Out-of-Band management feature, see Avaya Aura® Communication
Manager Feature Description and Implementation.
Login messages
The system displays either of the two messages during login:
• Issue of the Day: Displays warnings to users about unauthorized access. The system
displays this message before a successful login.
• Message of the Day (MOTD): Informs authorized users about matters, such as
upcoming outages and impending disk full conditions. The system displays this message
immediately after a user logs in.
• cp /etc/issue.avaya /etc/issue.net
You can set up 15 customized rules for daylight-saving time. With this setting, Communication
Manager administrators with servers in different time zones can set up a rule for each server. A
daylight-saving time rule specifies the exact time when you want to transition to and from daylight-
saving time. The rule also specifies the increment at which to transition, for example, 1 hour.
Note:
Whenever you change the time of day, the time zone, or daylight saving rules, you
must reboot the server for the changes to take effect. See the documentation for
information on rebooting the server for your system.
The system displays the Daylight Saving Rules screen. Verify the information you entered
is correct.
Disabling synchronization
About this task
Use this procedure to prevent switching between clock sources.
Procedure
Type disable synchronization media-gateway n, where n is the number of the gateway.
Displaying messages
Procedure
1. Type display bulletin-board.
2. Press Enter.
Posting a message
About this task
Post a message to the bulletin board. The message can be about a problem related to a new trunk
group. A representative from Avaya will reply to your message.
Procedure
1. Type change bulletin-board.
2. Press Enter.
The Bulletin Board screen displays.
The message space within the bulletin board contains three pages. The first page has 19
lines, however you can only enter text on lines 11-19. The first 10 lines on page 1 are for
high-priority messages from Avaya personnel, and these are noted with an asterisk (*). The
second and third pages each have 20 lines, and you can enter text on any of these lines.
The system automatically enters the date the message was posted or last changed to the
right of each message line.
3. Type your message.
You can enter up to 40 characters of text for each line. You also can enter one blank line. If
you enter more than one blank line, the system consolidates them and displays only one.
The system also deletes any blank line if it is line 1 of any page. You cannot indent text on
the bulletin board. The Tab key moves the cursor to the next line.
4. Save the changes.
Deleting messages
Procedure
1. Type change bulletin-board.
2. Press Enter.
The system displays the Bulletin Board screen.
3. Enter a space as the first character on each line of the message you want to delete.
4. Press Enter.
5. Save the changes.
Save translations
Use save translation to commit the active server translations (volatile) in memory to a file
(non-volatile). The translation will either complete or fail. For Linux platforms, a filesync process
copies the translation file to the standby server.
All translation data is kept in volatile system memory or on the hard drive during normal operation.
In the event of a power outage or certain system failures, data in memory is lost. Save
translation stores on disk the translation data currently in memory.
When a SAT user issues save translation on a duplicated system, translations are saved on both
the active and standby servers. If an update of the standby server is already in progress,
subsequent save translation commands fail with the message save translations has a
command conflict.
The save translation command does not run and the system displays an error message in
the following cases:
• An administration command changes the translation data.
• The Communication Manager Web interface Pre-Upgrade Step locks the translations.
Run save translation as part of scheduled background maintenance or on demand.
For information on the save translation command and the command syntax descriptions,
see Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and
Servers.
Performing backups
For information on performing backups to your system, see Maintenance Procedures
forMaintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and
Servers.
Communication Manager consists of hardware to perform call processing, and the software to make
it run. You use the administration interface to let the system know what hardware you have, where it
is located, and what you want the software to do with it. You can find out which circuit packs are in
the system and which ports are available by entering the command list configuration. All there are
variations on this command that display different types of configuration information. Use the help
function to experiment, and see which command works for you.
System configuration
Planning Your System
At a very basic level, Communication Manager consists of hardware to perform call processing
and the software to make it run. You use the administration interface to check what hardware you
have, where it is located, and what you want the software to do with it.
You can find out which circuit packs are in the system and which ports are available by entering
the command list configuration. There are variations on this command that display different types
of configuration information. Use the help function to experiment and see which command works
for you.
To view a list of port boards on your system:
1. Type list configuration port-network.
2. Press Enter.
You will find many sections in the administration interface where you will be asked to enter a port
or slot. The port or slot is an address that describes the physical location of the equipment you are
using. A port address consists of four parts:
cabinet the main housing for all the server equipment. Cabinets are numbered starting with
01.
carrier the rack within the cabinet that holds a row of circuit packs. Each carrier within a
cabinet has a letter: A to E.
slot the space in the carrier that holds an individual circuit pack. Slots are numbered
01-16.
port the wire that is connected to an individual piece of equipment (such as a telephone or
data module). The number of ports on a circuit pack varies depending on the type.
Therefore if you have a single-carrier cabinet, the circuit pack in slot 06 would have the address
01A06. If you want to attach a telephone to the 3rd port on this board, the port address is 01A0603
(01=cabinet, A=carrier, 06=slot, 03=port).
Dial plan
Dial Plan
The system interprets dialed digits based on the dial plan. If you dial 9 on your system to access
an outside line, the system finds an external trunk for that number because the dial plan is set that
way.
The dial plan also defines the number of digits that indicate certain types of calls. For example, the
dial plan might indicate that all internal extensions are four-digit numbers that start with either 1 or
2. An example will illustrate how to read the dial plan of your system.
Note:
In Communication Manager 8.0 the maximum length of a displayed extension (including
punctuation) is 16 characters.
Dial plan access table
The Dial Plan Analysis Table defines the dialing plan for your system. The Call Type column in the
Dial Plan Analysis Table indicates what the system does when a user dials the digit or digits
indicated in the Dialed String column. The Total Length column indicates how long the dialed
string will be for each type of call.
Dial plan parameters table
The Dial Plan Analysis table and the Dial Plan Parameters table define your dial plan. You can set
system-wide parameters for your dial plan, or define a Dial Plan Parameters table according to
each location.
Uniform dial plan
To Administer a Uniform Dial Plan, you can set up a Uniform Dialing Plan that can be shared
among a group of servers. For more information, see Avaya Aura® Communication Manager
Feature Description and Implementation, 555-245-205.
As an example, add a new set of extensions that start with 3 each and are four-digit long (3000 to
3999).
Procedure
1. Go to the administration interface.
2. Enter change dialplan analysis or change dialplan analysis location n,
where n represents the number of a specific location. Press Enter.
3. Move the cursor to an empty row.
4. Type 3 in the Dialed String column. Click Tab to move to the next field.
5. Type 4 in the Total Length column. Press Tab to move to the next field.
6. Type ext in the Call Type column.
7. To save your changes, press Enter.
Location numbers
The equipment gets location numbers as follows:
• IP telephones obtain their location numbers indirectly. A location number is administered on
the IP Network Region screen that applies to all the telephones in that IP region.
• Non-IP telephones and trunks inherit the location numbers of the hardware they are
connected to, such as the cabinet, remote office, or gateway.
• IP trunks obtain their location from the location of the associated signaling group. Direct
administration, which is only possible for signaling groups for remote offices or the methods
described for IP telephones above determine the location.
Location administration
A location number administered on the IP Network Region screen applies to all telephones in that
IP region. If a Location field is left blank on an IP Network Region screen, an IP telephone derives
its location from the cabinet. The CLAN board is located in the cabinet and the telephone is
registered to it.
For information on how to administer the location per station, see the Administer location per
station on page 144 section.
For information on the description of the Location field on the Stations with Off-PBX Telephone
Integration screen, see the Avaya Aura® Communication Manager Screen Reference.
The system adds some or all the leading digits to the front of the dialed number as
specified on the Uniform Dial Plan screen. The system then routes the call based on the
analysis of the entire dialed string and the administration on the Dial Plan Parameters
screen.
For more information about DPT, see Avaya Aura® Communication Manager Feature Description
and Implementation, 555-245-205.
Packet Size 10 ms 20 ms 30 ms 40 ms 50 ms 20 ms
G.711 102 83 77 74 72 71
G.729 46 27 21 18 16 15
G.723-6.3 NA NA 19 NA NA 13
G.723-5.3 NA NA 18 NA NA 12
G.722.2 NA 43 NA 34 NA 31
These values are not significantly different from the actual bandwidth used for 8–byte L2 WAN
headers and 10–byte L2 WAN headers. In some cases, the rounded up values shown above are
greater than the values used for 10 bytes.
The bandwidth usage numbers shown above have 6 bytes for Multilink Point-to-Point Protocol
(MP) or Frame Relay Forum (FRF), 12 Layer 2 (L2) header, and 1–byte for the end-of-frame flag
on MP and Frame Relay frames for a total of 7–byte headers only. They do not account for silence
suppression or header compression techniques, which might reduce the actual bandwidth. For
other types of networks (such as Ethernet or ATM) or for cases where there is a lot of silence
suppression or header compression being used, the network is modeled by administering the
CAC-BL limits in terms of number of connections rather than bandwidth used.
Music-on-hold
Music-on-Hold automatically provides music to a caller placed on hold. Music lets the caller know
that the connection is still active. The system does not provide music to callers in a multiple-party
connection who are in queue, on hold, or parked. Avaya Aura® Media Server is used as a
repository for announcements and music sources for the Music-on-hold feature.
For more information on locally sourced Music-on-Hold, see Avaya Aura® Communication
Manager Feature Description and Implementation, 555-245-205.
Locally sourced announcements and music
The Locally Sourced Announcements and Music feature is based on the concept of audio source
groups. Use this feature to provide announcement and music sources to be located on any or all
of the Voice Announcement with LAN (VAL) boards or on virtual VALs (vVAL) in a gateway. The
VAL or vVAL boards are assigned to an audio group. The audio group is then assigned to an
announcement or audio extension as a group sourced location. When an incoming call requires an
announcement or Music-on-Hold, the audio source that is closest to the incoming call trunk plays.
Storing audio locally minimizes audio distortion because the audio is located within the same port
network or gateway as the caller. Therefore, this feature improves the quality of announcements
and music on hold. This feature also reduces resource usage, such as VoIP resources, because
the nearest available audio source of an announcement or music is played. Locally Sourced
Announcements and Music also provides a backup for audio sources because multiple copies of
the audio files are stored in multiple locations. Audio sources are assigned either to an audio
group or a Music-on-Hold group.
Audio groups
An audio group is a collection of identical announcements or music recordings stored on one or
more VAL or vVAL boards. The audio group can contain announcements and music. The nearest
recording to a call plays for that call.
With centralized SIP trunking, the chances of having the closest audio source to the caller at the
main or survivable core data centers are high. For playback to occur in survivable mode, remote
gateways must be configured with the announcement and music files. So, it is recommended that
Audio Groups are configured to ensure the solution is capable of playing announcements and
music, regardless of the survivability status of the system.
For example: if a solution consists of a main data center with media server, survivable core data
center with media server and survivable remote with gateway, then an audio group containing the
three audio source locations of a media server from each data center and the remote gateway
should be constructed. This ensures playback capability regardless of whether the solution is in
normal or rainy day mode.
Music-on-hold groups
A Music-on-Hold (MOH) group is a collection of externally connected and continuously playing
identical music sources. An example of a Music-on-Hold source is a radio station connected to a
gateway using an analog station port. Multiple Music-on-Hold sources can be used in the same
system. Like the audio group, the nearest music source to a call plays for that call.
Music-on-hold sources
As with the Music-on-Hold feature, only one music source is defined for a system or for a tenant
partition. However, you can define a music source as a group of Music-on-Hold sources.
Therefore, both non-tenant and tenant systems can use the group concept to distribute Music-on-
Hold sources throughout a system.
Procedure
1. Type change ars analysis n on Administration interface. Press Enter. The system
displays the ARS Digit Analysis Table screen.
2. In the Dialed String field, type 5555.
5555 is the number in our example that end-users dial to reach emergency services.
3. In the Total Min and Max fields, type 4.
In this example, the user must dial all 4 digits for the call to be treated as an emergency
call.
4. In the Route Pattern field, type 1.
In this example, use route pattern 1 for local calls.
5. In the Call Type field, type alrt.
The system identifies the dialed string 5555 as one that activates emergency notification.
6. Press Enter to save your changes. Now set up the attendant console to receive
emergency notification.
7. Type change attendant 1. Press Enter.
The system displays the Attendant Console screen.
8. In the feature button area, assign a crss-alert button.
9. Press Enter to save your changes.
10. Assign a crss-alert button to each security guard’s telephone.
You cannot assign this button to a soft key.
Finally, ensure that all security personnel and the attendant must acknowledge the alert.
11. Type change system-parameters crisis-alert. Press Enter.
The system displays the Crisis Alert System Parameters screen.
12. Go to the Every User Responds field and type y.
13. In the SNMP Inform to Notify Adjunct When DCP and H.323 Stations Go In-Service
field, do the following:
• If you have DCP or H.323 phones and Emergency Location Management Solution, type
y.
• If you do not have DCP or H.323 phones, or Emergency Location Management Solution,
leave this field as n.
When a DCP station comes into service, or a H.323 station registers, the Communication
Manager sends SNMP messages to a trap receiver.
14. (Optional) In the SNMP Inform to Notify Adjunct When SIP Station Dials Emergency
Call field, do the following:
• If you have configured Emergency Location Management Solution as an ELIN server
with Session Manager, leave this field as n.
• If you have Emergency Location Management Solution, but have not configured it as an
ELIN server, type y.
For more information on administering the ELIN server, see Administering Avaya Aura®
Session Manager.
When a SIP station places an emergency call, the Communication Manager sends SNMP
messages to a trap receiver.
15. Press Enter to save your changes.
Automatic hold
You can set a system-wide parameter for users to initiate a call on a second line without putting
the first call on hold. This is called Automatic Hold, and you enable it on the Feature-Related
System Parameters screen. If you do not enable this feature, the active call drops when the user
presses the second line button.
Distinctive ringing
You can establish different ringing patterns for different types of calls. For example, you can
administer your system so that internal calls ring differently from external calls or priority calls. For
more information about the Distinctive Ringing feature, see Avaya Aura® Communication Manager
Feature Description and Implementation, 555-245-205.
You can use class of restriction (COR) to define the types of calls your users can place and
receive. Your system might have only a single COR, a COR with no restrictions, or as many CORs
as necessary to affect the required restrictions.
You will see the COR field in many different places throughout Communication Manager when
administering telephones, trunks, agent logins, and data modules, to name a few. You must enter
a COR on these screens, although you control the level of restriction the COR provides.
Station Lock
Use the Station Lock feature to lock a telephone to prevent others from making outgoing calls from
the telephone. You can activate the Station Lock feature by using a button or feature access code.
You can lock and unlock the telephones remotely.
Using Station Lock, users can:
• Change their Class of Restriction (COR). The lock COR is set to fewer calling permissions
than the usual COR of the station.
• Lock their telephones to prevent unauthorized outgoing calls.
• Block outgoing calls, and still receive incoming calls.
• Block all outgoing calls except for emergency calls.
Station Lock is activated by pressing a telephone button, which lights the button indicator, or by
dialing a Feature Access Code (FAC).
Analog and XMOBILE stations must dial a FAC to activate the feature. The user hears a special
dial tone on subsequent origination attempts from the telephone to indicate that the lock feature is
active.
Digital stations including DCP, BRI, IP hardphones and softphones access Station Lock with a
feature button or through a FAC. H.323 or DCP phones support the station lock functionality of
Communication Manager. SIP phones do not support the functionality. The Station Lock feature is
activated in the following cases:
• If a digital or IP telephone has a feature button for Station Lock but uses a FAC to activate
the feature, the LED lights up. The system generates the special tone.
• If a digital or IP telephone has a feature button for Station Lock and uses this button to
activate the feature, the LED lights up. The system generates the special tone.
• If a digital or IP telephone does not have a feature button for Station Lock and uses a FAC to
activate the feature, the system generates the special tone.
You can lock or unlock a station from any other station if the FAC is used and the Station Security
Code is known. You cannot lock the attendant console but you can use it to lock or unlock other
stations. You can lock or unlock a station using a remote access trunk.
For more information about Station Lock, see Avaya Aura® Communication Manager Feature
Description and Implementation.
Note:
The attendant console cannot be locked by TOD or manual station lock.
The chapter describes how to administer Communication Manager on the supported Avaya servers
after the product is installed and tested.
The target audience includes system administrators and users with data-networking experience in
data products and technology, and knowledge of the call processing engine of Communication
Manager. In a converged network, voice and data are both sent over a corporate local area network
(LAN). Such a configuration can provide primary or standby telephony, and communication-
processing capabilities.
For more information, see Avaya Aura® Communication Manager Hardware Description and
Reference.
survivable remote server during the configuration process using a web server interface. The
license file determines the mode in which the server operates, and the Configure Server web page
provides supplementary instruction.
If the S8300E or Avaya common server loses contact with its gateway, the gateway retains its last
status until Link Loss Delay Timer (LLDT) expires. The default for LLDT is five minutes, however,
this interval is administrable using the Link Loss Delay Timer (minutes) field on the IP-Options
System Parameters screen. After LLDT expires, the system removes all boards and deletes all the
call-processing information. However, if the gateway loses contact with the S8300E or Avaya
common server, the gateway tries to reconnect for a period of one minute. If reconnection fails, the
gateway then tries to connect with another server in its controller list. If the primary server is a
survivable remote server, it starts looking at the top of its MGC list to reconnect to the primary
server. Otherwise, it starts down the list of alternative servers. When a functional S8300E or
Avaya common server is located, the gateway indicates the server of its current call state, and the
server maintains the connections until the users disconnect.
If the primary call-processing server goes offline and a survivable remote server is available as a
standby unit, call processing happens as follows:
• IP telephones and gateways that were previously using the primary server try to register with
the standby server for call processing, provided that they have been administered to do so in
the controller list by using the set mgc list command.
• The standby server, which is the survivable remote server, goes into license error mode, and
then starts call processing. The standby server cannot preserve any calls set up by the
primary server. IP telephone connections can stay up until the call is completed if they are
shuffled, however, no features are supported on the call.
Note:
The license error mode runs for up to 30 days, and if the problem is unresolved, the
system goes into No License Mode and administration and some commands are
restricted.
• If the standby server is rebooted, all devices return to using the primary server for call-
processing service. Any calls in progress on the standby survivable remote server are
dropped when the reboot occurs as the change back to the primary server is not call
preserving.
The survivable remote server provides full functionality and feature.
For more information about Avaya servers, see Avaya Aura® Communication Manager Hardware
Description and Reference
For more information about SNMP alarms, see Avaya Aura® Communication Manager SNMP
Administration and Reference.
4. On the Server Access page, in the SAT over Telnet (5023) field, do one of the following:
• Select Enable to enable SAT over Telnet (5023).
• Select Disable to disable SAT over Telnet (5023).
5. Click Submit.
Supported browsers
Following are the minimum supported versions of the supported browsers:
• Internet Explorer 11
• Mozilla Firefox 65, 66, and 67
Note:
If your browser does not have a valid security certificate, you see a warning with
instructions to load the security certificate. If you are certain your connection is secure,
accept the server security certificate to access the Logon screen. If you plan to use this
computer and browser to access this or other S8300E or Avaya common servers
again, click the main menu link to Install Avaya Root Certificate after you log in.
The system displays the Logon screen.
4. In the Logon ID field, type your user name.
Note:
If you use an Avaya services login that is protected by the Enhanced Access Security
Gateway (EASG), you must have an EASG tool to generate a response for the
challenge that the Logon page generates.
5. Click Continue.
6. Type your password, and click Logon.
After successful authentication, the system displays the home page of the Communication
Manager SMI.
SNMP setup
You can set up Simple Network Management Protocol (SNMP) services on the server to provide a
means for a corporate NMS to monitor the server, and send alarm notifications to a services
agency, to a corporate NMS, or both. For more information about administering SNMP, see Avaya
Aura® Communication Manager SNMP Administration and Reference guide.
To activate SNMP alarm notification for devices, use the SNMP Traps screen and set up SNMP
destinations in the corporate NMS. SNMP traps for other devices on the network can be
administered using Device Manager. For more information about, see Avaya Aura®
Communication Manager SNMP Administration and Reference.
Note:
UDP port 162 for snmptrap must be opened to provide reception of traps (from gateways) and
transmission of traps to your trap receiver. Certain trap categories from gateways must be
administered “on” by gateway administration. Use gateway commands set snmp trap
enable auth and tcp syn-cookies for this. For more information on gateways, see
Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and
Servers guide and Maintenance Procedures for Avaya Aura® Communication Manager,
Branch Gateways and Servers guide.
and gateways to register with active survivable servers. The split registration prevention feature
keeps branch-oriented operations intact with local trunk and VoIP resources.
cannot reregister to the main server or the survivable server till one of the following conditions is
satisfied:
• At least one gateway reaches the time configured in time-day-window.
• The administrator runs the mg-return command.
• Re-registration to the main server or the survivable server ends in the following situations:
- The survivable server becomes inactive.
- One hour elapses after the administrator runs the enable-mg return command.
- The administrator runs the disable mg-return command.
• The survivable server deregisters from the main server or the survivable server.
deregisters from the main server, the main server does not receive information about the status of
the survivable server. If the main server does not receive information about the status of the
survivable server, the main server activates all the network regions associated with the survivable
server.
Note:
If there are port networks on SCS, it will remain active even if all the gateways reregister to
the main server. You can use the get forced-takeover ipserver-interface
command to force registration to the main server.
Telephones in a network region automatically deregister when all the gateways and port
networks deregister from the survivable server.
• The AGL that IP telephones receive after they reboot must contain the address of the
survivable server at the end of the list. If the IP address of the survivable server is not listed
in AGL and the main server is unreachable, telephones cannot register with the survivable
server.
Note:
If Communication Manager is upgrading to a newer version, the pre-upgrade AGL lists are not
disturbed unless the administrator makes changes to the AGL fields and enters new values.
For more information on the administration procedures for this feature, see Administrable
Alternate Gatekeeper List administration.
When sending the AGL list with the administrable AGL feature, the system uses each network
region (home, direct, indirect) and sends a subset of the C-LANs starting at a random place in the
C-LAN array.
The figure on page 77 shows how unwanted C-LANs can end up in the Alternate Gatekeeper
List.
In this configuration, the IP telephones in NR1 through NR3 have C-LANs in their network regions
as there are no C-LANs that are directly connected to NR200. You can add a few C-LANs in
NR200 to share with NR1-NR3 as they are directly connected. NR 200 consolidates traffic from
NR1-NR3 to obtain access to WAN. Using NR 200 also isolates C-LANS in each network region to
IP telephones in that particular network region.
NR4 and NR5 are Survivable Core Server locations, and the IP telephones in these two locations
need local C-LANs that are in NR4 and NR5.
NR101 and NR102 are Gateway or Survivable Remote Server locations and should share pooled
C-LANS. In this case, C-LANS are placed in NR201 as it is directly connected to the two NRs.
These C-LANs are physically at the main location. Before Communication Manager Release 5.1
C-LANs could be in home region of the IP Phone or in a directly connected NR. The IP telephones
in NR101 and NR102 now receive AGL information that contain C-LANs from NR201.
The IP telephones in NR4 and NR5 receive C-LANs in NR201 in the AGL as that NR is directly
connected. The IP telephones can end up with C-LANS in their AGL that cannot be used in a
WAN failure. This can significantly delay IP telephones in NR4 and NR5 from recovering to a C-
LAN that can be used in a WAN failure. This could also significantly delay IP telephones in NR4
and NR5 in recovering to a Survivable Core Server.
The figure on page 78 shows a workaround supported on Communication Manager 5.1 and
earlier. You can implement this workaround using another virtual network region.
Figure 3: Improved configuration for unwanted C-LANs using the enhanced AGL feature
The figure on page 78 shows the configuration in which the IP telephones in NR4 and NR5 are
administered to only use C-LANS in their native NR, and not use C-LANs in NR201. The IP
telephones AGLs in NR4 and NR5 contain local C-LANs. The IP telephones in NR101 and NR102
share C-LANs in NR201. Those C-LANS are physically located at location 1. A large number of C-
LANs in NR201, might result in large AGLs, and delay recovery to the Survivable Core Server.
With this enhancement, administrators can specify the number of C-LANs in NR201 and control
the size of AGL.
The figure shows a network configuration with numerous gateway or survivable remote server
locations, some of which are directly connected to the WAN, and others that are indirectly
connected to the WAN. All these gateways need to share a pool of C-LANS located at location 1.
The IP telephones in NR151 and NR152 are indirectly connected to NR200. Also, the system
cannot specify the number of C-LANs in NR200 to be used to control size of AGL.
The figure on page 80 shows the workaround that you can use in the pre-Communication
Manager 5.1 implementation.
In this configuration, all the IP telephone network regions are directly connected to a new NR201.
The AGL now contains C-LANs in NR201. But you cannot specify number of C-LANs in NR201
that you can use to control size of AGL. This configuration does not reflect the WAN topology.
The figure on page 80 shows the improved configuration using the Communication Manager 5.1
Administrable AGL feature.
All IP telephones AGL contain C-LANs in NR200, including the direct and indirect network regions.
You can specify the number of C-LANs in NR200 and control the size of the AGL.
Considerations
If the telephone IP address is not in one of the ranges in the IP network map, the AGL entries
consist of C-LANs or PE from the telephone home region only. Note that when administering an IP
address of a telephone in a network map, the associated AGL works robustly by accessing
connected regions and the homed region directly and indirectly.
Interactions
This section provides information about how the Administrable AGL feature for Communication
Manager 5.1 interacts with other features on the system.
• You can have some regions that use the pre-Communication Manager 5.1 nonadministrable
AGL implementation, and some other regions that use the new administrable AGL
implementation. But you cannot have a single network region that use a combination of the
two methods. The AGL column can either contain numbers or alphabets, but not both. The
field can also contain blanks. Blanks are ignored by both the old and the new implementation
of this feature.
• This feature only applies to H.323 IP telephone registrations and H.323 IP telephone AGLs.
The H.323 gateways also register to Communication Manager. This feature does not affect
how the gateways obtain and use their own lists of gatekeepers. This feature does not impact
on how IP (SIP) telephones register to SM 6.0 or SES 5.2 and earlier.
• If an extension number has shared control using the server between an H.323 IP telephone
and an H.323 IP softphone, Communication Manager displays both the AGLs that were sent
to the H.323 telephone and H.323 softphone.
• In prior releases of Communication Manager, the AGL feature only included C-LANs from the
same region and from directly connected regions. The AGL feature included C-LANs from all
indirectly connected regions if there were no C-LANS in the same or directly connected
regions. With this enhancement, it is now possible to explicitly administer Communication
Manager to include C-LANs from indirectly connected regions as well. Also, if you administer
a non-zero value in the AGL column for an indirectly connected region, it opens that indirectly
connected region C-LANs to be eligible to be used for load balancing.
• In general, when using the Communication Manager 5.1 Administrable AGL feature, C-LAN
priorities should not be used. Note the following information:
- For TTS telephones, Communication Manager 5.1 enhanced feature considers priorities,
C-LAN socket load, C-LAN’s service state, and whether the H.323 IP telephone
registration can use C-LANs for load balancing.
- For non-TTS telephones, priorities and C-LAN socket load are taken into account when
load balancing.
- For TTS and non-TTS telephones, the Communication Manager 5.1 enhanced feature
does not take either priorities or C-LAN socket load into consideration when building the
AGL.
Requirements
Procedure
1. Verify that your system is running Communication Manager Release 5.1 or later.
2. Complete basic administration procedures for H.323 telephones.
c. If the value is all for any row, you cannot enter a number into any of the other rows.
In this case, set them to all or blank. Note that if the value for every row is all or
blank, the system automatically uses the Release 5.1 or earlier version of this feature
to determine AGL.
3. Select Enter to save your changes.
Dynamic Host Configuration Protocol (DHCP) server can differ from the one displayed by the
status station command.
• If the gatekeeper sending the RCF to the telephone is not in the AGL, some telephones add
that particular gatekeeper address to their local AGL copy.
warm interrupt rather than a reset. This results in less drastic action being taken to recover links
and H.323 IP telephones.
During the network outage, only stable calls in progress have their bearer connections preserved.
A stable call is a call for which the talk path between the parties in the call is established. Call
control is unavailable during the network outage, and this means that any call in a changing state
is most likely not preserved.
Some examples are:
• Calls with dial tone
• Calls in dialing stage
• Calls in ringing stage
• Calls transitioning to or from announcements
• Calls transitioning to or from music-on-hold
• Calls on hold
• Calls in ACD queues
• Calls in vector processing
Further, you cannot change the state of a preserved call. So, features such as conference or
transfer are unavailable on the preserved calls. Button pushes are not recognized. Invocation of a
feature by the user is denied. In a conference call, if a party in the call drops, the call is dropped.
The following are additional improvements:
• Improve TCP Recovery Time
• Increase IPSI Local Buffering to prevent data loss
• Reduce escalation impact between 15 and 60 seconds by using warm interrupt of PKTINT
instead of PKTINT application reset (hardware interrupt).
• Reduce escalation impact between 60 and 90 seconds by extending PN cold reset action
from 60 seconds to 90 seconds
• Reduce Survivable Core Server No Service Timer minimum value from 3 minutes to 2
minutes to reduce local outage in case of prolonged network outage
• List measurements for the PCD-PKTINT socket for improved troubleshooting
For more information on System parameters screen, see Avaya Aura® Communication Manager
Screen Reference.
endpoints using TTS do not regenerate sockets. H.323 endpoints that do not use TTS do not
reregister or regenerate sockets.
Security considerations
Administration login passwords are passed in plain text with no encryption. The exceptions to this
no-encryption policy include:
• The EASG program installed on all Avaya servers.
• An encrypted web interface to the Avaya server. See the security certificate information in the
server online help.
• Optional encryption for data backups. See data backup and restore.
• Support for RADIUS authentication for gateways.
Example, 001V205
In this example, 001 represents the gateway number, V2 represents the slot number (possibly V1
through V9), and 05 represents the port number.
Enabling transmission over IP networks for TTY and fax calls example
Before you begin
The endpoints sending and receiving calls must be connected to a private network that uses
H.323 trunking or LAN connections between gateways and/or port networks. Calls must be able to
either be passed over the public network using ISDN-PRI trunks or passed over an H.323 private
network to Communication Manager switches that are similarly enabled.
Therefore, you must assign the IP codec you define in this procedure to the network gateways.
For our example, the network region 1 will be assigned codec set 1, which you are enabling to
handle fax and TTY calls.
Procedure
1. Enter change ip-codec-set 1 or change ip-media-parameters 1.
2. Complete the fields as required for each media type you want to enable.
3. Select Enter to save your changes.
For more information abou fax or TTY over IP, see Administering Network Connectivity on
Avaya Aura® Communication Manager.
• - Port Network Support: set to n indicates that traditional port networking is disabled. An
S8300E Server is the primary call controller.
- Processor Ethernet: set to y indicates the presence of an S8300E.
If you change the Enable Voice or Network Stats field from n to y, the system checks the
compatibility of the installed media processor boards and checks if the board is specified on the
Media Processor Measurement Selection screen. If the media processor board is not a valid
TN2302 or TN2602 board, the system displays the Board must be a valid TN2302 or
TN2602 error message.
If you change the Enable Voice or Network Stats field from y to n, the system checks to ensure
that the board is removed from the Media Processor Measurement Selection screen. If the media
processor board is not removed, the system displays the This board(s) will
automatically be removed from the meas-selection media-processor form
warning message. If you press enter again, the media processor board is removed from the Media
Processor Measurement Selection screen.
Note:
Before measuring the voice or network statistics for up to 50 boards, you must administer
media processor boards on the Circuit Packs screen, IP Interface screen and Measurement
Selection screen. To avoid having to go back and forth between the IP Interface screen and
the Media Processor Measurement Selection screen for each media processor board, you
must administer all boards for which you want to collect data on the Media Processor
Measurement Selection screen.
You can generate the report to record the voice statistics for each of the threshold criteria and for
the data calls at both an hourly and summary level. You can view this report at both a network
region and media processor board level. Report reflects data for up to 24 hours period. You can
generate the following reports:
• Hourly Jitter Network Region report – The Hourly Jitter Network Region report assess the
jitter at the network region per hour during calls.
• Hourly Delay Network Region report – The Hourly Delay Network Region report assess the
round trip delay at the network region per hour during calls.
• Hourly Packet Loss Network Region report – The Hourly Packet Loss Network Region report
assess the packet loss at the network region per hour during calls.
• Hourly Data Network Region report – The Hourly Data Network Region report assess the
data calls which exceeded a threshold event at the network region. This report is not applied
to the specific threshold exceeded, but applies only to pass-through and TTY relay calls,
which exceed any one of the three thresholds.
• Hourly Jitter Media Processor report – The Hourly Jitter Media Processor report assess the
jitter at the media processor region per hour during calls.
• Hourly Delay Media Processor report – The Hourly Delay Media Processor report assess the
round trip delay at the media processor region per hour during calls.
• Hourly Packet Loss Media Processor report – The Hourly Packet Loss Media Processor
report assess the packet loss at the media processor region per hour during calls.
• Hourly Data Media Processor report – The Hourly Data Media Processor report assess the
data calls which exceeded a threshold event at the media processor region. This report is not
applied to the specific threshold exceeded, but applies only to pass-through and TTY relay
calls which exceed any one of the three thresholds.
• Summary Jitter report – The summary jitter report summarizes up to five worst jitter calls for
the corresponding peak hour for a given media processor board in the network region.
• Summary Round Trip Delay report – The summary round trip delay report summarizes up to
five worst round trip delay calls for the corresponding peak hour for a given media processor
board in the network region.
• Summary Packet Loss report – The summary packet loss report summarizes up to five worst
packet loss calls for the corresponding peak hour for a given media processor board in the
network region.
• Summary Data report – The summary data report summarizes up to five worst data calls for
the corresponding peak hour for a given media processor board in the network region.
You can also view a near real time voice statistics on the Status Station screen that includes any
threshold exception data gathered during a call in progress.
For more information on the voice or network statistics reports, see Avaya Aura® Communication
Manager Reports.
SNMP administration
For more information, Avaya Aura® Communication Manager SNMP Administration and
Reference.
Network preemption
Communication Manager supports network preemption. For network preemption to work,
Communication Manager must be configured to use Session Manager as the bandwidth manager.
To configure Session Manager as the bandwidth manager, see Avaya Aura® Communication
Manager Feature Description and Implementation. The security administrator can assign
bandwidth budgets for audio and video, to each network link on Session Manager. When server or
network resources are running too low to allow additional calls, call preemption occurs. For more
information, see Administering Avaya Aura® Session Manager.
Processor Ethernet (PE) provides connectivity to IP endpoints, gateways, and adjuncts. The PE
interface is a connection in the Communication Manager software that uses a port on NIC in the
server. No additional hardware is required to implement PE, but the feature must be enabled using a
license file.
During the configuration of a server, PE is assigned to Computer Ethernet (CE). Both PE and CE
share the same IP address. However the CE interface is a native computer interface while the PE
interface is the logical appearance of the CE interface within the Communication Manager software.
The PE interface can either be a control network or a corporate LAN. The selected interface
determines the physical port that PE uses on the server.
Note:
The PE interface is enabled automatically on a survivable remote or a survivable core server. If
the PE interface is disabled, the survivable remote or survivable core server cannot register with
the main server and becomes nonfunctional.
In Communication Manager Release 5.2, PE is supported on duplicated servers for the connection
of H.323 devices, branch gateways, SIP trunks, and most adjuncts.
The capabilities of survivable core servers are enhanced to support the connection of IP devices to
the PE interface and to C-LAN interfaces located in G650 Media Gateway.
Note:
When you use PE on duplicated servers, you can use the following IP telephone models to
ensure an optimal system performance:
• 9610, 9620, 9630, 9640, and 9650 telephones with firmware 3.0 or later, or any future 96xx
and 96x1 models that support Time to Service (TTS) to work optimally.
• 4601+, 4602SW+, 4610SW, 4620SW, 4621SW, 4622SW, and 4625SW Broadcom telephones
with firmware R 2.9 SP1 or later, provided the 46xx telephones are not in the same subnet as
the servers.
All other IP telephone models reregister in case of server interchange. If not in the same subnet, the
46xx telephones reregister as the servers.
When PE is used on duplicated servers, PE must be assigned to an active server IP address. The
active server IP address is shared between the servers. In networking technology, this address is
called IP-alias. The active server is the only server that responds to the IP-alias.
A survivable remote or a single survivable core server can use the PE interface to connect to CDR,
AESVCS, and CMS. Duplicated survivable core servers can use the PE interface to connect to
CDR, Messaging, and SIP Enablement Server (SES).
For more information about survivable core servers, see Avaya Aura® Communication Manager
Survivable Options.
6. For adjunct connectivity to a survivable core server or survivable remote server, use the
Survivable Processor - Processor Channels screen to:
• Use the same processor channels information as the main server by entering
i(nherit) in the Enable field.
• Use different translations than that of the main server by entering o(verwrite) in the
Enable field. After entering o(verwrite), you can enter information specific to the
survivable core server or survivable remote server in the remaining fields.
• Disable the processor channel on the survivable core server or survivable remote server
by entering n(o) in the Enable field.
7. Execute a save translations all, save translations ess, or save
translations lsp command to send (file sync) the translations from the main server to
the survivable core server or survivable remote server.
Related links
Configuring the PE interface on the server using the server SMI on page 97
survivable remote server must be able to ping to the address. Select which interface you
want the file sync to use. Use the customer LAN for file sync.
To configure the ports on your server, click Firewall under the Security heading in the Server
Administration Interface.
Network Configuration
Use the Network Configuration page to configure the IP-related settings for the server.
Note:
Some of the changes made on the Network Configuration page can affect the settings on
other pages under Server Configuration. Make sure that all the pages under Server
Configuration have the proper and related configuration information.
Use the Network Configuration page to configure or view the settings for the hostname, alias Host
Name, DNS domain name, DNS search list, DNS IP addresses, server ID, and default gateway.
• If the configuration setting for a field is blank, you can configure that setting from the Network
Configuration page.
• If the configuration setting for a field is already obtained from an external source, such as
Avaya Aura® System Manager Solution Deployment Manager, that field is view-only.
• If you want to change the configuration setting obtained from an external source, you must
navigate to the external source, such as Avaya Aura® System Manager Solution Deployment
Manager used to configure the settings.
You can also configure the IP-related settings for each Ethernet port to determine how each
Ethernet port is to be used (functional assignment). Typically, you can configure an Ethernet port
without a functional assignment. However, any Ethernet port intended for use with Communication
Manager must be assigned the correct functional assignment. Make sure that the Ethernet port
settings in the Network Configuration page match with the physical connections to the Ethernet
ports. Ethernet ports can be used for multiple purposes, except for the service's laptop port.
However, currently there is no laptop service port within Communication Manager.
The number of entries for the Ethernet ports in the Network Configuration page corresponds with
the number of Network Interface Cards (NICs) the server has.
To activate the new settings in the server, you must restart Communication Manager. Make sure
that you restart Communication Manager only after configuring the complete settings of the server.
Too many restarts can escalate to a full Communication Manager reboot.
Important:
The IPv6 Address field is limited to a specific customer set and not for general use.
Duplication parameters
Communication Manager supports two server duplication types:
• software-based duplication
• encrypted software-based duplication
Note:
The server duplication type must be the same for both the active and standby servers.
Duplication parameters of the other server include the host name, server ID, corporate LAN IP
address, and duplication link IP address for the other server.
PE parameters include configuring the printer change priority level for the server and the IP
address. The IP address facilitates the server to determine whether the PE interface is working or
not.
To activate the new settings in the server, you must restart Communication Manager. Restart
Communication Manager only after configuring the complete settings of the server. Several
restarts can escalate to a complete Communication Manager reboot.
See the Duplication parameters page to configure the following settings for the server.
Adjuncts with PE
For the single main server, adjuncts that use the C-LAN can use the PE interface of the main
server for connectivity to the main server. For the Survivable Remote and Survivable Core
Servers, there are three adjuncts, the CMS, AESVCS, and the CDR, that are supported using the
Survivable Remote or Survivable Core Server’s PE interface. This section provides a high-level
overview of the adjuncts supported by the Survivable Core and Survivable Remote Servers and
how they are administered to use the PE interface.
• Survivable CMS
Starting with CMS Release 13.1, you can use a Survivable CMS co-located at the site of the
Survivable Core or Survivable Remote Server. A Survivable CMS is a standby CMS that
collects data from a Survivable Remote or Survivable Core Server when the main server is
not operational or when the customer is experiencing a network disruption. A Survivable
CMS should not be located at the same location as the main server.
During normal operations, the Survivable CMS has a connection to the Survivable Core or
Survivable Remote Server but does not collect data or support report users. Only the main
CMS server collects data. When a Survivable Core Server assumes control of one or more
port networks, or a Survivable Remote Server is active, the Survivable Core Server and/or
the Survivable Remote Server sends data to the Survivable CMS.
• CDR
The server initiates the connection to the CDR unit and sends call detail information over the
configured link. The link remains active at all times while the CDR unit waits for data to be
sent by a connected server. In the case of a Survivable Core or Survivable Remote Server,
data will not be sent until the survivable server becomes active. Some CDR units can collect
data from multiple servers in a configuration, separately or all at once. For information on the
capability of your CDR unit, check with your CDR vendor.
The CDR unit is administered on the IP Services screen. To use the PE interface, procr
must be entered in the Local Node field.
• AESVCS
AESVCS (Application Enablement Services) supports connectivity with a maximum of 16
servers. Since AESVCS cannot tell which server is active in a configuration, it must maintain
a constant connection with any server from which it might receive data. An Avaya Server
“listens” for AESVCS after it boots up. The AESVCS application establishes the connection to
the server.
If the adjunct terminates solely on the main server’s PE interface, you do not have to
administer the Survivable Processor screen. If AESVCS connects to a Survivable Remote or
Survivable Core Server, you must administer the Survivable Processor screen in addition to
the IP Services screen.
be entered in this field to receive an error or a warning alarm if the targeted value is
exceeded.
Related links
Connecting new telephones on page 109
Note:
The telephone type must match the board type. For example, match a two-wire digital
telephone with a port on a two-wire digital circuit pack. Use this procedure with all
circuit-switched telephones except BRI (ISDN) and model 7103A.
Caution:
You can destroy your hardware if you attempt to connect an analog telephone to a
digital port.
Note:
Fixing problems: If you misdial and the wrong extension is activated for the telephone
you are using, use the terminal translation initialization (TTI) unmerge feature access
code to “uninstall” the telephone before you try again.
Procedure
1. Find an available port.
2. Connect the port to the cross-connect field or termination closet.
3. Type the telephone details in the system.
Related links
Manage telephones on page 108
8. Press Enter.
9. To print to the system printer that you use for scheduled reports, type list
configuration station schedule immediate.
10. Press Enter.
11. Choose an extension number for the new telephone.
The extension you choose must not be assigned and must conform to your dial plan. You
should also determine whether this user needs an extension that can be directly dialed
(DID) or reached via a central telephone number. Be sure to note your port and extension
selections on your system’s paper records.
Telephone installation
After reading the relevant information about how to connect a telephone, you are ready to connect
the port to the cross-connect field. You can configure the system to set up the new telephone.
To request Avaya to install the new connections, go to the Avaya Support website at http://
support.avaya.com. Notify the Avaya technical support representative or on-site technician that
you are ready to add the telephone to the system.
If you are making the connections yourself, see the system installation guide for any questions.
5. Press Enter. When the system displays the Station screen, you see the extension number
and some default field values.
The name you enter displays on called telephones that have display capabilities. Some
messaging applications, such as INTUITY, recommend that you enter the last name of the
user and their extension to identify the telephone. The name entered is also used for the
integrated directory.
Tip:
To hide a name in the integrated directory, enter two tildes before the name when you
assign it to the telephone. Set Display Character Set on the System Parameters
Country-Options screen to Roman. The tildes are not displayed with caller ID name.
Also, if a name is entered with only one tilde, the name is converted to Eurofont
characters.
Note:
For 4610SW, 4620SW, 4621SW, and 4622SW, Sage, Spark, and 9600-series Spice
telephones, the Name field is supported by Unicode language display. You must use
MSA. For more information about Unicode language display, see Administering
Unicode display. Unicode is also an option for the 2420J telephone when Display
Character Set on the System Parameters Country-Options screen is Katakana. For
more information about the 2420J, see 2420 Digital Telephone User's Guide,
555-250-701.
The name you enter displays on called telephones that have display capabilities. Some
messaging applications, such as INTUITY, recommend that you enter the last name of the
user and their extension to identify the telephone. The name entered is also used for the
integrated directory.
Tip:
To hide a name in the integrated directory, enter two tildes before the name when you
assign it to the telephone. Set Display Character Set on the System Parameters
Country-Options screen to Roman. The tildes are not displayed with caller ID name.
Also, if a name is entered with only one tilde, the name is converted to Eurofont
characters rewrite in active voice
Note:
For 4610SW, 4620SW, 4621SW, and 4622SW, Sage, Spark, and 9600-series Spice
telephones, the Name field is supported by Unicode language display. You must use
MSA. For more information about Unicode language display, see Administering
Unicode display. Unicode is also an option for the 2420J telephone when Display
Character Set on the System Parameters Country-Options screen is Katakana. For
more information about the 2420J, see 2420 Digital Telephone User's Guide,
555-250-701.
6. Press Enter to save your changes.
Changing a station
About this task
You can make changes to a new telephone, such as assigning a coverage path or adding feature
buttons.
Procedure
1. Enter change station nnnn where nnnn is the extension of the new telephone.
2. Change the necessary fields as described in the previous section, and then press Enter.
Duplicating telephones
About this task
A quick way to configure telephones is by copying the information from an existing telephone and
modifying it for each new telephone. For example, you can configure one telephone as a template
for an entire work group. Then you can duplicate the template Station screen to add all the other
extensions in the group.
Note:
Only telephones of the same model can be duplicated. The duplicate command copies all
the feature settings from the template telephone to the new telephones.
Procedure
1. Type display station nnnn, where nnnn is the extension of the Station screen you
want to duplicate to use as a template.
2. Click Enter.
For more information, see Avaya Call Center Release 4.0 Automatic Call Distribution
(ACD) Guide, 07-600779.
Using an alias
About this task
Not every telephone model or device has a unique Station screen in the system. You might have
to use an available model as an “alias” for another. If you need to enter a telephone type that the
system does not recognize or support, use an alias. Defining aliases is also a useful method to
identify items that act as analog stations on Communication Manager, such as fax machines,
modems, or other analog device.
If you purchase a telephone model that is newer than your system, you can alias this telephone to
an available model type that best matches the features of your new telephone. See your
telephone manual to determine which alias to use. If your manual does not have this information,
you can contact the DEFINITY® helpline for an appropriate alias.
For example, you can create two aliases: one to add a new 6220 telephone and one to add
modems to your system.
Procedure
1. See your new telephone manual to find the correct alias.
To add a new 6220 telephone, you can find that the 6220 should be administered on an
older system as a 2500 telephone. To do this:
2. Type change alias station on CLI.
3. Press Enter.
The system displays the Alias Station screen.
4. In the Alias Set Type field, type 6220.
This is the name or model of the unsupported telephone.
5. In the Supported Set Type field, type 2500.
This is the name or model of the supported telephone.
6. In the Alias Set Type field, type modem.
You can call the alias set anything you like. Once you define the alias, you can use the
alias set in the Type field on the Station screen.
7. In the Supported Set Type field, type 2500.
Entering 2500 indicates to the system that these models are basic analog devices.
8. Press Enter to save your changes.
Now you can follow the instructions for adding a new telephone (or adding a fax or
modem). Avaya Communication Manager now recognizes the new type (6220 or modem)
that you enter in the Type field.
Be sure to see your telephone manual for instructions on how to set feature buttons and
call appearance buttons.
Note:
If you need to use an alias for a telephone, you might be unable to take advantage of
all the features of the new telephone.
Upgrading telephones
About this task
You can change the telephone type for a user without changing the location. You can access the
Station screen for the extension and enter the new model number.
Note:
This method is used only if the new telephone type matches the existing port type, such as a
digital telephone with a digital port.
For example, a user at extension 4556 who has a 7410+ telephone and wants to replace the
phone with a new 8411D telephone.
Procedure
1. On the command line interface, type change station 4556.
2. Press Enter.
The system displays the Station screen for 4556.
3. In the Type field, overwrite 7410+ with 8411D.
4. Press Enter.
Now you can access the functions and feature buttons that correspond to an 8411D
telephone.
Swapping telephones
About this task
Moving a telephone from one location to another or swapping telephones between two locations is
only possible if the two telephones are of the same type. Swapping telephones between two
locations is possible if the telephones are both digital or both analog. You can use X ports to easily
swap the telephones, A and B. Change port assignment of telephone A to X, change telephone
port assignment of B to old port of A, and finally, replace the X on telephone A to old port of B.
For example, to swap telephones for extension 4567 (port 01C0505) and extension 4575 (port
01C0516), complete the following steps:
Procedure
1. Type change station 4567.
2. Press Enter.
3. Record the current port address (01C0505), and type x in the Port field.
4. Press Enter to save your changes.
5. Type change station 4575.
6. Press Enter.
7. Record the current port address (01C0516).
8. Type 01C0505 in the Port field.
9. Update the Room and Jack fields.
10. Press Enter to save your changes
11. Type change station 4567 again.
12. Press Enter.
13. Type 01C0516 in the Port field.
This is the port that was assigned to extension 4575.
14. Update the Room and Jack fields.
15. Press Enter to save your changes.
16. Physically unplug the telephones, and move them to their new locations.
When you swap telephones, the system keeps the old button assignments. If you are
swapping to a telephone with softkeys, the telephone could have duplicate button
assignments because soft keys have default assignments. You can check your button
assignments and modify them as necessary.
the Public Safety Access Point (PSAP) could send the emergency response personnel to the
wrong location.
On the Feature-Related System Parameters screen, set the Terminal Translation Initialization
(TTI) Enabled field to y and the TTI State field to voice.
When you enter always or once in the Automatic Moves field on the Station screen,
Communication Manager obtains the serial number from the telephone and records it to ACTR
Move List. If you change the entry in the Automatic Moves field from always or once to no,
Communication Manager removes the extension from the Move List.
2. Dial the TTI security code from the telephone you want to merge.
• If the code is correct, you receive the dial tone.
• If the code is incorrect, you receive the intercept tone.
3. Dial the extension of the telephone you want to merge.
• If the extension is valid, you receive confirmation tone, which might be followed by dial
tone.
• If the extension is valid and you receive the intercept tone immediately following the
confirmation tone, attempt the merge again.
• If the extension is valid but the extension is being administered, you receive the reorder
tone. Try the merge again later.
• If the extension is invalid, you receive the intercept tone.
• If the system is busy and cannot complete the merge, you receive the reorder tone. Try
the merge again later.
• If the telephone has a download status of pending, you receive the reorder tone.
Change the download status to complete, and try the merge again.
Troubleshooting TTI
If you have difficulty in using TTI, review the following system restrictions.
Problem Restriction
The TTI Ports field This field shows only the number of TTI ports being administered. If a TTI exceeds
on the System the maximum number of ports, the port is not administered and cannot be added. In
Capacity screen that case, a telephone cannot be added. For details on the System Capacity screen,
(type display see Maintenance Commands for Avaya Aura® Communication Manager, Branch
capacity) shows Gateways and Servers.
the number of TTI
BRI endpoints are only counted as one TTI port. For example, for every two BRI
ports used in a
endpoints, one TTI port is counted. As such, you can have two telephones assigned
server running
to one port. If either endpoint is administered, the TTI port count is reduced by 1.
Communication
Manager.
The total number of The total number of translated data terminals and Data TTI ports in a system is
translated limited to the maximum number of administered data modules allowed in the system.
telephones and
Voice TTI ports in a
system is limited to
the maximum
number of
administered
telephones
supported in the
system.
When you use this This can happen when the number of telephones allowed by the system is twice the
order, voice and number of data terminals. For example, if the system limit for telephones is 15,000
then data (Set the and 7,500 for data, then when TTI was turned on for data first, only the first 7,500
TTI State field to unadministered ports would get TTI port translations.
voice and then set
the TTI State field
to data), you
reduce the chance
of a user trying to
use TTI on a data-
only terminal that
does not have TTI
port translation
When TTI is • If the TTI State field was previously activated but in a different state (such as, a
activated for the voice to data state), the old TTI translations are removed and the new ones added
system, these on a board-by-board basis.
actions take place
• If the TTI State field is set to voice, then default TTI translations are generated for
every unadministered port on all digital, hybrid, and analog boards.
• If the TTI State field is set to data, then default TTI translations are generated for
every unadministered port on all digital and data line boards in the system.
• Whenever a new digital board is inserted when the system is in TTI Data mode, or
when a digital, hybrid, or analog board is inserted when the system is in TTI Voice
mode, the unadministered ports on the board become TTI ports.
• When TTI is deactivated, all translation for the TTI ports is removed in the system,
and the ports return to an unadministered state.
Removing telephones
Before you begin
Before you physically remove a telephone from your system, check the telephone’s status,
remove it from any group or usage lists, and then delete it from the system’s memory. For
example, to remove a telephone at extension 1234:
Procedure
1. Type status station 1234.
2. Press Enter.
The system displays the General Status screen.
3. Make sure that the telephone is in the following state:
a. Plugged into the jack
b. Idle and not receiving calls
c. No messages waiting
d. No active buttons, such as Send All Calls or Call Forwarding
4. Type list groups-of-extension 1234.
5. Press Enter.
The Extension Group Membership screen shows whether the extension is a member of
any groups on the system.
6. Press Cancel.
7. If the extension belongs to a group, access the group screen and delete the extension from
that group.
If extension 1234 belongs to pickup group 2, type change pickup group 2 and delete
the extension from the list.
8. Type list usage extension 1234.
9. Press Enter.
The Usage screen shows where the extension is used in the system.
10. Press Cancel.
11. If the system displays the extension on the Usage screen, access the appropriate feature
screen and delete the extension.
If extension 1234 is bridged onto extension 1235, type change station 1235 and
remove the appearances of 1234.
12. Type change station 1234.
13. Press Enter.
Note:
You do not need to delete the extension from coverage paths. The system
automatically adjusts coverage paths to eliminate the extension.
Next steps
Now you can unplug the set from the jack and store it for future use. You do not need to
disconnect the wiring at the cross-connect field. The extension and port address remain available
for assignment at a later date.
Once you successfully remove a set, that set is permanently erased from system memory. If you
want to reactivate the set, you have to add it again as though it were a new telephone.
As an example, if you have already defined an alias for “fax” as a 2500 station type and want to
add a fax machine to extension 4444, the steps are as follows:
Procedure
1. Type add station 4444.
2. Press Enter.
3. In the Type field, type fax.
4. In the Port field, type the port address.
5. In the Name field, type a name to associate with this fax.
6. Move to the Data Restriction field and type y.
Entering y in this field prevents calls to and from this extension from being interrupted by
tone signals. This is important for fax machines and modems as these signals can disrupt
transmissions of data.
7. In the Distinctive Audible Alert field, type n.
This eliminates the distinct 2-burst ring for external calls, which often interferes with the
auto-answer function on fax machines or modems.
8. Press Enter to save changes.
For more information on modem or fax or TTY over IP, see Administering Network
Connectivity on Avaya Aura® Communication Manager, 555-233-504.
IP Softphones
Using Avaya IP Softphones, the end user can control telephone calls directly from a Personal
Computer (PC). An end user can log in remotely to the company server running Communication
Manager, and then make and receive telephone calls from the telephone extension.
Avaya IP Softphones support the following modes:
• Road-Warrior
You typically use this mode for laptop users who are travelling. In this mode, the Personal
Computer LAN connection carries both the call control signaling and the voice path. Because
the audio portion of the voice call is handled by the Personal Computer, you must have some
kind of audio device (e.g., handset, headset) Personal Computer to provide the audio
connection.
• Telecommuter or Avaya IP Agent
For the telecommuter or Avaya IP Agent mode, you make two separate connections to the
Avaya DEFINITY® server. The signaling path is carried over an IP network, and the voice
path is carried over the standard circuit-switched telephone network (PSTN). Since you are
using a telephone for audio, you do not need an H.323 Personal Computer audio application.
The telecommuter mode uses the Avaya IP Softphone interface on the user Personal
Computer and a standard telephone. The Avaya IP Agent mode uses the Avaya IP Agent
interface on the agent Personal Computer and a call center telephone.
• Native H.323 (only available with Avaya IP Softphone R2)
Using the standalone H.323 mode, the travelers can use some Communication Manager
features from a remote location. This mode uses a Personal Computer running an H.323 v2-
compliant audio application, such as Microsoft NetMeeting. The H.323 mode controls the call
signaling and the voice path. However, since it does not use the IP Softphone interface, this
configuration is capable of operating only as an analog or single-line telephone making one
call at a time without any additional assigned features. You can provide standalone H.323
users only features that they can activate with dial access codes.
• Control of IP Telephone (only available with IP Softphone R4 and later)
You can use this mode to make and receive calls under the control of the IP Softphone - just
like in the Telecommuter or Road Warrior mode. The big difference is that you have a real
digital telephone under your control. In the Road Warrior mode, there is no telephone. In the
Telecommuter mode, the telephone you are using whether analog, digital, or IP telephone is
brain dead. In this mode if you have an IP telephone, you get the best of both worlds.
• Control of DCP Telephone (only available with IP Softphone R5 and later)
This feature provides a registration endpoint configuration. With this new configuration, an IP
softphone and a non-softphone telephone can be in service on the same extension at the
same time. Also, the call control is executed by both the softphone and the telephone
endpoint, and the audio is monitored by the telephone endpoint.
Tip:
Use status station to show the part (product) ID, serial number, and the audio connection
method used by existing stations.
Note:
Beginning with the November 2003 release of Communication Manager, R1 and R2 IP
Softphone and IP Agent, which use a dual connect (two extensions) architecture, are no
longer supported. R3 and R4 IP Softphone and IP Agent, which use a single connect (one
extension) architecture, continue to be supported. This applies to the RoadWarrior and the
Telecommuter configurations for the IP Softphone. Native H.323 registrations for R1 and R2
Softphones continue to be supported.
2. Press Enter.
The system displays the Station screen.
3. In the Type field, enter H.323.
4. Press Enter to save your work.
Note:
Port 1720 is turned off by default to minimize denial of service situations. This applies
to all IP softphones release 5.2 or later. You can change this setting, if you have root
privileges on the system, by typing the command: /opt/ecs/ sbin ACL 1720 on
or off.
5. In the Security Code field, enter the password for this remote user.
For example, enter 1234321.
This password can be 3-8 digits in length.
6. In the Media Complex Ext field, type 3000.
This is the H.323 extension just administered.
7. In the IP Softphone field, type y.
8. On page 2, in the Service Link Mode field, type as-needed.
Set this field to permanent only for extremely busy remote telephone users, such as call
center agents.
9. In the Multimedia Mode field, type enhanced.
10. Press Enter to save your work.
Now you can install and configure the software on the user’s Personal Computer. In this
example, the user logs in by entering their DCP extension (3001) and password
(1234321).
Troubleshooting IP softphones
Problem
Display characters on the telephone cannot be recognized.
Cause
Microsoft Windows is not set to use Eurofont characters.
Solution
Set the Microsoft Windows operating system to use Eurofont.
For more information on how to install and configure the IP Softphone software, see user
documentation on Avaya IP softphone.
Problem
The call is not connecting to the home phone that is configured as Other Phone.
Cause
The Facility Restriction Level (FRL) of the desk phone is too low for the route pattern over which
the trunk call to the home phone is sent.
Solution
Increase the FRL of the desk phone or reduce the FRL on the route pattern.
IP Telephones
The 4600-series IP Telephones are physical sets that connect to Communication Manager via
TCP/IP.
Caution:
An Avaya IP endpoint can dial emergency calls (for example, 911 calls in the U.S.). It only
reaches the local emergency service in the Public Safety Answering Point area where the
telephone system has local trunks. Please be advised that an Avaya IP endpoint cannot dial
to and connect with local emergency service when dialing from remote locations that do not
have local trunks. You should not use an Avaya IP endpoint to dial emergency numbers for
emergency services when dialing from remote locations. Avaya Inc. is not responsible or liable
for any damages resulting from misplaced emergency calls made from an Avaya endpoint.
Your use of this product indicates that you have read this advisory and agree to use an
alternative telephone to dial all emergency calls from remote locations.
Adding an IP telephone
Before you begin
Verify the system has a:
• TN2302 IP Media Processor circuit pack for audio capability
• TN799 Control-LAN circuit pack for signaling capability (for CSI Servers only)
Make sure that you can use IP Telephones on your system. Display the System-Parameters
Customer-Options (Optional Features) screen, and verify the following field settings.
• Maximum Concurrently Registered IP Stations is greater than 0
• IP Stations field is y
• Information has been entered in the fields on the Maximum IP Registrations by Product ID
page.
About this task
These steps show how to add an IP telephone at extension 4005 and how to assign an extension.
Procedure
1. Type add station 4005.
2. Press Enter.
The system displays the Station screen.
Note:
When adding a new 4601 or 4602 IP telephone, you must use the 4601+ or 4602+
station type. This station type enables the Automatic Callback feature. When making a
change to an existing 4601 or 4602, you receive a warning message, stating that you
should upgrade to the 4601+ or 4602+ station type to access the Automatic Callback
feature.
The system displays the Port field as display only, and IP.
3. In the Security Code field, enter the password for the IP telephone user.
Although the system accepts a null password, the IP telephone does not work unless you
assign a password.
4. Press Enter to save your work.
Procedure
1. Type change station nnnn where nnnn is the extension number of the original dual-
connect telephone that you are replacing with a single-connect telephone.
The system displays the Station screen.
2. Move to the Media Complex Extension field.
3. Write down the number in the Media Complex field, then delete the number from the field.
4. Press Enter to save your work.
5. Remove the extension you recorded. Before you remove an H.323 extension from your
system, check the status, remove it from any group or usage lists, and then delete it from
the system’s memory.
For example, if you wrote down extension 1234 before you removed it from the Media
Complex field on the Station screen, then remove extension 1234 using these steps:
a. Type status station 1234.
b. Press Enter.
The system displays the General Status screen.
c. Make sure that the extension is idle and not making or receiving calls, has no
messages waiting and has no active buttons, such as Send All Calls or Call
Forwarding.
d. Type list groups-of-extension 1234.
e. Press Enter.
The Extension Group Membership screen shows whether the extension is a member
of any groups on the system.
f. Press Cancel.
g. If the extension belongs to a group, access the group screen and delete the extension
from that group.
If extension 1234 belongs to pickup group 2, type change pickup group 2 and
delete the extension from the list.
h. Type list usage extension 1234.
i. Press Enter.
The Usage screen shows where the extension is used in the system.
j. Press Cancel.
k. If the system displays the extension on the Usage screen, access the appropriate
feature screen and delete the extension.
If extension 1234 belongs to hunt group 2, type change hunt group 2 and delete
the extension from the list.
Note:
You do not need to delete the extension from coverage paths. The system
automatically adjusts coverage paths to eliminate the extension.
Once you successfully remove the extension, it is permanently erased from system
memory. If you want to reactivate the extension, you have to add it again as though it
were new.
Caution:
An Avaya IP endpoint can dial emergency calls, such as 911 in the U.S., but it only
reaches the local emergency service in the Public Safety Answering Point area where
the telephone system has local trunks. Please be advised that an Avaya IP endpoint
cannot dial to and connect with local emergency service when dialing from remote
locations that do not have local trunks. You should not use an Avaya IP endpoint to
dial emergency numbers for emergency services when dialing from remote locations.
Avaya Inc. is not responsible or liable for any damages resulting from misplaced
emergency calls made from an Avaya endpoint. Use of this product indicates that you
have read this advisory and agree to use an alternative telephone to dial all emergency
calls from remote locations. Go to the Avaya Support website at http://
support.avaya.com if you have questions about emergency calls from IP telephones.
Also, install and administer your Remote Office hardware at the remote location and obtain the
following information from the remote administration:
• IP address
• Password
About this task
As an example, set up a remote-office location using Avaya R300 Remote Office Communicator
hardware, add a new node, and set up the signaling group and trunk group.
Procedure
1. Type change node-names IP.
2. Press Enter.
The system displays the Node Name screen.
3. In the Name field, type in a word to identify the node.
Type Remote 6.
4. In the IP address field, type in the IP address to match the one on the Avaya R300
administration.
5. Press Enter to save your changes.
6. Type add remote office, and the number for this remote office.
7. Press Enter.
The system displays the Remote Office screen.
8. Fill in the following fields.
• Node Name - match the name on the IP Node Names screen.
• Network Region - this must match the network region on the IP Interfaces screen for
the circuit packs that connect this remote office. Use display ip-interfaces to find this
information.
• Location - match the one set up on the Location screen for this remote office.
• Site Data - identify the street address or identifier you want to use.
9. Press Enter to save your changes.
Tip:
Use status remote office to verify that your server running Communication Manager
recognizes the Remote Office information. It also displays the extensions and signaling
group you administer next.
6. In the Far-end Node Name field, identify the node name assigned to the CLAN that
supports the R300.
7. In the Near-end Listen Port field, type a port number in the 5000-9999 range.
8. In the Far-end Listen Port field, type 1720.
9. In the RRQ field, type y.
10. Tab to the Direct IP-IP Audio Connection field on another page of this screen, and type
y.
11. Press Enter to save your changes.
2. Press Enter.
The system displays the Firmware Station Download screen.
3. In the Schedule Download field, type y.
The Start Date/Time and Stop Date/Time fields appear.
4. In the Start Date/Time field, enter the month (mm), day (dd), year (yyyy), and time
(hh:mm) that you want the download to start.
5. In the Stop Date/Time field, enter the month (mm), day (dd), year (yyyy), and time
(hh:mm) that you want the download to stop.
6. In the Continue Daily Until Completed field, enter y if you want the system to execute the
firmware download each day at the scheduled time until all specified telephones have
received the firmware.
7. In the Beginning Station field, enter the first extension number in the range of telephones
to which you want to download the firmware.
Up to 1000 stations can be included in a scheduled download.
8. In the Ending Station field, enter the last extension number in the range of telephones to
which you want to download firmware.
Up to 1000 stations can be included in a scheduled download.
Note:
Although you can specify a range of up to 1000 extensions, all 1000 stations are not
downloaded simultaneously because there is a limit of how many concurrent
telephones will be downloaded on a board, gateway, and port network. These limits will
likely result in multiple "passes" required to attempt a download to the telephone. Also
note that on the first "pass", only two telephones will be attempted, and if multiple
telephones fail, then the schedule may stop.
9. Press Enter.
The firmware download is set to run at the scheduled time. If you entered n in the
Schedule Download? field, pressing Enter immediately initiates the download to the
specified range of telephones.
Note:
If you add the qualifier last to the status firmware download command, status
information on the last download schedule is displayed.
For more information about the features of Avaya J100 Series IP Phones, see Avaya Aura®
Communication Manager Hardware Description and Reference and Avaya J100 Series IP Phones
Overview and Specifications.
Note:
If the station extension is a SIP telephone type and if the application type is OPS on
the Stations with Off-PBX Telephone Integration screen, then the Off-PBX screen's
Location field is display-only and displays the value of the Location field of the
corresponding Station screen.
Once you add a telephone to the system, you can use the Station screen to change the settings,
such as adding or changing feature button assignments. You can assign features or functionality to
each programmable button according to your choice. If you have 6400-series telephones, you can
administer some of their own feature buttons. For more information, see Setting up Terminal Self-
Administration for more information.
Note:
An NI-BRI telephone with Communication Manager has only the Conference, Transfer, Hold,
and Drop feature buttons, none of which requires administration. On an NI-BRI telephone, you
can assign additional feature buttons only as call appearances. As a result, NI-BRI telephone
users must access all other features of Communication Manager using feature access codes.
Additionally, the number of call appearance buttons administered in Communication Manager
(the default is three) must match the number of call appearances programmed on the
telephone. Finally, Communication Manager does not support bridged call appearances for NI-
BRI telephones.
Note:
Vector Directory Numbers do not appear in the
integrated directory. Also, if you assign a name
beginning with two tildes (~~} to a telephone,
and Display Character Set on the System
Parameters Country-Options screen is set to
Roman, the name does not appear in the
integrated directory. Note that this is the only
way to hide a name in the integrated directory.
dir-pkup Dir Pickup Directed call pickup: Users uses this to answer a
call ringing at another extension without having to
be a member of a pickup group.
disp-chrg Disp Charges Provides your display telephone with a visual 1 per station
display of accumulated charges on your current
telephone call. Used exclusively outside the U.S.
and Canada.
disp-norm Local/ Normal Normal (display button): Toggles between LOCAL 1 per station
display mode (displays time and date) and
NORMAL mode (displays call-related data). LED off
= LOCAL mode and LED on = NORMAL.
dn-dst DoNotDisturb Places the user in the do not disturb mode. 1 per station
drop Drop User can drop calls. Users can drop calls from
automatic hold or drop the last party they added to a
conference call.
Table continues…
Note:
Go to Cover cannot be activated for calls
placed to a Vector Directory Number extension.
Go to Cover can be used to force a call to
cover to a VDN if the called principal has a
VDN as a coverage point.
grp-dn-dst GrpDoNotDstrb Group Do Not Disturb (display button): Places a 1 per station
group of telephones into the do not disturb mode.
grp-page GrpPg Using this users can make announcements to
(Number:___) groups of stations by automatically turning on their
speakerphones. Number: The extension of the page
group.
headset Headset Signals onhook or offhook state changes to 1 per station
Communication Manager. The green LED is on for
offhook state and off (dark) for onhook state.
hunt-ns (Grp: HuntNS Hunt-Group Night Service: Places a hunt-group into 3 per hunt
___) night service. Grp: Hunt group number. group
hntpos-bsy Busy (Hunt Grp Using this button, non-ACD hunt group users can 1 per station
(Grp: ___ ) #) opt-in or opt-out of hunt group calls. per group
in-call-id (Type: INCallID (group A member of a coverage answer group or hunt 1 per group-
__ Grp: ___) #, type, name, group can use the Coverage Incoming Call type per group
or ext #) Identification (ICI) button to identify an incoming call
to that group even though the member does not
have a display telephone. In the Type field, enter c
for coverage answer groups and type of h for a hunt
group. In the Grp field, enter the group number.
inspect Inspect Inspect (display button): Users use this on an active 1 per station
call to display the identification of an incoming call.
Users can also use this to determine the
identification of calls they placed on Hold.
Table continues…
Related links
Increasing Text Fields for Feature Buttons on page 146
Adding feature buttons on page 145
3. Review the current settings of the List1, List2, and List3 fields to determine if the list the
user wants to access is assigned to their telephone.
Proposed solution–Verify feature access code
Procedure
1. Type display feature-access-codes.
2. Press Enter.
3. Verify that the user is dialing the appropriate feature access code.
Proposed solution–Verify feature button assignment
Procedure
1. Type display station nnnn, where nnnn is the user’s extension.
2. Press Enter.
3. Review the current feature button assignments to determine whether:
• The user was pressing the assigned button.
• The list number and dial code are correct.
You can use a bridged call appearance to perform operations such as conference, transfer, hold,
drop, and priority calling.
The enhanced Bridged Call Appearance feature is introduced for Communication Manager
Release 6.3.2 and later. With this enhancement, Communication Manager matches the caller
information on the bridged lines with the caller information on the principal stations.
The following table depicts the display on bridged call appearance for an incoming call when the
enhanced Bridged Call Appearance feature is active.
Calling party name is available Calling party name is unavailable
Calling party number (CPN) is <calling name> <calling number> CALL FROM <calling number>
available
Calling party number (CPN) is <calling name> <incoming trunk name>
unavailable <incoming trunk access code>
Note:
SIP phones do not support the enhanced Bridged Call Appearance feature.
Related links
Enabling Enhanced Bridged Call Appearance on page 172
8. If you want the bridged appearance to ring when a call arrives at the primary telephone,
type y. Otherwise, leave the default n.
9. Complete the appropriate field for your telephone type.
• If your primary telephone is analog, move to the Line Appearance field and enter
abrdg-appr
• If your primary telephone is digital, move to the BUTTON ASSIGNMENTS field and
enter brdg-appr.
10. Press Enter.
Btn and Ext fields appear. If Per Button Ring Control is set to y on the Station screen for
the digital set, Btn, Ext, and Ring fields appear
11. Enter the primary telephone’s button number that you want to assign as the bridged call
appearance.
This button flashes when a call arrives at the primary telephone.
12. Enter the primary telephone extension.
13. If the system displays the Ring field, one of the following can be set:
• If you want the bridged appearance to ring when a call arrives at the primary telephone,
type y.
• If you do not want the bridged appearance to ring, leave the default n.
14. Press Enter to save your changes.
15. To see if an extension has any bridged call appearances assigned, type list bridge and the
extension.
16. Press Enter.
The user at extension 4567 can now use this list by dialing the feature access code for the
list and the dial code for the number they want to dial. Alternatively, you can assign an
abbreviated dialing button to this station using which the user can press one button to dial
a specific stored number on one of their three assigned abbreviated lists.
2. On page 2 of the Class of Service screen, set the Match BCA Display with Principal field
to y.
Extension to Cellular
Use the Extension to Cellular feature to extend your office calls and Communication Manager
features to a cellular telephone. For a detailed description of the Extension to Cellular feature and
how to administer it, see Extension to Cellular in Avaya Aura® Communication Manager Feature
Description and Implementation or Avaya Extension to Cellular User Guide.
Note:
Do not use station type XMOBILE. Endpoints
configured as XMOBILE cannot access
important enhancements to EC500, such as
support for SIP trunk groups.
Language To review the office telephone feature button All
Translations assignments
Numbering-Public/ Assign 10-digit caller identification. All
Unknown Format
Coverage Path Set up number of unanswered rings prior to Number of Rings
coverage.
Trunk Group Enable Call Detail Recording for outgoing trunk. CDR Reports
DS1 Circuit Pack Administer a DS1 Circuit pack for R2MFC for Signaling Mode: CAS
EC500 use.
Interconnect: CO
Trunk Group Administer a trunk group for EC500 use. Group Type
Note:
The Timer subfield displays, and defaults to n. Leaving the default setting of n
excludes the timer state
5. Set the optional Timer subfield to y to include an Extension to Cellular timer state for the
administered feature button
When the timer state is included, the Extension to Cellular user can activate a one-hour
timer to temporarily disable Extension to Cellular through this administered feature button.
6. Press Enter.
The corresponding feature button on the office telephone is now administered for
Extension to Cellular.
Note:
The feature status button on the office telephone indicates the current state of
Extension to Cellular regardless of whether the feature was enabled remotely or
directly from the office telephone.
For additional information, see the Avaya Extension to Cellular User’s Guide, 210-100-700.
Terminal Self-Administration
Before a user can enter the TSA Admin mode, their telephone must be completely idle. After a
user presses the Admin button and enters a security code (if necessary), they are prompted, via
the telephone’s display, to choose features to administer to buttons on their telephone. The user
can add, replace, or delete any of the following feature-button types from their telephone.
• CDR Account Code
• Automatic Dial
• Blank
• Call Forwarding
• Call Park
• Call Pickup
• Directed Call Pickup
• Group Page
• Send All Calls
• Toggle Swap
• Activate Whisper Page
• Answerback for Whisper Page
• Whisper Page Off
End-user button changes are recorded to the Communication Manager server’s history log so that
remote services can know what translations are changed.
A home station can be visited by another EMU user while the user is registered as an EMU visitor
elsewhere. A home station can be used as a visited station while the principal user’s EC500 or
other Off-PBX applications are active. And the principal user can activate an Off-PBX application
even if their home station is being visited by another EMU user.
Note:
In this document, any telephone that is not the primary telephone is referred to as the “visited”
telephone and any server that is not the home server of the primary telephone is referred to as
the “visited server.”
If the registration is successful, the visited telephone assumes the primary telephone’s
extension number and some specific administered button types. The display on the primary
telephone shows Visited Registration Active: <Extension>: The extension number that
displays is the extension number of the visited telephone
Note:
The speed dialing list that is stored on the primary telephone and the station logs are not
downloaded to the visited telephone.
2. Enter the security code of your primary telephone when you activate or deactivate EMU.
The security code is administered on page one of the Station screen. The security code
can be up to eight numbers. No letters or special characters are allowed. Once the security
code is entered, the system displays a * in the Security Code field.
3. On the Station screen, scroll down till you find the EMU Login Allowed field.
The EMU Login Allowed field applies to the visited station and must be set to y for EMU.
The valid entries to this field are y or n, with n as the default. You must set this field to y to
allow this telephone to be used as a visited station by an EMU user.
4. Select Enter to save your changes.
Note:
If you set the Send EMU Visitor CPN field to y, you must set the Format field on the
same page to either public or unk-pvt.
4. Click Enter to save your changes.
Activating EMU
Procedure
1. At the visited telephone, enter the EMU activation Feature Access Code (FAC).
You must enter the EMU activation FAC of the server in the location where you are dialing
from.
2. Enter the extension of your primary telephone set.
3. Enter the security access code of your primary telephone set. This is the security code
administered on the primary telephone’s station form on the home server.
• If the registration is successful, you hear confirmation tone.
• If the registration is unsuccessful, you hear audible intercept.
Audible intercept is provided when:
• The registration was rejected by the home server.
• The telephone where the registration attempt is made is not administered for EMU use.
• The 15 second timer expires at the visited server.
If the home server receives a request from a visited server for a telephone that already has
an EMU visitor registration active, the old registration is terminated and the new
registration is approved. If the primary telephone is in-use when a registration attempt is
made, the registration attempt fails.
Deactivating EMU
Procedure
1. At the visited telephone, enter the EMU deactivation FAC.
You must enter the EMU deactivation FAC of the server in the location where you are
dialing from.
2. Enter the extension number of the primary telephone.
3. Enter the security code of the visited telephone.
If the visited telephone does not deactivate, the telephone remains in the visited state.
4. To deactivate the visited telephone you can perform a busy-out, release busy-out at the
visited server.
5. Enter the EMU feature deactivation code and the security code of the visited telephone at
the home server location.
6. Press the <mute>RESET function on the IP telephone.
Note:
Anytime the visited telephone performs a reset, the EMU registration is deactivated.
7. Unplug the visited DCP set for a period of one minute
Unplugging or disconnecting a 4600 series set will not deactivate the set.
Attendant Consoles
The attendant console is the main answering position for your organization. The console operator
is responsible for answering incoming calls and for efficiently directing or "extending" calls to the
appropriate telephone. Using the attendant console your attendants can monitor:
• system problems
• toll fraud abuse
• traffic patterns
The number of consoles you can have in your organization varies depending on your Avaya
solution.
302 attendant consoles
Avaya Communication Manager supports the following 302 attendant consoles: the 302A/B, 302C,
and 302D consoles. You might have a basic or enhanced version of these consoles.
To compare and contrast the consoles, view the diagrams below.
• 302A/B
• 302C
• 302D
302D Console
The 302D console provides the following enhancements to the 302C console:
• Modular handset or headset connection
The console accepts a standard RJ11, 4-pin modular handset or headset. This connection
replaces the quarter-inch, dual-prong handset or headset connection.
• Activate or deactivate push-button
You can use the push-button on the left side of the console to activate or deactivate the
console. The system displays a message on the console identifying that the button must be
pressed to activate the console.
• Two-wire DCP compatibility
The console is compatible with two-wire DCP circuit packs only, not four-wire DCP circuit
packs.
• Headset volume control
302A/B Console
Note:
Button numbers map to physical positions on the console.
Figure notes:
1. Call processing area
2. Handset
3. Handset cradle
4. Warning lamps and call waiting lamps
5. Call appearance buttons
6. Feature area
7. Trunk group select buttons
8. Volume control buttons
9. Select buttons
10. Console display panel
302C Console
Note:
Button numbers map to physical positions on the console.
Figure notes:
1. Handset
2. Handset cradle
3. Call processing area
4. Warning lamps and call waiting lamps
5. Outside-line buttons
6. Display buttons
7. Display
8. Select buttons
9. Volume control buttons
10. Outside-line buttons
11. Feature buttons
12. Call appearance buttons
Related links
Attendant Consoles on page 183
302D Console
Note:
Button numbers map to physical positions on the console.
Related links
Attendant Consoles on page 183
COS you assigned on the Console Parameters screen. To avoid unexpected behavior,
you should assign the same COR and same COS on both screens.
If you give your attendants an individual extension, users can call the attendant directly by
dialing the extension.
Attendants can use Individual attendant extensions to use features that an attendant group
cannot use — for example, you can assign them to hunt groups.
5. In the Console Type field, enter night-only.
This indicates how this console is used in your organization—as a principal, day only, night
only, or day/night console. You can have only one night-time console (night only or day/
night) in the system.
6. In the Port field, enter the port address for this console.
7. Type a name to associate with this console in the Name field.
8. In the DIRECT TRUNK GROUP SELECT BUTTON ASSIGNMENTS fields, enter trunk
access codes for the trunks you want the attendant to be able to select with just one
button.
9. If you are using the Enhanced Selector console, set the HUNDREDS SELECT BUTTON
ASSIGNMENTS that you want this console to have.
If you want this console to be able to access extensions in the range 3500 to 3999, you
need to assign them 5 Hundreds Select Buttons: 35 for extensions 3500 to 3599, 36, 37,
38, and 39.
10. Assign the Feature Buttons that you want the 302 console to have.
To determine which buttons you can assign to a console, see Attendant Console Feature
Buttons.
Tip:
Feature buttons are not numbered top-to-bottom on the attendant console, as you
might expect.
11. Press Enter to save your changes.
Related links
Attendant Console Feature Buttons on page 189
1. List: List number 1 to 3 where the destination number is stored. DC: Dial codes of
destination number.
2. Grp: The split group number for ACD.
3. Code: Enter a stroke code (0 through 9).
4. TAC: local-tgs — TAC of local TG
remote-tgs — (L-TAC) TAC of TG to remote PBX
remote-tgs — (R-TAC) TAC of TG on remote PBX
The combination of local-tgs/remote-tgs per console must not exceed 12 (maximum).
Label associated button appropriately so as to easily identify the trunk group.
5. Grp: Enter a hundreds group number (1 through 20).
6. Enhanced Hospitality must be enabled on the System-Parameters Customer-Options
(Optional Features) screen.
7. Ext: Can be a VDN extension.
8. Link: A link number — 1 to 8 for multi-carrier cabinets, 1 to 4 for single-carrier cabinets.
9. Grp: A trunk group number.
10. Grp: Group number of the hunt group.
11. The attendant can alternate between active and split calls.
12. VDN of Origin must be enabled.
Procedure
1. Type change console-parameters.
2. Press Enter
The system displays the Console Parameters screen.
3. In the Calls in Queue Warning field, enter 3.
The system lights the console’s second call waiting lamp if the number of calls waiting in
the attendant queue exceeds 3 calls. Click Next to display page 2.
4. In the Time in Queue Warning field, enter 20.
The system issues a reminder tone if a call waits in the attendant queue for more than 20
seconds.
5. Press Enter to save changes.
Note:
Some of the settings on the individual Attendant Console screens can override your
system-wide settings.
The system displays the Attendant Console screen, so you can verify that you are
removing the correct attendant.
6. If the attendant that you have chosen is the correct attendant, save the changes.
If the system displays an error message that the attendant group must be taken out of
night service before removal or change, deactivate the Night Service feature.
7. If the extension has a voice mailbox, remove the extension from voice mail service.
8. Type save translations and press Enter.
9. Unplug the console from the jack and store it for future use.
Note:
You do not need to:
• Delete the extension from the coverage paths. The system automatically adjusts
coverage paths to eliminate the extension.
• Disconnect the wiring at the cross-connect field.
Note:
The extension and port address remain available for assignment at a later date.
Tip:
You can find more information about attendant backup in the GuestWorks Technician
Handbook.
Procedure
1. Type change console-parameters.
2. Press Enter.
The system displays the Console Parameters screen.
3. In the Backup Alerting field, enter y.
4. Press Enter to save changes.
The system will now notify anyone with an attendant queue calls button when the attendant
queue reaches the warning level or when the console is in night service.
5. Type change station 4345.
6. Press Enter.
The system displays the Station screen.
7. In one of the Button Assignment fields, enter atd-qcalls.
The atd-qcalls button provides the visual alerting for this telephone. When this button is
dark (idle state), there are no calls in the attendant queue. When the button shows a
steady light (busy state), there are calls in the attendant queue. When button shows a
flashing light (warning state), the number of calls in the attendant queue exceeds the
queue warning. The backup-telephone user also hears an alerting signal every 10
seconds.
8. Press Enter to save changes.
Now you need to train the user how to interpret the backup alerting and give them the
TAAS feature access code so that they can answer the attendant calls.
Displaying administration
This chapter provides information about the messages that appear on the screens of display
telephones.
Your system uses automatic incoming call display to provide information about incoming calls to a
display telephone that is active on a call. The information is displayed for 30 seconds on all
telephones except Call Master telephones. The display goes blank after 30 seconds on Call
Master telephones. However the information for each new call overrides the existing message.
The system displays the call information on the display only if the call terminates at the telephone.
For example, if the call is forwarded to another extension, the system does not display the call
information.
See the Telephone Displays feature description in the Avaya Aura® Communication Manager
Feature Description and Implementation, 555-245-505.
• Polish
• Romanian
• Russian
• Servian
• Slovak
• Swedish
• Ukrainian
6. Set the Display Language field on the Station screen to unicode. Note that the Station
screen displays the unicode keyword only if a Unicode-capable telephone is entered in the
Station screen Type field. To use a user-defined file, set the Display Language field on
the Station screen to user-defined.
Note:
There is no uninstall option for Phone Message files. You can reload a new Phone
Message file. This will overwrite existing Phone Message files.
Script Script Tag Start Code.. Script or Block Name SAT Screen
Number Name
Bit (hex) End Code
31 40000000 Viet
27 04000000 Jpan
28 08000000 ChiS
29 10000000 2F00..2FDF Kangxi Radicals ChiT
30 20000000 Korn
31 40000000 Viet
27 04000000 Jpan
28 08000000 ChiS
29 10000000 3000..303F CJKV Symbols and Punctuation ChiT
30 20000000 Korn
31 40000000 Viet
27 04000000 3040..309F Hiragana Jpan
27 04000000 30A0..30FF Katakana Jpan
29 10000000 3100..312F Bopomofo ChiT
32 80000000 3130..318F Hangul Compatibility Jamo Hang
29 10000000 31A0..31BF Bopomofo Extended ChiT
27 04000000 31F0..31FF Katakana Phonetic Extensions Jpan
27 04000000 Jpan
28 08000000 ChiS
29 10000000 3200..32FF Enclosed CJK Letters and Months ChiT
30 20000000 Korn
31 40000000 Viet
27 04000000 Jpan
28 08000000 ChiS
29 10000000 3300..33FF CJKV Compatibility ChiT
30 20000000 Korn
31 40000000 Viet
27 04000000 Jpan
28 08000000 CJKV Unified Ideographs ChiS
29 10000000 3400..4DBF Extension ChiT
30 20000000 A Korn
31 40000000 Viet
Table continues…
Script Script Tag Start Code.. Script or Block Name SAT Screen
Number Name
Bit (hex) End Code
27 04000000 Jpan
28 08000000 ChiS
29 10000000 4E00..9FFF CJKV Unified Ideographs ChiT
30 20000000 Korn
31 40000000 Viet
32 80000000 AC00..D7AF Hangul Syllables Hang
27 04000000 Jpan
28 08000000 ChiS
29 10000000 F900..FAFF CJK Compatibility Ideographs ChiT
30 20000000 Korn
31 40000000 Viet
00000100 FB50..FDFF Arabic Presentation Forms-A Arab
27 04000000 Jpan
28 08000000 ChiS
29 10000000 FE30..FE4F CJK Compatibility Forms ChiT
30 20000000 Korn
31 40000000 Viet
00000100 FE70..FEFF Arabic Presentation Forms-B Arab
26 02000000 FF00..FFEF Halfwidth and Fullwidth Forms Kana
Fixing Problems
Symptom Cause and Solution
Characters that display are not what you thought you This feature is case sensitive. Check the table to
entered. make sure that you entered the right case.
If you enter ~c, the system will display * instead. Lower-case “c” has a specific meaning in Avaya
Communication Manager, and therefore cannot be
mapped to any other character. The system
displays an asterisk “*” in its place.
If you enter ~-> or ~<-, the system does not display These characters do not exist as single keys on
anything. the standard US-English keyboard. Therefore the
system is not programmed to handle them.
Enhanced display characters appear in fields that If an existing display field contains a tilde (~)
you did not update. followed by Roman characters, and you update
and submit that screen after this feature is
activated, that field will display the enhanced
character set.
Nothing displays on the terminal at all. Some unsupported terminals do not display
anything if a special character is presented. Check
the model of display terminal that you are using.
If you enter a character with a descender then the Some of the unused characters in Group2a have
system displays it with part of it cut off. descenders that do not appear entirely within the
display area. These characters are not included in
the character map. For these characters (g,j,p,q,y),
use Group1 equivalents.
Related Topics
See the Telephone Displays and the Administrable Display Languages feature descriptions in the
Avaya Aura® Communication Manager Feature Description and Implementation, 555-245-205 for
more information.
To view the dial pad letter/number/symbol mapping tables used for the integrated directory, see
Telephone Display in Avaya Aura® Communication Manager Feature Description and
Implementation, 555-245-205.
the directory, use the touch-tone buttons to key in a name, and retrieve an extension from the
directory.
Note:
When you assign a name beginning with two tildes (~~) to a telephone, and Display
Character Set on the System Parameters Country-Options screen is set to Roman, the name
does not appear in the integrated directory. Note that this is the only way to hide a name in the
integrated directory.
The example below shows how to assign directory telephone buttons for extension 2000.
Our button assignment plan is set up so that telephone buttons 6, 7, and 8 are used for the
directory. Remember, the name you type in the Name field on the first page of the Station screen
is the name that the system will display when the integrated directory is accessed on a telephone
display, except when the name is “hidden”, as described in the Note above.
Procedure
1. Type change station 2000.
2. Press Enter.
3. Press Next Page to move to the BUTTON ASSIGNMENTS section on Station screen
(page 4).
4. In Button Assignment field 6, type directory.
5. In Button Assignment field 7, type next.
6. In Button Assignment field 8, type call-display.
7. Press Enter to save your changes.
2. If you require to customize your system, carefully read and understand each field
description before you make any changes.
For more information on redirecting calls, see Covering calls redirected to an off-site
location.
For information on setting the Caller Response Interval before a call goes to coverage, see
“Caller Response Interval” in the Call Coverage section of Avaya Aura® Communication
Manager Feature Description and Implementation, 555-245-205.
The next path is optional. It is the coverage path to which calls are redirected if the current
path’s coverage criteria does not match the call status. If the next path’s criteria matches
the call status, it is used to redirect the call; no other path is searched.
4. Fill in the Coverage Criteria fields.
You can see that the default sets identical criteria for inside and outside calls. The system
sets coverage to take place from a busy telephone, if there is no answer after a certain
number of rings, or if the DND (do not disturb), SAC (send all calls), or Go to Cover button
has been pressed or corresponding feature-access codes dialed.
5. Fill in the Point fields with the extensions, hunt group number, or coverage answer group
number you want for coverage points.
Each coverage point can be an extension, hunt group, coverage answer group, remote
number, or attendant.
6. Click Enter to save your changes.
Tip:
If you want to see which extensions or groups use a specific coverage path, type
display coverage sender group n, where n is the coverage path number. For
example, you should determine which extensions use a coverage path before you
make any changes to it.
7. Press Enter.
The system displays the Coverage Path screen.
Tip:
Before making changes, you can use display coverage sender group n, to
determine which extensions or groups use path n.
8. In the Coverage Point field, type the remote code number that you use for the external
number.
9. Save the changes.
Note:
If you do not have the Coverage of Calls Redirected Off-Net feature, the system
cannot monitor the call once it leaves the network. The call ends at the remote
coverage point.
In this example, the coverage rings at extension 4101, then redirects to the external
number. If you administer Coverage of Calls Redirected Off-Net and the external number is
unanswered or is busy, the call redirects to the next coverage point. In this example, the
next point is Point 3 (h77 or hunt group 77).
For more information on coverage, see Call Coverage in Avaya Aura® Communication
Manager Feature Description and Implementation, 555-245-205.
3. To define your coverage plan, enter the time of day and path number for each day of the
week and period of time.
Enter time in a 24-hour format from the earliest to the latest. For this example, assume that
coverage path 1 goes to a coworker, path 2 to home, and path 3 to voice mail.
Define your path for the full 24 hours (from 00:01 to 23:59) in a day. If you do not list a
coverage path for a period of time, the system does not provide coverage for that time.
4. Click Enter to save your changes.
5. Now assign time-of-day coverage to a user. For example, we use extension 2054:
a. Type change station nnnn, where nnnn is the extension number.
b. Press Enter.
The system displays the Station screen.
c. Move your cursor to Coverage Path 1 and type t plus the number of the Time of Day
Coverage Table.
d. Click Enter to save your changes.
Now calls to extension 2054 redirect to coverage depending on the day and time that each
call arrives.
Call Forwarding
This section explains how to administer various types of automatic call forwarding. To provide call
forwarding to your users, assign each extension a class of service (CoS) that allows call
forwarding. Then assign call-forwarding buttons to the user telephones (or give them the feature
access code (FAC) for call forwarding) so that they can easily forward calls.
Use the Station screen to assign the COS and any call-forwarding buttons. Call Forwarding can be
enabled using a button or a feature access code.
All Call Forward buttons have an Ext sub-field. The Ext sub-field allows you to apply Call
Forward button to a station other than the station on which the Call Forward button is configured.
For example, you can have a Call Forward button that applies to station X, and you can forward
calls on behalf of X.
Within each class of service, you can determine whether the users in that COS have the following
call forwarding features:
• Call Forwarding All Calls — Users can use this to redirect all incoming calls to an extension,
attendant, or external telephone number.
• Call Forwarding Busy/Don’t Answer — Users can use this to redirect calls only if their
extensions are busy or they do not answer.
• Restrict Call Fwd-Off Net — This prevents users from forwarding calls to numbers that are
outside your system network.
As the administrator, you can administer system-wide call-forwarding parameters to control when
calls are forwarded. Use the System Parameters Call Coverage/Call Forwarding screen to set the
number of times an extension rings before the system redirects the call because the user did not
answer (CFWD No Answer Interval). For example, if you want calls to ring 4 times at an extension
and, if the call is unanswered, redirect to the forwarding number, set this parameter to 4.
You also can use the System Parameters Call Coverage/Call Forwarding screen to determine
whether the forwarded-to telephone can override call forwarding to allow calls to the forwarded-
from telephone (Call Forward Override). For example, if an executive forwards incoming calls to
an attendant and the attendant needs to call the executive, the call can be made only if the Call
Forwarding Override field is set to y.
5. When they get dial tone, they dial their "forwarding-to" off-site or on-site number.
In this example, enter 9-555-2081.
6. When they hear the 3-beep confirmation tone, they disconnect.
3. Press
• 1 for internal calls
• 2 for external calls
• 3 for all calls
You hear a confirmation tone if the deactivation was successful.
2. Press
• 1 for Enhanced Call Forwarding Unconditional
• 2 for Enhanced Call Forwarding Busy
3. Press
• 1 to forward internal calls
• 2 to forward external calls
• 3 to forward all calls
4. Optionally, dial the destination number to which calls will be forwarded.
If you do not enter a destination number, the previous destination number will be used.
Dial # at the end of an external destination number, or wait for the timeout to expire.
You hear a confirmation tone if the action was successful.
Procedure
1. Dial the remote access number, including barrier code or authentication code.
2. Dial the feature access code to activate the Enhanced Call Forwarding feature.
3. Press one of the following numbers for the required enhanced call forwarding options:
• 1 for Enhanced Call Forwarding Unconditional.
• 2 for Enhanced Call Forwarding Busy.
• 3 for Enhanced Call Forwarding No Reply.
4. Press one of the following numbers for the required call type:
• 1 to forward internal calls.
• 2 to forward external calls.
• 3 to forward all calls.
5. Dial the forwarding station extension.
6. Dial the destination number to which calls will be forwarded.
Note:
After dialing the external destination number, press the pound key (#) or wait for the
timer to expire.
The system generates the confirmation tone.
4. Press one of the following numbers for the required call type:
• 1 for internal calls.
• 2 for external calls.
• 3 for all calls.
5. Dial the forwarding station extension.
6. Dial the destination number to which calls must be forwarded.
Note:
After dialing the external destination number, dial the pound key (#) or wait for the timer to
expire.
The system generates the confirmation tone.
Night Service
You can use night service to direct calls to an alternate location when the primary answering group
is unavailable. For example, you can administer night service so that anyone in your marketing
department can answer incoming calls when the attendant is at lunch or has left for the day.
Once you administer night service to route calls, your end-users merely press a button on the
console or a feature button on their telephones to toggle between normal coverage and night
service.
There are five types of night service:
• Night Console Night Service — directs all attendant calls to a night or day/night console
• Night Station Night Service — directs all incoming trunk or attendant calls to a night service
destination
• Trunk Answer from Any Station (TAAS) — directs incoming attendant calls and signals a bell
or buzzer to alert other employees that they can answer the calls
• Trunk Group Night Service — directs incoming calls to individual trunk groups to a night
service destination
• Hunt Group Night Service — directs hunt group calls to a night service destination
13. From a telephone with console permissions, dial the call forwarding feature access code,
then the hunt group’s extension, followed by the main number of AUDIX.
In our example, dial 51002.
Note:
You should receive the confirmation tone (3 beeps). This step is very important as calls
to the LDN night service extension do not follow coverage.
14. In voice mail, build your auto attendant with the extension of the Listed Directory Number,
not the hunt group.
The originally dialed number was the LDN. That is what Communication Manager passes
to the voice mail. In the case of the INTUITY and newer embedded AUDIX Voice Mail
systems, you can use the Auto Attendant routing table to send the calls to a common Auto
Attendant mailbox.
Any calls to extension 2000 now go to extension 3000 (the guard’s desk).
Any “0” seeking calls go to extension 3000 (the guard’s desk).
Sending LDN calls to the attendant during the day and to the
TAAS bell at night
Procedure
1. Type change console-parameters.
2. Press Enter.
The system displays the Console Parameters screen.
3. In the DID-LDN Only to Night Ext?field, type y.
Using this only listed directory number calls (LDN) go to the listed directory night service
number extension.
4. In the Ext Alert Port (TAAS) field, type 01A070.
This is the port address assigned to the external alerting device.
5. Click Enter to save your changes.
Any DNIS extension 2000 calls now go to the TAAS bell.
Any “0” seeking calls now go to the TAAS bell.
Procedure
To deactivate the Night Service feature, disable the Night Service feature button on the principal
attendant console or on the designated phone.
Call Pickup
Users might need to answer a call that is ringing at a nearby desk. With Communication Manager,
a user can answer a call that is ringing at another telephone in three ways:
• Use Call Pickup. With Call Pickup, you create one or more pickup groups. A pickup group is
a collection, or list, of individual telephone extensions. A pickup group is the way to connect
individual extensions together. For example, if you want everyone in the payroll department
to be able to answer calls to any other payroll extension, you can create a pickup group that
contains all of the payroll extensions.
A user extension can belong to only one pickup group. Also, the maximum number of pickup
groups might be limited by your system configuration.
Using their own telephones, all members in a pickup group can answer a call that is ringing at
another group member telephone. If more than one telephone is ringing, the system selects
the extension that has been ringing the longest.
• Use Extended Call Pickup. With Extended Call Pickup, you can define one or more extended
pickup groups. An extended pickup group is the way to connect individual pickup groups
together.
There are two types of extended pickup groups: simple and flexible. You administer the type
of extended pickup groups on a system-wide basis. You cannot have both simple and flexible
extended pickup groups on your system at the same time.
Based on the type of extended pickup group that you administer, members in one pickup
group can answer calls to another pickup group.
For more information, see Setting up simple extended pickup groups, Setting up flexible
extended pickup groups, and Changing extended pickup groups.
• Use Directed Call Pickup. With Directed Call Pickup, users specify what ringing telephone
they want to answer. A pickup group is not required with Directed Call Pickup. You must first
administer Directed Call Pickup before anyone can use this feature.
For more information, see Setting up Directed Call Pickup.
Throughout this procedure on pickup groups and extended pickup groups, we show examples to
make Call Pickup easier to understand.
the Call Pickup button status lamp flash and presses the Call Pickup button to answer the
call, the:
- Call Pickup button status lamp of the answering group member goes out.
- Call Pickup button status lamp of the called group member goes out.
- Call Pickup button status lamps of the other pickup group members go out.
- Telephone of the called group member stops ringing.
If another call comes into the pickup group,
• The call will alert to the answering group member. However, the answering group member
cannot answer the call using the call pickup button unless the member puts the original call
on hold. Once the group member is off the original call, that member is alerted for
subsequent group calls and can answer the call using the call pickup button.
• The call alerts to all other group members and can be answered by any of these other group
members.
In all scenarios, the call appearance button on the telephone of the called group member:
• Stays steadily lit if the Temporary Bridged Appearance on Call Pickup? field on the
Feature-Related System Parameters screen is set to y. The called group member can join
the call in progress by pressing the lit call appearance button. The person who picked up the
call can either stay on the call or disconnect the call.
• Goes out if the Temporary Bridged Appearance on Call Pickup? field on the Feature-
Related System Parameters screen is set to n. The called group member cannot join the call
in progress.
The system uses an algorithm to select what call is answered when multiple calls ring or alert in a
call pickup group at the same time. The system searches the extensions of the call pickup group
until the system finds an extension with a call that is eligible to be answered with Call Pickup. The
system selects this call to be answered. The next time that a group member answers a call with
Call Pickup, the system bypasses the extension that was answered most recently, and starts the
search at the next extension.
For example, if a group member attempts to use Call Pickup when two calls are ringing at
extension A and one call is ringing at extension B, the system selects the calls in the following
order:
• One of the calls to extension A
• The call to extension B
• The remaining call to extension A
The system also determines which call that a group member answers when multiple calls ring or
alert at the same telephone. The system selects the call with the lowest call appearance, which is
usually the call appearance that is nearest to the top of the telephone.
For example, when calls ring or alert at the second and the third call appearances, the system
selects the call on the second call appearance for the user to answer.
With Communication Manager Release 6.3.6, call pickup alerting has changed. If the calling
station and the called station belong to the same pickup group, both the stations will not get the
pickup notification. However, other members of the pickup group will receive the notification. This
behavior is applicable to all types of stations, such as DCP, H.323, and SIP. For example, Station
A, Station B, and Station C are in a pickup group. If Station A is used to call to Station B, Station C
will get the pickup notification. But, Station A and Station B will not get the pickup notification.
Example
This procedure shows how to set up a new pickup group 11 for Accounting. For the rest of these
procedures, let us say that you also set up these pickup groups:
• 12 for Billing
• 13 for Credit Services
• 14 for Delinquency Payments
• 15 for Executives
• 16 for Finance
Related links
Simple extended pickup groups on page 242
Flexible Extended Pickup Groups on page 245
3. Click Enter.
The system deletes the pickup group.
Related links
Simple extended pickup groups on page 242
Flexible Extended Pickup Groups on page 245
Members of two or more individual pickup groups can answer each others calls using simple
extended pickup groups. In a simple extended pickup group, an individual pickup group can be
assigned to only one extended pickup group.
All members of one pickup group can answer the calls to the other pickup groups within the simple
extended pickup group.
Caution:
Before you administer what type of extended pickup group to use (none, simple, or flexible),
be sure that your pickup group objectives are well thought out and defined.
In this exercise, you will:
• Set up the system for simple extended pickup groups.
• Assign a FAC so that users can answer calls.
• Add pickup groups, if needed
• Assign two pickup groups to an extended pickup group.
Related links
Adding Pickup Groups on page 237
Deleting a pickup group on page 241
Pickup Numbers
The Pickup Number column that is associated with the Pickup Group Number is the unique
number that users must dial after dialing the Extended Group Call Pickup Access Code FAC to
answer a call in that pickup group.
For example, let us say that the Extended Group Call Pickup Access Code FAC is *39. In the
above example:
• A user in pickup group 13 must dial *391 to answer a call to pickup group 14, because pickup
group 14 is assigned to Pickup Number 1.
• A user in pickup group 14 must dial *390 to answer a call to pickup group 13, because pickup
group 13 is assigned to Pickup Number 0.
Note:
To minimize the number of digits that a user has to dial, first assign pickup groups to Pickup
Numbers 0 to 9.
• By assigning Pickup Numbers 0 to 9, all users only needs to dial a single digit (0 to 9)
after the FAC to answer the call.
• If you assign a number greater than 9 (10 to 24) to any pickup group, all users must dial
two digits (00 to 24) after the FAC to answer the call.
Caution:
Before you administer what type of extended pickup group to use (none, simple, or flexible),
be sure that your pickup group objectives are well thought out and defined.
In this exercise, you will:
• Set up the system for flexible extended pickup groups.
• Assign a FAC so that users can answer calls.
• Add or change pickup groups, and "point" a pickup group to an extended pickup group.
Related links
Adding Pickup Groups on page 237
(Executives) to extended pickup group 4. For this example, let us say that you followed the
same procedure for pickup group 16 (Finance).
Note:
You do not have to populate the Extended Group Number field. You can leave the
Extended Group Number field blank. You can just as easily point the pickup group to
a different extended pickup group. For example, you can point pickup group 13 (Credit
Services) to extended pickup group 2, even though pickup group 13 is not a member
of extended pickup group 2.
4. Click Enter to save your changes.
Assigning pickup groups to a flexible extended pickup group
Procedure
1. Type change extended-pickup-group n, where n is the number of the extended
pickup group.
In this example, type change extended-pickup-group.
2. Press Enter.
The system displays the Extended Pickup Group screen for extended pickup group 4
3. Add pickup group 16 (Finance) to this extended pickup group.
4. Click Enter to save your changes.
Example
Here is how flexible extended pickup groups work.
Notice that pickup groups 13, 14, and 16 are now members of extended pickup group 4. On the
Pickup Group screen for pickup groups 13, 14, and 16, you also pointed each pickup group to
extended pickup group 4.
Pickup group 15 (Executives) is not a member of extended pickup group 4. However, on the
Pickup Group screen for group 15 (Figure 96: Pickup Group screen on page 266), you pointed
pickup group 15 to extended pickup group 4.
In addition to answering calls to their own pickup group:
Notice that pickup groups 13, 14, and 16 are now members of extended pickup group 4. On the
Pickup Group screen for pickup groups 13, 14, and 16, you also pointed each pickup group to
extended pickup group 4.
Pickup group 15 (Executives) is not a member of extended pickup group 4. However, on the
Pickup Group screen for group 15 (Figure 96), you pointed pickup group 15 to extended pickup
group 4.
In addition to answering calls to their own pickup group:
• Any member of pickup group 13 can answer calls to pickup groups 14 and 16.
• Any member of pickup group 14 can answer calls to pickup groups 13 and 16.
• Any member of pickup group 16 can answer calls to pickup groups 13 and 14.
• Any member of pickup group 15 can answer calls to pickup groups 13, 14, and 16 because
pickup group 15 points to extended pickup group 4.
• Any member of pickup groups 13, 14 and 16 cannot answer calls to pickup group 15 because
pickup group 15 is not a member of extended pickup group 4.
Hunt Groups
A hunt group is a group of extensions that receive calls according to the call distribution method
you choose. When a call is made to a certain telephone number, the system connects the call to
an extension in the group.
Use hunt groups when you want more than one person to be able to answer calls to the same
number. For example, set up a hunt group for:
• a benefits department within your company
• a travel reservations service
We’ll type ucd-loa so a call goes to the agent with the lowest percentage of work time
since login.
Note:
The COS for all hunt groups defaults to 1. Therefore, any changes to COS 1 on the
Class of Service screen changes the COS for all your hunt groups. A COS field does
not appear on the Hunt Group screen.
6. Click Next Page to find the Group Member Assignments screen.
7. In the Ext field, type the extensions of the agents you want in the hunt group.
We’ll type 1011, 1012, and 1013.
Tip:
For a ddc group type (also known as "hot seat" selection), the call is sent to the
extension listed in the first Ext field. The system uses this screen to determine the
hunting sequence.
8. Click Enter to save your changes.
The Name fields are display-only and do not appear until the next time you access this
hunt group.
Setting up a queue
About this task
You can tell your server running Communication Manager how to handle a hunt-group call when it
cannot be answered right away. The call waits in "queue."
We will tell Communication Manager that as many as 10 calls can wait in the queue, but that you
want to be notified if a call waits for more than 30 seconds.
You also want Communication Manager to send a warning when 5 or more calls are waiting in the
queue. This warning flashes queue-status buttons on telephones that have a status button for this
hunt group. When the buttons flash, everyone answering these calls can see that the help-line
calls need more attention.
Procedure
1. Type change hunt-group n, where n is the number of the hunt group to change.
2. Press Enter.
In our example, type change hunt-group 5.
The system displays the Hunt Group screen.
3. In the Queue field, type y.
4. In the Queue Length field, type the maximum number of calls that you want to wait in the
queue.
In our example, type 10.
5. In the Calls Waiting Threshold field, type the maximum number of calls that can be in the
queue before the system flashes the queue status buttons.
In our example, type 5.
6. In the Time Warning Threshold field, type the maximum number of seconds you want a
call to wait in the queue before the system flashes the queue status buttons.
In our example, type 30.
7. Click Enter to save your changes.
TTY callers can be accommodated by creating a hunt group that includes TTY-equipped agents.
The TTY itself looks a little like a laptop computer, except that it has a one- or two-line
alphanumeric display instead of a computer screen. The cost of a typical TTY is approximately
three hundred dollars. Although many TTYs can connect directly with the telephone network via
analog RJ-11 jacks, Avaya recommends that agents be equipped with TTYs that include an
acoustic coupler that can accommodate a standard telephone handset. One reason for this
recommendation is that a large proportion of TTY users are hearing impaired, but still speak
clearly. These individuals often prefer to receive calls on their TTYs and then speak in response.
This requires the call center agent to alternate between listening on the telephone and then typing
on the TTY, a process made considerably easier with an acoustically coupled configuration.
Although TTY-emulation software packages are available for Personal Computers, most of these
do not have the ability to intermix voice and TTY on the same call.
For a TTY hunt group, you can record TTY announcements and use them for the hunt group
queue. To record announcements for TTY, simply follow the same steps as with voice recordings
from your telephone (see Managing Announcements). However, instead of speaking into your
telephone to record, you type the announcement with the TTY device.
Note:
For an alternative to simply creating a TTY hunt group, you can use vectors to process TTY
calls. With vectors, you can allow TTY callers and voice callers to use the same telephone
number. In this case, you can also record a single announcement that contains both TTY
signaling and a voice recording.
Tip:
If you set the delay announcement interval to 0, callers automatically hear the
announcement before anything else. This is called a “forced first announcement.”
6. Click Enter to save your changes.
You can use the same announcement for more than one hunt group.
Tip:
Remember, Communication Manager automatically fills in some of the information
when you type your vector step. Press Tab.
Playing an Announcement
About this task
Write a vector to play an announcement for callers in a queue. Use the announcement to ask
callers to wait. You need to record the announcement before the vector can use it.
Let us play our announcement 4001, asking the caller to wait, then play music for 60 seconds,
then repeat the announcement and music until the call is answered. The goto command creates
the loop to repeat the announcement and the music. Unconditionally means under all conditions.
Tip:
Rather than loop your vectors directly back to the announcement step, go to the previous
queue-to step. This way, if for some reason the call does not queue the first time,
Communication Manager can attempt to queue the call again. If the call successfully queued
the first time though, it merely skips the queue-to step and plays the announcement. The
system cannot queue a call more than once in the exact same priority level.
To play and repeat an announcement, write this vector (steps 3-5):
Procedure
1. Keep it Blank.
2. Type queue-to main split 47 pri 1.
3. Type announcement 4001 (All agents are busy, please wait...).
4. Type wait-time 60 secs hearing music.
5. Type goto step 2 if unconditionally.
Caution:
Add a stop vector step only after calls are routed to a queue. If a stop vector is
executed for a call not in queue, the call drops.
Procedure
1. Type goto step 10 if staff agents split 10 > 0.
2. Type goto step 8 if time-of-day is all 17:00 to all 8:00.
3. Type queue-to main split 47 pri 1.
4. Type announcement 4001 (All agents are busy, please wait...).
5. Type wait-time 60 secs hearing music.
6. Type goto step 2 if unconditionally.
7. Type stop.
8. Type messaging split 99 for extension 2000.
9. Type stop.
10. Type messaging split 99 for extension 2001.
11. Type stop.
When there is an emergency, fire drill, or holiday, the supervisor or agent logs into this split.
When an agent logs into split 10, the system looks at vector step 1, sees that more than 0
people are logged into split 10, and sends calls to step 10 (which sends to messaging split
99). When your business returns to normal and the agent logs out of split 10, call handling
returns to normal.
5. Type stop.
Inserting a Step
About this task
It is easy to change a vector step and not have to retype the entire vector. We will add
announcement 4005 between step 3 and step 4 in vector 20.
Procedure
1. Type change vector 20. Press Enter.
The system displays the Call Vector screen.
2. Click Edit.
3. Type i followed by a space and the number of the step you want to add.
In our example, type i 4.
4. Type the new vector step.
We will type announcement 4005 (Please wait...).
5. Click Enter to save your changes.
Tip:
When you insert a new vector step, the system automatically renumbers the rest of the
vector steps and all references to the vector steps. Communication Manager inserts a
“*” when the numbering needs more attention.
Deleting a Step
Procedure
1. Type change vector 20. Press Enter.
The system displays the Call Vector screen.
2. Click Edit.
3. Type d followed by a space and the number of the step you want to delete.
In our example, type d 5.
Tip:
You can delete a range of vector steps. For example, to delete steps 2 through 5, type
d 2-5. Click Enter.
4. Click Enter to save your changes.
Tip:
When you delete a vector step, the system automatically renumbers the rest of the
vector steps and all references to the vector steps. An asterisk (*) is inserted when the
numbering needs more attention.
Variables in Vectors
You can use Call Vectoring feature called Variables in Vectors (VIV) to create variables that can be
used in vector commands to:
• Improve the general efficiency of vector administration
• Provide increased manager and application control over call treatments
• Create more flexible vectors that serve the needs of your customer and contact center
operations
The vector variables are defined in a central variable administration table. Values assigned to
some types of variables can also be quickly changed by means of special vectors, Vector
Directory Numbers (VDNs), or Feature Access Codes (FACs) that you administer specifically for
that purpose. Different types of variables are available to meet different types of call processing
needs. Vector variables can be added to “consider location,”“messaging,” and "“adjunct routing”
vector steps when the Call Center Release is 3.0 or later. Depending on the variable type,
variables can use either call-specific data or fixed values that are identical for all calls. In either
case, an administered variable can be reused in many vectors. For a more detailed description of
variable types and purposes, see Avaya Call Center Call Vectoring and Expert Agent Selection
(EAS) Guide, 07-600780.
Note:
For voice callers, the burst of TTY tones lasts about one second and sounds like a bird
chirping.
2. In vector step 3, since the TTY caller entered 1 in vector step 2, the TTY caller is sent to
vector step 8, at which point the caller is put in queue for a TTY-enabled agent in split 47.
Note:
The voice caller is sent to vector step 3 also, but a voice caller does not go to vector
step 8 because the caller did not enter 1 at vector step 2. Instead, voice callers
continue on to vector step 4, where they connect to split 48.
3. While the TTY caller waits in queue, he or she hears silence from vector step 9, then the
announcement in vector step 10, and is then looped back to wait with silence by vector
step 11.
See the Avaya Call Center Call Vectoring and Expert Agent Selection (EAS) Guide,
07-600780, for more information.
Automated Attendant competes with several features for ports on the Call Classifier —
Detector circuit pack or equivalent. For more information on circuit pack, see Avaya Aura®
Communication Manager Hardware Description and Reference.
Security alert:
Vector fraud is one of the most common types of toll fraud because vectors route calls based
on the class of restriction (COR) assigned to the VDN. See the Avaya Toll Fraud and Security
Handbook, 555-025-600 for more information.
Tip:
BCMS must be enabled to use both. Use display system-parameters
customer-options to see if BCMS is enabled.
7. Click Enter to save your changes.
Procedure
1. Type add term-ext-group next.
2. Press Enter.
The system displays the Terminating Extension Group screen.
3. In the Group Extension field, type 6725.
This is the extension for the advertising group.
4. In the Group Name field, type advertising.
This is the name of the group.
5. In the Coverage Path field, type 5.
This is the number of the call coverage path for this group.
For example, you might want executives in your company to have different calling privileges than
offered to receptionists.
When you set up a COR, you specify a Facility Restriction Level (FRL) on the Class of Restriction
screen. The FRL determines the calling privileges of a user. The levels of FRL are ranked from 0
to 7 where 7 has the highest level of privileges.
On the Route Pattern screen, you can assign an FRL to each route pattern preference. The
system checks the COR when the user makes a call. The system facilitates the call if the FRL of
the caller is higher than or equal to the FRL of the route pattern.
Changing station
About this task
Use the following procedure to set up a new telephone for an executive and change station 1234
from COR 1 to COR 7.
The latest translations are assigned for COR 1 with outward restrictions. You must assign a COR
with the highest level of permissions to station 1234.
FRL 0 is the lowest permission level.
FRL 7 is the highest permission level.
Procedure
1. On the SAT command line interface, type change station 1234.
2. Press Enter.
The system displays the Station screen.
3. In the COR field, type 7.
4. To save the changes that you make, press Enter.
5. To change the permission level from FRL 0 to FRL 7, type change cor 7.
6. Click Enter.
The system displays the Class of Restriction screen.
7. In the FRL field, type 7.
8. To save the changes that you make, press Enter.
Users with COR 7 have the highest level of calling permissions.
ARS Analysis
With ARS, Communication Manager checks the digits in the number called against the ARS Digit
Analysis Table to determine how to handle the dialed digits. Communication Manager also uses
Class of Restriction (COR) and Facility Restriction Level (FRL) to determine the calling privileges.
Let us look at a very simple ARS digit analysis table. Your system likely has more defined dialed
strings than this example. Refer to the following screenshot for ARS Digit Analysis Table.
In the ARS Digit Analysis Table screen, the far-left column of lists the first digits in the dialed
string. When a user makes an outgoing call, the system analyzes the digits, looks for a match in
the ARS Digit Analysis Table, and uses the information in the matching row to determine how to
route the call.
Let us say a caller places a call to 1-303-233-1000. Communication Manager matches the dialed
digits with those in the first column of the ARS Digit Analysis Table screen.
In this example, the dialed string matches the “1”. Then Communication Manager matches the
length of the entire dialed string (11 digits) to the minimum and maximum length columns. In our
example, the 11-digit call that started with 1 follows route pattern 30 as an fnpa call.
Tip:
The first dialed digit for an external call is often an access code. If ‘9’ is defined as the ARS
access code, Communication Manager drops this digit and analyzes the remaining digits with
the ARS Analysis Table.
The Route Pattern points to the route that handles the calls that match this dial string. Call Type
tells what kind of call is made with this dial string.
Call type helps Communication Manager decide how to handle the dialed string.
tell Communication Manager to delete a 1 and an area code on calls to one of your locations, and
avoid long-distance charges by routing the call over your private network.
ARS digit conversion examples
The ARS digit conversion table reflects these values:
• ARS feature access code = 9
• AAR feature access code = 8
• Private Network Office Code (also known as Home RNX) = 222
• Prefix 1 is required on all long-distance DDD calls
• Dashes (-) are for readability only
Communication Manager maps the dialed digits to the matching pattern that most closely matches
the dialed number.
Example:
If the dialed string is 957-1234 and matching patterns 957-1 and 957-123 are in the table, the
match is on pattern 957-123.
ARS digit conversion examples table:
Operation Actual Digits Matching Replacement Modified Notes
Dialed Pattern String Address
DDD call to 9-1-303-538-1 1-303-538 362 362-1345 Call routes via
ETN 345 AAR for RNX
362
Long-distance 9-10222+DDD 10222 (blank) (blank) Call routes as
call to specified dialed with
carrier DDD # over
private network
Terminating a 9-1-201-957-5 1-201-957-5 or 222-5 222-5567 Call goes to
local DDD call 567 or 957-5 home RNX 222,
to an internal 9-957-5567 ext. 5567
station
Unauthorized 9-1-212-976-1 1-XXX-976 # (blank) “#” means end
call to intercept 616 of dialing. ARS
treatment ignores digits
dialed after
976. User gets
intercept
treatment.
International 9-011-91-6725 011-91 222-0111# 222-0111 Call routes to
calls to an 30 local server
attendant (RNX 222),
then to
attendant
(222-0111).
Table continues…
3. Press Enter to view the ARS Digit Analysis Table screen beginning with 120.
The table (on the screen) in this example shows two translations for calls that begin with
120.
First, follow the routing for a long-distance call that begins with 120 and is allowed. The
120 translation handles all dial strings 1-201 through 1-209, and there are many matches.
• A user dials 9 120 plus 8 digits (the first of the 8 digits is not 0).
• Communication Manager drops the ARS FAC (9 in our example), looks at the ARS Digit
Analysis Table for 120, and analyzes the number. It determines the call is long-distance
and sends the call over route pattern 4
Now we will follow a call that begins with the restricted area code 200. Only one string
matches this translation.
• A user dials 9 1200 plus 7 digits.
• Communication Manager drops the ARS FAC (9), and looks at the ARS Digit Analysis
Table for 1200. It determines that the call type is deny, and the call does not go through.
5. Type the number of digits for the system to delete or insert and select the call type. the
system will delete, the number of digits the system will insert, and the call type against
which the system will test the modified digit string.
You can create numbering plans and time zone and daylight saving plans that are specific for
each location. Choose your main location, and offset the local time for each location relative to the
system clock time. The main location is typically set to have offset 0.
For example, we will set up multiple locations for Communication Manager server with cabinets in
Chicago and New York. Location 1 is assigned to the cabinet in Chicago, our main office, so
Central Standard Time is used for our main location. Location 2 is assigned to the cabinet in New
York. We’ll define the numbering plan area (NPA) for the Chicago and New York locations, and set
the time zone offset for NY to show the difference in time between Eastern Standard Time and
Central Standard Time.
Tip:
Type list cabinets to see the Cabinet screen and a list of cabinets and their locations.
To define locations for cabinets in Chicago and New York:
Procedure
1. Type change locations.
2. Press Enter.
The system displays the Locations screen.
3. Type y in the ARS Prefix 1 required for 10-digit NANP calls field.
Our dial plan requires users to dial a 1 before all 10-digit (long distance) NANP calls.
4. Type Chicago in the Name field in the Number 1 row.
Use this field to identify the location.
5. Type +00:00 in the TimeZone Offset field in the Number 1 row.
In our example, the system time and the Chicago location time are the same.
6. Type 1 in the Daylight Saving Rule field in the Number 1 row.
In our example, daylight saving rule 1 applies to U.S. daylight saving time.
Tip:
Use the display daylight-savings-rules command to see what rules have
been administered on Communication Manager.
7. Type 312 in the Number Plan Area Code field in the Number 1 row.
In our example, 312 is the local area code for Chicago, location 1.
8. Type New York in the Name field in the Number 2 row
9. Type -01:00 in the TimeZone Offset field in the Number 2 row.
In our example, subtract one hour from the system clock in Chicago to provide the correct
time for the location in New York.
10. Type 1 in the Daylight Saving Rule field in the Number 2 row.
In our example, daylight saving rule 1 applies to U.S. daylight saving time, and both
locations use the same rule.
11. Type 212 in the NANP field in the Number 2 row.
In our example, 212 is the local area code for New York, location 2.
12. Press Enter to save your changes.
See Avaya Aura® Communication Manager Feature Description and Implementation,
555-245-205, for more information on the Multiple Locations feature.
Tip:
Create location-specific routing by assigning different route patterns for each location.
To define local calls for servers in Chicago and New York:
Procedure
1. Type change ars analysis location 1.
2. Press Enter.
The system displays the ARS Digit Analysis Table screen for location 1.
3. Type the information for local dialed strings and service calls in each row on the screen.
In our example, for location 1 (Chicago) local HNPA calls:
a. Type the appropriate digit in the Dialed String field.
b. Type 7 in the Total Min field.
c. Type 7 in the Total Max field.
d. Type 2 in the Route Pattern field.
e. Type hnpa in the Call Type field.
In our example, for location 1 (Chicago) local service calls:
a. Type the appropriate digits in the Dialed String field.
b. Type 3 in the Total Min field.
c. Type 3 in the Total Max field.
d. Type 1 in the Route Pattern field.
e. Type svcl in the Call Type field.
4. Press Enter to save your changes.
5. Type change ars analysis 4 location 2.
6. Press Enter.
The system displays the ARS Digit Analysis Table for location 2.
7. Type in the local HNPA and service call routing information for New York.
8. Press Enter to save your changes.
See Automatic Routing in Avaya Aura® Communication Manager Feature Description and
Implementation, 555-245-205, for more information on ARS.
See Multiple Locations in Avaya Aura® Communication Manager Feature Description and
Implementation, 555-245-205 for more information on the Multiple Locations feature.
Tip:
If you change an existing area code for a network with multiple locations, be sure to
change the Number Plan Area Code field on the Locations screen.
Procedure
1. Type change ars analysis 01157.
2. Press Enter.
a. Enter 011 (international access)
b. Enter the country code (57)
The system displays the ARS Digit Analysis Table screen.
3. Move to a blank Dialed String field.
Skip to Step 6 to deny calls to this dialed string
If the dialed string is already defined in your system, the system displays the cursor in the
appropriate Dialed String field.
4. Enter 01157 in the Dialed String field.
5. Enter 10 in the Total Min and 23 in Total Max fields.
6. Enter deny (denied) in the Route Pattern field.
7. Enter intl in the Call Type field.
8. Press Enter to save your changes.
Procedure
1. Type change authorization-code 4395721.
2. Press Enter.
The system displays the Authorization Code - COR Mapping screen.
3. In the AC field, type 4395721.
4. In the COR field, enter 2.
5. Press Enter to save your changes.
ARS Partitions
Most companies want all their users to be able to make the same calls and follow the same route
patterns. However, you might find it helpful to provide special calling permissions or restrictions to
a group of users or to particular telephones.
With ARS partitioning, you can provide different call routing for a group of users or for specific
telephones.
Note:
If you used partitioning on a prior release of Avaya Communication Manager and you want to
continue to use partitioning, please read this section carefully. In this release of Avaya
Communication Manager, partition groups are defined on the Partition Route Table. If you
want to define routing based on partition groups, use the Partition Route Table. Partition
groups are no longer defined on the Digit Analysis Table.
Related links
Setting up Time of Day Routing on page 287
To restrict the lobby telephone, you modify the routing for a partition group to enable only specific
calls, such as U.S. based toll-free 1-800 calls, and then assign this partition group to the lobby
telephone.
To enable 1-800 calls for partition group 2:
Procedure
1. Type list ars route-chosen 18002221000.
2. Press Enter.
You can use any 7-digit number following the 1800 to create an example of the dialed
string.
The system displays the ARS Route Chosen Report screen for partition group 1.
3. Record the route pattern for the selected dialed string.
In our example, the route pattern for 1800 is p1. This indicates that the system uses the
Partition Routing Table to determine which route pattern to use for each partition.
Note:
If there was a number (with no p) under Route Pattern on the Route Chosen Report,
then all partitions use the same route pattern. You need to use the Partition Routing
Table only if you want to use different route patterns for different partition groups.
4. Press Cancel to return to the command prompt.
5. Type change partition-route-table index 1.
6. Press Enter.
The system displays the Partition Routing Table screen. In our example, partition group 1
can make 1800 calls and these calls use route pattern 30.
7. In the PGN2 column that corresponds to Route Index 1, type 30.
8. Press Enter.
This tells the system to use route pattern 30 for partition group 2 and allow partition group
2 members to make calls to 1800 numbers.
Procedure
1. Type display time-of-day 1.
2. Press Enter.
The system displays the Time Of Day Routing Plan screen for plan 1.
Note:
Make a note of the routing plan that is currently in effect. In our example, this plan is
for employees who can only make local calls.
You can see that in our example, two partition group numbers control time of day
routing. PGN 1 begins one minute after midnight (00:01) every day of the week, and is
used for after-business hours and all day Saturday and Sunday. PGN 2 is assigned to
office hours Monday through Friday, not including noon (12:00) to 1:00 p.m. (13:00).
3. Press Cancel to clear the screen.
Because Jim has a COR of 2 with Time of Day Plan Number 1, the system uses Time of
Day Routing Plan 1 to route the call.
According to Time of Day Routing Plan 1, calls made between 8:00 a.m. and 11:59 a.m.
route according to the route pattern set up on PGN 1.
If Jim makes a call between 12:00 p.m. and 1:00 p.m. on Monday, the Time of Day Routing
Plan 1 is used again. However, this time the call is routed according to PGN 2.
Intercept Extension field, type the extension number that you want to assign. The number
can be a recorded announcement, a vector directory number, or a hunt group extension.
3. Press Enter to save your changes.
Note:
Once a telephone has been associated with an extension, anyone using the terminal has the
capabilities of the associated station. Be sure to execute a dissociate request if the terminal
can be accessed by unauthorized users. This is particularly important if you use PSA and
DCP extenders to permit remote DCP access.
This is the feature access code you will use to deactivate Personal Station Access at
a telephone.
See Telecommuting settings changes for information on how to associate or disassociate
PSA.
See Enterprise Mobility User for information on how to set up the Enterprise Mobility User
feature.
Related links
Telecommuting settings changes on page 306
Enterprise Mobility User on page 178
This means that the password is a system generated random number. The system
displays a 10-digit number in the Password field. Take note of this number, your user will
need it at home to access the server running Communication Manager.
5. Enter Doe, John and click OK.
This is the last name and first name of the user. The system returns you to the Password
Manager screen.
6. Select CommLink:Select Cards.
The system displays a screen containing a list of cards (for example, Card A, Card B, and
so on). Each card corresponds to a port on your Avaya Server.
7. Select Card A and click OK.
8. Select CommLink:Upload Password.
The system displays the error message screen with the message “Administrator
password not loaded".
9. Click OK.
10. Enter 123456 and click OK.
11. Select CommLink:Upload Password.
12. When upload is complete, click OK.
13. Select File:Save As.
14. Enter doe.fil in the File field and click OK to save your changes.
The tenant number of the extension being administered must be accessible by the tenant
number from which the Extended User Administration of Redirected Calls FAC is dialed or
the request is denied. If the FAC is dialed on site, the tenant number of the station or
attendant must have access to the tenant number of the extension administered. If the FAC is
dialed off site, the tenant number of the incoming trunk must have access to the tenant
number of the extension administered.
This is the station extension you configured for telecommuting. The system displays the
Station screen.
6. Complete the following fields:
a. Enter 2 in the Coverage Path 1 field.
b. Enter 8 in the Coverage Path 2 field.
See Coverage Path in Avaya Aura® Communication Manager Screen Reference, for
information about and field descriptions on the Coverage Path screen.
See Avaya Aura® Communication Manager Feature Description and Implementation,
555-245-205, for a description of the Call Coverage feature.
See Avaya Aura® Communication Manager Feature Description and Implementation,
555-245-205, for information about the Extended User Administration of Redirected Calls
feature.
Caution:
Your attempt to disable the Remote Access feature will be lost if the server running
Communication Manager is rebooted without saving translations. Therefore, execute a
save translation command after permanently disabling the Remote Access
feature.
Note:
You must set up the client device for remote login and configure the device for SSH. For
information about understanding the relevant commands for SSH, see your client PC
documentation.
Press #.
This is to confirm your new security code. You hear the confirmation tone.
Note:
If you cannot change your security code, Manager 1 can clear the problem using the
Clear Audit Summary command.
System security
You can refer to the following checklist to keep your system secure.
No. Task Description Notes
System security
You can refer to the following checklist to keep your system secure. For more information about
the various features related to security, see the Avaya Toll Fraud and Security Handbook,
555-025-600 Guide.
1. Use Enhanced Access Security Gateway to log in to Communication Manager, and secure
both system administration and maintenance ports. Customers with maintenance contracts
have access to the optional password authentication interface program.
2. Activate Security Violations Notification (SVN) to report unsuccessful attempts to access
the system. Security Violations Notification lets you automatically disable a valid login ID
following a security violation involving that login ID and disable remote access following a
security violation involving a barrier code or authorization code.
3. Secure trunks using Automatic Route Selection (ARS), Class of Restriction (COR), Facility
Restriction Levels (FRLs) and Alternate Facility Restriction Levels (AFRLs), Authorization
Codes, Automatic Circuit Assurance (ACA), and Forced Entry of Account Codes (see Call
Security Enforcement
To include an added layer of security, the user must implement the following safeguards to ensure
physical security for Communication Manager:
• Unplug and secure attendant console handsets when the attendant position is not in use.
• Lock wiring closets and server rooms.
• Keep a log book register of technicians and visitors.
• Shred information about Communication Manager from folders that you discarded.
• Always demand verification of every technician or visitor by asking for a valid identification
proof.
• Keep any reports related to trunk access codes, screen barrier codes, authorization codes, or
password information secure.
• Keep the attendant console and supporting documentation in the office secured with a
changeable combination lock.
Provide the changeable combination number to people who need to enter the office.
• Keep any documentation related to the Communication Manager operation secure.
• Label all backup tapes or flash cards with correct dates to avoid using an outdated tape or
flash card while restoring data.
You must ensure that all backup media have the correct generic software load.
Note:
Avaya recommends that you permanently disable Remote Access using the change
remote-access command. If you do permanently disable Remote Access, the code
is removed from the software. Avaya charges a fee to restore the Remote Access
feature.
3. If you use Remote Access, but only for internal calls, change announcements or remote
service observing.
a. Use a 7-digit barrier code.
b. Assign a unique COR to the 7-digit barrier code.
The unique COR must be administered where the FRL is 0, the Calling Party
Restriction field is outward, and the Calling Permissions field is n on all unique
Trunk Group COR.
c. Assign Security Violation Notification Remote to 10 attempts in 2 minutes.
d. Set the aging cycle to 90 days with 100 call limit per barrier code.
4. If you use Remote Access to process calls off-net or in any way access the public network:
a. Use a 7-digit barrier code.
b. Assign a unique COR to the barrier code.
c. Restrict the COR assigned to each barrier code by FRL level to only the required
calling areas to conduct business.
d. Set the aging cycle to 90 days with 100 call limit per barrier code.
e. Suppress dial tone where applicable.
f. Administer Authorization Codes.
g. Use a minimum of 11 digits (combination of barrier codes and authorization codes).
h. Assign Security Violation Notification Remote to 10 attempts in 2 minutes.
5. If you use vectors:
a. Assign all Vector Directory Numbers (VDN) a unique COR.
See Avaya Aura® Call Center 5.2 Automatic Call Distribution (ACD) Reference,
07-602568, and Avaya Aura® Call Center 5.2 Call Vectoring and Expert Agent
selection (EAS) Reference, 07-600780, for more information.
Note:
The COR associated with the VDN dictates the calling privileges of the VDN/
vector. High susceptibility to toll fraud exists on vectors that have “collect digits”
steps. When a vector collects digits, it processes those digits back to
Communication Manager and if the COR of the VDN allows it to complete the call
off-net, it will do so. For example, the announcement “If you know your party’s 4-
digit extension number, enter it now” results in 4 digits being collected in step 6. If
you input “90##” or “900#”, the 4 digits are analyzed and if “9” points towards ARS
and “0” or “00” is assigned in the ARS Analysis Tables and the VDN COR allows
it, the call routes out of the server to an outside local exchange or long distance
operator. The operator then connects the call to the requested number.
b. If vectors associated with the VDN do not require routing the call off-net or via AAR,
assign a unique COR where the FRL is 0, the Calling Party Restriction field is
outward, the Calling Permissions field is n on all unique Trunk Group COR.
c. If the vector has a “route-to” step that routes the call to a remote server via AAR,
assign a unique COR with a unique ARS/AAR Partition Group, the lowest FRL to
complete an AAR call, and n on all unique COR assigned to your public network
trunking facilities on the Calling Permissions.
Assign the appropriate AAR route patterns on the AAR Partition Group using the
change aar analysis partition x 2 command.
Tip:
You can use the display aar analysis print command to print a copy of
your Automatic Alternate Routing (AAR) setup before making any changes. You
can use the printout to correct any mistakes.
d. If the vector has a “route-to” step that routes the call to off-net, assign a unique COR
with a unique ARS/AAR Partition Group, the lowest FRL to complete an ARS call, and
n on all unique COR assigned to your public network trunking facilities on the Calling
Permissions.
Assign the appropriate complete dial string in the “route-to” step of the vector the
unique ARS Partition Group using the change ars analysis partition x 2
command.
6. On the Feature Access Code (FAC) screen, Facility Test Calls Access Code, the Data
Origination Access Code, and the Data Privacy Access Code fields, change from the
default or remove them.
For information about the Feature Access Code (FAC) screen, see Avaya Aura®
Communication Manager Screen Reference.
Note:
These codes, when dialed, return system dial tone or direct access to outgoing
trunking facilities. Transfers to these codes can take place via an unsecured vector
with “collect digits” steps or an unsecured voice mail system.
7. Restrict Call Forwarding Off Net on every class of service.
See Avaya Aura® Communication Manager Screen Reference, for more information on
Class of Service.
Note:
You cannot administer loop-start trunks if Call Forwarding Off Net is required.
8. If loop start trunks are administered on Communication Manager and cannot be changed
by the Local Exchange Company, block all class of service from forwarding calls off-net.
In the Class of Service screen, Restriction Call Fwd-Off Net field, set to y for the 16
(0-15) COS numbers.
See Avaya Aura® Communication Manager Screen Reference, for more information on
Class of Service.
Note:
If a station is call forwarded off-net and an incoming call to the extension establishes
using a loop-start trunk, incorrect disconnect supervision can occur at the Local
Exchange Central Office when the call terminates. This gives the caller recall or
transfer dial tone to establish a fraudulent call.
9. Administer Call Detail Recording on all trunk groups to record both incoming and outgoing
calls.
See Call information collection for more information.
10. On the Route Pattern screen, be careful assigning route patterns with an FRL of 0; these
allow access to outgoing trunking facilities.
Avaya recommends assigning routes with an FRL of 1 or higher.
Note:
An exception might be assigning a route pattern with an FRL of 0 to be used for 911
calls so even restricted users can dial this in emergencies.
Tip:
You can use the list route-pattern print command to print a copy of your
FRLs and check their status.
11. On all Trunk Group screens, set the Dial Access field to n.
If set to y, users can dial Trunk Access Codes, thus bypassing all the ARS call screening
functions.
See the Trunk Group section of Avaya Aura® Communication Manager Screen Reference,
for more information.
12. On the AAR and ARS Digit Analysis Table, set all dial strings not required to conduct
business to den (deny).
For information about this screen, see Avaya Aura® Communication Manager Screen
Reference.
13. If you require international calling, on the AAR and ARS Digit Analysis Table, use only the
011+ country codes/city codes or specific dial strings.
14. Assign all trunk groups or same trunk group types a unique Class of Restriction.
If the trunk group does not require networking through Communication Manager,
administer the Class of Restriction of the trunk group where the FRL is 0, the Calling
Party Restriction field is outward, and all unique Class of Restriction assigned to your
outgoing trunk groups are n. See Class of Restriction in Avaya Aura® Communication
Manager Screen Reference, for more information.
Tip:
You can use the list trunk-group print command to have a printout of all your
trunks groups. Then, you can use the display trunk-group x command (where x
is the trunk group) to check the COR of each trunk group.
15. Avaya recommends you administer the following on all voice mail ports:
• Assign all voice mail ports a unique COR. See Class of Restriction in Avaya Aura®
Communication Manager Screen Reference, for more information.
• If you are not using outcalling, fax attendant, or networking, administer the unique COR
where the FRL is 0, the Calling Party Restriction field is outward, and all unique
trunk group COR on the Calling Permissions are n. See Class of Restriction in Avaya
Aura® Communication Manager Screen Reference, for more information.
Note:
Avaya recommends you administer as many layers of security as possible. You can
implement Step 9 and Step 16 as a double layer of security. In the event that the voice
mail system becomes unsecured or compromised for any reason, the layer of security
on Communication Manager takes over, and vice versa.
16. Administer all fax machines, modems, and answering machines analog voice ports as
follows:
• Set the Switchhook Flash field to n.
• Set the Distinctive Audible Alert field to n. See Station in Avaya Aura® Communication
Manager Screen Reference, for more information.
17. Install a Call Accounting System to maintain call records.
In the CDR System Parameters screen, Record Outgoing Calls Only field, set to y. See
CDR System Parameters in Avaya Aura® Communication Manager Screen Reference, for
more information.
18. Call Accounting Systems produce reports of call records.
It detects telephones that are being hacked by recording the extension number, date and
time of the call, and what digits were dialed.
Sample scenario to use the Busy Verify feature for toll fraud
detection
Procedure
1. Type change station xxxx. The busy verify button is assigned to the xxxx station.
2. Press Enter.
The system displays the Station screen. For this example, type extension 1014.
3. Click Next Page until you see the Site Data fields.
4. In the BUTTON ASSIGNMENTS area, type verify, and press Enter to save your
changes.
5. To activate the feature on the telephone, click the Verify button. Then type the Trunk
Access Code and the member number to be monitored.
For more information about authorization codes setup, see Avaya Toll Fraud and Security
Handbook, 555-025-600.
In this example, type 4285193. The number of digits entered must agree with the number
assigned in the Feature-Related System Parameters screen, Authorization Code Length
field.
Note:
Remember, all authorization codes used in the system must be the same length.
11. In the COR field, enter the required Class of Restriction number from 0 through 95.
In our example, type 1.
12. Enter change trunk-group n, where n is the assigned trunk group number, and press
Enter.
13. In the Auth Code field, enter y to require callers to enter an authorization code to tandem
a call through an AAR or ARS route pattern.
The code will be required even if the facility restriction level of the incoming trunk group is
normally sufficient to send the call out over the route pattern.
14. Select Enter to save your changes.
Note:
If you are not using Remote Access, go to Step 9.
6. (Optional) Type change remote-access and press Enter.
7. (Optional) In the Disable Following A Security Violation field, type y.
This disables Remote Access following detection of a remote access security violation.
8. (Optional) Press Enter to save your changes.
9. Type change station xxxx, where xxxx is the station to be assigned the notification
halt button and press Enter.
Procedure
1. Log in to Communication Manager System Management Interface.
2. On the Administration menu, click Server (Maintenance).
3. In the left navigation pane, under Security, click Server Log Files and do the following:
4. In Enabled, select Yes or No.
5. In Protocol, click the method to transfer the syslogs.
The options are:
• UDP
• TCP
• TLS
6. In Port, type the port number of the remote syslog server.
7. In Server IP/FQDN, type the FQDN or IP address of the remote syslog server.
8. Click Submit.
Station lock
The attendant console can lock or unlock other telephones. The attendant console cannot be
locked.
• For analog stations or without a display, the user hears a special dial tone. The special dial
tone has to be administered, and the user hears the special dial tone when the station is off
hook.
After a station is locked by TOD, it can be unlocked from any other station if the Feature Access
Code (FAC) or button is used. You have to also know the Station Security Code, and that the
Manual-unlock allowed? field on the Time of Day Station Lock Table screen is set to y.
Once a station has been unlocked during a TOD lock interval, the station remains unlocked until
next station lock interval becomes effective.
If the station was locked by TOD and by Manual Lock, an unlock procedure will unlock the Manual
Lock as well as the TOD Lock (“Manual-unlock allowed?” field on the Time of Day Station Lock
Table screen is set to y).
The TOD feature does not unlock a manually locked station.
Note:
The attendant console cannot be locked by TOD or manual station lock.
Station Lock
Use the Station Lock feature to lock a telephone to prevent others from placing outgoing calls from
the telephone.
HDE Feature 96xx and 96xx and Other sets Other sets
96x1 H.323 96x1 H.323
with without
with FW without FW
display display
changes changes
Station Lock X X X X
Extension to Cellular blocked (Note 2)
(no make, answer and bridge)
Station Lock X X X X
Bridged appearances blocked (Note 3)
Station Lock X X X X
Limited Access to Feature Access
Codes and Feature Buttons
Note 1: Telephone capabilities are call log, Avaya menu, contact list, USB access and redial
button.
Note 2: If the set offers Extension to Cellular.
Note 3: If the set offers bridged appearances.
Note:
From Release 8.1.2, Avaya Aura® applications support the file system data encryption feature.
This requires a new encryption capable variant of Release 8.1E OVA as prerequisite. The
encryption can be enabled only at the time of deploying Avaya Aura® application 8.1E OVA. If
you want to execute data encryption commands, then you must deploy Release 8.1E OVA and
then apply Release 8.1.2 or later patch on it.
For more information on the encryption commands, see Maintenance Commands for Avaya
Aura® Communication Manager, Branch Gateways and Servers.
From Release 8.1.2, you can enable or disable data encryption for Avaya Aura® applications at the
time of deployment. Data Encryption is supported only for Appliance Virtualization Platform and
VMware Virtualized Environment. Once you deploy the application with data encryption, you cannot
disable data encryption after deployment.
In a software-only environment, the customer must enable the encryption at the operating system
(OS) level. To be Data Privacy compliant, the customer must first encrypt the OS and apply the
Release 8.1.2 or later patch.
For Data Privacy configuration, the software-only customer has the ability to protect data-in-transit,
by utilizing the configuration to specify that TLS connections will be used in all situations like
signaling, control, and log transport. Communication Manager Release 8.1.2 provides some further
enhancements for TLS coverage to include CDR streaming and Communication Manager-to-CMS
control channel.
For Data Privacy configuration for log retention, the software-only customer has the ability to deploy
Release 8.1.2 features.
By enabling Data Encryption, your Communication Product's certain Operational data and Log Files
will be encrypted. You will be prompted to enter a passphrase that will be used to create or access
an encryption key. You must remember the encryption passphrase, if not it can result in locking up
the system. Secondly, you will be asked to configure the option for local key storage.
It is important to note that the encryption of the disk may have a performance impact. For further
information, refer to the Avaya Product Administration guide(s). Before you select an encryption
option, please read the Data Privacy Guideline so that you may better understand these options.
By disabling Data Encryption, your Communication Product's Operational data and Log Files will not
be stored in encrypted partitions.
If encryption is enabled and the Require Encryption Pass-Phrase at Boot-Time check box is
selected, you need to reenter the encryption passphrase whenever the application reboots.
During reboot, the application prompts you to enter the encryption passphrase on VM console at
first boot and upon entering the correct encryption passphrase, the system mounts all the encrypted
disks.
If encryption is enabled and the Require Encryption Pass-Phrase at Boot-Time check box is not
selected during OVA deployment, the application creates the Local Key Store and the system does
not prompt you to type the encryption passphrase whenever the application reboots to mount the
encrypted disks. You can also set up the remote key server by using the encryptionRemoteKey
command after the deployment of the application.
The following users can run the data encryption commands:
• Customer defined privileged account
• Any profile 18 user
• Root user
Encryption of Communication Manager partitions
When you enable data encryption for Communication Manager, the system encrypts the following
partitions that have personal data.
• /var/home
• /var/log
• /var/log/audit
• /var/xln
encryptionPassphrase command
Using the encryptionPassphrase command you can manage the encryption passphrase after
deploying the application.
Syntax
encryptionPassphrase [add | change | remove | list]
Considerations
You must deploy the application with data encryption.
encryptionRemoteKey command
Using the encryptionRemoteKey command you can manage the remote key server after
deploying the application.
Syntax
encryptionRemoteKey [add | remove | list]
Considerations
You must deploy the application with data encryption.
Where:
Address is the IP address or FQDN of the remote key server.
Port is the port number of the remote key server. If you do not enter the port number the
application uses the value of default port as 80.
3. In Enter existing passphrase, type the existing encryption passphrase.
If the remote key server is not configured, the application displays the following message.
Remote key server not found
If the remote key server is configured, the application adds the remote key server. When
you add a remote key server for the first time, the application disables the local key store.
encryptionLocalKey command
Using the encryptionLocalKey command you can enable or disable the local key store after
deploying the application with data encryption.
Syntax
encryptionLocalKey [enable | disable]
Considerations
You must deploy the application with data encryption.
For example, if the remote key server is configured, the system displays the following
status:
Data Encryption: enabled
Local Key Store: disabled
Encryption Passphrase Required at Boot-time: yes
remoteKeyServers: <remoteServer1: <remoteServerIPAddress> accessible>
Once you decide that you want to add a new trunk, contact your vendor. The vendor should
confirm the type of signal you want and provide you with a circuit identification number for the new
trunk. Be sure to record any vendor-specific ID numbers or specifications in case you ever have
any problems with this trunk.
• Trunk Type
Caution:
Use the list above as a starting point and talk to your service provider. Depending on your
particular application, you might need to coordinate additional administration with your service
provider.
This field defines the direction of traffic flow on this trunk group.
7. In the Night Service field, enter 1234.
This field assigns an extension to which calls are routed outside of business hours.
8. In the Incoming Destination field, enterattd.
This field assigns an extension to which incoming calls are routed during business hours.
By entering attd in this field, incoming calls go to the attendant and the system treats the
calls as Listed Directory Number calls.
9. In the Comm Type field, enter voice.
This field defines whether a trunk group can carry voice, data, or both. Analog trunks only
carry voice and voice-grade data.
10. In the Trunk Type field, enter ground-start.
This field tells the system what kind of signaling to use on this trunk group. To prevent
glare, Avaya recommends ground start signaling for most two-way CO, FX, and WATS
trunk groups.
11. Press Next Page until you find the Outgoing Dial Type field.
12. In the Outgoing Dial Type field, enter tone.
This field tells Communication Manager how digits are to be transmitted for outgoing calls.
Entering tone actually allows the trunk group to support both dual-tone multifrequency
(DTMF) and rotary signals, so Avaya recommends that you always put tone in this field.
13. In the Trunk Termination field, enter rc.
Use rc in this field when the distance to the central office or the server at the other end of
the trunk is more than 3,000 feet. If you do not know the distance to your central office,
check with your service provider.
14. Select Enter to save your changes.
Now you are ready to add trunks to this trunk group. See Adding trunks to a trunk group
example.
• Trunk Type
• Expected Digits (only if the digits your provider sends do not match your dial plan)
Caution:
Use the list above as a starting point and talk to your service provider. Depending on your
particular application, you might need to coordinate additional administration with your service
provider.
This field tells the system what kind of signaling to use on this trunk group. In most
situations, use wink start for DID trunks to minimize the chance of losing any of the
incoming digit string.
7. In the Incoming Dial Type field, enter tone.
This field tells Communication Manager how digits are transmitted for incoming calls.
Entering tone actually allows the trunk group to support both DTMF and rotary signals, so
Avaya recommends that you always put tone in this field.
8. In the Trunk Termination field, enterrc.
Use rc in this field when the distance to the central office or the server at the other end of
the trunk is more than 3,000 feet. If you do not know the distance to your central office,
check with your service provider.
9. Select Enter to save your changes.
Now you’re ready to add trunks to this trunk group. See Adding trunks to a trunk group
example.
See Digit insertion and absorption with trunk groups for instructions on matching modifying
incoming digit strings to match your dial plan.
Caution:
Use the list above as a starting point and talk to your service provider. Depending on your
particular application, you might need to coordinate additional administration with your service
provider.
To find out what circuit packs you need, see the Avaya Aura® Communication Manager Hardware
Description and Reference, 555-245-207.
PCOL restrictions
• Abbreviated Dialing can be used with a personal CO line, but the accessed lists are
associated with the individual telephones.
• Auto Hold and Leave Word Calling do not work with calls on a personal CO line.
• Send All Calls cannot be activated for a personal CO line.
• Avaya Aura® Messaging and Avaya Messaging cannot be in the coverage path of a PCOL
group.
• Only telephones in the same PCOL group can bridge onto calls on the personal CO line. If a
user is active on his or her primary extension number on a PCOL call, bridged call
appearances of that extension number cannot be used to bridge onto the call.
• When a user puts a call on hold on a personal CO line, the status lamp associated with the
PCOL button does not track the busy or idle status of the line.
Caution:
Use the list above as a starting point and talk to your service provider. Depending on your
particular application, you might need to coordinate additional administration with your service
provider.
This field defines the direction of traffic flow on this trunk group.
7. In the Night Service field, enter 1234.
This field assigns an extension to which calls are routed outside of business hours.
8. In the Comm Type field, enter voice.
This field defines whether a trunk group can carry voice, data, or both. Analog trunks only
carry voice and voice-grade data. If you’re administering a T1 connection in North America,
enter rbavd in this field.
9. In the Trunk Type field, enter wink/wink.
This field tells the system what kind of signaling to use on this trunk group. Because we’re
receiving and sending digits over this trunk group, we’re using wink/wink signaling to
minimize the chance of losing part of the digit string in either direction.
10. Enter tone in both the Outgoing Dial Type and Incoming Dial Type fields.
These fields tell Communication Manager how digits are transmitted for incoming calls.
Entering tone actually allows the trunk group to support both DTMF and rotary signals, so
Avaya recommends that you always put tone in this field.
11. Select Enter to save your changes.
Now you’re ready to add trunks to this trunk group. See Adding trunks to a trunk group
example.
counterpart, but digital trunks must connect to a DS1 circuit pack and this circuit pack must be
administered separately. The example in this section shows you how to do this.
In most cases, Avaya recommends leaving the default settings in fields that aren’t specifically
mentioned in the following instructions. Go to the Avaya Support website at http://
support.avaya.com for more information.
Your settings in the following fields must match your provider’s settings:
• Bit Rate
• Line Coding (unless you’re using a channel service unit to convert between your line coding
method and your provider’s)
• Framing Mode
• Signaling Mode
• Interface Companding
Caution:
Use the list above as a starting point and talk to your service provider. Depending on your
particular application, you might need to coordinate additional administration with your service
provider.
See DS1 Circuit Pack in Avaya Aura® Communication Manager Screen Reference for information
on administering DS1 service.
See DS1 Trunk Service in Avaya Aura® Communication Manager Feature Description and
Implementation, 555-245-205, for detailed information on DS1 service.
Caution:
If enhanced DS1 administration is disabled, you cannot make changes to the DS1
Circuit Pack screen before you remove related member translations of all trunks from
the trunk group. See Enhanced DS1 administration.
2. Before you administer a digital trunk group, verify you have one or more circuit packs that
support DS1 with enough open ports to handle the number of trunks you need to add.
For more information about what circuit packs you need, see Avaya Aura® Communication
Manager Hardware Description and Reference.
If you use b8zs line coding and esf framing, it will be easier to upgrade your T1 facility to ISDN
should you want to. You can upgrade without reconfiguring external channel service units, and
your service provider won’t have to reconfigure your network connection.
E1 recommended settings
DS1 administration for E1 service varies from country to country. Go to the Avaya Support website
at http://support.avaya.com for more information.
Note:
Remember that the central office serving your switching system might be emulating another
country’s network protocol. If so, you’ll have to administer your circuit packs and trunk groups
to match the protocol used by your central office.
Specific combinations of settings for some of these fields are shown below.
ITC, Bit Rate, and Line Coding values for enhanced DS1 administration
The system displays ITC (Information Transfer Capability) field on the Route Pattern screen,
Trunk Group screen, and Access Endpoint screen. The Line Coding and the Bit Rate fields
appear on the DS1 Circuit Pack screen. The settings for these fields on all the screens must be
coordinated as shown in the following tables.
ITC field Bit Rate Line Coding field
restricted 1.544 Mbps ami-zcs
2.048 Mbps ami-basic
unrestricted 1.544 Mbps b8zs
2.048 Mbps hdb3
Caution:
An entry in this field is only required for some circuit packs. Dip switch settings on the
circuit pack control the signalling mode used on the trunk group, so the entry in the
Mode field must correspond to the actual setting on the circuit pack.
6. In the Type field, enter t1-comp.
An entry in this field is only required for some circuit packs.
7. Repeat steps 3 to 6, as appropriate, for the remaining trunks.
Notice that you can assign trunks in the same trunk group to ports on different circuit
packs.
8. Select Enter to save your changes.
Trunk resets
To “reset” a trunk, use the busyout command followed by the release command, both executed
in a SAT window. You can run these commands on a board, a port, a trunk group, or an individual
trunk. The availability of these commands depends on your login permissions.
Note:
These commands can tear calls down, so use them with great caution. Go to the Avaya
Support website at http://support.avaya.com for details.
central office delivers the string “4444,” Communication Manager will change it to “64444,”
an extension that fits your dial plan.
4. In the Expected Digits field, enter 4.
This field tells Communication Manager how many digits the central office sends.
Note:
The Expected Digits field does not appear on the screen for tie trunk groups.
5. Select Enter to save your changes.
directory numbers go to the same attendant group. How you administer your system for LDN calls
depends on whether the calls are coming in over DID and tie trunks or over CO and FX trunks.
As an example, let us say that one attendant group answers calls for 3 different businesses, each
with its own listed directory number:
Procedure
1. Company A — 855-2020
2. Company B — 855-1000
3. Company C — 855-1111
DID trunks and some tie trunks transmit part or all of the dialed digit string to
Communication Manager. If you want these calls to different numbers to go to one
attendant group, you must identify those numbers for Communication Manager on the
Listed Directory Numbers screen.
We will take the 3 businesses listed above as an example. We will assume your server
receives 4 digits from the central office on a DID trunk group and that you’re not using
Tenant Partitioning. To make these calls to different listed directory numbers terminate to
your attendant group:
a. Enter change listed-directory-numbers.
b. In the Ext 1 field, enter2020.
This is the LDN for Company A.
c. In the Name field, enter Company A.
The system displays the name on the console display so the attendant knows which
business the call is for and how to answer it.
d. Repeat steps 2 and 3 for the other two businesses.
You can enter up to 20 different listed directory numbers on this screen.
e. Select Enter to save your changes.
To make LDN calls over a CO or FX trunk group terminate to an attendant group, you must
type attd in the Incoming Destination field on the Trunk Group creen for that group.
When you use the Listed Directory Number screen to assign some extensions to the
attendant group, or when you enter attd in the Incoming Destination field on the Trunk
Group screen for CO or FX trunks, Communication Manager treats these calls as LDN
calls.
See Listed Directory Numbers in Avaya Aura® Communication Manager Screen Reference
for detailed information about this feature.
Procedure
1. On the Trunk Group screen for group 5, enter y in the Receive Answer Supervision field.
2. Select Enter to save your change.
Now we will administer answer supervision by timeout. We’ll set the timer to 15 seconds.
a. On the Trunk Group screen for group 5, type 15 in the Answer Supervision Timeout
field.
b. Select Enter to save your change.
For more information about this feature, see Avaya Aura® Communication Manager
Feature Description and Implementation.
• The MM720 BRI circuit pack implements both sides of the interface. You can select the
options from the BRI Trunk Circuit Pack screen
For BRI trunk connections to a public ISDN, use the TN2185, MM722, or MM720. For BRI tie
trunks between systems, use the TN2185, MM722, or MM720 on one side and the TN556B/C/D or
TN2198 on the other side. The TN2464 circuit supports T1 and E1 digital facilities.
ISDN-PRI interfaces are supported by the TN767 circuit pack (for assignment of a T1 signaling
link and up to 24 ISDN-PRI trunk group members), or the TN464C or later circuit pack (for
assignment of a T1 or E1 signaling link and up to 24 or 31 ISDN-PRI trunk group members,
respectively). The TN2464 and TN2207 circuit pack can also be used with ISDN-PRI.
• The D-channel for ISDN-PRI interfaces switches through either the TN765 Processor
Interface (PI) circuit pack or the TN778 Packet Control (PACCON) circuit pack. The D-
channel for ISDN-BRI interfaces only switches through the TN778 Packet Control (PACCON)
circuit pack.
Note:
You cannot use the TN765 circuit pack with ISDN-BRI interfaces.
• A TN780 or TN2182 Tone Clock circuit pack provides synchronization for the DS1 circuit
pack.
Note:
The TN767 cannot be used to carry the D-channel if either the TN778 (PACCON) or TN1655
(PKTINT) circuit packs are used to switch the D-channel. However, in these circumstances,
the TN767 can be used for NFAS interfaces carrying only B-channels.
Screen Field
ISDN-BRI Circuit Pack screen (if using ISDN- All
BRI interfaces) or
All
DS1 Circuit Pack screen (if using ISDN-PRI
interfaces)
ISDN Numbering - Private All
Route Pattern All
Hunt Groups ISDN Caller Display
Signaling Group (if using ISDN-PRI interfaces) All
Terminating Extension Group ISDN Caller Display
Table Notes:
• System Parameters Customer-Options (Optional Features) — The ISDN-BRI Trunks or
ISDN-PRI fields must be set to y. For a TN778 and if using ISDN-PRI interfaces, the PRI
Over PACCON field must be set to y. These features are provided via license file. To enable
these features, go to the Avaya Support website at http://support.avaya.com.
• QSIG Optional Features fields can be enabled to allow appropriate administration for
Supplementary Service Protocol.
• Feature-Related System-Parameters — Set the Send Non-ISDN Trunk Group Name as
Connected Name and Display Connected Name/Number for ISDN DCS Calls fields.
• ISDN-BRI Trunk Circuit Pack — This screen is required if using ISDN-BRI trunk interfaces.
Assign all fields as required.
• DS1 Circuit Pack — This screen is required if using ISDN-PRI interfaces.
- DS1 (T1) Circuit Pack
Assign all fields as required. For Facility Associated Signaling, up to 23 ports are
available for administration as trunk members in an associated ISDN-PRI trunk group. The
24th port is used as a signaling channel. For Non-Facility Associated Signaling, all 24
ports can be used on certain DS1 circuit packs. The D-channel signaling function for these
packs must be provided by a designated DS1 pack on its 24th channel.
- E1 Circuit Pack
Assign all fields as required. For Facility Associated Signaling, up to 30 ports are
available for administration as trunk members in an associated ISDN-PRI trunk group. Port
number 16 is used as a signaling channel.
• Maintenance-Related System-Parameters — Use this screen only for a TN778. Set the
Packet Bus Maint field to y.
• ISDN Trunk Group — Enter information in all the fields except the trunk group members.
When using ISDN-PRI interfaces, enter the members after you establish the signaling links.
• Signaling Group — This screen is required if ISDN-PRI interfaces are used. Complete all
fields. This screen identifies groups of ISDN-PRI DS1 interface B-channels for which a given
D-channel (or D-channel pair) will carry the associated signaling information (supports the
Facility and Non-Facility Associated Signaling feature). Each DS1 board that is required to
have a D-channel must be in a different signaling group by itself (unless D-channel backup is
needed, in which case a second DS1 is administered as a backup D-channel). You are not
required to select a channel for a trunk group, but if you do, you must have already defined
the trunk group as type ISDN.
Note:
The following three screens, Processor Interface Data Module, Communication Interface
Links, and Communication Processor Channel Assignment are used only to support the
ISDN-PRI interfaces using PI TN765.
• Processor Interface Data Module — Use this screen only for a TN765. Assign up to 8
interface links using 8 Processor Interface Data Module screens for multi-carrier cabinet
systems, and up to 4 links for single-carrier cabinet systems. One Processor Interface Data
Module screen must be completed for each interface link to be assigned.
• Communication Interface Links — Use this screen only for a TN765. Assign link numbers 01
to 08 for a multi-carrier cabinet system or links 01 to 04 for a single-carrier cabinet system as
required. When first administering this screen for ISDN in Communication Manager, do not
administer the Enable field.
• Communication Processor Channel Assignment — Use this screen only for a TN765. Enter
assigned link numbers and assign associated channel numbers to each link. Complete all
fields of the screen as required. When first administering this screen for ISDN in
Communication Manager, you need to:
- First, administer the Interface Links screen, except the Enable field.
- Second, administer the ISDN fields on the Processor Channel screen.
- Last, go back to the Interface Links screen and administer the Enable field.
• ISDN Numbering - Public/Unknown — Complete all fields. This screen supports the ISDN
Call Identification Display.
• ISDN Numbering - Private — Complete all fields. This screen supports the ISDN Call
Identification Display.
• Routing Pattern — Complete all fields including the Supplemental ISDN Routing
Information fields as required.
• Hunt Group — Complete the ISDN Caller Display field by entering either grp-name or
mbr-name to specify whether the hunt group name or member name, respectively, is sent to
the originating user (supports the ISDN Call Identification Display feature).
• Terminating Extension Group — Complete the ISDN Caller Display field by entering either
grp-name or mbr-name to specify whether the group name or member name, respectively,
is sent to the originating user (supports the ISDN Call Identification Display feature).
• Synchronization Plan — Assigns primary and secondary external synchronization sources for
the ISDN-BRI Trunk or DS1 circuit pack. Complete all screen fields as required.
Note:
ISDN-BRI and ISDN-PRI interfaces cannot be mixed in the same trunk group. Therefore,
consider the following:
• The earliest trunk member (the lowest numbered one) administered is considered
correct.
• If an offending member is subsequently found (meaning the first member was BRI and a
later member was PRI, or vice versa), the cursor positions on the offending member, and
the system displays the following error message: You cannot mix BRI and PRI
ports in the same trunk group.
5. Ensure that the Maximum Administered SIP Trunks field is set to greater than 1 and
include enough trunks for Q-SIP trunk group use.
6. Scroll through the screens to find the IP Trunks field.
7. Ensure that the IP Trunks field is set to y.
Note:
If the Maximum Administered IP Trunks and Maximum Administered SIP Trunks
fields are set to less than 1, or the IP Trunks field is set to n, your system is disabled
for the QSIG over SIP feature. Go to the Avaya Support website at http://
support.avaya.com for assistance.
8. Select Enter to exit the screen.
For details of hardware components, see Avaya Aura® Communication Manager Hardware
Description and Reference.
For more information about managing media gateways, see the following:
• Administering Avaya G450 Branch Gateway
• Administering Avaya G430 Branch Gateway
• Avaya Branch Gateway G450 CLI Reference
• Avaya Branch Gateway G430 CLI Reference
• Avaya G450 Branch Gateway Overview and Specification
• Avaya G430 Branch Gateway Overview and Specification
SIGNALING GROUP
Related links
Detailed description of Avaya Aura Media Server (MS) on page 374
Note:
• The Far-end Node Name can only contain a node name that has an IPv4 address.
The system displays an error if the node name has an IPv4 address.
• The Far-end Domain is auto-populated and viewable as read-only with the IP
address of the media server based on the Far-end Node Name.
• The Near-end Node Name is read-only and is auto-populated with the string procr.
change signaling-group 10 Page 1 of 2
SIGNALING GROUP
Related links
Detailed description of Avaya Aura Media Server (MS) on page 374
Adding a media-server
About this task
Use the task to add an Avaya Aura® Media Server (MS) media-server.
Before you begin
Ensure that you have added the Avaya Aura® Media Server (MS) signaling group. For more
information, see Administering Avaya Aura® Media Server signaling group Communication
Manager.
Procedure
1. Type the add media-server xx command.
2. On the MEDIA SERVER screen, in the Signaling Group field, type the signaling group of
the Avaya Aura® Media Server (MS) signaling-group created in the Adding an Avaya Aura®
Media Server (MS) signaling-group section.
add media-server 22 Page 1 of 1
MEDIA SERVER
Signaling Group:
Voip Channel License Limit:
Dedicated Voip Channel Licenses:
The VoIP Channel License Limit field can be left blank. Type the value if you want to limit
the number of channels that can be established on the specified media-server. A blank
field indicates that the channel limit is limited only by the physical capacity of the specific
Avaya Aura® Media Server (MS).
The aggregate of dedicated channels administered across all media-servers must not
exceed the number of licensed VoIP channels.
Related links
Detailed description of Avaya Aura Media Server (MS) on page 374
Related links
Detailed description of Avaya Aura Media Server (MS) on page 374
Signaling Group: 1
Voip Channel License Limit:
Dedicated Voip Channel Licenses:
Related links
Detailed description of Avaya Aura Media Server (MS) on page 374
An announcement is a recorded message that a caller can hear while the call is in a queue, or if the
caller receives an intercept message for any reason. An announcement is often used in conjunction
with music.
The source for announcements can be either integrated or external.
• Integrated announcements are integrated in a circuit pack in the carrier, such as the TN2501AP
circuit pack or embedded in a gateway processor board. This board is called a vVAL source.
• External announcements are stored on a separate piece of equipment called an adjunct and
played back from the adjunct equipment.
This chapter uses the term announcement source to refer to integrated or external sources of
announcements.
For information on music streaming from media server, see chapter “Music streaming configuration”
of the documentImplementing and Administering Avaya Aura® Media Server.
2. Verify that your users have 6400-, 7400-, 8400-, or 9400-series DCP (digital) telephones.
When using feature access codes, observers must choose a mode at the start of the
session. They cannot switch to the other mode without ending the session and
beginning another.
Note:
Feature access codes are required for remote observing.
• Telephone dialing
• Hayes AT command dialing
• Administered connections
• Hotline dialing
Character Use
SPACE, -, (, and) improves legibility. Communication Manager ignores these
characters during dialing.
+ character (wait) interrupts or suspends dialing until the user receives dial tone
, (pause) inserts a 1.5-second pause
% (mark) indicates digits for end-to-end signaling (touch-tone). This is required
when the trunk is rotary. It is not required when the trunk is touch-
tone.
UNDERLINE or BACKSPACE corrects previously typed characters on the same line
@ deletes the entire line and starts over with a new DIAL: prompt
Each line of dialing information can contain up to 42 characters (the + and % characters count as
two each).
Examples of dialing are:
• DIAL: 3478
• DIAL: 9+(201) 555-1212
• DIAL: 8, 555-2368
• DIAL: 9+555-2368+%9999+123 (remote access)
2. If the CMD: prompt does not appear, press Break A + T at the same time, and then press
Enter..
3. At the CMD: prompt, the user types and presses au Enter.
4. To disconnect, enter +++.
Analog modems
When a telephone user places a data call with a modem, the user dials the data-origination
access code assigned in the system before dialing the endpoint.
Alphanumeric Dialing
Alphanumeric Dialing enhances data-terminal dialing using which users can place data calls by
entering an alphanumeric name rather than a long string of numbers.
For example, a user could type 9+1-800-telefon instead of 9+1-800-835-3366 to make a call.
Users need to remember only the alpha-name of the far-end terminating point.
You can use Alphanumeric Dialing to change a mapped string (digit-dialing address) without
having to inform all users of a changed dial address. Users dial the alpha name.
When a user enters an alphanumeric name, the system converts the name to a sequence of digits
according to an alphanumeric-dialing table. If the entered name is not found in the table, the
system denies the call attempt and the user receives either an Invalid Address message
(DCP) or a Wrong Address message (ISDN-BRI).
Because data terminals access Communication Manager via DCP or ISDN-BRI data modules,
dialing procedures vary:
• For DCP, at the DIAL: prompt users type the alphanumeric name. Press Enter.
• For ISDN-BRI, at the CMD:prompt users type d, a space, and the alphanumeric name. Press
Enter.
More than one alphanumeric name can see the same digit string.
Data Hotline
Data Hotline provides for automatic-nondial placement of a data call preassigned to an endpoint
when the originating server goes off-hook. Use for security purposes.
The endpoint can be used for hotline dialing if the users can use the endpoint software to select
the dial function without entering a number.
Data Privacy
Data Privacy protects analog data calls from being disturbed by any of the system’s overriding or
ringing features.
To activate this feature for a call, the user must dial the Data Privacy FAC in the beginning
of the call. If Data Privacy is disabled on the calling station’s COS, the user hears intercept
tone immediately after dialing the Data Privacy FAC.
Default Dialing
Default Dialing provides data-terminal users who dial a specific number the majority of the time a
very simple method of dialing that number. Normal data terminal dialing and alphanumeric dialing
are unaffected.
Default Dialing enhances data terminal (keyboard) dialing using which a data terminal user can
place a data call to a pre-administered destination by either pressing Enter at the DIAL: prompt
(for data terminals using DCP data modules) or typing d and pressing Enter at the CMD: prompt
(for data terminals using ISDN-BRI data modules). The data-terminal user with a DCP data
module can place calls to other destinations by entering the complete address after the DIAL:
prompt (normal data terminal dialing or alphanumeric dialing). The data-terminal user with an
ISDN-BRI data module can place calls to other destinations by typing d, a space, the complete
address. Press Enter after the CMD: prompt.
Note:
DU-type hunt groups connecting the system to a terminal server on a host computer have
hunt-group extensions set to no keyboard dialing.
For the AT command interface supported by the 7400A/7400B/8400B data module, to dial the
default destination, enter the ATD command (rather than press return).
Data Restriction
Data Restriction protects analog-data calls from being disturbed by any of the system’s overriding
or ringing features or system-generated tones.
Data Restriction applies to both voice and data calls.
Once you administer Data Restriction for an analog or multiappearance telephone or trunk group,
the feature is active on all calls to or from the terminal or trunk group.
Note:
Do not assign Data Restriction to attendant consoles.
• Priority Calls
Priority Calls are allowed if the analog station is idle. Call Waiting (including Priority Call
Waiting) is denied if the station is busy. However, Priority Calls appear on the next available
line appearance on multiappearance telephones.
• Service Observing
A data-restricted call cannot be service observed.
remote user share control of the data module. Actions of the user at the telephone might
affect the remote user.
- 1-Button Transfer to Data
The telephone user can transfer a call to the Data-Only Off-Premises Extension. The Data
Extension button lamp on the telephone lights and the Call in Progress lamp on the data
module lights during a data call.
- Data Call Preindication
The multiappearance telephone user presses the idle associated Data Extension button to
reserve a data module. The data module is busy to all other users. When the user
reserves a data module, the lamp associated with the Data Extension button winks and
lights at any other associated telephones. A remote user receives the BUSY message
when attempting to originate a call.
- Return-to-Voice
To establish a data call, the telephone user presses the associated busy Data Extension
button to transfer the call to the telephone. The data module associated with the Data
Extension button is disconnected from the call. The Call in Progress lamp on the data
module goes dark.
built-in DTDM), and the 7505D, 7506D, and 7507D have a DCP interface but have built-in
data module functionality. Both are administered by means of the Station screen in
Communication Manager.
7400D
This data module supports synchronous operation with CMS and DCS. It provides synchronous
data transmissions at speeds of 19.2-Kbps full duplex.
Administered Connections
Use the Administered Connections (AC) feature to establish an end-to-end connection between
two access or data endpoints. Communication Manager automatically establishes the connection
based on the attributes that you administer. The Administered Connections feature provides the
following abilities:
• Support of both permanent and scheduled connections
• Autorestoration (preserving the active session) for connections that are routed over Software
Defined Data Network (SDDN) trunks
• An administrable retry interval from 1 to 60 minutes for each AC
• An administrable alarm strategy for each AC
• An establish, retry, autorestoration order that is based on administered priority
connections are maintained between the server and both endpoints. In addition to maintaining the
active session, AC also provides a high level of security by prohibiting other connections from
intervening in active sessions. Autorestoration is usually complete before the 60-second endpoint
holdover interval. If autorestoration is successful, the call might be maintained, but this is not
guaranteed. The restoration is transparent to the user, with the exception of a temporary disruption
of service while restoration is in progress. A successful restoration is indicated by the restored
value in the Connection State field on the Administered-Connection Status screen. Although a
restoration is successful, the data session might not be preserved.
If autorestoration is inactive, or if the AC is not routed over SDDN trunks, the server immediately
attempts a fast retry to reestablish the connection. The server also attempts a retry if the
originating endpoint caused the drop. With fast retry, connections are not maintained on both
ends. Fast retry is not attempted for an AC that was last established with fast retry, unless that AC
is active for at least 2 minutes. If autorestoration or fast retry fails to restore or reestablish the
connection, the call drops, and the AC goes into retry mode. Retry attempts continue, at the
administered retry interval, as long as the AC is scheduled to be active.
• ISDN-PRI
• Tie
5. On the Class of Restriction screen, administer all fields.
6. On the Class of Service screen, administer all fields.
7. On the Dial Plan Parameters screen, administer the Local Node Number field with a
number from 1-63 that matches the DCS switch node number and the CDR node number.
8. On the Administered Connection screen, administer all fields.
9. On the Station screen, assign one button as ac-alarm.
10. On the Attendant Console screen, assign one button as ac-alarm.
Modem Pooling
Modem Pooling allows switched connections between digital-data endpoints (data modules) and
analog-data endpoints via pods of acoustic-coupled modems. The analog-data endpoint is either a
trunk or a line circuit.
Data transmission between a digital data endpoint and an analog endpoint requires conversion
through a modem, because the DCP format used by the data module is incompatible with the
modulated signals of an analog modem. A modem translates DCP format into modulated signals
and vice versa.
Modem Pooling feature provides pools of integrated-conversion modems and combined-
conversion modems.
Integrated-conversion modem pools have functionality integrated on the Pooled Modem circuit
pack, providing two modems. Each one emulates a TDM cabled to a 212 modem. Integrated are
modem pools unavailable in countries that use A-law companding.
Combined-conversion modem pools are TDMs cabled to any TDM-compatible modem.
Combined-conversion modem pools can be used with all systems.
The system can detect the needs for a modem. Data calls from an analog-data endpoint require
that the user indicate the need for a modem, because the system considers such calls to be voice
calls. Users indicate this need by dialing the data-origination access code field on the Feature
Access Code (FAC) screen before dialing the digital-data endpoint.
The system provides a Hold Time parameter to specify the maximum time any modem can be
held but not used (while a data call is in queue).
Two groups of different configurations are available for Personal Computer Interface: group 1 uses
DCP and group 2 uses the ISDN-BRI (Basic Rate Interface) protocol.
The group 1 configurations consist of DCP configurations that use a DCP expansion card in the
PC to link to the Avaya Server. Group 1 (shown in DCP PC interface configuration (Group 1) on
page 420) uses the following connections:
• The Personal Computer Interface card plugs into an expansion slot on the Personal
Computer. The card has 2 standard 8-pin modular jacks (line and telephone).
• The digital telephone plugs into the telephone jack on the Personal Computer Interface card.
• The line jack on the card provides a digital port connection to Avaya DEFINITY®servers.
• The distance between the Personal Computer Interface card and the PBX should be no more
than 1524m for 24-gauge wire or 1219m for 26-gauge wire.
The group 2 configurations link to the server using a Personal Computer/ISDN Interface card
installed in the Personal Computer. This group can include a stand-alone Personal Computer
terminal, or up to 4 telephones, handsets, or headsets. Group 2 (shown in the figure on page 421)
uses Personal Computer/ISDN Interface cards (up to four cards) which plug into expansion slots
on the Personal Computer. These cards each provide 2 standard 8-pin modular-jack connections
for both line connections (to the Avaya Server) and telephone connections. A standard 4-pin
modular jack is also available for use with a handset or headset.
Personal Computer Interface users have multiple appearances (depending on the software
application used) for their assigned extension. Designate one or more of these appearances for
use with data calls. With the ISDN-BRI version, you can use up to 4 separate Personal Computer/
ISDN Interface cards on the same Personal Computer. Assign each card a separate extension,
and assign each extension one or more appearances. The availability of specific features depends
on the COS of the extension and the COS for Communication Manager. Modem Pooling is
provided to ensure general availability of off-net data-calling services.
Administering a PC interface
Procedure
On the Station screen, set the Type field to pc.
Wideband Switching
Wideband Switching provides the ability to dedicate 2 or more ISDN-PRI B-channels or DS0
endpoints for applications that require large bandwidth. It provides high-speed end-to-end
communication between endpoints where dedicated facilities are not economic or appropriate.
ISDN-BRI trunks do not support wideband switching.
Wideband Switching supports:
• High-speed video conferencing
• WAN disaster recovery
• Scheduled batch processing (for example, nightly file transfers)
• LAN interconnections and imaging
• Other applications involving high-speed data transmission, video transmission, or high
bandwidth
have 30 B-channels, 1 D-channel, and a framing channel. Data flows bidirectionally across the
facility between the server that is running Communication Manager and the ISDN-PRI terminal
adapter.
PRI endpoints with Wideband Switching
A PRI-endpoint (PE) is a combination of DS0 B-channels on a line-side ISDN-PRI facility to which
an extension is assigned.
A PE can support calls of lower bandwidth. In other words, a PE that has a width of six DS0
channels can handle a call of one channel of 64 Kbps, up to and including six channels totaling
384 Kbps. Also, a PE can support calls on nonadjacent channels. For example, an endpoint
application that is connected to a PE that is defined as using B-channels 1 through 6 of an ISDN-
PRI facility could use B-channels 1, 3, and 5 successfully to originate a call.
If the PE is administered to use flexible channel allocation, the algorithm for offering a call to the
PE starts from the first DS0 that is administered to the PE. Since only one active call is permitted
on a PE, contiguous B-channels are always selected unless one or more B-channels are not in
service.
A PE remains in service unless all the B-channels are out of service. In other words, if B-channel 1
is out of service and the PE is five B-channels wide, the PE can still handle a wideband call of up
to four B-channels wide. A PE can only be active on a single call at any given time. That is, the PE
is considered to be idle, active or busy, or out of service.
One facility can support multiple separate and distinct PEs within a single facility. Non-overlapping
contiguous sets of B-channel DS0s are associated with each PE.
Universal digital signal level 1 board
The universal digital signal level 1 (UDS1) board is the interface for line-side and network facilities
that carries wideband calls.
Wideband Switching nonsignaling endpoint applications
Wideband Switching can also support configurations that use nonsignaling, non-ISDN-PRI line-
side T1 or E1 facilities. The endpoint applications are the same as those that are defined for
configurations with signaling.
Data service unit/channel service unit with Wideband Switching
The device service unit (DSU)/channel service unit (CSU) passes the call to the endpoint
application. Unlike terminal adapters, the DSU/CSU does not have signaling capability.
Note:
No DSU/CSU is needed if the endpoint application has a fractional T1 interface.
Line-side (T1 or E1) facility with Wideband Switching
This facility, like the ISDN-PRI facility, is composed of a group of DS0s (24 for a T1 facility and 32
for an E1 facility; both T1 and E1 use 2 channels for signaling purposes). Line-side facilities are
controlled solely from the server or Avaya Server. Through the access-endpoint command, a
specific DS0 or group of DS0s is assigned an extension. This individual DS0 or group, along with
the extension, is known as a Wideband Access Endpoint (WAE).
Wideband access endpoint
WAEs have no signaling interface to the server or Avaya Server. These endpoints simply transmit
and receive wideband data when the connection is active.
Note:
Communication Manager can determine if the connection is active, but this does not
necessarily mean that data is actually coming across the connection.
A WAE is treated as a single endpoint and can support only one call. If all DS0s comprising a
wideband access endpoint are in service, then the wideband access endpoint is considered in
service. Otherwise, the wideband access endpoint is considered out of service. If an in-service
wideband access endpoint has no active calls on its DS0s, it is considered idle. Otherwise, the
wideband access endpoint is considered busy.
Multiple WAEs are separate and distinct within the facility and endpoint applications must be
administered to send and receive the correct data rate over the correct DS0s. An incoming call at
the incorrect data rate is blocked.
Given this facility list concept, the algorithms have the ability to search for trunks, by facility, in an
attempt to satisfy the bandwidth requirements of a given wideband call. If one facility does not
have enough available bandwidth to support a given call, or it is not used for a given call due to
the constraints presented in the following section, then the algorithm searches the next facility in
the trunk group for the required bandwidth (if there is more than one facility in the trunk group).
In addition to searching for channels based on facilities and required bandwidth, Port Network
(PN) preferential trunk routing is also employed. This PN routing applies within each algorithm at a
higher priority than the constraints put on the algorithm by the parameters listed later in this
section. In short, all facilities that reside on the same PN as the originating endpoint are searched
in an attempt to satisfy the bandwidth of a given call, prior to searching any facilities on another
PN.
H11 channels
When a trunk group is administered to support H11, the algorithm to satisfy a call requiring 1,536
Kbps of bandwidth uses a fixed allocation scheme. That is, the algorithm searches for an available
facility using the following facility-specific channel definitions:
• T1: H11 can only be carried on a facility without a D-channel being signaled in an NFAS
arrangement (B-channels 1-24 are used).
• E1: Although the 1,536 Kbps bandwidth could be satisfied using a number of fixed starting
points (for example, 1, 2, 3, and so forth), the only fixed starting point being supported is 1.
Hence, B-channels 1-15 and 177-25 always are used to carry an H11 call on an E1 facility.
If the algorithm cannot find an available facility within the trunk that meets these constraints, then
the call is blocked from using this trunk group. In this case, the call can be routed to a different
trunk group preference via Generalized Route Selection (GRS), at which time, based on the
wideband options administered on that trunk group, the call would be subject to another hunt
algorithm (that is, either the same H11 algorithm or perhaps an N x DS0 algorithm described in a
later paragraph).
Note that on a T1 facility, a D-channel is not considered a busy trunk and results in a facility with a
D-channel always being partially contaminated. On an E1 facility, however, a D-channel is not
considered a busy trunk because H11 and H12 calls can still be placed on that facility; an E1
facility with a D-channel and idle B-channels is considered an idle facility.
H12 channels
Since H12 is 1,920 Kbps, which is comprised of 30 B-channels, a 1,920-Kbps call can be carried
only on an E1 facility. As with H11, the hunt algorithm uses a fixed allocation scheme with channel
1 being the fixed starting point. Hence, an H12 call is always carried on B-channels 1 through 15
and 17 through 31 on an E1 facility, as the following table shows. When the system is offered any
other call other than a 1,536-Kbps call, the algorithm behaves as it does when H11 is optioned.
DS0s that comprise each channel
Facility ISDN interface H11 H12
T1 23B + D - -
T1 24B (NFAS) 1-24 -
E1 30B + D 1 through 15, 17 through 25 1 through 15, 17 through 31
E1 31B (NFAS) 1 through 15, 17 through 25 1 through 15, 17 through 31
H0 channels
When a trunk group is administered to support H0, the algorithm to satisfy a call requiring 384
Kbps of bandwidth also uses a fixed allocation scheme. Unlike the H11 fixed scheme which only
supports a single fixed starting point, the H0 fixed scheme supports 4 (T1) or 5 (E1) starting
points. The H0 algorithm searches for an available quadrant within a facility based on the direction
of trunk or hunt administered. If the algorithm cannot find an available quadrant within any facility
allocated to this trunk group, then the call is blocked from using this trunk group. Again, based on
GRS administration, the call might route to a different trunk group preference and be subject to
another algorithm based on the wideband options administered.
Note that a D-channel is considered a busy trunk and results in the top most quadrant of a T1, B-
channels 19 to 24, always being partially contaminated. This is not true for NFAS.
If this H0 optioned trunk group is also administered to support H11, H12, or N x DS0, then the
system also attempts to preserve idle facilities. In other words, when offered a narrowband, H0, or
N x DS0 call, the system searches partially-contaminated facilities before it searches to idle
facilities.
N x DS0 channels
For the N x DS0 multi-rate service, a trunk group parameter determines whether a floating or a
flexible trunk allocation scheme is to be used. The algorithm to satisfy an N x DS0 call is either
floating or flexible.
• Floating (Contiguous) — In the floating scheme, an N x DS0 call is placed on a contiguous
group of B-channels large enough to satisfy the requested bandwidth without any constraint
being put on the starting channel (that is, no fixed starting point trunk).
• Flexible — In the flexible scheme, an N x DS0 call is placed on any set of B-channels as long
as the requested bandwidth is satisfied. There is absolutely no constraint such as contiguity
of B-channels or fixed starting points. Of course, as with all wideband calls, all the B-
channels comprising the wideband call must reside on the same ISDN facility.
Regardless of the allocation scheme employed, the N x DS0 algorithm, like the H11 and H12
algorithms, attempts to preserve idle facilities when offered B, H0, and N x DS0 calls. This is
important so that N x DS0 calls, for large values of N, have a chance of being satisfied by a
given trunk group. However, if one of these calls cannot be satisfied by a partially-
contaminated facility and an idle facility exists, a trunk on that idle facility is selected, thus
contaminating that facility.
There are additional factors to note regarding specific values of N and the N x DS0 service:
- N = 1 — this is considered a narrowband call and is treated as any other voice or
narrowband-data (B-channel) call.
- N = 6 — if a trunk group is optioned for both H0 and N x DS0 service, a 384-kbps call
offered to that trunk group is treated as an H0 call and the H0 constraints apply. If the H0
constraints cannot be met, then the call is blocked.
- N = 24 — if a trunk group is optioned for both H11 and N x DS0 service, a 1,536-kbps call
offered to that trunk group is treated as an H11 call and the H11 trunk allocation
constraints apply.
- N = 30 — if a trunk group is optioned for both H12 and N x DS0 service, a 1,920-kbps call
offered to that trunk group is treated as an H12 call and the H12 trunk allocation
constraints apply.
The following table describes the blocking strategy for the different channel types.
Channel type Blocking minimization strategy
H0 Preserve idle quadrants
H11 Preserve idle facilities
H12 Preserve idle facilities
Flexible N x DS0 Preserve idle facilities
Floating N x DS0 Preserve idle facilities as first priority
Call Coverage
A WAE cannot be administered as a coverage point in a call-coverage path.
Call Detail Recording (CDR)
When CDR is active for the trunk group, all wideband calls generate CDR records. The CDR
feature flag indicates a data call, and CDR records contain bandwidth and Bearer Capability Class
(BCC).
Call Forwarding
You must block Call Forwarding through Class of Service (COS).
Call Management System (CMS) and Basic Call Management System (BCMS)
Wideband calls can be carried over trunks that are measured by CMS and BCMS. Wideband
endpoints are not measured by CMS and BCMS.
Call Vectoring
PRI endpoints use a vector directory number (VDN) to dial. For example, PRI endpoint 1001 dials
VDN 500. VDN 500 points to Vector 1. Vector 1 can point to other PRI endpoints such as route-to
1002, or route-to 1003, or busy.
Certain applications use Call Vectoring. When an incoming wideband call hunts for an available
wideband endpoint, the call can point to a VDN, that sends the call to the first available PRI
endpoint.
Class of Restriction (COR)
COR identifies caller and called-party privileges for PRI endpoints. Administer the COR so that
account codes are not required. Forced entry of account codes (FEAC) is turned off for wideband
endpoints.
Class of Service (COS)
COS determines the class of features that a wideband endpoint can activate.
Facility Associated Signaling (FAS) and Non-Facility Associated Signaling (NFAS)
FAS and NFAS with or without D-Channel Backup requires administration by way of signaling
groups for trunk-side wideband interfaces.
Facility Busy Indication
You can administer a busy-indicator button for a wideband-endpoint extension, but the button
does not accurately track endpoint status.
Facility Test Calls
Use Facility Test Calls to perform loop-back testing of the wideband call facility.
Generalized Route Selection (GRS)
GRS supports wideband BCC to identify wideband calls. GRS searches a route pattern for a
preference that has wideband BCC. Route preferences that support wideband BCC also support
other BCCs for different call types to share the same trunk group.
ASAI Capabilities
For information concerning the types of associations over which various event reports can be sent,
see Communication Manager ASAI Technical Reference, 555-230-220.
Setting up ASAI
Procedure
1. Type add cti-link n, where n is a CTI link number from 1 to 64.
2. Press Enter.
The system displays the CTI Link screen.
3. In the Type field, type:
• asai-ip if the adjunct platform is not CentreVu Computer Telephony.
• adj-ip if the adjunct platform is CentreVu Computer Telephony.
4. Save the changes.
Setting up ASAI
Procedure
1. Type add cti-link n, where n is a CTI link number from 1 to 64.
2. Press Enter.
The system displays the CTI Link screen.
3. In the Type field, type:
• asai-ip if the adjunct platform is not CentreVu Computer Telephony.
• adj-ip if the adjunct platform is CentreVu Computer Telephony.
4. Save the changes.
Caution:
When migrating a platform from a legacy system to a Linux-based system of Communication
Manager 3.0 or newer, where both the old and new systems use CDR, ensure that the older
CDR parsing scripts correctly use all of the characters identified in each of the fields contained
in the applicable format table (see the Format Tables in the Avaya Aura® Communication
Manager Feature Description and Implementation, 555-245-205).
This tells Communication Manager to create records for both incoming and outgoing calls
over all trunk groups that use CDR.
13. In the Outg Trk Call Splitting and Inc Trk Call Splitting fields, type y.
This tells the system to create a separate record for any portion of an incoming or outgoing
call that is transferred or conferenced.
14. In the Outg Att Call Record and Inc Att Call Record fields, type y.
This tells the system to create a separate record for the attendant portion of any incoming
or outgoing call.
You can also administer Communication Manager to produce separate records for calls
that are conferenced or transferred. This is called Call Splitting. There are many other
variations that you can administer for CDR.
For additional information on Call Detail Recording (CDR), see Avaya Aura®
Communication Manager Feature Description and Implementation, 555-245-205.
• Periodic Pulse Metering (PPM, for non-ISDN trunks) accumulates pulses transmitted from the
public network at periodic intervals during an outgoing trunk call. At the end of the call, the
number of pulses collected is the basis for determining charges.
For more information about AOC and PPM, see Call Charge Information in Avaya Aura®
Communication Manager Feature Description and Implementation, 555-245-205.
providers supply is extended from a gateway enterprise system to the end user’s enterprise
system. The charging information can then be displayed on the user’s desktop.
Information can be extended and displayed either:
• At intervals during the call and at the end of the call, or
• Only at the end of the call
QSIG stands for Q-Signaling, which is a common channel signal protocol based on ISDN Q.931
standards and used by many digital telecommunications systems. Only charge information
received from the public network with ETSI Advice of Charge, and Japan Charge Advice is
extended into the QSIG private network.
Procedure
1. Type change trunk-group 3.
The system displays the Trunk Group screen with existing administration for this trunk
group. Click the numbered page tabs or Next Page to find fields that appear on
subsequent pages of the Trunk Group screen.
2. In the CDR Reports field, type y.
This ensures that the system displays the PPM information on the CDR report.
3. In the Direction field, enter two-way.
4. Click Next Page to find the PPM field.
5. In the PPM field, enter y.
6. In the Frequency field, enter 50/12.
This is the signal frequency (in kHz). The frequency you will use depends on what the
circuit pack you use is able to accept. See Tone Generation in Avaya Aura®
Communication Manager Screen Reference, for more information.
7. In the Administrable Timers section, set the Outgoing Glare Guard timer to 5 seconds
and select Enter to save your changes.
8. You also need to ensure that the values of the Digital Metering Pulse Minimum,
Maximum and Value on the DS1 Circuit Pack screen are appropriate to the values offered
by your service provider.
5. In the Charge Type field, enter Lira and select Enter to save your changes.
6. Enter change system-parameters features.
7. In the Charge Display Update Frequency (seconds) field, enter 30 and select Enter to
save your changes.
Frequent display updates might have considerable performance impact.
8. Now assign extension 5040 a disp-chrg button to give this user the ability to control the
charge display.
See Adding Feature Buttons for more information.
If you want end users to control when they view this information, you can assign a display
button that they can press to see the current call charges. If you want call charges to
display automatically whenever a user places an outgoing call, you can set Automatic
Charge Display to y on the COR screen.
• ssssss is populated with the least significant six digits of the System ID or SID. The SID is a
unique number in the RFA license file used to identify the system. The SID for a server can
be viewed by using one of the following methods:
- Use the statuslicense -v BASH command.
- Use the command display system-parameters customer-options on the SAT.
• cccc is populated with the least significant four digits of the Cluster ID (CL ID) or Module ID
(MID). To display the MID for the server:
- Use the statuslicense -v BASH command.
• YY is populated with the two digit number of the year the file was created.
• MM is populated with the two digit number of the month the file was created.
• DD is populated with the two digit day of the month the file was created.
• hh is populated with the hour of the day the file was created based on a 24 hour clock.
• mm is populated with the number of minutes after the hour when the file was created.
The Current CDR file uses the same naming convention except the name is prefixed with a “C-”.
Directory Application
With the Directory Application feature, you can search an LDAP database by using browsers that
are compatible with the 46xx and the 96xx deskphones. You can use the Web pages to configure
the Directory Application feature to connect to the LDAP database and to customize user search.
The Directory Application feature supports 250 instances of the directory configuration and
provides multilingual support for the instances.
For a more information about the Directory Application feature and the required configuration, see
Directory Application Job Aid on the Avaya Support website at https://support.avaya.com.
6. (Optional) Use the Search Screen Settings screen to customize the Search screen of the
deskphone.
7. (Optional) Use the Details Screen Settings screen to customize the Details screen of the
deskphone.
8. (Optional) Use the Ldap Filter Settings screen to customize the LDAP filter attributes.
General Settings
Use the General Settings page to administer the general settings and the LDAP connection
settings of the Directory Application feature.
Warning:
When you enable the Directory Application feature on the HTTP port or the unsecured
port 80, any browser can gain access to the Directory Application feature without
The directory application denies a search operation with a search string containing less
than the minimum number of characters required.
Translation Language
Use the Language Translation Settings screen for the translation language that you selected on
the General Administration screen for the selected Directory number. The system has 11
predefined translation languages. When you select a language, the system writes the language
translation to the file and the system displays the details in the translation column. If you select a
language other than the 11 predefined languages, the system displays the English text mapping
with English translations. In either of the scenarios, you can edit the English text in the translation
mapping column.
The following list includes the 11 predefined language translations:
• Brazilian-Portuguese
• English
• French
• German
• Italian
• Japanese
• Korean
• Lat-Spanish
• Russian
• Simplified Chinese
• Traditional Chinese
The user can edit the value in the Translation column of each English string.
External Numbers
Use the External Numbers Administration screen to view entries in the External Number search
root of the LDAP database, based on the specification in the General Settings screen. You can
also perform the following operations:
• Add external numbers.
• Edit external numbers.
• Delete external numbers.
For information about Communication Manager objects, see Administering Avaya Aura® System
Manager.
For information about templates, see Administering Avaya Aura® System Manager.
You can use the Inventory Management feature to configure Avaya Aura® System Manager to
discover specific devices within the network. By using the Inventory Management feature, you can
manage the SNMP access parameters that are used for the inventory collection process.
By using Inventory Management, you can do the following:
• Detect or discover your network that includes subnets and nodes.
• Discover your network by using Simple Network Management Protocol (SNMP)
For more information about the following, see the Administering Avaya Aura® System Manager and
SNMP Administration and Reference guides:
• Discovering elements
• Discovering SRS and SCS servers
• Creating profiles
• Overview of Inventory Management
Subscriber Management
With System Manager, you can perform messaging system administration activities, such as add,
view, edit, and delete subscribers. You can also administer mailboxes, and modify mailbox settings
for a messaging system.
System Manager supports:
• Communication Manager 5.0 and later and
• Avaya Aura® Messaging 6.0 and later
Adding a subscriber
Procedure
1. On the System Manager web console, click Elements > Messaging.
2. Click Subscriber in the left navigation pane.
3. Select one or more messaging systems from the list of Messaging Systems.
4. Click Show List.
5. Click New.
6. Complete the Basic Information, Subscriber Directory, Mailbox Features, Secondary
Extensions, and Miscellaneous sections.
7. Complete the Add Subscriber page and click Commit to add the subscriber.
Note:
If you select more than one Messaging or Modular Messaging from the list of
messaging systems, and then click New, the system displays the Add Subscriber page
with the first Messaging or Modular Messaging in context.
Related links
Subscribers (Avaya Aura Messaging) field descriptions on page 466
Editing a subscriber
Procedure
1. On the System Manager web console, click Elements > Messaging.
2. Click Subscriber in the left navigation pane.
3. Select a messaging system from the list of Messaging Systems.
4. Click Show List.
5. From the Subscriber List, choose the subscriber you want to edit.
6. Click Edit or View > Edit.
7. Edit the required fields in the Edit Subscriber page.
8. Click Commit to save the changes.
Related links
Subscribers (Avaya Aura Messaging) field descriptions on page 466
Viewing a subscriber
Procedure
1. On the System Manager web console, click Elements > Messaging.
2. Click Subscriber in the left navigation pane.
3. Select a messaging system from the list of Messaging Systems.
4. Click Show List.
5. Select the subscriber you want to view from the Subscriber List.
6. Click View.
Note:
You cannot edit any field on the View Subscriber page.
Related links
Subscribers (Avaya Aura Messaging) field descriptions on page 466
Deleting a subscriber
Procedure
1. On the System Manager web console, click Elements > Messaging.
2. Click Subscriber in the left navigation pane.
Subscriber list
The subscriber list displays all subscribers in a messaging version, such as Messaging,
Communication Manager Messaging, or Modular Messaging. You can apply filter to each column
in the subscriber list. You can also sort subscribers according to each of the column in the
subscriber list. You must refresh the page to view the information that is updated after the last
synchronization.
Name Description
Name The name of the subscriber.
Mailbox Number The mailbox number of the subscriber.
Email Handle The email handle of the subscriber.
Telephone Number The telephone number of the mailbox.
Last Modified The time and date when the subscriber details were
last modified.
User The name of the user to which the subscriber is
associated.
System The messaging system of the subscriber.
Filtering subscribers
Procedure
1. On the System Manager web console, click Elements > Messaging.
2. Click Subscriber in the left navigation pane.
3. Select a messaging system from the list of Messaging Systems.
Basic Information
Name Description
Class Of Service Name The name of the class of service (CoS) for this
subscriber.
CoS controls subscriber access to many features
and provides general settings, such as mailbox
size. The value that you select must be available in
the messaging system.
Community ID The default community ID for the subscriber.
Community IDs are used to control message
sending and receiving among groups of
subscribers. The default value is 1.
Numeric Address The unique address in the voice mail network. The
numeric address can be from 1 to 50 digits and can
contain the Mailbox Number.
Time zone The time zone for Avaya Aura® Messaging time
subscribers.
The value must be in the standardized name
format, America/Phoenix. Otherwise, the system
sets the Avaya Aura® Messaging subscriber time
zone to the System Manager server time zone.
PBX Extension The primary telephone extension of the subscriber.
For a Multisite system subscriber, this number is up
to 50 digits in length.
Site The name of the site. Avaya Aura® Messaging
includes a site named Default. Change the default
name when you set site properties for the first time.
Subscriber Directory
Field Description
Email Handle The name that the system displays before the
computer name and domain in the subscriber's
email address.
Telephone Number The telephone number of the subscriber as
displayed in address book listings and client
applications. The entry can be a maximum of 50
characters in length and can contain any
combination of digits (0-9), period (.), hyphen (-),
plus sign (+), and left and right parentheses ([) and
(]).
Table continues…
Field Description
Common Name The display name of the subscriber in address book
listings, such as those for email client applications.
The name can be 1 to 64 characters in length. This
field is automatically populated when you add a new
subscriber.
ASCII version of name If the subscriber name is entered in multi-byte
character format, then this field specifies the ASCII
translation of the subscriber name.
Pronounceable Name The pronounceable name of the user.
The name of a user, info mailbox, or distribution list
might not follow the pronunciation rules of the
primary language for your system. To increase the
likelihood of the Speech Recognition feature
recognizing the name, spell the name as you would
pronounce the name.
For example, if the primary language of your system
is English, spell Dan DuBois as Dan Doobwah. You
can enter an alternative name for the user. For
example, William Bell might also be known as Bill
Bell. If you enter William in the First name field, Bell
in the Last name field, and Bill Bell in the
Pronounceable name field, the speech engine
recognizes both William Bell and Bill Bell.
Include in Auto Attendant directory The option to add the messaging system to the auto
attendant directory.
Subscriber Security
Name Description
Expire Password An option to set the password expiry. The options
are:
• yes: for password to expire
• no: if you do not want your password to expire
Is Mailbox Locked? The option to lock your mailbox. A subscriber
mailbox can get locked after two unsuccessful login
attempts. The options are:
• no: To unlock your mailbox
• yes: To lock your mailbox and prevent access to it
Mailbox Features
Name Description
Personal Operator Mailbox The mailbox number or transfer dial string of the
subscriber's personal operator or assistant. This
field also indicates the transfer target when a caller
to this subscriber presses 0 while listening to the
subscriber greeting.
Personal Operator Schedule The option to specify when to route calls to the
backup operator mailbox. The default value is
Always Active.
TUI Message Order The order in which the subscriber hears the voice
messages. The options are:
• urgent first then newest: to direct the system to
play any messages marked as urgent prior to
playing non-urgent messages. Both the urgent
and non-urgent messages are played in the
reverse order of how they were received.
• oldest messages first: to direct the system to
play messages in the order they were received.
• urgent first then oldest: to direct the system to
play any messages marked as urgent prior to
playing non-urgent messages. Both the urgent
and non-urgent messages are played in the order
of how they were received.
• newest messages first: to direct the system to
play messages in the reverse order of how they
were received.
Intercom Paging The intercom paging settings for a subscriber. The
options are:
• paging is off: Disables intercom paging for this
subscriber.
• paging is manual: Callers can page the
subscriber with Subscriber Options or TUI if the
subscriber can modify.
• paging is automatic: Callers automatically page
the subscriber with TUI.
Table continues…
Name Description
VoiceMail Enabled The option to specify if a subscriber can receive
messages, email messages, and call-answer
messages from other subscribers. The options are:
• yes: To create, forward, and receive messages.
• no: To prevent the subscriber from receiving call-
answer messages and to hide the subscriber from
the telephone user interface (TUI). The subscriber
cannot use TUI to access the mailbox, and other
TUI users cannot address messages to the
subscriber.
MWI enabled The option to enable the message waiting indicator
(MWI) light feature. The options are:
• No: The user has a voice mailbox only.
• ByCOS: CoS controls how the system enables
MWI. The MWI enabled field overrides the MWI
setting defined by the CoS to which the user is
associated.
Secondary Extensions
Field Description
Secondary Extension One or more alternate number to reach a
subscriber. You can use secondary extensions to
specify a telephone number for direct reception of
faxes, to allow callers to use an existing Caller
Application, or to identify each line appearance on
the subscriber's telephone set if they have different
telephone numbers.
For Avaya Aura® Messaging 6.3, you can add a
maximum eight secondary extensions.
Miscellaneous
Field Description
Miscellaneous 1 Useful information about a subscriber template. The
messaging system does not use this information.
Entries in this field are for convenience and are not
used by the messaging system.
Miscellaneous 2 Useful information about a subscriber template. The
messaging system does not use this information.
Entries in this field are for convenience and are not
used by the messaging system.
Table continues…
Field Description
Miscellaneous 3 Useful information about a subscriber template. The
messaging system does not use this information.
Entries in this field are for convenience and are not
used by the messaging system.
Miscellaneous 4 Useful information about a subscriber template. The
messaging system does not use this information.
Entries in this field are for convenience and are not
used by the messaging system.
Button Description
Commit Saves all the changes.
Edit Allows you to edit the fields.
Reset or Clear Clears all changes.
Cancel Returns to the previous page.
7. (Optional) To customize the LDAP filter attributes, use the Ldap Filter Settings section.
IP DECT
Use the IP DECT (Digital Enhanced Cordless Telecommunications) feature to support an IP DECT
system, an IP-based cordless telephony and messaging system for connection to private
telephone exchanges.
Note:
The Far-end Network Region that has to be configured in the signaling-group must be
linked to this codec.
2. Fill in the following fields:
• Codec Set: 1 to 7 depending on the codec set to be used for the network region.
• RSVP Enabled: n.
3. Click Next until you see the Inter Network Region Connection Management section.
Avaya recommends you to use the same codec set which you already assigned, see
Assigning the codec task.
For information on administering the IP network regions, see the Administering IP network
regions section of Administering Network Connectivity on Avaya Aura® Communication
Manager, 555-233-504.
4. Select Enter to save your changes.
Note:
You must administer this trunk group to use an H.323 signaling group of x-mobility type
of DECT.
2. Ensure that the Group Type field is set to isdn.
3. Fill in the following fields:
• Direction: two-way.
• Carrier Medium: H.323.
• Service Type: tie.
4. Click Next until you see the Trunk Parameters section.
5. Fill in the following fields:
• Codeset to Send Display: 0.
Note:
Administer the Location for Routing Incoming Calls field only when the multiple
locations feature is enabled for IP DECT.
• Near-end Listen Port: Port of the CLAN or PE.
• Far-end Listen Port: Port of the ADMM or RFS.
• Far-end Network Region: Point to the associated network region.
• Calls Share IP Signaling Connection: n.
• Interworking Message: PROGress.
• Enable Layer 3 Test: y for IP trunk supervision.
4. Select Enter to save your changes.
Similarly, if Communication Manager wants to reach the blue agents (agent pool 2), it can do so
by utilizing the blue trunk between Communication Manager and Session Manager-2.
In the second scenario, administering additional trunks provide a solution for giving service to red
and blue agents, but introduces few other problems.
• The additional trunk members administered for redundancy remain unused in the first
scenario, when entity links to Session Manager-1 and Session Manager-2 are in service.
• Double the number of trunks need to be provisioned to cover a rarely occurring second
scenario. Given the limited trunk members on Communication Manager, using trunks for
redundancy reduces the trunks required for actual traffic.
• Routing and administration of route-patterns becomes complex.
For provisioning connectivity to Session Manager-1 and Session Manager-2, Communication
Manager has to create two signaling groups:
• Signaling group to Session Manager-1: Near-End as procr and Far-End as Session
Manager-1
• Signaling group to Session Manager-2: Near-End as procr and Far-End as Session
Manager-2
Each signaling must have 5000 trunks provisioned with Session Manager-1 and 5000 trunks to be
provisioned with Session Manager-2 as described earlier.
SIP trunk optimization feature allows each signaling group to point to multiple Session Managers.
In this particular case, a signaling group will point to both Session Manager-1 and Session
Manager-2. This is achieved by pointing the signaling group to a cluster of Session Managers. An
SM cluster can have as many as 28 Session Managers. With a Session Manager cluster, it is
assumed that all Session Managers share similar configuration and any Session Manager can
route a call to the far end station or far end trunk.
The ability of signaling group to point to both Session Managers reduces the required trunks by
half to be administered on Communication Manager while achieving full redundancy. If the link to
the Session Manager-1 fails, then the Signaling group uses the link to Session Manager-2 to route
all the outgoing traffic. The effect of having one signaling group pointing to multiple Session
Managers is as follows:
• Signaling group remains in service if at least one Session Manager administered in the
cluster is reachable.
• Trunk group remains in service and all members administered in the trunk group can be used
to deliver traffic.
• For example, trunk group with 5000 members in the first scenario can service 2500 agents
on Session Manager-1 and 2500 agents on Session Manager-2. The same trunk group with
5000 members can services 2500 agents on Session Manager-1 and 2500 agents on
Session Manager-2 through the link between Communication Manager and Session
Manager-2, if the connectivity between Communication Manager and Session Manager-1
goes down. Even if Session Manager-1 goes down and all agents move to Session
Manager-2, the same 5000 members will be able to reach all the 5000 agents.
Additional enhancements made in SIP trunk optimization feature are as follows:
• Number of trunk members have been increased to 9,999 for SIP trunk groups.
• Number of SIP agents have been increased to 10,000.
• System wide trunk members have been increased to 30,000.
• Measured trunks have been increased to 30,000.
• TLS connections for SIP have been increased to 56 from the current value of 32, to support
28 Session Managers. Because, two links are required to support each Session Manager.
• SIP Station form directly points to its Primary and Secondary Session Manager to support 28
Session Managers, because two links are required to support each Session Manager. For
more details on the capacities, see Avaya Aura® Communication Manager System Capacities
Table.
• Look Ahead Routing feature is deprecated for SIP station calls if routed over clustered
signaling group.
• Route pattern can now specify a network region.
Related links
Adding Session Managers to a cluster on page 482
Administering the number of members on a trunk group on page 483
Procedure
1. In a SAT session, enter change cluster session manager.
2. Enter the names of the Session Managers that you want the cluster to point to.
Related links
SIP trunk optimization on page 480
Certificate Management provides for support of importing new identity certificates and trusted CA
certificates with enhanced signatures, such as SHA2 and 2048 key length.
Certificate Management enables receiving and validating both existing certificates with SHA1-1024
signature and new certificates with SHA2-2048 signature.
Certificate installation activity is a maintenance activity and must be performed during maintenance
window when there is no call traffic running on the Communication Manager system. Performing
installation activity on live system can yield undesired system behavior like service disruption and
system overload.
Communication Manager uses four application directories to hold certificates.
Application Directory Service/Interface Peer entity Usage
C Communication Manager Session Manager SIP trunk, H.323 over
telephony another peer CM server, TLS and others
AES, CM Duplication
link, FIleSync links,
H.248 gateways and
96x1 H.323 phone.
W Administration Web PC Communication Manager
Server Web Administration
R Remote logging Syslog server and logging and services
general Services access access
A Authentication, External AAA server Administration accounts
Authorization and authentication
Accounting (AAA)
services (for example,
LDAP)
Related links
Generating a CSR on page 491
Generating a CSR when third-party signed certificate is unavailable on page 492
Identity Certificates
Overview
Each Communication Manager is installed with a unique identity certificate for Communication
Manager Telephony service, including the SIP TLS link. This certificate is unique for each
customer, and is secure. Three methods exist for certificate creation/signing:
1. Import a 3rd party hosted certificate pair (Trusted CA chain and the Identity cert)
2. Create a signed cert on SMGR and import to CM
3. Use CM’s Certificate Signing Request (CSR) and point to SMGR’s CA or to some other CA
for signing.
You should generate 2048 bits and SHA-2 hash identity certificates with your CA for all
Communication Manager services/interfaces, including Telephony service.
The Communication Manager Messaging service has its non-unique identity certificate that is
signed by Avaya SIP Product Certificate Authority, and automatically installed during installation.
Communication Manager Web Server is the same.
Each service or interface can only have ONE identity certificate, but one identity certificate may be
copied into multiple repositories.
Displaying a certificate
Procedure
1. On the Communication Manager SMI interface, click Administration > Server
(Maintenance).
2. In the Security section, click Server/Application Certificates.
3. Select a certificate from the list that has been, and click Display.
8. Click Add.
Communication Manager does not needs to be restarted for the certificate to take effect.
Trust certificates
Displaying a certificate
Procedure
1. On the Communication Manager SMI interface, click Administration > Server
(Maintenance).
2. In the Security section, click Trusted Certificates.
3. Select a certificate, and click Display.
Related links
Certificate Management on page 484
Removing a certificate
Procedure
1. On the Communication Manager SMI interface, click Administration > Server
(Maintenance).
2. In the Security section, click Trusted Certificates.
3. Select a certificate entry and click Remove.
4. Select the appropriate repositories from which you want to remove the certificate.
5. Click Remove.
Related links
Certificate Management on page 484
Copying a certificate
Procedure
1. On the Communication Manager SMI interface, click Administration > Server
(Maintenance).
2. In the Security section, click Trusted Certificates.
3. Select a certificate entry and click Copy.
4. Select the appropriate repositories in which you want to install the selected certificate.
5. Click Copy to install the selected certificate in the selected repositories.
For each repository where the certificate is installed, the system overwrites or creates the
server.crt and server.key files.
Related links
Certificate Management on page 484
Generating a CSR
Procedure
1. On the Communication Manager SMI interface, click Administration > Server
(Maintenance).
2. In the Security section, click Certificate Signing Request.
3. Enter the appropriate details in the fields.
4. Click Generate.
Related links
Certificate Management on page 484
Certificate Signing Request field descriptions on page 492
Related links
Generating a CSR on page 491
Note:
If you are using a duplex server, you must first install the certificate on a standby server and
perform a reset system 4. Later, you must interchange the server and apply the certificate on
the new standby server and perform a reset system 4. You must install the certificate on
Enterprise Survivable Server (ESS) and Local Survivable Processor (LSP) separately.
Procedure
1. On the Communication Manager SMI, click Administration > Server (Maintenance).
2. In the Security section, click Certificate Signing Request.
3. Enter the appropriate details in the fields.
For more information, see the “Certificate Signing Request field descriptions” section.
4. Click Generate Request.
5. Copy the content between “BEGIN CERTIFICATE REQUEST” and “END CERTIFICATE
REQUEST”, and click Continue.
6. Log on to System Manager web interface, and navigate to Security > Certificates >
Authority.
7. Click RA Functions > Add End Entity, and enter the required details.
For more information, see the “Creating an end entitiy” section in Administering Avaya
Aura® System Manager.
8. In the left navigation pane, click Public Web.
9. In the new window, click Create Certificate from CSR.
10. In the Certificate enrollment from a CSR page, enter the user name and password that
you created in Step 7.
11. Paste the CSR content that you copied in Step 5.
12. To generate and download the certificate on to your local machine, click OK.
13. Go to Communication Manager SMI, and download the certificate from your local machine.
14. In the Security section, click Server/Application Certificates.
15. Click Add, and copy the certificate which was downloaded.
16. Click Open, and check Communication Manager.
17. Click Add.
18. For the certificate to take effect, perform reset system 4 on Communication Manager.
Related links
Certificate Management on page 484
Navigate to the Manage Content > My Docs menu, and do any of the following:
- Create, rename, and delete a collection.
- Add topics from various documents to a collection.
- Save a PDF of selected content in a collection and download it to your computer.
- Share content in a collection with others through email.
- Receive collection that others have shared with you.
• Add yourself as a watcher using the Watch icon ( ).
Navigate to the Manage Content > Watchlist menu, and do the following:
- Enable Include in email notification to receive email alerts.
- Unwatch selected content, all content in a document, or all content on the Watch list page.
As a watcher, you are notified when content is updated or deleted from a document, or the
document is removed from the website.
• Share a section on social media platforms, such as Facebook, LinkedIn, and Twitter.
• Send feedback on a section and rate the content.
Note:
Some functionality is only available when you log on to the website. The available functionality
depends on the role with which you are logged in.
Training
The following courses are available on the Avaya Learning website at www.avaya-learning.com.
After logging into the website, enter the course code or the course title in the Search field and
click Go to search for the course.
Course code Course title
20970W Introducing Avaya Device Adapter
20980W What's New with Avaya Aura® Release 8.1
71200V Integrating Avaya Aura® Core Components
72200V Supporting Avaya Aura® Core Components
20130V Administering Avaya Aura® System Manager Release 8.1
21450V Administering Avaya Aura® Communication Manager Release 8.1
Support
Go to the Avaya Support website at https://support.avaya.com for the most up-to-date
documentation, product notices, and knowledge articles. You can also search for release notes,
downloads, and resolutions to issues. Use the online service request system to create a service
request. Chat with live agents to get answers to questions, or request an agent to connect you to a
support team if an issue requires additional expertise.
Communication Manager server separation ........................39 data line data module ........................................................ 410
configure 46xx and 96xx telephones using the WMLHOME data modules
property ..............................................................................473 7400A ..........................................................................411
configuring ........................................................................... 32 7400B+ ....................................................................... 411
log retention period .....................................................325 7400C High Speed Link ..............................................412
syslog server .............................................................. 324 7400D ......................................................................... 411
configuring Administrable Alternate Gatekeeper Lists .........82 7500 ....................................................................410, 412
configuring a DS1 circuit pack example .............................353 8400B ..........................................................................411
configuring telecommuting example .................................. 294 announcement ............................................................410
Configuring the IP synchronization for the network region .. 32 asynchronous ............................................................. 412
Configuring the synchronization reference for the BRI trunk BRI ..............................................................................410
board ....................................................................................31 data line ...................................................................... 410
Configuring the synchronization reference for the gateway .30 data-terminal dialing ................................................... 400
configuring the telephones .................................................453 DCP
Configuring your system .................................................... 180 telephone dialing ................................................. 401
considerations for Alphanumeric Dialing ........................... 404 detailed description ..................................................... 411
considerations for ASAI ..................................................... 435 Ethernet ...................................................................... 410
Considerations for Data Call Setup ................................... 403 ISDN-BRI ....................................................................402
Considerations for Data Privacy ........................................ 406 PPP ............................................................................ 410
considerations for Modem Pooling .................................... 419 processor/trunk ...........................................................410
Considerations for Personal Computer Interface ...............422 types ........................................................................... 410
Console Parameters data-only off-premises extensions ..................................... 409
setting ......................................................................... 194 Data-Only Off-Premises Extensions .................................. 409
content Data-Only Off-Premises Extensions administering ............409
publishing PDF output ................................................ 497 data privacy, administering ................................................ 405
searching .................................................................... 497 data privacy administration ................................................ 405
sharing ........................................................................497 Data Privacy considerations .............................................. 406
sort by last updated .................................................... 497 Data Privacy interactions ................................................... 406
watching for updates .................................................. 497 data restriction ................................................................... 407
Controlling Calls Users Can Make and Receive ..................57 data restriction, administering ............................................408
CO trunks ...........................................................................343 Data Restriction interactions ..............................................408
Coverage of Calls Redirected Off Net (CCRON) ............... 292 data terminal (keyboard) dialing
coverage option changing ..................................................307 alphanumeric .............................................................. 404
coverage options, assigning .............................................. 300 default dialing ............................................................. 407
Creating a New Time of Day Routing Plan ........................ 288 ISDN-BRI data modules ............................................. 402
creating a Station Security Code example .........................297 daylight saving rules ...................................................... 28, 29
CTI link .......................................................................435, 436 Daylight Saving Rules ..........................................................29
DCP and ISDN-BRI module call-progress messages ........399
DCP data modules .............................................................400
D Deactivate Night Service ................................................... 233
data calls Deactivation ....................................................................... 182
characters used .......................................................... 399 default dialing .....................................................................407
overview ..................................................................... 397 Defining options for calling party identification ...................181
setup ...........................................................................397 delete
Data Call Setup Administration ..........................................398 subscribers ................................................................. 464
Data Call Setup for data-terminal dialing ........................... 398 Deleting messages .............................................................. 34
Data Call Setup for telephone dialing ................................ 398 Dell™ PowerEdge ................................................................ 62
Data Call Setup interactions .............................................. 403 detailed description of Wideband Switching ...................... 423
Data Call Setup port assignments ..................................... 398 Detail Screen settings ........................................................456
data connection types ........................................................397 dialing
data encryption .................................................................. 335 alphanumeric .............................................................. 404
overview ..................................................................... 333 default .........................................................................407
password policy .......................................................... 335 DID trunks ..........................................................................345
remote key server .......................................................334 digital trunks .......................................................................351
DataHotline ........................................................................ 404 digits
Data Hotline administering .................................................405 absorbing ....................................................................359
Data Hotline interactions ....................................................405 inserting .............................................................. 358, 359
Extension to Cellular Setup Table ......................................173 Hayes command set .......................................................... 411
External Numbers home equipment, installing ................................................ 301
adding external number ..............................................457 Hunt Groups
deleting external number ............................................ 458 adding announcements .............................................. 255
editing external number .............................................. 458 changing group ...........................................................253
dynamic hunt group .................................................... 253
setting ......................................................................... 252
F setting queue .............................................................. 254
fax TTY callers ................................................................. 254
enabling transmission over IP networks ....................... 88
Fax I
Adding ........................................................................ 125
Enabling transmission over IP networks .....................126 ICLID Information
feature buttons table .......................................................... 148 Displaying ................................................................... 199
filtering subscribers Identify Certificates ............................................................ 485
using filters; subscribers ............................................. 465 identity certificate
finding content on documentation center ...........................497 adding for duplex server ............................................. 487
finding port matrix .............................................................. 497 adding for simplex server ........................................... 486
Fixing Problems in Terminal Self-Administration ............... 178 copying ....................................................................... 488
Flexible Extended Pickup Group removing .....................................................................488
assigning pickup groups ............................................. 247 Improved port network recovery from control network outages
Flexible Extended Pickup Groups ......................................245 ..............................................................................................84
following a process when working with trunk groups .........341 improved survivability administration ................................... 86
Forced Entry of Account Codes administering .................. 441 Incoming Calls
fraud ACD
system security ...........................................................312 automatic call distribution .................................... 266
FX trunk group ................................................................... 344 advanced call coverage .............................................. 211
FX trunks ........................................................................... 343 assigning terminating extension group ....................... 266
basic call coverage ..................................................... 208
call forwarding ............................................................ 214
G call pickup ...................................................................234
Gateway serviceability commands ...................................... 91 hunt groups .................................................................252
Gateway Virtual Val ............................................................381 night service ............................................................... 226
General Administration ...................................................... 453 Vectors
General Settings VDNs ................................................................... 256
Detail Screen settings .................................................456 Increasing Text Fields for Feature Buttons ........................ 146
General Administration ............................................... 453 InSite Knowledge Base ......................................................499
LDAP Administration .................................................. 454 Installing
LDAP Filter Settings ................................................... 456 phone message files ...................................................201
Search Screen settings .............................................. 455 installing home equipment example .................................. 302
Generating a CSR interactions for Administered Connections ........................ 417
CSR ............................................................................ 491 interactions for ASAI .......................................................... 435
when signed certificate is unavailable ........................ 492 interactions for Call Forwarding ......................................... 299
glare, prevention ................................................................ 431 interactions for Data Call Setup ......................................... 403
group communications .......................................................384 interactions for Data Hotline .............................................. 405
automatic answer intercom calls ................................ 393 interactions for Data-Only Off-Premises Extensions ..........409
chime paging over loudspeakers ................................386 interactions for Data Privacy ..............................................406
paging over speakerphones ....................................... 389 interactions for Data Restriction .........................................408
voice paging over loudspeakers ................................. 384 intercom
whisper paging ........................................................... 390 automatic answer calls ............................................... 393
using telephone as ..................................................... 391
intercom feature buttons .................................................... 392
H intercom group example .................................................... 393
Interconnect and Group Type entries for enhanced DS1
H0 channels .......................................................................430
administration .................................................................... 355
H11 channels ..................................................................... 429
Inter-exchange carrier calls ............................................... 275
hardware requirements ISDN trunk groups ....................... 362
setting Up Authorization Codes example ...........................321 Station Lock by time of day ..........................................60, 328
setting up Call Forwarding for telecommuting example .....298 Station Lock set up preparation ......................................... 327
setting up Chime Paging Over Loudspeakers example .....387 Station Lock with a Station Lock button— setting up .........327
Setting up emergency calls on IP telephones ....................134 Station Lock without a Station Lock button-setting up ....... 328
Setting Up Extension To Cellular Feature Access Button ..175 Stations ..............................................................................180
setting up intra-switch CDR example .................................439 station security code
Setting up IP synchronization .............................................. 32 creating .......................................................................296
setting up Personal Station Access example .....................295 Station Security Code example ......................................... 297
setting up Personal Station Access preparation ................ 295 Strategies for assigning CORs ............................................ 58
Setting up Remote Office on network regions ................... 138 subscriber; view
setting up Security Violations Notification example ........... 323 viewing subscribers .................................................... 464
setting up Service Observing .............................................395 subscriber list .....................................................................465
setting up speakerphone paging example ......................... 389 subscribers
setting up Station Lock with a Station Lock button example delete ..........................................................................464
............................................................................................327 deleting ....................................................................... 464
setting up Station Lock without a Station Lock button example removing .....................................................................464
............................................................................................328 subscribers; add
Setting Up Terminal Self-Administration ............................ 177 adding subscribers ..................................................... 463
setting up the DS1 board as a sync Source reference ...... 353 subscribers; new .........................................................463
Setting Up Voice Paging Over Loudspeakers example ..... 384 subscribers; edit
setup Authorization Codes .................................................321 editing a subscriber .................................................... 464
sharing content .................................................................. 497 editing subscribers ......................................................464
signing up support ...............................................................................499
PCNs and PSNs ......................................................... 502 supported browsers ............................................................. 65
Simple extended pickup groups .........................................242 survivable CMS ..................................................................105
creating .......................................................................243 Survivable Core Servers administration for PE ................. 105
Simple Network Management Protocol, see SNMP ............ 68 survivable remote server ..................................................... 62
SIP trunk optimization ........................................................480 Survivable Remote Servers administration for PE .............105
smi pe server configuration ..................................................97 SVN ........................................................................... 319, 323
SNMP swapping ............................................................................ 118
administering ................................................................ 68 switching, wideband ...........................................................423
administration ............................................................... 93 syslog server
sort documents by last updated .........................................497 sending logs to external serverr ................................. 324
Source-based Routing ....................................................... 361 system ................................................................................. 24
speakerphone, paging over ............................................... 389 System Access Terminal (SAT) ............................................86
speakerphone paging capacities ....................................... 390 system administration
Speakerphone paging troubleshooting .............................. 390 security ....................................................................... 310
Speed dialing ..................................................................... 167 system login .........................................................................24
split ...................................................................................... 68 system-parameters Customer-Options (Optional Features)
split registration ....................................................................68 screen .................................................................................. 89
split registration prevention ..................................................69 System Requirements ....................................................... 179
split registration prevention activation ..................................69 system security, see security ............................................. 310
split registration prevention feature ......................................72 system security checking ...................................................315
split registration prevention solution prerequisites and
constraints ........................................................................... 73
starting
T
SAT session ..................................................................65 T1 .......................................................................................354
station .................................................................................114 T1, recommended settings for digital trunks ......................354
Station Telecommuter mode
access a new group list .............................................. 167 Adding ........................................................................ 130
station lock telecommuting
lock ............................................................................. 326 Answer Supervision ....................................................292
Station Lock ................................................................. 59, 331 assigning coverage options ........................................ 300
hot desking enhancement .......................................... 330 associating office phone number to home station ...... 302
hot desking with station lock restrictions .................... 331 changing settings ........................................................306
interaction with PSA ................................................... 331 configuring Communication Manager for ....................292
Station Lock administering screens ................................... 329 Coverage of Calls Redirected Off Net (CCRON) ........292