Administering Avaya IP Office Platform with Manager

Administering Avaya IP Office™ Platform
with Manager
Release 9.1.2
Issue 10.38
February 2016
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Contents
Chapter 1: Introduction.......................................................................................................... 25
Purpose................................................................................................................................ 25
Intended audience................................................................................................................. 25
Documentation resources...................................................................................................... 25
Support................................................................................................................................ 26
Document changes since last issue........................................................................................ 26
What's New in Release 9.1.................................................................................................... 29
Chapter 2: Overview............................................................................................................... 34
Manager Modes.................................................................................................................... 34
Security Configuration Mode............................................................................................ 35
Standard Mode Configuration Mode.................................................................................. 36
Server Edition Configuration Mode.................................................................................... 38
Shell Server Mode........................................................................................................... 39
Backward Compatibility.................................................................................................... 40
Chapter 3: Getting Started..................................................................................................... 41
PC requirements................................................................................................................... 41
Installing Manager................................................................................................................. 42
Starting Manager................................................................................................................... 43
Opening a Configuration........................................................................................................ 44
Login messages.............................................................................................................. 45
Changing the Manager Language........................................................................................... 46
Chapter 4: Menu Bar Commands.......................................................................................... 48
File Menu............................................................................................................................. 48
File > Open Configuration.............................................................................................. 49
File > Close Configuration............................................................................................. 49
File > Save Configuration.............................................................................................. 49
File > Save Configuration As......................................................................................... 49
File > Change Working Directory................................................................................... 50
File | Preferences............................................................................................................ 51
File | Offline.................................................................................................................... 59
File | Advanced............................................................................................................... 60
File | Backup/Restore...................................................................................................... 77
File | Import/Export.......................................................................................................... 77
File | Exit........................................................................................................................ 78
View Menu............................................................................................................................ 78
Tools Menu........................................................................................................................... 79
Tools | Extension Renumber............................................................................................ 80
Tools | Line Renumber..................................................................................................... 80
Tools | Connect To.......................................................................................................... 80
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Tools | Export | User........................................................................................................ 81
Tools | SCN Service User Management............................................................................ 81
Tools | Busy on Held Validation........................................................................................ 82
Tools | MSN Configuration............................................................................................... 82
Tools | Print Button Labels............................................................................................... 83
Tools | Import Templates.................................................................................................. 84
Security Mode Menus............................................................................................................ 84
Embedded File Management Menus....................................................................................... 85
Chapter 5: Manager User Interface....................................................................................... 88
Title Bar................................................................................................................................ 88
Toolbars............................................................................................................................... 88
The Main Toolbar............................................................................................................ 89
The Navigation Toolbar.................................................................................................... 90
The Details Toolbar......................................................................................................... 90
The Navigation Pane............................................................................................................. 90
Expanding and Collapsing the Navigation Tree.................................................................. 91
The Group Pane.................................................................................................................... 91
Sorting the List................................................................................................................ 91
Customizing the Columns Displayed................................................................................. 92
Changing the Column Widths........................................................................................... 92
Adding a New Record...................................................................................................... 92
Deleting an Record.......................................................................................................... 93
Validating an Record....................................................................................................... 93
Show in Groups.............................................................................................................. 93
The Details Pane................................................................................................................... 93
Managing Records.......................................................................................................... 94
The Error Pane..................................................................................................................... 95
Altering the Automatic Validation Settings......................................................................... 96
Revalidating Configuration Settings.................................................................................. 96
Viewing an Error or Warning............................................................................................. 96
The Status Bar...................................................................................................................... 97
Configuring the Interface........................................................................................................ 97
Resizing the Manager Window......................................................................................... 98
Moving the Border Between the Panes............................................................................. 98
Showing or Hiding Toolbars............................................................................................. 98
Moving Toolbars.............................................................................................................. 99
Showing or Hiding Panes................................................................................................. 99
Changing the Position of the Details Pane......................................................................... 99
Changing the Size of Configuration Icons........................................................................ 100
Changing Tab Display................................................................................................... 100
Chapter 6: Working with the Server Edition Manager User Interface.............................. 101
Server Edition Solution View................................................................................................ 101
System Inventories.............................................................................................................. 103
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Default Settings................................................................................................................... 103
Record Consolidation.......................................................................................................... 104
Configuring Telephony Operation......................................................................................... 105
Incoming Call Routing.................................................................................................... 106
Outgoing Call Routing.................................................................................................... 111
Telephone Features Supported Across Server Edition and SCN Networks............................... 132
Chapter 7: Security Administration..................................................................................... 134
Security Users.................................................................................................................... 134
Default Service Users and Rights Groups........................................................................ 135
Access Control.................................................................................................................... 140
Encryption.......................................................................................................................... 140
Message Authentication....................................................................................................... 141
Certificates......................................................................................................................... 142
Implementing Security......................................................................................................... 147
SRTP................................................................................................................................. 149
Chapter 8: Editing IP Office Security Settings in Manager............................................... 152
Loading Security Settings..................................................................................................... 152
Saving Security Settings...................................................................................................... 153
Resetting a System's Security Settings................................................................................. 153
Chapter 9: Security Mode Field Descriptions.................................................................... 155
General Security Field Descriptions...................................................................................... 156
System............................................................................................................................... 160
System Details.............................................................................................................. 160
Unsecured Interfaces..................................................................................................... 162
Certificates................................................................................................................... 163
Security Services Settings.................................................................................................... 168
Rights Groups..................................................................................................................... 169
Group Details................................................................................................................ 170
Configuration................................................................................................................ 170
Security Administration.................................................................................................. 171
System Status............................................................................................................... 172
Enhanced TSPI............................................................................................................. 172
HTTP........................................................................................................................... 173
Web Services................................................................................................................ 173
External........................................................................................................................ 174
Service Users..................................................................................................................... 174
Chapter 10: Editing Configuration Settings....................................................................... 177
Mergeable Settings.............................................................................................................. 179
Configuration Size............................................................................................................... 188
Setting the Discovery Addresses.......................................................................................... 190
Opening a Configuration from a System................................................................................ 191
Opening a Configuration Stored on PC................................................................................. 193
Known System Discovery..................................................................................................... 194
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Contents
Configuring Manager for Known System Discovery.......................................................... 194
Using Known System Discovery..................................................................................... 195
Creating New Records......................................................................................................... 196
Adding a New Record Using the Details Pane................................................................. 196
Adding a New Record Using the Group Pane.................................................................. 196
Adding a New Record Using the Navigation Pane............................................................ 197
Other record creation methods....................................................................................... 197
Creating an Offline Configuration.......................................................................................... 198
Importing and Exporting Settings.......................................................................................... 199
Exporting Settings......................................................................................................... 201
Importing Settings......................................................................................................... 201
Copying and Pasting........................................................................................................... 202
Saving a Configuration onto PC............................................................................................ 202
Sending a Configuration....................................................................................................... 203
Erasing the Configuration..................................................................................................... 205
Default Settings................................................................................................................... 205
Chapter 11: Configuration Mode Field Descriptions......................................................... 208
Configuration field display.................................................................................................... 209
Configuration field display in Standard mode................................................................... 209
Configuration field display in Server Edition mode............................................................ 210
BOOTP | BOOTP Record..................................................................................................... 212
System .............................................................................................................................. 214
System | System........................................................................................................... 215
System | LAN1.............................................................................................................. 220
System | LAN2.............................................................................................................. 232
System | DNS............................................................................................................... 232
System | Voicemail........................................................................................................ 233
System | Telephony....................................................................................................... 241
System | Directory Services........................................................................................... 266
System | System Events................................................................................................ 271
System | SMTP............................................................................................................. 278
System | SMDR............................................................................................................. 279
System | Twinning......................................................................................................... 280
System | VCM............................................................................................................... 280
System | CCR............................................................................................................... 283
System | Codecs........................................................................................................... 284
System | VoIP Security.................................................................................................. 285
Dialer........................................................................................................................... 286
System | Contact Center................................................................................................ 289
Line.................................................................................................................................... 289
Analog Line................................................................................................................... 291
BRI Line....................................................................................................................... 299
PRI Trunks................................................................................................................... 304
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S0 Line......................................................................................................................... 330
H.323 Line.................................................................................................................... 333
IP DECT Line................................................................................................................ 345
SIP Line....................................................................................................................... 353
Line | SIP DECT Line..................................................................................................... 380
Line | SM Line............................................................................................................... 382
Line | IP Office Line....................................................................................................... 389
Control Unit | Control Unit.................................................................................................... 396
Extension........................................................................................................................... 397
Extension | Extn............................................................................................................ 398
Extension | Analog........................................................................................................ 401
Extension | VoIP............................................................................................................ 403
T38 Fax........................................................................................................................ 415
Extension | IP DECT...................................................................................................... 416
Extension | SIP DECT Base........................................................................................... 417
User................................................................................................................................... 418
User | User................................................................................................................... 419
User | Voicemail............................................................................................................ 425
User | DND................................................................................................................... 429
User | Short Codes........................................................................................................ 430
Source Numbers........................................................................................................... 431
User | Telephony........................................................................................................... 435
User | Forwarding.......................................................................................................... 444
User | Dial In................................................................................................................. 447
User | Voice Recording.................................................................................................. 447
User | Button Programming............................................................................................ 449
User | Menu Programming............................................................................................. 450
User | Mobility............................................................................................................... 452
User | Hunt Group Memberships..................................................................................... 455
User | Announcements.................................................................................................. 455
User | SIP..................................................................................................................... 457
User | Personal Directory............................................................................................... 457
User | Web Self Administration....................................................................................... 459
Group................................................................................................................................. 460
Group........................................................................................................................... 463
Group | Queuing............................................................................................................ 468
Overflow....................................................................................................................... 471
Group | Fallback............................................................................................................ 473
Group | Voicemail.......................................................................................................... 476
Group | Voice Recording................................................................................................ 480
Announcements............................................................................................................ 481
Hunt Group | SIP........................................................................................................... 484
Short Code......................................................................................................................... 484
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Contents
Service............................................................................................................................... 485
Normal, WAN or Intranet Services.................................................................................. 486
SSL VPN Service.......................................................................................................... 494
RAS | RAS.......................................................................................................................... 498
RAS | PPP.................................................................................................................... 498
Incoming Call Route............................................................................................................ 500
Incoming Call Route | Standard...................................................................................... 502
Incoming Call Route | Voice Recording........................................................................... 506
Incoming Call Route | Destinations................................................................................. 507
WAN Port........................................................................................................................... 508
WAN Port | WAN Port.................................................................................................... 509
WAN Port | Frame Relay................................................................................................ 509
WAN Port | DLCIs......................................................................................................... 510
WAN Port | Advanced.................................................................................................... 512
Directory............................................................................................................................. 513
Directory | Directory Entry.............................................................................................. 514
Time Profile........................................................................................................................ 516
Firewall Profile.................................................................................................................... 517
Firewall | Standard......................................................................................................... 518
Firewall | Custom........................................................................................................... 519
Static NAT.................................................................................................................... 521
IP Route............................................................................................................................. 522
IP Route | IP Route........................................................................................................ 523
RIP Dynamic Routing.................................................................................................... 524
Account Code..................................................................................................................... 524
Account Code | Account Code........................................................................................ 525
Account Code | Voice Recording.................................................................................... 525
License............................................................................................................................... 527
License | License........................................................................................................... 528
License | Remote Server................................................................................................ 530
Tunnel................................................................................................................................ 531
L2TP Tunnel................................................................................................................. 531
IP Security Tunnel......................................................................................................... 534
Auto Attendant.................................................................................................................... 536
Auto Attendant | Auto Attendant...................................................................................... 538
Actions......................................................................................................................... 539
Authorization Codes............................................................................................................ 541
Authorization Codes | Authorization Codes (9.0).............................................................. 542
User Rights......................................................................................................................... 542
User Rights | User......................................................................................................... 543
User Rights | Short Codes.............................................................................................. 543
User Rights | Button Programming.................................................................................. 544
User Rights | Telephony................................................................................................. 544
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User Rights | User Rights Membership............................................................................ 548
User Rights | Voicemail.................................................................................................. 548
User Rights | Forwarding................................................................................................ 550
ARS................................................................................................................................... 550
ARS............................................................................................................................. 551
RAS Location Request......................................................................................................... 555
Location............................................................................................................................. 556
Location (9.0)................................................................................................................ 558
Chapter 12: Configure general system settings................................................................ 560
On-boarding........................................................................................................................ 560
Configuring an SSL VPN using an on-boarding file........................................................... 560
System Date and Time........................................................................................................ 561
Time Profile........................................................................................................................ 563
Overriding a Time Profile............................................................................................... 564
Working with Templates....................................................................................................... 565
Importing Trunk Templates............................................................................................. 566
Creating a Trunk Template............................................................................................. 567
Creating a New SIP Trunk from a Template..................................................................... 567
Applying a Template to an Analog Trunk......................................................................... 568
Creating Server Edition Templates.................................................................................. 568
Creating a New Server Edition Record from a Template................................................... 569
Centralized System Directory............................................................................................... 570
Advice of Charge................................................................................................................. 573
Emergency Call................................................................................................................... 574
Fax Relay........................................................................................................................... 575
Caller Display...................................................................................................................... 577
Parking Calls....................................................................................................................... 578
Configuring Call Admission Control....................................................................................... 579
Manager location tab..................................................................................................... 580
Assigning a network entity to a location...................................................................... 580
System actions at maximum call threshold...................................................................... 581
Example....................................................................................................................... 581
Ring Tones......................................................................................................................... 583
Media Connection Preservation............................................................................................ 584
Music On Hold.................................................................................................................... 585
System Source.............................................................................................................. 587
Alternate Source........................................................................................................... 588
Conferencing...................................................................................................................... 591
Conference Phones....................................................................................................... 592
Ad-Hoc Conferencing.................................................................................................... 593
Meet Me Conferencing................................................................................................... 594
Routing External Callers................................................................................................ 596
Context Sensitive Conferencing...................................................................................... 597
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Contents
Paging................................................................................................................................ 599
Paging Via Voicemail Pro............................................................................................... 601
Automatic Intercom Calls..................................................................................................... 603
Wide Band Audio Support.................................................................................................... 604
Configuring Remote H.323 Extensions.................................................................................. 605
System Configuration.................................................................................................... 606
Phone Configuration...................................................................................................... 608
Creating a Virtual WAN Port................................................................................................. 608
Configuring authorization codes............................................................................................ 609
Entering an Authorization Code...................................................................................... 610
Configuring ARS................................................................................................................. 611
Example ARS Operation................................................................................................ 612
ARS Operation.............................................................................................................. 613
System Events.................................................................................................................... 623
Configuring Alarm Destinations...................................................................................... 624
Preventing Toll Bypass........................................................................................................ 624
Configuring unknown locations....................................................................................... 625
Call Barring......................................................................................................................... 625
Applying Call Barring..................................................................................................... 625
Overriding call barring.................................................................................................... 626
Chapter 13: Configure Server Edition system settings..................................................... 628
Opening the System Configurations...................................................................................... 628
Configuring the Systems...................................................................................................... 630
Saving Configuration Changes............................................................................................. 630
Starting System Status........................................................................................................ 632
Voicemail Administration...................................................................................................... 632
Configuring Resiliency......................................................................................................... 633
Setting Up Resilience.................................................................................................... 634
Configuring Location Based Extension Resiliency.................................................................. 635
Adding a Secondary Server.................................................................................................. 635
Adding a Secondary Server............................................................................................ 636
Adding an Expansion System............................................................................................... 636
Adding an Expansion System......................................................................................... 637
Displaying the System Inventories........................................................................................ 638
Removing an Expansion/Secondary Server........................................................................... 638
Synchronizing the Configurations.......................................................................................... 639
Displaying the Solution View................................................................................................ 639
Starting Web Control........................................................................................................... 639
Shared Administration......................................................................................................... 640
Chapter 14: Configure user settings................................................................................... 642
Configuring User Rights....................................................................................................... 642
Adding User Rights........................................................................................................ 644
Creating a User Right Based on an Existing User............................................................ 644
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Contents
Associating User Rights to a User................................................................................... 645
Copy User Rights Settings over a User's Settings............................................................ 645
Account Code Configuration................................................................................................. 645
Setting a User to Forced Account Code........................................................................... 646
Mobile Call Control.............................................................................................................. 647
Mobile Direct Access (MDA)........................................................................................... 650
Mobile Callback............................................................................................................. 651
Twinning............................................................................................................................. 652
Centralized Personal Directory............................................................................................. 655
Centralized Call Log............................................................................................................ 656
Coverage Groups................................................................................................................ 660
Hunt Group Operation......................................................................................................... 660
Hunt Group Types......................................................................................................... 661
Call Presentation........................................................................................................... 662
Hunt Group Member Availability..................................................................................... 663
Example Hunt Group..................................................................................................... 664
CBC/CCC Agents and Hunt Groups................................................................................ 667
Malicious Call Tracing (MCID).............................................................................................. 667
Call Restriction.................................................................................................................... 668
Call Intrusion....................................................................................................................... 669
Private Calls....................................................................................................................... 673
Call Waiting........................................................................................................................ 674
Message Waiting Indication.................................................................................................. 675
Message Waiting Indication for Analog Phones................................................................ 675
Message Waiting Indication for Analog Trunks................................................................. 676
System Phone Features....................................................................................................... 677
The 'No User' User.............................................................................................................. 679
Suppressing the NoCallerId alarm.................................................................................. 679
DND, Follow Me and Forwarding.......................................................................................... 680
Do Not Disturb (DND).................................................................................................... 682
Follow Me..................................................................................................................... 683
Forward Unconditional................................................................................................... 685
Forward on Busy........................................................................................................... 687
Forward on No Answer.................................................................................................. 689
Determining a User's Busy Status................................................................................... 691
Chaining....................................................................................................................... 692
Transferring Calls................................................................................................................ 693
Off-Switch Transfer Restrictions..................................................................................... 694
Context Sensitive Transfer............................................................................................. 695
Dial Tone Transfer......................................................................................................... 696
Handsfree Announced Transfers.................................................................................... 698
One Touch Transferring................................................................................................. 699
Centrex Transfer........................................................................................................... 700
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Contents
Hot Desking........................................................................................................................ 701
Remote Hot Desking...................................................................................................... 703
Call Center Agents........................................................................................................ 704
Hot Desking Examples................................................................................................... 704
Automatic Log Out......................................................................................................... 706
Chapter 15: Configuring the Avaya Session Border Controller for IP Office Remote
Workers.................................................................................................................................. 708
Overview............................................................................................................................ 708
Remote access............................................................................................................. 708
Licencing...................................................................................................................... 709
Remote Worker best practices........................................................................................ 709
Provisioning SIP Phones................................................................................................ 710
Configuring Session Border Controller Enterprise for IP Office Remote Workers...................... 711
Network interfaces......................................................................................................... 712
Creating a backup......................................................................................................... 713
Configuring network address translation.......................................................................... 713
Enabling interfaces........................................................................................................ 714
Configuring media interfaces.......................................................................................... 714
Configuring signalling interfaces..................................................................................... 715
Configuring server interworking profiles........................................................................... 716
Configuring phone interworking profiles........................................................................... 716
Configuring the call server.............................................................................................. 717
Configuring routing profiles............................................................................................. 717
Configuring topology hiding............................................................................................ 718
Configuring endpoint policy groups................................................................................. 719
Configuring endpoint policy groups application rules........................................................ 719
Configuring endpoint policy groups media rules............................................................... 720
Configuring endpoint policy groups signalling rules.......................................................... 720
Configuring server flows................................................................................................. 721
Configuring user agent profiles....................................................................................... 722
Configuring subscriber flows........................................................................................... 723
Chapter 16: Configuring SIP Trunks................................................................................... 725
Overview............................................................................................................................ 725
Configuring a SIP Trunk....................................................................................................... 726
SIP Line Requirements........................................................................................................ 727
SIP Incoming Call Routing................................................................................................... 729
SIP Prefix Operation............................................................................................................ 730
SIP messaging.................................................................................................................... 731
Outgoing call message details........................................................................................ 731
Incoming call message details........................................................................................ 736
Codec selection............................................................................................................ 741
DTMF transmission....................................................................................................... 742
Fax over SIP................................................................................................................. 742
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Contents
Hold scenarios.............................................................................................................. 742
SIP REFER................................................................................................................... 744
IP Office SIP trunk specifications.......................................................................................... 745
RFCs............................................................................................................................ 746
Transport protocols........................................................................................................ 747
Request methods.......................................................................................................... 747
Response methods........................................................................................................ 747
Headers....................................................................................................................... 748
Chapter 17: Configuring Small Community Networking................................................... 749
Supported Small Community Network Network Layouts......................................................... 750
Telephone Features Supported Across Server Edition and SCN Networks............................... 751
Voicemail Support............................................................................................................... 753
Enabling Small Community Networking................................................................................. 753
Setup the VoIP Line from System A to System B............................................................. 753
Setup the VoIP Line from System B to System A............................................................. 755
Small Community Network Management............................................................................... 755
Enabling SCN Discovery................................................................................................ 756
Creating a Common Admin Account............................................................................... 756
Loading a Small Community Network Configuration......................................................... 757
Editing a Small Community Network Configuration........................................................... 758
Using the Network Viewer.............................................................................................. 759
System Inventory.......................................................................................................... 764
Small Community Network Remote Hotdesking..................................................................... 764
Small Community Network Fallback...................................................................................... 765
SCN Short Code Programming............................................................................................. 766
Chapter 18: Short Code Overview....................................................................................... 768
Short Code Characters........................................................................................................ 769
User Dialing........................................................................................................................ 772
Application Dialing............................................................................................................... 774
Secondary Dial Tone........................................................................................................... 775
? Short Codes..................................................................................................................... 776
Short Code Matching Examples............................................................................................ 777
Default System Short Code List............................................................................................ 781
Chapter 19: Short Code Features........................................................................................ 787
Acquire Call........................................................................................................................ 787
AOC Previous Call.............................................................................................................. 787
AOC Reset Total................................................................................................................. 787
AOC Total........................................................................................................................... 787
Auto Attendant.................................................................................................................... 788
Auto Intercom Deny Off....................................................................................................... 788
Auto Intercom Deny On....................................................................................................... 788
Break Out........................................................................................................................... 789
Barred................................................................................................................................ 789
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Contents
Call Intrude......................................................................................................................... 790
Call Listen........................................................................................................................... 790
Call Park............................................................................................................................. 792
Call Park and Page.............................................................................................................. 793
Call Pickup Any................................................................................................................... 793
Call Pickup Extn.................................................................................................................. 794
Call Pickup Group............................................................................................................... 794
Call Pickup Line.................................................................................................................. 795
Call Pickup Members........................................................................................................... 795
Call Pickup User.................................................................................................................. 796
Call Queue......................................................................................................................... 797
Call Record......................................................................................................................... 797
Call Steal............................................................................................................................ 798
Call Waiting On................................................................................................................... 799
Call Waiting Off................................................................................................................... 799
Call Waiting Suspend.......................................................................................................... 800
Cancel All Forwarding.......................................................................................................... 800
Cancel Ring Back When Free.............................................................................................. 801
Change Login Code............................................................................................................. 801
Clear After Call Work........................................................................................................... 802
Clear Call........................................................................................................................... 802
Clear CW............................................................................................................................ 803
Clear Hunt Group Night Service............................................................................................ 803
Clear Hunt Group Out Of Service.......................................................................................... 804
Clear Quota........................................................................................................................ 805
Coaching Intrusion.............................................................................................................. 805
Conference Add.................................................................................................................. 806
Conference Meet Me........................................................................................................... 806
CW.................................................................................................................................... 808
Dial.................................................................................................................................... 808
Dial 3K1............................................................................................................................. 809
Dial 56K............................................................................................................................. 809
Dial 64K............................................................................................................................. 810
Dial CW.............................................................................................................................. 810
Dial Direct........................................................................................................................... 811
Dial Direct Hot Line.............................................................................................................. 811
Dial Emergency................................................................................................................... 812
Dial Extn............................................................................................................................. 812
Dial Fax.............................................................................................................................. 813
Dial Inclusion...................................................................................................................... 813
Dial Paging......................................................................................................................... 814
Dial Physical Extension by Number....................................................................................... 815
Dial Physical Extension By ID............................................................................................... 815
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Contents
Dial Speech........................................................................................................................ 816
Dial V110............................................................................................................................ 816
Dial V120............................................................................................................................ 816
Dial Video........................................................................................................................... 817
Disable ARS Form............................................................................................................... 817
Disable Internal Forwards.................................................................................................... 817
Disable Internal Forward Unconditional................................................................................. 818
Disable Internal Forward Busy or No Answer......................................................................... 818
Display Msg........................................................................................................................ 818
Do Not Disturb Exception Add.............................................................................................. 820
Do Not Disturb Exception Delete.......................................................................................... 820
Do Not Disturb On............................................................................................................... 821
Do Not Disturb Off............................................................................................................... 821
Enable ARS Form............................................................................................................... 822
Enable Internal Forwards..................................................................................................... 822
Enable Internal Forward Unconditional.................................................................................. 823
Enable Internal Forward Busy or No Answer.......................................................................... 823
Extn Login.......................................................................................................................... 823
Extn Logout........................................................................................................................ 825
Flash Hook......................................................................................................................... 825
FNE Service....................................................................................................................... 826
Follow Me Here................................................................................................................... 826
Follow Me Here Cancel........................................................................................................ 826
Follow Me To...................................................................................................................... 827
Forward Hunt Group Calls On.............................................................................................. 828
Forward Hunt Group Calls Off.............................................................................................. 828
Forward Number................................................................................................................. 829
Forward On Busy Number.................................................................................................... 829
Forward On Busy On........................................................................................................... 830
Forward On Busy Off........................................................................................................... 830
Forward On No Answer On.................................................................................................. 831
Forward On No Answer Off.................................................................................................. 831
Forward Unconditional On.................................................................................................... 832
Forward Unconditional Off.................................................................................................... 833
Group Listen Off.................................................................................................................. 833
Group Listen On.................................................................................................................. 834
Headset Toggle................................................................................................................... 834
Hold Call............................................................................................................................. 835
Hold CW............................................................................................................................. 835
Hold Music.......................................................................................................................... 836
Hunt Group Disable............................................................................................................. 836
Hunt Group Enable.............................................................................................................. 837
Last Number Redial............................................................................................................. 838
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Contents
MCID Activate..................................................................................................................... 838
Mobile Twinned Call Pickup................................................................................................. 838
Off Hook Station.................................................................................................................. 839
Outgoing Call Bar Off........................................................................................................... 839
Outgoing Call Bar On........................................................................................................... 840
Private Call......................................................................................................................... 841
Private Call Off.................................................................................................................... 841
Private Call On.................................................................................................................... 841
Priority Call......................................................................................................................... 842
Record Message................................................................................................................. 842
Relay On............................................................................................................................ 843
Relay Off............................................................................................................................ 844
Relay Pulse........................................................................................................................ 844
Resume Call....................................................................................................................... 845
Retrieve Call....................................................................................................................... 845
Ring Back When Free.......................................................................................................... 846
Secondary Dial Tone........................................................................................................... 847
Set Absent Text.................................................................................................................. 847
Set Account Code............................................................................................................... 848
Set Authorization Code........................................................................................................ 849
Set Hunt Group Night Service.............................................................................................. 849
Set Hunt Group Out Of Service............................................................................................ 850
Set Mobile Twinning Number................................................................................................ 851
Set Mobile Twinning On....................................................................................................... 851
Set Mobile Twinning Off....................................................................................................... 851
Set No Answer Time............................................................................................................ 852
Set Time Profile.................................................................................................................. 852
Set Wrap Up Time............................................................................................................... 854
Speed Dial.......................................................................................................................... 854
Shutdown Embedded Voicemail........................................................................................... 855
Stamp Log.......................................................................................................................... 856
Startup Embedded Voicemail............................................................................................... 856
Suspend Call...................................................................................................................... 856
Suspend CW....................................................................................................................... 857
Toggle Calls........................................................................................................................ 857
Unpark Call......................................................................................................................... 858
Voicemail Collect................................................................................................................. 858
Voicemail Node................................................................................................................... 861
Voicemail On...................................................................................................................... 861
Voicemail Off...................................................................................................................... 862
Voicemail Ringback On........................................................................................................ 862
Voicemail Ringback Off........................................................................................................ 863
Whisper Page..................................................................................................................... 863
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Contents
Chapter 20: Button Programming Overview...................................................................... 865
Programming Buttons with Manager..................................................................................... 865
Programming Button via the Menu Key................................................................................. 867
Setting a Button to Dial a Number................................................................................... 867
Setting a Button to a Switch Function.............................................................................. 867
Setting Buttons to Admin Function.................................................................................. 868
Programming Button via an Admin Button............................................................................. 869
Using an Admin Button.................................................................................................. 870
BST Button Programming.................................................................................................... 871
T3 Self-Administration......................................................................................................... 873
Interactive Button Menus..................................................................................................... 875
Label Templates.................................................................................................................. 875
Chapter 21: Button Programming Actions......................................................................... 877
Abbreviated Dial.................................................................................................................. 885
Abbreviated Dial Pause....................................................................................................... 885
Abbreviated Dial Program.................................................................................................... 886
Abbreviated Dial Stop.......................................................................................................... 887
Account Code Entry............................................................................................................. 887
ACD Agent Statistics........................................................................................................... 888
ACD Stroke Count............................................................................................................... 889
Acquire Call........................................................................................................................ 890
AD Special Functions.......................................................................................................... 890
AD Special Function Mark.................................................................................................... 890
AD Special Function Wait..................................................................................................... 891
AD Suppress....................................................................................................................... 892
Appearance........................................................................................................................ 893
Automatic Callback.............................................................................................................. 895
Auto-Intercom Deny............................................................................................................. 896
Automatic Intercom.............................................................................................................. 897
Break Out........................................................................................................................... 898
Bridged Appearance............................................................................................................ 899
Busy................................................................................................................................... 899
Busy On Held...................................................................................................................... 900
Call Forwarding All.............................................................................................................. 900
Call Intrude......................................................................................................................... 901
Call List.............................................................................................................................. 902
Call Listen........................................................................................................................... 903
Call Log.............................................................................................................................. 904
Call Park............................................................................................................................. 904
Call Park and Page.............................................................................................................. 906
Call Park To Other Extension............................................................................................... 907
Call Pickup......................................................................................................................... 908
Call Pickup Any................................................................................................................... 908
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Contents
Call Pickup Group............................................................................................................... 909
Call Pickup Members........................................................................................................... 910
Call Queue......................................................................................................................... 911
Call Record......................................................................................................................... 912
Call Screening.................................................................................................................... 913
Call Steal............................................................................................................................ 915
Call Waiting Off................................................................................................................... 916
Call Waiting On................................................................................................................... 916
Call Waiting Suspend.......................................................................................................... 917
Cancel All Forwarding.......................................................................................................... 918
Cancel Leave Word Calling.................................................................................................. 919
Cancel Ring Back When Free.............................................................................................. 920
Clear Call........................................................................................................................... 920
Clear CW............................................................................................................................ 921
Clear Hunt Group Night Service............................................................................................ 922
Clear Hunt Group Out Of Service.......................................................................................... 923
Clear Quota........................................................................................................................ 924
Coaching Intrusion.............................................................................................................. 924
Conference......................................................................................................................... 925
Conference Add.................................................................................................................. 926
Conference Meet Me........................................................................................................... 927
Consult............................................................................................................................... 929
Coverage Appearance......................................................................................................... 929
Dial.................................................................................................................................... 930
Dial 3K1............................................................................................................................. 931
Dial 56K............................................................................................................................. 932
Dial 64K............................................................................................................................. 932
Dial CW.............................................................................................................................. 933
Dial Direct........................................................................................................................... 934
Dial Emergency................................................................................................................... 934
Dial Inclusion...................................................................................................................... 935
Dial Intercom....................................................................................................................... 936
Dial Paging......................................................................................................................... 937
Dial Physical Extn by Number.............................................................................................. 938
Dial Physical Number by ID.................................................................................................. 939
Dial Speech........................................................................................................................ 940
Dial V110............................................................................................................................ 940
Dial V120............................................................................................................................ 941
Display Msg........................................................................................................................ 942
Dial Video........................................................................................................................... 942
Directed Call Pickup............................................................................................................ 943
Directory............................................................................................................................. 944
Do Not Disturb Exception Add.............................................................................................. 945
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Contents
Do Not Disturb Exception Delete.......................................................................................... 946
Do Not Disturb Off............................................................................................................... 947
Do Not Disturb On............................................................................................................... 947
Drop................................................................................................................................... 948
Extn Login.......................................................................................................................... 949
Extn Logout........................................................................................................................ 950
Flash Hook......................................................................................................................... 951
Follow Me Here................................................................................................................... 952
Follow Me Here Cancel........................................................................................................ 953
Follow Me To...................................................................................................................... 953
Forward Hunt Group Calls Off.............................................................................................. 954
Forward Hunt Group Calls On.............................................................................................. 955
Forward Number................................................................................................................. 956
Forward On Busy Number.................................................................................................... 957
Forward On Busy Off........................................................................................................... 958
Forward On Busy On........................................................................................................... 959
Forward On No Answer Off.................................................................................................. 959
Forward On No Answer On.................................................................................................. 960
Forward Unconditional Off.................................................................................................... 961
Forward Unconditional On.................................................................................................... 962
Group................................................................................................................................. 963
Group Listen On.................................................................................................................. 964
Group Paging...................................................................................................................... 965
Headset Toggle................................................................................................................... 966
Hold Call............................................................................................................................. 967
Hold CW............................................................................................................................. 967
Hold Music.......................................................................................................................... 968
Hunt Group Enable.............................................................................................................. 969
Hunt Group Disable............................................................................................................. 970
Inspect............................................................................................................................... 970
Internal Auto-Answer........................................................................................................... 971
Last Number Redial............................................................................................................. 972
Leave Word Calling............................................................................................................. 972
Line Appearance................................................................................................................. 973
Manual Exclude.................................................................................................................. 974
MCID Activate..................................................................................................................... 974
Monitor Analogue Trunk MWI............................................................................................... 975
Off Hook Station.................................................................................................................. 976
Pause Recording................................................................................................................. 976
Priority Call......................................................................................................................... 977
Priority Calling..................................................................................................................... 978
Private Call......................................................................................................................... 978
Relay Off............................................................................................................................ 979
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Contents
Relay On............................................................................................................................ 980
Relay Pulse........................................................................................................................ 981
Resume Call....................................................................................................................... 981
Request Coaching Intrusion................................................................................................. 982
Retrieve Call....................................................................................................................... 983
Ring Back When Free.......................................................................................................... 984
Ringer Off........................................................................................................................... 985
Self-Administer.................................................................................................................... 986
Send All Calls..................................................................................................................... 987
Set Absent Text.................................................................................................................. 988
Set Account Code............................................................................................................... 990
Set Hunt Group Night Service.............................................................................................. 990
Set Hunt Group Out Of Service............................................................................................ 992
Set Inside Call Seq.............................................................................................................. 993
Set Night Service Destination............................................................................................... 993
Set No Answer Time............................................................................................................ 994
Set Out of Service Destination.............................................................................................. 994
Set Outside Call Seq........................................................................................................... 995
Set Ringback Seq................................................................................................................ 995
Set Wrap Up Time............................................................................................................... 996
Speed Dial.......................................................................................................................... 996
Stamp Log.......................................................................................................................... 997
Stored Number View............................................................................................................ 998
Suspend Call...................................................................................................................... 999
Suspend CW....................................................................................................................... 999
Swap CLID Name/Number................................................................................................. 1000
Time of Day...................................................................................................................... 1000
Time Profile...................................................................................................................... 1001
Timer................................................................................................................................ 1003
Transfer............................................................................................................................ 1004
Toggle Calls...................................................................................................................... 1004
Twinning........................................................................................................................... 1005
Unpark Call....................................................................................................................... 1006
User................................................................................................................................. 1007
Visual Voice...................................................................................................................... 1009
Voicemail Collect............................................................................................................... 1011
Voicemail Off.................................................................................................................... 1012
Voicemail On.................................................................................................................... 1013
Voicemail Ringback Off...................................................................................................... 1014
Voicemail Ringback On...................................................................................................... 1014
Whisper Page................................................................................................................... 1015
Chapter 22: Appearance Button Operation...................................................................... 1017
Appearance Button Features.............................................................................................. 1018
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Contents
Call Appearance Buttons................................................................................................... 1019
Call Appearance Example 1......................................................................................... 1019
Call Appearance Example 2......................................................................................... 1020
How are Call Appearance Buttons Treated?.................................................................. 1021
Call Appearance Button Indication................................................................................ 1022
Bridged Appearance Buttons.............................................................................................. 1024
Bridged Appearance Example 1................................................................................... 1024
Bridged Appearance Example 2................................................................................... 1025
Bridged Appearance Example 3................................................................................... 1026
How are Bridged Appearances Treated?....................................................................... 1027
Bridged Appearance Button Indication.......................................................................... 1028
Call Coverage Buttons....................................................................................................... 1029
Call Coverage Example 1............................................................................................. 1029
Call Coverage Example 2............................................................................................. 1030
How is Call Coverage Treated?.................................................................................... 1031
Call Coverage Button Indication.................................................................................... 1032
Line Appearance Buttons................................................................................................... 1033
Line Appearance Example 1........................................................................................ 1034
Line Appearance Example 2........................................................................................ 1035
How are Line Appearances Treated?............................................................................ 1035
Line Appearance Button Indication................................................................................ 1037
T3 Phone Line Appearances........................................................................................ 1038
Selected Button Indication.................................................................................................. 1039
Idle Line Preference.......................................................................................................... 1040
Ringing Line Preference..................................................................................................... 1043
Answer Pre-Select............................................................................................................. 1045
Auto Hold......................................................................................................................... 1046
Ring Delay........................................................................................................................ 1047
Delayed Ring Preference................................................................................................... 1049
Collapsing Appearances.................................................................................................... 1050
Joining Calls..................................................................................................................... 1051
Multiple Alerting Appearance Buttons.................................................................................. 1054
Twinning........................................................................................................................... 1054
Busy on Held.................................................................................................................... 1055
Reserving a Call Appearance Button................................................................................... 1055
Logging Off and Hot Desking.............................................................................................. 1056
Applications...................................................................................................................... 1056
Programming Appearance Buttons..................................................................................... 1057
Appearance Function System Settings.......................................................................... 1058
Appearance Function User Settings.............................................................................. 1059
Programming Line Appearance ID Numbers.................................................................. 1060
Outgoing Line Programming......................................................................................... 1062
Chapter 23: Overview of Data Routing............................................................................. 1063
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Contents
Network Address Translation (NAT).................................................................................... 1064
Dynamic Host Configuration Protocol (DHCP)..................................................................... 1064
Simple ISDN Internet Connection....................................................................................... 1065
ISDN Link Between IP Offices............................................................................................ 1065
Using a Dedicated T1/PRI ISP Link..................................................................................... 1066
Tasks for Using a Dedicated T1/PRI ISP Link................................................................ 1066
Create a New WAN Service......................................................................................... 1066
Create the Virtual WAN Port......................................................................................... 1068
Create an IP Route...................................................................................................... 1068
T1 PRI Trunk.............................................................................................................. 1069
Remote Access................................................................................................................. 1069
Creating a VoIP Link via the WAN Port Using PPP............................................................... 1072
Chapter 24: Appendix: SMDR............................................................................................ 1074
SMDR Fields..................................................................................................................... 1075
SMDR Examples............................................................................................................... 1079
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Chapter 1: Introduction
Related links
Purpose on page 25
Intended audience on page 25
Documentation resources on page 25
Support on page 26
Document changes since last issue on page 26
What's New in Release 9.1 on page 29
Purpose
This document contains descriptions of the configuration fields and the configuration procedures for
administering Avaya IP Office using the Manager application.
Related links
Introduction on page 25
Intended audience
The primary audience for the Administering Avaya IP Office using Manager is the customer system
administrator. Implementation engineers and support and services personnel may also find this
information helpful, however, they are not the primary audience.
Related links
Introduction on page 25
Documentation resources
For a listing of documentation resources related to IP Office, see IP Office Documentation Library.
Download documents from the Avaya Support website at http://support.avaya.com.
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Introduction
Related links
Introduction on page 25
Support
Visit the Avaya Support website at http://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.
Related links
Introduction on page 25
Document changes since last issue
New Reference Document
Content previously included in this document has been moved into separate reference documents.
This content is available in pdf format from the Avaya support site and in the IP Office Knowledge
Base.
Locales:
Locale information has been moved to Avaya IP Office Platform™ Locale Settings Reference.
Table 1: Changes summary for release 9.1
Section
Summary of changes
Incoming Call Route
In the Incoming Call Route section, under the subtitle Outgoing Caller ID Matching,
the following statement has been added.
For internal calls being forwarded or twinned, if multiple incoming call route entries
match the extension number used as caller ID, the first entry created is used. This
entry should start with a “-” character (meaning fixed length) and provide the full
national number. These entries do not support wildcards. If additional entries are
required for incoming call routing, they should be created after the entry required
for reverse lookup.
Incoming Call Route
consolidated records
Starting with release 9.1, Incoming Call Routes are not longer supported as
consolidated records.
This change is reflected in the description for File | Preferences | Consolidate
Solution to Primary Settings and in the description for Record Consolidation in the
chapter Working with the Server Edition Manager User Interface.
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Document changes since last issue
Section
Summary of changes
NoUser User Source
Numbers
The following NoUser User Source Number has been added:
REPEATING_BEEP_ON_LISTEN
Configuring General
System Settings
Template information has been consolidated in the Configuring General System
Settings, under Working with Templates on page 565.
Configuration Mode
Field Descriptions
Default settings on System | Telephony | Telephony
The following settings are now set by default to On.
• Inhibit Off-Switch Forward/Transfer
• Drop External Only Impromptu Conference
The following statement has been added to the field Line | SIP Line | Transport |
Use Network Topology Info.
If no STUN server address is set for the interface, then the Binding Refresh Time
(System | LAN | Network Topology) is ignored by SIP Lines when calculating the
periodic OPTIONS timing unless the Firewall/NAT Type is set to Open Internet.
The tab System | ACCS has been renamed System | Contact Center.
Security Mode Field
Descriptions
The field description information for the tab System | Certificates has been
revised and updated.
File | Advanced |
Initial Configuration
The Initial Configuration section now contains the following note.
Note:
The Initial Configuration utility changes the security settings. Therefore, the
user running the utility must have security read/write rights.
Server Edition remote
worker capability
For release 9.0 and higher, on Server Edition systems, all users can be configured
for remote H.323 extension usage.
The licensing information in Remote H.323 Extensions on page 605 has been
updated.
Extension | IP DECT
settings
The Extension | IP DECT settings page now states “These settings are mergeable
with the exception of the Reserve License setting. Changing the Reserve
License settings requires a reboot of the system.”
ARS | ARS Rout ID
The ARS | ARS Rout ID field can now be edited. The field description has been
updated as follows.
The default value is automatically assigned. Range = 0 to 99999.
For most deployments, do not edit this field.
For those conditions where it is necessary to edit this field,the value must be
unique within ARS and within the line Outbound GroupIDs.
System | LAN | VoIP
DiffServ Settings
The description for System | LAN | DiffServ Settings has been updated to
remove out of date content. The description is now as follows.
DiffServ Settings
When transporting voice over low speed links it is possible for normal data packets
(1500 byte packets) to prevent or delay voice packets (typically 67 or 31 bytes)
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Introduction
Section
Summary of changes
from getting across the link. This can cause unacceptable speech quality.
Therefore it is important that all traffic routers and switches in a network to have
some form of Quality of Service mechanism (QoS). QoS routers are essential to
ensure low speech latency and to maintain sufficient audible quality.
The system applies the DiffServ settings to outgoing traffic on any SIP lines which
have their User Network Topology Info setting (Line | Transport) set to match
the LAN interface.
The system supports the DiffServ (RFC2474) QoS mechanism. This uses a Type
of Service (ToS) field in the IP packet header.
The hex and decimal entry fields for the following values are linked, the hex value
being equal to the decimal multiplied by 4.
User | Web Self
Administration
The login URL has been added to the User | Web Self Administration page.
Call Barring
information
Call barring information was mistakenly removed from previous issues. The
content has been added back in. See Call Barring on page 625.
File | Preferences |
Preferences | SE
Central Access
A field description has been added for File | Preferences | Preferences | SE
Central Access.
Directory
The Directory record capacity has been increased to 7500.
Extension |
Extension | Phone
Password
The field description has been changed to the following.
H.323 Extensions only. Does not apply to T3 series phones and DECT phones.
The password that must be entered as part of phone registration. This password is
used to secure registration of H.323 phones when there is no matching user to
authenticate against. Required if Media Security is enabled.
SMDR Fields
Fields 31 to 34 have added. These fields are used to uniquely identify a call made
in an IP Office Server Edition solution.
System | Unsecured
Interfaces | User
Service User
Credentials
The field description has been updated.
Services | Service
Details | Service
Access Source
The field description has been updated with new options.
System | Directory
Services | HTTP
The following fields have been added:
• HTTPS Enabled
• Port Number
Group | Group
The following note has been added to the Group | Profile option XMPP Group.
Important:
Before adding a user to an XMPP group, the user must be added to the
configuration and the configuration saved. If the user is added to the group
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What's New in Release 9.1
Section
Summary of changes
before the directory is synchronized, the user will not be visible in one-X
Portal.
Related links
Introduction on page 25
What's New in Release 9.1
SIP Line Simplification
SIP Line configuration parameters have been reorganized. The standard line tabs now contain only
those parameters that are commonly used for most deployment scenarios. A new SIP Advanced tab
and a new SIP Engineering tab now contain the less common configuration parameters, usually
reserved for expert users and support personnel. See SIP Line on page 353.
SIP Line Silence Suppression
The SIP Line | Advanced tab now contains a Send SilenceSupp=Off check box. This is used for
the G711 codec. When checked, the silence suppression off attribute is sent in SDP on this trunk.
Resiliency
Location Based Resiliency: Location based resiliency allows you to create a group of H.323
extensions by applying a common Location value to each extension. You can then specify the
fallback IP Office server for the group. Using location based resiliency
• Phones on the Primary and Secondary server can fall back to an Expansion system
• Phones on an Expansion system can fall back to another Expansion system
Automatic Fallback Recovery: After a fallback event, if the new setting System | Telephony |
Phone Failback is set to Automatic, and the phone’s primary gatekeeper has been up for more
than 10 minutes, the system causes idle phones to perform a failback recovery to the original
system.
IP DECT Line Resiliency: The IP DECT Line | Gateway tab now contains configuration fields for
Enable Resiliency. If resiliency is enabled, you can enable the option Backs up my IP DECT
Phones on the IP Office Line tab.
Suppress NoCallerId Alarm
The NoCallerId alarm can be suppressed using a NoUser source number. For a description of
source numbers, see User | Source Numbers.
For the procedure to suppress the alarm, see Suppressing the NoCallerId Alarm on page 679.
Directory Overrides Barring
Call barring can be overridden for numbers entered in the external directory. The configuration
setting Directory Overrides Barring has been added to the System | Telephony | Telephony tab.
For configuration information, see Overriding Call Barring on page 626.
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Introduction
Message Waiting Indicator for Analog Trunks
A Message Waiting Indicator (MWI) using a Bellcore FSK MWI signal is now supported for analog
trunks that terminate on an ATM4U-V2 card.
The configuration setting is located at Extension | Analog | Message Waiting Lamp Indication
Type.
For a configuration procedure, see Message Waiting Indication for Analogue Trunks on page 676.
Toll Bypass Prevention
A configuration procedure has been added for preventing toll bypass. See PreventingToll Bypass on
page 624.
Web Collaboration
The User | User tab, now contains a Web Collaboration check box. When enabled, it allows the
user to use the Web Collaboration application.
Answer and Disconnect Supervision
The System | Telephony | Tones and Music tab now contains a Analog Trunk VAD check box.
Select this option to enable Voice Activity Detection (VAD) for analog trunks terminating on the
ATM4U-V2 card. VAD functionality provides a Call Answer signal triggered by voice activity.
Third Party Voice Quality Monitoring
The RTCP collector IP address for phones field has been added to the LAN | VoIP tab. This
setting enables you to send the RTCP data collected to a third party QoS monitoring application.
Enable Remote Working
On the User | User tab, the Enable Remote Worker option does not need to be enabled for users
with SIP phones if an Avaya Session Border Controller for Enterprise (ASBCE) is deployed in the
network to allow remote workers to register their SIP phone from a remote location.
Outcalling Control
On the System | Voicemail tab, you can use the Outcalling Control setting to enable or disable
system wide outcalling on Voicemail Pro.
Authorization Codes
The following changes have been made to authorization codes.
• Authorization codes are now enabled by default.
• SMDR field 19 shows n/a regardless of whether an authorization code was used.
• Authorization codes can no longer be associated with User Rights. An authorization code must
be associated with a user.
Note:
In release 9.1, authorization codes can no longer be associated with User Rights. If an
authorization code was configured in relationship with User Rights in an earlier release
configuration, this authorization code will be lost during upgrade. The administrator must reconfigure the authorization code, after upgrade. The authorization code must be associated with
a user.
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What's New in Release 9.1
The authorization code configuration settings are described on the page for the Authorization
Code tab.
For a configuration procedure, see Configuring Authorization Codes on page 609.
Mergeable Settings
On the System | LAN1 | VoIP tab, under the SIP Registrar Enable settings, the Auto-create Extn/
User setting is now mergeable. Changing this setting does not require a reboot.
When creating an IP DECT line, the settings are now mergeable. You can also remove an IP DECT
line without rebooting.
Support for IP400 and IP500 Hardware Discontinued
The IP400 and IP 500 platforms are not supported in release 9.1.
Alarms
The following alarms have been added. For details, see System | System Events.
• Log stamped
• CPU warning/critical
• Memory use warning/critical
Security
User accounts:
• The Manager and Operator default service users have been removed.
• The following administrative accounts are disabled by default:
- IPDECTService
- BranchAdmin
- BuisinessPartner
- Maintainer
Phone Login PIN:
Voicemail PIN:
You can now configure a phone login PIN using the Login Code Complexity fields on the System |
Telephony tab.
You can now configure a voicemail PIN using the Voicemail Code Complexity fields on the
System | Voicemail tab.
Auto-create User and Auto-create Extension:
The default setting is now Off for all Auto-create fields.
LAN Settings:
On the System | LAN1 / LAN2 tabs, the following settings have changed.
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Introduction
Field
Default
SIP Registrar Enable
Default = Off.
RTP Port Number Range
IP 500 v2 default = 4000. Range = 46750 to 50750.
Linux default = 10000. Range = 40750 to 50750
Security General Settings:
On the Security Settings | General Settings tab, the following default values have been changed.
Field
Default
Security Administrator
Password
Range = 8 to 31 characters.
Minimum Password
Complexity
Default = Medium.
Previous Password Limit
(Entries)
Default = 4.
Service User Details
Minimum Password Length
Range 8 to 31 characters.
Password Reject Action
Default = Log and Temporary Disable.
Minimum Password
Complexity
Default = Medium.
Expiry Reminder Time
Default = 10.
IP Office User Details
Password Enforcement
Default = On.
Minimum Password Length
Range 8 to 31 characters.
Minimum Password
Complexity
Default = Medium
Password Reject Limit
Default = 5.
Password Reject Action
Default = Log and Temporary Disable.
Security Settings | System | Unsecured Interfaces:
The following values have changed.
Field
Default
The fields
Default = On.
TFTP Configuration Read and
TFTP Configuration Write
replaced with TFTP Server
This setting enables or disables the TFTP server.
TFTP Directory Read
Default = Off. Also disabled if TFTP Server is set to Off.
TFTP Voicemail
Disabled if TFTP ServerIt is set to Off.
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What's New in Release 9.1
Field
Default
The EConf field has been
removed.
DevLink
Default = On.
The Real Time Interface field
has been removed.
Related links
Introduction on page 25
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Chapter 2: Overview
This documentation covers the use of the Avaya IP Office Manager. Manager runs on a Windows
PC and connects to the IP Office system via Ethernet LAN or WAN connections.
Important:
Manager is an off-line editor. It receives a copy of the system's current configuration settings.
Changes are made to that copy and it is then sent back to the system for those changes to
become active. This means that changes to the active configuration in the system that occur
between Manager receiving and sending back the copy may be overwritten. For example, this
may affect changes made by a user through their phone or voicemail mailbox after the copy of
the configuration is received by Manager.
Related links
Manager Modes on page 34
Manager Modes
The menus and options displayed by Manager vary depending on the actions you are performing.
Manager runs in the following modes.
Basic Edition Mode
This is the mode used when a Basic Edition configuration is opened. Basic Mode includes systems
running Partner, Norstar, or Quick Mode. For information on administering a Basic Edition system,
see the IP Office Basic Edition Manager.
Security Configuration Mode
Manager can be used to edit the security settings of IP Office systems.
Standard Mode Configuration Mode
This is the mode used when a Standard Mode configuration is opened. Standard Mode includes
systems running Standard, Preferred, or Advanced Edition.
Server Edition Configuration Mode
This is the mode used when a Server Edition network configuration is opened.
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Manager Modes
Small Community Network Management
Manager supports loading the combined configurations from systems in a Small Community
Network.
IP Office Shell Server Mode
The IP Office Shell Server is a single installation of selected IP Office applications running on Linux.
You can use Manager to administer an IP Office Shell Server.
Embedded File Management
For systems with a memory card installed, Manager can be used to view and manage the files
stored on the card. Embedded File Management can be accessed by selecting File | Advanced |
Embedded File Management.
Upgrade Wizard
The Upgrade Wizard is a component of Manager used to upgrade the firmware run by the system.
Related links
Overview on page 34
Security Configuration Mode on page 35
Standard Mode Configuration Mode on page 36
Server Edition Configuration Mode on page 38
Shell Server Mode on page 39
Backward Compatibility on page 40
Security Configuration Mode
When Manager is in Security Mode, the screen elements shown are available.
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Overview
Related links
Manager Modes on page 34
Standard Mode Configuration Mode
When Manager is in configuration mode, the screen elements shown are available. Some of these
elements can be customized, moved and hidden.
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Manager Modes
Manager Configuration Mode Screen Elements
1
Title Bar
In addition to the application name, when configuration settings are loaded from a system, the title
bar displays the user name used to load the settings and the operator view applied.
2
Menu Bar
The options available with the drop down menus provided here change according to whether
Manager has a set of configuration or security settings loaded or not.
3
Main Toolbar
This toolbar provides icon shortcuts to the most frequently required configuration setting actions.
4
Navigation Pane
This pane shows icons for the different types of record that the configuration can contain. Each type
is followed by the number of records of that type already in the configuration. Selecting an icon
displays the matching records in the group pane and navigation toolbar.
5
Group Pane
This pane lists all the records that match the type selected in the navigation pane or navigation
toolbar. The list can be sorted by clicking on column heading. Selecting a record in this pane
displays its details in the details pane.
6
Details Pane
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Overview
This pane shows the configuration settings for a particular record within the configuration. The
record is selected using the navigation toolbar or using the navigation pane and group pane.
7
Navigation Toolbar
This toolbar provides a set of drop downs which can be used to navigate to particular records in the
configuration settings. The selected options in the navigation pane, the group pane and the details
pane are synchronized with the navigation toolbar and vice versa. This toolbar is particularly useful
if you want to work with the group pane and or navigation pane hidden in order to maximize the
display space for the details pane.
8
Error Pane
This pane shows errors and warnings about the configuration settings. Selecting an item here loads
the corresponding record into the details pane.
9
Status Bar This bar display messages about communications between Manager and systems. It
also displays the security level of the communications by the use of a padlock icon.
Related links
Manager Modes on page 34
Server Edition Configuration Mode
When the configuration from a Server Edition solution is loaded into Manager, Manager switches to
Server Edition mode operation.
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Manager Modes
Related links
Manager Modes on page 34
Shell Server Mode
An IP Office Shell Server is a single installation of selected IP Office applications running on Linux.
You can use Manager to configure and administer a Shell Server. Application Servers and Unified
Communications Modules (UCM) run on an IP Office Shell Server.
Since a Shell Server does not provide telephony, when you open a Shell Server configuration in
Manager, all telephony functions are disabled. The following Manager functions are supported for
Shell Servers:
• Discovery
• Initial configuration utility.
• System status.
• Load, edit and save security settings.
• Load, edit, and save the configuration.
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Overview
• Erase configuration and security settings.
• Audit trail display.
• Web Control.
For more information on the management of an IP Office Shell Server, see Installing and
Maintaining Avaya IP Office™ Platform Application Server and Installing Avaya IP Office™ Platform
Unified Communications Module.
Related links
Manager Modes on page 34
Backward Compatibility
Manager is part of the IP Office Admin Suite of programs. The Manager application can be used to
manage configurations from systems running earlier software releases. Manager adjusts the
settings and fields that it shows to match the core software level of the system.
Manager is able display systems with software levels it does not support in the Select IP Office
discovery menu, however those systems are indicated as not supported.
Backwards compatibility is only supported for General Availability releases of IP Office software. It is
not supported for private builds.
Note that this document describes the current release. If you are running an earlier software
release, obtain the Manager document for the specific release from the Avaya support site.
Related links
Manager Modes on page 34
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Chapter 3: Getting Started
Related links
PC requirements on page 41
Installing Manager on page 42
Starting Manager on page 43
Opening a Configuration on page 44
Changing the Manager Language on page 46
PC requirements
Supported Operating Systems
• Windows 7 (32/64 bit)
• Windows 8.1 (32/64 bit)
• Server 2008 R2 (32/64 bit)
• Server 2112 R2 (64 bit)
Minimum PC Requirements
IP Office
System
System RAM
(minimum or
higher)
Available
memory
required for
Manager
operations
Minimum free
hard disk
space
Processor
(similar or
higher)
Network size
supported
Standard Mode
4 GB
2 GB
6 GB
Intel® Core™ i3
or equivalent, 2
GHz minimum
Not applicable.
Server Edition
4 GB (32 bit
OS)
2 GB
6 GB
Intel® Core™ i3
or equivalent, 2
GHz minimum
Up to 32 nodes
Server Edition
8 GB (64 bit
OS)
4 GB
6 GB
Intel® Core™ i5
or equivalent, 2
GHz minimum
Up to 150
nodes
Applications
If not already present, the .NET Framework 4.0 application is installed with Manager.
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Getting Started
Ports
For information on port usage see the IP Office Avaya Port Matrix document on the Avaya support
site at https://support.avaya.com/helpcenter/getGenericDetails?detailId=C201082074362003
Related links
Getting Started on page 41
Installing Manager
Manager is a component of the IP Office Admin suite of applications. This suite is supplied on the
Software DVD (Disk 1). Alternatively, the IP Office Admin Suite can be downloaded from Avaya's
support website http://support.avaya.com.
In addition to Manager, the Admin suite includes options to install the following applications:
• System Monitor This is a tool for system installers and maintainers. Interpreting the
information output by System Monitor requires detailed data and telecoms knowledge.
• System Status Application This is a Java application that can be used to monitor the status
of the system such as extension, trunks and other resources. It displays current alarms and
most recent historical alarms.
Note:
This installation process will install Windows .NET2 if not already present. The installation
of .NET2 may require some systems to restart and the installation process to then be restarted.
Procedure
1. If installing from the Admin DVD, insert the DVD and when the page is displayed click on the
link for the Admin suite.
This will open a file windows showing the installation files for the suite. Locate and doubleclick on the setup.exe file.
2. Select the language you want to use for the installation process.
This does not affect the language used by Manager when it is run. Click Next >.
3. If an upgrade menu appears, it indicates that a previous installation has been detected.
Select Yes to upgrade the existing installed applications.
4. If required select the destination to which the applications should be installed.
We recommend that you accept the default destination. Click Next >.
5. The next screen is used to select which applications in the suite should be installed.
Clicking on each will display a description of the application. Click on the next to each
application to change the installation selection. When you have selected the installations
required, click Next >.
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Starting Manager
6. The applications selected are now ready to be installed.
Click Next >.
7. Following installation, you will be prompted whether you want to run Manager.
Selecting Yes runs Manager.
8. On some versions of Windows, you may be required to restart the PC.
Allow this to happen if required.
Related links
Getting Started on page 41
Starting Manager
No name or password is required to start Manager. A name and password is only required when
connecting with a system.
When started, by default Manager will attempt to discover any systems on the network. If it finds any
it will display a list from which you can select the system required.
1. Select Start and then Programs or All Programs depending on the version of Windows.
Select the IP Office program group.
2. Select
to run.
Manager. If a Windows Security Alert appears select Unblock to allow Manager
3. By default Manager will scan the network for any systems. What appears next depends on
whether it finds any systems.
• If Manager finds multiple systems, the Select IP Office window displays a list of those systems
from which you can select the one whose configuration you want to edit. If you want to open a
configuration go to Opening a Configuration. If you don't want to load a configuration click on
Cancel.
• If it finds a single system, it will attempt to open the configuration of that system by displaying
the Configuration Service User Login window..
• If no systems are found or you cancel the steps above, the Manager simplified view is
displayed.
Use the simplified view to select one of the following action.
• Create an Offline Configuration
• Open a Configuration from a System
• Read a Configuration from a File
Related links
Getting Started on page 41
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Getting Started
Opening a Configuration
The initial IP address ranges in which Manager searches for systems is set through the File |
Preferences | Discovery. By default, Manager scans the local network of the Manager PC.
1. Start Manager. If Manager is already started and a configuration is open in it, that
configuration must be closed first.
If Manager is set to Auto Connect on start up, it will scan for systems automatically and
either display the list of systems discovered or automatically start login to the only system
discovered.
Otherwise, select File | Open Configuration.
2. The Select IP Office window opens, listing those systems that responded.
• If Server Edition systems are detected, they are grouped together. By default the
configuration of those systems cannot be opened using Manager in Advanced View
mode and the configuration of a Primary Server can only be opened if the Open with
Server Edition Manager option is also selected.
• If Manager has been set with SCN Discovery enabled, systems in a Small Community
Network are grouped together. The checkbox next to the network name can be used to
load the configurations of all the configurations into Small Community Network
management mode.
• If the system required was not found, the Unit/Broadcast Address used for the search
can be changed. Either enter an address or use the drop-down to select a previously used
address. Then click Refreshto perform a new search.
• A list of known systems can be stored using Known System Discovery.
• Manager can be configured to search using DNS names.
• Systems found but not supported by the version of Manager being used will be listed as
Not Supported.
• If the system detected is running software other than from its primary folder, a
warning
icon will be shown next to it. The configuration can still be opened but only as a read-only
file.
3. When you have located the system required, check the box next to the system and click OK.
If the system selected is a Server Edition system and Manager is not running in Server
Edition mode, an Open with Server Edition Manager checkbox is shown and pre-selected.
Clicking OK will switch Manager to its Server Edition mode before loading the configuration.
4. The system name and password request is displayed. Enter the required details and click
OK.
The name and password used must match a service user account configured within the
system's security settings.
5. Additional messages will inform you about the success or failure of opening the configuration
from the system.
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Opening a Configuration
The method of connection, secure or insecure, attempted by Manager is set the applications
Secure Communications preferences setting.
• When Secure Communications is set to On, a
the lower right Manager status field.
padlock icon is displayed at all times in
• New installations of Manager default to having Secure Communications enabled. This
means Manager by default attempts to use secure communications when opening a
configuration.
• For Server Edition systems, Manager will always attempt to use secure communications
regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds, Manager
will offer to fallback to using unsecured communications.
6. Following a successful log in, the configuration is opened in Manager. The menus and
options displayed will depend on the type of system configuration loaded.
Related links
Getting Started on page 41
Login messages on page 45
Login messages
While attempting to login to a system, various messages may be displayed.
Configuration Not Loaded Messages
Access Denied
Displayed as the cause if the service user name/password were incorrect, or the service user has
insufficient rights to read the configuration. The Retry option can be used to log in again but multiple
rejections in a 10 minute period may trigger events, such as locking the user account, set by the
Password Reject Limit and Password Reject Action options in the systems security settings.
Failed to communicate with system
Displayed as the cause if the network link fails, or the secure communication mode is incorrect (for
example Manager is set to unsecured, but the system is set to secure only).
Account Locked
The account of the service user name and password being used is locked. This can be caused by a
number of actions, for example too many incorrect password attempts, passing a fixed expiry date,
etc. The account lock may be temporary (10 minutes) or permanent until manually unlocked. An
account can be enabled again through the system's security settings.
Additional Messages
Your service user account will expire in X days
Indicates that an Account Expiry date has been set on the system service user account and that
date is approaching. Someone with access to the system's security settings will be required unlock
the account and set a new expiry date.
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Getting Started
Your password will expire in X days. Do you wish to change it now?
Indicates that password ageing has been configured in the system's security settings. If your
password expires, someone with access to the system's security settings will be required to unlock
the account.
Change password
Through the system's security settings, a service user account can be required to change their
password when logging in. The menu provides fields for entering the old password and new
password.
Contact Information Check - This configuration is under special control
This message displays if a Manager user with administrator rights has entered their contact
information into the configuration. For example to indicate that they do not want the configuration
altered while a possible problem is being diagnosed. The options available are:
• Cancel Select this option to close the configuration without making any changes.
• Set configuration alteration flag Select this option if the configuration is being opened
because some urgent maintenance action. When the configuration is next opened, the fact that
it has been altered will be indicated on the System | System tab.
• Delete Contact Information Select this option to take the system out of special control.
• Leave contact information and flags unchanged (Administrators only) This option is only
available to service users logging in with administrator rights.
Related links
Opening a Configuration on page 44
Changing the Manager Language
About this task
The Manager application can run in multiple languages. By default it tries to use the best match to
the PC's regional location settings, otherwise it will use UK English.
The process below can be used to run Manager in one of its supported languages. However it does
not change the language used for help file content.
Procedure
1. Create a Windows shortcut to the Manager application .exe file. By default this file is located
in C:\Program Files\Avaya\IP Office\Manager\Manager.exe.
2. Right-click on the shortcut and select Properties.
3. The Target field can be used to specify the locale setting that Manager should use.
For example, for Italian the Target should have -locale:it-IT added to the end. For example:
"C:\Program Files\Avaya\IP Office\Manager\Manager.exe" -locale:it-IT.
4. Click OK.
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Changing the Manager Language
5. Manager should now run in the selected language when launched using the updated
shortcut.
Example
Table 2: Manager Locales
Manager Language
Shortcut Locale Setting
Brazilian Portuguese
locale:pt-Br
Chinese (Simplified)
-locale:zh-Hans
French
-locale:fr-FR
German
-locale:de-DE
Italian
-locale:it-IT
Mexican Spanish
-locale:es-MX
Russian
-locale:ru-RU
US English
-locale:en-US
Related links
Getting Started on page 41
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Chapter 4: Menu Bar Commands
The commands available through the Manager's menu bar change according to the mode in which
Manager is running. Commands may also be grayed out if not currently applicable. For some
commands, an arrow symbol indicates that there are sub-commands from which a selection can be
made.
The following sections outline the functions of each command. The Edit and Help menus are not
included.
Related links
File Menu on page 48
View Menu on page 78
Tools Menu on page 79
Security Mode Menus on page 84
Embedded File Management Menus on page 85
File Menu
Related links
Menu Bar Commands on page 48
File > Open Configuration on page 49
File > Close Configuration on page 49
File > Save Configuration on page 49
File > Save Configuration As on page 49
File > Change Working Directory on page 50
File | Preferences on page 51
File | Offline on page 59
File | Advanced on page 60
File | Backup/Restore on page 77
File | Import/Export on page 77
File | Exit on page 78
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File Menu
File > Open Configuration
This command displays the Select IP Office window used to receive a systems configuration
settings.
The same action is performed by the
icon in the Main Toolbar.
The Select IP Office menu is also used for other actions such as reboot and sending a
configuration. If the unit required is not found, the Unit/Broadcast Address can be changed and then
Refresh clicked. To change the TCP addresses scanned, select File | Preferences | Discovery and
enter the required addresses in the IP Search Criteria.
Known Units is not available unless configured
Related links
File Menu on page 48
File > Close Configuration
This command closes the currently loaded configuration without saving it.
Related links
File Menu on page 48
File > Save Configuration
The File > Save Configuration command saves the amended configuration.
If the configuration has been received from a system, the Send Config menu is displayed.
If the configuration file has been opened offline or created from new, the file is saved to disk only.
Related links
File Menu on page 48
File > Save Configuration As
The File > Save Configuration As command allows you to save a configuration a file on the
Manager computer. Note that dynamic configuration data, for example hunt groups advertised from
other systems in a network, are not included in a configuration file saved onto PC and then
reopened.
The command displays the Save As window. Select the drive and directory. and then enter the new
file name. Once you have entered the file name, the Save Configuration File window opens, where
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Menu Bar Commands
you have the option to encrypt the file by entering a password. Leave the password blank if you do
not want to encrypt the file.
Important:
Encrypted configuration files can only be opened with Manager 9.1 or later. In earlier versions of
Manager, the file will open but it is empty.
Configurations saved onto the PC in this way can be reopened using the
icon or the File >
Offline > Open File command. If the file has been encrypted, you must enter the password.
When Manager is running in Server Edition mode, the Save command operates differently. Multiple
files are saved, one .cfg file for each server in the network plus a single .cfi file for the whole
network.
The .cfi file can be used with the File > Offline > Open File Set command to open the whole set of
files in a single action.
Related links
File Menu on page 48
File > Change Working Directory
These settings allow you to change the default locations where Manager looks for and saves files.
These fields set the default location where Manager will look for and save files.
Directory
Description
Working Directory (.cfg
files)
Sets the directory into which Manager saves .cfg files. By default this is the
Manager application's program directory.
Note:
On Windows 7 systems, the default folder for Manager .cfg files is C:
\Program Files (x86)\Avaya\IP Office\Manager. On some Windows 7
systems, the file is saved to the user's profile folder at C:\Users
\<user_name>\AppData\Local\VirtualStore\Program Files (x86)\Avaya\IP
Office\Manager. You must turn on Show hidden files to access this folder.
Alternatively, you can set the working directory to an alternate location.
Binary Directory (.bin files)
Sets the directory in which the Manager upgrade wizard, HTTP, TFTP and
BOOTP functions look for firmware files requested by phones and other
hardware components. That includes .bin file, .scr files and .txt files. By
default this is the Manager application's program directory.
Tip:
In the Upgrade Wizard, right-clicking and selecting Change Directory also
changes this setting.
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File Menu
Directory
Description
Warning:
Historically, by default the Working Directory and Binary Directory are
the same. This is deprecated as it potentially allows remote TFTP/HTTP file
access to the folder containing copies of configuration files. Therefore it is
recommended that either of the folders is changed to an alternate location.
Known Units File
Sets the file and directory into which Manager can record details of the systems
it has discovered. Once a file location has been specified, a Known Units button
becomes available on the discovery menu used for loading system configuration.
Pressing that button displays the known units file as a list from which the
required system can be selected. It also allows sorting of the list and records to
be removed.
Related links
File Menu on page 48
File | Preferences
This command displays a window for configuring various aspects of Manager's operation. The
window is divided into a number of tabs.
Related links
File Menu on page 48
File | Preferences | Preferences on page 51
File | Preferences | Directories on page 53
File | Preferences | Discovery on page 54
File > Preferences > Visual Preferences on page 55
File | Preferences | Security on page 56
File | Preferences | Validation on page 58
File | Preferences | Preferences
This tab is accessed through File | Preferences and then selecting the Preferences tab.
Setting
Description
Edit Services Base TCP
Port:
Default = On.
Service Base TCP Port
Default = 50804.
This field shows or hides the base communication port settings.
Access to the configuration and security settings on a system requires Manager
to send its requests to specific ports. This setting allows the TCP Base Port used
by Manager to be set to match the TCP Base Port setting of the system. The
system's TCP Base Port is set through its security settings.
Service Base HTTP Port
Default = 80.
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Menu Bar Commands
Setting
Description
Access to the HTTP server on a system requires Manager to send its requests
to specific ports. This setting allows the HTTP Base Port used by Manager to be
set to match the HTTP Base Port setting of the system. The system’s HTTP
Base Port is set through its security settings.
Enable Time Server
Default = On.
This setting allows Manager to respond to RFC868 Time requests from systems.
It will provide the system with both the UTC time value and the local time value
of the PC on which it is running.
Enable BootP and TFTP
Servers
Default = Off.
Enable Port for Serial
Communication
Not used. This is a legacy feature for some older control units that were
managed via the serial port rather than the LAN.
Enter Port Number to be
used for Serial
Communication
Used with the setting above to indicate which serial port Manager should use.
Auto Connect on start up
Default = On
This setting allows Manager to respond to BOOTP request from systems for
which it also has a matching BOOTP record. It also allows Manager to respond
to TFTP requests for files.
If on, when Manager is started it will automatically launch the Select IP Office
menu and display any discovered systems. If only one system is discovered,
Manager will automatically display the login request for that system or load its
configuration if the security settings are default.
Set Simplified View as
default
Default = On
Default to Standard Mode
Default = Off
If on, the Manager will start in simplified view mode if no configuration is loaded.
If on, when a configuration from a new or defaulted system running in Basic
mode is loaded, Manager will automatically convert the configuration to
Standard mode. Sending the configuration back to the system will restart it in
Standard mode. Only select this option if the only systems you expect to install
are Standard systems.
This setting does not affect existing systems with non-default configurations.
Use Remote Access
Default = Off.
If selected, access to all the configurations of a multi-site network is allowed via
remote access to the primary server on the multi-site network. When selected,
an additional Use Remote Access check box option is displayed on the Select
IP Office menu when the Open with Server Edition Manager checkbox option
is selected or if Manager is already running in Server Edition mode.
Consolidate Solution to
Primary Settings
This setting is used by Manager when in Server Edition mode.
If Consolidate Network to Primary Settings is selected:
• Entry and administration of Short Code, Time Profile, Account Code and
User Rights records is performed only at the solution level.
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File Menu
Setting
Description
• Those records are then automatically replicated in the configurations of all the
systems in the solution but are still only visible and editable at the solution
level.
• When the configurations are loaded into Manager or when this setting is
changed to become selected, if any inconsistency between records are found,
a Consolidation Report is displayed. This report allows selection of whether
to update the system to match the primary or to update the primary to match.
If Consolidate Network to Primary Settings is not selected:
Entry and administration of Short Code, Time Profile, Account Code and User
Rights records can be performed at both the solution and individual system
levels.
• Records entered and edited at the solution level are automatically replicated in
the configurations of all the systems in the solution. Manager displays a label
on the record indicating that it is a record that is shared across the solution.
• If a shared record is edited at the individual system level, that copy of the
record is no longer shared with the other systems. It will not be updated by any
changes to the solution level version of the same record.
• No consolidation checking for inconsistencies is done by Manager when the
configurations are loaded.
SE Central Access
Default = Off. Displayed on Server Edition systems only.
If On, all Server Edition systems in the network obtain their configuration data
from a central location on the Primary Server. As a result, the display of
configuration changes is delayed until a synchronization process runs. The
synchronization process runs every 40 seconds. If the configuration change
requires a system restart, a refreshed configuration display is delayed until 40
seconds after system restart.
This setting can be used to drive configuration changes into expansion systems
when the expansion systems are not reachable through Manager and the only
accessible system is the Primary Server.
SE Central Access Port
Default = 7070.
When SE Central Access is set to On, the port used for routing HTTPS
requests for configuration synchronization.
Related links
File | Preferences on page 51
File | Preferences | Directories
These settings allow you to change the default locations where Manager looks for and saves files.
These fields set the default location where Manager will look for and save files.
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Menu Bar Commands
Directory
Description
Working Directory (.cfg
files)
Sets the directory into which Manager saves .cfg files. By default this is the
Manager application's program directory.
Note:
On Windows 7 systems, the default folder for Manager .cfg files is C:
\Program Files (x86)\Avaya\IP Office\Manager. On some Windows 7
systems, the file is saved to the user's profile folder at C:\Users
\<user_name>\AppData\Local\VirtualStore\Program Files (x86)\Avaya\IP
Office\Manager. You must turn on Show hidden files to access this folder.
Alternatively, you can set the working directory to an alternate location.
Binary Directory (.bin files)
Sets the directory in which the Manager upgrade wizard, HTTP, TFTP and
BOOTP functions look for firmware files requested by phones and other
hardware components. That includes .bin file, .scr files and .txt files. By
default this is the Manager application's program directory.
Tip:
In the Upgrade Wizard, right-clicking and selecting Change Directory also
changes this setting.
Warning:
Historically, by default the Working Directory and Binary Directory are
the same. This is deprecated as it potentially allows remote TFTP/HTTP file
access to the folder containing copies of configuration files. Therefore it is
recommended that either of the folders is changed to an alternate location.
Known Units File
Sets the file and directory into which Manager can record details of the systems
it has discovered. Once a file location has been specified, a Known Units button
becomes available on the discovery menu used for loading system configuration.
Pressing that button displays the known units file as a list from which the
required system can be selected. It also allows sorting of the list and records to
be removed.
Related links
File | Preferences on page 51
File | Preferences | Discovery
These settings affect the Select IP Office menu used by Manager to discovery systems.
Setting
Description
TCP and HTTP
Discovery
Default = On.
NIC IP/NIC Subnet
This area is for information only. It shows the IP address settings of the LAN network
interface cards (NIC) in the PC running Manager. Double-click on a particular NIC to
add the address range it is part of to the IP Search Criteria. Note that if the address of
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This setting controls whether Manager uses TCP to discover systems. The addresses
used for TCP discovery are set through the IP Search Criteria field below.
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File Menu
Setting
Description
any of the Manager PC's NIC cards is changed, the Manager application should be
closed and restarted.
IP Search Criteria
This section is used to enter TCP addresses to be used for the TCP discovery
process. Individual addresses can be entered separated by semi-colons, for example
135.164.180.170; 135.164.180.175. Address ranges can be specified using dashes,
for example 135.64.180.170 - 135.64.180.175.
UDP Discovery
Default = On
This settings controls whether Manager uses UDP to discover systems.
Enter Broadcast IP
Address
Default = 255.255.255.255
Use DNS
Selecting this option allows Manager to use DNS name (or IP address) lookup to
locate a system. Note that this overrides the use of the TCP Discovery and UDP
Discovery options above. This option requires the system IP address to be assigned
as a name on the users DNS server. When selected, the Unit/Discovery Address
field on the Select IP Office window is replaced by a Enter Unit DNS Name or IP
Address field.
SCN Discovery
If enabled, when discovering systems, the list of discovered systems will group
systems in the same Small Community Network and allow them to be loaded as a
single configuration. At least one of the systems in the Small Community Network
must be running Release 6.0 or higher software. See Small Community Network
Management. This does not override the need for each system in the Small
Community Network to also be reachable by the TCP Discovery and or UDP
Discovery settings above and accessible by the router settings at the Manager
location.
The broadcast IP address range that Manager should used during UDP discovery.
Since UDP broadcast is not routable, it will not locate systems that are on different
subnets from the Manager PC unless a specific address is entered.
Related links
File | Preferences on page 51
File > Preferences > Visual Preferences
Setting
Description
Icon size
Sets the size for the icons in the navigation pane between Small, Medium or Large.
Multiline Tabs
Default = Off.
In the details pane, for record types with more than two tabs, Manager can either use
buttons to scroll the tabs horizontally or arrange the tabs into multiple rows. This
setting allows selection of which method Manager uses.
Related links
File | Preferences on page 51
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Menu Bar Commands
File | Preferences | Security
Controls the various security settings of Manager. To control the security settings of the system, see
the information on Security mode.
All settings, except Secure Communications, can only be changed when a configuration has been
opened using a user name and password with Administrator rights or security administration rights.
Setting
Description
Request Login on
Save
Default = On
Close Configuration/
Security Settings
After Send
Default = On.
By default a valid user name and password is required to receive a configuration
from a system and also to send that same configuration back to the system.
Deselecting this setting allows Manager to send the configuration back without
having to renter user name and password details. This does not apply to a
configuration that has been saved on PC and then reopened. This setting can only
be changed when a configuration has been opened using a user name and
password with Administrator rights or security administration rights.
When selected, the open configuration file or security settings are closed after
being sent back to the system. This is the normal default. This setting does not
affect multi-site network modes of Manager which always close the configuration
after saving.
Before disabling this setting, you should recall that the configuration held by a
running system can be changed by actions other than Manager, for example
changes made by users through their phone. Keeping a configuration open in
Manager for longer than necessary increases the chances that that copy of the
configuration differs from the current configuration of the running system and will
overwrite those changes when sent back to the system.
Save Configuration
File After Load
Default = Off.
Backup Files on
Send
Default = Off.
Backup File
Extension
Default = .BAK
Number of Backup
Files to keep
Default = Unlimited.
When selected, a copy of the configuration is saved to Manager's working
directory. The file is named using the system name and the suffix .cfg. This setting
can only be changed when a configuration has been opened using a user name
and password with Administrator rights or security administration rights.
If selected, whenever a copy of a configuration is sent to a system, a backup copy
is saved in Manager's working directory. The file is saved using the system name,
date and a version number followed by the Backup File Extension as set below.
This setting can only be changed when a configuration has been opened using a
user name and password with Administrator rights or security administration rights.
Sets the file extension to use for backup copies of system configurations
generated by the Backup Files on Send option above.
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File Menu
Setting
Description
This option allows the number of backup files kept for each system to be limited. If
set to a value other then Unlimited, when that limit would be exceeded, the file
with the oldest backup file is deleted.
Enable Application
Default = On.
Idle Timer (5 minutes)
When enabled, no keyboard or mouse activity for 5 minutes will cause the
Manager to grey out the application and re-request the current service user
password. This setting can only be changed when a configuration has been
opened using a user name and password with Administrator rights or security
administration rights.
Secure
Communications
Default = On
When selected, any service communication from Manager to the system uses the
TLS protocol. This will use the ports set for secure configuration and secure
security access. It also requires the configuration and or security service within the
system's security configuration settings to have been set to support secure access.
Depending on the level of that secure access selected, it may be necessary for the
Manager Certificate Checks below to be configured to match those expected by
the system for configuration and or security service. See Security Administration.
• When Secure Communications is set to On, a
all times in the lower right Manager status field.
padlock icon is displayed at
• For Server Edition systems, Manager will always attempt to use secure
communications regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds,
Manager will offer to fallback to using unsecured communications.
Manager Certificate
Checks
When the Secure Communications option above is used, Manager will process
and check the certificate received from the system. This setting can only be
changed when a configuration has been opened using a user name and password
with Administrator rights or security administration rights. The options are:
• Low: Any certificate sent by the system is accepted.
• Medium: Any certificate sent by the system is accepted if it has previously been
previously saved in the Windows' certificate store. If the certificate has not been
previously saved, the user has the option to review and either accept or reject
the certificate.
• High: Any certificate sent by the system is accepted if it has previously been
previously saved in the Windows' certificate store. Any other certificate cause a
log in failure.
Certificate Offered to
IP Office
Default = none Specifies the certificate used to identify Manager when the Secure
Communications option is used and the system requests a certificate. Use the Set
button to change the selected certificate. Any certificate selected must have an
associated private key held within the store:
• Select from Current User certificate store - Display certificates currently in the
currently logged-in user store.
• Select from Local Machine certificate store.
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Menu Bar Commands
Setting
Description
• Remove Selection – do not offer a Manager certificate.
:
Security – Registry Settings
Warning:
Avaya accept no liability for any issues arising from the editing of a PC’s registry settings. If you
are in any doubt about how to perform this process you should not proceed. It is your
responsibility to ensure that the registry is correctly backed up before any changes are made.
Note:
Before manually editing any registry entry, the following Microsoft support articles should be
read:
• http://support.microsoft.com/kb/256986
• http://www.microsoft.com/resources/documentation/windows/xp/all/proddocs/en-us/
regedit_permit_key.mspx
Manager stores it’s security preferences in the Windows Registry. The following key affects
manager security operation; it’s values may only be changed by a configuration or security
administrator:
HKEY_CURRENT_USER\Software\Avaya\IP400\Manager\Security\
In order to prevent circumvention by manual editing of the Windows Registry, Regedt32.exe, the
native registry editor, allows an operator user (with Full Control permissions) to edit permissions on
a per key basis.
To prevent a user from manually editing the security preferences, the HKEY_USERS\User GUID
\Software\Avaya\IP400\Manager\Security key permission should be set to ‘Read’ only for
that user. Ensure that all child object permissions are replaced as well by using the ‘Advanced’
button.
To allows the security policy of all local PC users to be fixed, a set of values in the key
HKEY_CURRENT_USER\Software\Avaya\IP400\Manager\Security\ may be created. This is
tested and used in preference to any value found under HKEY_CURRENT_USER\Software\Avaya
\IP400\Manager\Security\.
This key is not created by the manager application.
Related links
File | Preferences on page 51
File | Preferences | Validation
By default Manager validates the whole configuration when it is loaded and individual fields
whenever they are edited. This tab allows selection of when automatic validation should be applied
to configuration files loaded into Manager.
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File Menu
Setting
Description
Validate
configuration on
open
Automatically validate configuration files when they are opened in Manager.
Validate
configuration on edit
Validate the whole configuration when OK is clicked after editing a record. For large
configurations, disabling this option removes the delay caused by validating the
configuration after every edit.
Prompt for
configuration
validation on save or
send
If selected, when saving or sending a configuration, a prompt is displayed asking
whether the configuration should be validated. If validation is selected and error are
found, the send or save process is canceled. This option is disabled if Validate
configuration on edit is selected.
Related links
File | Preferences on page 51
File | Offline
Related links
File Menu on page 48
File > Offline > Create New Config on page 59
File | Offline | Open File on page 59
Open File Set on page 59
File | Offline | Send Config on page 60
File | Offline | Receive Config on page 60
File > Offline > Create New Config
Used to create an offline configuration.
This command starts a dialog that allows you to create a default offline configuration by specifying
the system locales, the type of control unit and expansion modules and the trunk cards fitted. The
same action is performed by the
icon in the Main Toolbar.
Related links
File | Offline on page 59
File | Offline | Open File
This command allows a configuration file stored on PC to be opened in Manager.
Related links
File | Offline on page 59
Open File Set
This command is only available when manager is running in Server Edition mode. It can be used to
load a set of files previously saved offline using the File | Save Configuration As command.
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Menu Bar Commands
When selected, browse to the location of the saved .cfi file and associated .cfg files and select
the .cfi file.
Related links
File | Offline on page 59
File | Offline | Send Config
This command is used to send an offline configuration to a system.
Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
After sending the configuration, you should receive the configuration back from the system and note
any new validation errors shown by Manager. For example, if using Embedded Voicemail, some
sets of prompt languages may need to be updated to match the new configurations locale setting
using the Add/Display VM Locales option.
Related links
File | Offline on page 59
File | Offline | Receive Config
This command displays the Select IP Office menu used to receive a systems configuration settings.
Once the configuration has been received, you are prompted to save it on the PC.
Related links
File | Offline on page 59
File | Advanced
Related links
File Menu on page 48
File | Advanced | Erase Configuration on page 61
File | Advanced | Reboot on page 61
File | Advanced | System Shutdown on page 62
File | Advanced | Upgrade on page 63
File | Advanced | Change Mode on page 65
File | Advanced | Switch to Standard Mode on page 66
File > Advanced > Audit Trail on page 67
File | Advanced | Security Settings on page 68
File | Advanced | Erase Security Settings (Default) on page 68
File | Advanced | Embedded File Management on page 68
File | Advanced | Format IP Office SD Card on page 69
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File Menu
File > Advanced > Recreate IP Office SD Card on page 70
File | Advanced | Memory Card Command on page 72
File | Advanced | Launch Voicemail Pro on page 72
File | Advanced | System Status on page 72
File | Advanced | LVM Greeting Utility on page 73
File | Advanced | Initial Configuration on page 73
File | Advanced | Add/Display VM Locales on page 77
File | Advanced | Erase Configuration
This command returns the configuration settings of a system back to their default values. It does not
affect the system's security settings or audit trail record.
When this command is used, the Select IP Office menu is displayed. Once a system is selected, a
valid configuration user name and password are required to complete the action.
IP500 V2 systems using IP Office A-Law or IP Office U-Law System SD cards will default to Quick
mode. Loading the configuration will switch Manager to simplified view. To change the system back
to operating in Standard mode, use either of the following methods:
• Change Mode
This will change the operating mode of the system and create a default configuration
appropriate to that mode. For example, this method can be used to change a Standard mode
system to a Basic Edition system.
• Switch to Standard Mode
This option (only shown in Manager simplified view) will change the operating mode of a Basic
Edtion system to Standard mode.
For systems running in Server Edition mode, this command can normally only be used when
Manager is also running in Server Edition mode.
Related links
File | Advanced on page 60
File | Advanced | Reboot
When this command is used, the Select IP Office window is displayed. Once a system is selected,
a valid user name and password are required. The type of reboot can then be selected in the
Reboot window.
When the reboot occurs can be selected as follows:
• Immediate Send the configuration and then reboot the system.
• When Free Send the configuration and reboot the system when there are no calls in progress.
This mode can be combined with the Call Barring options.
• Timed The same as When Free but waits for a specific time after which it then wait for there to
be no calls in progress. The time is specified by the Reboot Time. This mode can be combined
with the Call Barring options.
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Menu Bar Commands
Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time for
the reboot. If the time is after midnight, the system's normal daily backup is canceled.
Call Barring These settings can be used when the reboot mode When Free is selected. They bar
the sending or receiving of any new calls.
Related links
File | Advanced on page 60
File | Advanced | System Shutdown
This command can be used to shutdown systems. The shut down can be either indefinite or for a
set period of time after which the system will reboot. For Linux based telephone systems, the
shutdown command is applied to the telephony service on the server and not to the whole sever. In
that case, if the system is shutdown indefinitely, it can be restarted using the server's web control
pages to either restart the service or to restart the whole server.
Warning:
A shutdown must always be used to switch off the system. Simply removing the power cord or
switching off the power input may cause the loss of configuration data.
This is not a polite shutdown, any user calls and services in operation will be stopped. Once
shutdown, the system cannot be used to make or receive any calls until restarted.
The shutdown process takes up to a minute to complete. When shutting down a system with a
Unified Communications Module installed, the shutdown can take up to 3 minutes while the card
safely closes all open files and closes down its operating system. During this period the module's
LED 1 remains green.
When shutdown, the LEDs shown on the system are as follows. Do not remove power from the
system or remove any of the memory cards until the system is in this state:
• LED1 on each IP500 base card installed will also flash red rapidly plus LED 9 if a trunk
daughter card is fitted to the base card.
• The CPU LED on the rear of the system will flash red rapidly.
• The System SD and Optional SD memory card LEDs on the rear of the system are
extinguished.
To restart a system when shutdown indefinitely, or to restart a system before the timed restart,
switch power to the system off and on again.
Once you have selected the system from the Select IP Office window, the System Shutdown Mode
window opens. Select the type of shutdown required:
• If a Timed shutdown is selected, the system will reboot after the set time has elapsed.
• If Indefinite is used, the system can only be restarted by having its power switched off and
then on again. For Linux based telephone systems, the telephony service must be restarted
through the server's web control pages.
Related links
File | Advanced on page 60
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File Menu
File | Advanced | Upgrade
Note:
This command is not used with Linux based systems. Linux systems are updated through the
server's webcontrol menus.
This command starts the Upgrade Wizard tool. The Upgrade Wizard is used to compare the
software level of the control unit and modules within systems against the software level of the .bin
binary files Manager has available. The Upgrade Wizard can then be used to select which units to
upgrade.
Warning:
• Incorrect use of the upgrade command can halt system operation and render units in the
system unusable. You must refer to the Technical Bulletins for a specific release for full
details of performing software upgrades to that release. There may be additional steps
required such as defaulting the security settings.
• Performing any other actions on a system during an upgrade or closing the upgrade wizard
and Manager during an upgrade may render systems unusable.
• During an upgrade the system may restrict calls and services. It will reboot and disconnect
all current calls and services.
• The Validate option must remain selected wherever possible. Use of unvalidated upgrades
is subject to a number of conditions outlined in the IP Office Installation Manual and
Technical Bulletins.
The list area shows details of systems found by the Upgrade Wizard and the software currently held
by those systems. The check boxes are used to select which units should be upgraded. Upgrading
will require entry of a valid name and password for the selected system.
Column
Description
Name
The name of the system as set in its configuration (System | System | Name) .
IP Address
The IP address of the system.
Type
The type of system and the names of the various firmware files used by external
expansion systems supported by the system type.
Version
Details the current software each unit in the systems is running.
Edition
Indicates the operation mode of the system.
Licensed
Indicates the highest value software upgrade license present in the system's
configuration. The IP Office Release that is supported by that license is also
indicated in brackets.
Required License
Indicates the software upgrade license required for the current level of software
the system is running. The IP Office Release that is supported by that license is
also indicated in brackets.
Table continues…
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Menu Bar Commands
Column
Description
It does not refer to the software upgrade license required for the level of software
which is available for upgrade. The system must include a license for the specific
level of software it is required to run.
For IP500 V2 systems, a value of 255 indicates that the control unit is still in its
initial 90 days where it can be upgraded to a higher level without requiring an
upgrade license.
Available
Shows the version of the matching firmware files that Manager has available (a –
indicates no file available) in its current working directory. Upgrading to a release
higher than that supported by the current Licensed level will leave the system
unable to support any functions until the appropriate upgrade license is added to
the system configuration.
The Upgrade Wizard includes a number of check boxes that can be used to include other actions as
part of the upgrade process:
• Validate
• The Validate option should remain selected wherever possible. When selected, the upgrade
process is divided as follows: transfer new software, confirm transfer, delete old software,
restart with new software. If Validate is not selected, the old software is deleted before the new
software is transferred.
• Backup System Files
• For any IP500 V2 systems being upgraded, the Backup system files option will cause the
system to backup its memory card files as part of the upgrade.
• Upload System File
• For any IP500 V2 system being upgraded, the Upload system files option will upload various
files:
- It copies the binary files for the system control unit and possible external expansion
modules.
- It copies the firmware files used by phones supported by the system.
- For systems running Basic Edition, the files for Basic Edition Web Manager are copied.
- For systems configured to run Embedded Voicemail, the Embedded Voicemail prompts for
those supported languages set as the system locale, user locales, incoming call route
locales and short code locales are upgraded. In addition the English language prompts are
upgraded as follows: IP Office A-Law/Norstar SD Cards - UK English, IP Office U Law/
PARTNER SD Cards - US English.
• Restart IP Phones For Manager 8.1 and higher the Restart IP Phones option can be used.
This will cause those phone to load any upgrade phone firmware included in the system
upgrade (if using the system's memory card as their firmware file source).
Related links
File | Advanced on page 60
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File Menu
Searching for Systems
The default address used by the Upgrade Wizard is the address shown in the Manager title bar,
which is selected through File | Preferences. If the unit required is not found, the address used can
be changed.
Procedure
1. Enter or select the required address in the Unit/Broadcast Address field.
2. Click Refreshto perform a new search.
Changing the .bin File Directory
The directory in which the Upgrade Wizard looks for .bin files is set through Manager's Binary
Directory setting. This can be changed using Files | Change Working Directory or File |
Preferences | Directories.
Use this procedure to change it directly from the Upgrade Wizard.
Procedure
1. Right-click on the list area.
2. Select Select Directory.
3. Browse to and highlight the folder containing the .bin files. Click OK.
4. The list in the Available column will be updated to show the .bin files in the selected
directory that match units or modules listed.
File | Advanced | Change Mode
This command can be used to change the operating mode of an IP500 V2 System SD card and thus
of the system. For example, it can be used to switch a system currently running Standard Mode to
Basic Mode.
Important:
Using this command will default the configuration. Therefore ensure that you have a backup
copy of the configuration before using this command in case it is necessary to return to the
previous mode.
Do not use this command if the system includes components not supported by the mode to which
you want to switch. If that is the case, the system may not restart correctly. For example, BRI cards
are not supported by Basic Edition Partner mode.
In order to use this command, the system security settings must be at their default settings. The
current setting can be defaulted using the Erase Security Settings (Default) command.
Follow the command, the system is restarted.
For an IP500 V2 system to run in Standard Mode, its configuration must include an Essential
Edition license. A Standard mode system without this license will not allow any telephony functions.
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Menu Bar Commands
For the mode change, you should perform a system upgrade or upload system files action as the
files included in any previous upgrade or upload will have been based on the system's previous
mode setting and configuration settings.
After a mode change, the system restarts. If the system does not restart, the most likely cause is
that the systems security settings were not at their default settings.
The window that opens after selecting a system will indicate the modes available.
Standard mode
Mode selection is only possible for systems fitted with IP Office A-Law or IP Office U-Law SD
cards. For systems fitted with IP Office Partner Edition or IP Office Norstar Edition SD cards, the
systems will default to the respective mode of that card regardless of the mode selection.
Note that this process does not change the formatted type of the System SD card. For example, if a
system fitted with an IP Office A-Law card has its mode changed to Standard mode, if that system
is then defaulted again it will restart as a Basic Edition system as appropriate to the card type.
Basic Edition
There is no mode selection. The system will be changed to a defaulted Standard mode configuration
Related links
File | Advanced on page 60
File | Advanced | Switch to Standard Mode
This option will change the operating mode of the configuration loaded in Manager to that of a
Standard Mode system. Manager will automatically switch to its advanced view mode. When the
configuration is sent back to the system, it will restart in Standard Mode.
For an IP500 V2 system to run in Standard Mode, its configuration must include an Essential
Edition license. A Standard Mode system without this license will not allow any telephony functions.
The command provides two options:
• Default Using this method will default the configuration. It is the recommended method for
installation of a new installation or for when a Standard Mode system has been defaulted and
needs to be returned to Essential Edition operation.
• Best Match Using this method will attempt to preserve configuration settings; for examples
user names, extension numbers, licenses, SIP trunks, etc. However, many settings will be
flagged as errors by Manager. These errors should then be resolved before sending the
configuration to the system.
If this is an existing system, it is recommended that you first use Manager to receives and save a
copy of the current configuration locally using Save Configuration As. This process does not
default the security settings of the system.
Do not use this command if the system includes components not supported by the Essential Edition.
(currently IP500 ETR6 base cards for ETR phones). The system may not restart correctly if that is
the case.
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File Menu
When this command is selected, Manager will first browse for available systems. When a system is
selected from those found, load its configuration. If this cannot be done using the default password
(password) it may not be possible to complete the process.
! Automatic Conversion to Standard Mode This process can be applied automatically when a
configuration for a new or defaulted system running in Basic Edition — Quick mode is loaded. This
is done by selecting the Default to Standard Mode option in the Manager Preferences. Only select
this option if the only systems you expect to install are Standard mode systems.
Related links
File | Advanced on page 60
File > Advanced > Audit Trail
The audit trail lists the last 16 actions performed on the system from which the configuration loaded
into Manager was received. It includes actions by service users such as sending a configuration
back, reboots, upgrades and defaulting the system.
Audit trail events can be output to a Syslog server through the system's System | System Events
settings.
The last failed action is always recorded and shown in red. It is kept even if there have been 16
subsequent successful actions.
The Audit Trail is part of the system configuration file received from the system. If the configuration
is kept open between send and reboot operations (ie. if Close Configuration/Security Setting After
Send is not selected), the Audit Trail will not show details of those operations. It will only show
details of those operations if the configuration is closed and then a new copy of the configuration is
received from the system.
Audit Details
When a specific access event is selected from the list, the following information is shown in the
Audit Details section:
• The Security User shows the service user name used for the access action.
• The Data and Time of Access indicate the local system time when the recorded event
occurred.
• The PC Login is the computer name of the PC used for the access.
• The PC IP Address and PC MAC Address are the IP address and MAC address of the PC
used for access.
• The Access Type details the type of action that was performed.
• The Outcome shows the system's response to the access. The outcome Success (Warning)
refers to the sending of a configuration that contains fields marked as errors or warnings by
Manager's validation function. Success (Clean) refers to the sending of a configuration that
does not contain any validation errors or warnings.
Items Changed
The Items Changed area summarizes the changes contained in a sent configuration. Where
changes to a single record of a particular type are made, the Item Name field lists the individual
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Menu Bar Commands
record changed. Where changes are made to several records of the same type, the Item Name field
displays Multiple items.
Related links
File | Advanced on page 60
File | Advanced | Security Settings
This command is used to switch the Manager application to security mode. In that mode, Manager is
used to edit the security settings of a system.
Related links
File | Advanced on page 60
File | Advanced | Erase Security Settings (Default)
This command returns the security settings of a system back to their default values. This action
does not affect the system's configuration or audit trail record. When this command is used, the
Select IP Office menu is displayed. Once a system is selected, a valid security user name and
password are required to complete the action.
The system's security settings are returned to their defaults as indicated in the Security Mode
chapter.
Note that any security certificates stored and being used by the system are deleted. Any services
currently using those certificates are disconnected and disabled until the appropriate certificates are
added back to the system's security configuration. That includes SSL VPN connections being used
to perform system maintenance.
For Standard mode and Server Edition systems, the name and password used for this command
are those required for security configuration access which are different from those used for normal
configuration access.
For Basic Edition systems, the name and password required are those of the Administrator
account used for configuration access.
For IP500 V2 control units, if the security settings cannot be defaulted using this command, they can
be defaulted using a DTE cable connection to the system. Refer to the IP Office Installation manual
for details.
Related links
File | Advanced on page 60
File | Advanced | Embedded File Management
For control units with a memory card installed, the contents of the card can be viewed using
Manager. This view can also be used to add and remove files from the card. This may be useful
when the memory card is being used to store Music on Hold or IP phone firmware files.
For Linux based systems, the folder /opt/ipoffice is used as the file repository for embedded
file management actions.
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File Menu
Related links
File | Advanced on page 60
File | Advanced | Format IP Office SD Card
This command allows suitable SD cards to be formatted by the Manager PC. The system supports
SD cards with the following format: SDHC minimum 4GB FAT32 format (Single partition, SDHC,
class2+, FAT32, SPI & SD bus). Non-Avaya supplied cards of the same format can be used a
system's Optional SD slot for additional actions such as backup.
Warning:
• Do not re-purpose a Enterprise Branch SD card for use with any other IP Office mode.
Doing so may damage the SD card and make it unusable for your Enterprise Branch
system.
• All File Will Be Erased Note that this action will erase any existing files and folders on the
card. If the requirement is just to update the card, use Recreate IP Office SD Card without
reformatting. Once a card has been formatted, the folders and files required for operation
can be loaded onto the card from the Manager PC using the Recreate IP Office SD Card
command.
• Avaya supplied SD cards should not be formatted using any other method than the format
commands within Manager and System Status Application. Formatting the cards using any
other method will remove the feature key used for system licensing from the card.
Related links
File | Advanced on page 60
Formating the SD card on page 69
Formating the SD card
Procedure
1. Insert the SD card into a reader slot on the Manager computer.
2. Using Manager, select File | Advanced | Format IP Office SD Card.
3. Select the type of card.
This selection just sets the card label shown when viewing the card details. It does not affect
the actual formatting. Select the label that matches the file set you will be placing on the
card.
• IP Office A-Law A system fitted with this type of card will default to A-Law telephony. The
system will default to IP Office Basic Edition - Quick Mode PBX System operation.
• IP Office U-Law A system fitted with this type of card will default to A-Law telephony. The
system will default to IP Office Basic Edition - Quick Mode Key System operation.
• IP Office Partner Edition A system fitted with this type of card will default to A-Law
telephony and IP Office Basic Edition - PARTNER® Mode operation.
• IP Office Norstar Edition A system fitted with this type of card will default to U-Law
telephony and IP Office Basic Edition - Norstar® Mode operation.
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Menu Bar Commands
• Enerprise Branch Use this option for an SD card intended to be used with an IP Office
system running in Enterprise Branch Mode. There is a separate SD card for Enterprise
Branch. The Enterprise Branch SD card can only be used for Enterprise Branch operation
and cannot be used to change modes to IP Office. You also cannot use or change an IP
Office SD card for use with an Enterprise Branch system.
Warning:
Do not re-purpose a Enerprise Branch card for use with any other IP Office mode.
Doing so may damage the SD card and make it unusable for your Enterprise Branch
system.
4. Browse to the card location and click OK.
5. The status bar at the bottom of Manager will display the progress of the formatting process.
6. When the formatting is complete, you can use the Recreate IP Office SD Card command to
load the system folders and files onto the card from the Manager PC.
Related links
File | Advanced | Format IP Office SD Card on page 69
File > Advanced > Recreate IP Office SD Card
This command can be used with the System SD cards used by IP500 V2 control units. It allows
Manager to copy all the files and folders used by a system when starting onto the card that has
been place into the card slot of the PC running Manager. It updates the card with the version of
those files installed with the Manager application. It includes the binary files for the system, external
expansion modules and phones. The command also copies all language prompt sets used by
Embedded Voicemail.
If the card contains dynamic system files such as SMDR records, they are temporarily backed up by
Manager and then restored after the card is recreated. For the card to be used in a system's
System SD slot the card must be Avaya SD Feature Key card. The card must be correctly
formatted, however a reformat of an existing working card is not necessary before using recreate to
update the card contents.
The source for the files copied to the SD card are the sub-folders of the \Memory Cards folder
under Manager's Working Directory (normally C:\Program Files\Avaya\IPOffice
\Manager). However, if the Working Directory is changed to a location without an appropriate set of
\Memory Cards sub-folders, the required set of files will not be copied onto the SD card.
Related links
File | Advanced on page 60
Recreating the IP Office SD Card on page 71
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File Menu
Recreating the IP Office SD Card
About this task
Note:
This process can take up to 20 minutes depending on the PC. Once started, the process should
not be interrupted.
Procedure
1. Insert the SD card into a reader slot on the Manager computer.
2. Using Manager, select File | Advanced | Recreate IP Office SD Card.
3. Select the type of system for which the card is intended.
This selection will affect how the system operates when defaulted with this card present in its
System SD card slot.
• IP Office A-Law A system fitted with this type of card will default to A-Law telephony. The
system will default to IP Office Basic Edition - Quick Mode PBX System operation.
• IP Office U-Law A system fitted with this type of card will default to ULAW telephony. The
system will default to IP Office Basic Edition - Quick Mode Key System operation.
• IP Office Partner Edition A system fitted with this type of card will default to A-Law
telephony and IPOffice Basic Edition - PARTNER® Mode operation.
• IP Office Norstar Edition A system fitted with this type of card will default to U-Law
telephony and IPOffice Basic Edition - Norstar® Mode operation.
• Enterprise Branch Use this option for an SD card intended to be used with an IP Office
system running in Enterprise Branch mode. There is a separate SD card for IP Office. The
Enterprise Branch SD card can only be used for IP Office operation and cannot be used
tochange modes to IP Office. You also cannot use or change an IP Office SD card for use
withan Enterprise Branch system.
Warning:
Do not re-purpose a Enterprise Branch SD card for use with any other IP Office mode.
Doing so may damage the SD card and make it unusable for your Enterprise Branch
system.
4. Browse to the card location and click OK.
5. Manager will prompt whether you want to include Basic Edition Web Manager files as part of
the recreate process.
6. For systems that will be running in Basic Edition mode, these files are necessary if you want
to use Web Manager to configure the system.
7. For all systems, these files are necessary if you want to go through the process of onboarding registration.
8. Manager will start creating folders on the SD card and copying the required files into those
folders.
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Menu Bar Commands
9. Do not remove the card until the process is completed and Manager displays a message that
the process has been completed.
Related links
File > Advanced > Recreate IP Office SD Card on page 70
File | Advanced | Memory Card Command
These commands are used with the memory cards installed in IP500 V2 control units.
Shutdown
This command can be used to shutdown the operation of IP500 V2 unit memory cards.
This action or a system shutdown must be performed before a memory card is removed from the
unit. Removing a memory card while the system is running may cause file corruption. Card services
can be restarted by either reinserting the card or using the Start Up command.
Shutting down the memory card will disable all services provided by the card including Embedded
Voicemail if being used. Features licensed by the memory card will continue to operate for up to 2
hours.
Start Up
This command can be used to restart operation of an IP500 V2 memory card that has been shut
down. >The command will start the Select IP Office discovery process for selection of the system.
Related links
File | Advanced on page 60
File | Advanced | Launch Voicemail Pro
If the Voicemail Pro client is installed on the same PC as Manager, this link can be used to launch
the Voicemail Pro client. This can also be done by clicking on the
icon in the Manager toolbar.
Related links
File | Advanced on page 60
File | Advanced | System Status
System Status is an application that can be used to monitor and report on the status of a system.
This is a separate application from Manager but if installed on the same PC, it can be started using
the File | Advanced | System Status link within Manager. Use of the application requires a service
user name and password configured on the system for System Status Access within the system's
security settings.
Related links
File | Advanced on page 60
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File Menu
File | Advanced | LVM Greeting Utility
This command launches a utility that can be used to convert .wav files to the formats used by
Embedded Voicemail (c11). The source file must be in the standard format used for all system
applications: PCM, 8kHz 16-bit, mono.
The resulting named greeting files can then be transferred to the Embedded Voicemail memory card
and selected as auto attendant greetings. That is done using the Recording Name field on the Auto
Attendant | Auto Attendant tab. The same named greeting file can be used in several auto
attendants.
The utility can be run separately using the file LVMGreeting.exe found in the LVMGreeting subfolder of the Manager application.
Note:
The LVM Greeting Utility option is not selectable (grayed out) when Voicemail Pro is selected as
the system's voicemail type.
Related links
File | Advanced on page 60
File | Advanced | Initial Configuration
Note:
The Initial Configuration utility changes the security settings. Therefore, the user running the
utility must have security read/write rights.
Basic Mode and Standard Mode Initial Configuration
The Initial Configuration menu is displayed for all new or fully defaulted IP500 V2 systems. It allows
the required operating mode for the system to be selected.
.
For Quick Mode, Partner Mode and Norstar Mode, leave the selection set to Basic Mode.
For a system that you want to run in Essential Edition, Preferred Edition or Advanced Edition
modes, select IP Office Standard Mode.
For an IP500 V2 system to run in Standard Mode, its configuration must include an Essential
Edition license. A Standard Mode system without this license will not allow any telephony functions.
For a system that is being installed as an expansion server for a Server Edition solution, select
Server Edition Expansion.
Server Edition Initial Configuration
On a Basic or Standard Mode system, use the Initial Confguration option to convert the existing
system configuration into a Server Edition system configuration. It will effectively default the
configuration and reload it in Manager in Server Edition mode. Once Server Edition Expansion is
selected as the System Type, the Initial Configuration menu is displayed. If Server Edition
Expansion is selected in that menu, following selection of the various menu options, the system is
rebooted as a Expansion System (V2) for a Server Edition network.
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Menu Bar Commands
For systems being configured for operation in a Server Edition solution, the Initial Configuration
menu is used to set or confirm a range of settings. The field shown and accessible in the form
depend on the selected System Type.
Once the menu is completed and Save is clicked, the values entered are written into the system
configuration and the system is restarted. The menu is also displayed when creating an offline
configuration for a Server Editionsystem. The configuration of an existing non-Server Edition
system can be converted to a Server Edition configuration, invoking this menu, using the File |
Advanced | Initial Configuration menu option.
System Type Indicate the type of sever role the system will perform.
Retain Configuration Data This option is shown for IP500 V2 units being converted to become
Expansion System (V2)s in a Server Edition solution.
If left unselected, the default, the existing configuration of the system is defaulted as per a standard
Server Edition expansion system.
If selected, the existing configuration is retained. However, some elements of that configuration may
be invalid or ignored in a Server Edition solution. It is the installers responsibility to ensure that the
final configuration is valid for use in the solution. See IP500 V2 Conversion.
Option
Description
Name
A name to identify this system. This is typically used to identify the configuration by the
location or customer's company name. Some features such as Gatekeeper require the
system to have a name. This field is case sensitive and within any network of systems
must be unique. Do not use <, >, |, \0, :, *, ?, . or /.
Locale
This setting sets default telephony and language settings based on the selection. It also
sets various external line settings and so must be set correctly to ensure correct
operation of the system. See Supported Country and Locale Settings. For individual users
the system settings can be overridden through their own locale setting (User | User |
Locale).
Services Device ID Set a Device ID for the system. This ID is displayed on the Solution View and System
Inventory pages and on the System | System tab in the configuration. The value can be
changed using the Device ID field on the System | System Events | Configuration tab. If
an SSL VPN is configured, , Avaya recommends that the Device ID match an SSL VPN
service Account Name. Each SSL VPN service account name has an associated SSL
VPN tunnel IP address. Having the displayed Device ID match an SSL VPN service
account name helps identify a particular SSL VPN tunnel IP address to use for remotely
managing IP Office.
LAN Interface
This IP Address, IP Mask, Gateway and DHCP Mode settings can be set for the systems
two LANs, LAN1 and LAN2. These radio buttons are used to switch between displaying
the LAN1 details or the LAN2 details.
IP Address
LAN1 Default = 192.168.42.1. LAN2 Default = 192.168.43.1.
This is the IP address of the Control Unit on LAN1. If the control unit is also acting as a
DHCP server on the LAN, this address is the starting address for the DHCP address
range.
IP Mask
Default = 255.255.255.0. This is the IP subnet mask used with the IP address.
Table continues…
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File Menu
Option
Description
Gateway
The address of the default gateway for routing traffic not in the same subnet address
range of the IP Address/IP Mask set above. A default IP Route for this address is added
to the systems configuration.
DHCP Mode:
Default = Server.
This controls the control unit's DHCP mode for the LAN. When doing DHCP:
• LAN devices are allocated addresses from the bottom of the available address range
upwards.
• Dial In users are allocated addresses from the top of the available range downwards.
• If the control unit is acting as a DHCP server on LAN1 and LAN2, Dial in users are
allocated their address from the LAN1 pool of addresses first.
• Server When this option is selected, the system will act as a DHCP Server on this LAN,
allocating address to other devices on the network and to PPP Dial in users.
• Disabled When this option is selected, the system will not use DHCP. It will not act as a
DHCP server and it will not request an IP address from a DHCP server on this LAN.
• Dial In When this option is selected, the system will allocate DHCP addresses to PPP
Dial In users only. On systems using DHCP pools, only addresses from a pool on the
same subnet as the system's own LAN address will be used.
• Client When this option is selected, the system will request its IP Address and IP Mask
from a DHCP server on the LAN.
Server Edition
Primary Server
The IP address of the Primary Server. This address is used to add an IP line to the
Primary Server to the configuration.
Server Edition
Secondary Server
The IP address of the Secondary Server. This address is used to add an IP line to the
Secondary Server to the configuration.
DNS Server
This is the IP address of a DNS Server. If this field is left blank, the system uses its own
address as the DNS server for DHCP client and forwards DNS requests to the service
provider when Request DNS is selected in the service being used (Service | IP).
Enterprise Branch Initial Configuration
The Initial Installation utility provides a default configuration and security settings that minimize initial
installation activities and maximize security. The system must be configured with the default settings
before the system can be administered by System Manager. This utility is used for new installations
and after an upgrade to enable System Manager administration of the IP Office.
1. Select File > Advanced > Launch Initial Installation Utility.
2. In the System Name field, enter the appropriate system name.
3. For the WAN Interface, select LAN1 or LAN2. If you select LAN1, the DHCP Mode is
disabled.
4. In the IP Address field, enter the appropriate IP address.
5. In the IP Mask field, enter the appropriate IP mask.
6. In the Gateway field, enter the appropriate gateway. Manager will create an IP route using
this gateway with the selected WAN as the destination.
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Menu Bar Commands
7. In the DHCP Mode section, if you selected LAN1, select the appropriate DHCP option. If you
selected LAN2, DHCP Mode is disabled.
8. Select the Under Centralized Management? check box if you want the IP Office system to
be managed by System Manager.
9. If you selected the Under Centralized Management? check box, a number of additional
fields are shown, configure these additional fields as appropriate:
• SMGR Address - the IP address of the server running System Manager
• SNMP Community
• SNMP Device ID
• Trap Community
• SCEP Domain Certificate Name
• Certificate Enrollment (SCEP) Password
Select Save.
When you run the Initial Installation Utility, the Initial Installation utility also configures the following:
• System Status Interface (SSA) service security level – Unsecure only
• Configuration service security level –Secure, Medium
• Security Administration service security level – Secure, Medium
• OAMP Web Services service security level – Secure, Low (if locally administered)
• OAMP Web Services service security level – Secure, High (if administered by System
Manager)
• Admin Client Certificate checks:-- High (if administered by System Manager)
• SCEP client active (if administered by System Manager)
• SCEP server IP address from SMGR IP address (if administered by System Manager)
• Legacy Program Code – Active (if locally administered)
If the system is administered by System Manager, the following is automatically configured:
• SNMP enabled
• SNMP trap destination 1 from System Manager IP address
• All SNMP traps active
• WebLM client active
• WebLM service address from System Manager IP address
• Remove all default extension users, leaving “NoUser” and “RemoteManager”
Related links
File | Advanced on page 60
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File Menu
File | Advanced | Add/Display VM Locales
This option is only displayed displayed when the configuration from an IP500 V2 systems with its
Voicemail Type set to Embedded Voicemail is received in Manager. It is not shown for off-line
configuration or configurations loaded from a PC file.
Selecting this option displays a list of the Embedded Voicemail prompt languages. Those languages
already present on the System SD card or not supported are greyed out. Additional languages can
be selected and then uploaded from Manager to the system.
When editing the system configuration in Manager, if the locale language selected for the system, a
user, a short code or an incoming call route is not already present on the System SD card, Manager
will display an error. Add/Display VM locales can then be used to upload the prompts for the
required language in order to correct the error.
You can reload languages that are already installed on the System SD card. For example, you may
want to reload the languages if new prompts have been added in a maintenance release. To reload
existing languages, upgrade the system (File | Advanced | Upgrade) with the Upload System Files
option checked. You can also choose Upload System Files from the Embedded File Management
utility (File | Advanced | Embedded File Management).
The Recreate IP Office SD Card command can be used to locally load all available languages onto
an SD card.
Related links
File | Advanced on page 60
File | Backup/Restore
Backup Binaries and Configurations
This command copies all configuration files (.cfg) and software binary files (.bin) stored in Manager's
working directory to a selected folder.
Restore Binaries and Configurations
This command copies all configuration files (.cfg) and software files (.bin) stored in a selected folder
to the Manager's working directory.
Related links
File Menu on page 48
File | Import/Export
Export
This command allows you to export the selected parts of the configuration to either a set of CSV text
files (.csv) or a single binary file (.exp).
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Menu Bar Commands
The display shows those exportable record types for which the configuration contains records. The
File Type and the Save In path can be selected at the base. The default location used is subdirectory of the Manager application directory based on system name of the currently loaded
system.
Manager imports and exports CSV files using UTF8 character encoding which uses a double byte to
support characters with diacritic marks such as ä. Other applications such as Excel, depending on
the user PC settings, may use different single-byte encoding which will cause such characters to be
removed. Care should be taken to ensure that any tool used to create or edit a CSV supports all the
characters expected and is compatible with UTF8.
Import
This command allows you to import configuration settings. Two formats are supported. Binary files
(.exp) are settings previously exported from a system using File | Import /Export | Export. CSV text
files (.csv) can also be exported from a system or can be created using a plain text editor.
For the selected File Type and the Look In path, the window displays the file or files found. The
default location used is sub-directory of the Manager application directory based on system name of
the currently loaded system.
Manager imports and exports CSV files using UTF8 character encoding which uses a double byte to
support characters with diacritic marks such as ä. Other applications such as Excel, depending on
the user PC settings, may use different single-byte encoding which will cause such characters to be
removed. Care should be taken to ensure that any tool used to create or edit a CSV supports all the
characters expected and is compatible with UTF8.
Related links
File Menu on page 48
File | Exit
The File | Exit command exits the Manager application.
Related links
File Menu on page 48
View Menu
Toolbars
Allows selection of which toolbars should be shown or hidden in configuration mode. A tick mark is
displayed next to the name of those toolbars that are currently shown.
Tooltip
Controls whether additional tooltips are displayed when Manager is running in simplified view mode.
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Tools Menu
Navigation Pane
Shows or hides the Navigation Pane. A tick mark appears next to the command when the pane is
shown.
Group Pane
Shows or hides the Group Pane. A tick mark appears next to the command when the pane is
shown.
Details Pane
Sets the location of the Details Pane when the Group Pane is also shown. The Details Pane can be
placed either below or to the right of the Group Pane.
Error Pane
Shows or hides the Error Pane. A tick mark appears next to the command when the pane is shown.
Advance View
Causes Manager to switch from its simplified view to advanced view mode. Manager automatically
switches to advanced view mode if a Standard Edition configuration is loaded.
Hide Admin Tasks
This settings shows or hides the Admin Tasks List available when Manager has a Basic Edition
configuration loaded.
Simplified View
If Manager has no configuration loaded, this command switches it from advanced view to simplified
view.
TFTP Log
This command displays the TFTP Log window. This window shows TFTP traffic between Manager
and devices that uses TFTP to send and receive files. For example, the TFTP Log below shows an
Avaya IP phone requesting and then being sent its software files.
Related links
Menu Bar Commands on page 48
Tools Menu
Related links
Menu Bar Commands on page 48
Tools | Extension Renumber on page 80
Tools | Line Renumber on page 80
Tools | Connect To on page 80
Tools | Export | User on page 81
Tools | SCN Service User Management on page 81
Tools | Busy on Held Validation on page 82
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Menu Bar Commands
Tools | MSN Configuration on page 82
Tools | Print Button Labels on page 83
Tools | Import Templates on page 84
Tools | Extension Renumber
This command allows the extension numbering of user extensions to be changed. The existing
extension number range to be adjusted can be specified followed by the new start point for the
range after renumbering.
The command does not alter the extension number used for hunt groups but does adjust the
extension numbers of hunt group members.
Related links
Tools Menu on page 79
Tools | Line Renumber
On external trunks Line appearance ID numbers can be assigned to each channel supported in
order to allow that channel or line to be associated with a Line Appearance button on phones that
support button programming. By default all lines are automatically numbered from 701 upwards
when added to the system. This command allows the lines to be renumbered from a different
starting point.
Related links
Tools Menu on page 79
Tools | Connect To
This option can be used to create H.323 IP line connections between two systems in a multi-site
network, one being the system with its configuration currently loaded in Manager, the other being
selected from a discovery dialog. This option is not available for Server Edition mode.
Important:
This process will require the systems to be rebooted.
Procedure
1. With the configuration of the first system received from that system and displayed in
Manager, clicking on
or Tools | Connect To
2. A discovery menu is displayed and will list any other systems discovered.
3. Select the system to which connection is required.
4. Enter the login name and password for configuration access to that system.
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Tools Menu
5. Manager will switch to Small Community Network management mode, displaying the
configuration of both systems.
6. Click
to save the new configuration back to each system.
Related links
Tools Menu on page 79
Tools | Export | User
When performing an upgrade from B5800 Branch Gateway 6.2 or IP Office 8.1 to IP Office 9.0,
users are not automatically created in System Manager. Before an upgrade, IP Office users must be
exported to a file and then the file must be imported to System Manager. This feature allows you to
export all users or selected users from a loaded configuration to an XML file that is then imported in
System Manager. The default filename is <SystemName>_Users.xml.
User Name – select this check box to export all users.
User Name/Extension – select the appropriate check boxes to export those users.
Related links
Tools Menu on page 79
Tools | SCN Service User Management
When managing multiple systems, it may be useful to create a common user name and password
on all the systems for configuration access. This tool can be used to create a new service user
account, SCN_Admin, for configuration access.
This process requires you to have a user name and password for security configuration access to
each of the systems.
Select Tools | SCN Service User Management.
The option is not shown if a Basic Mode system configuration is loaded. If no configuration is
loaded, and the option is not shown, select View | Advanced View.
Procedure
1. The Select IP Office menu displays the list of discovered systems.
2. Select the systems for which you want to create a common configuration account.
Click OK.
3. A user name and password for security configuration access to each system is requested.
Enter the values and click OK. If the same values can be used for all systems enter those
values, select Use above credentials for all remaining, selected IPOs. If each system
requires a different security user names and password, deselect Use above credentials for
all remaining, selected IPOs.
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Menu Bar Commands
4. The systems will be listed and whether they already have an SCN_Admin account is shown.
5. To create the SCN_Admin account on each system and set the password for those account
click on Create Service User.
6. Enter the common password and click OK.
7. The password can be changed in future using the Change Password option.
8. Click Close.
Related links
Tools Menu on page 79
Tools | Busy on Held Validation
Busy on Held is a user feature where, when the user has a call on hold, the system indicate the
user as being busy to any further calls.
The use of Busy on Held in conjunction with multiple call appearance buttons is deprecated. This
command can be used to identify those users who have multiple call appearance buttons and for
whom Busy on Held is currently set.
When run, it shows a list of the users affected and if selected their Busy on Held setting will be
switched off.
Related links
Tools Menu on page 79
Tools | MSN Configuration
Used to populate the Incoming Call Route table with a range of MSN or DID numbers.
Setting
Description
MSN/DID
The first number in the set of MSN numbers for which you have subscribed.
Note:
If you require to find an exact match between the MSN numbers and the
destination numbers, enter a minus (-) sign before the first MSN number.
Destination
Where incoming calls with matching digits should be routed. The drop-down list
contains the extensions and groups on the system.
Line Group ID
Specifies the incoming line group ID of the trunks to which the DID routing is
applied.
Presentation Digits
Set to match the number of digits from the MSN/DID number that the central office
exchange will actually present to the system.
Range
How many MSN or DID number routes to create in sequence using the selected
MSN/DID and Destination as start points. Only routing to user extensions is
supported when creating a range of records.
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Tools Menu
Related links
Tools Menu on page 79
Tools | Print Button Labels
This option is only enabled if a version of DESI software is also installed on the same PC as
Manager. It can then be used when a system configuration is loaded in Manager.
DESI software can be obtained from the Avaya support web site (http://support.avaya.com) or from
DESI (http//www.desi.com). Currently, though all users are shown, only ETR, M Series, T-Series,
1400 and 1600 phones are supported by DESI templates.
The text used on the labels:
• If a text label has been added in the user's Button Programming settings, that text label is
passed to the DESI application.
• Note that the DESI application cannot import non-ASCII characters and may render them
incorrectly.
• Manager will display a warning if it estimates that the user's current text for some buttons may
exceed the label space of the phone type.
• If no text label has been set, the default label for the action currently assigned to the button is
passed to the DESI application.
• Once the labels are shown in the DESI application, the label text can be changed.
1. Load the configuration of the system for which you want to print button labels.
2. Select Tools and then Print Button Labels.
• Name/Extn These are the user name and extension number details of the users in the system
configuration currently loaded in Manager.
• Phone Type This field shows the type of phone, if known, that the user is currently associated
with. The drop down can be used to change the selection if required.
• Expansion Modules If the phone type supports additional button modules, this drop down can
be used to select the type and number of button modules.
• Print Extn This check box is used to select whether the phone button details should be
included in the output passed to the DESI software.
• Print BM1/Print BM2/Print BM3 These check boxes are used to select whether button module
button details should be included in the output passed to the DESI software. These button will
only be selectable if the user's Expansion Modules is set to the number of button modules.
Click Print via DESIto transfer the information to the DESI application. Within DESI, edit the labels
as required and then print the labels.
Related links
Tools Menu on page 79
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Menu Bar Commands
Tools | Import Templates
Manager can be used to import and use trunk templates. SIP trunk templates can be used to add
SIP trunks. Analog trunk templates can also be applied to existing analog trunks. This option does
not affect the additional template options used for Server Edition mode.
The templates need to be stored in a specific Manager sub-folder \Templates.
Some templates may be supplied with the Manager application and will be automatically installed to
the correct location. This command can be used to select a folder containing other template files
and will copy those files into the correct Manager sub-folder.
Related links
Tools Menu on page 79
Security Mode Menus
These commands are available when the Manager is in security configuration mode.
Open Security Settings
Displays the Select IP Office menu to select and load a system's security settings. This requires
entry of a user name and password with rights to access security settings of the selected system.
This behavior changes when configuration settings have already be received from a system using a
service user name and password that also has security access rights for that system. In that case,
the system's security settings are automatically loaded without requiring name and password entry.
Close Security Settings
Close the currently open set of security settings received from a system without saving those
settings.
Save Security Settings
Send edited security settings back to the system. Requires re-entry of a service user name and
password with access rights for security settings.
Reset Security Settings
Reset the security settings of the selected system to defaults. Requires entry of a service user name
and password with access rights for resetting the security settings. This option is not usable while a
set of security configuration settings is loaded.
The command File | Advanced | Erase Security Settings (Default) performs the same action from
Manager configuration mode.
Preferences
Displays a window for configuring various aspects of Manager's operation. The window is divided
into a number of tabs.
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Embedded File Management Menus
For a description of the Preferences options, see File | Preferences.
Exit
This command closes Manager.
Configuration
Returns Manager to configuration mode.
Related links
Menu Bar Commands on page 48
Embedded File Management Menus
For control units with a memory card installed, the contents of the card can be viewed using
Manager. This view can also be used to add and remove files from the card. This may be useful
when the memory card is being used to store Music on Hold or IP phone firmware files. For Linux
based systems access to the /opt/ipoffice folder is provided.
Embedded Voicemail Files
When viewing the memory card, the files related to Embedded Voicemail are visible, however these
files are greyed out (ie. cannot be deleted, downloaded or overwritten).
• Mailbox greetings and messages are shown as .clp files.
• The language prompts for Embedded Voicemail functions are stored in separate language subfolders of lvmail. These are .c11 files.
• Named prompt files for use by Embedded Voicemail auto attendants are stored in the lvmail
\AAG folder and use the same .c11 or .c23 file formats as the language prompts. These files
can be created from standard .wav files before being downloaded to the memory card by using
the LVM Greeting Utility.
Avaya IP Phone Files
The memory card can be used as the source of files requested by IP Phones when rebooting. For
phones using system DHCP, once the files are loaded onto the card, the TFTP Server IP Address
and HTTP Server IP Address on the System | System tab must be set to match the system's LAN
address.
Viewing a Memory Card
When Advanced | Embedded File Management is selected, the Manager will go through normal
system discovery but will only allow selection of systems which can support a memory card. When a
system is selected, a valid service user name and password for configuration access to that system
is requested. If the system selected does not have a memory card installed, the files view remains
blank and the message TFTP:Received TFTP Error "Not Found" appears in Manager's status
bar.
Changing the Files View
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Menu Bar Commands
The type of display used in the Files pane can be changed by selecting from the View menu in the
toolbar.
Open File Settings
Select a system and display the contents of its memory cards if any are present and in use.
Close File Settings
Close the current memory card contents listing without exiting embedded file management mode.
Refresh File Settings
This command can be used to request a file update from the system.
Upload File
This command can be used to select and upload a file to the memory card in the system.
Upload System Files
This command is available with IP500 V2 systems. When this command is selected, Manager will
upload the software files for operation to the System SD card.
Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
• It copies the binary files for the system control unit and possible external expansion modules.
• It copies the firmware files used by phones supported by the system.
• For systems configured to running in Basic Edition mode, the files for Basic Edition Web
Manager are copied.
• For systems configured to run Embedded Voicemail, the Embedded Voicemail prompts for
those supported languages set as the system locale, user locales, incoming call route locales
and short code locales are upgraded. In addition the English language prompts are upgraded
as follows: IP Office A-Law/Norstar SD Cards - UK English, IP Office U Law/PARTNER SD
Cards - US English.
Backup System Files
This command is available with IP500 V2 systems. When selected, Manager copies the folders and
files from the System SD card's /primary folder to its /backup folder. Any matching files and
folders already present are overwritten. This action can be included as part of the system's
automatic daily backup process (System | System | Automatic Backup).
Restore System Files
This command is available with IP500 V2 systems. When selected, Manager copies the folders and
files from the System SD card's /backup folder to its /primary folder. Any matching files and
folders already present are overwritten.
Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
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Embedded File Management Menus
Upgrade Binaries
This command is available for IP500 V2 systems that have a system SD card and Optional SD card
installed.
When this command is selected, all files except config.cfg and keys.txt files in the Optional SD
card's \primary folder are copied to the System SD card.
Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
Upgrade Configuration
This command is not used with IP Office systems. This command is available for IP500 V2 systems
that have a system SD card and Optional SD card installed.
When this command is selected, any config.cfg and keys.txt files in the Optional SD card's
\primary folder are copied to the System SD card.
Warning:
After this command is completed, the system is rebooted. This will end all calls and services in
progress.
Upload Phone Files
This command is available for IP500 V2 control units. When this command is selected, Manager
copies the software files relating to phone firmware to the memory card. For IP500 V2 control units,
use Upload System Files.
Copy System Card
This command is available for IP500 V2 systems that have an Optional SD card installed in addition
to the mandatory System SD card. When this command is selected, the system will copy the folders
and files on its System SD card to the Optional SD card. Any matching files and folders already
present on the Optional SD card are overwritten.
This process takes at least 90 minutes and can take longer.
Configuration
This command will exit Embedded File Management and return Manager to configuration editing
mode.
Related links
Menu Bar Commands on page 48
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Chapter 5: Manager User Interface
This section of the documentation covers the operation of Manager when being used to edit the
configuration of a system running in Standard Mode. Much of it is also applicable for when also
editing the configuration of systems running in Server Edition mode. Additional Server Edition Mode
functions are detailed in the next chapter.
Related links
Title Bar on page 88
Toolbars on page 88
The Navigation Pane on page 90
The Group Pane on page 91
The Details Pane on page 93
The Error Pane on page 95
The Status Bar on page 97
Title Bar
The Manager title bar shows the following information.
• The Manager application version.
• The system name of the system from which the currently loaded configuration was received.
• The software level of the system's control unit.
• The service user name used to receive the configuration and that user's associated operator
rights.
Related links
Manager User Interface on page 88
Toolbars
Manager displays the following toolbars:
• Main Toolbar
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Toolbars
• Navigation Toolbar
• Details Toolbar
Related links
Manager User Interface on page 88
The Main Toolbar
The Main toolbar is displayed at the top of the Manager window, just below the menu bar. This
toolbar is also available when Manager is in security mode. However many of the controls will not
function in security mode.
Open Configuration from a System Advertises to the address currently shown in the Manager's
title bar for any available systems. A list of responding systems is then displayed. When a system is
selected from this list, a valid user name and password must be entered. Equivalent to File | Open
Configuration.
Open Configuration File Open a configuration file stored on a PC. The button can be clicked to
display a browse window. Alternatively the adjacent arrow can be used to drop-down a list of the
last 4 previously opened configuration files. Equivalent to File | Offline | Open File.
Save Configuration File The action of this icon depends on whether the currently loaded
configuration settings were received from a system or opened from a file stored on PC. If the former
applies, the menu sending the configuration back to the system is displayed. In the latter case, the
file changes are saved to the original file. Equivalent to File | Save Configuration.
Collapse All Groups Causes all
symbols in the navigation pane to be collapsed to
symbols.
Show/Hide the Navigation Pane
Show/Hide the Group Pane
Show/Hide the Error Pane
Validate Configuration Runs a validation on all the currently loaded configuration settings. The
results appear in the error pane. By default the configuration is automatically validated when loaded
and changes are validated when made, however the validation preferences can be changed through
File | Preferences | Validation.
Create New Configuration Runs a series of dialogs that create a new configuration from scratch.
Connect To For a standalone system, start the process of adding it to a multi-site network. Not
available in Server Edition mode.
Voicemail Pro Client Launch the Voicemail Pro client if also installed on the Manager PC.
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Manager User Interface
Server Edition Solution View Switch to the solution view. This option is only shown when
Manager is running in Server Edition mode.
The Navigation Toolbar
This toolbar provides drop down lists which can be used to navigate to particular records in the
configuration settings. The selected options in the navigation pane, group pane and the details pane
are synchronized with the navigation toolbar and vice versa. This toolbar is particularly useful if you
want to work with the group pane and or navigation pane hidden in order to maximize the display
space for the details pane.
This toolbar is not available when Manager is in security mode.
The Details Toolbar
This toolbar is shown in the top-right of the details pane. The options within the toolbar may vary or
be greyed out depending on the actions allowed for a particular configuration record.
Create a New Record The arrow is used to select the record type to be created. For example;
when adding an extension clicking may allow selection of a VoIP Extension or IP DECT
Extension.
Export as Template (Binary) This option is shown for Server Edition mode. It allows certain
types of records to be saved as templates. These can then be used to create new records.
Delete Current Record Delete the currently displayed record.
Validate Current Record By default records are validated when opened and when edited. This
is set through the Manager application's validation settings.
< > Previous Record/Next Record Click < or > at the top-right to move to the previous or next
record.
The Navigation Pane
This pane shows icons for the different types of record that the configuration can contain. Each type
is followed by the number of records of that type already in the configuration. When Manager is
used in security mode, this pane is also used by Manager in security mode to display records for
security settings.
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The Group Pane
Selecting an icon displays the matching records in the group pane, navigation toolbar and details
pane. Note that Manager is used to configure different types of system. Therefore the icons shown
may vary depending on the type of system you are configuring. For descriptions of the different
icons refer to Configuration Settings.
The information in the pane also depends on whether the group pane is visible or not. If the group
pane is visible, the navigation pane just shows icons for accessing which types of records should be
shown in the group pane. The group pane can then be used to select which of those records is
currently shown in the details pane. If the group pane is not visible, the navigation pane shows icons
for each type of records and under those icons for each individual record. The navigation pane can
then be used to select which of those records is currently shown in the details pane.
Related links
Manager User Interface on page 88
Expanding and Collapsing the Navigation Tree
Where or icons appear in the pane, they allow the structure to be expanded or collapsed. When
the group pane is hidden, and icons are shown for each record type and allow the record type
to be expanded to display all the existing records of that type.
The
icon in the main toolbar can also be used to collapse all the expanded record types shown in
the navigation pane.
The Group Pane
This pane lists all the records that match the type selected in the navigation pane or navigation
toolbar. The list can be sorted by clicking on a column heading. Selecting a record in this pane
displays its details in the details pane.
The icons used in the pane may vary according to the state of the record. For example, some of the
users shown in this example have been configured for hot desking. This pane is also used by
Manager in security mode to display records for security settings.
Related links
Manager User Interface on page 88
Sorting the List
About this task
The records shown in the group pane can be sorted using any of the columns displayed.
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Manager User Interface
Procedure
1. To sort the list using the details in a particular column, click on the column header.
2. Clicking on the same column header again reverses the sort order.
Customizing the Columns Displayed
About this task
For each record type, which details are shown in the group pane can be customized. Also the order
of the column can be adjusted.
Procedure
1. Right-click on the pane and select Customize Columns.
2. To add a column, selects its name in the left-hand Available Columns list and click >>to
move it to the right-hand Selected Columns list.
3. To remove a column, select its name in the right-hand Selected Columns list and click << to
move it to the left-hand Available Columns list.
4. To change the order of the Selected Columns, click on a column name and use the ^ and V
controls.
5. Click OK.
Changing the Column Widths
About this task
Procedure
1. In the column headers, place the cursor over the border between two columns.
2. When the cursor changes to a double headed arrow with a bar through it, click and hold the
cursor.
3. Drag the border to the required position and release the cursor.
Adding a New Record
About this task
The group pane can be used to add a new record of the type currently displayed.
Right-click on the pane and select New.
A arrow symbol next to New indicates that you can select a particular type of new record to create.
Click the arrow and select an option from the list.
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The Details Pane
Procedure
1. Use the details pane to configure the new record.
2. Click OKin the details pane.
Deleting an Record
About this task
Procedure
1. Select the record to be deleted by clicking on it.
2. Right-click on the pane and select Delete.
Validating an Record
About this task
Procedure
1. Select the record to be validated by clicking on it.
2. Right-click on the pane and select Validate.
Show in Groups
About this task
This command groups the items shown in the group pane. The grouping method will vary depending
on the record type being listed. For example, short codes are grouped based on short code feature
type such as all forwarding short codes together.
Procedure
Right-click on the pane and select Show In Groups.
The Details Pane
Whenever a selection is made through the group pane or the navigation toolbar, the settings for the
matching record are shown in the details pane. This pane is also used by Manager in security mode
to display records for security settings.
The details are grouped into tabs. The tabs available may vary depending on what particular type of
record is being viewed.
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Manager User Interface
Individual settings may also be grayed out. This indicates that they are either for information only or
that they cannot be used until another setting is enabled.
The top-left icon indicates the following:
Locked Indicates that you can view the settings but
cannot change them.
Editable Indicates that you can change the settings
if required.
Changed Indicates that the settings have been
changed since the tab was opened. Click OK to save
the changes or Cancel to undo.
Various icons may appear adjacent to settings:
Locked Setting The setting cannot be changed
through this tab. This icon appears on user settings
where the user is associated with User Rights that
controls the setting.
Information Indicates a value which does not have
to be set but may be useful if set.
Warning A warning indicates a configuration setting
value that is not typical and may indicate
misconfiguration.
Error An error indicates a configuration setting value
that is not supported by the system. Such settings
may cause the system to not operate as expected.
Related links
Manager User Interface on page 88
Managing Records on page 94
Managing Records
Procedure
1. Edit a record
a. The method of entering a record varies as different fields may use different methods.
For example text record boxes or drop down lists.
b. By default when changes are made, they are validated once another field is selected.
See File | Preferences | Validation.
c. Click on OK at the base of the details pane to accept the changes or click on Cancel to
undo the changes.
2. Add a record.
a. Click
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The Error Pane
b. Select the type of record required. For example, with extensions you can select from H.
323 Extension or SIP Extension.
3. Delete a record.
Click
at the top-right of the details pane.
4. Validate a record.
Click
at the top-right of the details pane.
5. Move to the previous or next record.
Click <or > at the top-right to move to the previous or next record.
6. Select a new tab.
a. To view the detail stored on a particular tab, click on the name of that tab.
b. If the tab required is not shown, use the
controls if shown on the right to scroll
through the available tabs. The tabs available may vary depending on what particular
type of record is being viewed.
Related links
The Details Pane on page 93
The Error Pane
Validation is a process where Manager checks configuration records for errors or for values for
which it regards as requiring a warning. The results of this checking are shown by icons next to the
field that caused the error or warning, All errors and warnings are also listed in the Error Pane.
By default validation is performed automatically whenever a configuration file is opened and when
any field is edited. However, if required, the use of automatic validation can be controlled through
the settings on the File | Preference | Validation tab.
Icons
The icons used for errors and warnings are as follows. These are shown in the error pane and also
next to the related field in the details pane. In the details pane, the error or warning description is
shown when the cursor is hovered over the icon.
Error An error indicates a configuration setting value that is not supported by the system. Such
settings are likely to cause the system to not operate as expected.
Warning A warning indicates a configuration setting value that is not typical and may indicate
misconfiguration.
Information Typically indicates a setting which may be useful to set.
Related links
Manager User Interface on page 88
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Manager User Interface
Altering the Automatic Validation Settings
About this task
The settings for automatic validation are adjustable.
Procedure
1. Select File | Preferences.
2. Select the Validation tab.
Select the options required.
• Validate configuration on open Automatically validate configuration files when they are
opened in Manager.
• Validate configuration on edit Validate the whole configuration when OK is clicked after
editing a record. For large configurations, disabling this option removes the delay caused
by validating the configuration after every edit.
• Prompt for configuration validation on save or send If selected, when saving or
sending a configuration, a prompt is displayed asking whether the configuration should be
validated. If validation is selected and error are found, the send or save process is
canceled. This option is disabled if Validate configuration on edit is selected.
3. Click OK.
Revalidating Configuration Settings
About this task
If necessary, you can force a validation check of the whole configuration or of the current record
shown in the details pane.
Procedure
1. To validate the whole configuration, click
2. For a particular record, click
in the main toolbar.
in the details pane.
Viewing an Error or Warning
About this task
Procedure
1. Clicking on an error or warning in the error pane will load the matching record tab into the
details pane.
2. The < and > can be used to move to the next error or warning in the error pane.
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The Status Bar
The Status Bar
The status bar at the base of the Manager screen is used to display icons and messages about
communications between Manager and systems. If the Manager is also acting as a BOOTP and
TFTP server it will also show BOOTP and TFTP messages.
A padlock icon is displayed whenever the Manager communications settings are set to secure. This
indicates all attempted configuration and security settings exchanged will be attempted over a
secure TLS link:
Status bar messages display information about communications the Manager application receives.
Some typical status bar messages are listed below.
Ready
This message is normally seen when Manager has just started and no configuration has been
received.
Received BOOTP request for 001125465ab2, unable to process
Manager is acting as a BOOTP server. It has received a BOOTP request that does not match a
system listed in its BOOTP records. The cause may be a device or application, other than an IP
Office, that also uses BOOTP.
TFTP: Received TFTP Error "NotFound" from 192.168.42.1
An attempt to receive settings from or send settings to the system failed. The most probable cause
is a name or password error.
TFTP: Received 17408 bytes for Marks_Test
Manager has received configuration settings from the named system using TFTP.
Sent 100% of C:\Program Files\Avaya\IP Office\Manager\b10d01b2_3.bin
Manager has sent the indicated file in response to a BOOTP request.
Related links
Manager User Interface on page 88
Configuring the Interface
The Manager configuration settings interface can be customized in a number of ways. These
changes are remembered the next time Manager is started.
Related links
Manager User Interface on page 88
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Manager User Interface
Resizing the Manager Window
About this task
When the Manager window is not maximized or minimized, it size can be adjusted.
Procedure
1. Place the cursor over the edge of the current window.
2. When the cursor changes to a double-headed arrow, click and hold the cursor.
3. Drag the edge to the required position and then release the cursor.
Moving the Border Between the Panes
About this task
The border between the visible panes can be adjusted. Note that this is a proportional rather than
exact position. If the whole window size is altered, the border position may also move.
Procedure
1. Place the cursor over the border between two panes.
2. When the cursor changes to a double headed arrow with a bar through it, click and hold the
cursor.
3. Drag the border to the required position and release the cursor.
Showing or Hiding Toolbars
About this task
The different toolbars can be hidden if not required.
Procedure
1. Select View and then Toolbars.
Those toolbars currently shown are indicated by a tick mark.
2. To show or hide a toolbar, click on its name.
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Configuring the Interface
Moving Toolbars
About this task
The position of the Manager toolbars can be moved. Note that when moving a toolbar, the other
toolbars and panes may adjust their size or position to ensure that all the toolbar icons remain
visible.
Procedure
1. Place the cursor over the end of the toolbar.
2. When the cursor changes to a four-way arrow, click and hold the cursor.
3. Move the toolbar to the required position and release the cursor.
Showing or Hiding Panes
About this task
The details pane cannot be hidden. The navigation pane, group pane and error pane can be shown
or hidden. To do this use either of the following methods.
From the main toolbar, use the following icons:
•
Hide/Show Navigation Pane.
•
Hide/Show Group Pane.
•
Hide/Show Error Pane.
or
Procedure
1. Select View.
Those panes currently shown are indicated by a tick mark.
2. To show or hide a pane, click on its name.
Changing the Position of the Details Pane
About this task
When the group pane is visible, the details pane is shown either below it or to its right. This position
can be adjusted.
Procedure
1. Select View and then Details Pane.
2. The current position setting is indicated by a tick mark.
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Manager User Interface
3. To select a position, click on it.
Changing the Size of Configuration Icons
About this task
The size of the icons used on the navigation pane and details pane can be adjusted.
Procedure
1. Select File and then Preferences.
2. Select the Visual Preferences tab.
3. Select the required icon size from Small, Medium or Large.
4. Click OK.
Changing Tab Display
About this task
For records with more than two tabs, you can select whether Manager should use
arrange the tabs as multiple rows when necessary.
controls or
Procedure
1. Select Files | Preferences | Visual Preferences.
2. Select Multiline Tabs.
3. Click OK.
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Chapter 6: Working with the Server Edition
Manager User Interface
Related links
Server Edition Solution View on page 101
System Inventories on page 103
Default Settings on page 103
Record Consolidation on page 104
Configuring Telephony Operation on page 105
Telephone Features Supported Across Server Edition and SCN Networks on page 132
Server Edition Solution View
When the configuration of a Server Edition solution is loaded into Manager, Manager starts with the
Server Edition Solution View menu. This menu includes the system inventory of the servers, links
for launching various functions and a summary table of the servers and the links between the
servers.
Displaying the Server Edition Solution View
Manager normally starts with the Server Edition Solution View when the configuration for a Server
Edition network is loaded. However, if required, to return to the solution view do one of the following.
• Click on the
• Click on the
Server EditionSolution View icon in the toolbar.
Solution icon in the navigation pane.
Interpreting and Using the Network Table
The table at the bottom of the solution view give a quick overview of all the servers and whether
their configuration was loaded into Manager.
Description
This column describes the type of server being detailed by the row. It also includes a
status indicator for the configuration file that Manager has loaded for the server.
•
Green - Configuration Loaded The configuration of the server has been
successfully retrieved and can be edited in Manager.
Table continues…
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Working with the Server Edition Manager User Interface
•
Yellow - Offline Configuration Loaded The configuration loaded is an offline
configuration. This will appear for a server that has been added to the solution
when the physical server is not currently connected on the network and Create
Offline Configuration was selected. The offline configuration file is stored on and
retrieved from the primary server until it can be replaced by or replace the actual
server configuration.
•
Red - Configuration Not Loaded There is no configuration for the system
loaded even though the solution configuration includes an entry for the server.
This will appear for a server that has been added to the network when the physical
server is not currently connected on the network and Create Offline
Configuration was not selected. If may also appear if the server is currently not
contactable.
•
Grey - No Connection This icon is used in conjunctions with the others to
indicate that there is no current connection to the server. For example:
• In conjunction with a green icon, it indicates that the server for which a
configuration has been loaded cannot be detected on the network. This may be a
temporary issue caused by that particular server rebooting following a
configuration change.
• In conjunction with a red icon, it indicates that the server for which a configuration
has not been loaded has now been detected on the network. Saving and reloading
the solution configuration may resolve the issue.
Name
This is the server name as taken from its configuration file. Offline is shown if no
configuration file is available.
Address
The IP address of the server. This is the address that is used when Manager
attempts to retrieve the servers configuration when loading the solution
configuration.
Primary Link
This value indicates the configuration settings of the H.323 IP trunk between the
primary server and the server indicated by the row. It should state Bothway. If it
states anything other, that indicates a mismatch in H.323 IP trunk configuration
between the system and the primary server. To correct this, right-click on the row
and select Connect to Primary.
Secondary Link
This column is only shown after a secondary server has been added to the
configuration of the solution. The value indicates the configuration settings of the H.
323 IP trunk between the secondary server and the server indicated by the row. It
should state Bothway. If it states anything other, that indicates a mismatch in H.323
IP trunk configuration between the system and the secondary server. To correct this,
right-click on the row and select Connect to Secondary.
Users Configured
This column summarizes the number of users (other than NoUser) configured on
the server. A total for the whole network is shown in the Solution row.
Extensions Configured
This column summarizes the number of extensions configured on the server. A total
for the whole network is shown in the Solution row.
Right-clicking on a server in the table may present a number of action. The actions available vary
with the current state of the network configuration.
• Remove Remove the server from the solution configuration.
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System Inventories
• Connect to Primary Repair the configuration of the H.323 IP trunks between the server and
the primary server.
• Connect to Secondary Repair the configuration of the H.323 IP trunks between the server
and the secondary server.
• Create Offline Configuration Create an offline configuration file for a server for which no
actual configuration has been loaded. The Offline Configuration menu will be displayed
followed by the Initial Configuration menu for the server type. The offline configuration file is
saved on the primary server.
Related links
Working with the Server Edition Manager User Interface on page 101
System Inventories
Manager can be used to display a system inventory for any of the servers in the Server Edition
solution. The system inventory is a quick summary of key settings and information about the server.
It can also display an overview system inventory for the whole Server Edition solution.
Displaying a Server's System Inventory
The method for displaying the system inventory depends on what is currently being displayed by
Manager.
In the Server Edition Solution View, using the table at the bottom of the menu, click on the server for
which you want to display the system inventory. Click on Network for the inventory of the Server
Edition network.
or
icon of the server for which you want to display the system
In the navigation pane, click on the
inventory. Click on the
Network icon for the inventory of the Server Edition network.
Related links
Working with the Server Edition Manager User Interface on page 101
Default Settings
Most of the defaults for systems in a Server Edition solution match those of individual IP Office
systems as detailed in the Configuration Settings section. The table lists some differences.
All auto-create extension and auto-create user settings for IP devices are set to off.
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Working with the Server Edition Manager User Interface
Settings
Primary Server
Secondary Server
Time
Settings
Hidden. Time taken
from host server.
SNTP from the primary server.
Voicemail
Voicemail Pro
Centralized Voicemail to the primary server
Alarms
Syslog relay all alarms
to the local host.
Syslog relay all alarms to the primary server.
IP Address
Specified during initial configuration menu.
Physical
–
–
Auto-created
IP Lines
H.323 line to the
secondary and each
expansion system.
H.323 line to the
primary and each
expansion system.
H.323 line to the
primary and to the
secondary if present.
Backup to secondary.
Backup to primary.
No backup.
Physical
–
–
Auto-created but no
base extension setting.
IP
None
None
None
User
None
None
None
Hunt Group
None
Not allowed
Not allowed
Incoming Call Route
None
Replicated from primary.
Directory
Stored on the primary
Configured to obtain system directory from the
primary server.
User Rights
None
Replicated from primary.
Short code to
secondary
Short code to primary
System
Lines
Extension
ARS
50:Main
Expansion System
Short code to primary
and to secondary if
present.
Related links
Working with the Server Edition Manager User Interface on page 101
Record Consolidation
Note:
For release 9.1 and higher, record consolidation is no longer supported for Incoming Call
Routes.
By default, to maintain the configurations of the systems in a Server Edition solution, certain types of
configuration records are treated differently. Short Code, Time Profile, Account Code and User
Rights records are only shown at the solution level and cannot be edited in individual system
configurations. However, Manager invisibly replicates these records, adding a copy to the
configuration of each system in the solution and updating those copies when necessary.
In Web Manager, consolidated records are shown at the top the Solutions page, under Solution
Objects.
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Configuring Telephony Operation
In Manager, operation of record consolidation is controlled by the File > Preferences >
Preferences setting Consolidate Solution to Primary Settings. By default that setting is selected.
The setting has the following effects.
If Consolidate Network to Primary Settings is selected:
• Entry and administration of Short Code, Time Profile, Account Code and User Rights
records is performed only at the solution level.
• Those records are then automatically replicated in the configurations of all the systems in the
solution but are still only visible and editable at the solution level.
• When the configurations are loaded into Manager or when this setting is changed to become
selected, if any inconsistency between records are found, a Consolidation Report is
displayed. This report allows selection of whether to update the system to match the primary or
to update the primary to match.
If Consolidate Network to Primary Settings is not selected:
Entry and administration of Short Code, Time Profile, Account Code and User Rights records
can be performed at both the solution and individual system levels.
• Records entered and edited at the solution level are automatically replicated in the
configurations of all the systems in the solution. Manager displays a label on the record
indicating that it is a record that is shared across the solution.
• If a shared record is edited at the individual system level, that copy of the record is no longer
shared with the other systems. It will not be updated by any changes to the solution level
version of the same record.
• No consolidation checking for inconsistencies is done by Manager when the configurations are
loaded.
Related links
Working with the Server Edition Manager User Interface on page 101
Configuring Telephony Operation
Each server in the network acts as a separate telephone system and supports most of the features
detailed for standalone IP Office systems in the Configuration Settings, Button Programming,
Appearance Button Operation and Telephone Features sections of this documentation. For features
that are supported between systems in the network, refer to Network Telephony Features.
Each system in the network automatically shares the extension numbers of users and hunt groups.
These can be dialed by any user regardless of which system they are hosted on and appear in the
internal directories on Avaya phones and applications.
The external system directory held by the primary server is shared with all systems on the network.
It is accessible by all users through the directory on their Avaya phones and applications.
Related links
Working with the Server Edition Manager User Interface on page 101
Incoming Call Routing on page 106
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Working with the Server Edition Manager User Interface
Outgoing Call Routing on page 111
Incoming Call Routing
The routing of incoming external calls is controlled by Incoming Call Route entries added to the
configuration of the network.
Determining which incoming call route is used is based on the call matching a number of possible
criteria. In order of highest priority first, the criteria, which if set must be matched by the call in order
for the call to use that route are:
1. The Bearer Capability indicated, if any, with the call. For example whether the call is a
voice, data or video call.
2. The Incoming Group ID of the trunk or trunk channel on which the call was received.
3. The Incoming Number received with the call.
4. The Incoming Sub Address received with the call.
5. The Incoming CLI of the caller.
The entries are shared by all systems in the network. If a specific route is required for a particular
system, it should be recalled that the Incoming Group ID assigned to a trunk or trunks can be used
as one of the criteria that must be matched on an incoming call route for calls to use that route.
The following are a number of examples that summarize the configuration changes necessary for
incoming call routing. These are simple examples and cannot cover all possible scenarios.
Related links
Configuring Telephony Operation on page 105
Example 1: Routing All Calls to an Operator on page 106
Example 2: Routing All Calls to an Operator Group on page 108
Example 3: Routing DID Calls to Matching Extensions on page 109
Example 1: Routing All Calls to an Operator
In this scenario, all incoming external calls are routed to a single user extension number. Depending
on the system licensing, that user can be configured as a receptionist, using the IP Office
SoftConsole application in parallel with their phone to answer and distribute the incoming calls.
1. Setup the Trunk Incoming Group IDs.
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Configuring Telephony Operation
The Incoming Group setting of the external line on the Primary Server has been set to 8.
The same incoming group ID can be used on multiple lines and on lines on different
systems.
The location of the Incoming Group field varies depending on the type of trunk. For SIP
trunks, it is set as part of the SIP URI settings used by the trunk.
2. Configure the Call Destinations.
A new user with extension number 201 is added to the configuration of the primary, along
with a matching extension for their IP phone.
If the user is to use the IP Office SoftConsole application, they must be added to the
configuration of the primary server. Receptionist users are not supported on other systems in
the network. If the user is not intending to use IP Office SoftConsole, they could be located
on any system in the network. External calls would be routed to the system hosting the user.
3. Create an Incoming Call Route to Match the Calls.
A new Incoming Call Route record is added to the configuration settings. The key settings
used in this scenario are listed below. All other fields are left at their defaults:
a. The fields on the Standard tab are used to set the criteria that are used to match
incoming calls to call routes.
• Bearer Capability: Any Voice This will match the incoming voice calls.
• Line Group ID: 8 This will match only calls on trunks where the trunk has its
Incoming Group setting also set to 8.
• Incoming Number: blank This will match any call.
b. The Destination tab is used to set the destination for calls that are matched to the
incoming call route. If necessary multiple destinations can be set, with Time Profiles
used to set when the different destinations are used. For this scenario we only have one
destination we want to use.
Using the Destination drop-down list, the 201 Receptionist user has been selected as
the destination for all calls. If they do not answer, the calls will go to their voicemail
mailbox. This is the simplest form of call routing. If required options such as a fallback
destination could be specified and different destinations to be automatically used
outside the operators known hours of work.
4. Save the configuration.
Result
All incoming voice calls on the primary system's SIP trunks are now matched to the new incoming
call route and are directed to the receptionist to be answered or to leave a message.
Since the Incoming Call Route entry is by default replicated as part of the configuration used by all
the servers, applying the same Incoming Group setting to any trunk hosted by any other system in
the network will route calls to the same destination without any further configuration.
Related links
Incoming Call Routing on page 106
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Working with the Server Edition Manager User Interface
Example 2: Routing All Calls to an Operator Group
In this scenario, all incoming external calls are routed to a single hunt group extension number. That
hunt group can be used to contain a number of user extension numbers and to specify the order in
which new calls should be presented to available users, whether calls should queue for an available
user and a range of other features. This is an ideal solution for scenarios where you have several
receptionists who can answer incoming external calls.
The configuration is very similar to that used in Example 1. Once the additional users have been
added and the hunt group created, the hunt group extension number automatically becomes
available as a selectable destination for Incoming Call Routes.
1. Setup the Trunk Incoming Group IDs
The Incoming Group setting of the external line on the Primary Server has been set to 8.
The same incoming group ID can be used on multiple lines and on lines on different
systems.
The location of the Incoming Group field varies depending on the type of trunk. For SIP
trunks, it is set as part of the SIP URI settings used by the trunk.
2. Configure the Call Destinations
Any existing receptionist user plus any new receptionists need to be added as members of a
hunt group.
New extension users with extension numbers 201, 202 and 203 are added to the
configuration of the primary, along with a matching extension for their IP phones.
A new hunt group with extension number 200 is added to the configuration of the primary
and users 201, 202 and 203 are specified as members of the group.
3. Create an Incoming Call Route to Match the Calls
A new Incoming Call Route record is added to the configuration settings. The key settings
used in this scenario are listed below. All other fields are left at their defaults.
a. The fields on theStandard tab are used to set the criteria that are used to match
incoming calls to call routes.
• Bearer Capability: Any Voice This will match the incoming voice calls.
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Configuring Telephony Operation
• Line Group ID: 8 This will match only calls on trunks where the trunk has its
Incoming Group setting also set to 8.
• Incoming Number: blank This will match any call.
b. The Destination tab is used to set the destination for calls that are matched to the
incoming call route. If necessary multiple destinations can be set, with Time Profiles
used to set when the different destinations are used. For this scenario we only have one
destination we want to use.
Using the Destination drop-down list, the 200 Reception group has been selected as
the destination for all calls. If they do not answer, the calls will go to their voicemail
mailbox. This is the simplest form of call routing. If required options such as a fallback
destination could be specified and different destinations to be automatically used
outside the operators known hours of work.
4. Save the configuration.
Result
All incoming voice calls on the primary system's SIP trunks are now matched to the new incoming
call route and are directed to the receptionist to be answered or to leave a message.
Since the Incoming Call Route entry is by default replicated as part of the configuration used by all
the servers, applying the same Incoming Group setting to any trunk hosted by any other system in
the network will route calls to the same destination without any further configuration.
Related links
Incoming Call Routing on page 106
Example 3: Routing DID Calls to Matching Extensions
In this scenario, the customer has subscribed to receive incoming number digits in the range 300 to
399 on certain calls. They want those calls routed to match extension numbers on the system, ie. to
users and hunt groups with the extension numbers in the range 300 to 399.
1. Setup the Trunk Incoming Group IDs
The Incoming Group setting of the external line on the Primary Server has been set to 8.
The same incoming group ID can be used on multiple lines and on lines on different
systems.
The location of the Incoming Group field varies depending on the type of trunk. For SIP
trunks, it is set as part of the SIP URI settings used by the trunk.
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2. Configure the Call Destinations
New extension users and hunt groups with extension numbers in the range 300 to 399 have
been added to the configurations of the systems as required to meet the customers needs.
3. Create an Incoming Call Route to Match the Calls
A new Incoming Call Route record is added to the configuration settings. The key settings
used in this scenario are listed below. All other fields are left at their defaults.
• The fields on the Standard tab are used to set the criteria that are used to match
incoming calls to call routes.
- Bearer Capability: Any Voice This will match the incoming voice calls.
- Line Group ID: 8 This will match only calls on trunks where the trunk has its Incoming
Group setting also set to 8.
- Incoming Number: 3XX This will match any call where the incoming number received
with the call ends in 300 to 399. If no Incoming Number match occurs, the call will be
matched to a default incoming call route with a blank Incoming Number field as setup
in examples 1 and 2.
Note that this is acting of digits supplied by the line provider, it is not caller ID (ICLID)
matching. Caller ID matching can be done using the Incoming CLI field.
• The Destination tab is used to set the destination for calls that are matched to the
incoming call route. If necessary multiple destinations can be set, with Time Profiles used
to set when the different destinations are used. For this scenario we only have one
destination we want to use.
In the Destination field, manually enter 3#. When this route is matched by a call, the # in
the destination is replaced by the incoming digits that matched the XX wildcards in the
Incoming Number field.
4. Save the configuration.
Result
Any incoming voice calls on the primary system's SIP trunks that now include 3XX at the end of an
incoming number supplied with the call are matched to this call route rather than to the route
previously setup (Example 1 or Example 2) and are routed to the matching extension number. Calls
where no incoming number match occurs are still matched to the previous incoming call route with a
blank Incoming Number field.
Related links
Incoming Call Routing on page 106
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Outgoing Call Routing
When a user dials a number, it is checked in a number of ways:
• The number dialed is always checked first for a match to an extension number. If a match
occurs, the call is routed to the matching user or hunt group. This matching is against any
extension number in the network, not just local system extension numbers.
• If the dialing does not match an extension number, it is checked for a matching short code.
Depending on the source of the dialing, the checking is made against user short codes, user
rights short codes and finally against system short codes. See Short Codes for full details.
• Dialing that is matched to a short code that uses the Dial feature is assumed to potentially be
an outgoing external call. The matching short code defines which where the call should be sent
and what number to dial should be sent.
• The normal practice is to route calls that match a Dial feature short code to an ARS form. The
ARS form can contain additional short codes to determine to which lines particular numbers are
routed. Doing this also helps keep external call short codes separate from short codes for other
functions and thus easier to maintain.
Call Routing Recommendations
Calls can be routed to a trunk using short codes set in the configuration of a particular user, users
(using User Rights), system (using system short codes) or ARS short codes. The following are
recommendations for the configuration on external call routing in a network:
1. Use ARS short codes wherever possible. This simplifies configuration and maintenance by
keeping call routing short codes separate from any other short codes, making the
configuration easier to implement and to understand. It also means that the full range of
other ARS features such as overflow and fallback routing can be used.
2. By default all calls are routed to the primary server and then fallback to the secondary.
Implementing as much call routing in the ARS settings of the primary as possible saves on
having to implement multiple matching settings on all the expansion servers in the network.
This eases maintenance and the addition of new expansion servers.
Short Codes and ARS Forms
Outgoing call routing relies on short codes and ARS records in the system configuration. Refer to
the relevant chapters on Short Codes and ARS to understand how those types of configuration
records are used.
The short method for describing short codes in this manual, for example 9N/Dial/./0, indicates the
settings of the following fields of a short code: Code/Feature/Telephone Number/Line Group ID.
For a description of the individual fields see Short Code.
Related links
Configuring Telephony Operation on page 105
Default Call Routing on page 112
Example 1: Using SIP Trunks Hosted by the Primary on page 114
Example 2: Primary Trunk Fallback to Secondary Trunks on page 117
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Example 3: Using Trunks on an Expansion System (V2) on page 121
Example 4: Local Branch Override on page 125
Example 5: PSTN Tail-End-Hop-Off on page 128
Default Call Routing
For outgoing call routing, a combination of system short codes and ARS entries are used. The
default operation is listed below. The default configuration is not sufficient to complete call routing,
additional configuration is required to route the calls from the Primary Server to the external trunks
hosted by the network.
The default settings send all potential external calls to the Primary Server where it is assumed those
calls will be routed to SIP trunks hosted by the primary. Additional configuration is necessary to
complete the routing from the Primary Server. The configuration to do that will vary depending on
which system is hosting the external trunk or trunks. Examples of typical configuration changes
required are given below.
The above default operation is achieved by the following defaults:
Primary Server:
The following external call routing is configured by default on the Primary Server:
• Default System Short Code
• The server has a default system short code that is used as a match for any dialing that does
not match an extension number or any other short code. This default system short code is also
used for matching to digits received on calls from other systems in the network. The default
system short code used depends on the server's companding (A-Law or U-Law) setting.
- A-Law On A-Law systems, a default ? short code is used to route any external dialing to
ARS record 50:Main. This will include matching any digits received on calls from other
servers in the network that don't match extension numbers.
- U-Law On U-Law systems, it is assumed that external calls are indicated by a 9 prefix. A
default short code 9N is used to route the digits N to ARS record 50:Main.
• Default ARS 50:Main
A first ? short code in the ARS form routes calls to the H.323 line that goes to the Primary
Server server by using the Line Group ID of 0.
All Other Servers:
On all other server types, the system and ARS defaults are set to route all potential external calls to
the Primary Server.
• Default System Short Code
• A default system ? short code is present in the configuration. This routes any dialing that has
no other match to the ARS record 50:Main in the configuration of the system where the dialing
occurred.
• Default ARS 50:Main
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Configuring Telephony Operation
A default ? short code in the ARS record is used to route all calls to the Primary Server. On
expansion systems, an additional ? short code is used to route calls to the Secondary Server if
the Primary Server is not available for some reason.
Default Settings Summary
The table below summarizes the settings described above.
A-Law
System Short Code
ARS 50:Main Short
Codes
Description
Primary Server
?/Dial/./50:Main
?/Dial/./0
Send all calls with no
other match to any trunks
in with the Outgoing
Group setting of 0. Since
0 is not a valid value, this
needs to be changed to
match the Outgoing
Group setting actually
used for the external
trunks hosted by the
Primary Server.
Secondary Server
?/Dial/./50:Main
?/Dial/./99999
Send all calls with no
other match to the
Primary Server.
Expansion
?/Dial/./50:Main
?/Dial/./99999 ?/Dial/./
99998
Send all calls with no
other match to the
Primary Server if
available, else to the
Secondary Server.
U-Law
System Short Code
ARS 50:Main Short
Codes
Description
Primary Server
9N/Dial/N/50:Main
?/Dial/./0
Send all calls with no
other match to any trunks
in with the Outgoing
Group setting of 0. Since
0 is not a valid value, this
needs to be changed to
match the Outgoing
Group setting actually
used for the external
trunks hosted by the
Primary Server.
Secondary Server
?/Dial/./50:Main
?/Dial/./99999
Send all calls with no
other match to the
Primary Server.
Expansion
?/Dial/./50:Main
?/Dial/./99999 ?/Dial/./
99998
Send all calls with no
other match to the
Table continues…
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U-Law
System Short Code
ARS 50:Main Short
Codes
Description
Primary Server if
available, else to the
Secondary Server.
The short method for describing short codes in this manual, for example 9N/Dial/./0, indicates the
settings of the following fields of a short code: Code/Feature/Telephone Number/Line Group ID.
For a description of the individual fields see Short Code.
The different destinations used for the dialing short codes in the Line Group ID field are:
• 50:Main Route the call to the ARS record 50:Main on the system where the short code match
has occurred. If additional ARS records are created on a system they can also be selected as
the destination for calls to be routed.
• 99999 Route the call to the Primary Server.
• 99998 Route the call to the Secondary Server.
• 99901 to 99930 Route the call to one of the expansion systems. These destinations can only
be used on short codes on the Primary Server and Secondary Server.
• 0 Seize a trunk from those trunks on the system that have their Outgoing Group setting set to
0. Since 0 is not a valid value, this needs to be changed to match the Outgoing Group setting
actually used for the external trunks hosted by the Primary Server.
Examples
The following are a number of examples that summarize the configuration changes necessary for
outgoing call routing. These are simple examples and cannot cover all possible scenarios. There are
additional examples of how the settings in the ARS forms can be used in the ARS section of this
document.
Related links
Outgoing Call Routing on page 111
Example 1: Using SIP Trunks Hosted by the Primary
This is the simplest example. It assumes that a SIP trunk has been added to the configuration of the
Primary Server, along with the necessary licenses.
In the Primary Server configuration:
1. In the configuration of the SIP trunk on the Primary Server, the SIP URI form includes a field
for setting the Outgoing Group. Set this to a unique ID, by default it is set to 1.
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Configuring Telephony Operation
2. In the ARS record 50:Main on the Primary Server, select and edit the exist ? short code to
change it from Line Group ID of 0 to a Line Group ID of 1.
• Code: ? The ? short code character matches any digits for which no other short code
match is present in this ARS form.
• Feature: Dial
• Telephone Number: . The . short code character matches all the digits dialled, not just
those that matched the Code field.
• Line Group ID: 1 (or whichever number was set for the SIP trunk Outgoing Group on the
Primary Server) This field is used to match outgoing calls (ie. calls using the Dial feature)
to an available trunk on the system with the matching number as its Outgoing Group
setting. Multiple trunks on the same system can have the same Outgoing Group setting.
3. Save the configuration.
Results: A-Law Systems
For systems operating in A-Law, it is assumed that no dialing prefix is used for external calls.
If a user hosted on an expansion system dials 555 123 4567:
Server
Event
Digits
Expansion System
A user dials 555 123 4567.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to the
primary.
555 123 4567
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Server
Event
Digits
Primary Server
The call is received on the H.323
line.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available SIP trunk channel.
555 123 4567
If a user hosted on the primary system dials 555 123 4567:
Server
Event
Digits
Primary Server
A user dials 555 123 4567.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available SIP trunk channel.
555 123 4567
Results: U-Law Systems
For systems operating in U-Law, it is assumed that a 9 prefix is used for external calls.
If a user hosted on an expansion system dials 9 555 123 4567:
Server
Event
Digits
Expansion System
A user dials 9 555 123 4567.
9 555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to the
primary.
9 555 123 4567
The call is received on the H.323
line.
9 555 123 4567
The digits are matched to the 9N/
Dial/N system short code. This
routes the call to ARS 50:Main.
having removed the 9 prefix.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available SIP trunk channel.
555 123 4567
Primary Server
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Configuring Telephony Operation
If a user hosted on the primary system dials 9 555 123 4567:
Server
Event
Digits
Primary Server
A user dials 9 555 123 4567.
9 555 123 4567
The digits are matched to the 9N/
Dial/N system short code. This
routes the call to ARS 50:Main.
having removed the 9 prefix.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available SIP trunk channel.
555 123 4567
Related links
Outgoing Call Routing on page 111
Example 2: Primary Trunk Fallback to Secondary Trunks
This example is builds on the previous example. In this case the network has also has a Secondary
Server and SIP trunks have been added to the Secondary Servers configuration using the
Outgoing Group setting of 2.
In the Primary Server configuration:
Configure the Primary Server as per example 1. This will route all outgoing calls to the SIP trunk
hosted by the Primary Server.
In the Secondary Server configuration:
1. In the configuration of the SIP trunk on the Secondary Server, the SIP URI form includes a
field for setting the Outgoing Group. Set this to a unique ID, for this example we assume it
is 2.
2. Select ARS and click on the
icon to add a new ARS record.
3. Set the Route Name to something suitable descriptive such as Fallback.
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4. Add a short code that will route calls that use this ARS record to the SIP trunk hosted by the
secondary. The short code depends on whether we need to remove a dial 9 prefix or not:
• If no dialing prefix is being used, add a ?/Dial/./2 short code.
• If a dial 9 prefix is being used, add a 9N/Dial/N/2 short code.
5. Click OK.
In the ARS record 50:Main on the Secondary Server, we need to set the record to fallback to
using the new ARS form when a route to the primary cannot be seized.
6. In the Alternate Route drop down select the fallback ARS.
7. Set the Alternate Route Priority Level to 1. This is the lowest level of priority so no users
will need to wait to use the alternate ARS is the connection to the primary is not available.
8. Save the configurations.
Results: A-Law Systems
For systems operating in A-Law, it is assumed that no dialing prefix is used for external calls.
If a user hosted on an expansion system dials 555 123 4567:
Server
Event
Digits
Expansion System
A user dials 555 123 4567.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
Table continues…
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Configuring Telephony Operation
Server
Secondary Server
Event
Digits
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This tries to route the call to
the primary (99999).
555 123 4567
A trunk to the primary cannot be
seized. The digits are matched to
the next ?/Dial/. short code in the
ARS form. This routes the call to
the secondary (99998).
555 123 4567
The call is received on the H.323
line.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The ARS cannot seize a trunk to
the primary so fallback to its
alternate ARS, 51:Fallback.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available SIP trunk channel.
555 123 4567
If a user hosted on the secondary system dials 555 123 4567:
Server
Event
Digits
Secondary Server
A user dials 555 123 4567.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The ARS cannot seize a trunk to
the primary so fallback to its
alternate ARS, 51:Fallback.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available SIP trunk channel.
555 123 4567
Results: U-Law Systems
For systems operating in U-Law, it is assumed that a 9 prefix is used for external calls.
If a user hosted on an expansion system dials 555 123 4567:
Server
Event
Digits
Expansion System
A user dials 555 123 4567.
9 555 123 4567
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Server
Secondary Server
Event
Digits
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This tries to route the call to
the primary (99999).
9 555 123 4567
A trunk to the primary cannot be
seized. The digits are matched to
the next ?/Dial/. short code in the
ARS form. This routes the call to
the secondary (99998).
9 555 123 4567
The call is received on the H.323
line.
9 555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 555 123 4567
The ARS cannot seize a trunk to
the primary so fallback to its
alternate ARS, 51:Fallback.
9 555 123 4567
The digits are matched to the first 555 123 4567
9N/Dial/N short code in the ARS
form. This routes the call to an
available SIP trunk channel having
removed the 9 prefix.
If a user hosted on the secondary system dials 555 123 4567:
Server
Event
Digits
Secondary Server
A user dials 555 123 4567.
9 555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 555 123 4567
The ARS cannot seize a trunk to
the primary so fallback to its
alternate ARS, 51:Fallback.
9 555 123 4567
The digits are matched to the first 555 123 4567
9N/Dial/N short code in the ARS
form. This routes the call to an
available SIP trunk channel having
removed the 9 prefix.
Related links
Outgoing Call Routing on page 111
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Configuring Telephony Operation
Example 3: Using Trunks on an Expansion System (V2)
Primary Server, Secondary Server and Linux Expansion systems only support SIP trunks for
external trunk connections. An Expansion System (V2) must be used for other trunk types such as
analog, BRI or PRI. In this example we assume that the Expansion System (V2) is being used for all
outgoing external calls from the network.
1. In the Expansion A configuration: No changes are required, this and other expansion
systems still send their calls to the primary which then routes the calls to the system which is
hosting the external trunks.
2. In the configuration of the Primary Server, identify the H.323 IP trunk that is used for calls to
the Expansion System (V2) and note the Outgoing Group ID setting, it will be in the range
99901 to 999930. The trunk can be recognized by the Gateway IP Address on the VoIP
Settings tab matching the IP address of the Expansion System (V2).
3. In the ARS record 50:Main on the Primary Server, select and edit the default ? short code to
route calls to the expansion system:
• Code: ?
• Feature: Dial
• Telephone Number: .
• Line Group ID: 99901 (or whichever number was set for the Outgoing Group ID to the
Expansion System (V2)).
4. In the configuration of the system hosting the external trunk:
a. Set the Outgoing Group ID of the external trunks to a unique value. For this example
we assume 3 is used. The field where this is set depends on the trunk type but always
has the same name.
b. [Optional] If the network expects a dial 9 prefix on external calls, we need to add a new
system short code that matches and removes that prefix from calls dialed on the local
system as for call made on other systems this prefix is removed by the primary. Add a
system short code similar to the following:
• Code: 9N
• Feature: Dial
• Telephone Number: N
• Line Group ID: 50:Main
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5. In the ARS record 50:Main, select and edit the first default ? short code to use the Line
Group ID of 3.
• Code: ?
• Feature: Dial
• Telephone Number: .
• Line Group ID: 3 (or whichever number was set for the Outgoing Group ID on the
system's external trunks).
6. Save the configuration.
Results: A-Law
For systems operating in A-Law, it is assumed that no dialing prefix is used for external calls.
If a user hosted on other expansion systems dials 555 123 4567:
Server
Event
Digits
Expansion System A
A user dials 555 123 4567.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to the
primary.
555 123 4567
The call is received on the H.323
line.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to
expansion system B.
555 123 4567
The call is received on the H.323
line.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available trunk.
555 123 4567
Primary Server
Expansion System B
If a user hosted on the primary server dials 555 123 4567:
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Configuring Telephony Operation
Server
Event
Digits
Primary Server
A user dials 555 123 4567.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to
expansion system B.
555 123 4567
The call is received on the H.323
line.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available trunk.
555 123 4567
Expansion System B
If a user hosted on expansion system B dial 555 123 4567:
Server
Event
Digits
Expansion System B
A user dials 555 123 4567.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available trunk.
555 123 4567
Results: U-Law
For systems operating in U-Law, it is assumed that a 9 prefix is used for external calls.
If a user hosted on other expansion systems dials 9 555 123 4567:
Server
Event
Digits
Expansion System A
A user dials 9 555 123 4567.
9 555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to the
primary.
9 555 123 4567
Table continues…
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Server
Event
Digits
Primary Server
The call is received on the H.323
line.
9 555 123 4567
The digits are matched to the 9N/
Dial/N system short code. This
routes the call to ARS 50:Main.
having removed the 9 prefix.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to
expansion system B.
555 123 4567
The call is received on the H.323
line.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available trunk.
555 123 4567
Expansion System B
If a user hosted on the primary server dials 9 555 123 4567:
Server
Event
Digits
Primary Server
A user dials 9 555 123 4567.
9 555 123 4567
The digits are matched to the 9N/
Dial/N system short code. This
routes the call to ARS 50:Main.
having removed the 9 prefix.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to
expansion system B.
555 123 4567
The call is received on the H.323
line.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available trunk.
555 123 4567
Expansion System B
If a user hosted on expansion system B dial 9 555 123 4567:
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Configuring Telephony Operation
Server
Event
Digits
Expansion System B
A user dials 9 555 123 4567.
9 555 123 4567
The digits are matched to the 9N/
Dial/N system short code. This
routes the call to ARS 50:Main.
having removed the 9 prefix.
9 555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available trunk.
9 555 123 4567
Related links
Outgoing Call Routing on page 111
Example 4: Local Branch Override
This example builds on Example 1 where all external calls are routed to SIP trunks hosted by the
primary server.
There may be cases where an expansion system (Expansion System (L) and Expansion System
(V2)) can route calls through its own local trunks more cheaply than through trunks located on
another system such as the primary. For example, if the site where the expansion system is located
can make local calls free of charge or at a much lower cost. If that is the case, then for certain
dialing we would want to have those calls use the local trunk rather than following the default routing
to the primary server.
Suppose for this example, expansion system A is in national area code 444. The customer wants
any dialing of numbers prefixed with 444 by the users hosted on that system to be routed out on the
external trunks connected to that expansion system. All other external dialing should continue to
follow the defaults of being routed to the primary.
1. In the configuration of the expansion system, set the Outgoing Group ID of the external
trunks to a unique value. For this example we assume 4 is used. The field where this is set
depends on the trunk type but always has the same name.
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Working with the Server Edition Manager User Interface
2. In the ARS record 50:Main on the expansion system we need to add a new short code that
matches and routes calls that include the 444 area code. The short code required depend on
whether an external dialing prefix is used or not.
Add a new short code:
• Code: 444N (A-Law) or 9444N (U-Law)
• Feature: Dial
• Telephone Number: N
• Line Group ID: 4 (or whichever number was set for the Outgoing Group ID on the
system's external trunks).
The new code will match any dialing sent to the ARS prefixed with 444. Depending on local
dialing patterns it may also be necessary to add similar codes to match dialing prefixed with
a national prefix before the local area code. For example short codes for dialing prefixed
1444N or dialing of a particular length that would imply its a local number, eg. XXXXXXX.
3. Save the configuration.
Results: A-Law Systems
For systems operating in A-Law, it is assumed that no dialing prefix is used for external calls.
If a user hosted on expansion system A dials 555 123 4567:
This routing matches the example setup in Example 1 where the main trunks and call routing
settings for external calls by any system are configured on the primary server.
Server
Event
Digits
Expansion System A
A user dials 555 123 4567.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to the
primary.
555 123 4567
The call is received on the H.323
line.
555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available SIP trunk channel.
555 123 4567
Primary Server
If a user hosted on expansion system A dials 444 123 4567:
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Configuring Telephony Operation
Server
Event
Digits
Expansion System A
A user dials 444 123 4567.
444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
444 123 4567
The digits are matched to the first
444N/Dial/N short code in the
ARS form. This routes the call to
the external trunk on the system.
444 123 4567
Results: U-Law
For systems operating in U-Law, it is assumed that a 9 prefix is used for external calls.
If a user hosted on expansion system A dials 9 555 123 4567:
This routing matches the example setup in Example 1 where the main trunks and call routing
settings for external calls by any system are configured on the primary server.
Server
Event
Digits
Expansion System A
A user dials 9 555 123 4567.
9 555 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to the
primary.
9 555 123 4567
The call is received on the H.323
line.
9 555 123 4567
The digits are matched to the 9N/
Dial/N system short code. This
routes the call to ARS 50:Main.
having removed the 9 prefix.
555 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to an
available SIP trunk channel.
555 123 4567
Primary Server
If a user hosted on expansion system A dials 9 444 123 4567:
Server
Event
Digits
Expansion System A
A user dials 9 444 123 4567.
9 444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 444 123 4567
Table continues…
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Working with the Server Edition Manager User Interface
Server
Event
Digits
The digits are matched to the first
9444N/Dial/N short code in the
ARS form. This routes the call to
the external trunk on the system.
444 123 4567
Related links
Outgoing Call Routing on page 111
Example 5: PSTN Tail-End-Hop-Off
This example builds on Example 4 where calls local to a particular expansion system are routed to
external trunks hosted by that expansion system. Having setup routing similar to that example, it
may be advantageous to route any dialing of the same are code on other systems in the network to
the trunks on the local expansion system.
Suppose for this example, we want users on any system in the network that dial a call prefixed with
national area code 444 to have that call routed to external trunks on expansion system A which is in
that local area. We can do this by adding additional short codes to the ARS on the primary system
as the default routing on all systems already sends all calls to the primary.
Important:
Check Local and National Call Routing Regulations
This type of operation may be subject to nation restrictions, especially where the expansion
system trunks are being used to make calls in a different country code to other systems in the
network. Such operation is not allowed by the telecommunications regulations in some
countries.
1. In the configuration of the Primary Server, identify the H.323 IP trunk that is used for calls to
the expansion system and note the Outgoing Group ID setting, it will be in the range 99901
to 999930. The trunk can be recognized by the Gateway IP Address on the VoIP Settings
tab matching the IP address of the expansion system.
2. In the ARS record 50:Main on the primary server:
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Configuring Telephony Operation
Add a new short code:
• Code: 444N (A-Law) or 9444N (U-Law)
• Feature: Dial
• Telephone Number: .
• Line Group ID: 99901 (or whichever number was set for the Outgoing Group ID to the
expansion system).
The new code will match any dialing sent to the ARS prefixed with 444. Again, as in Example
4, it may also be necessary to add similar codes to match dialing prefixed with a national
prefix before the local area code. For example short codes for dialing prefixed 1444N or
dialing of a particular length that would imply its a local number, eg. XXXXXXX.
3. In the expansion system configuration, apply the same short codes as used for Example 4 to
route dialing to area code 444 dialed by users on that system. Those short codes are also
applied to calls received on the network trunks that arrive with those digits.
4. Save the configuration.
Results: A-Law Systems
Any dialing prefixed with 444 on any system is routed to the primary server from where it is rerouted
to the expansion system hosting trunks local to that area code.
If a user hosted on expansion system A dials 444 123 4567:
Server
Event
Digits
Expansion System A
A user dials 444 123 4567.
444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
444 123 4567
The digits are matched to the first
444N/Dial/N short code in the
ARS form. This routes the call to
the external trunk on the system.
444 123 4567
If a user hosted on the primary server dials 444 123 4567:
Server
Event
Digits
Primary Server
A user dials 444 123 4567.
444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
444 123 4567
The digits are matched to the first
444N/Dial/. short code in the ARS
form. This routes the call to
expansion system A.
444 123 4567
Table continues…
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Working with the Server Edition Manager User Interface
Server
Event
Digits
Expansion System A
The call is received on the H.323
line.
444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
444 123 4567
The digits are matched to the first
444N/Dial/N short code in the
ARS form. This routes the call to
the external trunk on the system.
123 4567
If a user hosted on other expansion systems dials 444 123 4567:
Server
Event
Digits
Expansion System B
A user dials 444 123 4567.
444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
444 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to the
primary.
444 123 4567
The call is received on the H.323
line.
444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
444 123 4567
The digits are matched to the first
444N/Dial/. short code in the ARS
form. This routes the call to
expansion system A.
444 123 4567
The call is received on the H.323
line.
444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
444 123 4567
The digits are matched to the first
444N/Dial/N short code in the
ARS form. This routes the call to
the external trunk on the system.
123 4567
Primary Server
Expansion System A
Results: U-Law Systems
Any dialing prefixed with 9444 on any system is routed to the primary server from where it is
rerouted to the expansion system hosting trunks local to that area code.
If a user hosted on expansion system A dials 9 444 123 4567:
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Configuring Telephony Operation
Server
Event
Digits
Expansion System A
A user dials 9 444 123 4567.
9 444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 444 123 4567
The digits are matched to the first
9444N/Dial/N short code in the
ARS form. This routes the call to
the external trunk on the system.
123 4567
If a user hosted on the primary server dials 9 444 123 4567:
Server
Event
Digits
Primary Server
A user dials 9 444 123 4567.
9 444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 444 123 4567
The digits are matched to the first
9444N/Dial/. short code in the
ARS form. This routes the call to
expansion system A.
9 444 123 4567
The call is received on the H.323
line.
9 444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 444 123 4567
The digits are matched to the first
9444N/Dial/N short code in the
ARS form. This routes the call to
the external trunk on the system.
123 4567
Expansion System A
If a user hosted on other expansion systems dials 9 444 123 4567:
Server
Event
Digits
Expansion System B
A user dials 9 444 123 4567.
9 444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 444 123 4567
The digits are matched to the
first ?/Dial/. short code in the ARS
form. This routes the call to the
primary.
9 444 123 4567
The call is received on the H.323
line.
9 444 123 4567
Primary Server
Table continues…
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Working with the Server Edition Manager User Interface
Server
Expansion System B
Event
Digits
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 444 123 4567
The digits are matched to the first
9444N/Dial/. short code in the
ARS form. This routes the call to
expansion system A.
9 444 123 4567
The call is received on the H.323
line.
9 444 123 4567
The digits are matched to the ?/
Dial/. system short code. This
routes the call to ARS 50:Main.
9 444 123 4567
The digits are matched to the first
9444N/Dial/N short code in the
ARS form. This routes the call to
the external trunk on the system.
123 4567
Related links
Outgoing Call Routing on page 111
Telephone Features Supported Across Server Edition and
SCN Networks
Each system running IP Office in a multi-site network acts as a self-contained IP Office telephone
system. In addition to the remote systems sharing knowledge of user and hunt group extension
numbers, the following additional telephony features are supported between systems in a multi-site
network. Features not listed are not supported across the multi-site network.
Absence Text
Advertised Hunt Groups Hunt groups set to advertised can be dialed by users on other systems
Anti-tromboning Calls routed across the multi-site network and back to the originating system are
turned back into internal calls on the originating system only.
Break Out Dialing
Call Park / Unpark Call
Call Pick-up Extension
Call Tagging
Callback When Free
Centralized Call Log
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Telephone Features Supported Across Server Edition and SCN Networks
Centralized Personal Directory
Conference
Distributed Hunt Groups
Distributed Voicemail Server Support When using Vociemail Pro, each system can support its
own Voicemail Pro server. See the Voicemail Pro Installation Manual.
Enable ARS / Disable ARS
Extension Dialing Each system automatically learns the user extension numbers available on other
systems and routes calls to those numbers.
Fallback Server Edition Fallback SCN Fallback
Fax Relay
Follow Me Here / Follow Me To
Forwarding
Hold Held calls are signalled across the network.
Internal Twining
Intrusion Features
Mobile Call Control Licensed mobile call control users who remote hot desk to another system take
their licensed status with them.
Music On Hold Source Selection
Relay On / Relay Off / Relay Pulse
Remote Hot Desking
Set Hunt Group Out of Service / Clear Hunt Group Out of Service
Transfer Calls can be transferred to network extensions.
User DSS/BLF Monitoring of user status only. The ability to use additional features such as call
pickup via a USER button will differ depending on whether the monitored user is local or remote.
Indication of new voicemail messages provided by SoftConsole user speed dial icon is not
supported.
User Profile Resilence When a user hot desks to another system, they retain their Profile settings
and rights.
Related links
Working with the Server Edition Manager User Interface on page 101
Configuring Small Community Networking on page 749
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Chapter 7: Security Administration
The security settings are stored on the system and are separate from the system's configuration
settings. To change a system's security settings, Manager must first be switched to security mode
by selecting File | Advanced | Security Settings from the menu bar.
Security settings can only be loaded directly from a system. These settings cannot be saved as a
file on the local PC, nor do they appear as a temporary file at any time. You can optionally secure
the link between the system and Manager for configuration and security settings exchanges. By
default Manager and the system will always attempt to use the original, unsecured link.
Administration security is achieved using a number of optional cryptographic elements:
•
•
•
•
Access control to prevent unauthorized use.
Encryption to guarantee data remains private.
Message Authentication ensures data has not been tampered with.
Identity assures the source of the data.
Related links
Security Users on page 134
Access Control on page 140
Encryption on page 140
Message Authentication on page 141
Certificates on page 142
Implementing Security on page 147
SRTP on page 149
Security Users
Access to system settings is controlled by Service Users and Rights Groups stored in the control
unit's security settings. These are stored separately from the system's configuration settings. All
actions involving communications between Manager and the system require a service user name
and password. That service user must be a member of a Rights Group with permissions to perform
the required action.
Security Administrators:
By default the security administrator is the only user who can access the system's security settings
using Manager's security mode.
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Security Users
Service Users:
Each service user has a name, a password and is a member of one or more Rights Groups.
Rights Groups:
The Rights Groups to which a service user belongs determine what actions they can perform.
Actions available to Rights Groups include configuration actions, security actions and system status
actions. Where a service user has been configured as a member of more than one Rights Group,
they combine the functions available to the separate Rights Groups.
Related links
Security Administration on page 134
Default Service Users and Rights Groups on page 135
Default Service Users and Rights Groups
Security Administrator Account
The following Security Administrator account is present on first startup and security settings reset.
Name
Default
Account
Status
Usage
Rights Group
Membership
Notes
Security
Enabled,
This is the default security
administration account.
Implied all
security rights
Cannot be removed or
disabled
Force
password
change
Has all rights to all security
management and
maintenance services
Should not be renamed
Service User Accounts
The following Service User accounts are present on first start-up and security settings reset.
Name
Default
Account
Status
Usage
Rights Group
Membership
Administrator
Enabled,
This is the default account
used for system
configuration using the IP
Office and Web Manager
applications, including oneX Portal/Voicemail Pro
administration.
Administrator,
Should not be removed or
System Status,
disabled
Business Partner
Should not be renamed
Force
password
change
Notes
Has all rights to all
management and
maintenance services
including security settings.
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Security Administration
Name
Default
Account
Status
Usage
Rights Group
Membership
Notes
EnhTcpaService
Enabled
This account is used for
one-X Portal for IP Office
connections to the system.
TCPA Group
Although not enforced, the
password should be
change as soon as possible
in both IP Office and one-X
Portal
Enable only when one-X
Portal deployed
IPDECTService
Disabled
This account is used for
DECT R4 system
provisioning
IPDECT Group
Enable only when DECT
R4 deployed and
provisioning mode active
BranchAdmin
Disabled
This account is used for
System Manager (SMGR)
access in a branch
deployment
SMGR Admin
Enable only when SMGR
deployed; will be enabled
when the Initial
Configuration Utility (ICU)
run and SMGR
administration selected.
Must not be renamed
BusinessPartner
Disabled
Similar access rights to
Administrator and can be
used as a separate
account for Business
Partners
Business Partner Should be removed/
disabled unless required
Maintainer
Disabled
Maintenance account
Maintainer
without edit configuration or
security access.
Should be removed/
disabled unless required
Can be used for Manager
(read-only), Web Manager
(read-only), System Status
Application (SSA), Backup/
Restore, System Monitor,
Upgrade
Rights Groups
The following Rights Groups are present on first start-up and security settings reset.
Name
Usage
Rights Group
User
Notes
Administrator
Group
Allows full access to the IP
Office Manager application
to configure the system.
Administrator
All IP Office Manager operations are
permitted
No security or maintenance
access
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Security Users
Name
Usage
Rights Group
User
Notes
Manager Group
Allows limited access to the
IP Office Manager
application to configure the
system.
–
All IP Office Manager operations
permitted except:
• Delete Short Code
• View LAN2 Settings
Operator Group
Allows limited access to the
IP Office Manager
application to configure the
system.
–
All IP Office Manager operations
permitted except:
• New object creations
• View LAN2 Settings
• Delete Directory
• Delete ICR
System Status
Group
Allows limited access to the
SSA and Sys Monitor
applications.
Administrator
TCPA Group
This group is used by the
one-X Portal for IP Office
application.
EnhTcpaService
IPDECT Group
This group is used by the
DECT R4 master base
station to extract DECT
settings from IP Office.
IPDECTService
SMGR Admin
This group is used by
SMGR to configure IP
Office.
BranchAdmin
Security Admin
Allows access to security
settings only
–
Backup Admin
Allows access to all backup
and restore services only,
including one-X Portal
–
Upgrade Admin
Allows access to the
upgrade service
–
System Admin
Allows configuration of IP
Office, one-X Portal and
Voicemail Pro
–
Maint Admin
Allows configuration of IP
Office, one-X Portal and
Voicemail Pro along with
backup, restore and
upgrade
–
Sys Monitor access right only checked
when using service users with Sys
Monitor
Do not change the access rights
Allows upgrade of both IP Office
applications and operating system
Typically used for maintenance
personnel
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Security Administration
Name
Usage
Rights Group
User
Business Partner
Full access to all
configuration, security and
maintenance services.
Administrator,
BusinessPartner
Customer Admin
Web Management , one-X
Portal and Voicemail Pro
administration
–
Maintainer
Allows configuration view
only, along with SSA, Sys
Monitor backup, restore and
upgrade
Notes
No IP Office manager access
Typically used for maintenance
personnel with no need for
configuration changes
Rights Group Assignment
Service
Access Right
Rights Group
• 1 = Administrator
Group
• 2 = Manager Group
• 5 = TCPA Group
• 6 = IPDECT Group
• 7 = SMGR Admin
• 3 = Operator Group
• 4 = System Status
Group
1
Configuration
2
3
4
5
6
7
Read all configuration
Write all configuration
Merge configuration
Default configuration
Reboot/Shutdown
immediately
Reboot when free
Reboot at time of day
Security Admin
Read all security
settings
Write all security
settings
Reset all security
settings
Write own service user
password
System Status
System Status Access
Read all configuration
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Security Users
Service
Access Right
Rights Group
• 1 = Administrator
Group
• 2 = Manager Group
• 5 = TCPA Group
• 6 = IPDECT Group
• 7 = SMGR Admin
• 3 = Operator Group
• 4 = System Status
Group
1
2
3
4
5
6
7
System Control
Sys Monitor
Enhanced TSPI
Enhanced TSPI
Access
HTTP
DECT R4 Provisioning
Web Services
Security Read All
Security Write All
Security Write Own
Password
Config Read All
Config Write All
Backup
Restore
Upgrade
External
Voicemail Pro Basic
Voicemail Pro
Standard
Voicemail Pro
Administrator
One-X Portal
Administrator
one-X Portal Super
User
Web Control
Administrator
Web Control Security
WebRTC Administrator
Related links
Security Users on page 134
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Security Administration
Access Control
Access to configuration, security settings and SSA is controlled by the use of service users,
passwords and Rights Groups. All actions involving communications between the Manager user and
the system require a service user name and password. That service user must be a member of a
Rights Group configured to perform the required action.
In the example illustrated above:
• Service user X can read and write the configuration. However they can only edit Operator
settings and can only make changes that can be merged.
• Service user Y can read and write the configuration, edit all settings and make changes that
require reboots.
• Service user Z can read and write the configuration, edit all settings and make changes that
require reboots. They can also access the security settings.
• The Security Administrator can only access the security settings.
Related links
Security Administration on page 134
Encryption
Encryption ensures that all data sent by either the system or Manager cannot be ‘read’ by anyone
else, even another copy of Manager. Encryption is the application of a complex mathematical
process at the originating end, and a reverse process at the receiving end. The process at each end
uses the same ‘key’ to encrypt and decrypt the data:
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Message Authentication
Any data sent may be optionally encrypted using a number of well known and cryptographically
secure algorithms:
Algorithm
Effective key size (bits)
Use
DES-40
40
Not supported.
DES-56
56
Not supported.
3DES
112
‘Minimal’ security.
RC4-128
128
‘Acceptable’ security.
AES-128
128
‘Strong’ security.
AES-256
256
‘Strong’ security.
In general the larger the key size, the more secure the encryption. However smaller key sizes
usually incur less processing. The system supports encryption using the Transport Layer Security
(TLS) v1.0 protocol. In addition, many cryptographic components of the TLS module have been
FIPS 140-2 certified, indicating the accuracy of implementation.
Related links
Security Administration on page 134
Message Authentication
Message authentication ensures that all data sent by either the system or Manager cannot be
tempered with (or substituted) by anyone else without detection. This involves the originator of the
data producing a signature (termed a hash) of the data sent, and sending that as well. The receiver
gets the data and the signature and check both match.
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Any data sent may be optionally authenticated using a number of well known and cryptographically
secure algorithms:
Algorithm
Effective hash size (bits)
Use
MD5
128
Not recommended.
SHA-1
160
‘Acceptable’ security.
SHA-2
224, 256, 384, 512
‘Strong’ security
In general the larger the hash size, the more secure the signature. However smaller hash sizes
usually incur less processing.
IP Office supports message authentication using the Transport Layer Security (TLS) 1.0, 1.1, and
1.2 protocol. In addition, many cryptographic components of the TLS module have been FIPS 140-2
certified, indicating the accuracy of implementation.
Related links
Security Administration on page 134
Certificates
Public key cryptography is one of the ways to maintain a trustworthy networking environment. A
public key certificate (also known as a digital certificate or identity certificate) is an electronic
document used to prove ownership of a public key. The certificate includes information about the
key, information about its owner's identity, and the digital signature of an entity that has verified the
certificate's contents are correct. If the signature is valid, and the person examining the certificate
trusts the signer, then they know they can use that key to communicate with its owner.
The system used to provide public-key encryption and digital signature services is called a public
key infrastructure (PKI). All users of a PKI should have a registered identity which is stored in a
digital format and called an Identity Certificate. Certificate Authorities are the people, processes and
tools that create these digital identities and bind user names to public keys.
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Certificates
There are two types of certificate authorities (CAs), root CAs and intermediate CAs. In order for a
certificate to be trusted and for a secure connection to be established, that certificate must have
been issued by a CA that is included in the trusted certificate store of the device that is connecting.
If the certificate was not issued by a trusted CA, the connecting device then checks to see if the
certificate of the issuing CA was issued by a trusted CA, and so on until either a trusted CA is found.
The trusted certificate store of each device in the PKI must contain the required certificate chains for
validation.
IP Office Root Certificate Authority
IP Office generates a self-signed certificate. For IP500 V2 systems, a certificate is generated
automatically on the first start up. On Linux systems, a certificate is generated during the ignition
process.
The following entities can act as the certificate authority.
• The Server Edition Primary Server, an Application Server, or a Unified Communication Module
(UCM) can act as the root certificate authority for all nodes in the system.
• In Enterprise Branch deployments, the System Manager can act as the root certificate
authority.
• Identity certificates can also be purchased and issued by a third party certificate authority.
Regardless of the method used to provide the IP Office identity, the certificate authority which signs
the IP Office identity certificate must be trusted by all the clients and endpoints that need to
establish a secure connection with IP Office. They must be a part of the PKI. Therefore, the root CA
certificate must be downloaded to client devices and placed in the trusted certificate store. If there
are intermediate CAs in the certificate chain, either the intermediate CAs must be added to the client
device Trusted Certificate Store or the certificate chain must be advertised by IP Office in the initial
TLS exchange.
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Certificates and TLS
Telephony signaling like SIP messaging is secured using Transport Layer Security (TLS). TLS
provides communication security using certificates to authenticate the other end of the IP Link.
The message exchange in TLS is aimed at verifying the identity of the communicating parties and
establishing the keys that will be used to encrypt the signaling data between the two parties.
Typically, the server sends its identity certificate, either self-signed or signed by the CA, to the client.
The client must have the CA certificate in its trusted certificate store.
IP Office acts as the TLS server in its interactions with SIP telephony clients. This means that the
TLS application on the IP Office must be configured to listen for client connections by enabling TLS
in the SIP Registrar on the LAN1 and LAN2 interfaces.
Windows Certificate Store
The certificate store used by Manager to save and retrieve X509 certificates is the default one
provided by the Windows operating system. The Windows certificate store is relevant to any
application running on Windows that uses certificates for security, either TLS or HTTPS. For
example, the Avaya Communicator Client.
Warning:
Avaya accepts no responsibility for changes made by users to the Windows operating system.
Users are responsible for ensuring that they have read all relevant documentation and are
sufficiently trained for the task being performed.
Windows Certificate Store Organization
By default, certificates are stored in the following structure:
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Certificates
Each of the sub folders has differing usage. The Certificates - Current User area changes with the
currently logged-in windows user. The Certificate (Local Computer) area does not change with the
currently logged-in windows user.
Manager only accesses some of the certificate sub folder:
Certificates (Local Computer) Folder
Manager Use
Personal | Certificates
Folder searched by Manager 1st for matching
certificate to send to the system when requested.
Certificate matched by the subject name contained in
File | Preferences | Security | Certificate offered to
the system.
Folder accessed whenever ‘Local Machine
certificate store’ used for Security Settings.
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Security Administration
Certificates (Local Computer) Folder
Manager Use
Folder searched by Manager for matching certificate
when certificate received from the system, and File |
Preferences | Security | Manager Certificate
Checks = Medium or High.
Trusted Root Certification Authorities |
Certificates
Folder searched by Manager for matching parent
certificates when non-self signed certificate received
from the system, and File | Preferences | Security |
Manager Certificate Checks = Medium or High.
Certificates – Current User Folder
Manager Use
Personal | Certificates
Folder searched by Manager 2nd for matching
certificate (subject name) to send to the system
when requested. Certificate matched by the subject
name contained in File | Preferences | Security |
Certificate offered to the system.
Folder accessed whenever ‘Current User
certificate store’ used for Security Settings.
Folder searched by Manager for matching certificate
when certificate received from IP Office, and File |
Preferences | Security | Manager Certificate
Checks = Medium or High.
Trusted Root Certification Authorities |
Certificates
Folder searched by Manager for matching parent
certificates when non-self signed certificate received
from the system, and File | Preferences | Security |
Manager Certificate Checks = Medium or High.
Other People | Certificates
Folder searched by Manager for matching parent
certificates when non-self signed certificate received
from the system, and File | Preferences | Security |
Manager Certificate Checks = Medium or High.
Windows Certificate Store Import
In order to use certificates – either for security settings or Manager operation – they must be present
in the windows certificate store. Certificates may be placed in the store by the Certificate Import
Wizard. The Certificate Import Wizard can be used whenever a certificate is viewed. In order for
Manager to subsequently access this certificate the Place all certificate in the following store
option must be selected:
• If the certificate is to subsequently identify the system, the Other People folder should be used.
• If the certificate is to subsequently identify the Manager, the Personal folder should be used,
and the associated private key saved as well.
Certificate Store Export
Any certificate required outside of the Manager PC must be first saved in the Certificate store, then
exported.
If the certificate is to be used for identity checking (i.e. to check the far entity of a link) the certificate
alone is sufficient, and should be saved in PEM or DER format.
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Implementing Security
If the certificate is to be used for identification (i.e. to identify the near end of a link) the certificate
and private key is required, and should be saved in PKCS#12 format, along with a password to
access the resultant .pfx file.
Related links
Security Administration on page 134
Implementing Security
IP Office can be made a very secure. However, only a certain number of features are active by
default in order to ease the initial installation. If all Manager and system security settings are left at
default, no security mechanisms are active, other than the use of default service user names and
passwords. In addition, all legacy interfaces are active, and all configuration and security data is
sent unencrypted. Therefore, it is necessary to implement the configuration options listed here.
Additional setting may be necessary to further secure the individual deployment. Avaya is
presenting this information for guidance only; the customer is responsible for ensuring their system
is secure.
To improve IP Office security in practice, two main mechanisms are used:
• Activation of IP Office security features.
• Reduction of exposure to external or internal attack.
Minimum Security
A minimum security scenario could be where configuration data is open, but the security settings are
constrained: Any individual with the correct service user name and password can access the
configuration from any PC installation of Manager, no logging of access: Passwords can be simple,
and will never age.
• Change all default passwords of all service users and Security Administrator.
• Set the system Security Administration service security level to Secure, Low.
• Set the system service user Password Reject Action to None.
• Set the system Client Certificate Checks level to None (default).
• Set the system Minimum Password Complexity to Low (default).
• Set the system Previous Password Limit to zero (default).
• Set the system Password Change Period to zero (default).
• Set the system Account Idle Time to zero (default).
• Set certificate check level to low in Manager Security Preferences (default).
In addition, any PC installation of Manager can manage any IP Office.
Medium Security
A medium security scenario could be where both configuration and security settings are constrained
and a level of logging is required: Any individual with the correct service user name and password
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Security Administration
can access the configuration from any PC installation of Manager: Passwords cannot be simple, and
will age.
• Change all default passwords of all service users and Security Administrator
• Set the system Security Administration service security level to Secure, Medium.
• Set the system Configuration service security level to Secure, Medium.
• Set the system service user Password Reject Action to Log to Audit Trail (default).
• Set the system Client Certificate Checks level to None (default).
• Set the system Minimum Password Complexity to Medium.
• Set the system Previous Password Limit to non zero.
• Set the system Password Change Period to non zero.
• Set the system Account Idle Time to zero (default).
• Disable all the system Unsecured Interfaces.
• Set certificate check level to low in Manager Security Preferences (default).
Maximum Security
A maximum security scenario could be where both configuration and security settings are
constrained and a full level of logging is required: Certified individuals with the correct service user
name and password can access the configuration from specific PC installations of Manager:
Passwords cannot be simple, and will age: Manager can managed specific systems.
• Change all default passwords of all service users and Security Administrator
• Set the system Security Administration service security level to Secure, High.
• Set the system Configuration service security level to Secure, High.
• Set the system service user Password Reject Action to Log and Disable Account.
• Set the system Client Certificate Checks level to High.
• Set the system Minimum Password Complexity to High.
• Set the system Minimum Password Length to >8.
• Set the system Previous Password Limit to non zero (>5).
• Set the system Password Change Period to non zero.
• Set the system Account Idle Time to non zero.
• Set the system Session ID Cache to zero.
• Install valid, 1024 bits+, non self signed certificates (+private key) in all IP Office server
certificates, derived from a trusted certificate authority.
• Install the corresponding trusted CA certificate in each of the Manager’s windows certificate
stores.
• Install valid, 1024 bits+, non self signed certificate (+ private key) in all Manager Certificate
Stores.
• Install the corresponding certificates in all the system Certificate Stores of all permissible
Manager entities, and the trusted CA certificate.
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SRTP
• Disable all the system Unsecured Interfaces.
• Set Manager Certificate Checks level to high in Manager Security Preferences.
• Set Certificate offered to the system in Manager Security Preferences.
The above essentially locks the systems and corresponding Managers together. Only recognized
(by strong certificate) entities may communicate successfully on the service interfaces. All services
use strong encryption and message authentication.
The use of intermediate CA certificates can be used to overcome the limit of 6 maximum certificates
in each system Certificate Store.
Related links
Security Administration on page 134
SRTP
Secure Real-Time Transport Protocol (SRTP) refers to the application of additional encryption and
or authentication to VoIP calls (SIP and H.323). SRTP can be applied between telephones, between
ends of an IP trunk or in various other combinations.
IP Office supports:
• Individual configuration for RTP and RTCP authentication and encryption
• HMAC SHA1 as the authentication algorithm
• AES-CM as the encryption algorithm
• 80 (default) or 32 bit authentication tag
• Master key length of 128 bits
• Master salt length of 112 bits.
Configuring the use of SRTP at the system level is done on the System | VoIP Security tab using
the Media Security setting. The options are:
• Best Effort
• Disabled (default)
• Enforced
When enabling SRTP on the system, the recommended setting is Best Effort. In this scenario, IP
Office uses SRTP if supported by the other end, and otherwise uses RTP. If the Enforced setting is
used, and SRTP is not supported by the other end, the call is not established.
The system level setting can be overridden at the trunk or extension level. This can be used for
special cases where the trunk or extension setting must be different from the system settings.
If the system level setting is Enforced, and devices that do not support SRTP are connected to the
system, their extension level configuration must be Disabled or calls will fail. This extra
configuration would typically not be required if the system level setting is Best Effort.
SRTP is supported on SIP Lines, SM Lines, and IP Office Lines. SRTP is not supported on H.323 IP
trunks.
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Encrypted RTCP
IP Office supports unencrypted RTCP by default. This default is compatible with most Avaya
endpoints which do not currently support encrypted RTCP. To the extent possible, any type of
endpoint using SRTP with IP Office should use unencrypted RTCP for consistency with other
endpoints to allow for direct media.
IP Office supports RTCP encryption as a configurable option. In addition to system level
configuration, it can be turned on at the trunk and extension level. Therefore, RTCP encryption can
be configured as an exception for an entity which only supports encrypted RTCP. In such case there
will be no direct media SRTP between that entity and one that does not support encrypted RTCP,
and IP Office will relay the SRTP media.
Authentication
Authentication can be applied to both the voice part of calls (the RTP stream) and or to the control
signal associated with the call (the RTCP stream). By default, IP Office supports RTP encryption,
RTP authentication, RTCP authentication. Authentication is applied after encryption so that packets
can be authenticated at the remote end without having to be decrypted first.
• The method used for the initial exchange of authentication keys during call setup depends on
whether the call is using SIP or H.323. The IP Office system uses SDESC for SIP calls and
H235.8 for H.323 calls.
• SRTP is only supported when using an addition method such as TLS or a VPN tunnel to
establish a secure data path before call setup.
• A replay attack is when someone intercepts packets and then attempts to use them to for a
denial-of-service or to gain unauthorized access. Replay protection records the sequence of
packets already received. If a packed has been received previously, it is ignored. If packets
arrive outside a specified sequence range, the security device rejects them. All packets in a
stream (RTP and RTCP) have a sequential index number, however packets may not be
received in sequential order. SRTP protects against replay attacks by using a moving replay
window containing the index numbers of the last 64 authenticated packets received or
expected. Any packet received that has an index older than the current window is ignored. Only
packets with an index ahead of the window or inside the window but not already received are
accepted. Separate replay protection is used for the RTP and the RTCP streams.
• Rekeying is the sending of new authentication keys at intervals during an secure call. This
option is not supported by the IP Office system which just sends authentication keys at the start
of the call.
SRTP sessions can use direct media between the devices or can be relayed via the IP Office
system. In some scenarios the IP Office system can be one end of the SRTP part of a call that then
continues to a non-SRTP destination.
If both the call originator and target require SRTP: A direct media is made if supported, using
SRTP. If direct media is not supported, the call is relayed via the IP Office system. In either case
SRTP parameters are negotiated end to end with the IP Office system translating and forwarding
them from one end to other end if necessary.
If only the originator or target requires SRTP: A non-direct media call is setup with with SRTP
negotiated between the IP Office system and the party which requires SRTP.
Emergency Calls
Emergency calls from an extension are not blocked even if SRTP is required but cannot be
established.
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SRTP
Calls using SRTP do not use any special indication on the user's telephone. Normal call functions
(conference, transfer, etc) remain available to the user. SRTP alarms and details of when SRTP is
being used are shown by the System Status Application and System Monitor.
Related links
Security Administration on page 134
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Chapter 8: Editing IP Office Security
Settings in Manager
The following conditions apply when editing the IP Office security settings.
• Editing of security settings may only be done online to a system.
No offline saving or editing is allowed for security purposes.
• No errors in the security settings are allowed to persist.
This prevents the system becoming inaccessible through operator error.
• Sets of changes to security objects may be made without the need for the OK button to be
selected every time.
This allows a coordinated set of changes to be accepted or canceled by the operator.
Loading Security Settings
About this task
The address ranges in which Manager searches for systems are set through the Manager
preferences (File | Preferences | Discovery). The security mechanism used for security settings
transfer between Manager and a system are set through the Secure Communications attribute of
Manager preferences (File | Preferences | Security).
If not already done, switch Manager to security mode by selecting File | Advanced | Security
Settings.
Note:
If the system's configuration settings have already been loaded using a service user name and
Password that also has security access, then the security settings are automatically loaded
when Manager is switched to security mode.
Procedure
1. If already in security mode, click
Settings from the menu bar.
in the main toolbar or select File | Open Security
2. The Select IP Office window appears, listing those systems that responded.
The list can be sorted by clicking on the column names.
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Saving Security Settings
3. If the system required was not found, the address used for the search can be changed.
Enter or select the required address in the Unit/Broadcast Address field and then click
Refresh to perform a new search.
4. When the system required is located, check the box next to the system and click OK.
5. The user name and password request for the system is then displayed.
Enter the required details and click OK. By default this is a different user name and
password from those that can be used for configuration access.
6. If the security settings are received successfully, they appear within Manager.
• If the service user name/password is incorrect, or the service user has insufficient rights to
read the security settings, "Access Denied" is displayed.
• If the network link fails, or the secure communication mode is incorrect (for example
Manager is set to unsecured, but the system is set to secure only), "Failed to
communicate with IP Office" is displayed.
Saving Security Settings
About this task
Procedure
1. Click
in the Main Toolbar or select File | Save Security Settings from the menu bar.
These options are only available when some change has been made.
2. The user name and password request for the system is then displayed.
Enter the required details and click OK. By default this is a different user name and
password from those that can be used for configuration access.
Resetting a System's Security Settings
About this task
Procedure
1. Select File | Reset Security Settings (if in security mode), or File | Advanced | Erase
Security Settings (if in configuration mode).
2. The Select IP Office window appears, listing those systems that responded.
The list can be sorted by clicking on the column names.
3. When the system required is located, check the box next to the system and click OK.
4. The user name and password request for the system is then displayed.
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Editing IP Office Security Settings in Manager
Enter the required details and click OK. By default this is a different user name and
password from those that can be used for configuration access.
5. Manager will indicate if the security settings are reset successfully.
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Chapter 9: Security Mode Field
Descriptions
The Manager Security Mode is used to load and edit the security settings of a system. How the
controls operate is similar to Manager in configuration mode.
To switch to Security Mode, select File | Advanced | Security Settings.
To switch back to Configuration Mode, select File | Configuration.
Security Mode Screen Elements
Table 3: Toolbar icons
Icon
Action
Get the Security Settings
Save the Security Settings
Not Used in security mode
Show/Hide the Navigation Pane
Show/Hide the Group Pane
Not used in security mode
Not used in security mode
Security Settings Pane: This pane is used to select the type of security records that should be
displayed in the group pane or details pane.
•
•
•
•
General Defines general security controls for the system. When selected, the settings are
displayed in the details pane.
System Defines security settings for the system such as application access. When
selected, the settings are displayed in the details pane.
Services Secure services supported by the system. Currently these are access to security
settings and access to configuration settings.
Rights Groups Create groups with different access rights. When selected, the existing
Rights Groups are displayed in the group pane.
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Security Mode Field Descriptions
•
Service Users Sets the name and password for an administrator. Also allows selection of
the Rights Groups to which the user belongs. When selected, the existing service users are
displayed in the group pane.
Group Pane: This pane is used to display the existing Right Groups or Service Users when those
options are selected in the security settings pane.
Details Pane: This pane shows the settings selected in the security settings pane or the group
pane.
Status Bar: This bar display messages about communications between Manager and systems. It
also displays the security level of the communications by the use of a padlock icon.
Related links
General Security Field Descriptions on page 156
System on page 160
Security Services Settings on page 168
Rights Groups on page 169
Service Users on page 174
General Security Field Descriptions
Field
Description
Security Administrator
The Security Administrator is a special service user who does not belong to any Rights Groups. The Security
Administrator is able to access the system's security settings but cannot access its configuration settings. By
default they are the only service user able to access to the security settings.
Unique Security
Administrator
Default = Off
Name:
Default = 'security'. Range = 6 to 31 characters.
When selected, only the Security Administrator is able to access the system's security
settings. When this is selected, the security options for Rights Groups are disabled.
When not selected, the ability to access security settings can also be assigned to Rights
Groups.
The name for the Security Administrator.
Password
Default = 'securitypwd'. Range = 8 to 31 characters.
The password for the Security Administrator. In order to change the Security
Administrator password, the current password must be known.
Minimum Password
Complexity
February 2016
Default = Medium.
The password complexity requirements for the Security Administrator. This setting is
active for attempted password changes on both Security Manager and the system. The
options are:
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General Security Field Descriptions
Field
Description
Low:
Any password characters may be used without constraint.
Medium:
The password characters used must include characters from at least 2 of the 'code
point sets' listed below. For example a mix of lower case and upper case. In addition, 3
or more consecutive identical characters of any type is not allowed.
High:
The password characters used must include characters from at least 3 of the 'code
point sets' listed below. For example a mix of lower case, upper case and numbers. In
addition, 3 or more consecutive identical characters of any type is not allowed.
Code Point Sets:
• Lower case alphabetic characters.
• Upper case alphabetical character.
• Numeric characters.
• Non-alphanumeric characters, for example # or *.
Previous Password
Limit (Entries)
Default = 4. Range = 0 (Off) to 10 records.
The number of previous password to check for duplicates against when changing the
password. When set to 0, no checking of previous passwords takes place. This setting
is active for attempted password changes on both Security Manager and the system.
Service User Details
These settings control service user names and password/account policies. This setting is active for attempted
password changes on all administration interfaces.
Minimum Name
Length
Default = 6, Range 1 to 31 characters.
Minimum Password
Length
Default = 8, Range 1 to 31 characters. This field sets the minimum password length for
service user passwords.
Password Reject
Limit
Default = 3, Range 0 to 255 failures.
Password Reject
Action
Default = Log and Temporary Disable.
This field sets the minimum name length for service user names.
Sets how many times an invalid name or password is allowed within a 10 minute period
before the Password Reject Action is performed. Selecting 0 indicates never perform
the Password Reject Action.
The action performed when a user reaches the Password Reject Limit. The options
are:
• No Action
• Log to Audit Trail Log to Audit Trail creates a record indicating the service user
account name and time of last failure.
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Security Mode Field Descriptions
Field
Description
• Log and Disable Account:. Log and Disable Account creates an audit trail record
and additionally permanently disables the service user account. This account can only
be enabled using the Security Manager Service User settings.
• Log and Temporary Disable: Log and Temporary Disable creates an audit trail
record and additionally temporarily disables the service user account for 60 seconds.
This account can only be enabled using the Security Manager Service User settings.
Minimum Password
Complexity
Default = Medium.
The password complexity requirements for the Security Administrator. This setting is
active for attempted password changes on both Security Manager and the system. The
options are:
Low:
Any password characters may be used without constraint.
Medium:
The password characters used must include characters from at least 2 of the 'code
point sets' listed below. For example a mix of lower case and upper case. In addition, 3
or more consecutive identical characters of any type is not allowed.
High:
The password characters used must include characters from at least 3 of the 'code
point sets' listed below. For example a mix of lower case, upper case and numbers. In
addition, 3 or more consecutive identical characters of any type is not allowed.
Code Point Sets:
• Lower case alphabetic characters.
• Upper case alphabetical character.
• Numeric characters.
• Non-alphanumeric characters, for example # or *.
Previous Password
Limit (Entries)
Default = 4. Range = 0 (Off) to 10 records.
Password Change
Period
Default = 0 (Off). Range 0 to 999 days.
The number of previous password to check for duplicates against when changing the
password. When set to 0, no checking of previous passwords takes place. This setting
is active for attempted password changes on both Security Manager and the system.
Sets how many days a newly changed password is valid. Selecting 0 indicates any
password is valid forever. This setting is active for password changes through this form
or prompted by Manager. Note that the user must be a member of a Rights Group that
has the Security Administration option Write own service user password enabled. If
this timer expires, the service user account is locked. The account may only be reenabled using the Service User Settings. To prompt the user a number of days before
the account is locked set a Expiry Reminder Time (see below).
Whenever this setting is changed and the OK button is clicked, the Security Manager
recalculates all existing service user password timers.
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General Security Field Descriptions
Field
Description
Account Idle Time
Default = 0 (Off). Range 0 to 999 days.
Sets how many days a service user account may be inactive before it becomes
disabled. Selecting 0 indicates an account may be idle forever. If this timer expires, the
service user account is permanently disabled. The account may only be re-enabled
using the Service User Settings. The idle timer is reset whenever a service user
successfully logs in.
Whenever this setting is changed and the OK button is clicked, the Security Manager
recalculates all existing service user idle timers.
Expiry Reminder
Time
Default = 10. Range 0 (Off) to 999 days.
Sets the period before password or account expiry during which a reminder indication if
the service user logs in. Selecting 0 prevents any reminders. Reminders are sent, for
password expiry due to the Password Change Period (above) or due to the Account
Expiry date (see Service User setting) – whichever is the sooner. Currently Manager
displays reminders but System Status does not.
IP Office User Details
These settings control IP Office user password/account policies.
Password
Enforcement
Default = On.
Minimum Password
Length
Default = 8, Range 1 to 31 characters.
Minimum Password
Complexity
Default = Medium.
When enabled, password settings are enforced. When disabled, password
requirements are not enforced and the remaining settings are not editable
This field sets the minimum password length for user passwords
The password complexity requirements for the Security Administrator. This setting is
active for attempted password changes on both Security Manager and the system. The
options are:
Low:
Any password characters may be used without constraint.
Medium:
The password characters used must include characters from at least 2 of the 'code
point sets' listed below. For example a mix of lower case and upper case. In addition, 3
or more consecutive identical characters of any type is not allowed.
High:
The password characters used must include characters from at least 3 of the 'code
point sets' listed below. For example a mix of lower case, upper case and numbers. In
addition, 3 or more consecutive identical characters of any type is not allowed.
Code Point Sets:
• Lower case alphabetic characters.
• Upper case alphabetical character.
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Security Mode Field Descriptions
Field
Description
• Numeric characters.
• Non-alphanumeric characters, for example # or *.
Password Reject
Limit
Default = 5, Range 0 to 255 failures.
Password Reject
Action
Default = Log and Temporary Disable. The action performed when a user reaches the
Password Reject Limit. The options are:
Sets how many times an invalid name or password is allowed within a 10 minute period
before the Password Reject Action is performed. Selecting 0 indicates never perform
the Password Reject Action.
• No Action
• Log to Audit Trail Log to Audit Trail creates a record indicating the user account
name and time of last failure.
• Log and Disable Account Log and Disable Account creates an audit trail record and
additionally permanently disables the user account. The account can be enabled
using the Account Status field on the User | User page.
• Log and Temporary Disable Log and Temporary Disable creates an audit trail
record and additionally temporarily disables the user account for 60 seconds. The
account can be enabled using the Account Status field on the User | User page.
Related links
Security Mode Field Descriptions on page 155
System
Related links
Security Mode Field Descriptions on page 155
System Details on page 160
Unsecured Interfaces on page 162
Certificates on page 163
System Details
Field
Description
Base Configuration
Services Base TCP
Port
February 2016
Default = 50804. Range = 49152 to 65526.
This is the base TCP port for services provided by the system. It sets the ports on which
the system listens for requests to access those services, using its LAN1 IP address.
Each service uses a port offset from the base port value. If this value is changed from
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System
Field
Description
its default, the Manager application must be set to the same Base TCP Port through its
Services Base TCP Port setting (File | Preferences).
For information on port usage see the IP Office Port Matrix document on the Avaya
support site at https://support.avaya.com/helpcenter/getGenericDetails?
detailId=C201082074362003
Maximum Service
Users
Default = 64.
Maximum Rights
Groups
Default = 32.
This is a fixed value for indication purposes only. This value is the maximum number of
service users that can be stored in a system's security settings
This is a fixed value for indication purposes only. This value is the maximum number of
Rights Groups that can be stored in a system's security settings.
System Discovery
System discovery is the processes used by applications to locate and list available systems. The IP Office can
be disabled from responding to this process if required. If this is done, access to the IP Office requires its
specific IP address to be used.
TCP Discovery
Active
Default = On.
UDP Discovery
Active
Default = On.
Selecting TCP Discovery Active allows the system to respond to those requests.
Selecting UDP Discovery Active allows the system to respond to those requests.
Security
These settings cover the per-system security aspects, primarily TLS settings.
Session ID Cache
Default = 10 hours, Range 0 to 100 hours.
This sets how long a TLS session ID is retained by the system. If retained, the session
ID may be used to quickly restart TLS communications between the system and a reconnecting application. When set to 0, no caching takes place and each TLS
connection must be renegotiated.
HTTP Challenge
Timeout (Seconds)
Default = 10.
RFC2617 Session
Cache (Minutes)
Default = 10.
For HTTP/HTTPS connection attempts, this field sets the timeout for connection
validation responses.
For HTTP/HTTPS sessions, this field sets the allowed duration for successful logins as
per RFC2617.
HTTP Ports
HTTP Port
Default = 80.
HTTPS Port
Default = 443.
Web Services Port
Default =8443.
WebSocket Proxy
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Security Mode Field Descriptions
Field
Description
These settings are applicable to WebSocket communication over IP Office lines.
Enabled
Default = On.
When set to On , Web Manager uses the proxy server to communicate between the
Server Edition Primaryserver and other nodes.
When set to Off, the WebSocket proxy is disabled. Any IP Office line WebSocket
communication over an HTTP session is closed with “404 NotFound”.
Enforce Secure
Default = On.
Applicable only when the Enabled check box is On.
When set to On, any proxy communication over IP Office line Websocket uses HTTPS.
When set to Off, any HTTPS communication over IP Office line Websocket is
disallowed and the session is closed with “403 Forbidden”.
Related links
System on page 160
Unsecured Interfaces
These features relate to applications that access the system configuration settings using older
security methods.
Field
Description
System Password
Default = 'password'. Range = 0 to 31 characters.
The system password is used by Manager to upgrade IP Office IP500 V2 systems. Also
used for Monitor when the Monitor password setting is blank.
Voicemail Password Default = Blank. Range = 0 to 31 characters.
This password is required if a matching password is also set through the Voicemail Pro
client application. Typically no password is set.
Monitor Password
Default = Blank. Range = 0 to 31 characters.
This password, if set, is used by the System Monitor application. If this password is not
set, those applications use the system password. If changing this password with no
previous password set, enter the system password as the old password.
Use Service User
Credentials
Default = Off.
Applications
Controls
These check boxes control which actions the system will support for legacy
applications. Different combinations are used by the different applications. A summary
of the applications affected by changes is listed in the Application Support list.
Set to On to enable log in to the System Monitor application using the service user
credentials.
• TFTP Server: Default = On.
• TFTP Directory Read: Default = Off.
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System
Field
Description
• TFTP Voicemail: Default = Off.
• Program Code: Default = On.
• DevLink: Default = On.
• TAPI/DevLink3: Default = Off.
• HTTP Directory Read: Default = On. Allow the system's current directory records to
be accessed using HTTP.
• HTTP Directory Write: Default = On. Allow HTTP import to be used to place
temporary directory records into the directory.
Application Support
This panel is shown for information only. It indicates the effect on various applications of
the Application Controls selections.
Related links
System on page 160
Certificates
Services between the system and applications may, depending on the settings of the service being
used for the connection, require the exchange of security certificates. The system can either
generate its own certificate or certificates provided from a trusted source can be loaded.
Warning:
The process of 'on-boarding' (see Deploying Avaya IP Office™ Platform SSL VPN Services)
automatically adds a certificate for the SSL VPN to the system's security settings when the onboarding file is uploaded to the system. Care should be taken not to delete such certificates
except when advised by Avaya.
Field
Description
Identity Certificate:
The Identity Certificate is an X.509v3 certificate that identifies the system to a connecting client device (usually
a PC running a application). This certificate is offered in the TLS exchange when the system is acting as a TLS
server, which occurs when accessing a secured service. An identity certificate can also be used when IPOffice
acts as TLS client and the TLS server requires IPOffice to send client certificate.
By default, the system's own self-generated certificate is used. A certificate is advertised when the Service
Security Level is set to a value other than Unsecure Only. The certificate can take up to one minute to
generate. During this time, normal system operation is suspended. You can regenerate the certificate by
clicking Delete. Regenerating a certificate may impact system performance. Perform this action during a
maintenance window.
Use the Set command to replace the system generated certificate with an external certificate.
Offer Certificate
Default = On.
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Security Mode Field Descriptions
Field
Description
This is a fixed value for indication purposes only. This sets whether the system will offer
a certificate in the TLS exchange when the IP Office is acting as a TLS server, which
occurs when accessing a secured service.
Offer ID Certificate
Chain
Default = Off.
Signature
Default = SHA256/RSA2048.
When set to On, this setting instructs IP Office to advertise a chain of certificates in the
TLS session establishment. The chain of certificates is built starting with the identity
certificate and adding to the chain all certificates it can find in the IP Office Trusted
Certificate Store based on the Common Name found in the "Issued By" Subject
Distinguished Name field in each of the certificates in the chain. If the Root CA
certificate is found in the IP Office Trusted Certificate Store, it will be included in the
chain of certificates. A maximum of six certificates are supported in the advertised chain
of certificates.
This setting configures both the signature algorithm and the RSA key length to use
when generating the IP Office identity certificate. The options are:
• SHA256/RSA2048
• SHA1/RSA1024
If any other combinations are needed, the Security Administrator will need to construct
the IP Office identity certificate outside of Manager and use the Set action to install it.
Private Key
Default = System generated random value. A blank field is displayed.
Use this field to enter a private key. If you set a private key, it is only used in the case of
self-signed certificates. To set the private key, you must click Delete to generate a new
certificate.
Issued to
Default = IP Office identity certificate.
Common name of issuer in the certificate.
Default Subject
Name
Default = Blank.
Specifies a common name for the subject of this certificate. The subject is the endentity or system that owns the certificate (public key). Example:
ipoffice-0123456789AB.avaya.com
If the field is left blank, a system generated subject name is used.
Subject Alternative
Name(s)
Default = Blank.
The Subject Alternative Name (SAN) field allows a list of alternate names to be bound
to the subject of the certificate.
The input field will allow the user to enter multiple Subject Alternate Names, each
separated by the comma “,” character. Each SAN consists of a PREFIX, followed by the
colon “:” character, followed by VALUE. The list of allowed PREFIX strings are “DNS”,
“URI”, “IP”, “SRV”, and “email”. The VALUE can be any sequence of characters except
the comma. (The comma is reserved as a field separator.) The input field has a
maximum size limit of 511 characters.
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System
Field
Description
Example: ”DNS: ipoffice-0123456789AB.avaya.com, IP:
192.168.137.29, URI:http://avaya_example_url.com/,
email:jack@my_email_server.com”
Supported Subject Alternative Name types include: a DNS Name, a Uniform Resource
Identifier, an IP Address, an SRV record, or an electronic mail address. Although a
PREFIX must be specified to select the type of name, no validation is performed on the
value.
If this field is left blank, a system generated subject alternative name field value is used.
Set
Set the current certificate and associated private key. The certificate and key must be a
matching pair. The source may be
• Current User Certificate Store.
• Local Machine Certificate Store.
• File in the PKCS#12 (.pfx) format
• Pasted from clipboard in PEM format, including header and footer text.
This method must be used for PEM (.cer) and password protected PEM (.cer) files.
The identity certificate requires both the certificate and private key. The .cer format
does not contain the private key. For these file types select Paste from clipboard
and then copy the certificate text and private key text into the Certificate Text Capture
window.
IP Office supports certificates with RSA key sizes of 1024, 2048 and 4096 bits. The use
of RSA key size 4096 may impact system performance. The recommended key size is
2048.
IP Office supports signature algorithms of SHA-1, SHA-224, SHA-256, SHA-384 and
SHA-512. Using signature size larger than SHA-256 may impact system performance.
The recommended signature algorithm is SHA-256.
Using a file as the certificate source:
In Manager, when using the file option, the imported "p12" "pfx" or "cer" file for setting
the identity certificate can only contain the private key and identity certificate data. It
cannot contain additional Intermediate CA certificates or the Root CA certificate. The
Intermediate CA certificates or the Root CA certificate must be imported separately in
the IP Office Trusted Certificate Store.
This does not apply to Web Manager.
Note:
Web Manager does not accept the file of type "cer" with extension ".cer". This file
type can only be used in Manager.
View
View the current certificate. The certificate (not the private key) may also be installed
into the local PC certificate store for export or later use when running the manager in
secured mode.
Delete
Deletes the current certificate and the system generates a new certificate. This can take
up to one minute to generate. During this time, normal system operation is suspended.
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Security Mode Field Descriptions
Field
Description
Regenerating a certificate may impact system performance. Perform this action during a
maintenance window.
Use Different
Identity Certificate
for Telephony
Default = Off.
When set to Off, all secure communications use the default identity certificate and
settings.
When set to On, telephony related secure communications use a separate identity
certificate that must by set by the Security Administrator.
Received Certificate
Checks
(Management
Interface)
Default = None.
This setting is used configuration administration connections to the system by
applications such as Manager. When the Service Security Level of the service being
used is set to High, a certificate is requested by the system. The received certificate is
tested as follows:
• None: No extra checks are made (The certificate must be in date).
• Low: Certificate minimum key size 1024 bits, in date.
• Medium: Certificate minimum key size 1024 bits, in date, match to store.
• High: Certificate minimum key size 2048 bits, in date, match to store, no self signed,
no reflected, chain validation.
Received Certificate
Checks (Telephony
Endpoints)
Default = None.
This setting is used with IP telephony endpoints connecting to the system.
This setting is used by IP Office to validate the identity certificate offered by the other
end of TLS connection. IP Office does not support mutual authentication for SIP
terminals (an identity certificate is not installed in all SIP terminals). Therefore, IP Office
does not require a client certificate from a SIP terminal, only SIP and SM trunks.
The received certificate is tested as follows:
• None: No extra checks are made (The certificate must be in date).
• Low: Certificate minimum key size 1024 bits, in date.
• Medium: Certificate minimum key size 1024 bits, in date, match to store.
• High: Certificate minimum key size 2048 bits, in date, match to store, no self signed,
no reflected, chain validation.
Trusted Certificate
Store: Installed
Certificates
Default = A set of fixed Avaya provided Intermediate CA or Root CA certificates.
The certificate store contains a set of trusted certificates used to evaluate received
client certificates. Up to 25 X.509v3 certificates may be installed. The source may be:
• Current User Certificate Store.
• Local Machine Certificate Store.
• File in one of the following formats:
- PKCS#12 (.pfx)
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System
Field
Description
- PEM (.cer)
- password protected PEM (.cer)
- DER (.cer)
- password protected DER (.cer)
• Pasted from clipboard in PEM format, including header and footer text.
Add
Add a trusted certificate. The source may be:
• Current User Certificate Store.
• Local Machine Certificate Store.
• File in one of the following formats:
- PEM (.cer)
- password protected PEM (.cer)
- DER (.cer)
- password protected DER (.cer)
• Pasted from clipboard in PEM format, including header and footer text.
This method must be used for PKCS#12 (.pfx) files. The PKCS#12 (.pfx) format
contains a private key and a trusted certificate cannot contain a private key. For this
file type, select Paste from clipboard and then copy the certificate text into the
Certificate Text Capture window.
IP Office supports certificates with RSA key sizes of 1024, 2048 and 4096 bits. The use
of RSA key size 4096 may impact system performance. The recommended key size is
2048.
IP Office supports signature algorithms of SHA-1, SHA-224, SHA-256, SHA-384 and
SHA-512. Using signature size larger than SHA-256 may impact system performance.
The recommended signature algorithm is SHA-256.
View
View the current certificate. The certificate (not the private key) may also be installed
into the local PC certificate store for export or later use when running the manager in
secured mode.
Delete
Delete the current certificate.
SCEP Settings
The Simple Certificate Enrollment Protocol is a protocol intended to ease the issuing of certificates in a network
where numerous devices are using certificates. Rather than having to individually administer the certificate
being used by each device, the devices can be configured to request a certificate using SCEP.
These settings are relevant for IP Office Branch deployments.
These settings are not used in IP Office Standard mode.
Active
Default = Off.
Request Interval
(seconds)
Default = 120 seconds. Range = 5 to 3600 seconds.
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Security Mode Field Descriptions
Field
Description
SCEP Server IP/
Name
Default = Blank.
SCEP Server Port
Default = 80 for HTTP and 443 for HTTPS.
SCEP URI
Default = /ejbca/publicweb/apply/scep/pkiclient.exe
SCEP Password
Default = Blank.
Related links
System on page 160
Security Services Settings
This tab shows details of the services that the system runs to which service users can communicate.
Field
Description
Name
The name of the service. This is a fixed value for indication purposes only.
Host System
This field shows the system's name. This is a fixed value for indication purposes only.
Service Port
This is the port on which the system listens for attempts to access the service. The
routing of traffic to this port may need to be enabled on firewalls and network devices
between the service users and the system. The base port (TCP or HTTP) for each
service is offset by a fixed amount from the ports set in System Settings.
For information on port usage see the IP Office Port Matrix document on the Avaya
support site at https://support.avaya.com/helpcenter/getGenericDetails?
detailId=C201082074362003
Service Security
Level
Default = ‘Unsecure Only’.
Sets the minimum security level the service will support. See File | Preferences |
Security for the corresponding Manager application setting, which must be changed to
match the appropriate service access security settings.
Warning:
If the system does not already have an X509 security certificate, selecting a setting
other than Unsecure Only will cause the system to stop responding for a period
(less than a minute) while the system generates its own unique security certificate.
The options are:
• Unsecure Only This option allows only unsecured access to the service. The service's
secure TCP port, if any, is disabled. This or disabled are the only options supported for
the System Status Interface and Enhanced TSPI services.
• Unsecure + Secure This option allows both unsecured and secure (Low) access. In
addition, TLS connections are accepted without encryption, just authentication.
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Rights Groups
Field
Description
• Secure, Low This option allows secure access to that service using TLS, and
demands weak (for example DES_40 + MD5) encryption and authentication or higher.
The service's unsecured TCP port is disabled.
• Secure, Medium This option allows secure access to that service using TLS, and
demands moderate (for example DES_56 + SHA-1) encryption and authentication or
higher. The service's unsecured TCP port is disabled.
• Secure, High This option allows secure access to that service using TLS and
demands strong (for example 3DES + SHA-1) encryption and authentication, or higher.
In addition, a certificate is required from the client (usually Manager). See System
Details | Client Certificate Checks for tests made on the received certificate. The
service's unsecured TCP port is disabled.
• Disabled This option is available for the System Status Interface and Enhanced TSPI
services. If selected, access to the service is disabled.
Service Access
Source
For Server Edition systems, it is defaulted to Server Edition Manager. When set to
Server Edition Manager, the system can only be configured using Manager in its Server
Edition mode. When set to Unrestricted, the system can be configure using Manager in
its normal Simplified View or Advanced View modes.
Warning:
Opening the configuration of a Server Edition system in Manager running in any
mode other than Server Edition mode should be avoided unless absolutely
necessary for system recovery. Even in that case, Manager will not allow
renumbering, changes to the voicemail type and changes to H.323 lines.
For systems centrally managed using SMGR , it is defaulted to Avaya Aura System
Manager. When set to Avaya Aura System Manager, the system can only be
configured using SMGR in Branch Mode. When set to Unrestricted, the system can be
configured using Manager in its normal Simplified View or Advanced View modes.
Related links
Security Mode Field Descriptions on page 155
Rights Groups
Related links
Security Mode Field Descriptions on page 155
Group Details on page 170
Configuration on page 170
Security Administration on page 171
System Status on page 172
Enhanced TSPI on page 172
HTTP on page 173
Web Services on page 173
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Security Mode Field Descriptions
External on page 174
Group Details
This tab sets the name of the Rights Group.
Field
Description
Name
: Range = Up to 31 characters
The name for the Rights Group should be unique. The maximum number of rights groups
is 32.
Related links
Rights Groups on page 169
Configuration
This tab sets the configuration settings access for service user's who are members of this Rights
Group.
Field
Description
IP Office Service
Rights
This setting controls what action on the system can be performed by members of the
Rights Group.
Manager
Operator Rights
This setting controls what types of configuration records Manager will allow members of the
Rights Group to viewed and what actions they can perform with those types of records.
Operator
View/Edit/ New/Delete
Configuration Record
Types
Administrator
All
View, edit create and delete
all configuration records.
Manager
View
View all except WAN Port.
Edit
Extension, User, Hunt
Group, Short Code, Service,
RAS, Incoming Call Route,
Directory, Time Profile,
Firewall Profile, IP Route,
Least Cost Route, Account
Code, ARS.
New
Operator
February 2016
Delete
As edit except Short Code.
View
View all except WAN Port.
Edit
Extension, User, Hunt
Group, Short Code, Service,
RAS, Incoming Call Route,
Time Profile, Firewall Profile,
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Rights Groups
Field
Description
IP Route, Least Cost Route,
Account Code, License,
ARS.
User & Group Edit
New
None.
Delete
Delete Incoming Call Route
and Directory.
View
Edit
User and Hunt Group
records only.
New
None
Delete
User & Group Admin
All
User and Hunt Group
records only.
Dir & Account Admin
All
Directory and Account Code
records only.
Time & Attendant Admin
All
Time Profile and Auto
Attendant records only.
ICR & User Rights Admin
All
Incoming Call Route and
User Rights records only.
Read Only
View
View all configuration
records.
Edit
None.
New
Delete
Related links
Rights Groups on page 169
Security Administration
This tab sets the security settings access for Service user's who are members of this Rights Group.
These settings are ignored and greyed out if a Unique Security Administrator has been enabled in
General Settings.
Field
SDescription
Read all security
settings
Members of the Rights Group can view the system's security settings.
Write all security
settings
Members of the Rights Group can edit and return changes to the system's security
settings.
Reset all security
settings
If selected, members of the Rights Group can reset the security settings to default values.
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Security Mode Field Descriptions
Field
SDescription
Write own service
user password
If selected, members of the Rights Group can change their own password when requested
to do so by the system. That request may be the result of a Password Change Period,
Force new password or Account Expiry. The new password change is requested
automatically at login time.
Related links
Rights Groups on page 169
System Status
This tab sets whether members of the group can access the system using the System Status
Application (SSA).
Field
Description
System Status
Access
If selected, members of the Rights Group can view the system's current status and
resources using the System Status Application (SSA).
Read all
configuration
The System Status application includes tools to take a snapshot of the system for use by
Avaya for diagnostics. That snapshot can include a full copy of the system's configuration
settings. This setting must be enabled for the SSA user to include a copy of the
configuration in the snapshot.
System Control
If enabled, the SSA user is able to use SSA to initiate system shutdowns and memory
card shutdown/restarts.
SysMonitor
Access
If enabled, members of the Rights Group can use the System Monitor application to
perform detailed diagnosis of system problems.
Related links
Rights Groups on page 169
Enhanced TSPI
This tab sets whether members of the group can access the system using the Enhanced TSPI
application interface.
Field
Description
Enhanced TSPI
Access
If selected, applications in this rights group are able to use the system's Enhanced
TSPI interface. This interface is currently used by the one-X Portal application server
for its connection to the system.
Related links
Rights Groups on page 169
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Rights Groups
HTTP
This tab sets the HTTP services supported for members of the group.
Field
Description
DECT R4
Provisioning
This service is used to allow the system to configure the DECT R4 master base station
and to respond to handsets subscribing to the DECT R4 system. It requires both the
system and DECT R4 master base station to be configured to enable provisioning. For
full details refer to the DECT R4 Installation Manual.
Related links
Rights Groups on page 169
Web Services
These settings are used by users in rights groups using web services to configure and manage the
system. These are currently not used on Standard Mode systems
IP Office Service Rights
Field
Description
Security Read All
If selected, the rights group members can view system security settings.
Security Write All
If selected, the rights group members can change system security settings.
Security Write
Own Password
If selected, members of the Rights Group can change their own password when
requested to do so by the system. That request may be the result of a Password Change
Period, Force new password or Account Expiry. The new password change is requested
automatically at login time.
Config Read All
If selected, the rights group members can view system configuration settings
Config Write All
If selected, the rights group members can change system configuration settings.
Backup
If selected, the rights group members can initiate the system backup process.
Restore
If selected, the rights group members can initiate the system restore process.
Upgrade
If selected, the rights group members can initiate the system upgrade process.
Web Manager Rights
These rights are used with web service access to a system such as the IP Office web manager used
with IP Office Basic Edition systems.
Field
Description
Business Partner
Role
Service Provider
Role
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Security Mode Field Descriptions
Field
Description
Administrator Role
End User Role
Related links
Rights Groups on page 169
External
These settings are used by users in rights groups for external components using web services to
configure and manage the system.
IP Office Service Rights
Field
Description
Voicemail Pro
Basic
If selected, the rights group members can read the configuration and perform backup,
restore, and upgrade.
Voicemail Pro
Standard
If selected, the rights group members can update the configuration and perform backup,
restore, and upgrade.
Voicemail Pro
Administrator
If selected, the rights group members can update the configuration and security settings.
one-X Portal
Administrator
If selected, the rights group members can update the configuration and security settings.
Does not include backup and restore.
one-X Portal
Super User
If selected, the rights group members can perform backup and restore.
Web Control
Administrator
If selected, the rights group members can update the configuration settings.
Web Control
Security
If selected, the rights group members can update the security settings.
WebRTC Gateway
Administrator
If selected, the rights group members can update the configuration settings.
Related links
Rights Groups on page 169
Service Users
These settings are displayed when Service Users is selected in the navigation pane and a
particular service user is selected in the group pane.
The maximum number of service users is 64.
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Service Users
Field
Description
Name:
Range = Up to 31 characters. Sets the service user's name.
The minimum name length is controlled through General settings.
Note:
If changing the user name and/or password of the current service user used to load
the security settings, after saving the changes Manager should be closed. Not
closing Manager will cause error warnings when attempting to send any further
changes.
Password:
Range = Up to 31 characters. Sets the service user's password.
To change the current password click Change. Enter and confirm the new password. Note
that an error will be indicated if the password being entered does not meet the password
rules set through General settings.
To clear the cache of previous password details used by the password rules setting, click
Clear Cache. For example, if the rule restricting the reuse of old passwords has been
enabled, clearing the cache allows a previous password to be used again.
Account Status
Default = Enabled.
Displays the current service user account status (correct at the time of reading from the
system). The options are:
• Enabled This status is the normal non-error state of a service user account. This setting
can be selected manually to re-enable an account that has been disabled or locked.
Note that re-enabling a locked account will reset all timers relating to the account such
as Account Idle Time.
• Force New Password This status can be selected manually. The service user is then
required to change the account password when they next log in. Until a password
change is successful, no service access is allowed. Note that the user must be a
member of a Rights Group that has the Security Administration option Write own
service user password enabled.
• Disabled This status prevents all service access. This setting can be selected manually.
The account can be enabled manually by setting the Account Status back to Enabled.
• Locked – Password Error This status indicates the account has been locked by the
Password Reject Action option Log and Disable Account on the security General
Settings tab. The account can be enabled manually by setting the Account Status back
to Enabled.
• Locked - Temporary This status indicates the account is currently locked temporarily
by the Password Reject Action option Log and Temporary Disable on the security
General Settings tab. The account can be enabled manually by setting the Account
Status back to Enabled, otherwise the service user must wait for the 10 minute period
to expire.
• Locked - Idle This status indicates the account has been locked by passing the number
of days set for the Account Idle Time on the security General Settings tab without being
used. The account can be enabled manually by setting the Account Status back to
Enabled.
• Locked - Expired This status indicates the account has been locked after passing the
Account Expiry date set below. The account can be enabled manually by setting
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Security Mode Field Descriptions
Field
Description
Account Status back to Enabled, and resetting the Account Expiry date to a future
date or to No Account Expiry.
• Locked – Password Expired This status indicates the account has been locked after
having not been changed within the number of days set by the Password Change
Period option on the security General Settings tab. The account can be enabled
manually by setting the Account Status back to Enabled.
Account Expiry
Default = <None> (No Expiry).
Not applicable to Web Manager.
This option can be used to set a calendar date after which the account will become
locked. The actual expiry time is 23:59:59 on the selected day. To prompt the user a
number of days before the expiry date, set an Expiry Reminder Time on the security
General Settings tab.
Rights Group
Membership
The check boxes are used to set the Rights Groups to which the user belongs. The
user's rights will be a combination of the rights assigned to the groups to which they
belong.
Related links
Security Mode Field Descriptions on page 155
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Chapter 10: Editing Configuration Settings
Before editing the system's configuration settings, it is important to understand how those settings
are stored and used by the system.
The control unit holds copies of its configuration in both its internal non-volatile and RAM memory. A
copy is also held on the System SD card (IP500 V2).
The copies in non-volatile memory and System SD card, are retained even if power to the control
unit is removed.During power up, the system loads the configuration file stored on the System SD
card into its RAM memory. Other systems load the configuration stored in non-volatile memory into
RAM memory. The copy in RAM memory is then used to control the system's operation.
If the system encounters a problem using the configuration file in its System SD card's /primary
folder, it attempt to use the copy in its non-volatile memory. For fully details of the IP500 V2 boot
process and SD card usage refer to the IP Office Installation Manual.
Users actions such as changing their forward destinations or mailbox passcode are written to the
configuration in RAM memory.
Changes made using Manager are written to the configuration in non-volatile memory and then
copied into the RAM memory and System SD.
Between 00:00 and 00:30, a daily backup occurs which copies the configuration in the system's
operation RAM memory back into its non-volatile memory and, on IP500 V2 system's, the System
SD card.On IP500 V2 system, the contents of the system memory cards /primary folder can then
also be automatically copied to the /backup folder by enabling System | System | Automatic
Backup.
When the system is shutdown using the correct shutdown method, the configuration in RAM
memory is copied to the non-volatile memory and System SD card.
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Editing Configuration Settings
Using Manager
When using Manager to edit the configuration settings, the following need to be remembered:
• Manager receives the current configuration settings from RAM memory. Therefore the
configuration it receives includes any changes made by users up to that time. However it will
not contain any subsequent changes made by users.
• When sending the configuration settings back to the system, Manager allows two choices,
reboot or merge.
• Reboot sends the configuration to the system's non-volatile memory along with an instruction
to reboot. Following the reboot, the new configuration in non-volatile memory is copied to the
RAM memory and used.
• Merge sends the configuration to the system's non-volatile memory without rebooting. The
system then copies those changes that are mergeable into the RAM memory. A key point here
is that not all configuration settings are mergeable, see the Mergeable Settings list.
As a result of the above, it is important to bear the follow scenarios in mind:
• Changes made by users after a configuration is received by Manager may be lost when the
configuration is sent back from Manager. Therefore it is preferable to always edit a recently
received copy of the configuration rather than one that has been open for a period of time.
• If a merge is attempted with non-mergeable changes, those items will be written to the nonvolatile memory but will not be copied to RAM memory. If a daily backup occurs, they will then
be overwritten by the RAM. If a power loss reboot occurs, they will be written to RAM memory.
Related links
Mergeable Settings on page 179
Configuration Size on page 188
Setting the Discovery Addresses on page 190
Opening a Configuration from a System on page 191
Opening a Configuration Stored on PC on page 193
Known System Discovery on page 194
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Mergeable Settings
Creating New Records on page 196
Creating an Offline Configuration on page 198
Importing and Exporting Settings on page 199
Copying and Pasting on page 202
Saving a Configuration onto PC on page 202
Sending a Configuration on page 203
Erasing the Configuration on page 205
Default Settings on page 205
Mergeable Settings
The table below shows the configuration records for which changes can be merged and those that
require a system reboot. The Send Configuration menu shown when sending a configuration to the
system automatically indicates when the configuration is mergeable.
Configuration Setting
Merge
able
Notes
System | System
Yes
These settings are mergeable with the exception of
Locale and Favor RIP Routes over Static Routes.
Changing these settings requires a reboot of the
system.
System | LAN | LAN Settings
No
System | LAN | VoIP
No
The following settings are mergeable:
• Auto-create Extn
• Auto-create User
• H.323 Signalling over TLS
• Remote Call Signalling Port
• Enable RTCP Monitoring on Port 5005
• RTCP collector IP address for phones
• Scope
• Initial keepalives
• Periodic timeout
• VLAN
• 1100 Voice VLAN Site Specific Option Number
(SSON)
• 1100 Voice VLAN IDs
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Editing Configuration Settings
Configuration Setting
Merge
able
Notes
The remaining settings are not mergeable. Changes
to these settings will require a reboot of the system.
System | LAN | Network Topology
No
System | LAN | DHCP Pools
Yes
These settings are mergeable. However, the
following actions require a merge with service
disruption.
• Changing the Start Address, Subnet Mask or
Default Router value for an existing DHCP Pool
of addresses.
• Decreasing Pool Size for an existing DHCP Pool
of addresses.
• Deleting an existing DHCP Pool of addresses.
When these actions are performed, the DHCP
(Server or DialIn) is re-initialized which triggers a
reboot of the Avaya DHCP Clients (H.323 and SIP)
in order to force the Avaya DHCP clients to renew
their IP address lease and apply the new settings.
For the remaining Avaya and non-Avaya DHCP
clients, you must manually reboot the devices in
order to force the IP Addresses lease renewal.
Otherwise, the devices continue to use the allocated
IP addresses until the IP addresses lease time out
expires. IP address lease time out is set to three
days.
Note that IP Office supports phone reboot only for
E129 and B179 SIP phone models.
System | DNS
No
System | Voicemail
Yes
These settings are mergeable with the exception of
Voicemail Type and Voicemail IP Address.
Changes to these settings requires a reboot of the
system.
System | Telephony | Telephony
Yes
Changes to Companding LAW and Automatic
Codec Preference require a reboot.
System | Telephony | Park and Page
Yes
System | Telephony | Tones and Music
Yes
• Changes to Disconnect Tone requires a reboot.
• Changes to Busy Tone Detection requires a
reboot.
• Deleting any of the hold music Alternate Sources
requires a reboot.
System | Telephony | Ring Tones
Yes
System | Telephony | SM
Yes
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Mergeable Settings
Configuration Setting
Merge
able
System | Telephony | Call Log
Yes
System | Telephony | TUI
Yes
System | Directory Services | LDAP
Yes
System | Directory Services | HTTP
Yes
System | System Events | Configuration
No
System | System Events | Alarms
No
System | SMTP
No
System | SMDR
Yes
System | VCM
Yes
System | CCR
Yes
System | Codecs
Yes
System | VoIP Security
Yes
System | Dialer
Yes
Notes
Changes to the Operation field or to the Trunk
Range / IP Office table require a reboot.
It is recommended that you do not change the
mergeable settings while the system is in use.
System | Contact Center
Yes
Line | Analog Line | Line Settings
Yes
These settings are mergeable with the exception of
the Network Type setting. Changes to this setting
will require a reboot of the system.
Line | Analog Line | Analog Options
No
The Allow Analog Trunk to Trunk Connect setting
is mergeable. The remaining settings are not
mergeable. Changes to these settings will require a
reboot of the system.
Line | BRI Line
No
The following settings are not mergeable. Changes
to these settings will require a reboot of the system.
• Line Sub Type
• Network Type
• TEI
• Add 'Not-end-to-end ISDN' Information
Element
• Progress Replacement
• Clock Quality
• Force Number Plan to ISDN
Decreasing the Number of Channels setting
requires a a “merge with service disruption”. When
the configuration file is sent to the system, active
calls on the deleted channels are cleared.
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Editing Configuration Settings
Configuration Setting
Merge
able
Notes
The remaining settings are mergeable.
Line | BRI Line | Channels
Yes
Line | E1 PRI Line
No
The following settings are not mergeable. Changes
to these settings require a system reboot.
• Line Sub Type
• Network Type
• TEI
• Channel Allocation
• CRC Checking
• Clock Quality
• Add 'Not-end-to-end ISDN' Information
Element
• Progress Replacement
• Force Number Plan to ISDN
• Line Signalling
Decreasing the Number of Channels setting
requires a a “merge with service disruption”. When
the configuration file is sent to the system, active
calls on the deleted channels are cleared.
The remaining settings are mergeable.
Line | E1 Short Codes
Yes
Line | E1 PRI Channels
No
The following settings are mergeable:
• Line Appearance ID (ETSI, ETSI CHI)
• Admin (ETSI CHI)
• Incoming Group ID (ETSI CHI)
• Outgoing Group ID (ETSI CHI)
The remaining settings are not mergeable. Changes
to these settings require a system reboot.
Line | E1–R2 Options
No
Changing the Admin setting is mergeable.
The remaining settings are not mergeable. Changes
to these settings will require a reboot of the system.
Line | E1–R2 Channels
No
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Admin
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Mergeable Settings
Configuration Setting
Merge
able
Notes
The remaining settings are not mergeable. Changes
to these settings require a system reboot.
Line | E1–R2 MFC Group
No
Line | E1–R2 Advanced
No
Line | US T1 Line
No
The following settings are mergeable:
• Admin
• Prefix
• Send original calling party for forwarded and
twinning calls
• Originator number for forwarded and twinning
calls
The remaining settings are not mergeable. Changes
to these settings require a system reboot.
Line | T1 Channels
No
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
The remaining settings are not mergeable. Changes
to these settings require a system reboot.
Line | T1 ISDN Line
No
The following settings are mergeable:
• Prefix
• Send Redirecting Number
• Admin
• Send original calling party for forwarded and
twinning calls
• Originator number for forwarded and twinning
calls
The remaining settings are not mergeable. Changes
to these settings require a system reboot.
Line | T1 ISDN Channels
No
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
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Editing Configuration Settings
Configuration Setting
Merge
able
Notes
The remaining settings are not mergeable. Changes
to these settings require a system reboot.
Line | T1 ISDN TNS
No
Line | T1 ISDN Special
No
Line | T1 ISDN Call by Call
No
Line | H.323 Line | VoIP Line
No
Line | H.323 Line | Short Codes
No
Line | H.323 Line | VoIP Settings
No
Line | IP DECT Line | Line
Yes
Line | IP DECT Line | Gateway
Yes
Line | IP DECT Line | VoIP
Yes
Line | SIP Line | SIP Line
Yes
For line subtypes ETSI, ETSI CHI, QSIG A, and
QSIG B, these settings are mergeable. Changes to
these settings do not require a reboot of the system.
When creating an IP DECT line, these settings are
mergeable. You can also remove an IP DECT line
without rebooting. Changing an IP DECT line that
has been imported into the configuration is not
mergeable.
These settings are mergeable with the exception of
the Line Number setting. Changing the Line
Number setting requires a “merge with service
disruption”. When the configuration file is sent to the
system, the SIP trunk is restarted and all calls on the
line are dropped.
Deleting a SIP line requires a “merge with service
disruption”.
Line | SIP Line | Transport
Yes
Changing these settings requires a “merge with
service disruption”. When the configuration file is
sent to the system, the SIP trunk is restarted.
Note that ITSP Proxy Address and Calls Route via
Registrar are mergeable.
Deleting a SIP line requires a “merge with service
disruption”.
Line | SIP Line | SIP URI
Yes
These settings are mergeable, with the following
exceptions.
• Registration
• Identity Header
These settings require a “merge with service
disruption”. When the configuration file is sent to the
system, the SIP trunk is restarted and all calls on the
line are dropped.
Line | SIP Line | VoIP
Yes
Line | SIP Line | T38 Fax
Yes
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Mergeable Settings
Configuration Setting
Merge
able
Line | SIP Line | SIP Credentials
Yes
Line | SIP Line | SIP Advanced
Yes
Line | SIP Line | Engineering
Yes
Line | SIP DECT Line | SIP DECT Base
No
Line | SIP DECT Line | VoIP
No
Line | SM Line | Session Manager
No
Notes
These settings are mergable, with the exception of
the Media Connection Preservation setting.
Changing the Media Connection Preservation
setting requires a “merge with service disruption”.
When the configuration file is sent to the system, the
SIP trunk is restarted and all calls on the line are
dropped.
These settings are not mergeable. Changes to these
settings will require a reboot of the system.
Changing the In Service setting to Disabled (out of
service) requires a system reboot. However,
changing the In Service setting to Enabled is
mergeable. Configuration changes made while the
line is out of service are also mergeable.
Line | SM Line | VoIP
No
Line | SM Line | T38 Fax
Yes
Line | IP Office Line | Line
No
Line | IP Office Line | Short Codes
No
For line subtypes ETSI, ETSI CHI, QSIG A, and
QSIG B, these settings are mergeable. Changes to
these settings do not require a reboot of the system.
Line | IP Office Line | VoIP Settings
Yes
For line subtypes ETSI, ETSI CHI, QSIG A, and
QSIG B, these settings are mergeable. Changes to
these settings do not require a reboot of the system.
Line | IP Office Line | T38 Fax
Yes
Control Unit | Unit
No
Extension | Extn
No
Extension | Analog Extension | Analog
No
Extension | H323 Extension | VoIP
No
Extension | SIP Extension | VoIP
No
Extension | SIP Extension | T38 Fax
Yes
Extension | IP DECT Extension
Yes
Extension | SIP DECT Base
No
These settings are mergeable with the exception of
the Reserve License setting. Changing the
Reserve License settings requires a reboot of the
system.
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Editing Configuration Settings
Configuration Setting
Merge
able
User | User
Yes
User | Voicemail
Yes
User | DND
Yes
User | Short Codes
Yes
User | Source Numbers
Yes
User | Telephony | Call Settings
Yes
User | Telephony | Supervisor Settings
Yes
User | Telephony | Multi-line Options
Yes
User | Telephony | Call Log
Yes
User | Telephony | TUI
Yes
User | Forwarding
Yes
User | Dial In
Yes
User | Voice Recording
Yes
User | Button Programming
Yes
User | Menu Programming | T3
Telephony
Yes
User | Menu Programming | Hunt Group
Yes
User | Menu Programming | 4400/6400
Yes
User | Mobility
Yes
User | Group Membership
Yes
User | Announcements
Yes
User | SIP
Yes
User | Personal Directory
Yes
User | Web Self Administration
Yes
Group | Group
Yes
Group | Queuing
Yes
Group | Overflow
Yes
Group | Fallback
Yes
Group | Voicemail
Yes
Group | Voice Recording
Yes
Group | Announcements
Yes
Group | SIP
Yes
Short Code | Short Code
Yes
Service | Service
Yes
Service | Bandwidth
Yes
Notes
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Mergeable Settings
Configuration Setting
Merge
able
Service | IP
Yes
Service | Autoconnect
Yes
Service | Quota
Yes
Service | PPP
Yes
Service | Fallback
Yes
Service | Dial In
Yes
SSL VPN Service | Service
Yes
SSL VPN Service | Session
Yes
SSL VPN Service | NAPT
Yes
SSL VPN Service | Fallback
Yes
RAS | RAS
Yes
RAS | PPP
Yes
Incoming Call Route | Standard
Yes
Incoming Call Route | Voice Recording
Yes
Incoming Call Route | Destinations
Yes
WAN Port | WAN Port
No
WAN Port | Frame Relay
No
WAN Port | DLCIs
No
WAN Port | Advanced
No
Directory | Directory Entry
Yes
Time Profile | Time Profile
Yes
Firewall Profile | Standard
Yes
Firewall Profile | Custom
Yes
Firewall Profile | Static NAT
Yes
IP Route | IP Route
Yes
IP Route | RIP Dynamic Routing
Yes
Account Code | Account Code
Yes
Account Code | Voice Recording
Yes
License | License
Yes
License | Remote Server
No
Tunnel | Tunnel (L2TP)
No
Tunnel | Main (IPSec)
No
Tunnel | IKE Policies (IPSec)
No
Notes
The Reserved Licenses setting is mergeable. The
remaining settings are not mergeable. Changes to
these settings requires a reboot of the system.
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Editing Configuration Settings
Configuration Setting
Merge
able
Tunnel | IKE Policies (IPSec)
No
Auto Attendant | Auto Attendant
Yes
Auto Attendant | Actions
Yes
Authorization Codes
Yes
User Rights | User
Yes
User Rights | Short Codes
Yes
User Rights | Button Programming
Yes
User Rights | Telephony | Call Settings
Yes
User Rights | Telephony | Supervisor
Settings
Yes
User Rights | Telephony | Multi-line
Options
Yes
User Rights | Telephony | Call Log
Yes
User Rights | User Rights Membership
Yes
User Rights | Voicemail
Yes
User Rights | Forwarding
Yes
ARS | ARS
Yes
RAS Location Request
Yes
Location
Yes
Notes
Related links
Editing Configuration Settings on page 177
Configuration Size
The maximum size of the configuration file that can be loaded into an IP 500 V2 control unit is 2.0
MB.
When you attempt to save a configuration that is too large, you will be prompted and the save is
canceled.
During normal operation, additional configuration records can be added to the configuration without
using Manager (for example call log records and directory records made from phones). If, during the
overnight backup to flash memory, the configuration if found to be too large, records will be removed
until the configuration is sufficiently small to be backed up. The recordsremoved are call log
records, system directory records and then personal directory in that order. Note that those records
will still exist in the configuration running the system in its RAM memory, however if the system is
restarted they will disappear as the configuration is reloaded from the Flash memory.
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Configuration Size
Figures for all individual records in the configuration cannot be given as they vary. The list below
gives typical values, in bytes, for common records:
Physical Extension: 70.
IP Extension: 70.
User: 170.
User Short Code: 40.
DSS Button: 20.
Hunt Group: 100.
Hunt Group member: 10.
System Short Code: 10.
Normal Service: 220.
Intranet Service: 240.
WAN Service: 400.
RAS Service: 110.
Incoming Call Route: 30.
WAN Port (PPP): 70.
WAN Port (FR): 120.
Directory Record: 70.
Time Profile: 40.
Time Profile Record: 20.
Firewall Profile: 40.
Custom Firewall Record: 80.
IP Route (Static): 30.
License Key: 40.
Account Code: 40.
Logical LAN: 60.
Tunnel (L2TP): 200.
Tunnel (IPSec): 110.
Related links
Editing Configuration Settings on page 177
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Editing Configuration Settings
Setting the Discovery Addresses
By default, when
or File | Open configuration is selected, Manager's Select IP Office window
opens. It performs a UDP broadcast to the address 255.255.255.255. This broadcast will only locate
systems that are on the same network subnet as the PC running Manager.
For systems not located on the same subnet as the Manager PC, the following options are
supported.
Specific Addressing The Unit/Broadcast Address shown on the Select IP Office menu can be
changed to the specific IP address of the required system. A single address is routable and so can
be used to discover a system on another subnet.
TCP Discovery Address Ranges A set of TCP addresses and address ranges can be specified for
use by the Select IP Office discovery process.
Known System Discovery Manager can write the details of systems it discovers to a file. The list of
systems in that file can then be used for access to those systems. See Known System Discovery.
DNS Lookup Manager can be configured to locate systems using DNS name lookup. This requires
the systems on a customer network to be added as names on the customer's DNS server and the
Manager PC to be configured to use that server for DNS name resolution. The use of DNS is
configured through File | Preferences | Discovery.
Changing the Initial Discovery Settings The Discovery tab of the Preferences menu can be
used to set the UDP and TCP addresses used by the discovery process run by the Select IP Office
menu.
1. Select File | Preferences menu.
2. Select the Discovery tab.
TCP Discovery: Default = On. This setting controls whether Manager uses TCP to discover
systems. The addresses used for TCP discovery are set through the IP Search Criteria field below.
NIC IP/NIC Subnet This area is for information only. It shows the IP address settings of the LAN
network interface cards (NIC) in the PC running Manager. Double-click on a particular NIC to add
the address range it is part of to the IP Search Criteria. Note that if the address of any of the
Manager PC's NIC cards is changed, the Manager application should be closed and restarted.
IP Search Criteria This section is used to enter TCP addresses to be used for the TCP discovery
process. Individual addresses can be entered separated by semi-colons, for example
135.164.180.170; 135.164.180.175. Address ranges can be specified using dashes, for example
135.64.180.170 - 135.64.180.175.
UDP Discovery: Default = On This settings controls whether Manager uses UDP to discover
systems.
Enter Broadcast IP Address: Default = 255.255.255.255 The broadcast IP address range that
Manager should used during UDP discovery. Since UDP broadcast is not routable, it will not locate
systems that are on different subnets from the Manager PC unless a specific address is entered.
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Opening a Configuration from a System
Use DNS: Selecting this option allows Manager to use DNS name (or IP address) lookup to locate
a system. Note that this overrides the use of the TCP Discovery and UDP Discovery options above.
This option requires the system IP address to be assigned as a name on the users DNS server.
When selected, the Unit/Discovery Address field on the Select IP Office dialogue is replaced by a
Enter Unit DNS Name or IP Address field.
SCN Discovery: If enabled, when discovering systems, the list of discovered systems will group
systems in the same Small Community Network and allow them to be loaded as a single
configuration. At least one of the systems in the Small Community Network must be running
Release 6.0 or higher software. See Small Community Network Management. This does not
override the need for each system in the Small Community Network to also be reachable by the
TCP Discovery and or UDP Discovery settings above and accessible by the router settings at the
Manager location.
Related links
Editing Configuration Settings on page 177
Opening a Configuration from a System
The initial IP address ranges in which Manager searches for systems are set through the Manager
preferences (File | Preferences | Discovery). By default it scans the local network of the Manager
PC.
Start Manager. If Manager is already started and a configuration is open in it, that configuration must
be closed first.
• If Manager is set to Auto Connect on start up, it will scan for systems automatically and either
display the list of systems discovered or automatically start login to the only system discovered.
• Otherwise, click on
or select File | Open Configuration.
The Select IP Office window appears, listing those systems that responded.
• If Server Edition systems are detected, they are grouped together. By default the configuration
of those systems cannot be opened using Manager in Advanced View mode and the
configuration of a Primary Server can only be opened if the Open with Server Edition Manager
option is also selected. See Server Edition Mode.
• If Manager has been set with SCN Discovery enabled, systems in a Small Community Network
are grouped together. The checkbox next to the network name can be used to load the
configurations of all the configurations into Small Community Network management mode.
• If the system required was not found, the Unit/Broadcast Address used for the search can be
changed. Either enter an address or use the drop-down to select a previously used address.
Then click Refreshto perform a new search.
• The address ranges used by Manager for searching can be configured through the File |
Preferences | Discovery tab.
• A list of known systems can be stored and used. See Known System Discovery
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• Manager can be configured to search using DNS names. See the Use DNS option.
• Systems found but not supported by the version of Manager being used will be listed as Not
Supported.
• If the system detected is running software other than from its primary folder, a
warning icon
will be shown next to it. The configuration can still be opened but only as a read-only file.
When you have located the system required, check the box next to the system and click OK.
If the system selected is a Server Editionsystem and Manager is not running in Server Edition mode,
an Open with Server Edition Manager checkbox is shown and pre-selected. Clicking OK will
switch Manager to its Server Edition mode before loading the configuration.
The system name and password request is displayed. Enter the required details and click OK.
The name and password used must match a service user account configured within the system's
security settings.
Additional messages will inform you about the success or failure of opening the configuration from
the system.
The method of connection, secure or insecure, attempted by Manager is set the applications Secure
Communications preferences setting.
• When Secure Communications is set to On, a padlock icon is displayed at all times in the
lower right Manager status field.
• New installations of Manager default to having Secure Communications enabled. This means
Manager by default attempts to use secure communications when opening a configuration.
• For Server Edition systems, Manager will always attempt to use secure communications
regardless of the Secure Communications setting.
• If no response to the use of secure communication is received after 5 seconds, Manager will
offer to fallback to using unsecured communications.
Following a successful log in, the configuration is opened in Manager. The menus and options
displayed will depend on the type of system configuration loaded.
Login Messages
While attempting to login to a system, various additional messages may be displayed.
Configuration Not Loaded Messages
Access Denied This is displayed as the cause if the service user name/password were incorrect, or
the service user has insufficient rights to read the configuration. The Retry option can be used to log
in again but multiple rejections in a 10 minute period may trigger events, such as locking the user
account, set by the Password Reject Limit and Password Reject Action options in the systems
security settings.
Failed to communicate with system This is displayed as the cause if the network link fails, or the
secure communication mode is incorrect (for example Manager is set to unsecured, but the system
is set to secure only).
Account Locked The account of the service user name and password being used is locked. This
can be caused by a number of actions, for example too many incorrect password attempts, passing
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Opening a Configuration Stored on PC
a fixed expiry date, etc. The account lock may be temporary (10 minutes) or permanent until
manually unlocked. An account can be enabled again through the system's security settings.
Additional Messages
Your service user account will expire in X days This message indicates that an Account Expiry
date has been set on the system service user account and that date is approaching. Someone with
access to the system's security settings will be required unlock the account and set a new expiry
date.
Your password will expire in X days. Do you wish to change it now? This message indicates
that password ageing has been configured in the system's security settings. If your password
expires, someone with access to the system's security settings will be required to unlock the
account.
Change password Through the system's security settings, a service user account can be required
to change their password when logging in. The menu provides fields for entering the old password
and new password.
Contact Information Check - This configuration is under special control This message will
appear if a Manager user with administrator rights has entered their contact information into the
configuration. For example to indicate that they do not want the configuration altered while a
possible problem is being diagnosed. The options available are:
Cancel Select this option to close the configuration without making any changes.
Set configuration alteration flag Select this option if the configuration is being opened because
some urgent maintenance action. When the configuration is next opened, the fact that it has been
altered will be indicated on the System | System tab.
Delete Contact Information Select this option to take the system out of special control.
Leave contact information and flags unchanged (Administrators only) This option is only
available to service users logging in with administrator rights.
Related links
Editing Configuration Settings on page 177
Opening a Configuration Stored on PC
About this task
A configuration file previously saved on the PC can be reopened in Manager. This method of access
does not require entry of a service user name and password. All parts of the configuration are
visible.
Use either of the following processes to load a saved configuration file:
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Editing Configuration Settings
Procedure
1. Click
the main toolbar or select File | Offline | Open File from the menu bar. If the files is
one that has previously been opened offline, click the symbol next to
in the main
toolbar
2. An Open configuration file window appears.
Use this to browse to the required configuration file.
3. Select the file and click Open.
Related links
Editing Configuration Settings on page 177
Known System Discovery
The Select IP Office menu normally displays systems discovered by Manager using either UDP
broadcast and or TCP requests (see Setting the Discovery Addresses). Manager can be configured
to also record details of discovered units and then display a list of those previously discovered
('known') systems.
Related links
Editing Configuration Settings on page 177
Configuring Manager for Known System Discovery
About this task
Use of known systems discovery is not enabled by default. The Manager must be configured for the
feature with a file location to which it can store and retrieve known system details.
Procedure
1. Select File | Change Working Directory.
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Known System Discovery
2. In the Known Units File field, enter the directory path and file name for a CSV file into which
Manager can write details of the systems it discovers.
If the file specified does not exist it will be created by Manager.
3. Click OK.
Using Known System Discovery
About this task
Procedure
1. When the Select IP Office screen is displayed click on Known Units.
2. The screen displays the list of systems previously discovered and stored in the CSV file.
3. To select an control unit, highlight the row containing unit data and click OK.
The selected unit will appear in the Select IP Office window.
4. To filter displayed units, type the first few characters of the unit name in the Filter field.
Any unit whose name does not match the filter will be temporarily hidden.
5. Each discovery appends data to the known unit list.
It is possible that details of some records in the list may be out of date. Right clicking on the
leftmost (grey) column of any row will bring up a floating menu offering the options of
Refresh and Delete.
6. A new record may be manually added without having to access the system first through
normal discovery.
Enter the IP address of the new system in the IP Address column of the blank row shown
with a * and select Refresh from the floating menu. This will update the Known Units file with
data relating to the unit with the specified address.
7. Select Cancel to return to the Select IP Office menu.
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Editing Configuration Settings
Result
Note:
• The key used by the Known Systems CSV file is the IP address. The file cannot contain
records for separate systems that use the same IP address for access.
• The file can be made read only. In that case any attempts using Manager to update the file
will be ignored.
Creating New Records
There are a number of ways in which you can add new records to the configuration currently loaded
in Manager.
Related links
Editing Configuration Settings on page 177
Adding a New Record Using the Details Pane
About this task
Procedure
1. Use the navigation pane, groups pane or navigation toolbar, select an existing record of the
type required.
2. Click
at the top-right of the details pane.
3. Select the type of record required.
For example, with extensions you can select from H.323 Extension or SIP Extension.
4. Complete the settings for the new record and click OK.
Adding a New Record Using the Group Pane
About this task
Procedure
1. Using the navigation pane or navigation toolbar, select the type of record required.
2. Right-click on the group pane and select
New.
3. If a list is displayed, select the specific type of record required.
4. Complete the settings for the new record and click OK.
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Creating New Records
Adding a New Record Using the Navigation Pane
About this task
Procedure
1. In the navigation pane, right-click on the type of record required and select
New.
2. If a list is displayed, select the specific type of record required.
3. Complete the settings for the new record and click OK.
Other record creation methods
About this task
Procedure
1. Import records
New records can be created for users, hunt groups, short codes and directory entries by
importing files. See Importing and Exporting Settings.
2. Automatically create records
There are scenarios where new records are automatically added to the configuration:
• Certain default records are added to the configuration when a new system is started or
when a system configuration is erased.
• New records for extensions and lines are automatically added to match the hardware
present when a system is started or rebooted.
• For telephones connected via IP links (H.323, SIP and DECT), the system can be
configured to automatically create extension and or user records to match the telephone.
3. Use templates to create records
Manager supports a number of methods by which new records can be created from
templates.
Trunk Templates SIP trunks can be created from templates. For analog trunks, records are
automatically created by the presence of the trunk hardware, however a template can then
be applied to an analog trunk.
Server Edition When being used to edit the configuration of Server Edition systems,
Manager has a number of additional options to export existing records as templates and to
create new records from those templates.
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Editing Configuration Settings
Creating an Offline Configuration
About this task
Manager can be used to create a new configuration without connecting to a system. This allows the
creation of a configuration prior to installation of the real system and so can be used to speed up the
installation process.
The configuration created must match the physical equipment in the system into which the
configuration will be loaded. Doing otherwise may cause the system to reset and experience other
problems.
The Create Configuration tool includes all control units, external expansion modules and trunk
cards supported. It is your responsibility to confirm what equipment is supported in your locale.
Creating a New Configuration
Procedure
1. Close or save any configuration currently open.
2. Click
in the main toolbar or select File | Offline | Create New Config.
3. You should set the Configuration, Locale, Extension Number Length and System Unit
first.
Changing any of these after you start selecting other system hardware will reset the
hardware selections.
4. Select the type of Configuration you want to create.
The other options available will change depending on the selection. If the menu has been
started from Manager running in Server Edition mode, the only option is Server Edition
Edition.
5. Select the Locale for the system.
This defines a range of features such as default telephony settings.
6. The Extension Number Length setting value can be None or 3 to 15.
If a value is selected, all default extension, user and hunt group extension numbers created
by Manager will be that length. In addition Manager will display a warning if an extension
number of a different length is entered when editing the configuration.
7. Select the type of System Unit.
Select the hardware components for the system. For a Server Edition system this is only
necessary if a Expansion System (V2) is selected as the System Units option.
8. Select the additional cards to include in the control unit.
The number and type of cards selectable will depend on the control unit type.
9. Select the external expansion modules to also include in the system.
10. Click OK.
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Importing and Exporting Settings
11. For non-Server Edition systems, the configuration is created and loaded into Manager.
For Server Edition systems, the Initial Configuration menu for the selected type of system
unit is displayed. Complete the menu and click Save.
12. Once this configuration has been edited as required it can be saved on the PC or sent to a
system.
13. To Save a Configuration File on the PC Use File | Save Configuration.
14. To Send the Configuration to a System If the system which you want to use the
configuration is available, use File | Offline | Send Configuration to send the configuration to
it.
Warning:
This action will cause the system to reboot and will disconnect all current calls and
service.
• Ensure that you have a copy of the systems existing configuration before overwriting it
with the off-line configuration.
• After sending the configuration, you should receive the configuration back from the system
and note any new validation errors shown by Manager. For example, if using Embedded
Voicemail, some sets of prompt languages may need to be updated to match the new
configurations locale setting using the Add/Display VM Locales option.
Related links
Editing Configuration Settings on page 177
Importing and Exporting Settings
Manager can import configuration settings created elsewhere. This can be useful when setting up a
new system or sharing common settings such as a directory between systems.
The system supports LDAP (System | Directory Services | LDAP) for automatic importation of
directory records (LDAP Version 2).
The system also supports HTTP (System | Directory Services | HTTP) for automatic importation of
directory records.
Note:
You can also manage configuration settings using templates. See Working with Templates on
page 565.
Settings are imported and exported in the following formats:
• Binary Files (.exp) These are non-editable files. During import and export it is possible to
select what types of records should be included in the file. During import the whole file is
imported.
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Editing Configuration Settings
• Comma Separated Variable Text Files (.csv) These are plain text files. In addition to being
exported from a system these files can be created and edited using programs such as
WordPad or Excel.
When opening a .csv file in Excel it will alter the way some data is displayed, automatically changing
the display format of dates and long numbers such as phone numbers.
UTF-8 Character Encoding Manager imports and exports CSV files using UTF-8 character
encoding which uses a multiple bytes to support characters with diacritic marks such as ä. Other
applications, such as Excel, may, depending on the user PC settings, use different encoding which
will cause such characters to be removed or corrupted. Care should be taken to ensure that any tool
used to create or edit the CSV supports all the characters expected and uses UTF-8 format.
• Importing into Manager from Excel From Excel save the file as a .csv. This file will use ANSI
character encoding. Open the file in Notepad and use the Save As option to rename the file
and select UTF-8 encoding. Import the UTF-8 version of the file into Manager.
• Exporting from Manager into Excel Do not double-click on the file exported from Manager.
Start Excel and use File | Open to select the file. Excel will recognize that the file uses UTF-8
encoding and will start its text file importation wizard. Follow the wizard instructions and select
comma as the field delimiter.
CSV File Formats
The format is CSV using commas as field separator, no text delimiters and no header row. The
simplest way to check the required format for a CSV file prior to import, is to export and a file from
an existing system configuration.
File Name
Fields in Order
Configuration
Proprietary format. Note that this does not contain all configuration fields.
License
The License option is only available for export and only for ADI licenses
present on the system.
Short Code
Code, Telephone Number, Feature.
User
Name, Extension, User Rights, Email Address, Full Name, Password,
VoiceMail Code, Login Code, UserTemplate, ExtensionTemplate.
Directory
Name, Number, Speed Dial.
Group
Name, Extension, Group, Hunt, Rotary, Longest Waiting, Queuing On,
Voicemail On, Broadcast, Voicemail Email.
Note:
Group: Apart from Name, Extension and Voicemail Email, the fields use a 1 or 0 value for on or
off.
The format of the system CSV is too complex to be described. It is a full export of all the system's
configuration settings. This file format should only be used for export and import between systems
and not for any offline editing.
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Importing and Exporting Settings
Using the CSV Configurator spreadsheet
You can use the CSV Configurator spreadsheet to create or modify multiple configuration entries.
The CSV Configurator spreadsheet is available in the Manager application folder. By default,
Manager is installed under C:\Program Files or C:\Program Files (x86) on Windows 7.
The remaining path and file name is ...\Avaya\IP Office\Manager\IP Office User CSV
Configurator.xlsm.
Follow the procedure below to export configuration settings. You can then use the exported files
with the CSV Configurator spreadsheet. Follow the instructions in the spreadsheet.
Related links
Editing Configuration Settings on page 177
Exporting Settings
About this task
Procedure
1. Select File | Import/Export... from the menu bar.
2. Select Export.
3. Select the type of file.
The list of exportable record types will change to match the file type.
4. Select the types of items that should be exported.
5. Use the Save In path to select the location for the exported files.
The default location used is a subfolder in the Manager application directory based on
system name of the currently loaded system. For example, ...\Avaya\IP Office
\Manager\System_1.
6. Click OK.
Importing Settings
Importing settings will overwrite any existing records that match a record being imported
Procedure
1. Select File | Import/Export... from the menu bar.
2. Select Import.
3. Select the type of file.
The list of items will change to match the type of file selected and whether a matching file or
files is found in the current file path.
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Editing Configuration Settings
4. Use Look In to adjust the file path.
The default location used is a subfolder in the Manager application directory based on
system name of the currently loaded system. For example, ...\Avaya\IP Office
\Manager\System_1.
5. Select the types of items that should be imported.
6. Click OK.
Copying and Pasting
Manager supports the normal Windows methods of cutting, copying, pasting and deleting records
and settings. These can be accessed through the Edit menu in the menu bar or using the standard
Windows keyboard shortcuts for those actions. They can also be accessed by selecting a record or
text field and then right-clicking.
Copy and paste can be used with the navigation and group panes to create a new record with the
same settings as the original. The copy will be renamed as Copy of ... to avoid conflicting with the
original.
When using copy and paste between individual settings fields, whether on the same record or a
different record, care should be taken to ensure that the fields use the same type of data. Similarly
copying a record in the navigation or group pane and then pasting it into the details pane will prompt
Manager to paste the copied records data into the first field of the current record in the details pane.
As a general rule, cut and paste actions should be used with the same pane and within similar
record types.
For users and user rights, a number of controls have been provided to copy settings between a user
and a user right or vice versa. See User Rights Overview in the Configuration Settings section.
Related links
Editing Configuration Settings on page 177
Saving a Configuration onto PC
The system configuration settings shown within Manager can be saved as a .cfg file on the Manager
PC. These files can be used as backups or sent to other persons to aid problem diagnostics. Note
however that an offline configuration file does not include the Audit Trail records for the system.
Automatically Saving Sent Configurations
By default, Manager creates a file copy of the configuration before it is sent to the system. This copy
is stored in Manager's Working Directory using the system name and .cfg. This behavior is
controlled by the Backup File on Send (File | Preferences | Security) option.
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Sending a Configuration
The number of backups of each systems configuration can be limited to a set number of the most
recent copies.
Saving a Configuration Received from a System
Select File | Save Configuration as from the menu bar.
Saving a Configuration opened on the PC
Click
in the main toolbar or select File | Save Configuration from the menu bar.
Related links
Editing Configuration Settings on page 177
Sending a Configuration
The current configuration settings open within Manager can be sent to the system. The method
depends on whether Manager is being used to edit the configuration of a single system or a network
of systems.
Sending an Individual System Configuration
The first steps of this process depend on whether you are sending a configuration received from the
system or sending one opened offline/created new.
• A Configuration Opened from a System Click
Configuration from the menu bar.
in the main toolbar or select File | Save
• A Configuration Created Offline or Opened from a PC File Select File | Offline | Send
Config from the menu bar.
The Send Configuration menu is displayed.
Configuration Reboot Mode If Manager thinks the changes made to the configuration settings are
mergeable, it will select Merge by default, otherwise it will select Immediate.
• Merge Send the configuration settings without rebooting the system. This mode should only be
used with settings that are mergeable. Refer to Mergeable Settings.
• Immediate Send the configuration and then reboot the system.
• When Free Send the configuration and reboot the system when there are no calls in progress.
This mode can be combined with the Call Barring options.
• Timed The same as When Free but waits for a specific time after which it then wait for there to
be no calls in progress. The time is specified by the Reboot Time. This mode can be combined
with the Call Barring options.
• Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time for
the system reboot. If the time is after midnight, the system's normal daily backup is canceled.
• Call Barring These settings can be used when the reboot mode When Free or Timed is
selected. They bar the sending or receiving of any new calls.
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Editing Configuration Settings
Click OK. A service user name and password may be requested.
• If the service user name or password used do not valid, "Access Denied" is displayed.
• If the service user name used does not have rights to send a configuration or to request a
reboot or merge, "Insufficient service user rights" is displayed.
• If the service user name used does not have operator rights to make the changes that have
been made to the configuration, "Insufficient operator rights. Operator cannot modify xxxx
records" is displayed.
• The warning will appear if the configuration being sent contain any errors indicated by a
icon in the error pane. The configuration can still be sent by selected Yes.
• The message Failed to save the configuration data. (Internal error) may indicate that the
IP500 V2 system has booted using software other than that in its System SD card's primary
folder.
Sending Multiple Configurations
When Manager is running in Server Edition mode or SCN Management mode, it loads multiple
configurations at the same time.
1. Click
in the main toolbar or select File | Save Configuration from the menu bar.
2. The menu displayed only shows details for those systems where the system configuration
has been changed and needs to be sent back to the system.
• Select By default all systems with configuration changes are selected. If you want to exclude a
system from having its configuration updated, either deselect it or cancel the whole process.
• Change Mode If Manager thinks the changes made to the configuration settings are
mergeable, it will select Merge by default, otherwise it will select Immediate.
• Merge Send the configuration settings without rebooting the system. This mode should only be
used with settings that are mergeable. Refer to Mergeable Settings.
• Immediate Send the configuration and then reboot the system.
• When Free Send the configuration and reboot the system when there are no calls in progress.
This mode can be combined with the Incoming Call Barring and Outgoing Call Barring
options.
• Store Offline It is possible to add a reference for a Server Edition Secondary or for a Server
Edition Expansion System to create a configuration file for that system even though it is not
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Erasing the Configuration
physically present. Store Offline saves that configuration on the Server Edition Primary in its file
store. The same file is retrieved from there until such time as the physical server is present at
which time you are prompted whether to use the stored file or the actual servers current
configuration.
• Timed The same as When Free but waits for a specific time after which it then wait for there to
be no calls in progress. The time is specified by the Reboot Time. This mode can be combined
with the Incoming Call Barring and Outgoing Call Barring options.
• Reboot Time This setting is used when the reboot mode Timed is selected. It sets the time for
the system reboot. If the time is after midnight, the system's normal daily backup is canceled.
• Incoming Call Barring This setting can be used when the reboot mode When Free or Timed
is selected. It bars the receiving of any new calls.
• Outgoing Call Barring This setting can be used when the reboot mode When Free or Timed
is selected. It bars the making of any new calls.
Click OK. The progress of the sending of each configuration is displayed.
Related links
Editing Configuration Settings on page 177
Erasing the Configuration
About this task
The system configuration settings can be erased. During this process, the system is rebooted and
starts with a set of default settings. This process does not erase the security settings of the system.
Erasing the Configuration
Procedure
1. Select File | Advanced | Erase Configuration (Default).
2. Enter a valid user name and password.
3. The system will be rebooted.
Related links
Editing Configuration Settings on page 177
Default Settings
The following applies to new systems and those defaulted using the Erase Configuration command.
They also apply to IP500 V2 control units defaulted using the reset button on the rear of the unit
(refer to the Installation manual for details of using the reset button).
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Editing Configuration Settings
Mode
IP500 V2 control units can operate in a number of modes. The initial mode is determined by the type
of System SD card fitted and the level of software.
IP Office A-Law: A system fitted with this type of card will default to A-Law telephony. The system
will default to IP Office Basic Edition - Quick Mode PBX System operation.
IP Office U-Law: A system fitted with this type of card will default to ULAW telephony. The system
will default to IP Office Basic Edition - Quick Mode Key System operation.
IP Office Partner Edition: A system fitted with this type of card will default to A-Law telephony and
IP Office Basic Edition - PARTNER® Mode operation.
IP Office Norstar Edition: A system fitted with this type of card will default to U-Law telephony and
IP Office Basic Edition - Norstar® Mode operation.
Enterprise Branch: Use this option for an SD card intended to be used with an IP Office system
running in Enterprise Branch Mode. There is a separate SD card for Enterprise Branch. The
Enterprise Branch SD card can only be used for Enterprise Branch operation and cannot be used to
change modes to IP Office. You also cannot use or change an IP Office SD card for use with an
Enterprise Branch system.
Warning:
Do not re-purpose a Enerprise Branch card for use with any other IP Office mode. Doing so may
damage the SD card and make it unusable for your Enterprise Branch system.
Default Short Codes
For IP500 V2 control units, A-Law or U-Law operation is determined by the Feature Key dongle
installed in the system. Depending on the variant, a default system will use different sets of default
short codes. See the Default System Short Code List.
Default Data Settings
When a new or defaulted control unit is switched on, it requests IP address information from a
DHCP Server on the network. This operation will occur whether the LAN cable is plugged in or not.
If a DHCP server responds within approximately 10 seconds, the control unit defaults to being a
DHCP client and uses the IP address information supplied by the DHCP server.
If no DHCP Server responds, the control unit still defaults to being the DHCP client but assumes the
following default LAN addresses:
• For its LAN1 it allocates the IP address 192.168.42.1 and IP Mask 255.255.255.0.
• For its LAN2 if supported, it allocates the IP address 192.168.43.1 and IP Mask 255.255.255.0.
Once a control unit has obtained IP address and DHCP mode settings, it will retain those settings
even if rebooted without a configuration file present on the System SD card. To fully remove the
existing IP address and DHCP mode setting, the system must be defaulted using Manager.
Default Security Settings
Security settings are held separately from the configuration settings and so are not defaulted by
actions that default the configuration. To return the security settings to their default values the
separate Erase Security Settings command should be used.
Default Standard Mode Telephony Configuration Settings
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Default Settings
A hunt group Main is created with extension number 200. The first 16 extensions on the systems
are added to the group.
All physical extensions ports are numbered from extension number 201 upwards. A matching user
record for each extension is also created.
A default incoming call route for all voice calls is created, with the default hunt group Main as its
destination.
A default incoming call route for data calls is created with the default RAS record DialIn as its
destination.
All lines are defaulted to Incoming Group ID and Outgoing Group ID of 0.
Default short codes are created based on whether the system's locale is A-Law or U-Law. See
Default Short Codes.
Default Server Edition Telephony Configuration Settings
No users except NoUser.
All extensions are unnumbered.
No default hunt group or incoming calls routes are created.
All auto-create options are off by default.
Related links
Editing Configuration Settings on page 177
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Chapter 11: Configuration Mode Field
Descriptions
This following sections detail the configuration settings for the different record types within the
system. Depending on the type and locale of the system some settings and tabs may be hidden as
they are not applicable. Other settings may be grayed out. This indicates that the setting is either for
information only or that another setting needs to be enabled first.
Related links
Configuration field display on page 209
BOOTP | BOOTP Record on page 212
System on page 214
Line on page 289
Control Unit | Control Unit on page 396
Extension on page 397
User on page 418
Group on page 460
Short Code on page 484
Service on page 485
RAS | RAS on page 498
Incoming Call Route on page 500
WAN Port on page 508
Directory on page 513
Time Profile on page 516
Firewall Profile on page 517
IP Route on page 522
Account Code on page 524
License on page 527
Tunnel on page 531
Auto Attendant on page 536
Authorization Codes on page 541
User Rights on page 542
ARS on page 550
RAS Location Request on page 555
Location on page 556
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Configuration field display
Configuration field display
The way that the system configuration settings are grouped and displayed in the navigation pane
depends on whether Manager is running in its normal Standard mode or in Server Edition mode.
Related links
Configuration Mode Field Descriptions on page 208
Configuration field display in Standard mode on page 209
Configuration field display in Server Edition mode on page 210
Configuration field display in Standard mode
This order of display of different types of configuration record is used for Standard mode systems.
System Overall settings for the data and telephony operation of the system.
Line Settings for trunks and trunk channels within the system.
Control Unit Information summary of the system.
Extension Settings for extension ports.
User Settings for each system user. They may or may not be associated with an extension.
Hunt Group Collections of users to which calls can be directed for answer by any one of those
users.
Short Code These are numbers which when dialed trigger specific features or are translated for
external dialing. Short codes can be set at both the system wide level and locally for a particular
system.
Service Configuration settings such as user names and passwords needed for connections to
data services such as the Internet.
RAS Remote Access Service settings for connecting incoming data calls.
Incoming Call Route Records here are used to match incoming call details on external trunks to
destinations on the system.
WAN Port Configuration settings for the WAN ports provided on some units.
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Configuration Mode Field Descriptions
Directory External names and numbers. Used for matching names to incoming calls and for
dialing from user applications.
Time Profile Used to control when various functions are active.
Firewall Profile Use to control the types of data traffic that can cross into or out of the system.
IP Route These records are used to determine where data traffic on the system should be
routed.
Account Code Used for call logging and to control the dialing of certain numbers.
License License keys are used to enable system features and applications.
Tunnel Used to created IPSec and L2TP data tunnels.
User Rights Provide templates to control the settings applied to associated users.
Auto Attendant Used when an Avaya memory card is installed in the control unit.
ARS Automatic Route Selection is used by to control outgoing external calls.
Authorization Codes Authorization codes are similar to account codes. However, unlike account
codes which are usable by any user, each authorization code is only usable by a specific user or
users associated with a specific set of user rights.
Related links
Configuration field display on page 209
Configuration field display in Server Edition mode
When Manager is being used in Server Edition mode, the navigation pane works in the same way
as normal. However, the different types of configuration records are ordered and grouped differently.
This reflects the fact that some types of record are automatically shared across all systems in the
network.
Solution settings
The first 8 types are records have special behaviors that are different from normal records stored in
the configurations of individual systems in the network.
User These records show settings for system users. Each user may or may not be associated
with an extension. All the users configured on all systems are grouped here to allow easy
configuration access. The individual user records are still stored in the configuration of the particular
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Configuration field display
system on which the user was created and can also be accessed through that system's
configuration settings. New users are created through the User settings of the system that hosts the
user.
Hunt Group These records are groups of users to which calls can be directed for answering by
any one of those users. Hunt group records are stored in the configuration of the Primary Server but
those hunt groups are advertised for use by all systems in the network.
Directory External names and numbers. These records are used to match names to incoming
calls and for making calls by name selection from the directory on phones or in applications. These
directory records are stored in the configuration of the Primary Server. By default all other systems
in the network automatically import a copy of the Primary Server system directory at regular
intervals.
By default, the following types of records are all shared and replicated by each system in the
network and cannot be set at an individual system level. That operation can be changed using the
consolidation settings.
Short Code These are numbers which when dialed trigger specific features or are translated for
external dialing. These short codes are common to all systems in the network.
Incoming Call Route Records set here are used to match incoming call details on external
trunks to destinations. These incoming call routes are shared by all systems in the network.
Time Profile Used to control when various functions are active. The time profiles set here are
shared by all systems in the network.
Account Code Used for call logging and to control the dialing of certain numbers. The account
codes set here are shared by all systems in the network.
User Rights Provide templates to control the settings applied to users associated with a
particular set of user rights. These user rights are shared and replicated on all systems in the
network.
Individual system settings
In addition to the settings above, a range of other types of record can be configured for each
individual system in the network. Visibility and configuration of Short Code, Incoming Call Route,
Time Profile, Account Code and User Rights records is dependent on the consolidation settings
of Manager.
System A system icon is shown for each system in the network. That is, one for the Primary
Server, one for the the Secondary Server if installed and one for each Expansion System (L) and
Expansion System (V2) systems. Each can be expanded to allow configuration of records that are
particular to that system.
•
•
Line Settings for trunks and trunk channels within the system.
Control Unit Information summary of the system.
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Configuration Mode Field Descriptions
•
•
•
•
•
•
•
•
•
•
•
•
Extension Settings for extension ports.
User Settings for each system user. They may or may not be associated with an
extension.
Short Code These are numbers which when dialed trigger specific features or are
translated for external dialing.
Service Configuration settings such as user names and passwords needed for connections
to data services such as the Internet.
RAS Remote Access Service settings for connecting incoming data calls.
WAN Port Configuration settings for the WAN ports provided on some units.
Firewall Profile Use to control the types of data traffic that can cross into or out of the
system.
IP Route These records are used to determine where data traffic on the system should be
routed.
License License keys are used to enable system features and applications.
Tunnel Used to created IPSec and L2TP data tunnels.
ARS Automatic Route Selection is used by to control outgoing external calls.
Authorization Codes Authorization codes are similar to account codes. However, unlike
account codes which are usable by any user, each authorization code is only usable by a
specific user or users associated with a specific set of user rights.
Related links
Configuration field display on page 209
BOOTP | BOOTP Record
The BOOTP settings are used by the Manager application itself. They are not system configuration
settings.
BOOTP is protocol used by devices to request software when restarting. It is used when upgrading
the control unit within a system or when the core software within the control unit has been erased.
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BOOTP | BOOTP Record
When running, Manager can respond to BOOTP requests and, if it finds a matching BOOTP record
for the system, provide the software file indicated by that record.
BOOTP records are not part of a system's configuration settings, they are items saved on the
Manager PC. Normally Manager automatically creates a BOOTP record for each system with which
it has communicated, up to a maximum of 50 records. However BOOTP records can be added and
edited manually when necessary.
Field
Description
File Location
The location from which Manager provides files in response to BOOTP is its binaries
directory. This can be changed using File > Change Working Directory or File >
Preferences > Directories.
This directory is also the directory used by Manager when providing files by TFTP.
Disabling BOOTP
Manager can be disabled from providing BOOTP support for any systems. SelectFile >
Preferences > Preferences > Enable BOOTP and TFTP Server.
Enabled
Default = Enabled
If unchecked, BOOTP support for the matching system from this Manager PC is disabled.
System Name
This field is not changeable. It shows the system name.
MAC Address
The MAC address of the system. The address can be obtained and or verified in a number
of ways:
• When a system's configuration settings are loaded into Manager, it is shown as the
Serial Number on the Unit form. On defaulted systems, it is also used as the system
name.
• If the system is requesting software, the MAC address is shown as part of the request in
the status bar at the base of the Manager screen.
• If the system can be pinged, it may be possible to obtain its MAC address using the
command arp -a <ip address>.
IP Address
The IP address of the system's LAN1.
Filename
The name of the .bin software file used by that type of control unit. To be transferred to the
system, this file must exist in the Manager applications Working Directory.
Time Offset
: Default = 0.
In addition to performing BOOTP support for systems, the Manager application can also
act as a time server (RFC868). This field sets the offset between the time on the PC
running Manager and the time sent to the system in response to its time requests. The
field is not used if a specific Time Server IP Address is set through the System form in the
system's configuration settings.
Manager can be disabled from acting as an Internet Time (RFC868) server. SelectFile >
Preferences > Preferences and uncheck Enable time server.
Related links
Configuration Mode Field Descriptions on page 208
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Configuration Mode Field Descriptions
System
Configuration overview
There is one System record for each system being managed. When using Manager in Server
Edition or Small Community Network Management modes, clicking on the System icon for a
particular system displays a system inventory page for that system.
The following tabs are part of the System configuration.
Tab
Description
System
General settings for the system.
LAN1
Network settings for the RJ45 Ethernet LAN port on the control unit.
LAN2
Network settings for the RJ45 Ethernet WAN port on the control unit.
DNS
Specify the Domain Name Server addresses to use for address resolution.
Voicemail
Details the type and location of the voicemail server.
Telephony
System-wide telephony settings.
Directory Services
Settings to allow the system to import directory numbers from other sources.
System Events
Simple Network Management Protocol (SNMP), email (SMTP), and Syslog settings for
the sending of system events.
SMTP
Settings for SMTP email sending from the system.
Twinning
System wide controls for the use of Mobile Twinning.
SMDR
Settings for the sending of call records to a specified IP address.
CCR
This tab is used for settings specific to the Customer Call Reporter (CCR) application.
Not supported in multi-site networks.
Related links
Configuration Mode Field Descriptions on page 208
System | System on page 215
System | LAN1 on page 220
System | LAN2 on page 232
System | DNS on page 232
System | Voicemail on page 233
System | Telephony on page 241
System | Directory Services on page 266
System | System Events on page 271
System | SMTP on page 278
System | SMDR on page 279
System | Twinning on page 280
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System
System | VCM on page 280
System | CCR on page 283
System | Codecs on page 284
System | VoIP Security on page 285
Dialer on page 286
System | Contact Center on page 289
System | System
These settings are mergeable. However, changes to Locale or Favor RIP Routes over Static
Routes require a reboot.
Field
Description
Name
Default: = System MAC Address.
A name to identify this system. This is typically used to identify the configuration by the
location or customer's company name. Some features such as Gatekeeper require the
system to have a name. This field is case sensitive and within any network of systems
must be unique. Do not use <, >, |, \0, :, *, ?, . or /.
Contact
Information
Default = Blank.
This field is only be edited by service user with administrator rights. If Contact Information
is entered, it will set the system under 'special control'.
If the contact information is set using a standalone version of Manager, warnings that
"This configuration is under special control" are given when the configuration is opened
again. This can be used to warn other users of Manager that the system is being
monitored for some specific reason and provide them with contact details of the person
doing that monitoring. See Loading a Configuration.
Locale
Sets default telephony and language settings based on the selection. It also sets various
external line settings and so must be set correctly to ensure correct operation of the
system. See Supported Country and Locale Settings. For individual users, the system
settings can be overridden through their own locale setting Select User > User > Locale.
Location
Default = None.
Specify a location to associate the system with a physical location. Associating a system
with a location allows emergency services to identify the source of an emergency call.
The drop down list contains all locations that have been defined in the Location page.
Customize Locale Settings
The Customize locale matches the Saudi Arabia locale but with the following additional controls shown below.
For other locales, these are set on the System > Telephony > Tones and Music tab.
Tone Plan
Default = Tone Plan 1
The tone plan control tones and ringing patterns. The options are:
• Tone Plan 1: United States.
• Tone Plan 2: United Kingdom.
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Configuration Mode Field Descriptions
Field
Description
• Tone Plan 3: France.
• Tone Plan 4: Germany.
• Tone Plan 5: Spain.
CLI Type
Used to set the CLI detection used for incoming analogue trunks. The options are:
• DTMF
• FSK V23
• FSK BELL202
Device ID
Server Edition Only. Displays the value set for Device ID on the System | System Events
| Configuration tab. If an SSL VPN is configured, Avaya recommends that the Device ID
match an SSL VPN service Account Name. Each SSL VPN service account name has an
associated SSL VPN tunnel IP address. Having the displayed Device ID match an SSL
VPN service account name helps identify a particular SSL VPN tunnel IP address to use
for remotely managing IP Office.
TFTP Server IP
Address
Default = 0.0.0.0 (Disabled).
While running, Manager can act as a TFTP server and provides files from its configured
binaries directory.
Note:
The following Manager settings are disabled by default:
• Security Settings | Unsecured Interfaces | Applications Controls | TFTP
Directory Read
• File | Preferences | Preferences | Enable BootP and TFTP Servers
To set the address to broadcast, use the value 255.255.255.255. On Server Edition
Systems, the default setting on the Primary Server is 0.0.0.0 (disabled). The default
setting on Secondary and Expansion servers is the address of the Primary Server.
For Avaya IP phones using the system for DHCP, the address set here is used when they
make TFTP requests for software and settings files.
On systems with an Avaya memory card, the LAN1 IP Address can be entered to specify
that memory card as the TFTP file source. The files required for download must be
transferred onto the card using either TFTP from the PC command line (see File Writer
IP Address below) or through Embedded File Management.
1100, 1200, 1600 and 9600 Series IP phones do not support TFTP and require an HTTP
Server IP Address (see below) to be specified.
This address is only used in DHCP responses if the Phone File Server Type is set to
Custom.
HTTP Server IP
Address
February 2016
Default = 0.0.0.0 (Disabled).
For Avaya IP phones using the system for DHCP, the address set here is used when they
make HTTP requests for software and settings files. The files for download should be
placed in the HTTP server's root directory.
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System
Field
Description
Using the system's own memory card is supported as the source for HTTP files. This is
supported for up to 50 IP phones total. This is done by setting the TFTP Server IP
Address and HTTP Server IP Address to match the system IP address.
HTTP-TFTP Relay is support using Manager as the TFTP server (not supported by Linux
based systems). This is done by setting the TFTP Server IP Address to the address of
the Manager PC and the HTTP Server IP Address to the control unit IP address. This
method is supported for up to 5 IP phones total.
This address is only used in DHCP responses if the Phone File Server Type (see below)
is set to Custom.
Phone File Server
Type
Default = Memory Card (Disk on a Linux system).
For IP (H.323 and SIP) phones using the system as their DHCP server, the DHCP
response can include the address of a file server from which the phone should request
files. The setting of this field controls which address is used in the DHCP response. The
options are:
• Custom: The DHCP response provided to phones it is supporting contains the TFTP
Address from the TFTP Server IP Address above and the HTTP Address from the
HTTP Server IP Address above.
• Memory Card (Disk on a Linux system): The system will respond to file requests from
phones using files on its own memory card. The DHCP response provided to phones it
is supporting contains the LAN address of the system for both TFTP and HTTP
address.
• Manager: The system will forward any H.323 phone file request to the configured
Manager PC IP Address set below. HTTP-TFTP relay is used for HTTP requests. The
DHCP response provided to phones the system is supporting contains the LAN address
of the system for the HTTP Address. This option is not supported for Linux based
systems.
HTTP Redirection
Default = Off.
Allows for the use of an alternate HTTP file server for the download of large binary files.
This field is available when the Phone File Server Type is set to Memory Card (or Disk
on a Linux system). When this field is set to Phone Binaries, 96x1 H.323 phones
requesting their binary files are redirected to the HTTP server defined in the HTTP Server
IP Address field.
Manager PC IP
Address
Default = 0.0.0.0 (Broadcast).
Avaya HTTP
Clients Only
Default = Off.
This address is used when the Phone File Server Type is set to Manager.
When selected, the system only responds to HTTP requests from sources it identifies as
another system, an Avaya phone or application.
Enable SoftPhone Default = Off.
HTTP Provisioning
This option must be enabled if the IP Office Video Softphone is being supported.
Favour RIP Routes Default = Off
over Static Routes
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Configuration Mode Field Descriptions
Field
Description
RIP can be enabled on the system LAN1 and LAN2 interfaces, and on specific Services.
When this setting is on, the RIP route to a destination overrides any static route to the
same destination in the system's IP Routes, regardless of the RIP route's metric. The only
exception is RIP routes with a metric of 16 which are always ignored.
Note:
If a previously learnt RIP route fails, the system applies a metric of 16 five minutes
after the failure. When off, any RIP route to a destination for which a static route has
been configured is ignored. This option is not supported on Linux based systems.
Automatic Backup
Default = On.
This command is available with IP500 V2 systems. When selected, as part of its daily
backup process, the system automatically copies the folders and files from the System
SD card's /primary folder to its /backup folder. Any matching files and folders already
present in the /backup folder are overwritten.
Provider
Default = Not visible. This field is visible only if the system has been branded by addition
of a special license for a specific equipment provider. The branding is fixed, that is it
remains even if the license is subsequently removed. The number shown is a unique
reference to the particular equipment provider for whom the system has been branded.
When branded, the equipment provider's name is displayed on idle phone displays and
other provider related features are enabled.
Time Setting
Config Source
The time is either set manually (see Date and Time), obtained using Time protocol
(RFC868) requests or obtained using Network Time Protocol (RFC958) request. This field
is used to select which method is used and to apply ancillary settings based on the
selected method.
For Server Edition networks, the Primary Server is defaulted to use SNTP to
0.pool.ntp.org to obtain its time and date. The Secondary Server and expansion servers
are defaulted to use SNTP to obtain their time from the Primary Server.
For other IP Office systems the default is Voicemail Pro/Manager. This option should not
be used with Server Edition systems and systems with a Unified Communication Module
as in those scenarios, the voicemail server is being hosted by and getting its time from
the same server as the IP Office.
• None, the system to not make any time requests. The system time and date needs to
be set using a phone with System Phone Rights (User | User). The system can then
automatically apply daylight saving settings to the manually set time.
• Voicemail Pro/Manager Both the Voicemail Pro service and the Manager program can
act as RFC868 Time servers for the system. Use of other RFC868 server sources is not
supported. They provide both the UTC time value and the local time as set on the PC.
The system makes a request to the specified address following a reboot and every 8
hours afterwards. This option should not be used with Server Edition systems and
systems with a Unified Communication Module as in those scenarios the voicemail
server is being hosted by and getting its time from the same server as the IP Office.
- IP Address: Default = 0.0.0.0 (Broadcast) The address to which the RFC868 request
is sent. 0.0.0.0 means default operation. In this mode, following a reboot the control
unit will send out a time request on its LAN interfaces. It first makes the request to the
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System
Field
Description
Voicemail Server IP address in its configuration if set and, if it receives no reply, it
then makes a broadcast request.
- Time Offset: Default = 00:00 This value is not normally set as any time changes,
including daylight saving changes, that occur on the PC will be matched by the
system.
If you are running Manager when the voicemail server starts, voicemail does not start
as a time server. It is therefore recommended that you have no copy of Manager
running when you start or restart the voicemail server. Manager can be disabled from
acting as a RFC868 time server by deselecting the Enable Time Server option (File |
Preferences | Edit | Preferences).
• SNTP: Use a list of SNTP servers to obtain the UTC time. The records in the list are
used one at a time in order until there is a response. The system makes a request to
the specified addresses following a reboot and every hour afterwards.
Time Settings
Time Server
Address
Default = Blank
Displayed when the Time Setting Config Source is set to SNTP.
Enter a list of IP addresses, host names, or fully qualified domain names (FQDN) for the
SNTP servers. Separate each record with a space. The use of broadcast addresses is
not supported. The list is used in order of the records until a response is received.
Time Zone
Select a time zone from the list.
Local Time Offset
from UTC
Default is based on the currently selected time zone.
Automatic DST
Default = On.
This setting is used to set the local time difference from the UTC time value provided by
an SNTP server. For example, if the system is 5 hours behind UTC, this field should be
configured with -05:00 to make the adjustment. The time offset can be adjusted in 15
minute increments. If also using the daylight time saving settings below, use this offset to
set the non-DST local time.
When set to On, the system automatically corrects for daylight saving time (DST)
changes as configured in the Clock Forward/Back Settings below.
Clock Forward/
Back Settings
(Start Date — End
Date (DST Offset))
Default is based on the currently selected time zone.
Click Edit to configure the time and date for DST clock corrections. In the Daylight Time
Settings window, you can configure the following information:
• DST Offset: the number of hours to shift for DST.
• Clock Forward/Back: Select Go Forward to set the date when the clock will move
forward. Select Go Backwards to set the date when the clock will move backward.
• Local Time To Go Forward: The time of day to move the clock forward or backward.
• Date for Clock Forward/Back: Set the year, month and day for moving the clock
forwards and backwards.
Once you click OK, the forward and back dates, plus the DST offset, are displayed using
the format (Start Date — End Date (DST Offset)).
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Configuration Mode Field Descriptions
Field
Description
File Writer IP
Address
Default = 0.0.0.0 (Disabled)
Dongle Serial
Number
Displayed for IP 500 V2 systems. This field is for information only.
System
Identification
Displayed for Linux based systems. This field is for information only.
AVPP IP Address
Default = 0.0.0.0 (Disabled)
This field set the address of the PC allowed to send files to the System SD card installed
in the system using HTTP or TFTP methods other than embedded file management.On
systems with an Avaya memory card, this field sets the address of the PC allowed to
send files to the memory card using HTTP or TFTP methods other than embedded file
management. For Linux based systems it is applied to non-embedded file management
access to the /opt/ipoffice folder on the server. An address of 255.255.255.255 allows
access from any address. If embedded file management is used, this address is
overwritten by the address of the PC using embedded file management (unless set to
255.255.255.255).
Shows the serial number of the feature key dongle against which the system last
validated its licenses. Local is shown for a serial port, Smart Card or System SD feature
key plugged directly into the control unit. Remote is shown for a parallel or USB feature
key connected to a feature Key Server PC. The serial number is printed on the System
SD card and prefixed with FK.
This is the unique system reference that is used to validate licenses issued for this
particular system. For a physical server this is a unique value based on the server
hardware. For a virtual server this value is based on several factors including the LAN1
and LAN2 IP addresses, the host name and the time zone. If any of those are changed,
the System ID changes and any existing licenses become invalid.
Where Avaya 3600 Series SpectraLink wireless handsets are being used with the
system, this field is used to specify the IP address of the Avaya Voice Priority Processor
(AVPP)
Related links
System on page 214
System | LAN1
This tab is used to configure the behavior of the services provided by the system's first LAN
interface.
Up to 2 LAN's (LAN1 and LAN2) can be configured. The control unit has 2 RJ45 Ethernet ports,
marked as LAN and WAN. These form a full-duplex managed layer-3 switch. Within the system
configuration, the physical LAN port is LAN1, the physical WAN port is LAN2.
Configuring both interfaces with the same IP address on the same subnet is not supported.
However, no warning is issued when this configuration is implemented.
Related links
System on page 214
LAN Settings on page 221
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System
VoIP on page 222
Network Topology on page 228
DHCP Pools on page 231
LAN Settings
Navigation: System | LAN | LAN Settings
Used to set the general LAN settings for the LAN interface such as the IP address mode.
Configuration Settings
These settings are not mergeable. Changes to these settings require a reboot of the system.
Field
Description
IP Address
Default = 192.168.42.1 or DHCP client.
This is the IP address of the Control Unit on LAN1. If the control unit is also acting as
a DHCP server on the LAN, this address is the starting address for the DHCP
address range.
IP Mask
Default = 255.255.255.0 or DHCP client.
This is the IP subnet mask used with the IP address.
Primary Trans. IP
Address
Default = 0.0.0.0 (Disabled)
RIP Mode
Default = None.
This setting is only available on control units that support a LAN2. Any incoming IP
packets without a service or session are translated to this address if set.
Routing Information Protocol (RIP) is a method by which network routers can
exchange information about device locations and routes. Routes learnt using RIP are
known as 'dynamic routes'. The system also supports 'static routes' though its IP
Route records. For Server Edition systems this setting is only available on Expansion
System (V2) systems. The options are:
• None: The LAN does not listen to or send RIP messages
• Listen Only (Passive): Listen to RIP-1 and RIP-2 messages in order to learn RIP
routes on the network.
• RIP1: Listen to RIP-1 and RIP-2 messages and send RIP-1 responses as a subnetwork broadcast.
• RIP2 Broadcast (RIP1 Compatibility): Listen to RIP-1 and RIP-2 messages and
send RIP-2 responses as a sub-network broadcast.
• RIP2 Multicast: Listen to RIP-1 and RIP-2 messages and send RIP-2 responses to
the RIP-2 multicast address.
Enable NAT
Default = Off
This setting controls whether NAT should be used for IP traffic from LAN1 to LAN2.
This setting should not be used on the same LAN interface as a connected WAN3
expansion module.
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Configuration Mode Field Descriptions
Field
Description
Number of DHCP IP Default = 200 or DHCP client. Range = 1 to 999.
Addresses
This defines the number of sequential IP addresses available for DHCP clients.
DHCP Mode
Default = DHCP Client.
This controls the control unit's DHCP mode for the LAN. When doing DHCP:
• LAN devices are allocated addresses from the bottom of the available address
range upwards.
• Dial In users are allocated addresses from the top of the available range
downwards.
• If the control unit is acting as a DHCP server on LAN1 and LAN2, Dial in users are
allocated their address from the LAN1 pool of addresses first.
The options are:
• Server: When this option is selected, the system will act as a DHCP Server on this
LAN, allocating address to other devices on the network and to PPP Dial in users.
• Disabled When this option is selected, the system will not use DHCP. It will not act
as a DHCP server and it will not request an IP address from a DHCP server on this
LAN.
• Dial In When this option is selected, the system will allocate DHCP addresses to
PPP Dial In users only. On systems using DHCP pools, only addresses from a pool
on the same subnet as the system's own LAN address will be used.
• Client When this option is selected, the system will request its IP Address and IP
Mask from a DHCP server on the LAN.
Note:
Do not use this option with a limited time lease line.
• Advanced: (IP500 V2 only). The system can be configured with a number of
DHCP Pools from which it can issue IP addresses.
Related links
System | LAN1 on page 220
VoIP
This form is used to set the system defaults for VoIP operation on the LAN interface.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Note:
Under SIP Registrar Enable, the Auto-create Extn/User setting is mergeable. Changing this
setting does not require a reboot.
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System
Field
Description
H.323 Gatekeeper
Enable
Default = On
Auto-create Extn
Default = Off
This settings enables gatekeeper operation.
When this option is on, an extension record is automatically created for H.323 phones
registering themselves with the system as their gatekeeper. SIP Extensions use a
separate setting.
If using resilience backup to support Avaya IP phones, Auto-create Extn and Autocreate User should not be left enabled after initial configuration or any subsequent
addition of new extensions and users. Leaving auto-create options enabled on a
system that is a failover target may cause duplicate extension/user records on the
multi-site network under multiple failure scenarios.
For security, any auto-create settings set to On are automatically set to Off after 24
hours.
Auto-create User
Default = Off
When this option is on and H.323 Auto-create Extn is also on, when a new H.323
extension is created a matching user record is also created.
For security, any auto-create settings set to On are automatically set to Off after 24
hours.
H.323 Remote Extn
Enable
Default = Off.
The system can be configured to support remote H.323 extensions in the case where
NAT is used in the connection path. This could be the case where the IP Office is
located behind a corporate NAT/Firewall router and/or the H.323 phone is located
behind residential NAT enable router.
The use of this option and the interaction and configuration of external third party
elements is beyond the scope this help file.
In the case where the public IP address of the corporate router is unknown, the LAN's
Network Topology settings should be used to configure a STUN Server. Enabling H.
323 Remote Extn Enable allows configuration of the RTP Port number Range
(NAT) settings.
Currently, only 9600 Series phones are supported as H.323 remote extensions.
Remote Call
Signalling Port
Default = 1720.
SIP Trunks Enable
Default = On
This field is available when H.323 Remote Extn Enable is set to On. It specifies the
corporate router's signalling port for remote H.323 phones.
This settings enables support of SIP trunks. It also requires entry of SIP Trunk
Channels licenses.
Enabling SIP Trunks Enable allows configuration of the RTP Port number Range
(NAT) settings.
SIP Registrar Enable
Default = Off.
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Configuration Mode Field Descriptions
Field
Description
Used to set the system parameters for the system acting as a SIP Registrar to which
SIP endpoint devices can register. Separate SIP registrars can be configured on
LAN1 and LAN2. Registration of a SIP endpoint requires an available IP Endpoints
license. SIP endpoints are also still subject to the extension capacity limits of the
system.
Auto-create Extn/User
Default = Off.
If on, a new extension and user records are created by the registration of a new SIP
endpoint. If off, SIP endpoint can only register against existing configuration records.
If a password is requested by a SIP device during registration to create a new user
and extension records in the system configuration, 0000 should be used. If
registering against an existing user record, the user Login Code should be used if
set or 0000 if the user does not have a Login Code set.
For security, any auto-create settings set to On are automatically set to Off after 24
hours.
Note:
This setting is not applicable to the Avaya A175 Desktop Video Device with the
Avaya Communicator.
SIP Remote Extn
Enable
Default = Off.
The system can be configured to support remote SIP extensions in the case where
NAT is used in the connection path. This could be the case where the IP Office is
located behind a corporate NAT/Firewall router and/or the SIP phone is located
behind residential NAT enable router.
This option cannot be enabled on both LAN1 and LAN2.
The use of this option and the interaction and configuration of external third party
elements is beyond the scope this help file.
In the case where the public IP address of the corporate router is unknown, the LAN's
Network Topology settings should be used to configure a STUN Server. Enabling SIP
Remote Extn Enable allows configuration of:
• the Remote UDP Port, Remote TCP Port, Remote TLS Port settings
• the Port Number Range (NAT) settings
Currently, only Avaya Communicator for Windows, Avaya Communicator for iPad,
one-X Mobile iOS and one-X Mobile Android SIP clients are supported as SIP remote
extensions.
Domain Name
Default = Blank
This is the local SIP registrar domain name that will be needed by SIP endpoints in
order to register with the system. If this field is left blank, registration is against the
LAN IP address.
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System
Field
Description
Layer 4 Protocol
Default = TCP and UDP. This field is used to select which protocosl are supported for
SIP connections: TCP, UDP, or TLS.
• UDP Port: Default = 5060. The port to use for SIP UDP support if UDP is selected
as the Layer 4 Protocol above.
• TCP Port: Default = 5060. The port to use for SIP TCP support if TCP is selected
as the Layer 4 Protocol above.
• TLS Port: Default = 5061. The port to use for SIP TLS support.
• Remote UDP Port: Default = 5060. The port to use for SIP UDP support if UDP is
selected as the Layer 4 Protocol for remote SIP extension.
• Remote TCP Port: Default = 5060. The port to use for SIP TCP support if TCP is
selected as the Layer 4 Protocol for remote SIP extension.
• Remote TLS Port: Default = 5061. The port to use for SIP TLS support if TLS is
selected as the Layer 4 Protocol for remote SIP extension.
Challenge Expiry Time Default = 10.
(secs)
The challenge expiry time is used during SIP extension registration. When a device
registers, the system SIP Registrar will send a challenge back to the device and waits
for an appropriate response. If the response is not received within this timeout the
registration is failed.
RTP
Port Number Range
For each VoIP call, a receive port for incoming Real Time Protocol (RTP) traffic is
selected from a defined range of possible ports, using the even numbers in that
range. The Real Time Control Protocol (RTCP) traffic for the same call uses the RTP
port number plus 1, that is the odd numbers. For control units and Avaya H.323 IP
phones, the default port range used is 49152 to 53246. On some installations, it may
be a requirement to change or restrict the port range used. It is recommended that
only port numbers between 49152 and 65535 are used, that being the range defined
by the Internet Assigned Numbers Authority (IANA) for dynamic usage.
Important:
The minimum and maximum settings of the port range should only be adjusted
after careful consideration of the customer network configuration and existing
port usage. For pre-Release 8.1 systems, the gap between the minimum and
maximum port values must be at least 1024. For Release 8.1 and higher, the
gap between the minimum and maximum port values must be at least 254.
Port Range (minimum) IP 500 v2 default = 46750. Range = 46750 to 50750.
Linux default = 40750. Range = 40750 to 50750
This sets the lower limit for the RTP port numbers used by the system.
Port Range
(maximum)
IP 500 v2 default = 50750. Range = 46750 to 50750.
Linux default = 50750. Range = 40750 to 50750
This sets the upper limit for the RTP port numbers used by the system.
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Configuration Mode Field Descriptions
Field
Description
Port Number Range (NAT)
These settings are available when either H.323 Remote Extn Enable, SIP Trunks Enable, or SIP Remote
Extn Enable is set to On.
This option is not supported if the Firewall/NAT Type (System | LAN1 | Network Topology) is set to
Symmetric Firewall or Open Internet.
Port Range (minimum) IP 500 v2 default = 46750. Range = 46750 to 50750.
Linux default = 40750. Range = 40750 to 50750
This sets the lower limit for the RTP port numbers used by the system.
Port Range
(maximum)
IP 500 v2 default = 50750. Range = 46750 to 50750.
Linux default = 50750. Range = 40750 to 50750
This sets the upper limit for the RTP port numbers used by the system.
Enable RTCP Monitor
On Port 5005
Default = On.
For 1600, 4600, 5600 and 9600 Series H.323 phones, the system can collect VoIP
QoS (Quality of Service) data from the phones. For other phones, including non-IP
phones, it can collect QoS data for calls if they use a VCM channel. The QoS data
collected by the system is displayed by the System Status Application.
This setting is mergeable. However it only affects H.323 phones when the register
with the system. therefore any change to this setting requires H.323 phones that have
already been registered to be rebooted. Avaya H.323 phones can be remotely
rebooted using the System Status Application.
The QoS data collected includes: RTP IP Address, Codec, Connection Type, Round
Trip Delay, Receive Jitter, Receive Packet Loss.
This setting is not the same as the RTCPMON option within Avaya H.323 phone
settings. The system does not support the RTCPMON option.
RTCP Collector IP
address for phones
Default = Blank.
This setting defines an alternate destination for the RTCP Monitor data described in
the Enable RTCP Monitor On Port 5005 field above. This enables you to send the
data collected to a third party QoS monitoring application.
The Enable RTCP Monitor On Port 5005 must be turned Off to enable this field.
Changes to this setting requires a reboot of the phones.
Keepalives
These settings can be used with SIP trunks associated with the LAN through their Use Network Topology Info
setting (Line | Transport). For some scenarios, with frequent call forwarding on the same SIP trunk, speech path
may be lost during the connection. The use of periodic keepalive packets may prevent the issue.
Scope
Default = Disabled
Select whether the sending of keepalive packets should be disabled or sent for RTP
or for both RTP and RTCP.
Periodic timeout
Default = 0 (Off). Range = 0 to 180 seconds.
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System
Field
Description
Sets how long the system will wait before sending a keepalive if no other packets of
the select SCOPE are seen.
Initial keepalives
Default = Disabled.
If enabled, keepalives can also been sent during the initial connection setup.
DiffServ Settings
When transporting voice over low speed links it is possible for normal data packets (1500 byte packets) to
prevent or delay voice packets (typically 67 or 31 bytes) from getting across the link. This can cause
unacceptable speech quality. Therefore it is important that all traffic routers and switches in a network to have
some form of Quality of Service mechanism (QoS). QoS routers are essential to ensure low speech latency and
to maintain sufficient audible quality.
The system applies the DiffServ settings to outgoing traffic on any SIP lines which have their User Network
Topology Info setting (Line | Transport) set to match the LAN interface.
The system supports the DiffServ (RFC2474) QoS mechanism. This uses a Type of Service (ToS) field in the IP
packet header.
The hex and decimal entry fields for the following values are linked, the hex value being equal to the decimal
multiplied by 4.
DSCP (Hex)
Default = B8 (Hex)/46 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
The DiffServ Code Point (DSCP) setting applied to VoIP calls. By default, the same
setting is used for audio and video. If desired, you can configure separate values for
audio and video. For correct operation, especially over WAN links, the same value
should be set at both ends.
Video DSCP (Hex)
Default = B8 (Hex)/46 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
The DiffServ Code Point (DSCP) setting applied to video VoIP calls. For correct
operation, especially over WAN links, the same value should be set at both ends.
DSCP Mask (Hex)
Default = FC (Hex)/63 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
Allows a mask to be applied to packets for the DSCP value.
SIG DSCP (Hex)
Default = 88 (Hex)/34 (decimal). Range = 00 to FF (Hex)/0 to 63 (decimal)
This setting is used to prioritize VoIP call signaling.
DHCP Settings
Primary Site Specific
Option Number
(4600/5600)
Default = 176. Range = 128 to 254.
Secondary Site
Specific Option
Number (1600/9600)
Default = 242. Range = 128 to 254.
VLAN
Default = Not present. This option is applied to H.323 phones using the system for
DHCP support. If set to Disabled, the L2Q value indicated to phones in the DHCP
A site specific option number (SSON) is used as part of DHCP to request additional
information. 176 is the default SSON used by 4600 Series and 5600 Series IP
phones.
Similar to the primary SSON. 242 is the default SSON used by 1600 and 9600 Series
IP phones requesting installation settings via DHCP.
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Configuration Mode Field Descriptions
Field
Description
response is 2 (disabled). If set to Not Present, no L2Q value is included in the DHCP
response.
1100 Voice VLAN Site
Specific Option
Number (SSON)
Default = 232.
1100 Voice VLAN IDs
Default = Blank.
This is the SSON used for responses to 1100/1200 Series phones using the system
for DHCP.
For 1100/1200 phone being supported by DHCP, this field sets the VLAN ID that
should be provided if necessary. Multiple IDs (up to 10) can be added, each
separated by a + sign.
Related links
System | LAN1 on page 220
Network Topology
STUN (Simple Traversal of UDP through NAT) is a mechanism used with overcome the effect of
NAT firewalls. The network address translation (NAT) action performed by this type of firewall can
have negative effects on VoIP calls.
Test packets are sent by the system to the address of the external STUN server, those packets
crossing the firewall in the process. The STUN server replies and includes copies of the packets it
received in the reply. By comparing the packet sent and received, it is possible for the system to
determine the type of NAT firewall and to modify future packets to overcome the effects of the
firewall.
These settings are used for SIP trunk connections from the LAN. For further details of system SIP
operation refer to the SIP Line section. The use of STUN is unnecessary if the SIP ITSP uses a
Session Border Controller (SBC). Use of SIP requires entry of SIP Trunk Channels licenses.
The network topology settings are also used for H.323 remote extensions supported on the LAN.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
The following fields can be completed either manually or the system can attempt to automatically
discover the appropriate values. To complete the fields automatically, only the STUN Server IP
Address is required. STUN operation is then tested by clicking Run STUN. If successful the
remaining fields are filled with the results.
Field
Description
STUN Server IP
Address
Default = Blank
STUN Port
Default = 3478.
Enter the IP address or fully qualified domain name (FQDN) of the SIP ITSP's STUN
server. The system will send basic SIP messages to this destination and from data
inserted into the replies can try to determine the type NAT changes being applied by
any firewall between it and the ITSP.
Defines the port to which STUN requests are sent if STUN is used.
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System
Field
Description
Firewall/NAT Type
Default = Unknown
The settings here reflect different types of network firewalls. The options are:
• Blocking Firewall
• Symmetric Firewall: SIP packets are unchanged but ports need to be opened
and kept open with keep-alives. If this type of NAT is detected or manually
selected, a warning ‘Communication is not possible unless the STUN server is
supported on same IP address as the ITSP’ will be displayed as part of the
manager validation.
• Open Internet: No action required. If this mode is selected, settings obtained by
STUN lookups are ignored. The IP address used is that of the system LAN
interface.
• Symmetric NAT: A symmetric NAT is one where all requests from the same
internal IP address and port, to a specific destination IP address and port, are
mapped to the same external IP address and port. If the same host sends a
packet with the same source address and port, but to a different destination, a
different mapping is used. Furthermore, only the external host that receives a
packet can send a UDP packet back to the internal host. SIP Packets need to be
mapped but STUN will not provide the correct information unless the IP address
on the STUN server is the same as the ITSP Host. If this type of NAT/Firewall is
detected or manually selected, a warning ‘Communication is not possible unless
the STUN server is supported on same IP address as the ITSP’ will be displayed
as part of the manager validation.
• Full Cone NAT: A full cone NAT is one where all requests from the same internal
IP address and port are mapped to the same external IP address and port.
Furthermore, any external host can send a packet to the internal host, by sending
a packet to the mapped external address. SIP packets need to be mapped to NAT
address and Port; any Host in the internet can call in on the open port, that is the
local info in the SDP will apply to multiple ITSP Hosts. No warning will be
displayed for this type of NAT because the system has sufficient information to
make the connection).
• Restricted Cone NAT: A restricted cone NAT is one where all requests from the
same internal IP address and port are mapped to the same external IP address
and port. Unlike a full cone NAT, an external host (with IP address X) can send a
packet to the internal host only if the internal host had previously sent a packet to
IP address X. SIP packets needs to be mapped. Responses from hosts are
restricted to those that a packet has been sent to. So if multiple ITSP hosts are to
be supported, a keep alive will need to be sent to each host. If this type of NAT/
Firewall is detected or manually selected, no warning will be displayed for this type
of NAT.
• Port Restricted Cone NAT: A port restricted cone NAT is like a restricted cone
NAT, but the restriction includes port numbers. Specifically, an external host can
send a packet, with source IP address X and source port P, to the internal host
only if the internal host had previously sent a packet to IP address X and port P.
SIP packets needs to be mapped. Keep-alives must be sent to all ports that will be
the source of a packet for each ITSP host IP address. If this type of NAT/Firewall
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Configuration Mode Field Descriptions
Field
Description
is detected or manually selected, no warning will be displayed for this type of NAT.
However, some Port Restricted NAT's have been found to be more symmetric in
behavior, creating a separate binding for each opened Port, if this is the case the
manager will display a warning ‘Communication is not possible unless the STUN
server is supported on same IP address as the ITSP’ as part of the manager
validation.
• Static Port Block: Use the RTP Port Number Range specified on the VoIP tab
without STUN translation. Those ports must be fixed as open on any NAT firewall
involved
• One-To-One NAT: This setting supports IP Office cloud deployments where the
Primary server is behind a NAT that performs IP address translation but not port
mappings. All required ports must be open on the NAT.
When set to One-To-One NAT, the following configuration settings are applied
and cannot be edited.
- The LAN | Network Topology | Public Port values are set to 0.
- LAN | VoIP | SIP Registrar Enable remote protocol port values are set to equal
their corresponding local protocol port values.
- The LAN | VoIP | RTP | Port Number Range (NAT) Minimum and Maximum
values are set to equal the corresponding Port Number Range values.
• Unknown
Binding Refresh Time
(seconds)
Default = 0 (Never). Range = 0 to 3600 seconds.
Having established which TCP/UDP port number to use, through either automatic or
manual configuration, the system can send recurring ‘SIP OPTIONS requests’ to the
remote proxy terminating the trunk. Those requests will keep the port open through
the firewall. Requests are sent every x seconds as configured by this field.
Note:
If a binding refresh time has not been set you may experience problems
receiving inbound SIP calls as they are unable to get through the Firewall. In
these circumstances make sure that this value has been configured.
Public IP Address
Default = 0.0.0.0 This value is either entered manually or discovered by the Run
STUN process. If no address is set, the system LAN1 address is used.
Public Port
Default = 0
The public port value for UDP, TCP, and TLS. For each protocol, this value is either
entered manually or discovered by the Run STUN process.
Run STUN
This button tests STUN operation between the system LAN and the STUN Server IP
Address set above. If successful the results are used to automatically fill the
remaining fields with appropriate values discovered by the system. Before using Run
STUN the SIP trunk must be configured.
When this option is used, a
information icon is shown against the fields to
indicate that the values were automatically discovered rather than manually entered.
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System
Field
Description
Run STUN on startup
Default = Off
This option is used in conjunction with values automatically discovered using Run
STUN. When selected, the system will rerun STUN discovery whenever the system
is rebooted or connection failure to the SIP server occurs.
Related links
System | LAN1 on page 220
DHCP Pools
DHCP pools allows for the configuration of of IP address pools for allocation by the system when
acting as a DHCP server. On an IP500 V2 system, you can configure up to 8 pools. On Server
Edition Linux systems, you can configure up to 64 pools.
By default the DHCP settings (IP Address, IP Mask and Number of DHCP IP Addresses) set on the
LAN Settings tab are reflected by the first pool here. For support of PPP Dial In address requests, at
least one of the pools must be on the same subnet as the system's LAN. Only addresses from a
pool on the same subnet as the system's own LAN address will be used for PPP Dial In.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Apply to Avaya IP
Phones Only
Default = Off.
If this option is selected, the DHCP addresses are only used for requests from Avaya
IP phones. Other devices connected to the system LAN will have to use static
addresses or obtain their address from another DHCP server.
In addition to the above control, Avaya IP phones will only complete DHCP against a
DHCP server configured to supports a Site Specific Option Number (SSON) that
matches that set on the phone. The SSON numbers supported by the system DHCP
are set on the VoIP sub-tab.
DHCP Pool
Up to 8 pools can be added. The first pool matches the IP Address, IP Mask and
Number of DHCP IP Addresses on the LAN Settings sub-tab. When adding or editing
pools, Manager will attempt to warn about overlaps and conflicts between pools. The
options are:
• Start Address Sets the first address in the pool.
• Subnet Mask: Default = 255.255.255.0 Sets the subnet mask for addresses issued
from the pool.
• Default Router: Default = 0.0.0.0 For pools issuing IP addresses on the same subnet
as the system LAN's, 0.0.0.0 instructs the system to determined the actual default
router address to issue by matching the IP address/subnet mask being issued in the
IP Routing table. This matches the default behaviour used by systems without
multiple pools. For pools issuing addresses not on the same subnet as the system
LAN's, the default router should be set to the correct value for devices on that subnet.
• Pool Size: Default = 0 Set the number of DHCP client addresses available in the
pool.
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Configuration Mode Field Descriptions
Related links
System | LAN1 on page 220
System | LAN2
This set of tabs is used to configure the system's second LAN interface.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
The fields available for LAN2 are the same as for LAN1 except for the following additional field:
Field
Description
Firewall
Default = <None> (No firewall)
Allows the selection of a system firewall to be applied to traffic routed from LAN2 to
LAN1.
Related links
System on page 214
System | DNS
DNS is a mechanism through which the URL's requested by users, such as www.avaya.com, are
resolved into IP addresses. These requests are sent to a Domain Name Server (DNS) server, which
converts the URL to an IP address. Typically the internet service provider (ISP) will specify the
address of the DNS server their customers should use.
WINS (Windows Internet Name Service) is a similar mechanism used within a Windows network to
convert PC and server names to IP addresses via a WINS server.
If the system is acting as a DHCP server, in addition to providing clients with their own IP address
settings it can also provide them with their DNS and WINS settings if requested by the client.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
DNS Service IP Address
Default = 0.0.0.0 (Do not provide DNS/Use DNS forwarding)
This is the IP address of a DNS Server. Your Internet service provider or network
administrator provides this information. If this field is left blank, the system uses
its own address as the DNS server for DHCP client and forward DNS requests to
the service provider when Request DNS is selected in the service being used
(Service | IP).
Backup DNS Server IP
Address
Default = 0.0.0.0 (No backup)
DNS Domain
Default = Blank (No domain)
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System
Field
Description
This is the domain name for your IP address. Your Internet service provider or
network administrator provides this. Typically this field is left blank.
WINS Server IP Address Default = 0.0.0.0 (Do not provide WINS) This is the IP address of your local
WINS server. This is only used by Windows PCs, and normally points to an NT
server nominated by your network administrator as your WINS server. Setting a
value will result in also sending a mode of "hybrid". For Server Edition this field is
only available on Expansion System (V2) servers.
Backup WINS Server IP
Address
Default = 0.0.0.0 (No backup)
WINS Scope
Default = Blank (no scope) This is provided by your network administrator or left
blank. For Server Edition this field is only available on Expansion System (V2)
servers.
Related links
System on page 214
System | Voicemail
The following settings are used to set the system's voicemail server type and location. Fields are
enabled or grayed out as appropriate to the selected voicemail type. Refer to the appropriate
voicemail installation manual for full details.
Changes to Voicemail Type and the DTMF Breakout settings require a reboot.
Field
Description
Voicemail Type
Defaults: Non-Server Edition = Embedded Voicemail, Primary Server = Voicemail Pro,
Server Edition Secondary Server with independent voicemail or for Outbound Contact
Express = Voicemail Pro, Server Edition Others: Centralized Voicemail.
Sets the type of voicemail system being used. The options are:
• None: No voicemail operation.
• Analogue Trunk MWI (Release 9.0.3+): Select this option to support receiving a
message waiting indicator (MWI) signal from analog trunks terminating on the
ATM4U-V2 card. MWI is a telephone feature that turns on a visual indicator on a
telephone when there are recorded messages.
• Avaya Aura Messaging: Select this option if you want to configure the system to
use Avaya Aura Messaging as the central voicemail system. If you choose this
option, you are still able to use Embedded Voicemail or Voicemail Pro at each
branch to provide auto-attendant operation and announcements for waiting calls.
When selected, access to voicemail is routed via an SM line to the numbers
specified in the AAM Number field. The optional AAM PSTN Number can be
configured for use when the SM Line is not in service.
For a setup where the voicemail box numbers configured on Avaya Aura Messaging
or Modular Messaging are same as the caller's DID, the short code to route the
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Configuration Mode Field Descriptions
Field
Description
PSTN call should be such that the caller-id is withheld ( "W" in the telephonenumber of the shortcode ). This is to make sure that, during rainy day - the voicemail
system does not automatically go to the voicemail box of the caller based on the
caller id.
• CallPilot: Select this option if you want to configure the system to use CallPilot over
SIP as the central voicemail system. If you choose this option, you are still able to
use Embedded Voicemail or Voicemail Pro at each branch to provide autoattendant operation and announcements for waiting calls. When selected, access to
voicemail is routed via SM line to the numbers specified in the CallPilot Number
field.
Note:
The CallPilot PSTN Number field and associated Enable Voicemail
Instructions Using DTMF check box are not supported. IP Office cannot
access the CallPilot system over the PSTN when the Session Manager line is
down.
Note:
Users can access their CallPilot voicemail by dialing the Voicemail Collect
short code. Access to CallPilot voicemail from Auto Attendant cannot be
enabled by setting a Normal Transfer action to point to the Voicemail Collect
short code. If desired, it can be enabled by setting a Normal Transfer action to
point to the CallPilot number.
• Centralized Voicemail Select this option when using a Voicemail Pro system
installed and licensed on another system in a multi-site network. The outgoing line
group of the H.323 IP line connection to the system with the Voicemail Pro should
be entered as the Voicemail Destination. In a Server Edition network this option is
used on the Secondary Server and expansion systems to indicate that they use the
Primary Server for as their voicemail server.
• Distributed Voicemail: This option can be used when additional Voicemail Pro
voicemail servers are installed in a multi-site network and configured to exchange
messages with the central voicemail server using email. This option is used if this
system should use one of the additional servers for its voicemail services rather
than the central sever. When selected, the Voicemail Destination field is used for
the outgoing H.323 IP line to the central system and the Voicemail IP Address is
used for the IP address of the distributed voicemail server the system should use.
This option is not supported by Server Edition systems.
• Embedded Voicemail On systems with an Avaya memory card, select this option
to run Embedded Voicemail which stores messages and prompts on the memory
card. It also supports internal Auto Attendant configuration through the system
configuration. The IP500 V2 supports 2 simultaneous Embedded Voicemail calls by
default but can be licensed for up to 6. The licensed limit applies to total number of
callers leaving messages, collecting messages and or using an auto attendant. This
option is not supported by Server Edition systems.
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System
Field
Description
• Group Voicemail This option is used to support third-party voicemail systems
attached by extension ports in the group specified as the Voicemail Destination.
This option is not supported by Server Edition systems.
• Modular Messaging over SIP Select this option if you want to configure the system
to use Modular Messaging over SIP as the central voicemail system. If you choose
this option, you are still able to use Embedded Voicemail or Voicemail Pro at each
branch to provide auto-attendant operation and announcements for waiting calls.
When selected, access to voicemail is routed via an SM line to the numbers
specified in the MM Number field. The optional MM PSTN Number can be
configured for use when the SM Line is not in service.
Note:
Embedded Voicemail and Voicemail Pro are available only in Distributed
branch deployments. They are not available when there are centralized users
configured for an IP Office system that is deployed as either a Centralized
branch or a mixed branch.
The Embedded Voicemail option uses the Essential Edition and the Additional
Voicemail Ports licenses to control the number of ports that can be used. These
licenses are also used to control the number of ports on systems where Embedded
Voicemail is configured to provide local Auto Attendant and announcements while
the selected option for voicemail is one of the central voicemail options through the
Session Manager (i.e. Avaya Aura Messaging, Modular Messaging, or CallPilot).
Similarly, the Voicemail Pro option uses the Preferred Edition and the Incremental
Voicemail Ports licenses to control the number of ports that can be used. These
licenses are also used to control the number of ports on systems where Voicemail
Pro is configured to provide local Call Flow processing while the selected option for
voicemail is Avaya Aura Messaging, Modular Messaging or CallPilot.
- When the system routes a call to the voicemail server it indicates the locale for
which matching prompts should be provided if available. The locale sent to the
voicemail server by the system is determined as show below. If the required set of
prompts is not available, the voicemail will fallback to another appropriate
language and finally to English (refer to the appropriate voicemail installation
manual for details).
- Short Code Locale: The short code locale, if set, is used if the call is routed to
voicemail using the short code.
- Incoming Call Route Locale: The incoming call route locale, if set, is used if
caller is external.
- User Locale: The user locale, if set, is used if the caller is internal.
- System Locale: If no user or incoming call route locale is set, the system locale is
used unless overridden by a short code locale.
- Systems using Embedded Voicemail, if the required set of upgraded language
prompts to match the locale is not present on the system SD card, Manager will
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Configuration Mode Field Descriptions
Field
Description
display an error. The required prompt set can be uploaded from Manager using
the Add/Display VM Locales option.
• Remote Audix Voicemail: Select this option if using a remote Avaya Intuity Audix
or MultiMessage voicemail system. Requires entry of an Audix Voicemail license in
Licenses. This option is not supported by Server Edition systems.
• Voicemail Pro Select this option when using Voicemail Pro. The IP address of the
PC being used should be set as the Voicemail IP Address. In a Server Edition
network this option is used on the Primary Server. It can also be used on the
Secondary Server if the Secondary server is connected to its own voice mail server
or if the Secondary Server is part of an Outbound Contact Express deployment. Use
of Voicemail Pro requires licenses for the number of simultaneous calls to be
supported. Licenses are not required for an Outbound Contact Express deployment.
• Voicemail Mode: Default = IP Office Mode. Embedded Voicemail on IP500 V2
systems can use either IP Office Mode or Intuity Mode key presses for mailbox
functions. End users should be provided with the appropriate mailbox user guide for
the mode selected. You can switch between modes without losing user data, such
as passwords, greetings, or messages.
The following user guides are available from the Avaya support web site:
• IP Office Basic Edition - Embedded Voicemail User Guide (Intuity Mode)
• IP Office Basic Edition - Embedded Voicemail User Guide (IP Office Mode)
• IP Office Basic Edition Norstar Mode - Embedded Voicemail User Guide (Intuity
Mode)
• IP Office Basic Edition Norstar Mode - Embedded Voicemail User Guide (IP Office
Mode)
• IP Office Basic Edition Partner Mode - Embedded Voicemail User Guide (Intuity
Mode)
• IP Office Basic Edition Partner Mode - Embedded Voicemail User Guide (IP Office
Mode)
• IP Office Essential Edition - Embedded Voicemail User Guide (Intuity Mode)
• IP Office Essential Edition - Embedded Voicemail User Guide (IP Office Mode)
• IP Office Voicemail Pro Mailbox User Guide (Intuity Mode)
• IP Office Voicemail Pro Mailbox User Guide (IP Office Mode)
Add/Display VM
Locales
For new IP500 V2 SD cards and cards recreated using Manager, the following
Embedded Voicemail languages set are placed onto cards by default. Using this
option displays the list of languages that can be uploaded from Manager. Those
languages already present or not supported are greyed out. If a locale is selected for
the system, a user, a short code or an incoming call route which is not present on the
SD card, Manager will display an error. This command can be used to upload the
required language prompts to correct the error.
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System
Field
Description
Voicemail Destination
Defaults: Non-Server Edition = Blank, Server Edition = IP trunk connection to the
Primary Server.
• When the Voicemail Type is set to Remote Audix Voicemail, Centralized
Voicemail or Distributed Voicemail, this setting is used to enter the outgoing line
group of the line configured for connection to the phone system hosting the central
voicemail server.
• When the Voicemail Type is set to Group Voicemail, this setting is used to specify
the group whose user extensions are connected to the 3rd party voicemail system.
• When the Voicemail Type is set to Analogue Trunk MWI, this setting is used to
specify the phone number of the message center. All analogue trunks configured for
Analogue Trunk MWI must have the same destination.
Voicemail IP Address
Defaults: Non-Server Edition = 255.255.255.255, Primary Server = Primary Server IP
Address.
This setting is used when the Voicemail Type is set to Voicemail Pro or Distributed
Voicemail. It is the IP address of the PC running the voicemail server that the system
should use for its voicemail services. If set as 255.255.255.255, the control unit
broadcasts on the LAN for a response from a voicemail server. If set to a specific IP
address, the system connects only to the voicemail server running at that address. If
the system is fitted with an Unified Communication Module hosting Voicemail Pro, the
field should be set to 169.254.0.2.
Backup Voicemail IP
Address
Defaults: Primary Server = Secondary Server IP Address, All others = 0.0.0.0 (Off).
This option is supported with Voicemail Pro.
An additional voicemail server can be setup but left unused. If contact to the voicemail
server specified by the Voicemail IP Address is lost, responsibility for voicemail
services is temporarily transferred to this backup server address.
Audix UDP
Available if the voicemail type Remote Audix Voicemail is selected. Needs to be
completed with a four digit number from the Universal Dial Plan of the Avaya
Communication Manager system.
Voicemail Channel
Reservation
These settings allow the channels between the system and Voicemail Pro to be
reserved for particular functions. Unreserved channels can be used for any function
but reserved channels cannot be used for any function other than that indicated.
These settings are not available unless the configuration includes validated licenses
for the total number of voicemail channels.
Note that the voicemail server also restricts the maximum number of channels that
can be used for some services that would be taken from the Unreserved Channels
pool. Alarms and callbacks are each limited to up to 2 channels at any time. Outcalling
and conference invites are each limited to up to 5 channels at any time.
• Unreserved Channels: This setting cannot be changed and by default will show the
total number of licensed voicemail channels. This number will decrease as channels
are reserved for the following functions.
• Mailbox Access: Default = 0 This setting sets the number of channels reserved for
users accessing mailboxes to collect messages.
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Configuration Mode Field Descriptions
Field
Description
• Auto-Attendant:: Default = 0 This setting sets the number of channels reserved for
users directed to Voicemail Pro short code and module start points.
• Voice Recording: Default = 0 This setting sets the number of channels reserved for
voice recording other than mandatory voice recording (see below). If no channels
are available recording does not occur though recording progress may be indicated.
• Mandatory Voice Recording: Default = 0 This setting sets the number of channels
reserved for mandatory voice recording. When no channels are available for a call
set to mandatory recording, the call is barred and the caller hears busy tone.
• Announcements: Default = 0 This setting sets the number of channels reserved for
announcements. When no channels are available calls continue without
announcements.
Maximum Record
Time
Default = 120 seconds. Range = 30 to 180 seconds. This field is only available when
Embedded Voicemail is selected as the Voicemail Type. The value sets the
maximum record time for messages and prompts.
Messages Button
Goes to Visual Voice
Default = On.
Outcalling Control
Default = Off.
Visual Voice allows phone users to check their voicemail mailboxes and perform
action such as play, delete and forward messages through menus displayed on their
phone. By default, on phones with a MESSAGES button, the navigation is via spoken
prompts. This option allows that to be replaced by Visual Voice on phones that
support Visual Voice menus. For further details see Visual Voice.
This setting is used to enable or disable system wide outcalling on VMPro.
DTMF Breakout
Allows system defaults to be set. These are then applied to all user mailboxes unless the users own settings
differ.
The Park & Page feature is supported when the system voicemail type is configured as Embedded Voicemail
or Voicemail Pro. Park & Page is also supported on systems where Avaya Aura Messaging, Modular
Messaging over SIP, or CallPilot (for IP Office Aura Edition with CS 1000 deployments) is configured as the
central voice mail system and the local Embedded Voicemail or Voicemail Pro provides auto attendant
operation. The Park & Page feature allows a call to be parked while a page is made to a hunt group or
extension. This feature can be configured for Breakout DTMF 0, Breakout DTMF 2, or Breakout DTMF 3.
Reception/Breakout
(DTMF 0)
The number to which a caller is transferred if they press 0while listening to the
mailbox greeting rather than leaving a message (*0 on Embedded Voicemail in IP
Office Mode).
For voicemail systems set to Intuity emulation mode, the mailbox owner can also
access this option when collecting their messages by dialing *0.
If the mailbox has been reached through a Voicemail Pro call flow containing a Leave
Mail action, the option provided when 0 is pressed are:
• For IP Office mode, the call follows the Leave Mail action's Failure or Success
results connections depending on whether the caller pressed 0 before or after the
record tone.
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System
Field
Description
• For Intuity mode, pressing 0 always follows the Reception/Breakout (DTMF 0)
setting.
• When Park & Page is selected for a DTFM breakout, the following drop-down boxes
appear:
- Paging Number: Displays a list of hunt groups and users (extensions). Select a
hunt group or extension to configure this option.
- Retries: The range is 0 to 5. The default setting is 0.
- Retry TimeoutProvided in the format M:SS (minute:seconds). The range can be
set in 15-second increments. The minimum setting is 15 seconds and the
maximum setting is 5 minutes. The default setting is 15 seconds
Breakout (DTMF 2)
The number to which a caller is transferred if they press 2while listening to the
mailbox greeting rather than leaving a message (*2 on Embedded Voicemail in IP
Office Mode).
Breakout (DTMF 3)
The number to which a caller is transferred if they press 3while listening to the
mailbox greeting rather than leaving a message (*3 on Embedded Voicemail in IP
Office Mode).
Voicemail Code Complexity
Defines the requirements for the voicemail code.
For IP Office systems that have Voicemail Type set to Centralized, the Voicemail Code Complexity settings
must be the same as the IP Office system that is connected to Voicemail Pro.
Enforcement
Default = On.
When on, a user PIN is required.
Minimum Length
Default = 4. Maximum 15 digits.
Complexity
Default = On.
When on, the following complexity rules are enforced.
• The user extension number cannot be used.
• A PIN consisting of repeated digits is not allowed (1111).
• A PIN consisting of a sequence, forward or reverse, is not allowed (1234).
Call Recording
These setting relate to call recording. Call recording is only supported when using a Voicemail Pro voicemail
server.
Auto Restart Paused
Recording (secs)
Default = 15 seconds. Range = Never or 5 to 999 seconds.
Hide Auto Recording
Default = On (USA)/Off (Rest of World)
If recording, manual or automatic, of a call is halted using a Pause Recording button,
this timer determines when recording is restarted if the button is not pressed again.
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Configuration Mode Field Descriptions
Field
Description
During call recording by Voicemail Pro, some Avaya phones display REC or similar to
show that the call is being recorded. When on, hide auto recording suppresses this
recording indication.
SIP Settings
For Enterprise Branch deployments, these settings are used for calls made or
received on a SIP line where any of the line’s SIP URI fields are set to use internal
data. For Embedded Voicemail and Voicemail Pro, for calls made or received on a
SIP line where any of the line's SIP URI fields are set to Use Internal Data, that data
is taken from these settings. These options are shown if the system has SIP trunks
and is set to use Embedded Voicemail, Voicemail Lite/Pro, Centralized Voicemail
or Distributed Voicemail.
SIP Name
Default = Blank on Voicemail tab/Extension number on other tabs.
The value from this field is used when the From field of the SIP URI being used for a
SIP call is set to Use Internal Data.
SIP Display Name
(Alias)
Default = Blank on Voicemail tab/Name on other tabs.
Contact
Default = Blank on Voicemail tab/Extension number on other tabs. The value from this
field is used when the Contact field of the SIP URI being used for a SIP call is set to
Use Internal Data.
Anonymous
Default = On on Voicemail tab/Off on other tabs. If the From field in the SIP URI is set
to Use Internal Data, selecting this option inserts Anonymous into that field rather
than the SIP Name set above.
The value from this field is used when the Display Name field of the SIP URI being
used for a SIP call is set to Use Internal Data
Voicemail Language Prompts
When the system routes a call to the voicemail server it indicates the locale for which matching
prompts should be provided if available. The locale sent to the voicemail server by the system is
determined as show below. If the required set of prompts is not available, the voicemail will fallback
to another appropriate language and finally to English (refer to the appropriate voicemail installation
manual for details).
• Short Code Locale: The short code locale, if set, is used if the call is routed to voicemail using
the short code.
• Incoming Call Route Locale: The incoming call route locale, if set, is used if caller is external.
• User Locale: The user locale, if set, is used if the caller is internal.
• System Locale: If no user or incoming call route locale is set, the system locale is used unless
overridden by a short code locale.
Systems using Embedded Voicemail, if the required set of upgraded language prompts to match the
locale is not present on the system SD card, Manager will display an error. The required prompt set
can be uploaded from Manager using the Add/Display VM Locales option.
Related links
System on page 214
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System
System | Telephony
This tab is used to set the default telephony operation of the system. Some settings shown here can
be overridden for individual users through their User | Telephony tab. The settings are split into a
number of sub-tabs.
Related links
System on page 214
Telephony on page 241
Park and Page on page 252
Tones and Music on page 253
Ring Tones on page 259
SM on page 260
Call Log on page 263
TUI on page 264
Telephony
These settings have changed in release 9.1. View the 9.0 settings on page 247.
This page is used to configure a wide range of general purpose telephony settings for the whole
system.
These settings are mergeable. However, changes to Companding LAW and Automatic Codec
Preference require a reboot.
Field
Description
Analog Extensions
These settings apply only to analog extension ports provided by the system. For Server Edition this field is only
available on Expansion System (V2) systems
Default Outside Call
Sequence
Default = Normal
Default Inside Call
Sequence
Default = Ring Type 1
Default Ring Back
Sequence
Default = Ring Type 2
This setting is only used with analog extensions. It sets the ringing pattern used for
incoming external calls. For details of the ring types see Ring Tones. This setting
can be overridden by a user's User | Telephony | Call Settings | Outside Call
Sequence setting. Note that changing the pattern may cause fax and modem device
extensions to not recognize and answer calls.
This setting is only used with analog extensions. It sets the ringing pattern used for
incoming internal calls. For details of the ring types see Ring Tones. This setting can
be overridden by a user's User | Telephony | Call Settings | Inside Call Sequence
setting.
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Configuration Mode Field Descriptions
Field
Description
This setting is only used with analog extensions. It sets the ringing pattern used for
ringback calls such as hold return, park return, voicemail ringback, and Ring Back
when Free. For details of the ring types see Ring Tones. This setting can be
overridden by a user's User | Telephony | Call Settings | Ringback Call
Sequence setting.
Restrict Analog
Extension Ringer
Voltage
Default = Off.
Dial Delay Time (secs)
Supported on IP500 V2 systems only. If selected, the ring voltage on analogue
extension ports on the system is limited to a maximum of 40V Peak-Peak. Also
when selected, the message waiting indication (MWI) settings for analog extension
are limited to Line Reversal A, Line Reversal B or None. Any analog extension
already set to another MWI setting is forced to Line Reversal A.
Default = 4 (USA/Japan) or 1 (ROW). Range = 1 to 30 seconds.
This setting sets the time the system waits following a dialed digit before it starts
looking for a short code match. In situations where there are potential short codes
matches but not exact match, it also sets the delay following the dialing of a digit
before dialing complete is assumed. See the Short Codes section.
Dial Delay Count
Default = 0 digits (USA/Japan) or 4 digits (ROW). Range = 0 to 30 digits.
This setting sets the number of digits dialed after which the system starts looking for
a short code match regardless of the Dial Delay Time.
Default No Answer
Time (secs)
Default = 15 seconds. Range = 6 to 99999 seconds.
This setting controls the amount of time before an alerting call is considered as
unanswered. How the call is treated when this time expires depends on the call type.
For calls to a user, the call follows the user's Forward on No Answer settings if
enabled. If no forward is set, the call will go to voicemail if available or else
continues to ring. This timer is also used to control the duration of call forwarding if
the forward destination does not answer. It also controls the duration of ringback call
alerting. This setting is overridden by the User | Telephony | Call Settings | No
Answer Time setting for a particular user if different.
For calls to hunt groups, this setting controls the time before the call is presented to
the next available hunt group member. This setting is overridden by the Hunt Group
| Hunt Group | No Answer Time setting for a particular hunt group if different.
Hold Timeout (secs)
Default = Locale specific. Range = 0 (Off) to 99999 seconds.
This setting controls how long calls remain on hold before recalling to the user who
held the call. Note that the recall only occurs if the user has no other connected call.
Recalled calls will continue ringing and do not follow forwards or go to voicemail.
Park Timeout (secs)
Default = Locale specific. Range 0 (Off) to 99999 seconds.
This setting controls how long calls remain parked before recalling to the user who
parked the call. Note that the recall only occurs if the user has no other connected
call. Recalled calls will continue ringing and do not follow forwards or go to
voicemail.
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System
Field
Description
Ring Delay
Default = 5 seconds. Range = 0 to 98 seconds. This setting is used when any of the
user's programmed appearance buttons is set to Delayed ringing. Calls received on
that button will initially only alert visually. Audible alerting will only occur after the ring
delay has expired. This setting can be overridden by a ring delay set for an
individual user (User | Telephony | Multi-line Options | Ring Delay).
Call Priority Promotion
Time (secs)
Default = Disabled. Range = Disabled, 10 to 999 seconds.
When calls are queued for a hunt group, higher priority calls are placed ahead of
lower priority calls, with calls of the same priority sort by time in queue. External calls
are assigned a priority (1-Low, 2-Medium or 3-High) by the Incoming Call Route
that routed the call. Internal calls are assigned a priority of 1-Low. This option can
be used to increase the priority of a call each time it has remained queued for longer
than this value. The calls priority is increased by 1 each time until it reaches 3-High.
In situations where calls are queued, high priority calls are placed before calls of a
lower priority. This has a number of effects:
• Mixing calls of different priority is not recommended for destinations where
Voicemail Pro is being used to provided queue ETA and queue position messages
to callers since those values will no longer be accurate when a higher priority call
is placed into the queue. Note also that Voicemail Pro will not allow a value
already announced to an existing caller to increase.
• If the addition of a higher priority call causes the queue length to exceed the hunt
group's Queue Length Limit, the limit is temporarily raised by 1. This means that
calls already queued are not rerouted by the addition of a higher priority call into
the queue.
Default Currency
Default = Locale specific.
This setting is used with ISDN Advice of Charge (AOC) services. Note that changing
the currency clears all call costs stored by the system except those already logged
through SMDR. The currency is displayed in the system SMDR output.
Maximum SIP Sessions Default = 0.
This field is shown for Server Edition systems. On Server Edition systems, the
Maximum SIP Sessions value must match the total number of SIP set and trunk
calls that can occur at the same time.
The Maximum SIP Sessions setting determines the number of SIP Trunk Channel
licenses reserved for concurrent sessions on any SIP trunks provided by the server.
Those licenses are reserved from the pool of SIP Trunk Channel licenses in the
configuration of the Primary Server.
Default Name Priority
Default = Favour Trunk.
For SIP trunks, the caller name displayed on an extension can either be that
supplied by the trunk or one obtained by checking for a number match in the
extension user's personal directory and the system directory. This setting
determines which method is used by default. For each SIP line, this setting can be
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Configuration Mode Field Descriptions
Field
Description
overridden by the line's own Name Priority setting if required. Select one of the
following options:
• Favour Trunk: Display the name provided by the trunk. For example, the trunk
may be configured to provide the calling number or the name of the caller. The
system should display the caller information as it is provided by the trunk. If the
trunk does not provide a name, the system uses the Favour Directory method.
• Favour Directory: Search for a number match in the extension user's personal
directory and then in the system directory. The first match is used and overrides
the name provided by the SIP line. If no match is found, the name provided by the
line, if any, is used.
Media Connection
Preservation
Default = Enabled.
Phone Failback
Default = Automatic.
When enabled, attempts to maintain established calls despite brief network failures.
Call handling features are no longer available when a call is in a preserved state.
When enabled, Media Connection Preservation applies to SCN links and Avaya H.
323 phones that support connection preservation.
Applies to H.323 phones that support resiliency. The options are:
• Automatic
• Manual
Phones are permitted to failover to the secondary gatekeeper when the IP Office
Line link to the primary gatekeeper is down.
When set to Automatic, if a phone’s primary gatekeeper has been up for more than
10 minutes, the system causes the phone to failback if the phone is not in use. If the
phone is in use, the system will reattempt failback 10 seconds after the phone
ceases to be in use.
When set to Manual, phones remain in failover until manually restarted or reregistered, after which the phone attempts to fail back.
Login Code Complexity
Defines the requirements for the login code.
Enforcement
Default = On.
When on, a user PIN is required.
Minimum Length
Default = 4. Maximum 15 digits.
Complexity
Default = On.
When on, the following complexity rules are enforced.
• The user extension number cannot be used.
• A PIN consisting of repeated digits is not allowed (1111).
• A PIN consisting of a sequence is not allowed (1234).
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System
Field
Description
Companding Law
These settings should not normally be changed from their defaults. They should only
be used where 4400 Series phones (ULAW) are installed on systems which have ALaw digital trunks.
A-Law or U-Law> PCM (Pulse Code Modulation) is a method for encoding voice as
data. In telephony, two methods of PCM encoding are widely used, A-Law and ULaw (also called Mu-Law or µ-Law). Typically U-Law is used in North America and a
few other locations while A-Law is used elsewhere. As well as setting the correct
PCM encoding for the region, the A-Law or U-Law setting of a system when it is first
started affects a wide range of regional defaults relating to line settings and other
values.
For IP500 V2 systems, the encoding default is set by the type of Feature Key
installed when the system is first started. The cards are either specifically A-Law or
U-Law>. PARTNER Mode cards are U-Law. Norstar Mode cards are A-Law.
DSS Status
Default = Off
This setting affects Avaya display phones with programmable buttons. It controls
whether pressing a DSS key set to another user who has a call ringing will display
details of the caller. When off, no caller information is displayed.
Auto Hold
Default = On (Off for the United States locale).
Used for users with multiple appearance buttons. When on, if a user presses
another appearance button during a call, their current call is placed on hold. When
off, if a users presses another appearance button during a call, their current call is
disconnected.
Dial By Name
Default = On
When on, allows the directory features on various phones to match the dialing of full
names. This option is fixed as On and is not adjustable.
Show Account Code
Default = On This setting controls the display and listing of system account codes.
• When on: When entering account codes through a phone, the account code digits
are shown while being dialed.
• When off: When entering account codes through a phone, the account code digits
are replaced by s characters on the display.
Inhibit Off-Switch
Forward/Transfer
Default = On
Restrict Network
Interconnect
Default = Off.
When enabled, this setting stops any user from transferring or forwarding calls
externally. See Off-Switch Transfer Restrictions.
When this option is enabled, each trunk is provided with a Network Type option that
can be configured as either Public or Private. The system will not allow calls on a
public trunk to be connected to a private trunk and vice versa, returning number
unobtainable indication instead.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
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Configuration Mode Field Descriptions
Field
Description
Include location
specific information
Default = Off.
When set to On, this setting is available in the trunk configuration settings when
Network Type is set to Private.
Set to On if the PBX on the other end of the trunk is toll compliant.
Drop External Only
Impromptu Conference
Default = On.
Visually Differentiate
External Call
Default = Off.
Unsupervised Analog
Trunk Disconnect
Handling
Default = Off.
If selected, when the last remaining internal user in a conference exits the
conference, the conference is ended, regardless of whether it contains any external
callers. If not selected, the conference is automatically ended when the last internal
party or trunk that supports reliable disconnect exits the conference. The Inhibit OffSwitch Forward/Transfer option above is no longer applied to conference calls.
This setting is applied to the lamp flashing rate used for bridged appearance and call
coverage appearance buttons on 1400, 1600 and 9600 Series phones and on their
button modules. When selected, external calls alerting on those buttons will use a
slow flash (200ms on/50ms off). If not selected or if the call is internal, normal
flashing (500ms on/500ms off) is used.
When using analog trunks, various methods are used for trunk supervision, ie. to
detect when the far end of the trunk has disconnected and so disconnect the local
end of the call. Depending on the locale, the system uses Disconnect Clear
signalling and or Busy Tone Detection. This setting should only be enabled if it is
know that the analog trunks do not provide disconnect clear signalling or reliable
busy tone. For Server Edition this field is only available on Expansion System (V2)
systems. When enabled:
• Disconnect Clear signalling detection is disabled. Busy tone detection remains on.
• Unsupervised transfers and trunk-to-trunk transfers of analog trunk calls are not
allowed. The Allow Analog Trunk to Trunk Connect setting on analog trunks
(Line | Analog Options) is disabled.
• If Voicemail Pro is being used for external call transfers, Supervised Transfer
actions should be used in call flows rather than Transfer actions.
• All systems in the network must have this setting set to match each other.
High Quality
Conferencing
Digital/Analogue Auto
Create User
Default = On.
Supports the use of the G.722 codec. IP lines and extensions using G.722 are
provided with wide band audio. If High Quality Conferencing is enabled, when
several wide band audio devices are in the same conference, the system will ensure
that the audio between them remains wide band, even if the conference also
contains other lines and devices using narrow band audio (analog devices, digital
devices and IP devices using codecs other than G.722).
Default = On. (Not supported on Server Edition Linux systems. Default = Off for
Server Edition.)
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System
Field
Description
When enabled, an associated user is created for each digital/analogue extension
created. Digital/analogue extension creation occurs on initial start up, reset of
configuration, or addition of new digital/analogue expansion units or plug-in
modules.
Directory Overrides
Barring
Default = On.
When enabled, barred numbers are not barred if the dialed number is in the External
Directory.
Related links
System | Telephony on page 241
Telephony (9.0) on page 247
Telephony (9.0)
This page is used to configure a wide range of general purpose telephony settings for the whole
system.
These settings are mergeable. However, changes to Companding LAW and Automatic Codec
Preference require a reboot.
Field
Description
Analog Extensions
These settings apply only to analog extension ports provided by the system. For Server Edition this field is only
available on Expansion System (V2) systems
Default Outside Call
Sequence
Default = Normal
Default Inside Call
Sequence
Default = Ring Type 1
Default Ring Back
Sequence
Default = Ring Type 2
Restrict Analog
Extension Ringer
Voltage
Default = Off.
This setting is only used with analog extensions. It sets the ringing pattern used for
incoming external calls. For details of the ring types see Ring Tones. This setting
can be overridden by a user's User | Telephony | Call Settings | Outside Call
Sequence setting. Note that changing the pattern may cause fax and modem device
extensions to not recognize and answer calls.
This setting is only used with analog extensions. It sets the ringing pattern used for
incoming internal calls. For details of the ring types see Ring Tones. This setting can
be overridden by a user's User | Telephony | Call Settings | Inside Call Sequence
setting.
This setting is only used with analog extensions. It sets the ringing pattern used for
ringback calls such as hold return, park return, voicemail ringback, and Ring Back
when Free. For details of the ring types see Ring Tones. This setting can be
overridden by a user's User | Telephony | Call Settings | Ringback Call
Sequence setting.
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Configuration Mode Field Descriptions
Field
Description
Supported on IP500 V2 systems only. If selected, the ring voltage on analogue
extension ports on the system is limited to a maximum of 40V Peak-Peak. Also
when selected, the message waiting indication (MWI) settings for analog extension
are limited to Line Reversal A, Line Reversal B or None. Any analog extension
already set to another MWI setting is forced to Line Reversal A.
Dial Delay Time (secs)
Default = 4 (USA/Japan) or 1 (ROW). Range = 1 to 30 seconds.
This setting sets the time the system waits following a dialed digit before it starts
looking for a short code match. In situations where there are potential short codes
matches but not exact match, it also sets the delay following the dialing of a digit
before dialing complete is assumed. See the Short Codes section.
Dial Delay Count
Default = 0 digits (USA/Japan) or 4 digits (ROW). Range = 0 to 30 digits.
This setting sets the number of digits dialed after which the system starts looking for
a short code match regardless of the Dial Delay Time.
Default No Answer
Time (secs)
Default = 15 seconds. Range = 6 to 99999 seconds.
This setting controls the amount of time before an alerting call is considered as
unanswered. How the call is treated when this time expires depends on the call type.
For calls to a user, the call follows the user's Forward on No Answer settings if
enabled. If no forward is set, the call will go to voicemail if available or else
continues to ring. This timer is also used to control the duration of call forwarding if
the forward destination does not answer. It also controls the duration of ringback call
alerting. This setting is overridden by the User | Telephony | Call Settings | No
Answer Time setting for a particular user if different.
For calls to hunt groups, this setting controls the time before the call is presented to
the next available hunt group member. This setting is overridden by the Hunt Group
| Hunt Group | No Answer Time setting for a particular hunt group if different.
Hold Timeout (secs)
Default = Locale specific. Range = 0 (Off) to 99999 seconds.
This setting controls how long calls remain on hold before recalling to the user who
held the call. Note that the recall only occurs if the user has no other connected call.
Recalled calls will continue ringing and do not follow forwards or go to voicemail.
Park Timeout (secs)
Default = Locale specific. Range 0 (Off) to 99999 seconds.
This setting controls how long calls remain parked before recalling to the user who
parked the call. Note that the recall only occurs if the user has no other connected
call. Recalled calls will continue ringing and do not follow forwards or go to
voicemail.
Ring Delay
Default = 5 seconds. Range = 0 to 98 seconds. This setting is used when any of the
user's programmed appearance buttons is set to Delayed ringing. Calls received on
that button will initially only alert visually. Audible alerting will only occur after the ring
delay has expired. This setting can be overridden by a ring delay set for an
individual user (User | Telephony | Multi-line Options | Ring Delay).
Call Priority Promotion
Time (secs)
Default = Disabled. Range = Disabled, 10 to 999 seconds.
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System
Field
Description
When calls are queued for a hunt group, higher priority calls are placed ahead of
lower priority calls, with calls of the same priority sort by time in queue. External calls
are assigned a priority (1-Low, 2-Medium or 3-High) by the Incoming Call Route
that routed the call. Internal calls are assigned a priority of 1-Low. This option can
be used to increase the priority of a call each time it has remained queued for longer
than this value. The calls priority is increased by 1 each time until it reaches 3-High.
In situations where calls are queued, high priority calls are placed before calls of a
lower priority. This has a number of effects:
• Mixing calls of different priority is not recommended for destinations where
Voicemail Pro is being used to provided queue ETA and queue position messages
to callers since those values will no longer be accurate when a higher priority call
is placed into the queue. Note also that Voicemail Pro will not allow a value
already announced to an existing caller to increase.
• If the addition of a higher priority call causes the queue length to exceed the hunt
group's Queue Length Limit, the limit is temporarily raised by 1. This means that
calls already queued are not rerouted by the addition of a higher priority call into
the queue.
Default Currency
Default = Locale specific.
This setting is used with ISDN Advice of Charge (AOC) services. Note that changing
the currency clears all call costs stored by the system except those already logged
through SMDR. The currency is displayed in the system SMDR output.
Maximum SIP Sessions Default = 0.
This field is shown for Server Edition systems. On Server Edition systems, the
Maximum SIP Sessions value must match the total number of SIP set and trunk
calls that can occur at the same time.
The Maximum SIP Sessions setting determines the number of SIP Trunk Channel
licenses reserved for concurrent sessions on any SIP trunks provided by the server.
Those licenses are reserved from the pool of SIP Trunk Channel licenses in the
configuration of the Primary Server.
Default Name Priority
Default = Favour Trunk.
For SIP trunks, the caller name displayed on an extension can either be that
supplied by the trunk or one obtained by checking for a number match in the
extension user's personal directory and the system directory. This setting
determines which method is used by default. For each SIP line, this setting can be
overridden by the line's own Name Priority setting if required. Select one of the
following options:
• Favour Trunk: Display the name provided by the trunk. For example, the trunk
may be configured to provide the calling number or the name of the caller. The
system should display the caller information as it is provided by the trunk. If the
trunk does not provide a name, the system uses the Favour Directory method.
• Favour Directory: Search for a number match in the extension user's personal
directory and then in the system directory. The first match is used and overrides
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Configuration Mode Field Descriptions
Field
Description
the name provided by the SIP line. If no match is found, the name provided by the
line, if any, is used.
Media Connection
Preservation
Default = Enabled.
Companding Law
These settings should not normally be changed from their defaults. They should only
be used where 4400 Series phones ULAW are installed on systems which have ALaw digital trunks.
When enabled, attempts to maintain established calls despite brief network failures.
Call handling features are no longer available when a call is in a preserved state.
When enabled, Media Connection Preservation applies to SCN links and Avaya H.
323 phones that support connection preservation.
A-Law or U-Law> PCM (Pulse Code Modulation) is a method for encoding voice as
data. In telephony, two methods of PCM encoding are widely used, A-Law and ULaw (also called Mu-Law or µ-Law). Typically U-Law is used in North America and a
few other locations while A-Law is used elsewhere. As well as setting the correct
PCM encoding for the region, the A-Law or U-Law setting of a system when it is first
started affects a wide range of regional defaults relating to line settings and other
values.
For IP500 V2 systems, the encoding default is set by the type of Feature Key
installed when the system is first started. The cards are either specifically A-Law or
U-Law>. PARTNER Mode cards are U-Law. Norstar Mode cards are A-Law.
DSS Status
Default = Off
This setting affects Avaya display phones with programmable buttons. It controls
whether pressing a DSS key set to another user who has a call ringing will display
details of the caller. When off, no caller information is displayed.
Auto Hold
Default = On.
Used for users with multiple appearance buttons. When on, if a user presses
another appearance button during a call, their current call is placed on hold. When
off, if a users presses another appearance button during a call, their current call is
disconnected.
Dial By Name
Default = On
When on, allows the directory features on various phones to match the dialing of full
names. This option is fixed as On and is not adjustable.
Show Account Code
Default = On This setting controls the display and listing of system account codes.
• When on: When entering account codes through a phone, the account code digits
are shown while being dialed.
• When off: When entering account codes through a phone, the account code digits
are replaced by s characters on the display.
Inhibit Off-Switch
Forward/Transfer
Default = On
When enabled, this setting stops any user from transferring or forwarding calls
externally. See Off-Switch Transfer Restrictions.
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System
Field
Description
Restrict Network
Interconnect
Default = Off.
When this option is enabled, each trunk is provided with a Network Type option that
can be configured as either Public or Private. The system will not allow calls on a
public trunk to be connected to a private trunk and vice versa, returning number
unobtainable indication instead.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Include location
specific information
Default = Off.
Drop External Only
Impromptu Conference
Default = On.
Visually Differentiate
External Call
Default = Off.
Unsupervised Analog
Trunk Disconnect
Handling
Default = Off.
If selected, when the last remaining internal user in a conference exits the
conference, the conference is ended, regardless of whether it contains any external
callers. If not selected, the conference is automatically ended when the last internal
party or trunk that supports reliable disconnect exits the conference. The Inhibit OffSwitch Forward/Transfer option above is no longer applied to conference calls.
This setting is applied to the lamp flashing rate used for bridged appearance and call
coverage appearance buttons on 1400, 1600 and 9600 Series phones and on their
button modules. When selected, external calls alerting on those buttons will use a
slow flash (200ms on/50ms off). If not selected or if the call is internal, normal
flashing (500ms on/500ms off) is used.
When using analog trunks, various methods are used for trunk supervision, ie. to
detect when the far end of the trunk has disconnected and so disconnect the local
end of the call. Depending on the locale, the system uses Disconnect Clear
signalling and or Busy Tone Detection. This setting should only be enabled if it is
know that the analog trunks do not provide disconnect clear signalling or reliable
busy tone. For Server Edition this field is only available on Expansion System (V2)
systems. When enabled:
• Disconnect Clear signalling detection is disabled. Busy tone detection remains on.
• Unsupervised transfers and trunk-to-trunk transfers of analog trunk calls are not
allowed. The Allow Analog Trunk to Trunk Connect setting on analog trunks
(Line | Analog Options) is disabled.
• If Voicemail Pro is being used for external call transfers, Supervised Transfer
actions should be used in call flows rather than Transfer actions.
• All systems in the network must have this setting set to match each other.
High Quality
Conferencing
Default = On.
Supports the use of the G.722 codec. IP lines and extensions using G.722 are
provided with wide band audio. If High Quality Conferencing is enabled, when
several wide band audio devices are in the same conference, the system will ensure
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Configuration Mode Field Descriptions
Field
Description
that the audio between them remains wide band, even if the conference also
contains other lines and devices using narrow band audio (analog devices, digital
devices and IP devices using codecs other than G.722).
Strict SIPS
(Enterprise Branch deployments) Default = Off.
This option provides a system-wide configuration for call restrictions based on SIPS
URI.
When this option is off, calls are not rejected due to SIPS. A call is sent according to
the configuration of the outgoing trunk or line that it is routed to, regardless of the
way the call came in, even if the call came in as a SIP invite with SIPS URI and is
being sent with a SIP URI onto a non-secure SIP trunk.
When this option is on, an incoming SIP invite with SIPS URI if targeted to a SIP
trunk (SM line or SIP line) is rejected if the target trunk is not configured with SIPS in
the URI Type field.
Digital/Analogue Auto
Create User
Default = On. (Off for Server Edition)
Directory Overrides
Barring
Default = On.
When enabled, an associated user is created for each digital/analogue extension
created. Digital/analogue extension creation occurs on initial start up, reset of
configuration, or addition of new digital/analogue expansion units or plug-in
modules.
When enabled, barred numbers are not barred if the dialed number is in the External
Directory.
Related links
Telephony on page 241
Park and Page
The Park and Page tab allows for simple configuration of the of the short code and the
programmable button for the park and page function.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
Central Park Range
Default = Blank. Range = nX to nnnnnnnXX The park slot ID range definition,
where n is a digit sequence from 1 to 9999999 and X represents a park slot value
from 0 to 99. The Central Park Range cannot exceed 9 characters total length.
Examples:
• 1X defines range 10-19
• 3XX defines range 300-399
• 9876543XX defines range 987654300-987654399
Page Target Group List
February 2016
Default = Blank. The list of paging group targets that are presented on supported
phones if the Page action is requested after the Call Park.
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System
Field
Description
On some phones, only the first three groups can be presented as Page options
(via the Softkeys on the phone). On phones that support scrolling lists, a larger list
of possible Page targets can be presented.
Related links
System | Telephony on page 241
Tones and Music
These settings have changed in release 9.0. View the release 8.1 settings on page 247.
This page is used to configure the various tones and music on hold sources used by the system.
The settings are mergable with the following exceptions.
• Changes to Disconnect Tone requires a reboot.
• Changes to Busy Tone Detection requires a reboot.
• For Hold Music, changes to the System Source requires a reboot. Deleting any of the hold
music Alternate Sources also requires a reboot.
For additional information on configuring hold music, see Music On Hold on page 585.
Field
Description
Conferencing Tone
Default = Entry & Exit Tones.
This settings controls how conference tones are used. The options are:
• Entry & Exit Tones: Asingle tone is heard when a new party joins a conference and
double-tone is heard when a party leaves the conference.
• Repeating ToneA conference tone is heard every 10 seconds by all conference
parties.
Note that no conference tones are played in an a conference initiated by an Outbound
Contact Express agent.
Disconnect Tone
Default = Default (Use locale setting).
For digital and IP phones, when the system detects that the far end of a call has
disconnected, it can make the near end either go idle or play disconnect tone. By
default, the chosen behavior depends on the system locale. This field can be used to
override the locale's default action and force either disconnect tone or go idle. The
options are:
• Default: Use the system locale specific action for disconnected calls.
• On: Play disconnect tone when far end disconnection is detected.
• Off: Go idle when far end disconnection is detected.
Busy Tone Detection
Default = Off. Enables or disables the use of busy tone detection for call clearing. This
is a system wide setting.
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Configuration Mode Field Descriptions
Field
Description
CLI Type
This field is used to set the CLI detection used for incoming analogue trunks. Note that
the CLI Type field is shown for locales other than Customize. For the Customize
locale it is set through the System | System form.The options are:
• DTMF
• FSK V23
• FSK BELL202
Local Dial Tone
Default = On
For all normal operation this setting should be left enabled as it allows the system to
provide dial tone to users (essential for MSN working).
Local Busy Tone
Default = Off
This setting should only be used when the local exchange gives a busy signal via Q.
931 but does not provide busy tone.
Beep on Listen
Default = On
This setting controls whether call parties hear a repeating tone when their call is
monitored by another party using the Call Listen feature.
Warning:
The use of features to listen to a call without the other call parties being aware of
that monitoring may be subject to local laws and regulations. Before enabling the
feature you must ensure that you have complied with all applicable local laws
and regulations. Failure to do so may result in severe penalties.
GSM Silence
Suppression
Default = Off.
Analogue Trunk VAD
Default = Off.
This setting should only be selected if voice quality problems are experienced with
calls to voicemail or while recording calls. When on, the system signals silence by
generating silence data packets in periods when the voicemail system is not playing
prompts. Note that use of this option may cause some timeout routing options in
voicemail to no longer work.
Select this option to enable Voice Activity Detection (VAD) for analog trunks
terminating on the ATM4U-V2 card. VAD functionality provides a Call Answer signal
triggered by voice activity. This signal can be used for:
• Mobile Twinning
• SMDR
• Call Forwarding
• Call Display
• Mobile Call Control
• Transfer Ringing Call
• TAPI
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System
Field
Description
• Trunk to Trunk Call
Busy Tone Detection
Default = System Frequency (Tone defined by system locale) Allows configuration of
the system's busy tone detection settings on lines that do not provide reliable
disconnect signalling. In that case, the system will use tone disconnect clearing to
disconnect such lines after 6 seconds of continuous tone. The default tone (frequency
and on/off cadence) detection used is defined by the system locale. The settings
should not be adjusted unless advised by Avaya Technical Support. Changes to this
setting require a reboot rather than a merge when the new configuration is sent to the
system. .For Server Edition this field is only available on Expansion System (V2)
systems.
Hold Music
This section is used to define the source for the system's music on hold source. You must ensure that any MOH
source you use complies with copyright, performing rights and other local and national legal requirements.
Server Edition deployments support centralized music on hold, where the Primary Server streams music to the
Secondary Server and all expansion servers.
The WAV file properties must be:
• PCM
• 8kHz 16-bit
• mono
• maximum length 90 seconds (30 seconds on non-IP500 systems, 600 seconds on Linux based systems)
If the file downloaded is the incorrect format, it will be discarded from memory after the download.
The first WAV file, for the system source, must be called HoldMusic.wav. Alternate source WAV files:
• can be up to 27 IA5 characters
• cannot contain spaces
• any extension is allowed
• case sensitive
System Source
Default = WAV File.
Selects the default hold music source for most uses of music on hold. Note that
changes to the System Source requires a reboot. The options are:
• WAV: Use the WAV file HoldMusic.wav. This file is loaded via TFTP.
Note that on Linux systems, the file name is case sensitive.
• WAV (restart): Identical to WAV except that for each new listener, the file plays
from the beginning.
Not supported on IP500 V2 systems. Cannot be used as a centralized source.
• External: Applicable to IP 500 v2 systems. Use the audio source connected to the
back of the control unit.
• Tone: The use of a double beep tone (425Hz, 02./0.2/0.2/3.4 seconds on/off) can be
selected as the system source. The hold music tone is automatically used if the
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Configuration Mode Field Descriptions
Field
Description
system source is set to WAV File but the HoldMusic.wav file has not yet been
successfully downloaded.
Alternate Sources
Up to three additional hold music source can be specified. Note that adding and
changing a source can be done using a merge but deleting a source requires a
reboot.
• Number: Assigned automatically by the system. The alternate sources are
numbered 2, 3 and 4.
• Name: Up to 31 characters This field is used to associate a name with the alternate
source. That name is then used in the Hold Music Source field on Incoming Call
Routes and Hunt Groups.
• Source: Up to 31 characters. Defines the source for the music on hold.
The options are listed below with a brief description. For more information, see
Alternate Source on page 588.
- WAV: To specify a wav file, enter WAV: followed by the file name. Playback
resumes from where it left off the last time.
- XTN: Any analog extension.
Not applicable to Linux systems.
- WAVRST: To specify a wav file, enter WAVRST: followed by the file name.
Playback is started every time from the beginning.
- WAVDIR: Multiple WAV file source. The WAV files must be stored in the /opt/
ipoffice/tones/mohwavdir/ directory. Playback resumes from where it left
off the last time.
Not applicable to IP500 V2 systems.
- WAVDIRRST: Multiple WAV file source. The WAV files must be stored in
the /opt/ipoffice/tones/mohwavdir/ directory. Playback is started every
time from the beginning.
Not applicable to IP500 V2 systems.
- USB: Supports multiple USB inputs. Enter USB:<number>.
Not applicable to IP500 V2 systems.
- LINE: In Server Edition networks, setting the Secondary Server and Expansion
Server Alternate Source to Line allows the server to receive streamed audio
from a source on the Primary Server. On the Secondary Server and Expansion
Server, enter Line:x,ywhere x is the line number to the Primary Server and y is
the MOH source number on the Primary Server.
Related links
System | Telephony on page 241
Tones and Music (9.0) on page 257
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System
Tones and Music (9.0)
This page is used to configure the various tones and music on hold sources used by the system.
Changes to Busy Tone Detection requires a reboot.
For Hold Music, changes to the System Source requires a reboot. Deleting any of the hold music
Alternate Sources also requires a reboot.
Field
Description
Conferencing Tone
Default = Entry & Exit Tones.
This settings controls how conference tones are used. The options are:
• Entry & Exit Tones: Asingle tone is heard when a new party joins a conference and
double-tone is heard when a party leaves the conference.
• Repeating ToneA conference tone is heard every 10 seconds by all conference
parties.
Note that no conference tones are played in an a conference initiated by an Outbound
Contact Express agent.
Disconnect Tone
Default = Default (Use locale setting).
For digital and IP phones, when the system detects that the far end of a call has
disconnected, it can make the near end either go idle or play disconnect tone. By
default, the chosen behavior depends on the system locale. This field can be used to
override the locale's default action and force either disconnect tone or go idle. The
options are:
• Default: Use the system locale specific action for disconnected calls.
• On: Play disconnect tone when far end disconnection is detected.
• Off: Go idle when far end disconnection is detected.
Busy Tone Detection
Default = Off. Enables or disables the use of busy tone detection for call clearing. This
is a system wide setting.
CLI Type
This field is used to set the CLI detection used for incoming analogue trunks. Note that
the CLI Type field is shown for locales other than Customize. For the Customize
locale it is set through the System | System form.The options are:
• DTMF
• FSK V23
• FSK BELL202
Local Dial Tone
Default = On
For all normal operation this setting should be left enabled as it allows the system to
provide dial tone to users (essential for MSN working).
Local Busy Tone
Default = Off
This setting should only be used when the local exchange gives a busy signal via Q.
931 but does not provide busy tone.
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Configuration Mode Field Descriptions
Field
Description
Beep on Listen
Default = On
This setting controls whether call parties hear a repeating tone when their call is
monitored by another party using the Call Listen feature.
Warning:
The use of features to listen to a call without the other call parties being aware of
that monitoring may be subject to local laws and regulations. Before enabling the
feature you must ensure that you have complied with all applicable local laws
and regulations. Failure to do so may result in severe penalties.
GSM Silence
Suppression
Default = Off.
Busy Tone Detection
Default = System Frequency (Tone defined by system locale) Allows configuration of
the system's busy tone detection settings on lines that do not provide reliable
disconnect signalling. In that case, the system will use tone disconnect clearing to
disconnect such lines after 6 seconds of continuous tone. The default tone (frequency
and on/off cadence) detection used is defined by the system locale. The settings
should not be adjusted unless advised by Avaya Technical Support. Changes to this
setting require a reboot rather than a merge when the new configuration is sent to the
system. .For Server Edition this field is only available on Expansion System (V2)
systems.
This setting should only be selected if voice quality problems are experienced with
calls to voicemail or while recording calls. When on, the system signals silence by
generating silence data packets in periods when the voicemail system is not playing
prompts. Note that use of this option may cause some timeout routing options in
voicemail to no longer work.
Hold Music
This section is used to define the source for the system's music on hold source. You must ensure that any MOH
source you use complies with copyright, performing rights and other local and national legal requirements.
Server Edition deployments support centralized music on hold, where the Primary Server streams music to the
Secondary Server and all expansion servers.
System Source
Default = WAV File.
Selects the default hold music source for most uses of music on hold. Note that
changes to the System Source requires a reboot. The options are:
• WAV: Use the WAV file HoldMusic.wav. This file is loaded via TFTP.
Note that on Linux systems, the file name is case sensitive.
• WAV (restart): Identical to WAV except that for each new listener, the file plays
from the beginning.
Not supported on IP500 V2 systems. Cannot be used as a centralized source.
• External: For IP 500 v2 systems, use the audio source connected to the back of the
control unit.
For Linux servers, use the first available USB audio source.
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System
Field
Description
• Tone: The use of a double beep tone (425Hz, 02./0.2/0.2/3.4 seconds on/off) can be
selected as the system source. The hold music tone is automatically used if the
system source is set to WAV File but the HoldMusic.wav file has not yet been
successfully downloaded.
Alternate Sources
Up to three additional hold music source can be specified. Note that adding and
changing a source can be done using a merge but deleting a source requires a
reboot.
• Number: : Assigned automatically by the system. The alternate sources are
numbered 2, 3 and 4.
• Name:: Up to 31 characters This field is used to associate a name with the alternate
source. That name is then used in the Hold Music Source field on Incoming Call
Routes and Hunt Groups.
• Source:: Up to 31 characters Defines the source for the music on hold.
- WAV: To specify a wav file, enter WAV: followed by the file name. Playback
resumes from where it left off the last time.
- XTN: Any analog extension with its Equipment Classification set as MOH Source
can be entered as the alternate source. Enter XTN: followed by the extension's
Base Extension number. For example XTN:224.
- WAVDIR: Multiple WAV file source. The WAV files must be stored in the /opt/
ipoffice/tones/mohwavdir/ directory. The directory can contain up to 255 files and
each file can be up to 10 minutes in length.
Not applicable to Server Edition IP500 V2 expansion systems.
- USB: Supports multiple USB inputs. Enter USB:<number>.
USB:1 is the first source found and is treated as External when the System
Source is set to External. Additional devices are numbered sequentially. For
example USB:2, USB:3.
- LINE: In Server Edition networks, setting the Secondary Server and Expansion
Server Alternate Source to Line allows the server to receive streamed audio
from a source on the Primary Server. On the Secondary Server and Expansion
Server, enter Line:x,ywhere x is the line number to the Primary Server and y is
the MOH source number on the Primary Server.
Related links
Tones and Music on page 253
Ring Tones
Navigation: System | Telephony | Ring Tones
Additional configuration information
For additional ring tone configuration information, see Ring Tones. on page 583
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Configuration Mode Field Descriptions
Configuration settings
Used to configure distinct ring tones for groups and incoming call routes. Ring Tone configuration is
only supported on 1400 series and 9500 series phones.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
Available Ring
Tones
In this table, the Number, Name, and Source values are system supplied. The
Name value is used to create a ring tone plan.
Ring Tone Plan
Use this table to specify available ring tones. Ring tones in this table can be applied
to hunt groups and incoming call routes and by short codes.
• Number: System supplied.
The Number can be used in a short code by adding r(x) to the Telephone
Number field, where x = 1 to 8 and specifies which ring tone plan to use.
• Name: A descriptive name for where this ring tone is used. For example, the
name of a hunt group. Each name in the table must be unique. Once configured in
this table, ring tone names can be selected from the Ring Tone Override field at:
- Group | Group
- Incoming Call Route | Standard
Ring Tone Override is supported on 1400 and 9500 series phones.
• Ring Tone: The list of ring tone names from the Available Ring Tones table.
Related links
System | Telephony on page 241
SM
These settings have changed in release 9.1. View the 9.0 settings on page 262.
This page is used to configure settings that apply to both SM lines.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
Short Form Dialing
Length
Default = 0. Range = 0 to 14.
Branch Prefix
Default = Blank. Maximum range = 15 digits.
This number specifies the short-form dialing length for all Centralized users and
Groups. Configuration of this field allows IP Office to treat the last N digits (where N
is the number entered in this field) of each Centralized user’s extension number as
an alias to that user’s extension number. For example, if a Centralized user’s
extension number is 5381111 and the Short Form Dialing Length is 4, the system
will match calls to 1111 with this extension. When 1111 is dialed by another user on
the system, entered from the auto attendant, or comes from the ICR, then in sunnyday that call will be sent to Session Manager with the number converted to 5381111
and in rainy-day it will target the extension 5381111 locally.
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System
Field
Description
This number is used to identify the IP Office system within the Avaya Aura®
network. The branch prefix of each IP Office system must be unique and must not
overlap. For example 85, 861 and 862 are okay, but 86 and 861 overlap. On calls
routed via an SM Line, the branch prefix is added to the caller's extension number.
You have the option to leave the Branch Prefix field blank. If you do not configure
the branch prefix, the IP Office user extensions must be defined with the full
enterprise number.
Local Number
Length
Default = Blank (Off). Range = Blank or 3 to 9 in deployments with IP Office users
and blank or 3 to 15 in deployments with only centralized users.
This field sets the default length for extension numbers for extensions, users, and
groups added to the IP Office configuration. Entry of an extension number of a
different length will cause an error.
The number of digits entered in the Branch Prefix field plus the value entered in
the Local Number Length field must not exceed 15 digits. You have the option to
leave the Local Number Length field blank.
Proactive
Monitoring
Default = 60 seconds. Range = 60 seconds to 100000 seconds.
Monitoring Retries
Default = 1. Range = 0 to 5.
The Enterprise Branch system sends regular SIP OPTIONS messages to the SM
line in order to check the status of line. This setting controls the frequency of the
messages when the SM line is currently in service. Centralized SIP phones use
their own settings.
The number of times the Enterprise Branch system retrys sending an OPTIONS
request to Session Manager before the SM Line is marked out-of-service.
Reactive Monitoring
Default 60 seconds. Range = 10 to 3600 seconds.
The Enterprise Branch system sends regular SIP OPTIONS messages to the SM
line in order to check the status of line. This setting controls the frequency of the
messages when the SM line is currently out of service. Centralized SIP phones use
their own settings.
Failback Policy
Default = Auto.
This field allows the administrator to choose between an automatic or manual
failback policy on the IP Office. In deployments with Centralized phones, this field
must be set consistently with the Failback Policy of the phones, which is configured
via the Session Manager global settings in System Manager. The options are:
• Auto: IP Office automatically brings the SM Line to ‘In Service’ status as soon as
it detects via the Reactive Monitoring that the Session Manager is reachable
• Manual: When an SM line is in "Out of Service" state, IP Office does not bring it
back to "In Service" status based on automatic detection. IP Office keeps the SM
Line in "Out of Service" state until the administrator manually initiates Failback of
IP Office from Session Manager.
Related links
System | Telephony on page 241
SM (9.0) on page 262
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Configuration Mode Field Descriptions
SM (9.0)
This page is used to configure settings that apply to both SM lines.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
Short Form Dialing
Length
Default = 0. Range = 0 to 14.
Branch Prefix
Default = Blank. Maximum range = 15 digits.
This number specifies the short-form dialing length for all Centralized users and
Groups. Configuration of this field allows IP Office to treat the last N digits (where N
is the number entered in this field) of each Centralized user’s extension number as
an alias to that user’s extension number. For example, if a Centralized user’s
extension number is 5381111 and the Short Form Dialing Length is 4, the system
will match calls to 1111 with this extension. When 1111 is dialed by another user on
the system, entered from the autoattendant, or comes from the ICR, then in sunnyday that call will be sent to Session Manager with the number converted to 5381111
and in rainy-day it will target the extension 5381111 locally.
This number is used to identify the IP Office system within the Avaya Aura®
network. The branch prefix of each IP Office system must be unique and must not
overlap. For example 85, 861 and 862 are okay, but 86 and 861 overlap. On calls
routed via an SM Line, the branch prefix is added to the caller's extension number.
You have the option to leave the Branch Prefix field blank. If you do not configure
the branch prefix, the IP Officeuser extensions must be defined with the full
enterprise number.
Local Number
Length
Default = Blank (Off). Range = Blank (Off) or 3 to 9.
This field sets the default length for extension numbers for extensions, users,
andhunt groups added to the IP Office configuration. Entry of an extension number
of a different length will cause an error warning by Manager.
The number entered in the Branch Prefix field plus the number entered in the
Local Number Length field must not exceed 15 digits. You have the option to
leave the Local Number Length field blank.
Proactive
Monitoring
Default = 60 seconds. Range = 60 seconds to 100000 seconds.
Monitoring Retries
Default = 1. Range = 0 to 5.
The Enterprise Branch system sends regular SIP OPTIONS messages to the SM
line in order to check the status of line. This setting controls the frequency of the
messages when the SM line is currently in service. Centralized SIP phones use
their own settings.
The number of times the Enterprise Branch system retrys sending an OPTIONS
request to Session Manager before the SM Line is marked out-of-service.
Reactive Monitoring
Default 60 seconds. Range = 10 to 3600 seconds.
The Enterprise Branch system sends regular SIP OPTIONS messages to the SM
line in order to check the status of line. This setting controls the frequency of the
messages when the SM line is currently out of service. Centralized SIP phones use
their own settings.
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System
Field
Description
Failback Policy
Default = Auto.
This field allows the administrator to choose between an automatic or manual
failback policy on the IP Office. In deployments with Centralized phones, this field
must be set consistently with the Failback Policy of the phones, which is configured
via the Session Manager global settings in System Manager. The options are:
• Auto: IP Office automatically brings the SM Line to ‘In Service’ status as soon as
it detects via the Reactive Monitoring that the Session Manager is reachable
• Manual: When an SM line is in "Out of Service" state, IP Office does not bring it
back to "In Service" status based on automatic detection. IP Office keeps the SM
Line in "Out of Service" state until the administrator manually initiates Failback of
IP Office from Session Manager.
Media Security
Default = Disable.
Secure RTP (SRTP) can be used between IP devices to add additional security.
These setting control whether SRTP is used for this device and the setting used for
the SRTP. For further details of SRTP refer to Secure VoIP (SRTP). These settings
apply only to centralized phones. The options are:
• Disable: Media security is not required. All media sessions (audio, video, and
data) will be enforced to use RTP only.
• Enforce: Media security is required. All media sessions (audio, video, and data)
will be enforced to use SRTP only.
• Best Effort: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
Media Security
Options
Not displayed if Media Security is disabled. The options are:
• Encryptions: Default = RTP This setting allows selection of which parts of a
media session should be protected using encryption. The default is to encrypt just
the RTP stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication. The default
is authenticate just the RTCP stream (call control signals).
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80 and
SRTP_AES_CM_128_SHA1_32.
Related links
SM on page 260
Call Log
The system can store a centralized call log for users. Each users' centralized call log can contain up
to 30 call records for user calls. When this limit is reached, each new call records replaces the
oldest previous record.
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Configuration Mode Field Descriptions
On Avaya phones with a fixed Call Log or History button (1400, 1600, 9500 and 9600 Series), that
button can be used to display the user's centralized call log. The centralized call log is also used for
M-Series and T-Series phone. The user can use the call log to make calls or to store as a personal
speed dial. They can also edit the call log to remove records. The same call log is also used if the
user logs into one-X Portal for IP Office.
The centralized call log moves with the user if they log on and off from different phones. This
includes if they hot desk within a network.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
Default Centralized
Call Log On
Default = On.
Log Missed Calls
Answered at
Coverage
Default = Off.
Log Missed Hunt
Group Calls
When selected, each user is defaulted to have the system store a call log of their
calls. This call log is accessible on the phone when the user is using a phone with a
Call Log or History button. The use of centralized call logging can be enabled/
disabled on a per user basis using the Centralized Call Log user setting (User |
Telephony | Call Log).
This setting controls how calls to a user, that are answered by a covering user
should be logged in the centralized call log. This option applies for calls answered
elsewhere (covered) by pickup, call coverage (call coverage buttons or coverage
group), bridged appearance button, user BLF, voicemail, etc.
Setting
Targeted User
Covering User
Off
Nothing
Answered Call
On
Missed Call
Answered Call
Default = Off. By default, hunt group calls are not included in any user's centralized
call log unless answered by the user. If this option is selected, a separate call log is
kept for each hunt group of calls that are not answered by anyone. It includes hunt
group calls that go to voicemail.
If missed hunt group calls are also being logged, the system stores up to 10 call
records for each hunt group. When this limit is reached, new call records replace
the oldest record.
Within the user call log setting (User | Telephony | Call Log), the list of hunt groups
allows selection of which hunt groups' missed call records should be displayed as
part of the user's centralized call log.
Related links
System | Telephony on page 241
TUI
This page is used to configure system wide telephony user interface (TUI) options.
These settings are mergeable.
Default Phone Display Options:
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System
Use these settings to define the default phone display when feature menus are disabled. Note that
for new users, the default phone display options are set to the system default values.
Feature menus can be disabled in one of two ways.
On System | Telephony | TUI, set the Features Menu to off. On User | Telephony | TUI, set the
User Setting to Same as System.
On User | Telephony | TUI, set the User Setting to Custom and set the Features Menu to off.
Phone Type
Variable
Description
1400
Display Name Preference
Defines the default value of the User’s
Features > Phone User > Phone Screen
Settings > Display Name setting.
1600
Default = Off
When enabled, displays the user name.
9500
Column View Preference
9608
9611
Defines the default value of the User’s
Features > Phone User > Phone Screen
Settings > Display Mode setting.
Default = Dual
Column view can be Single or Dual.
9621
Quick Touch Panel Lines
9641
Defines the default value of the User’s
Features > Phone User > Phone Screen
Settings > Quick Touch Lines setting.
Default = Optimize
Sets the Quick Touch Panel number.
The options are 1, 2, and Optimize.
When set to Optimize:
• 9621 = 1
• 9641 = 2
Field
Description
Time Format
Default = Locale Defined.
Set the system time format display. The default time format is defined by the Locale
setting. You can override the default and set the time format to a 12- hour or 24hour clock.
Features Menu Controls
Features Menu
Default = On
When set to off, TUI feature menus are not available. When set to on, you can
select to turn individual feature menus off or on. The following feature menus are
listed:
Basic Call Functions (Transfer to Mobile, Pickup, Park)
Advanced Call Functions (Do Not Disturb, DNS Exceptions, Account Code)
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Configuration Mode Field Descriptions
Field
Description
Forwarding
Hot Desk Functions
Passcode Change
Phone Lock
Self Administration
Voicemail Controls
For information on telephony features, see the IP Office Product Description.
Related links
System | Telephony on page 241
System | Directory Services
Related links
System on page 214
System | Directory Services | LDAP on page 266
HTTP on page 269
System | Directory Services | LDAP
The system supports LDAP Version 2. LDAP (Lightweight Directory Access Protocol) is a software
protocol for enabling anyone to locate organizations, individuals, and other resources such as files
and devices in a network, whether on the Internet or on a corporate intranet. LDAP is a "lightweight"
(smaller amount of code) version of DAP (Directory Access Protocol), which is part of X.500, a
standard for directory services in a network. LDAP is lighter because in its initial version, it did not
include security features.
The system supports the import of directory records from one system to another using HTTP. That
includes using HTTP to import records that another system has learnt using LDAP. HTTP import,
which is simpler to configure, can be used to relay LDAP records with LDAP configured on just one
system.
LDAP records can contain several telephone numbers. Each will be treated as a separate directory
record when imported into the system directory.
In a network, a directory tells you where in the network something is located. On TCP/IP networks,
including the Internet, the Domain Name System (DNS) is the directory system used to relate the
domain name to a specific network address. However, you may not know the domain name. LDAP
allows you to search for an individual without knowing where they're located (although additional
information will help with the search).
An LDAP directory is organized in a simple "tree" hierarchy consisting of the following levels:
• The "root" directory (the starting place or the source of the tree), which branches out to
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System
• Countries, each of which branches out to
• Organizations, which branch out to
• Organizational units (divisions, departments, and so forth), which branches out to (includes an
entry for)
• Individuals (which includes people, files, and shared resources such as printers)
An LDAP directory can be distributed among many servers. Each server can have a replicated
version of the total directory that is synchronized periodically. An LDAP server is called a Directory
System Agent (DSA). An LDAP server that receives a request from a user takes responsibility for
the request, passing it to other DSA's as necessary, but ensuring a single coordinated response for
the user.
LDAP Directory Synchronization allows the telephone number Directory held in the Control Unit to
be synchronized with the information on an LDAP server. The feature can be configured to
interoperate with any server that supports LDAP Version 2.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
LDAP Enabled
Default = Off
This option turns LDAP support on or off. The system uses LDAP Version 2. If the
server being queried is an LDAP Version 3 server, support for LDAP Version 2
requests may need to be enabled on that server (all LDAP Version 3 servers
support LDAP Version 2 but do not necessarily have it enabled by default).
User Name
Default = Blank
Enter the user name to authenticate connection with the LDAP database. To
determine the domain-name of a particular Windows 2000 user look on the
"Account" tab of the user's properties under "Active Directory Users and
Computers". Note that this means that the user name required is not necessarily the
same as the name of the Active Directory record. There should be a built-in account
in Active Directory for anonymous Internet access, with prefix "IUSR_" and suffix
server_name (whatever was chosen at the Windows 2000 installation). Thus, for
example, the user name entered is this field might be:
IUSR_CORPSERV@example.com
Password
Default = Blank
Enter the password to be used to authenticate connection with the LDAP database.
Enter the password that has been configured under Active Directory for the above
user. Alternatively an Active Directory object may be made available for anonymous
read access. This is configured on the server as follows.
In "Active Directory Users and Computers" enable "Advanced Features" under the
"View" menu. Open the properties of the object to be published and select the
"Security" tab. Click "Add" and select "ANONYMOUS LOGON", click "Add", click
"OK", click "Advanced" and select "ANONYMOUS LOGON", click "View/Edit",
change "Apply onto" to "This object and all child objects", click "OK", "OK", "OK".
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Configuration Mode Field Descriptions
Field
Description
Once this has been done on the server, any record can be made in the User Name
field in the System configuration form (however this field cannot be left blank) and
the Password field left blank. Other non-Active Directory LDAP servers may allow
totally anonymous access, in which case neither User Name nor Password need be
configured.
Server IP Address
Default = Blank
Enter the IP address of the server storing the database.
Server Port
Default = 389
This setting is used to indicate the listening port on the LDAP server.
Authentication
Method
Default = Simple
Select the authentication method to be used. The options are:
• Simple: clear text authentication
• Kerberos: Kerberos 4 LDAP and Kerberos 4 DSA encrypted authentication (for
future use).
Resync Interval
(secs)
Default = 3600 seconds. Range = 1 to 99999 seconds.
The frequency at which the system should resynchronize the directory with the
server. This value also affects some aspects of the internal operation.
The LDAP search inquiry contains a field specifying a time limit for the search
operation and this is set to 1/16th of the resync interval. So by default a server
should terminate a search request if it has not completed within 225 seconds
(3600/16).
The client end will terminate the LDAP operation if the TCP connection has been up
for more than 1/8th of the resync interval (default 450 seconds). This time is also
the interval at which a change in state of the "LDAP Enabled" configuration item is
checked.
Search Base/Search
Filter
Default = Blank These 2 fields are used together to refine the extraction of directory
records. Basically the Base specifies the point in the tree to start searching and the
Filter specifies which objects under the base are of interest. The search base is a
distinguished name in string form (as defined in RFC1779).
The Filter deals with the attributes of the objects found under the Base and has its
format defined in RFC2254 (except that extensible matching is not supported). If the
Search Filter field is left blank the filter defaults to "(objectClass=*)", this will match
all objects under the Search Base. The following are some examples applicable to
an Active Directory database.
• To get all the user phone numbers in a domain:
Search Base: cn=users,dc=acme,dc=com
Search Filter: (telephonenumber=*)
• To restrict the search to a particular Organizational Unit (eg office) and get cell
phone numbers also:
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System
Field
Description
Search Base: ou=holmdel,ou=nj,DC=acme,DC=com
Search Filter: (|(telephonenumber=*)(mobile=*))
• To get the members of distribution list "group1":
Search Base: cn=users,dc=acme,dc=com
Search Filter:
(&(memberof=cn=group1,cn=users,dc=acme,dc=com)
(telephonenumber=*))
Number Attributes
: Default = see below
Enter the number attributes the server should return for each record that matches
the Search Base and Search Filter. Other records could be ipPhone, otherIpPhone,
facsimileTelephoneNumber, otherfacsimileTelephone Number, pager or
otherPager. The attribute names are not case sensitive. Other LDAP servers may
use different attributes.
By default the record is
"telephoneNumber,otherTelephone,homePhone=H,otherHomePhone=H,mobile=M,
otherMobile=M", as used by Windows 2000 Server Active Directory for Contacts.
The optional "=string" sub-fields define how that type of number is tagged in the
directory. Thus, for example, a cell phone number would appear in the directory as:
John Birbeck M 7325551234
Related links
System | Directory Services on page 266
HTTP
Navigation: System | Directory Services | HTTP
Additional configuration information
For additional configuration information, see Centralized System Directory on page 570.
Configuration settings
The system can use HTTP to import the directory records held by another system. Note that support
for HTTP can be disabled. The setting System | System | Avaya HTTP Clients Only can restrict a
system from responding to HTTP requests. The system's Unsecured Interface security settings
also included controls for HTTP access (HTTP Directory Read and HTTP Directory Write).
For Server Edition, on Secondary Server, Expansion System (L) and Expansion System (V2)
systems, the HTTP settings are automatically defaulted to obtain the system directory from the
Primary Server.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
Directory Type
Default = None (No HTTP import)/IP Office SCN on Server Edition.
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Configuration Mode Field Descriptions
Field
Description
Set whether HTTP import should be used and the method of importation. The options
are:
• None: Do not use HTTP import.
• IP Office: Import from the system at the IP address set in the Source field.
• IP Office SCN: Import from a system in a multi-site network. The Source field is used
to select the Outgoing Line ID that matches the H.323 line to the remote system.
Source
Default = Blank/9999 on Server Edition.
The form of this field changes according to the Directory Type selection above. For IP
Office this field requires the IP address of the other system. For IP Office SCN, the
outgoing group ID of the IP Office line to the remote system is used.
List
Default = All.
This field sets what types of directory record should be imported. The options are:
• All: Import the full set of directory records from the remote system.
• Config Only: Import just directory records that are part of the remote system's
configuration. Note that these will be treated as imported records and will not be added
to the local systems own configuration records.
• LDAP Only: Import just directory records that the remote system has obtained as the
result of its own LDAP import. This allows LDAP directory records to be relayed from
one system to another.
• HTTP Only: Import just directory records that the remote system has obtained as the
result of its own HTTP import. This allows HTTP directory records to be relayed from
one system to another.
URI
Default = /system/dir/complete_dir_list?sdial=true
This field is for information only and cannot be adjusted. The path shown changes to
match the List setting above.
Resync Interval
(secs)
Default = 3600 seconds.
HTTPS Enabled
Default = On.
Set how often the system should request an updated import. When a new import is
received, all previously imported records are discarded and the newly imported records
are processed.
Turns HTTPS support on or off for directory record import.
Port Number
Default = 443.
The port used for the Directory import.
When HTTPS Enabled is set to On, the default value is 443. When HTTPS Enabled is
set to Off, the default value is 80.
Related links
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System
System | System Events
The system supports a number of methods by which events occurring on the system can be
reported. These are in addition to the real-time and historical reports available through the System
Status Application (SSA).
Related links
System on page 214
System Events | Configuration on page 271
System Events | Alarms on page 272
System Events | Configuration
This form is used for general configuration related to system alarms. For email alarms the SMTP tab
is also used.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
SNMP Agent Configuration
SNMP Enabled
Default = Off.
Enables support for SNMP. This option is not required if using SMTP or Syslog.
Community
(Read-only)
Default = Blank.
SNMP Port
Default = 161. Range = 161, or 1024 to 65534. The port on which the system listens for
SNMP polling.
Device ID
This is a text field used to add additional information to alarms. If an SSL VPN is
configured, Avaya recommends that the Device ID match an SSL VPN service Account
Name. Each SSL VPN service account name has an associated SSL VPN tunnel IP
address. Having the displayed Device ID match an SSL VPN service account name helps
identify a particular SSL VPN tunnel IP address to use for remotely managing IP Office.
Contact
This is a text field used to add additional information to alarms.
Location
This is a text field used to add additional information to alarms.
The SNMP community name to which the system belongs.
QoS Parameters
These parameters are used if Enable RTCP Monitor on Port 5005 is selected (Systems | LAN1 | VoIP). They
are used as alarm thresholds for the QoS data collected by the system for calls made by Avaya H.323 phones
and for phones using VCM channels. If a monitored call exceeds any of the threshold an alarm is sent to the
System Status application. Quality of Service alarms can also be sent from the system using Alarms.
• The alarm occurs at the end of a call. If a call is held or parked and then retrieved, an alarm can occur for
each segment of the call that exceeded a threshold.
• Where a call is between two extensions on the system, it is possible that both extensions will generate an
alarm for the call.
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Configuration Mode Field Descriptions
Field
Description
• An alarm will not be triggered for the QoS parameters recorded during the first 5 seconds of a call.
Round Trip Delay
(msec)
Default = 350.
Jitter (msec)
Default =20.
Less than 160ms is high quality. Less than 350ms is good quality. Any higher delay will be
noticeable by those involved in the call. Note that, depending on the compression codec
being used, some delay stems from the signal processing and cannot be removed: G.711
= 40ms, G.723a = 160ms, G.729 = 80ms.
Jitter is a measure of the variance in the time for different voice packets in the same call to
reach the destination. Excessive jitter will become audible as echo.
Packet Loss (%)
Default = 3.0.
Excessive packet loss will be audible as clipped words and may also cause call setup
delays.
Good Quality
High Quality
Round Trip Delay
< 350ms
< 160ms
Jitter
< 20ms
< 20ms
Packet Loss
< 3%
< 1%
Related links
System | System Events on page 271
System Events | Alarms
This form is used to configure what can cause alarms to be sent using the different alarm methods.
• Up to 5 alarm traps can be configured for use with the SNMP settings on the System | System
Events | Configuration tab.
• Up to 3 email alarms can be configured for sending using the systems System | SMTP
settings. The email destination is set as part of the alarm configuration below.
• Up to 2 alarms can be configured for sending to a Syslog destination that is included in the
alarm settings.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
New Alarm
This area is used to show and edit the alarm.
Destination
To use SNMP or Email the appropriate settings must be configured on the Configuration sub-tab. Note that the
Destination type is grayed out if the maximum number of configurable alarms destinations of that type has been
reached. Up to 5 alarm destinations can be configured for SNMP, 3 for SMTP email, and 2 for Syslog
Trap
If selected, the details required in addition to the selected Events are:
• Server Address: Default = Blank. The IP address or fully qualified domain name
(FQDN) of the SNMP server to which trap information is sent.
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System
Field
Description
• Port: Default = 162. Range = 0 to 65535. The SNMP transmit port.
• Community: Default = Blank The SNMP community for the transmitted traps. Must be
matched by the receiving SNMP server.
• Format: Default = IP Office. The options are:
- IP Office SNMP event alarms format in accordance with IP Office.
- SMGR SNMP event alarms format in accordance with SMGR.
Syslog
If selected, the details required in addition to the selected Events are:
• IP Address: Default = Blank. The IP address of the Syslog server to which trap
information is sent.
• Port: Default = 514. Range = 0 to 65535. The Syslog destination port.
• Protocol: Default = UDP. Select UDP or TCP.
• Format: Default = Enterprise. The options are:
- Enterprise Syslog event alarms format in accordance with Enterprise.
- IP Office Syslog event alarms format in accordance with IP Office.
Email
If selected, the details required in addition to the selected Events are:
Email: The destination email address.
Minimum Security Default = Warnings.
Level
The options are:
• Warnings: All events, from Warnings to Critical, are sent.
• Minor: Minor, major, and critical events are sent. Warnings are not sent.
• Major: Major and critical events are sent. Warnings and minor events will not be sent.
• Critical: Only critical events are sent.
•
Events
Default = None
Sets which types of system events should be collected and sent. The table below lists the
alarms associated with each type of event. Text in italics in the messages is replaced with
the appropriate data. Items in [] brackets are included in the message if appropriate. The
subject line of SMTP email alarms takes the form "System name: IP address - System
Alarm".
Alarm Types
Note the following.
• Voicemail Pro Storage Alarms: The alarm threshold is adjustable through the Voicemail Pro
client.
• Embedded Voicemail Storage Alarms: A disk full alarm is generated when the Embedded
Voicemail memory card reaches 90% full. In addition a critical space alarm is generated at 99%
full and an OK alarm is generated when the disk space returns to below 90% full.
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Configuration Mode Field Descriptions
• Loopback: This type of alarm is only available for systems with a United States locale.
Type
Events
Event State
Message
Entity
Application
Voicemail operation
The Voicemail server is now
operational.
Voicemail Failure
The Voicemail server is down.
Voicemail Event - storage OK
The Voicemail server storage is OK.
Voicemail Event - storage nearly full
The Voicemail server storage is
nearly full.
Voicemail Event - storage full
The Voicemail server storage is full.
All licenses in use
The following licenses are all in use.
License Type: <name>.
All resources in use
The following system resources are
all in use: <resource type> will be
provided.
Authentication failure
Avaya Contact Center Select data
synchonization failed: authentication
failure.
Clock source changed
8kHz clock source changed. Details
will be provided.
CPU warning/critical
• Warning alarm: CPU utilization
near capacity.
Service
• Critical alarm: CPU exhausted.
Feature license missing
Attempt to use a feature for which no
license is installed. License Type:
<name>.
Hold music file failure
Failed to load Hold Music source file.
Log stamped
Log Stamp #nnn created in Monitor
logs.
Logon failed
Logon failure reason will be
provided.
Memory use warning/critical
• Warning alarm: Memory utilization
near capacity.
• Critical alarm: Memory exhausted.
Network interconnect failure
Details of the network
interconnection failure will be
provided.
No free channels available
No free channels were available.
Outgoing group ID: <number>.
OEM card slot error
System running secondary software
or error description with OEM card
will be provided.
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System
Type
Events
Event State
Message
SIP message too large
SIP message Rx error - too large ignored.
SIP Registration Expiry
• Avaya IP Office Contact Contact
Center SIP registration expired
• Avaya Contact Center Select SIP
registration expired.
Sync Request Timeout
Avaya Contact Center Select data
synchonization failed: no response
from CCMA server
Memory Card
Change
The PC card in name has changed.
Expansion
Module
Operational
Expansion module name link is up.
Failure
Expansion module name link is
down.
Error
Expansion module name link has a
link error.
Change
Expansion module name link has
changed.
Operational
Trunk number (name) [on expansion
module number] is now operational.
Failure
Trunk number (name) [on expansion
module number] is down.
Trunk seize failure
Seize failure: Channel [number] or
Port [number].
Incoming call outgoing trunk failure
Incoming call outgoing trunk:
Channel [number] or Port [number].
CLI not delivered
CLI not delivered: Channel [number]
or Port [number].
DDI incomplete
DDI incomplete. Expected Number of
digits: <number>.
LOS
LOS
OOS
OOS
Red Alarm
Red Alarm
Blue Alarm
Blue Alarm
Yellow Alarm
Yellow Alarm
IP connection failure
IP connection failure. IP Trunk Line
Number: <number> or Remote end
IP address: <IP address>.
Small Community Network invalid
connection
Small Community Network invalid
connection. IP trunk line number:
Trunk
Trunk
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Configuration Mode Field Descriptions
Type
Events
Event State
Message
<number> or remote end IP address:
<IP address>.
Link
Device changed
Device changed. Home Extension
Number: <number>.
LDAP server communication failure
LDAP server communication failure
Resource down
Link/resource down. Module type,
number and name
will be provided.
VCM
Memory Card
SMTP server communication failure
SMTP server communication failure
Voicemail Pro connection failure
Voicemail Pro connection failure
Operational
VCM module name is now
operational.
Failure
VCM module name has failed.
Non-primary location boot alarm
System running backup software.
Invalid SD Card
Incompatible or Invalid (System or
Optional) SD Card fitted.
Network link failure
Network Interface name (ip address)
has been disconnected.
Network link operational
Network Interface name (ip address)
has been connected.
System warm start
System has been restarted (warm
start).
System cold start
System has restarted from power fail
(cold start).
SNMP Invalid community
Invalid community specified in SNMP
request.
Server operational
The license server is now
operational.
Server failure
The license server is no longer
operational.
Invalid Card
Free Capacity
Generic
License
Loopback
Generic
License
Server
License Key
Failure
License Key Failure
Loopback
Near end line loopback
Trunk number (name) [on expansion
module number] is in near end
loopback.
Near end payload loopback
Trunk number (name) [on expansion
module number] is in near end
loopback with payload.
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System
Type
Phone
Change
Events
Phone
Change
Event State
Message
Loopback off
Trunk number (name) [on expansion
module number] has no loopback.
Phone has been unplugged
The phone with id n has been
removed from extension extension
(unit, port number).
Phone has been plugged in
The phone with type type (id
number) has been plugged in for
extension extension (unit, port
number).
Quality of
Service
QoS
Monitoring
If Enable RTCP Monitor on Port 5005 is selected, any monitored calls that
exceeds the set QoS Parameters will cause an alarm.
Syslog
Basic Audit
Events as written to the system Audit Trail. Available on Syslog output only.
System
Configuration
CCR group agent not targeted
CCR Group agent not targeted as it
is not an CCR Agent. Group :
<name> Agents: <name1, ..., name
n>.
Small Community Network dial plan
conflict
Small Community Network dial plan
conflict
No incoming call route for call
The following line had no Incoming
Call Route for a call. Line: <number>
or Line Group ID: <number>.
Installed hardware failure
Installed hardware failure details will
be provided.
Emergency call successful
Successful Emergency Call |
Emergency call! Location:location
Dialled:dialled number
Called:number sent on the line
CallerID:ID Usr:user Extn:extension
Emergency call failure
Failed Emergency Call | Emergency
call! Location:location Dialled:dialled
number FailCause:cause Usr:user
Extn:extension
System
Shutdown
Running
Backup
Emergency
Calls
Related links
System | System Events on page 271
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Configuration Mode Field Descriptions
System | SMTP
SMTP can be used as the method of sending system alarms. The email destination is set as part of
the email alarms configured in System | System Events | Alarms.
SMTP can be used with Embedded Voicemail for Voicemail Email. The voicemail destination is set
by the user's Voicemail Email address.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Server Address
Default = 0.0.0.0
This field sets the IP address of the SMTP server being used to forward SNMP
alarms sent by email.
Port
Default = 25. Range = 0 to 65534.
This field set the destination port on the SMTP server.
Email From Address
Default = Blank
This field set the sender address to be used with mailed alarms. Depending of
the authentication requirements of the SMTP server this may need to be a valid
email address hosted by that server. Otherwise the SMTP email server may
need to be configured to support SMTP relay.
Use STARTTLS
Default = Off. (Release 9.0.3).
Select this field to enable TLS/SSL encryption. Encryption allows voicemail-toemail integration with hosted email providers that only permit SMTP over a
secure transport.
Server Requires
Authentication
Default = On
User Name
Default = Blank This field sets the user name to be used for SMTP server
authentication.
Password
Default = Blank This field sets the password to be used for SMTP server
authentication.
Use Challenge
Response
Authentication (CRAMMD5)
Default = Off. This field should be selected if the SMTP uses CRAM-MD5.
This field should be selected if the SMTP server being used requires
authentication to allow the sending of emails. When selected, the User Name
and Password fields become available
Related links
System on page 214
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System
System | SMDR
Using a specified IP address, the system can send a call record for each completed call.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Note:
Outbound Contact Express does not generate SMDR records.
Field
Description
Output
Default = No Output.
Select the type of call record that the system should output via IP. The options
are:
• No Output
• SMDR Only : Send call records using the SMDR settings below.
SMDR: Station Message Detail Recorder Communications
This fields are available when SMDR is selected as the output. For information on SMDR record details, see
the appendix.
IP Address
Default = 0.0.0.0 (Listen).
The destination IP address for SMDR records. The address 0.0.0.0 puts the
control unit in listen mode on the specified TCP port. When a connection is
made on that port, all SMDR records in the buffer are provided.
TCP Port
Default = 0.
The destination IP port for SMDR records.
Records to Buffer
Default = 500. Range = 10 to 3000.
The system can cache up to 3000 SMDR records if it detects a communications
failure with destination address. If the cache is full, the system will begin
discarding the oldest records for each new record.
Call Splitting for Diverts
Default = Off.
When enabled, for calls forwarded off-switch using an external trunk, the SMDR
produces separate initial call and forwarded call records. This applies for calls
forwarded by forward unconditional, forward on no answer, forward on busy,
DND or mobile twinning. It also applies to calls forwarded off-switch by an
incoming call route. The two sets of records will have the same Call ID. The call
time fields of the forward call record are reset from the moment of forwarding on
the external trunk.
Related links
System on page 214
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Configuration Mode Field Descriptions
System | Twinning
These settings are used with Mobile Twinning, see the User | Mobility tab for further details.
On mobile twinned calls, if the original party information is used or a specific calling party
information CLI is set, that number overrides setting the outgoing CLI using short codes.
Warning:
Outgoing CLI
Changing the outgoing CLI for calls requires the line provider to support that function. You must
consult with your line provider before attempting to change the outgoing CLI, failure to do so may
result in loss of service. If changing the outgoing CLI is allowed, most line providers required that the
outgoing CLI used matches a number valid for return calls on the same trunks. Use of any other
number may cause calls to be dropped or the outgoing CLI to be replaced with a valid number.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
Send Original Party
Information for Mobile
Twinning
Default = Off
When on, the system will attempt to send the ICLID information provided with the
incoming call to the twinning destination.
• The SIP line Send Caller ID setting takes priority.
• The values on the System | Twinning tab override the SIP lines Send Caller ID
setting.
Calling Party Information
for Mobile Twinning
Default = Blank (Disabled). Range = Up to 32 digits.
This field is usable when Send Original Party Information for Mobile Twinning
is off. Note that the number entered here for use as the CLI must be a valid
number for return calls to the same site. Some line providers may reject calls that
use a number that is not valid for return calls to the same site. In addition
depending on the line type and line provider settings the maximum number of
digits may be limited.
Related links
System on page 214
System | VCM
This form allows adjustment of the operation of any Voice Compression Modules (VCM's) installed
in a control unit.
Calls to and from IP devices can require conversion to the audio codec format being used by the IP
device. For systems this conversion is done by voice compression channels. These support the
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System
common IP audio codecs G.711, G.723 and G.729a. For details of how to add voice compression
resources to a system, refer to the IP Office Installation Manual.
These settings should only be adjusted under the guidance of Avaya support.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
When are Voice Compression Channels Used
IP Device to Non-IP Device: These calls require a voice compression channel for the duration of
the call. If no channel is available, busy indication is returned to the caller.
IP Device to IP Device: Call progress tones (for example dial tone, secondary dial tone, etc) do not
require voice compression channels with the following exceptions:
• Short code confirmation, ARS camp on and account code entry tones require a voice
compression channel.
• Devices using G.723 require a voice compression channel for all tones except call waiting.
When a call is connected:
• If the IP devices use the same audio codec no voice compression channel is used.
• If the devices use differing audio codecs, a voice compression channel is required for each.
Non-IP Device to Non-IP Device: No voice compression channels are required.
Music on Hold: This is provided from the system's TDM bus and therefore requires a voice
compression channel when played to an IP device.
Conference Resources and IP Devices: Conferencing resources are managed by the conference
chip which is on the system's TDM bus. Therefore, a voice compression channel is required for
each IP device involved in a conference. This includes services that use conference resources such
as call listen, intrusion and silent monitoring. They also apply to call recording.
Page Calls to IP Device: Page calls require 1 voice compression channel per audio codec being
used by any IP devices involved. The system only uses G.729a for page calls, therefore only
requiring one channel but also only supporting pages to G.729a capable devices.
Voicemail Services and IP Devices: Calls to the system voicemail servers are treated as data
calls from the TDM bus. Therefore calls from an IP device to voicemail require a voice compression
channel.
Fax Calls: These are voice calls but with a slightly wider frequency range than spoken voice calls.
The system only supports fax across IP between systems with the Fax Transport option selected.
SIP Calls:
• SIP Line Call to/from Non-IP Devices: Voice compression channel required.
• Outgoing SIP Line Call from IP Device: No voice compression channel required.
• Incoming SIP Line Call to IP Device: Voice compression channel reserved until call
connected.
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Configuration Mode Field Descriptions
T38 Fax Calls: The system supports T38 fax on SIP trunks and SIP extensions. Each T38 fax call
uses a VCM channel.
• Within a multi-site network, an T38 fax call can be converted to a call across across an H.323
line between systems using the Fax Transport Support protocol. This conversion uses 2 VCM
channels.
• In order use T38 Fax connection, the Equipment Classification of an analog extension
connected to a fax machine can be set Fax Machine. Additionally, the short code feature Dial
Fax is available.
Note:
T3 IP devices must be configured to 20ms packet size for the above conditions to apply. If left
configured for 10ms packet size, a voice compression channel is needed for all tones and for
non-direct media calls.
Measuring Channel Usability
The System Status Application can be used to display voice compression channel usage. Within the
Resources section it displays the number of channel in use. It also displays how often there have
been insufficient channels available and the last time such an event occurred.
Field
Description
Echo Return Loss (dB)
Default = 6dB. IP500 VCM, IP500 VCM V2 and IP500 Combination Cards. This
control allows adjustment of expected echo loss that should be used for the echo
cancellation process.
Echoes are typically generated by impedance mismatches when a signal is
converted from one circuit type to another, most notably from analog to IP. To
resolve this issue, an estimated echo signal can be created from one output and
then subtracted from the input to hopefully remove any echo of the output.
The options are:
• 0dB
• 3dB
• 6dB
• 9dB
Nonlinear Processor
Mode
Default = Adaptive. I
A low level of comfort noise is required on digital lines during periods where there
would normally be just silence. This is necessary to reassure users that the call is
still connected. These controls allow adjustment of the comfort noise generated
by the nonlinear processor (NLP) component of the VCM. The options are:
• Adaptive: Adaptive means the comfort noise generated by the NLP will try to
match background noise.
• Silence: Silence means the NLP will not generate comfort noise at all
• Disabled: Nonlinear processing is not applied, in which case some residual
echo may be heard.
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System
Field
Description
NLP Comfort Noise
Attenuation
Default = -9dB.
The options are:
• -3dB
• -6dB
• -9dB
NLP Comfort Noise
Ceiling
Default =-30dB.
The options are:
• -30dB
• -55dB
Modem
For Fax relay, these settings allow adjustment of the TDM side operation applied to fax calls using VCM
channels.
Tx Level (dB)
Default = -9dB. Range = 0 to -13dB.
CD Threshold
Default = -43dB, Options = -26dB, -31dB or -43dB.
No Activity Timeout
(secs)
Default = 30 seconds. Range = 10 to 600 seconds.
Related links
System on page 214
System | CCR
Customer Call Reporter (CCR) is an application that collects and displays information on the current
status of hunt groups and users that have been configured for Customer Call Reporter operation.
CCR is not supported by Server Edition.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
Busy Not Available Reason Codes
Agents who indicate that they are in a 'busy not available' state can be prompt to also indicate the reason for
being in that state. This menu allows descriptions for the possible reasons to be entered. The descriptions
are then used in menus from which the Agent's make selections when setting themselves into busy not
available state and in reports on Agent status.
Code/Reason
Rows 1 to 8 can be used to contain descriptions of up to 31 characters each.
Rows 0 and 9 are fixed as Unsupported and Busy Not Available.
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Configuration Mode Field Descriptions
Field
Description
For Customer Call Reporter 6.1, the reason codes are used to categorize calls in
the Agent Time Card report. Reason 1 is used to define lunch. All other reason
codes are reported as breaks.
Default After Call Work
Time (seconds)
Default = 10. Range = 10 to 999 seconds.
If an agent goes into the After Call Work (ACW) state, either automatically or
manually, this field sets the duration of that state after which it is automatically
cleared. This duration can be overridden by the Agent's own setting (User |
Telephony | Supervisor Settings | After Call Work Time). During ACW state, hunt
group calls are not presented to the user.
Related links
System on page 214
System | Codecs
This tab is used to set the codecs available for use with all IP (H.323 and SIP) lines and extensions
and the default order of codec preference.
• Avaya H.323 telephones do not support G.723 and will ignore it if selected.
• For systems with H.323 lines and extensions, one of the G.711 codecs must be selected and
used.
• G.723 is not supported by Linux based systems.
• The number of channels provided by an IP500 VCM 32 or IP500 VCM 64 card, up to a
maximum of 32 or 64 respectively, depends on the actual codecs being used. This also applies
to IP500 VCM 32 V2 and IP500 VCM 64 V2 cards. The following table assumes that all calls
using the VCM use the same codec.
Codec
IP500 VCM 32 IP500 VCM 32
V2
IP500 VCM 64 IP500 VCM 64
V2
G.711
32
64
G.729a
30
60
G.723
22
44
G.722
30
60
When paging, always use only one codec (the preferred). It is the system administrator's
responsibility to ensure all the phones in the paging group support the codec.
Usability
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Field
Description
RFC2833 Default
Payload
Default = 101. Range = 96 - 127.
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System
Field
Description
This field specifies the default value for RFC2833 dynamic payload negotiation.
Service providers that do not support dynamic payload negotiation may require a
fixed value.
Available Codecs
This list shows the codecs supported by the system and those selected as usable.
Those codecs selected in this list are then available for use in other codec lists
shown in the configuration settings. For example the adjacent Default Selection list
and the individual custom selection list on IP lines and extensions.
Warning:
Removing a codec from this list automatically removes it from the codec lists of
any individual lines and extensions that are using it.
The available codecs in default preference order are: G.711 A-Law, G.711 U-Law,
G.729 and G.723.1. Note that the default order for G.711 codecs will vary to match
the system's default companding setting. G.723.1 is not supported on Linux based
systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition it is supported on Primary
Server, Secondary Server and Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
Default Codec
Selection
By default, all IP (H.323 and SIP) lines and extensions added to the system have
their Codec Selection setting set to System Default. That setting matches the
codec selections made in this list. The buttons between the two lists can be used to
move codecs between the Unused and the Selected parts of the list and to change
the order of the codecs in the selected codecs list.
Related links
System on page 214
System | VoIP Security
Use this tab to set system level media security settings. These settings apply to all lines and
extensions on which SRTP is supported and which have their Media Security settings configured to
be Same as System. Individual lines and extensions have media security settings that can override
system level settings.
Simultaneous SIP extensions that do not have physical extensions in the configuration use the
system security settings.
SM lines and all centralized user extensions must have uniform media security settings.
Field
Description
Media Security
Default = Disabled.
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Configuration Mode Field Descriptions
Field
Description
Secure RTP (SRTP) can be used between IP devices to add additional security. These
settings control whether SRTP is used for this system and the settings used for the
SRTP. The options are:
• Disabled: Media security is not required. All media sessions (audio, video, and data)
will be enforced to use RTP only.
• Enforce: Media security is required. All media sessions (audio, video, and data) will be
enforced to use SRTP only.
Warning:
Selecting Enforce on a line or extension that does not support media security will
result in media setup failures.
• Best Effort: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
Media Security
Options
Not displayed if Media Security is set to Disabled. The options are:
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which parts
of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option to
select SRTP_AES_CM_128_SHA1_32.
Strict SIPS
(Enterprise Branch deployments) Default = Off.
This option provides a system-wide configuration for call restrictions based on SIPS URI.
When this option is off, calls are not rejected due to SIPS. A call is sent according to the
configuration of the outgoing trunk or line that it is routed to, regardless of the way the
call came in, even if the call came in as a SIP invite with SIPS URI and is being sent with
a SIP URI onto a non-secure SIP trunk.
When this option is on, an incoming SIP invite with SIPS URI if targeted to a SIP trunk
(SM line or SIP line) is rejected if the target trunk is not configured with SIPS in the URI
Type field.
Related links
System on page 214
Dialer
Use this tab to configure the functions required for an Outbound Contact Express deployment.
These settings are mergeable. However, changes to the Operation field or to the Trunk Range / IP
Office table require a reboot.
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System
It is recommended that you do not change the mergeable settings while the system is in use.
Field
Description
Operation
Default = Off.
On the primary IP Office Server Edition server, set this field to Primary. For all other IP
Office servers, set this field to Child. When set to Off or Child, no other fields are
displayed.
Record Mode
Default = Off
Defines the automatic call recording function on VMPro. The options are:
• Whole Call: The entire call is recorded.
• Agent Connected: Recording starts once the conversation begins.
• Off
Record Controls
: Default = Full Defines what functions an agent can perform from WebAgent or from the
handset. The options are:
• Full
• Pause
• Off
Record Mode
and Record
Controls
Record Mode and Record Controls are related. The combined configuration settings are
listed below.
Note that stopping and starting the recording creates multiple recording files. Pausing and
resuming the recording keeps the recording in a single file.
Record Mode
Record Controls
Result
Off
Off
Calls are not recorded.
Agent Connected
Off
All calls are always recorded
from the time the agent joins
the call.
Agent Connected
Pause
All calls are always recorded
but the Agent can pause and
resume recording.
Agent Connected
Full
All calls are always recorded
from the time the Agent joins
the call. Agent has full
control on when calls get
recorded.
Whole Call
Off
All calls are always recorded
from the time the customer
answers.
Whole Call
Pause
All calls are always recorded
from the time the customer
answers but the Agent can
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Configuration Mode Field Descriptions
Field
Description
pause and resume the
recording.
Whole Call
Full
Call recording starts before
the agent is connected. All
calls are always recorded
but the Agent can pause and
resume the recording
Agent Call Back
Time
Default = 60. Range = 30 - 300.
Remote Agent
Display Text
Default = Blank. Maximum length = 33.
Remote Agent
Confirmation
Voice Prompt
Default = Blank. Maximum length = 31.
Remote Agent
First Extension
Number
Default = 0. The first extension number allocated to a remote agent. It must not conflict with
the existing dialing plan. If the range contains existing user extensions, they are used when
assigning extensions to remote users.
Remote Agent
Number of
Extensions
Default = 0. Maximum = 500.
The number of seconds an agent has to make a manual call after a customer hang up.
Used when a customer wants to be called on a different number.
Specify the text string displayed on the remote agent extension if that extension supports
displays and the protocol allows it to be transmitted.
Specify the Call Flow Entry point name used to play a greeting to the remote agent when
they log in. The actual Entry Point is added as a Modules Entry point using the VMPro
Client. The entry point cannot be added as a short code, user or group entry point.
The range of extensions starting from the one above. A user is created for every extension.
If the field is edited and the number of extensions is reduced, the number of remote agents
that can log in is reduced to the new setting. However, reducing the range does not
automatically delete previously created users. Users can only be deleted manually.
Use Custom Hold Default = unchecked. Defines system behavior when a call is placed on hold. When
Treatment
unchecked the the system Hold Music setting is used for the system's music on hold
source. When checked, the music on hold source is VMPro.
Record while on
Hold
Default = unchecked. When the Use Custom Hold Treatment box is checked, the Record
while on Hold setting can be enabled. When unchecked, recording is paused when the
call is on hold. When checked, recording continues when the call is on hold.
Trunk Range / IP
Office
The number of trunks used by Outbound Contact Express. The default entry is Trunk
Range: 1-250 for the Primary (Local) server. 250 is the maximum number of trunks
configured on a single server. Use this table to define the number of trunks managed by
the Primary and Secondary systems. The trunk range must match the line numbers used
by the Proactive Contact Dialer. Enter only one range per server.
Related links
System on page 214
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Line
System | Contact Center
The Contact Center tab contains the user information required by IP Office to synchronize account
information with an an Avaya Contact Center Select (ACCS) system. The information is
synchronized using the Contact Center Management Application (CCMA). These settings are only
used for the deployment of an ACCS system.
This tab is visible on the Server Edition Primary Server and Standard Mode IP 500 V2 systems.
These settings are mergeable.
Field
Description
Contact Center
Application
Default = None.
The options are
• Avaya Contact Center Select
• Avaya IP Office Contact Center
Synchronize to
this System
Default = Off.
CCMA Address
Default = Blank
When set to On, the CCMA fields below are enabled.
Address of the Contact Center Management Application system.
CCMA Username
Default = Blank
User name on the Contact Center Management Application system.
CCMA Password
Default = Blank
Password on the Contact Center Management Application system.
Related links
System on page 214
Line
The line settings shown in the system configuration will change according to the types of trunk cards
installed in the control unit or added using external expansion modules.
Warning:
Changing Trunk Cards Changing the trunk card installed in an control unit will result in line
settings for both the previous trunk card and the currently installed trunk card. In order to
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Configuration Mode Field Descriptions
change the trunk card type in a particular card slot, the configuration must be defaulted. This
does not apply if replacing an existing card with one of a higher capacity or fitting a trunk card
into a previously unused slot.
Trunk Incoming Call Routing
Each trunk type can be categorized as either an external trunk or internal trunk. The trunk type
affects how the system routes calls received on that trunk and the routing of calls to the trunk.
Trunk Types
External Trunks
Internal Trunks
Analog trunks
QSIG (T1, E1 or H.323)
T1 Robbed Bit
BRI So
E1R2
H.323
ISDN BRI (excluding So)
SCN
ISDN PRI T1
SES
ISDN PRI E1
SM Line
SIP
Incoming Calls Routed by
All incoming calls are routed by
comparison of call details for
matches within the system
Incoming Call Routes.
Incoming calls are routed by
looking for a match to the
incoming digits in the following
order:
Line short codes are not used.
• Extension number.
• Trunk short codes (excluding ?
short code).
• System short codes (excluding ?
short code).
• Trunk ? short code.
• System ? short code.
Line Groups
Each system trunk (or in some cases individual trunk channels) can be configured with an Incoming
Group ID and an Outgoing Group ID. These group IDs are used as follows:
•
•
Incoming Call Routes For incoming calls on external trunks, the Incoming Group ID of the
trunk is one of the factors used to match the call to one of the configured incoming call routes.
Short Codes - Routing Outgoing Calls For dialing which matches a short code set to a Dial
feature, the short codes Line Group ID can indicate either an ARS form or to use a trunk from
set to the same Outgoing Group ID. If the call is routed to an ARS form, the short codes in the
ARS form will specify the trunks to use by matching Outgoing Group ID.
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Line
Removing Unused Trunks
In cases where a trunk card is installed but the trunk is not physically connected, it is important to
ensure that the trunk is disabled in the configuration. This can be done on most trunks using by
setting the line's Admin setting to Out of Service.
This is especially important with analog trunks. Failure to do this may cause the system to attempt to
present outgoing calls to that trunk. Similarly, where the number of channels subscribed is less than
those supportable by the trunk type, the unsubscribed channels should be disabled.
Clock Quality
Calls between systems using digital trunks (for example E1, E1R2, T1 PRI and BRI) require an
common clock signal. The system will try to obtain this clock signal from an exchange through one
of its digital trunks. This is done by setting the Clock Quality setting of that Line to Network. If there
are multiple trunks to public exchanges, another trunk can be set as Fallback should the primary
clock signal fail. Other trunks should be set as Unsuitable.
Related links
Configuration Mode Field Descriptions on page 208
Analog Line on page 291
BRI Line on page 299
PRI Trunks on page 304
S0 Line on page 330
H.323 Line on page 333
IP DECT Line on page 345
SIP Line on page 353
Line | SIP DECT Line on page 380
Line | SM Line on page 382
Line | IP Office Line on page 389
Analog Line
Analog trunks can be provided within the systems in the following ways. In all cases the physical
ports are labeled as Analog. For full details of installation refer to the IP Office Installation manual.
Using ICLID: The system can route incoming calls using the ICLID received with the call. However
ICLID is not sent instantaneously. On analog trunks set to Loop Start ICLID, there will be a short
delay while the system waits for any ICLID digits before it can determine where to present the call.
Line Status: Analog line do not indicate call status other than whether the line is free or in use.
Some system features, for example retrieving unanswered forwards and making twinned calls make
use of the call status indicated by digital lines. This is not possible with analog lines. Once an analog
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Configuration Mode Field Descriptions
line has been seized, the system has to assume that the call is connected and treats it as having
been answered.
Dialing Complete: The majority of North-American telephony services use en-bloc dialing.
Therefore the use of a ; is recommended at the end of all dialing short codes that use an N. This is
also recommended for all dialing where secondary dial tone short codes are being used.
Ground Start: This type of analog trunk is only supported through the Analog Trunk external
expansion module.
Related links
Line on page 289
Line | Line Settings (Analog) on page 292
Line | Analog Options on page 293
Line | Line Settings (Analog)
This tab covers general settings for an analog line.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Line Number
This parameter is not configurable, it is allocated by the system.
Card/Module
Indicates the card slot or expansion module being used for the trunk device
providing the line. For IP500 V2 control units: 1 to 4 match the slots on the front
of the control unit from left to right. Expansion modules are numbered from 5
upwards, for example trunks on the module in Expansion Port 1 are shown as 5.
Port
Indicates the port on the Card/Module above to which the configuration settings
relate.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or
Private. The system will return number busy indication to any attempt to
connect a call on a Private trunk to a Public trunk or vice versa. This restriction
includes transfers, forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks,
VPNremote, application telecommuter mode.
Telephone Number
Used to remember the external telephone number of this line to assist with loopback testing. For information only.
Incoming Group ID
Default = 0, Range 0 to 99999. The Incoming Group ID to which a line belongs
is used to match it to incoming call routes in the system configuration. The
matching incoming call route is then used to route incoming calls. The same ID
can be used for multiple lines.
Outgoing Group ID
Default = 0, Range 0 to 99999.
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Line
Field
Description
Short codes that specify a number to dial also specify the line group to be used.
The system will then seize a line with a matching Outgoing Group ID.
For Basic Edition and Standard Edition deployments, the same ID can be used
for multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must
also be unique within the network, ie. the same ID cannot be used in the
configuration of any lines on another server system in the network.
Reserved Group ID
numbers
Description
90000 - 99999
Reserved for system use (not enforced).
99999 and 99998
In a Server Edition network, reserved for the H.323 lines
to the Primary Server and Secondary Server respectively.
99901 to 99930
In a Server Edition network, reserved for the H.323 lines
from the Primary Server to each expansion system in the
network.
0
In a Server Edition network, the ID 0 cannot be used.
98888
For IP Office deployed in an Enterprise Branch
environment, reserved for the SM line.
Outgoing Channels
Default = 1 (not changeable)
Voice Channels
Default = 1 (not changeable)
Prefix
Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is useful if
all users must dial a prefix to access an outside line. The prefix is automatically
placed in front of all incoming numbers so that users can dial the number back.
For outgoing calls: The system does not strip the prefix, therefore any prefixes
not suitable for external line presentation should be stripped using short codes.
Line Appearance ID
Default = Auto-assigned. Range = 2 to 9 digits. Allows a number to be assigned
to the line to identify it. On phone's that support call appearance buttons, a Line
Appearance button with the same number will show the status of the line and
can be used to answer calls on the line.The line appearance ID must be unique
and not match any extension number.
Admin
Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or
if the trunk is not connected.
Related links
Analog Line on page 291
Line | Analog Options
This tab covers analog line specific settings. The system wide setting CLI Type (System |
Telephony | Tones & Music) is used for to set the incoming CLI detection method for all analogue
trunks.
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Configuration Mode Field Descriptions
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Channel
Set by the system. Shown for information only.
Trunk Type
Default = Loop Start
Sets the analog line type. The options are:
• Ground Start: Ground Start is only supported on trunks provided by the Analog
Trunk 16 expansion module. It requires that the module and the control unit are
grounded. Refer to the IP Office installation manual.
• Loop Start
• Loop Start ICLID: As the system can use ICLID to route incoming calls, on
analog Loop Start ICLID trunks there is a few seconds delay while ICLID is
received before the call routing can be determined.
Signaling Type
Default = DTMF Dialing
Sets the signaling method used on the line. The options are:
• DTMF Dialing
• Pulse Dialing
Direction
Default = Both Directions
Sets the allowed direction of operation of the line. The options are:
• Incoming
• Outgoing
• Both Directions
Flash Pulse Width
Default = 500ms. Range = 0 to 2550ms.
Set the time interval for the flash pulse width.
Await Dial Tone
Default = 3000ms. Range = 0 to 25500ms.
Sets how long the system should wait before dialing out.
Echo Cancellation
Default = 16ms.
The echo cancellation should only be adjusted as high as required to remove echo
problems. Setting it to a higher value than necessary can cause other distortions.
Not used with external expansion module trunks. The options are (milliseconds):
• Off
• 8
• 16
• 32
• 64
• 128
Echo Reduction
Default = On. (ATM4Uv2 card only)
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Line
Field
Description
Used when impedance matching is not required but echo reduction is.
Mains Hum Filter
Default = Off.
If mains hum interference on the lines is detected or suspected, this settings can be
used to attempt to remove that interference. Useable with ATM16 trunks and IP500
ATM4U trunks. The options are:
• Off
• 50Hz
• 60Hz
Impedance
Set the impedance used for the line. This field is only available for certain system
locales. The value used for Default is set by the system Locale.
• Brazil: Default = 900R Adjustable between 600R and 900R as required by the
line provider.
• Korea: Default = Default (600ohms). In addition to the default impedance settings,
an alternate set of impedance values can be selected.
• United States: Default = Default (600 ohms). In addition to the default impedance
setting, the following alternate sets of impedance values Alternate1, Alternate2
and Alternate3 can be selected.
• The following values are used for Automatic Impedance Matching:
600+2150nF, 600, 900+2150nF, 900, 220+820||115nF, 370+620||310nF,
270+750||150nF, 320+1050||230nF, 350+1000||210nF, 800+100||210nF.
Quiet Line
This field is only available for certain system locales (see above). The setting may
be required to compensate for signal loss on long lines.
Digits to break dial tone
Default = 2. Range = Up to 3 digits.
During automatic impedance testing (see below), once the system has seized a
line, it dials this digit or digits to the line. In some cases it may be necessary to use
a different digit or digits. For example, if analog trunk go via another PBX system or
Centrex, it will be necessary to use the external trunk dialing prefix of the remote
system plus another digit, for example 92.
Automatic
Default = Yes. (ATM4Uv2 card only)
When set to Yes, the Default value is used. The value used for Default is set by
the system Locale.
When set to No, the Impedance value can be manually selected from the list of
possible values:
600
900 270+(750R || 150nF) and 275R + (780R || 150nF)
220+(820R || 120nF) and 220R+ (82R || 115nF)
370+(620R || 310nF)
320+(1050R || 230nF)
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Configuration Mode Field Descriptions
Field
Description
370+(820R || 110nF)
275+(780R || 115nF)
120+(820R || 110nF)
350+(1000R || 210nF)
200+(680R || 100nF)
600+2.16μF
900+1μF
900+2.16μF
600+1μF Global Impedance
Automatic Balance
Impedance Match
These controls can be used to test the impedance of a line and to then display the
best match resulting from the test. Testing should be performed with the line
connected but the system otherwise idle. To start testing click Start. The system will
then send a series of signals to the line and monitor the response, repeating this at
each possible impedance setting. Testing can be stopped at any time by clicking
Stop. When testing is complete, Manager will display the best match and ask
whether that match should be used for the line. If Yes is selected, Manager will also
ask whether the match should be applied to all other analog lines provided by the
same analog trunk card or module.
Note that on the Analog Trunk Module (ATM16), there are four control devices,
each supporting four channels. The impedance is set by the control device for all
four channels under its control. Consequently, the impedance match tool only
functions on lines 1, 5, 9, and 13.
Before testing, ensure that the system Locale setting (System | System) is
correctly set. Also check that the system Companding Law settings (System |
Telephony | Telephony) are set correctly. If either needs to be changed, make the
required change and save the setting to the system before proceeding with
impedance matching.
Due to hardware differences, the impedance matching result will vary slightly
depending on which type of trunk card or expansion module is being used.
Automatic Balance Impedance Matching, Quiet Line and Digits to break dial
tone are available for the Bahrain, Egypt, French Canadian, Kuwait, Morocco,
Oman, Pakistan, Qatar, Saudi Arabia, South Africa, Turkey, United Arab Emirates,
United States and Customize locales.
Allow Analog Trunk to
Trunk Connect
Default = Not selected (Off). When not enabled, users cannot transfer or forward
external calls back off-switch using an analog trunk if the call was originally made or
received on another analog trunk. This prevents transfers to trunks that do not
support disconnect clear.
If the Unsupervised Analog Trunk Disconnect Handling setting (System |
Telephony | Telephony) is enabled, this setting is greyed out and trunk to trunk
connections to any analog trunks are not allowed.
BCC
Default = Not selected [Brazil locale only]
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Line
Field
Description
A collect call is a call at the receiver's expense and by his permission. If supported
by the line provider, BCC (Block Collect Call) can be used to bar collect calls.
Secondary Dial Tone
Default = Off
Configures the use of secondary dial tone on analog lines. This is a different
mechanism from secondary dial tone using short codes. This method is used mainly
within the Russian locale. When selected, the options are:
• Await time: Default = 3000ms. Range = 0 to 25500ms. Used when secondary
dial tone (above) is selected. Sets the delay.
• After n Digits: Default = 1. Range = 0 to 10. Sets where in the dialing string, the
delay for secondary dial tone, should occur.
• Matching Digit: Default =8. Range = 0 to 9. The digit which, when first matched
in the dialing string, will cause secondary dial tone delay.
Long CLI Line
Default = Off
The CLI signal on some analog lines can become degraded and is not then
correctly detected. If you are sure that CLI is being provided but not detected,
selecting this option may resolve the problem.
Modem Enabled
Default = Off
The first analog trunk in a control unit can be set to modem operation (V32 with V42
error correction). This allows the trunk to answer incoming modem calls and be
used for system maintenance. When on, the trunk can only be used for analog
modem calls. The default system short code *9000* can be used to toggle this
setting.
For the IP500 ATM4 Uni Trunk Card Modem, it is not required to switch the card's
modem port on/off. The trunk card's V32 modem function can be accessed simply
by routing a modem call to the RAS service's extension number. The modem call
does not have to use the first analog trunk, instead the port remains available for
voice calls.
MWI Standard
Default = None. Release 9.0.3+
When the System | Voicemail | Voicemail Type is set to Analogue MWI, change
this setting to Bellcore FSK.
Pulse Dialing
These settings are used for pulse dialing.
• Mark: Default = 80 (80ms). Range = 0 to 255. Interval when DTMF signal is kept
active during transmission of DTMF signals.
• Space: Default = 80 (80ms). Range = 0 to 255. Interval of silence between DTMF
signal transmissions.
• Inter-Digit Pause: Default = 500ms. Range = 0 to 2550ms. Sets the pause
between digits transmitted to the line.
Ring Detection
These settings are used for ring detection.
• Ring Persistency: Default = Set according to system locale. Range = 0 to
2550ms. The minimum duration of signal required to be recognized.
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Configuration Mode Field Descriptions
Field
Description
• Ring Off Maximum: Default = Set according to system locale. Range = 0 to
25500ms. The time required before signaling is regarded as ended.
Disconnect Clear
Disconnect clear (also known as 'Line Break' or 'Reliable Disconnect') is a method
used to signal from the line provider that the call has cleared. The system also uses
'Tone Disconnect', which clears an analog call after 6 seconds of continuous tone,
configured through the Busy Tone Detection (System | Telephony | Tones &
Music) settings.
• Enable: Default = On Enables the use of disconnect clear.
• Units: Default = 500ms. Range = 0 to 2550ms. This time must be less than the
actual disconnect time period used by the line provider by at least 150ms.
If the Unsupervised Analog Trunk Disconnect Handling setting (System |
Telephony | Telephony) is enabled, this setting is greyed out and disconnect clear
disabled..
DTMF
These settings are used for DTMF dialing.
• On: Default = 40ms. Range = 0 to 255ms. The width of the on pulses generated
during DTMF dialing.
• Off: Default = 60ms. Range = 0 to 255ms. The width of the off pulses generated
during DTMF dialing.
BCC Flash Pulse Width
[Brazil locale only] Default = 100 (1000ms). Range = 0 to 255.
Sets the BCC (Block collect call) flash pulse width.
Gains
These settings are used to adjust the perceived volume on all calls.
• A | D: Default = 0dB. Range =-10.0dB to +6.0dB in 0.5dB steps. Sets the analog
to digital gain applied to the signal received from the trunk by the system. To
conform with the Receive Objective Loudness Rating at distances greater than
2.7km from the central office, on analog trunks a receive gain of 1.5dB must be
set.
• D | A: Default = 0dB. Range =-10.0dB to +6.0dB in 0.5dB steps. Sets the digital to
analog gain applied to the signal from the system to the trunk.
• Voice Recording: Default = Low Used to adjust the volume level of calls
recorded by voicemail. The ptions are:
- Low
- Medium
- High
Related links
Analog Line on page 291
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Line
BRI Line
BRI trunks are provided by the installation of a BRI trunk card into the control unit. The cards are
available in different variants with either 2 or 4 physical ports. Each port supports 2 B-channels for
calls. For full details of installation refer to the IP Office Installation manual.
Point-to-Point or Multipoint
BRI lines can be used in either Point-to-Point or Point-to-Multipoint mode. Point-to-Point lines are
used when only one device terminates a line in a customer's office. Point-to-Multipoint lines are
used when more than one device may be used on the line at the customer's premises. There are
major benefits in using Point-to-Point lines:• The exchange knows when the line/terminal equipment is down/dead, thus it will not offer calls
down that line. If the lines are Point-to-Multipoint, calls are always offered down the line and fail
if there is no response from the terminal equipment. So if you have two Point-to-Multipoint lines
and one is faulty 50% of incoming calls fail.
• You get a green LED on the Control Unit when the line is connected. With Point-to-Multipoint
lines some exchanges will drop layer 1/2 signals when the line is idle for a period.
• The timing clock is locked to the exchange. If layer 1/2 signals disappear on a line then the
Control Unit will switch to another line, however this may result in some audible click when the
switchover occurs.
The system's default Terminal Equipment Identifier (TEI) will normally allow it to work on Point-toPoint or Point-to-Multipoint lines. However if you intend to connect multiple devices simultaneously
to an BRI line, then the TEI should be set to 127. With a TEI of 127, the control unit will ask the
exchange to allocate a TEI for operation.
Note:
When connected to some manufactures equipment, which provides an S0 interface (BRI), a
defaulted Control Unit will not bring up the ISDN line. Configuring the Control Unit to a TEI of
127 for that line will usually resolve this.
Related links
Line on page 289
Line | BRI Line on page 299
Line | Channels (BRI) on page 304
Line | BRI Line
These settings are not mergeable. Changes to these settings will require a reboot of the system.
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Configuration Mode Field Descriptions
Field
Description
Card/Module
Indicates the card slot or expansion module being used for the trunk device
providing the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit
from left to right. Expansion modules are numbered from 5 upwards, for example
trunks on the module in Expansion Port 1 are shown as 5.
Port
Indicates the port on the Card/Module above to which the configuration settings
relate.
Line Number
This parameter is not configurable; it is allocated by the system.
Line Sub Type
Default = ETSI Select to match the particular line type provided by the line
provider.
IP500 BRI daughter cards can be configured for So (S-Bus) operation for
connection to ISDN terminal devices. Note that this requires the addition of
terminating resistors at both the system and remote ends, and the use of a suitable
cross-over cable. For full details refer to the IP Office Installation manual.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Telephone Number
Used to remember the external telephone number of this line to assist with loopback testing. For information only.
Incoming Group ID
Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used
to route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID
Default = 0, Range 0 to 99999. Short codes that specify a number to dial also
specify the line group to be used. The system will then seize a line with a matching
Outgoing Group ID.
For Basic Edition and Standard Edition deployments, the same ID can be used for
multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must also
be unique within the network, ie. the same ID cannot be used in the configuration
of any lines on another server system in the network.
Reserved Group ID numbers
Description
90000 - 99999
Reserved for system use (not enforced).
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Line
Field
Prefix
Description
99999 and 99998
In a Server Edition network, reserved for
the H.323 lines to the Primary Server
and Secondary Server respectively.
99901 to 99930
In a Server Edition network, reserved for
the H.323 lines from the Primary Server
to each expansion system in the
network.
0
In a Server Edition network, the ID 0
cannot be used.
98888
For IP Office deployed in an Enterprise
Branch environment, reserved for the
SM line.
Default = Blank. The prefix is used in the following ways:
• For incoming calls: The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the
Prefix field is added to the ICLID.
• For outgoing calls: The prefix is not stripped, therefore any prefixes not suitable
for external line presentation should be stripped using short codes.
National Prefix
Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number
is presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix
Default = 00
This indicates the digits to be prefixed to an incoming international call. When a
number is presented from ISDN as an "international number" this prefix is added.
For example 441923000000 is converted to 00441923000000.
TEI
Default = 0 The Terminal Equipment Identifier. Used to identify each device
connected to a particular ISDN line. For Point-to-Point lines this is 0. It can also be
0 on a Point to Multipoint line, however if multiple devices are sharing a Point-toMultipoint line it should be set to 127 which results in the exchange allocating the
TEI's to be used.
Number of Channels
Default = 2. Range = 0 to 2.
Defines the number of operational channels that are available on this line.
Outgoing Channels
Default = 2. Range = 0 to 2.
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.
Voice Channels
Default = 2. Range = 0 to 2.
The number of channels available for voice use.
Data Channels
Default = 2. Range = 0 to 2.
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Configuration Mode Field Descriptions
Field
Description
The number of channels available for data use. If left blank, the value is 0.
Clock Quality
Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether
the system should try to take its clock source for call synchronization and signalling
from this line. Preference should always be given to using the clock source from a
central office exchange if available by setting at least one exchange line to
Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network
source not be available.
• Lines from which the clock source should not be taken should be set as
Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock
source.
• In scenarios where several systems are network via digital trunk lines, care must
be taken to ensure that all the systems use the same clock source. The current
source being used by a system is reported within the System Status Application.
Add 'Not-end-to-end
Default = Never*. Sets whether the optional 'Not end-to-end ISDN' information
ISDN' Information Element element should be added to outgoing calls on the line. The options are Never,
Always or POTS (only if the call was originated by an analog extension). *The
default is Never except for the following locales; for Italy the default is POTS, for
New Zealand the default is Always.
Progress Replacement
Default = None.
Progress messages are defined in the Q.931 ISDN connection control signaling
protocol. Generally, If a progress message is sent, the caller does not get
connected and so typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to configure
alternative signaling to the ISDN line for internally generated Progress messages.
The options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not
hear the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically will
accrue call costs.
Supports Partial Rerouting Default = Off.
Partial rerouting (PR) is an ISDN feature. It is supported on external (non-network
and QSIG) ISDN exchange calls. When an external call is transferred to another
external number, the transfer is performed by the ISDN exchange and the channels
to the system are freed. Use of this service may need to be requested from the line
provider and may incur a charge.
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Line
Field
Description
Force Number Plan to
ISDN
Default = Off.
Send Redirecting Number
Default = Off.
This option is only configurable when Support Partial Rerouting is also enabled.
When selected, the plan/type parameter for Partial Rerouting is changed from
Unknown/Unknown to ISDN/Unknown.
This option can be used on ISDN trunks where the redirecting service is supported
by the trunk provider. Where supported, on twinned calls the caller ID of the
original call is passed through to the twinning destination. This option is only used
for twinned calls.
Support Call Tracing
Default = Off. The system supports the triggering of malicious caller ID (MCID)
tracing at the ISDN exchange. Use of this feature requires liaison with the ISDN
service provider and the appropriate legal authorities to whom the call trace will be
passed. The user will also need to be enabled for call tracing and be provider with
either a short code or programmable button to activate MCID call trace. Refer to
Malicious Call Tracing in the Telephone Features section for full details.
Active CCBS Support
Default = Off.
Call completion to a busy subscriber (CCBS). It allows automatic callback to be
used on outgoing ISDN calls when the destination is busy. This feature can only be
used on point-to-point trunks. Use of this service may need to be requested from
the line provider and may incur a charge.
Passive CCBS
Default = Off.
Cost Per Charging Unit
Advice of charge (AOC) information can be display on T3/T3IP phones and output
in SMDR. The information is provided in the form of charge units. This setting is
used to enter the call cost per charging unit set by the line provider. The values are
1/10,000th of a currency unit. For example if the call cost per unit is £1.07, a value
of 10700 should be set on the line. Refer to Advice of Charge.
Send original calling party Default = Off.
for forwarded and twinning
Use the original calling party ID when forwarding calls or routing twinned calls.
calls
This setting applies to the following ISDN lines:
• PRI24 with subtypes PRI, ETSI and ETSI CHI
• PRI30 with subtypes ETSI and ETSI CHI
Originator number for
forwarded and twinning
calls
Default = blank.
The number used as the calling party ID when forwarding calls or routing twinned
calls. This field is grayed out when the Send original calling party for forwarded
and twinning calls setting is enabled.
This setting applies to the following ISDN lines:
• PRI24 with subtypes PRI, ETSI and ETSI CHI
• PRI30 with subtypes ETSI and ETSI CHI
Related links
BRI Line on page 299
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Configuration Mode Field Descriptions
Line | Channels (BRI)
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel
either double-click on it or click the channel and then select Edit.
To edit multiple channels at the same time, select the required channels using Ctrl or Shift and then
click Edit. When editing multiple channels, fields that must be unique such as Line Appearance ID
are not shown.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Line Appearance
ID
Default = Auto-assigned. Range = 2 to 9 digits.
Used for configuring Line Appearances with button programming. The line appearance ID
must be unique and not match any extension number. Line appearance is not supported
for trunks set to QSIG operation and is not recommended for trunks be used for DID.
Related links
BRI Line on page 299
PRI Trunks
PRI trunks are provided by the installation of a PRI trunk card into the control unit. avThe IP500 PRIU trunk card can be configured (see below) to one of those line types. The cards are also available
with either 1 or 2 physical ports. The number of B-channels supported by each physical port
depends on the line type of the card.
• E1: 30 B-channels and 1 D-channel per port.
• T1: 24 B-channels per port.
• US PRI: 23 B-channels and 1 D-channel per port.
• E1-R2: 30 B-channels and 1 D-channel per port.
For full details of installation refer to the IP Office Installation manual.
IP500 PRI-U Trunk Card Line Type
The IP500 PRI-U card can be configured to support either E1, T1 or E1-R2 PRI line types. To select
the line type required, right-click on the line in the group or navigation pane and select Change
Universal PRI Card Line Type.
The control unit supports 8 B-channels on each IP500 PRI-U card fitted. Additional B-channels up to
the full capacity of IP500 PRI-U ports installed require licenses added to the configuration. Dchannels are not affected by licensing.
For ETSI and QSIG trunks, license instances are consumed by the number of calls in progress on
B-channels.
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Line
For T1, E1R2 and ETSI CHI trunks, licenses instances are consumed by the channels set as in
service.
Related links
Line on page 289
E1 Line on page 305
E1 R2 Line on page 312
T1 Line on page 317
T1 PRI Line on page 323
E1 Line
Related links
PRI Trunks on page 304
Line | E1 PRI Line on page 305
Line | E1 Short Codes on page 310
Line | E1 PRI Channels on page 311
Line | E1 PRI Line
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Line Number
This parameter is not configurable; it is allocated by the system.
Line Sub Type
Select to match the particular line type provided by the line provider. The options are:
• ETSI
• ETSI CHI
• QSIG A
• QSIG B
ETSI CHI is used to send the channel allocation ID (CHI) in the call setup signalling.
This is a request to use a particular B-channel rather than use any B-channel
allocated by the central office exchange.
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice
Networking licenses.
Card/Module
Indicates the card slot or expansion module being used for the trunk device providing
the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from
left to right. Expansion modules are numbered from 5 upwards, for example trunks
on the module in Expansion Port 1 are shown as 5.
Port
Indicates the port on the Card/Module above to which the configuration settings
relate.
Network Type
Default = Public.
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Configuration Mode Field Descriptions
Field
Description
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Telephone Number
Used to remember the external telephone number of this line to assist with loop-back
testing. For information only.
Channel Allocation
Default = 30|1.
For lines set to ETSI CHI, this option allows the system to select the default order in
which channels should be used for outgoing calls. Typically this is set as the
opposite of the default order in which the central office exchange uses channels for
incoming calls.
For lines set to the Line Sub Type of ETSI CHI, the Incoming Group ID is set as
part of the individual channel settings.
Incoming Group ID
Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID
Default = 0, Range 0 to 99999.
Short codes that specify a number to dial also specify the line group to be used. The
system will then seize a line with a matching Outgoing Group ID.
For Basic Edition and Standard Edition deployments, the same ID can be used for
multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must also be
unique within the network, ie.
the same ID cannot be used in the configuration of any lines on another server
system in the network.
Reserved Group ID numbers
Description
90000 - 99999
Reserved for system use (not enforced).
99999 and 99998
In a Server Edition network, reserved for
the H.323 lines to the Primary Server and
Secondary Server respectively.
99901 to 99930
In a Server Edition network, reserved for
the H.323 lines from the Primary Server
to each expansion system in the network.
0
In a Server Edition network, the ID 0
cannot be used.
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Line
Field
Description
98888
Prefix
For IP Office deployed in an Enterprise
Branch environment, reserved for the SM
line.
Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the
Prefix field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable
for external line presentation should be stripped using short codes.
National Prefix
Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number is
presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix
Default = 00
This indicates the digits to be prefixed to an incoming international call. When a
number is presented from ISDN as an "international number" this prefix is added. For
example 441923000000 is converted to 00441923000000.
TEI
Default = 0 The
Terminal Equipment Identifier. Used to identify each Control Unit connected to a
particular ISDN line. For Point to Point lines this is typically (always) 0. It can also be
0 on a Point to Multi-Point line, however if multiple devices are sharing a Point to
Multi-Point line it should be set to 127 which results in the exchange deciding on the
TEI's to be used.
Number of Channels
Defines the number of operational channels that are available on this line. Up to 30
for E1 PRI, 23 for T1 PRI.
Outgoing Channels
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls. Only available when the
Line Sub Type is set to ETSI.
Voice Channels
The number of channels available for voice use. Only available when the Line Sub
Type is set to ETSI.
Data Channels
The number of channels available for data use. Only available when the Line Sub
Type is set to ETSI.
CRC Checking
Default = On
Switches CRC on or off.
Line Signalling
Default = CPE This option is not used for lines where the Line SubType is set to
QSIG. Select either CPE (customer premises equipment) or CO (central office). The
CO feature is intended to be used primarily as a testing aid. It allows PRI lines to be
tested in a back-to-back configuration, using crossover cables.
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Configuration Mode Field Descriptions
Field
Description
The CO feature operates on this line type by modifying the way in which incoming
calls are disconnected for system configuration in Brazil and Argentina. In these
locales, the CO setting uses Forced-Release instead of Clear-Back to disconnect
incoming calls. The Brazilian Double-Seizure mechanism, used to police Collect
calls, is also disabled in CO mode.
Clock Quality
Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from
this line. Preference should always be given to using the clock source from a central
office exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network source
not be available.
• Lines from which the clock source should not be taken should be set as
Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must
be taken to ensure that all the systems use the same clock source. The current
source being used by a system is reported within the System Status Application.
Add 'Not-end-to-end
ISDN' Information
Element
Default = Never
Sets whether the optional 'Not end-to-end ISDN' information element should be
added to outgoing calls on the line. The options are:
• Never
• Always
• POTS(only if the call was originated by an analog extension).
The default is Never except for the following locales; for Italy the default is POTS, for
New Zealand the default is Always.
Progress Replacement
Default = None.
Progress messages are defined in the Q.931 ISDN connection control signaling
protocol. Generally, If a progress message is sent, the caller does not get connected
and so typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to configure
alternative signaling to the ISDN line for internally generated Progress messages.
The options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not
hear the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically will
accrue call costs.
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Line
Field
Description
Supports Partial
Rerouting
Default = Off.
Force Number Plan to
ISDN
Default = Off.
Send Redirecting
Number
Default = Off.
Support Call Tracing
Default = Off.
Partial rerouting (PR) is an ISDN feature. It is supported on external (non-network
and QSIG) ISDN exchange calls. When an external call is transferred to another
external number, the transfer is performed by the ISDN exchange and the channels
to the system are freed. Use of this service may need to be requested from the line
provider and may incur a charge.
This option is only configurable when Support Partial Rerouting is also enabled.
When selected, the plan/type parameter for Partial Rerouting is changed from
Unknown/Unknown to ISDN/Unknown.
This option can be used on ISDN trunks where the redirecting service is supported
by the trunk provider. Where supported, on twinned calls the caller ID of the original
call is passed through to the twinning destination. This option is only used for
twinned calls.
The system supports the triggering of malicious caller ID (MCID) tracing at the ISDN
exchange. Use of this feature requires liaison with the ISDN service provider and the
appropriate legal authorities to whom the call trace will be passed. The user will also
need to be enabled for call tracing and be provider with either a short code or
programmable button to activate MCID call trace. Refer to Malicious Call Tracing in
the Telephone Features section for full details.
Active CCBS Support
Default = Off.
Call completion to a busy subscriber (CCBS). It allows automatic callback to be used
on outgoing ISDN calls when the destination is busy. This feature can only be used
on point-to-point trunks. Use of this service may need to be requested from the line
provider and may incur a charge.
Passive CCBS
Default = Off.
Cost Per Charging Unit
Advice of charge (AOC) information can be display on T3/T3IP phones and output in
SMDR. The information is provided in the form of charge units. This setting is used to
enter the call cost per charging unit set by the line provider. The values are
1/10,000th of a currency unit. For example if the call cost per unit is £1.07, a value of
10700 should be set on the line. Refer to Advice of Charge.
Admin
Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
Send original calling
party for forwarded and
twinning calls
Default = Off.
Use the original calling party ID when forwarding calls or routing twinned calls.
This setting applies to the following ISDN lines:
• PRI24 with subtypes PRI, ETSI and ETSI CHI
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Configuration Mode Field Descriptions
Field
Description
• PRI30 with subtypes ETSI and ETSI CHI
Originator number for
forwarded and twinning
calls
Default = blank.
The number used as the calling party ID when forwarding calls or routing twinned
calls. This field is grayed out when the Send original calling party for forwarded
and twinning calls setting is enabled.
This setting applies to the following ISDN lines:
• PRI24 with subtypes PRI, ETSI and ETSI CHI
• PRI30 with subtypes ETSI and ETSI CHI
The following fields are shown for a US T1 trunk card set to ETSI or QSIG operation. These cards
have the same settings E1 PRI trunk cards set to ETSI or QSIG but only support 23 channels.
Field
Description
CSU Operation
Check this field to enable the T1 line to respond to loop-back requests from the line.
Haul Length
Default = 0-115 feet
Sets the line length to a specific distance.
Channel Unit
Default = Foreign Exchange This field should be set to match the channel signaling
equipment provided by the Central Office. The options are Foreign Exchange,
Special Access or Normal.
Related links
E1 Line on page 305
Line | E1 Short Codes
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those types
of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Related links
E1 Line on page 305
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Line
Line | E1 PRI Channels
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel
either double-click on it or click the channel and then select Edit.
To edit multiple channels at the same time, select the required channels using Ctrl or Shift and then
click Edit. When editing multiple channels, fields that must be unique such as Line Appearance ID
are not shown.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Line Appearance ID
Default = Auto-assigned. Range = 2 to 9 digits.
Used for configuring Line Appearances with button programming. The line
appearance ID must be unique and not match any extension number. Line
appearance is not supported for trunks set to QSIG operation and is not
recommended for trunks be used for DID.
If the trunk Line Sub Type is set to ETSI CHI, outgoing line appearance calls must
use the corresponding channel.
The following additional fields are shown for lines where the Line Sub Type is set to ETSI CHI.
Field
Description
Incoming Group ID
Default = 0, Range 0 to 99999. The Incoming Group ID to which a line belongs
is used to match it to incoming call routes in the system configuration. The
matching incoming call route is then used to route incoming calls. The same ID
can be used for multiple lines.
Outgoing Group ID
Default = 0, Range 0 to 99999.
Short codes that specify a number to dial also specify the line group to be used.
The system will then seize a line with a matching Outgoing Group ID.
For Basic Edition and Standard Edition deployments, the same ID can be used
for multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must
also be unique within the network, ie. the same ID cannot be used in the
configuration of any lines on another server system in the network.
Reserved Group ID
numbers
Description
90000 - 99999
Reserved for system use (not enforced).
99999 and 99998
In a Server Edition network, reserved for the H.323
lines to the Primary Server and Secondary Server
respectively.
99901 to 99930
In a Server Edition network, reserved for the H.323
lines from the Primary Server to each expansion
system in the network.
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Configuration Mode Field Descriptions
Field
Direction
Description
0
In a Server Edition network, the ID 0 cannot be
used.
98888
For IP Office deployed in an Enterprise Branch
environment, reserved for the SM line.
Default = Bothways
The direction of calls on the channel. The options are:
• Incoming
• Outgoing
• Bothways
Bearer
Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
Admin
Default = Out of Service. This field can be used to indicate whether the channel
is in use or not. On trunks where only a limited number of channels have been
requested from the trunk provider (known as sub-equipped trunks), those
channels not provided should be set as Out of Service. For channels that are
available but are temporarily not being used select Maintenance.
Tx Gain
Default = 0dB. Range = -10dBb to +5dB.
The transmit gain in dB.
Rx Gain
Default = 0dB. Range = -10dBb to +5dB.
The receive gain in dB.
Related links
E1 Line on page 305
E1 R2 Line
Related links
PRI Trunks on page 304
Line | E1-R2 Options on page 312
Line | E1-R2 Channels on page 314
Line | E1 R2 MFC Group on page 316
Line | E1-R2 Advanced on page 316
Line | E1-R2 Options
These settings are not mergeable. Changes to these settings will require a reboot of the system.
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Line
Field
Description
Card/Module
Indicates the card slot or expansion module being used for the trunk device providing
the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit from
left to right. Expansion modules are numbered from 5 upwards, for example trunks on
the module in Expansion Port 1 are shown as 5.
Port
Indicates the port on the Card/Module above to which the configuration settings relate.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private. The
system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Line Number
Allocated by the system.
Line SubType
Default = E1-R2
The options are:
• E1-R2
• ETSI
• QSIGA
• QSIGB
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice
Networking licenses.
Channel Allocation
Default = 30 | 1
The order, 30 | 1 or 1 | 30, in which channels are used.
Country (Locale)
Default = Mexico. Select the locale that matches the area of usage. Note that
changing the locale will return the MFC Group settings to the defaults for the selected
locale. Currently supported locales are:
• Argentina
• Brazil
• China
• India
• Korea
• Mexico
• None
Admin
Default = In Service.
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Configuration Mode Field Descriptions
Field
Description
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
The table at the base of the form displays the settings for the individual channels
provided by the line. For details of the channel settings see the E1-R2 Channel form.
To edit a channel, either double-click on it or right-click and select Edit. This will
display the Edit Channel dialog box. To edit multiple channels at the same time select
the channels whilst pressing the Shift or Ctrl key. Then right-click and select Edit.
Related links
E1 R2 Line on page 312
Line | E1-R2 Channels
This tab allows settings for individual channels within the trunk to be adjusted. To edit a channel,
select the required channel or channels and click Edit.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
The channel settings are split into two sub-tabs, E1R2 Edit Channel and Timers.
The Timers tab displays the various timers provided for E1-R2 channels. These should only be
adjusted when required to match the line provider's settings.
Field
Descriptions
Channel
The channel or channels being edited.
Incoming Group ID
Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming
call routes in the system configuration. The matching incoming call route is then
used to route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID
Default = 0, Range 0 to 99999.
Short codes that specify a number to dial also specify the line group to be used.
The system will then seize a line with a matching Outgoing Group ID.
For Basic Edition and Standard Edition deployments, the same ID can be used
for multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must
also be unique within the network, ie. the same ID cannot be used in the
configuration of any lines on another server system in the network.
Reserved Group ID numbers
Description
90000 - 99999
Reserved for system use (not enforced).
99999 and 99998
In a Server Edition network, reserved for the
H.323 lines to the Primary Server and
Secondary Server respectively.
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Line
Field
Direction
Descriptions
99901 to 99930
In a Server Edition network, reserved for the
H.323 lines from the Primary Server to each
expansion system in the network.
0
In a Server Edition network, the ID 0 cannot
be used.
98888
For IP Office deployed in an Enterprise
Branch environment, reserved for the SM line.
Default = Both Directions
The direction of calls on the channel. The options are:
• Incoming
• Outgoing
• Both Directions
Bearer
Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
Line Signaling Type
Default = R2 Loop Start
The signaling type used by the channel. Current supported options are:
• R2 Loop Start
• R2 DID
• R2 DOD
• R2 DIOD
• Tie Immediate Start
• Tie Wink Start
• Tie Delay Dial
• Tie Automatic
• WAN Service
• Out of Service
Dial Type
Default = MFC Dialing
The type of dialing supported by the channel. The options are:, or .
• MFC Dialing
• Pulse Dialing
• DTMF Dialing
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Configuration Mode Field Descriptions
Related links
E1 R2 Line on page 312
Line | E1 R2 MFC Group
These tabs show the parameter assigned to each signal in an MFC group. The defaults are set
according to the Country (Locale) on the Line tab. All the values can be returned to default by the
Default All button on the Advanced tab.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
To change a setting either double-click on it or right-click and select Edit.
Related links
E1 R2 Line on page 312
Line | E1-R2 Advanced
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Zero Suppression
Default = HDB3
Selects the method of zero suppression used (HDB3 or AMI).
Clock Quality
Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from
this line. Preference should always be given to using the clock source from a central
office exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network source
not be available.
• Lines from which the clock source should not be taken should be set as Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must be
taken to ensure that all the systems use the same clock source. The current source
being used by a system is reported within the System Status Application.
Clock Quality
Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether the
system should try to take its clock source for call synchronization and signalling from
this line. Preference should always be given to using the clock source from a central
office exchange if available by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network source
not be available.
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Line
Field
Description
• Lines from which the clock source should not be taken should be set as Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz clock source.
• In scenarios where several systems are network via digital trunk lines, care must be
taken to ensure that all the systems use the same clock source. The current source
being used by a system is reported within the System Status Application.
Pulse Metering Bit
Default = A Bit
Sets which bit should be used to indicate the pulse metering signal. The options are:
• A Bit
• B Bit
• C Bit
Line Signaling
Default = CPE
The options are:
• CPE
• CO
• CO
The feature is intended to be used primarily as a testing aid. It allows T1 and E1 lines
to be tested in a back-to-back configuration, using crossover (QSIG) cables.
The CO feature operates by modifying the way in which incoming calls are
disconnected for system configuration in Brazil and Argentina. In these locales, the
CO setting uses Forced-Release instead of Clear-Back to disconnect incoming calls.
The Brazilian Double-Seizure mechanism used to police Collect calls, is also disabled
in CO mode.
Incoming Routing
Digits
Default = 4
CRC Checking
Default = On
Sets the number of incoming digits used for incoming call routing.
Switches CRC on or off.
Default All Group
Settings
Default the MFC Group tab settings.
Line Signaling Timers
To edit one of these timers, either double-click on the timer or right-click on a timer
and select the action required.
Related links
E1 R2 Line on page 312
T1 Line
Related links
PRI Trunks on page 304
Line | US T1 Line on page 318
T1 Channels on page 320
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Configuration Mode Field Descriptions
Line | US T1 Line
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Line Number
Allocated by the system.
Card/Module
Indicates the card slot or expansion module being used for the trunk device
providing the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control
unit from left to right. Expansion modules are numbered from 5 upwards,
for example trunks on the module in Expansion Port 1 are shown as 5.
Port
Indicates the port on the Card/Module above to which the configuration
settings relate.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict
Network Interconnect is enabled. It allows the trunk to be set as either
Public or Private. The system will return number busy indication to any
attempt to connect a call on a Private trunk to a Public trunk or vice versa.
This restriction includes transfers, forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems
also using any of the following other system features: multi-site networks,
VPNremote, application telecommuter mode.
Line Sub Type
Default = T1
Set to T1 for a T1 line.
Channel Allocation
Default = 24 | 1
The order, 24 to 1 or 1 to 24, in which channels are used.
Prefix
Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is
useful if all users must dial a prefix to access an outside line. The prefix is
automatically placed in front of all incoming numbers so that users can dial
the number back.
Framing
Default = ESF
Selects the type of signal framing used. The options are:
• ESF
• D4
Zero Suppression
Default = B8ZS
Selects the method of zero suppression used. The options are:
• B8ZS
• AMI ZCS
Clock Quality
Default = Network
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Line
Field
Description
Refer to the IP Office Installation Manual for full details. This option sets
whether the system should try to take its clock source for call
synchronization and signalling from this line. Preference should always be
given to using the clock source from a central office exchange if available
by setting at least one exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are
used is described in the IP Office Installation Manual. If additional lines
are available, Fallback can be used to specify a clock source to use
should the Network source not be available.
• Lines from which the clock source should not be taken should be set as
Unsuitable.
• If no clock source is available, the system uses its own internal 8KHz
clock source.
• In scenarios where several systems are network via digital trunk lines,
care must be taken to ensure that all the systems use the same clock
source. The current source being used by a system is reported within the
System Status Application.
Haul Length
Default = 0-115 feet.
Sets the line length to a specific distance.
Channel Unit
Default = Foreign Exchange
This field should be set to match the channel signaling equipment provided
by the Central Office. The options are:
• Foreign Exchange
• Special Access
• Normal
CRC Checking
Default = On
Turns CRC on or off.
Line Signaling
Default = CPE
This field affects T1 channels set to Loop-Start or Ground-Start. The field
can be set to either CPE (Customer Premises Equipment) or CO (Central
Office). This field should normally be left at its default of CPE. The setting
CO is normally only used in lab back-to-back testing.
Incoming Routing Digits
Default=0 (present call immediately)
Sets the number of routing digits expected on incoming calls. This allows
the line to present the call to the system once the expected digits have
been received rather than waiting for the digits timeout to expire. This field
only affects T1 line channels set to E&M Tie, E&M DID, E&M Switched 56K
and Direct Inward Dial.
CSU Operation
Enable this field to enable the T1 line to respond to loop-back requests
from the line.
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Configuration Mode Field Descriptions
Field
Description
Enhanced Called Party Number
Default = Off
This option is not supported for systems set to the United States locale.
Normally the dialed number length is limited to 15 digits. Selecting this
option increases the allowed dialed number length to 30 digits.
Admin
Default = In Service.
This field allows a trunk to be taken out of service if required for
maintenance or if the trunk is not connected.
Related links
T1 Line on page 317
T1 Channels
Navigation: Line | T1 Channels
The settings for each channel can be edited. Users have the option of editing individual channels by
double-clicking on the channel or selecting and editing multiple channels at the same time. Note that
the Line Appearance ID cannot be updated when editing multiple channels.
When editing a channel or channels, the settings available are displayed on two sub-tabs; T1 Edit
Channel and Timers.
The following settings are mergeable:
• Incoming Group ID
• Outgoing Group ID
• Line Appearance ID
• Admin
The remaining settings are not mergeable. Changes to these settings require a system reboot.
Field
Description
Channel
Allocated by the system.
Incoming Group
ID
Default = 0, Range 0 to 99999.
Outgoing Group
ID
Default = 1. Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call routes
in the system configuration. The matching incoming call route is then used to route
incoming calls. The same ID can be used for multiple lines.
Short codes that specify a number to dial also specify the line group to be used. The
system will then seize a line with a matching Outgoing Group ID.
In a Server Edition network, the Outgoing Group ID used on a system must also be
unique within the network. The same ID cannot be used in the configuration of any lines
on another server system in the network. For Standard Edition deployments, the same ID
can be used for multiple lines.
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Line
Field
Description
Reserved Group ID Numbers:
• 90000 - 99999 Reserved for system use (not enforced).
• 99999 and 99998 In a Server Edition network, reserved for the IP Office lines to the
Primary Server and Secondary Server respectively.
• 99901 to 99930 In a Server Edition network, reserved for the IP Office lines from the
Primary Server to each expansion system in the network.
• 0 In a Server Edition network, the ID 0 cannot be used.
• 98888 For IP Office deployed in an Enterprise Branch environment, reserved for the SM
line.
Line Appearance
ID
Default = Auto-assigned. Range = 2 to 9 digits.
Direction
Default = Bothway
Used for configuring Line Appearances with button programming. The line appearance ID
must be unique and not match any extension number. Line appearance is not supported
for trunks set to QSIG operation and is not recommended for trunks be used for DID.
The direction of calls on the channel. The options are:
• Incoming
• Outgoing
• Bothway
Bearer
Default = Any
The type of traffic carried by the channel. The options are:
• Voice
• Data
• Any
Admin
Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
Type
Default = Loop-Start.
The T1 emulates the following connections:
• Ground-Start
• Loop-Start
• E&M - TIE
• E&M - DID
• E&M Switched 56K
• Direct Inward Dial
• Clear Channel 64K
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Configuration Mode Field Descriptions
Field
Description
Trunks set to E&M - DID will only accept incoming calls.
If E&M - TIE is selected and the Outgoing Trunk Type is set to Automatic, no
secondary dial tone is provided for outgoing calls on this line/trunk.
Dial Type
Default = DTMF Dial
Select the dialing method required. The options are:
• DTMF Dial
• Pulse Dial
Incoming Trunk
Type
Default = Wink-Start
Used for E&M types only. The handshake method for incoming calls. The options are:
• Automatic
• Immediate
• Delay Dial
• Wink-Start
Outgoing Trunk
Type
Default = Wink-Start
Used for E&M types only. The handshake method for outgoing calls. The options are:
• Automatic
• Immediate
• Delay Dial
• Wink-Start
If the line Type is set to E&M-TIE and the Outgoing Trunk Type is set to Automatic, no
secondary dial tone is provided for outgoing calls on this line/trunk.
Tx Gain
Default = 0dB.
The transmit gain in dB.
Rx Gain
Default = 0dB.
The receive gain in dB.
Admin
Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or if the
trunk is not connected.
Timer Settings
This sub-tab allows various timers relating to operation of an individual channel to be adjusted.
These should only be adjusted to match the requirements of the line provider. The following is a list
of the default values. To reset a value, click on the current value and then right click and select from
the default, minimize and maximize options displayed.
Incoming Automatic Delay: 410.
Silent Interval: 1100.
Incoming Wink Delay: 100.
Outgoing Seizure: 10.
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Line
Wink Signal: 200.
Wink Start: 5000.
Incoming Dial Guard: 50.
Wink Validated: 80.
First Incoming Digit: 15000.
Wink End: 350.
Incoming Inter Digit: 5000.
Delay End: 5000.
Maximum Inter Digit: 300.
Outgoing Dial Guard: 590.
Flash Hook Detect: 240.
Outgoing IMM Dial Guard: 1500.
Incoming Disconnect: 300.
Outgoing Pulse Dial Break: 60.
Incoming Disconnect Guard: 800.
Outgoing Pulse Dial Make: 40.
Disconnected Signal Error: 240000.
Outgoing Pulse Dial Inter Digit: 720.
Outgoing Disconnect: 300.
Outgoing Pulse Dial Pause: 1500.
Outgoing Disconnect Guard: 800.
Flash Hook Generation: 500.
Ring Verify Duration: 220.
Outgoing End of Dial: 1000.
Ring Abandon: 6300.
Answer Supervision: 300.
Ping Verify: 600.
Incoming Confirm: 20.
Long Ring Time: 1100.
Related links
T1 Line on page 317
T1 PRI Line
Related links
PRI Trunks on page 304
Line | Line (T1 ISDN) on page 323
Channels (T1 ISDN) on page 326
Line | TNS (T1 ISDN) on page 328
Line | Special (T1 ISDN) on page 328
Line | Call By Call (US PRI) on page 329
Line | Line (T1 ISDN)
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Variable
Description
Line Number
Allocated by the system.
Card/Module
Indicates the card slot or expansion module being used for the trunk device
providing the line.
For IP500 V2 control units: 1 to 4 match the slots on the front of the control unit
from left to right. Expansion modules are numbered from 5 upwards, for
example trunks on the module in Expansion Port 1 are shown as 5.
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Configuration Mode Field Descriptions
Variable
Description
Port
Indicates the port on the Card/Module above to which the configuration settings
relate.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or
Private. The system will return number busy indication to any attempt to connect
a call on a Private trunk to a Public trunk or vice versa. This restriction includes
transfers, forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks,
VPNremote, application telecommuter mode.
Line Sub Type
: Default = PRI
Set to PRI. If set to T1 see Line Form (T1). If set to ETSI, ETSI CHI, QSIG A or
QSIG B see Line (E1).
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice
Networking licenses.
Channel Allocation
Default = 23 | 1
The order, 23 to 1 or 1 to 23, in which channels are used.
Switch Type
Default = NI2
The options are
• 4ESS
• 5ESS
• DMS100
• NI2
Provider
Default = Local Telco
Select the PSTN service provider (AT&T, Sprint, WorldCom or Local Telco).
Prefix
Default = Blank
Enter the number to prefix to all incoming numbers for callback. This is useful if
all users must dial a prefix to access an outside line. The prefix is automatically
placed in front of all incoming numbers so that users can dial the number back.
Add 'Not-end-to-end ISDN'
Information Element
Default = Never*.
Sets whether the optional 'Not end-to-end ISDN' information element should be
added to outgoing calls on the line. The options are
• Never
• Always
• POTS (only if the call was originated by an analog extension)
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Line
Variable
Description
*The default is Never except for the following locales; for Italy the default is
POTS, for New Zealand the default is Always.
Progress Replacement
Default = None.
Progress messages are defined in the Q.931 ISDN connection control signaling
protocol. Generally, If a progress message is sent, the caller does not get
connected and so typically does not accrue call costs.
Not all ISDN lines support Q.931 Progress messages. Use this setting to
configure alternative signaling to the ISDN line for internally generated Progress
messages. The options are:
• Alerting: Map to Q.931 Alerting. The call is not connected. The caller does not
hear the message and typically does not accrue call costs.
• Connect: Map to Q.931 Connect. The caller hears the message and typically
will accrue call costs.
Send Redirecting Number
Default = Off.
This option can be used on ISDN trunks where the redirecting service is
supported by the trunk provider. Where supported, on twinned calls the caller ID
of the original call is passed through to the twinning destination. This option is
only used for twinned calls.
Send Names
This option is available when the Switch Type above is set to DMS100. If set,
names are sent in the display field. The Z shortcode character can be used to
specify the name to be used.
Names Length
Set the allowable length for names, up to 15 characters, when Send Names is
set above.
Test Number
Used to remember the external telephone number of this line to assist with loopback testing. For information only.
Framing
Default = ESF
Selects the type of signal framing used (ESF or D4).
Zero Suppression
Default = B8ZS
Selects the method of zero suppression used (B8ZS or AMI ZCS).
Clock Quality
Default = Network
Refer to the IP Office Installation Manual for full details. This option sets whether
the system should try to take its clock source for call synchronization and
signalling from this line. Preference should always be given to using the clock
source from a central office exchange if available by setting at least one
exchange line to Network.
• If multiple lines are set as Network, the order in which those lines are used is
described in the IP Office Installation Manual. If additional lines are available,
Fallback can be used to specify a clock source to use should the Network
source not be available.
• Lines from which the clock source should not be taken should be set as
Unsuitable.
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Configuration Mode Field Descriptions
Variable
Description
• If no clock source is available, the system uses its own internal 8KHz clock
source.
• In scenarios where several systems are network via digital trunk lines, care
must be taken to ensure that all the systems use the same clock source. The
current source being used by a system is reported within the System Status
Application.
CSU Operation
Tick this field to enable the T1 line to respond to loop-back requests from the
line.
Haul Length
Default = 0-115 feet
Sets the line length to a specific distance.
Channel Unit
Default = Foreign Exchange
This field should be set to match the channel signaling equipment provided by
the Central Office. The options are
• Foreign Exchange
• Special Access
• Normal
CRC Checking
Default = On
Turns CRC on or off.
Line Signaling
The field can be set to either CPE (Customer Premises Equipment) or CO
(Central Office). This field should normally be left at its default of CPE. The
setting CO is normally only used in lab back-to-back testing.
Incoming Routing Digits
Default=0 (present call immediately)
Sets the number of routing digits expected on incoming calls. This allows the line
to present the call to the system once the expected digits have been received
rather than waiting for the digits timeout to expire. This field only affects T1 line
channels set to E&M Tie, E&M DID, E&M Switched 56K and Direct Inward
Dial.
Admin
Default = In Service.
This field allows a trunk to be taken out of service if required for maintenance or
if the trunk is not connected.
Related links
T1 PRI Line on page 323
Channels (T1 ISDN)
This tab allows settings for individual channels within the trunk to be adjusted. This tab is not
available for trunks sets to ETSI or QSIG mode.
Usability
These settings are not mergeable. Changes to these settings will require a reboot of the system.
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Line
Configuration Settings
Channel Allocated by the system.
Incoming Group ID: Default = 0, Range 0 to 99999. The Incoming Group ID to which a line belongs
is used to match it to incoming call routes in the system configuration. The matching incoming call
route is then used to route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID: Default = 0, Range 0 to 99999. Short codes that specify a number to dial also
specify the line group to be used. The system will then seize a line with a matching Outgoing Group
ID.
For Basic Edition and Standard Edition deployments, the same ID can be used for multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must also be unique within
the network, ie. the same ID cannot be used in the configuration of any lines on another server
system in the network.
Reserved Group ID numbers
Description
90000 - 99999
Reserved for system use (not enforced).
99999 and 99998
In a Server Edition network, reserved for the H.323
lines to the Primary Server and Secondary Server
respectively.
99901 to 99930
In a Server Edition network, reserved for the H.323
lines from the Primary Server to each expansion
system in the network.
0
In a Server Edition network, the ID 0 cannot be used.
98888
For IP Office deployed in an Enterprise Branch
environment, reserved for the SM line.
Line Appearance ID: Default = Auto-assigned. Range = 2 to 9 digits. Used for configuring Line
Appearances with button programming. The line appearance ID must be unique and not match any
extension number.
Direction: Default = Both Directions The direction of calls on the channel (Incoming, Outgoing or
Both Directions).
Bearer: Default = Any The type of traffic carried by the channel (Voice, Data or Any).
Service: Default = None. If the line provider is set to AT&T, select the type of service provided by
the channel. The options are:
Call by Call
SDN (inc GSDN)
MegaCom 800
MegaCom
Wats
Accunet
ILDS
I800
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Configuration Mode Field Descriptions
ETN
Private Line
AT&T Multiquest
For other providers, the service options are None or No Service.
Admin: Default = Out of Service Used to indicate the channel status (In Service, Out of Service or
Maintenance).
Tx Gain: Default = 0dB The transmit gain in dB.
Rx Gain: Default = 0dB The receive gain in dB.
Related links
T1 PRI Line on page 323
Line | TNS (T1 ISDN)
This tab is shown when the line Provider is set to AT&T. It allows the entry of the Network Selection
settings. These are prefixes for alternative long distance carriers. When a number dialed matches
an entry in the table, that pattern is stripped from the number before being sent out. This table is
used to set field in the TNS (Transit Network Selection) information element for 4ESS and 5ESS
exchanges. It is also used to set fields in the NSF information element.
Usability
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Configuration Settings
TNS Code: The pattern for the alternate long distance carrier. For example: The pattern 10XXX is
added to this tab. If 10288 is dialed, 10 is removed and 288 is placed in the TNS and NSF
information.
Related links
T1 PRI Line on page 323
Line | Special (T1 ISDN)
This tab is shown when the line Provider is set to AT&T. This table is used to set additional fields in
the NSF information element after initial number parsing by the TNS tab. These are used to indicate
the services required by the call. If the channel is set to Call by Call, then further parsing is done
using the records in the Call by Call tab.
Usability
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Configuration Settings
Short code: The number which results from the application of the rules specified in the User or
System Short code tables and the Network Selection table and the Call-by-call table to the number
dialed by the user.
Number: The number to be dialed to line.
Special: Default = No Operator (No Operator, Local Operator or Presubscribed Operator).
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Line
Plan: Default = National (National or International).
Typical values are:
Short Code
Number
Service
011N
N
No Operator, International
010N
N
Local Operator, International
01N
N
Local Operator, National
00N
N
Presubscribed Operator, National
0N
N
Presubscribed Operator, National
1N
1N
No operator, National
Related links
T1 PRI Line on page 323
Line | Call By Call (US PRI)
This tab is shown when the line Provider is set to AT&T. Settings in this tab are only used when
calls are routed via a channel which has its Service set to Call by Call.
It allows short codes to be created to route calls to a different services according to the number
dialed. Call By Call reduces the costs and maximizes the use of facilities. Call By Call chooses the
optimal service for a particular call by including the Bearer capability in the routing decision. This is
particularly useful when there are limited resources.
Usability
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Configuration Settings
Short Code: The number which results from the application of the rules specified in the User or
System Short code tables and the Network Selection table to the number dialed by the user.
Number: The number to be dialed to line.
Bearer: Default = Any The type of channel required for the call (Voice, Data or Any).
Service: Default = AT&T The service required by the call. The options are:
Call by Call
SDN (inc GSDN)
MegaCom 800
MegaCom
Wats
Accunet
ILDS
I800
ETN
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Configuration Mode Field Descriptions
Private Line
AT&T Multiquest
Related links
T1 PRI Line on page 323
S0 Line
These settings are used for S0 ports provided by an S08 expansion module connected a control
unit. For full details of installation refer to the IP Office Installation manual.
Though displayed as lines, these BRI ports are used for connection of ISDN2 devices such as video
conferencing units or ISDN PC cards.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can be
used to modify the incoming digits.
Related links
Line on page 289
Line | S0 Line on page 330
Line | S0 Short Codes on page 332
Line | S0 Channels on page 332
Line | S0 Line
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Line Number
This parameter is not configurable. It is allocated by the system.
Telephone Number
Used to remember the telephone number of this line. For information only.
Prefix
Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type
(National, International or Unknown). If the call type is unknown, then the
number in the Prefix field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not
suitable for external line presentation should be stripped using short
codes.
National Prefix
Default = 0
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Line
Field
Description
This indicates the digits to be prefixed to a incoming national call. When a
number is presented from ISDN as a "national number" this prefix is added.
For example 1923000000 is converted to 01923000000.
International Prefix
Default = 00
This indicates the digits to be prefixed to an incoming international call.
When a number is presented from ISDN as an "international number" this
prefix is added. For example 441923000000 is converted to
00441923000000.
Incoming Group ID
Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to
incoming call routes in the system configuration. The matching incoming
call route is then used to route incoming calls. The same ID can be used for
multiple lines.
Outgoing Group ID
Default = 0, Range 0 to 99999.
Short codes that specify a number to dial also specify the line group to be
used. The system will then seize a line with a matching Outgoing Group ID.
For Basic Edition and Standard Edition deployments, the same ID can be
used for multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system
must also be unique within the network, ie. the same ID cannot be used in
the configuration of any lines on another server system in the network.
TEI
Reserved Group ID numbers
Description
90000 - 99999
Reserved for system use (not
enforced).
99999 and 99998
In a Server Edition network, reserved
for the H.323 lines to the Primary
Server and Secondary Server
respectively.
99901 to 99930
In a Server Edition network, reserved
for the H.323 lines from the Primary
Server to each expansion system in
the network.
0
In a Server Edition network, the ID 0
cannot be used.
98888
For IP Office deployed in an Enterprise
Branch environment, reserved for the
SM line.
Default = 0
Not used. The Control Unit will ignore any entry.
Number of Channels
Default = 2
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Configuration Mode Field Descriptions
Field
Description
Defines the number of operational channels that are available on this line. 2
for BRI and up to 30 for PRI - depending upon the number of channels
subscribed.
Outgoing Channels
Default = 2
This defines the number of channels available, on this line, for outgoing
calls. This should normally be the same as Number of Channels field, but
can be reduced to ensure incoming calls cannot be blocked by outgoing
calls.
Voice Channels
Default = 2
The number of channels available for voice use.
Data Channels
Default = 2
The number of channels available for data use. If left blank the value is 0.
Related links
S0 Line on page 330
Line | S0 Short Codes
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those types
of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Related links
S0 Line on page 330
Line | S0 Channels
For S0 channels this form is not used.
Related links
S0 Line on page 330
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Line
H.323 Line
These lines are added manually. They allow voice calls to be routed over data links within the
system. They are therefore dependent on the IP data routing between the system and the
destination having being configured and tested.
Calls received on IP, S0 and QSIG trunks do not use incoming call routes. Routing for these is
based on incoming number received as if dialed on-switch. Line short codes on those trunks can be
used to modify the incoming digits.
Network Assessments
Not all data connections are suitable for voice traffic. A network assessment is required for internal
network connections. For external network connections a service level agreement is required from
the service provider. Avaya cannot control or be held accountable for the suitability of a data
connection for carrying voice traffic.
QSIG trunks trunks are not supported on IP500 V2 systems without IP500 Voice Networking
licenses.
Related links
Line on page 289
Line | H.323 VoIP Line on page 333
Line | H.323 Short Codes on page 335
Line | H.323 VoIP Settings on page 335
Line | H.323 VoIP Line
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Line Number
Default = Auto-filled. Range = 1 to 349.
Enter the line number that you wish. Note that this must be unique.
Telephone Number
Used to remember the telephone number of this line. For information only.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
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Configuration Mode Field Descriptions
Field
Description
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Prefix
Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the
Prefix field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable
for external line presentation should be stripped using short codes.
National Prefix
Default = 0
This indicates the digits to be prefixed to a incoming national call. When a number is
presented from ISDN as a "national number" this prefix is added. For example
1923000000 is converted to 01923000000.
International Prefix
Default = 00
This indicates the digits to be prefixed to an incoming international call. When a
number is presented from ISDN as an "international number" this prefix is added.
For example 441923000000 is converted to 00441923000000.
Location
Default = Cloud.
Specify a location to associate the extension with a physical location. Associating
an extension with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined on Location |
Location.
Description
Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
Outgoing Group ID
Default = Auto-filled. Range 0 to 99999.
Short codes that specify a number to dial also specify the line group to be used. The
system will then seize a line with a matching Outgoing Group ID. The same ID can
be used for multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must also
be unique within the network, ie. the same ID cannot be used in the configuration of
any lines on another server system in the network. The IDs 99999 and 99998 which
are reserved for the H.323 lines to the Primary Server and Secondary Server
respectively. The IDs 99901 to 99930 are reserved for the H.323 lines from the
Primary Server to each expansion system in the network. The ID 0 cannot be used
in a Server Edition network.
Number of Channels
Default = 20, Range 0 to 250; 0 to 500 for Server Edition Select systems.
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Line
Field
Description
Defines the number of operational channels that are available on this line.
Outgoing Channels
Default = 20, Range 0 to 250; 0 to 500 for Server Edition Select systems..
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.
TEI
Default = 0. Range = 0 to 127.
The Terminal Equipment Identifier. Used to identify each Control Unit connected to
a particular ISDN line. For Point to Point lines this is typically (always) 0. It can also
be 0 on a Point to Multi-Point line, however if multiple devices are actually sharing a
Point to Multi-Point line it should be set to 127 which will result in the exchange
deciding on the TEI's to be used by this Control Unit.
Related links
H.323 Line on page 333
Line | H.323 Short Codes
For some types of line, Line short codes can be applied to any digits received with incoming calls.
The line Short Code tab is shown for the following trunk types which are treated as internal or
private trunks: QSIG (T1, E1, H.323), BRI S0, H.323, SCN, IP Office. Incoming calls on those types
of trunk are not routed using Incoming Call Route settings. Instead the digits received with
incoming calls are checked for a match as follows:
Extension number (including remote numbers in a multi-site network).
• Line short codes (excluding ? short code).
• System short codes (excluding ? short code).
• Line ? short code.
• System ? short code.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Related links
H.323 Line on page 333
Line | H.323 VoIP Settings
These settings have changed in release 9.1. View the 9.0 settings on page 338.
This form is used to configure the VoIP setting applied to calls on the H.323 line.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
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Configuration Mode Field Descriptions
Field
Description
Gateway IP Address
Default = Blank
Enter the IP address of the gateway device at the remote end.
Port
Default = 1720
The H.323 line is identified by the IP Address:Port value. Specifying a unique port
value for this IP address allows multiple lines to use the same IP address.
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
Supplementary
Services
Default = H450.
Selects the supplementary service signaling method for use across the H.323 trunk.
The remote end of the trunk must support the same option. The options are:
• None: No supplementary services are supported.
• H450: Use for H.323 lines connected to another PBX or device that uses H450.
• QSIG: Use for H.323 lines connected to another PBX or device that uses QSIG.
Call Initiation Timeout
Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
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Line
Field
Description
VoIP Silence
Suppression
Default = Off.
Enable FastStart for
non-Avaya IP Phones
Default = Off
Fax Transport Support
Default = Off
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP
lines using G.711 between systems. On trunk's between networked systems, the
same setting should be set at both ends.
A fast connection procedure. Reduces the number of messages that need to be
exchanged before an audio channel is created.
This option is only supported on trunks with their Supplementary Services set to
IP Office SCN or IP Office Small Community Network - Fallback. Fax relay is
supported across H.323 multi-site network lines with Fax Transport Support
selected. This will use 2 VCM channels in each of the systems. Fax relay is only
supported on IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or IP500
Combo cards. Fax relay is not supported on Server Edition Linux servers.
Local Tones
Default = Off
When selected, the tones are generated by the local system to which the phone is
registered. This option should not be used with lines being used for a multi-site
network.
DTMF Support
Default = Out of Band
DTMF tones can be sent to the remote end either as DTMF tones within the calls
audio path (In Band) or a separate signals (Out of Band). Out of Band is
recommended for compression modes such as G.729 and G.723 compression
modes where DTMF in the voice stream could become distorted.
Allow Direct Media Path
Default = On
This settings controls whether IP calls must be routed via the system or can be
routed alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
Media. Enabling this option may cause some vendors problems with changing the
media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the
system. RTP relay support allows calls between devices using the same audio
codec to not require a voice compression channel.
Progress Ends Overlap
Send
Default = Off.
Some telephony equipment, primarily AT&T switches, over IP trunks send a H.323
Progress rather than H.323 Proceeding message to signal that they have
recognized the digits sent in overlap state. By default the system expects an H.323
Proceeding message. This option is not available by default. If required, the value
ProgressEndsOverlapSend must be entered into the Source Numbers tab of the
NoUser user.
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Configuration Mode Field Descriptions
Field
Description
Default Name From
Display IE
Default = Off.
When set, the Display IE is used as the default source for the name.
Related links
H.323 Line on page 333
Line | H.323 VoIP Settings (Release 9.0) on page 338
Line | H.323 VoIP Settings (Release 9.0)
This form is used to configure the VoIP setting applied to calls on the H.323 line.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
The VoIP settings available for an H.323 trunk depend on the Supplementary Services setting.
• Standard H.323 line VoIP Settings are applicable to trunks with their Supplementary Services
set to H450, QSIG or None.
• SCN H.323 line VoIP Settings are applicable to H.323 trunks with their Supplementary
Services set to IP Office SCN or IP Office SCN - Fallback.
Related links
Line | H.323 VoIP Settings on page 335
Standard H.323 VoIP Settings (Release 9.0) on page 338
SCN H.323 VoIP Settings (Release 9.0) on page 341
Standard H.323 VoIP Settings (Release 9.0)
The following settings are applicable to trunks with their Supplementary Services set to other than
IP Office SCN or IP Office Small Community Network - Fallback.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Gateway IP Address
Default = Blank
Enter the IP address of the gateway device at the remote end.
Port
Default = 1720
The H.323 line is identified by the IP Address:Port value. Specifying a unique port
value for this IP address allows multiple lines to use the same IP address.
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
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Line
Field
Description
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
Supplementary Services
Default = IP Office SCN Selects the supplementary service signaling method for
use across the H.323 trunk. The remote end of the trunk must support the same
option. The options are:
• None: No supplementary services are supported.
• H450: Use for H.323 lines connected to another PBX or device that uses H450.
• QSIG: Use for H.323 lines connected to another PBX or device that uses QSIG.
• IP Office SCN: This option is used for H.323 trunks within a multi-site network.
The systems within a multi-site network automatically exchange information about
users and extensions, allowing remote users to be called without any additional
configuration on the local system. For full details of Small Community Network
operation see Small Community Networking. For full details of a Server Edition
network, see Server Edition Mode.
• IP Office SCN - Fallback: This option is used for a multi-site network trunk
connection as above, where the system at the end of the trunk will try to take over
the selected SCN Backup Options if this system is not visible within the multi-site
network for a period of more than 3 minutes. See Small Community Network
Fallback.
Note that both ends of the SCN trunk connection must be set to fallback.
On the system requesting backup, the required SCN Backup Options are selected,
indicating that it is requesting backup. A single system can only request backup
from one other system.
A system providing backup can provide backup for up to 7 other systems.
Location
Default = Cloud.
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Configuration Mode Field Descriptions
Field
Description
Specify a location to associate the extension with a physical location. Associating
an extension with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined in the Location
form.
Call Initiation Timeout
Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
VoIP Silence
Suppression
Default = Off
Enable FastStart for nonAvaya IP Phones
Default = Off
Fax Transport Support
Default = Off
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP
lines using G.711 between systems. On trunk's between networked systems, the
same setting should be set at both ends.
A fast connection procedure. Reduces the number of messages that need to be
exchanged before an audio channel is created.
This option is only supported on trunks with Supplementary Services set to IP
Office SCN or IP Office Small Community Network - Fallback. Fax relay is
supported across H.323 multi-site network lines with Fax Transport Support
selected. This will use 2 VCM channels in each of the systems. Fax relay is only
supported on IP500/IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or
IP500 Combo cards. Fax relay is not supported on Server Edition Linux servers.
Local Tones
Default = Off
When selected, the tones are generated by the local system to which the phone is
registered. This option should not be used with lines being used for a multi-site
network.
DTMF Support
Default = Out of Band
DTMF tones can be sent to the remote end either as DTMF tones within the calls
audio path (In Band) or a separate signals (Out of Band). Out of Band is
recommended for compression modes such as G.729 and G.723 compression
modes where DTMF in the voice stream could become distorted.
For trunks with Supplementary Services set to IP Office SCN or IP Office SCN Fallback, this option is fixed to Out of Band.
Allow Direct Media Path
Default = On
This settings controls whether IP calls must be routed via the system or can be
routed alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
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Field
Description
Media and be using the same protocol (H.323 or SIP). Enabling this option may
cause some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the
system. RTP relay support allows calls between devices using the same audio
codec to not require a voice compression channel.
Progress Ends Overlap
Send
Default = Off.
Default Name From
Display IE
Default = Off.
Some telephony equipment, primarily AT&T switches, over IP trunks send a H.323
Progress rather than H.323 Proceeding message to signal that they have
recognized the digits sent in overlap state. By default the system expects an H.323
Proceeding message. This option is not available by default. If required, the value
ProgressEndsOverlapSend must be entered into the Source Numbers tab of the
NoUser user.
When set, the Display IE is used as the default source for the name.
Related links
Line | H.323 VoIP Settings (Release 9.0) on page 338
SCN H.323 VoIP Settings (Release 9.0)
The following settings are applicable to H.323 trunks with their Supplementary Services set to IP
Office SCN or IP Office SCN - Fallback.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Server Edition Network Usage
In a Server Edition network, all systems are linked in a star topography to the the Primary Server,
using an H.323 IP trunk with its Outgoing Group ID set to 99999. If the network also includes a
Secondary Server, all systems are also linked to the Secondary Server using an H.323 IP trunk with
its Outgoing Group IP set to 99998.
These are the only H.323 IP trunks supported within the Server Edition network.
Field
Description
Gateway IP Address
Default = Blank
Enter the IP address of the gateway device at the remote end. This address must
not be shared by any other IP line (H.323, SIP, SES or IP DECT).
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
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Configuration Mode Field Descriptions
Field
Description
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
Supplementary Services
Default = IP Office SCN Selects the supplementary service signaling method for
use across the H.323 trunk. The remote end of the trunk must support the same
option. The options are:
• None: No supplementary services are supported.
• H450: Use for H.323 lines connected to another PBX or device that uses H450.
• QSIG: Use for H.323 lines connected to another PBX or device that uses QSIG.
• IP Office SCN: This option is used for H.323 trunks within a multi-site network.
The systems within a multi-site network automatically exchange information
about users and extensions, allowing remote users to be called without any
additional configuration on the local system. For full details of Small Community
Network operation see Small Community Networking. For full details of a Server
Edition network, see Server Edition Mode.
• IP Office SCN - Fallback: This option is used for a multi-site network trunk
connection as above, where the system at the end of the trunk will try to take
over the selected SCN Backup Options if this system is not visible within the
multi-site network for a period of more than 3 minutes. See Small Community
Network Fallback.
Note that both ends of the SCN trunk connection must be set to fallback.
On the system requesting backup, the required SCN Backup Options are
selected, indicating that it is requesting backup. A single system can only request
backup from one other system.
A system providing backup can provide backup for up to 7 other systems.
SCN Backup Options
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Field
Description
These options are only available when the Supplementary Services option is set to IP Office - Fallback. The
intention of this feature is to attempt to maintain a minimal level of operation while problems with the local
system are resolved.
Backs up my IP Phones
Default = On.
This option is used for Avaya 1600, 4600, 5600 and 9600 Series phones registered
with the system. When selected, it will share information about the registered
phones and users on those phones with the other system.
If the local system is no longer visible to the phones, the phones will reregister with
the other system. The users who were currently on those phones will appear on the
other system as if they had hot desked.
Note that when the local system is restored to the network, the phones will not
automatically re-register with it. A phone reset via either a phone power cycle or
using the System Status Application is required.
When phones have registered with the other system, they will show an R on their
display.
If using resilience backup to support Avaya IP phones, Auto-create Extn and
Auto-create User should not be left enabled after initial configuration or any
subsequent addition of new extensions and users. Leaving auto-create options
enabled on a system that is a failover target may cause duplicate extension/user
records on the multi-site network under multiple failure scenarios.
Backs up my Hunt Groups Default = On.
When selected, any hunt groups the local system is advertising to the network are
advertised from the other system when fallback is required. The trigger for this
occurring is Avaya H.323 phones registered with the local system registering with
the other system, ie. Backs up my IP Phones above must also be enabled. In a
Server Edition network this option is only available on the H.323 trunk from the
Primary Server to the Secondary Server.
When used, the only hunt group members that will be available are as follows:
• If the group was a distributed hunt group, those members who were remote
members on other systems still visible within the network.
• Any local members who have hot desked to another system still visible within the
network.
When the local system becomes visible to the other system again, the groups will
return to be advertised from the local system.
Backs up my Voicemail
Default = On.
This option can be used if the local system is hosting the Voicemail Pro server
being used by the network. If selected, when the local system is no longer visible to
the voicemail server, the other system will act as host for the voicemail server. In a
Server Edition network this option is only available on the H.323 trunk from the
Primary Server to the Secondary Server. It is assumed as being on an is
automatically set by the Resilience Administration tool.
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Configuration Mode Field Descriptions
Field
Description
The option:
• requires the other system to have licenses for the Voicemail Pro features that are
required to operate during any fallback period.
• requires Voicemail Pro 5.0+.
Call Initiation Timeout
Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
VoIP Silence Suppression
Default = Off.
When selected, this option will detect periods of silence on any call over the line
and will not send any data during those silent periods. This feature is not used on
IP lines using G.711 between systems. On trunk's between networked systems, the
same setting should be set at both ends.
Fax Transport Support
Default = Off
This option is only supported on trunks with their Supplementary Services set to
IP Office SCN or IP Office Small Community Network - Fallback. Fax relay is
supported across H.323 multi-site network lines with Fax Transport Support
selected. This will use 2 VCM channels in each of the systems. Fax relay is only
supported on IP500/IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or
IP500 Combo cards. Fax relay is not supported on Server Edition Linux servers.
Local Tones
Default = Off.
When selected, the tones are generated by the local system to which the phone is
registered. This option should not be used with lines being used for a multi-site
network.
DTMF Support
Default = Out of Band
DTMF tones can be sent to the remote end either as DTMF tones within the calls
audio path (In Band) or a separate signals (Out of Band). Out of Band is
recommended for compression modes such as G.729 and G.723 compression
modes where DTMF in the voice stream could become distorted.
For trunks with Supplementary Services set to IP Office SCN or IP Office SCN Fallback, this option is fixed to Out of Band.
Allow Direct Media Path
Default = On
This settings controls whether IP calls must be routed via the system or can be
routed alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes
the need for a voice compression channel. Both ends of the calls must support
Direct Media and be using the same protocol (H.323 or SIP). Enabling this option
may cause some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the
system. RTP relay support allows calls between devices using the same audio
codec to not require a voice compression channel.
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Related links
Line | H.323 VoIP Settings (Release 9.0) on page 338
IP DECT Line
This type of line can be manually added. They are used to route voice calls over an IP data
connection to an Avaya IP DECT system. Only one IP DECT line can be added to a system. Refer
to the IP DECT R4 Installation manual for full details.
Currently, only one IP DECT line is supported on a system.
Related links
Line on page 289
Line | IP DECT Line on page 345
Line | IP DECT Gateway on page 346
Line | IP DECT VoIP on page 350
Line | IP DECT Line
These settings have changed in release 9.1. View the 9.0 settings. on page 345
When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. The following actions are not meregable:
• Changing an existing IP DECT line.
• Changing an IP DECT line that has been imported into the configuration.
Field
Description
Line Number
This number is allocated by the system and is not adjustable.
Associated Extensions
Lists all the DECT extensions associated with the IP DECT line.
Description
Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
Related links
IP DECT Line on page 345
Line | IP DECT Line (9.0) on page 345
Line | IP DECT Line (9.0)
These settings are not mergeable. Changes to these settings will require a reboot of the system.
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Configuration Mode Field Descriptions
Field
Description
Line Number
This number is allocated by the system and is not adjustable.
Associated Extensions
Lists all the DECT extensions associated with the IP DECT line.
Related links
Line | IP DECT Line on page 345
Line | IP DECT Gateway
These settings have changed in release 9.1. View the release 9.0 settings. on page 348
This form is used to configure aspects of information exchange between the IP Office and IP DECT
systems.
When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. The following actions are not meregable:
• Changing an existing IP DECT line.
• Changing an IP DECT line that has been imported into the configuration.
Field
Description
Auto-Create
Extension
Default = Off.
If enabled, subscription of a handset with the DECT system causes the auto-creation of a
matching numbered extension within the system configuration if one does not already
exist.
For security, any auto-create settings set to On are automatically set to Off after 24 hours.
Auto-Create User
Default = Off.
This option is only usable if Auto-Create Extension is also enabled. If enabled,
subscription of a handset with the DECT system causes the auto-creation of a matching
user within the system configuration if one does not already exist.
For security, any auto-create settings set to On are automatically set to Off after 24 hours.
Enable DHCP
Support
Default = Off
This option is not supported for use with Avaya IP DECT R4. The IP DECT base stations
require DHCP and TFTP support. Enable this option if the system is being used to
provide that support, using IP addresses from its DHCP range (LAN1 or LAN2) and its
TFTP server setting. If not enabled, alternate DHCP and TFTP options must be provided
during the IP DECT installation.
• If it is desired to use the system for DHCP support of the ADMM and IP DECT base
stations only, the system address range should be set to match that number of
addresses. Those addresses are then taken during the system restart and will not be
available for other DHCP responses following the restart.
• For larger IP DECT installations, the use of a non-embedded TFTP software option
other than Manager is recommended.
Boot File
Default = ADMM_RFP_1_1_13.tftp. Range = Up to 31 characters.
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Field
Description
The name and path of the ADMM software file. The path is relative to the TFTP server
root directory.
ADMM MAC
Address
Default = 00:00:00:00:00:00
VLAN ID
Default = Blank. Range = 0 to 4095.
This field must be used to indicate the MAC address of the IP DECT base station that
should load the ADMM software file and then act as the IP DECT system's ADMM. The
address is entered in hexadecimal format using comma, dash, colon or period separators.
If VLAN is being used by the IP DECT network, this field sets the VLAN address assigned
to the base stations by the system if Enable DHCP Support is selected.
• The system itself does not apply or use VLAN marking. It is assumed that the addition
of VLAN marking and routing of VLAN traffic is performed by other switches within the
customer network.
• An ID of zero is not recommended for normal VLAN operation.
• When blank, no VLAN option is sent to the IP DECT base station.
Base Station
Address List
Default = Empty
This box is used to list the MAC addresses of the IP DECT base stations, other than the
base station being used as the ADMM and entered in the ADMM MAC Address field.
Right-click on the list to select Add or Delete. or use the Insert and Delete keys. The
addresses are entered in hexadecimal format using comma, dash, colon or period
separators.
Enable Provisioning
This option can be used with DECT R4 systems. It allows the setting of several values in the system
configuration that previously needed to be set separately in the master base stations configuration. For full
details refer to the DECT R4 Installation manual. The use of provisioning requires the system security settings
to include an IPDECT Group.
SARI/PARK
Default = 0
Enter the PARK (Portable Access Rights Key) license key of the DECT R4 system. DECT
handset users enter this key when subscribing to the DECT system.
Subscriptions
Default = Disabled
Select the method of subscription supported for handsets subscribing to the DECT R4
system. The options are:
• Disabled:Disables subscription of handsets.
• Auto-Create: Allow anonymous subscription of handsets. Once subscribed, the
handset is assigned a temporary extension number. That extension number can be
confirmed by dialing *#. A new extension number can be specified by dialing
<Extension Number>*<Login Code>#. The Auto-Create Extension and Auto-Create
User settings above should also be enabled. While configured to this mode, Manager
will not allow the manual addition of new IP DECT extensions.
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Configuration Mode Field Descriptions
Field
Description
• Preconfigured: Allow subscription only against existing IP DECT extensions records in
the system configuration. The handset IPEI number is used to match the subscribing
handset to a system extension.
Authentication
Code
Default = Blank.
Set an authentication code that DECT handset users should enter when subscribing to
the DECT system.
Enable Resiliency
Default = Off.
Enables resiliency on the IP DECT Line. To configure resiliency, you must also configure an IP Office Line with
Backs up my IP Dect Phones set to On.
Status Enquiry
Period
Default = 30 seconds.
Prioritize Primary
Default = Off.
The period between successive verifications on the H.323 channel. The smaller the
interval, the faster the IP DECT system recognizes that IP Office is down.
Only available when Enable Provisioning is set to On.
Set to On for automatic fail-over recovery. When on, the IP DECT system switches
automatically from the backup IP Office to the "primary" IP Office.
Note that the IP DECT system does not switch back automatically from the backup IP
Office to the primary. The IP DECT system must be manually switched using Web
Manager.
Supervision
Timeout
Default = 120 seconds.
Only available when Enable Provisioning is set to On.
The period of time the IP DECT system will wait between attempts to switch from the
backup IP Office to its "primary" IP Office.
Related links
IP DECT Line on page 345
Line | IP DECT Gateway (9.0) on page 348
Line | IP DECT Gateway (9.0)
This form is used to configure aspects of information exchange between the IP Office and IP DECT
systems.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Auto-Create
Extension
Default = Off.
If enabled, subscription of a handset with the DECT system causes the auto-creation of a
matching numbered extension within the system configuration if one does not already
exist.
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Field
Description
Auto-Create User
Default = Off.
This option is only usable if Auto-Create Extension is also enabled. If enabled,
subscription of a handset with the DECT system causes the auto-creation of a matching
user within the system configuration if one does not already exist.
Enable DHCP
Support
Default = Off
This option is not supported for use with Avaya IP DECT R4. The IP DECT base stations
require DHCP and TFTP support. Enable this option if the system is being used to
provide that support, using IP addresses from its DHCP range (LAN1 or LAN2) and its
TFTP server setting. If not enabled, alternate DHCP and TFTP options must be provided
during the IP DECT installation.
• If it is desired to use the system for DHCP support of the ADMM and IP DECT base
stations only, the system address range should be set to match that number of
addresses. Those addresses are then taken during the system restart and will not be
available for other DHCP responses following the restart.
• For larger IP DECT installations, the use of a non-embedded TFTP software option
other than Manager is recommended.
Boot File
Default = ADMM_RFP_1_0_0.tftp. Range = Up to 31 characters.
The name and path of the ADMM software file. The path is relative to the TFTP server
root directory.
ADMM MAC
Address
Default = 00:00:00:00:00:00
VLAN ID
Default = Blank. Range = 0 to 4095.
This field must be used to indicate the MAC address of the IP DECT base station that
should load the ADMM software file and then act as the IP DECT system's ADMM. The
address is entered in hexadecimal format using comma, dash, colon or period separators.
If VLAN is being used by the IP DECT network, this field sets the VLAN address assigned
to the base stations by the system if Enable DHCP Support is selected.
• The system itself does not apply or use VLAN marking. It is assumed that the addition
of VLAN marking and routing of VLAN traffic is performed by other switches within the
customer network.
• An ID of zero is not recommended for normal VLAN operation.
• When blank, no VLAN option is sent to the IP DECT base station.
Base Station
Address List
Enable
Provisioning
Default = Empty
This box is used to list the MAC addresses of the IP DECT base stations, other than the
base station being used as the ADMM and entered in the ADMM MAC Address field.
Right-click on the list to select Add or Delete. or use the Insert and Delete keys. The
addresses are entered in hexadecimal format using comma, dash, colon or period
separators.
This option can be used with DECT R4 systems. It allows the setting of several values in
the system configuration that previously needed to be set separately in the master base
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Configuration Mode Field Descriptions
Field
Description
stations configuration. For full details refer to the DECT R4 Installation manual. The use
of provisioning requires the system security settings to include an IPDECT Group.
SARI/PARK
Default = 0
Enter the PARK (Portable Access Rights Key) license key of the DECT R4 system. DECT
handset users enter this key when subscribing to the DECT system.
Subscriptions
Default = Disabled
Select the method of subscription supported for handsets subscribing to the DECT R4
system. The options are:
• Disabled:Disables subscription of handsets.
• Auto-Create: Allow anonymous subscription of handsets. Once subscribed, the
handset is assigned a temporary extension number. That extension number can be
confirmed by dialing *#. A new extension number can be specified by dialing
<Extension Number>*<Login Code>#. The Auto-Create Extension and Auto-Create
User settings above should also be enabled. While configured to this mode, Manager
will not allow the manual addition of new IP DECT extensions.
• Preconfigured: Allow subscription only against existing IP DECT extensions records in
the system configuration. The handset IPEI number is used to match the subscribing
handset to a system extension.
Authentication
Code
Default = Blank.
Set an authentication code that DECT handset users should enter when subscribing to
the DECT system.
Related links
Line | IP DECT Gateway on page 346
Line | IP DECT VoIP
These settings have changed in release 9.1. View the release 9.0 settings. on page 352
This form is used to configure the VoIP setting applied to calls on the IP DECT line.
When creating an IP DECT line, these settings are mergeable. You can also remove an IP DECT
line without rebooting. The following actions are not meregable:
• Changing an existing IP DECT line.
• Changing an IP DECT line that has been imported into the configuration.
Field
Description
Gateway IP Address
Default = Blank.
Enter the IP address of the gateway device at the remote end. This address must
not be shared by any other IP line (H.323, SIP, SES or IP DECT).
Standby IP Address
Default = Blank.
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Field
Description
IP Address of the Standby Master IP Base Station or the second Mirror Base
Station. When the primary Mirror Base Station or Master Base Station is offline the
second Mirror or the Standby Master will take over and the system will use this IP
address.
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to
change the order of the selected codecs.
TDM | IP Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
VoIP Silence
Suppression
Default = Off.
Allow Direct Media Path
Default = On
When selected, this option will detect periods of silence on any call over the line
and will not send any data during those silent periods. This feature is not used on
IP lines using G.711 between systems. On trunk's between networked systems,
the same setting should be set at both ends.
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Configuration Mode Field Descriptions
Field
Description
This settings controls whether IP calls must be routed via the system or can be
routed alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes
the need for a voice compression channel. Both ends of the calls must support
Direct Media and be using the same protocol (H.323 or SIP). Enabling this option
may cause some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the
system. RTP relay support allows calls between devices using the same audio
codec to not require a voice compression channel.
Related links
IP DECT Line on page 345
Line | IP DECT VoIP (9.0) on page 352
Line | IP DECT VoIP (9.0)
This form is used to configure the VoIP setting applied to calls on the IP DECT line.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Gateway IP Address
Default = Blank.
Enter the IP address of the gateway device at the remote end. This address must
not be shared by any other IP line (H.323, SIP, SES or IP DECT).
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
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Field
Description
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to
change the order of the selected codecs.
TDM | IP Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
VoIP Silence
Suppression
Default = Off.
Allow Direct Media Path
Default = On
When selected, this option will detect periods of silence on any call over the line
and will not send any data during those silent periods. This feature is not used on
IP lines using G.711 between systems. On trunk's between networked systems,
the same setting should be set at both ends.
This settings controls whether IP calls must be routed via the system or can be
routed alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes
the need for a voice compression channel. Both ends of the calls must support
Direct Media and be using the same protocol (H.323 or SIP). Enabling this option
may cause some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the
system. RTP relay support allows calls between devices using the same audio
codec to not require a voice compression channel.
Related links
Line | IP DECT VoIP on page 350
SIP Line
IP Office supports SIP voice calls through the addition of SIP lines to the system configuration. This
approach allows users with non-SIP phones to make and receive SIP calls.
For information on SIP line specifications and a procedure for configuring a SIP line, see
Configuring SIP Trunks on page 725.
Related links
Line on page 289
Line | SIP Line on page 354
Line | SIP Transport on page 363
Line | SIP URI on page 366
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Configuration Mode Field Descriptions
Line | SIP VoIP on page 368
Line | SIP T38 Fax on page 374
Line | SIP Credentials on page 375
Line | SIP Advanced on page 376
Line | SIP Engineering on page 380
Line | SIP Line
These settings have changed in release 9.1. View the release 9.0 settings on page 358.
These settings are mergeable with the exception of the Line Number setting. Changing the Line
Number setting requires a “merge with service disruption”. When the configuration file is sent to the
system, the SIP trunk is restarted and all calls on the line are dropped.
Deleting a SIP line requires a “merge with service disruption”.
Field
Description
Line Number
Default = Automatically assigned.
By default a value is assigned by the system. This value can be changed but it
must be unique.
ITSP Domain Name
Default = Blank.
This field is used to specify the default host part of the SIP URI in the From, To,
and R-URI fields for outgoing calls. For example, in the SIP URI
name@example.com, the host part of the URI is example.com. When empty, the
default host is provided by the Transport | ITSP Proxy Address field value. If
multiple addresses are defined in the Transport | ITSP Proxy Address field, then
this field must be defined.
For the user making the call, the user part of the From SIP URI is determined by
the settings of the SIP URI channel record being used to route the call (see SIP
URI | Local URI). This will use one of the following:
• a specific name entered in Local URI field of the channel record.
• or specify using the primary or secondary authentication name set for the line
below.
• or specify using the SIP Name set for the user making the call (User | SIP | SIP
Name).
For the destination of the call, the user part of the To and R-URI fields are
determined by dial short codes of the form 9N/N"@example.com” where N is the
user part of the SIP URI and "@example.com" is optional and can be used to
override the host part of the To and R-URI.
URI
Default = SIP.
When SIP or SIPS is selected in the drop-down box, the SIP URI format is used
(for example, name@example.com).
When Tel is selected in the drop-down box, the Tel URI format is used (for
example, +1-425-555-4567). This affects the From field of outgoing calls. The To
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Field
Description
field for outgoing calls will always use the format specified by the short codes used
for outgoing call routing. Recommendation: When SIP Secured URI is required,
the URI Type should be set to SIPS. SIPS can be used only when Layer 4 Protocol
is set to TLS.
Location
Default = Cloud.
Specify a location to associate the line with a physical location. Associating a line
with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined in the System |
Location form.
Prefix
Default = Blank.
This prefix is removed from the called number on outgoing calls if present.
National Prefix
Default = 0.
This prefix is added to calls identified as not being international.
International Prefix
Default = 00.
This prefix is added to calls identified as not being national.
Country Code
Default = Blank.
Set to match the local country code of the system location.
Name Priority
Default = System Default.
For SIP trunks, the caller name displayed on an extension can either be that
supplied by the trunk or one obtained by checking for a number match in the
extension user's personal directory and the system directory. This setting
determines which method is used by the line. The options are:
• System Default: Use the system's Default Name Priority setting (System |
Telephony | Telephony).
• Favour Trunk: Display the name provided by the trunk. For example, the trunk
may be configured to provide the calling number or the name of the caller. The
system should display the caller information as it is provided by the trunk. If the
trunk does not provide a name, the system uses the Favour Directory method.
• Favour Directory: Search for a number match in the extension user's personal
directory and then in the system directory. The first match is used and overrides
the name provided by the SIP line. If no match is found, the name provided by
the line, if any, is used.
Description
Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
Network Type
Default = Public.
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Configuration Mode Field Descriptions
Field
Description
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
In Service
Default = On.
When this field is not selected, the SIP trunk is unregistered and not available to
incoming and outgoing calls.
Check OOS
Default = On.
If enabled, the system will regularly check if the trunk is in service using the
methods listed below. Checking that SIP trunks are in service ensures that
outgoing call routing is not delayed waiting for response on a SIP trunk that is not
currently usable.
For UDP and TCP trunks, OPTIONS message are regularly sent. If no reply to an
OPTIONS message is received the trunk is taken out of service.
For TCP trunks, if the TCP connection is disconnected the trunk will be taken out
of service.
For trunks using DNS, if the IP address is not resolved or the DNS resolution has
expired, the trunk is taken out of service.
Session Timers
Refresh Method
Default = Auto.
The options are:
• Auto
• Reinvite
• Update
When Auto is selected, if UPDATE is in the Allow: header from the far SIP
endpoint, then it is used. Otherwise INVITE is used.
Timer (seconds)
Default = On Demand. Range = 90 to 64800
This field specifies the session expiry time. At the half way point of the expiry time,
a session refresh message is sent. When set to On Demand, IP Office will not
send a session refresh message but will respond to them.
Forwarding and Twinning
Originator number
Default = Blank.
This field can be used to set an originator number for forwarded and twinned calls
when using any of the Send Caller ID options above other than None. If exported
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Field
Description
or imported as part of a trunk template, this setting is not supported by Basic
Edition systems.
Send Caller ID
Default = None.
Set the type of header for to use for the originating calling party information. The
options are:
• Diversion Header: Use the information from the Diversion Header.
• Remote Party ID: Use the Remote Part ID.
• P Asserted ID: Use the contact information from the P Asserted ID.
• None: This option corresponds to the ISDN withheld setting.
This setting is also used for forwarded calls. Note that the values on the System |
Twinning tab override this if set. For incoming calls to a hunt group, the hunt
group details will be provided and not the details of the answering agent. This
setting is mergeable.
The SIP line Send Caller ID setting takes priority.
Redirect and Transfer
Redirection and blind transfer are configured separately. By default, they are disabled.
A supervised transfer occurs when a consultation call is made and the REFER contains a Replaces: header
indicating the CallID of another call leg which the REFERing agent has already initiated with the REFER target.
Note:
Do not change these settings unless directed to by the SIP service provider.
Incoming Supervised
REFER
Default = Auto.
Determines if IP Office will accept a REFER being sent by the far end. The options
are:
• Always: Always accepted.
• Auto: If the far end does not advertise REFER support in the Allow: header of
the OPTIONS responses, then IP Office will reject a REFER from that endpoint.
• Never: Never accepted.
Outgoing Supervised
REFER
Default = Auto.
Determines if IP Office will attempt to use the REFER mechanism to transfer a
party to a call leg which IP Office has already initiated so that it can include the
CallID in a Replaces: header. The options are:
• Always: Always use REFER.
• Auto: Use the Allow: header of the OPTIONS response to determine if the
endpoint supports REFER.
• Never: Never use REFER.
Send 302 Moved
Temporarily
Default = Off.
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Configuration Mode Field Descriptions
Field
Description
A SIP response code used for redirecting an unanswered incoming call. It is a
response to the INVITE, and cannot be used after the 200 OK has been sent as a
response to the INVITE.
Outgoing Blind REFER
Default = Off.
When enabled, a user, voicemail system or IVR can transfer a call by sending a
REFER to an endpoint that has not set up a second call. In this case, there is no
Replaces: header because there is no CallID to replace the current one. This
directs the far end to perform the transfer by initiating the new call and release the
current call with IP Office.
Related links
SIP Line on page 353
Line | SIP Line (Release 9.0) on page 358
Line | SIP Line (Release 9.0)
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Line Number
Default = Automatically assigned.
By default a value is assigned by the system. This value can be changed but it
must be unique.
ITSP Domain Name
Default = Blank.
This field is used to specify the default domain part of the SIP URI in the From, To
and R-URI fields for outgoing calls. For example, in the SIP URI
name@example.com, the domain part of the URI is example.com. When empty,
the default domain is provided by the Transport | ITSP Proxy Address field
value. If multiple addresses are defined in the Transport | ITSP Proxy Address
field, then this field must be defined.
For the user making the call, the user part of the From SIP URI is determined by
the settings of the SIP URI channel record being used to route the call. This will
use one of the following:
• a specific name entered in Local URI field of the channel record.
• or specify using the primary or secondary authentication name set for the line
below.
• or specify using the SIP Name set for the user making the call (User | SIP | SIP
Name).
For the destination of the call, the user part of the To and R-URI fields are
determined by dial short codes of the form 9N/N"@example.com” where N is the
user part of the SIP URI and "@example.com" is optional and can be used to
override the domain in To and R-URI.
Prefix
Default = Blank.
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Field
Description
This prefix is removed from the called number on outgoing calls if present.
National Prefix
Default = 0.
This prefix is added to calls identified as not being international.
Country Code
Default = Blank.
Set to match the local country code of the system location.
International Prefix
Default = 00.
This prefix is added to calls identified as not being national.
Send Caller ID
Default = None.
Select which value the SIP line should use for the original calling party ID when
routing twinned calls. The options are:
• Diversion Header: Use the information from the Diversion Header.
• Remote Party ID: Use the Remote Part ID.
• P Asserted ID: Use the contact information from the P Asserted ID.
• None: This option corresponds to the ISDN withheld setting.
This setting is also used for forwarded calls. Note that the values on the System |
Twinning tab override this if set. For incoming calls to a hunt group, the hunt
group details will be provided and not the details of the answering agent. This
setting is mergeable.
The SIP line Send Caller ID setting takes priority.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Association Method
Default = By Source IP Address.
This setting sets the method by which a SIP line is associated with an incoming
SIP request.
The match criteria used for each line can be varied. The search for a line match for
an incoming request is done against each line in turn using each lines Association
Method. The order of line matching uses the configured Line Number settings
until a match occurs. If no match occurs the request is ignored. This method allows
multiple SIP lines with the same address settings. This may be necessary for
scenarios where it may be required to support multiple SIP lines to the same ITSP.
For example when the same ITSP supports different call plans on separate lines or
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Configuration Mode Field Descriptions
Field
Description
where all outgoing SIP lines are routed from the system via an additional on-site
system. The options are:
• By Source IP Address: This option uses the source IP address and port of the
incoming request for association. The match is against the configured remote
end of the SIP line, using either an IP address/port or the resolution of a fully
qualified domain name.
• "From" header hostpart against ITSP domain: This option uses the host part
of the From header in the incoming SIP request for association. The match is
against the ITSP Domain Name above.
• R-URI hostpart against ITSP domain: This option uses the host part of the
Request-URI header in the incoming SIP request for association. The match is
against the ITSP Domain Name above.
• "To" header hostpart against ITSP domain: This option uses the host part of
the To header in the incoming SIP request for association. The match is against
the ITSP Domain Name above.
• "From" header hostpart against DNS-resolved ITSP domain: This option
uses the host part of the FROM header in the incoming SIP request for
association. The match is found by comparing the FROM header against a list of
IP addresses resulting from resolution of the ITSP Domain Name above or, if
set, the ITSP Proxy Address on the Transport tab.
• "Via" header hostpart against DNS-resolved ITSP domain: This option uses
the host part of the VIA header in the incoming SIP request for association. The
match is found by comparing the VIA header against a list of IP addresses
resulting from resolution of the ITSP Domain Name above or, if set, the ITSP
Proxy Address on the Transport tab.
• "From" header hostpart against ITSP proxy: This option uses the host part of
the “From” header in the incoming SIP request for association. The match is
against the ITSP Proxy Address on the Transport tab.
• "To" header hostpart against ITSP proxy: This option uses the host part of the
From header in the incoming SIP request for association. The match is against
the ITSP Proxy Address on the Transport tab.
• R-URI hostpart against ITSP proxy: This option uses the host part of the
Request-URI in the incoming SIP request for association. The match is against
the ITSP Proxy Address on the Transport tab.
REFER Support
Default = On.
REFER is the method used by many SIP device, including SIP trunks, to transfer
calls. These settings can be used to control whether REFER is used as the
method to transfer calls on this SIP trunk to another call on the same trunk. If
supported, once the transfer has been completed, the system is no longer involved
in the call. If not supported, the transfer may still be completed but the call will
continue to be routed via the system.
Incoming
Default = Auto
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Field
Description
Select whether REFER can or should be used when an attempt to transfer an
incoming call on the trunk results in an outgoing call on another channel on the
same trunk. The options are:
• Always: Always use REFER for call transfers that use this trunk for both legs of
the transfer. If REFER is not supported, the call transfer attempt is stopped.
• Auto: Request to use REFER if possible for call transfers that use this trunk for
both legs of the transfer. If REFER is not supported, transfer the call via the
system as for the Never setting below.
• Never: Do not use REFER for call transfers that use this trunk for both legs of
the transfer. The transfer can be completed but will use 2 channels on the trunk.
Outgoing
Default = Auto
Select whether REFER can or should be used when attempt to transfer an
outgoing call on the trunk results in an incoming call on another channel on the
same trunk. This uses system resources and may incur costs for the duration of
the transferred call. The options available are the same as for the Incoming
setting.
Method for Session
Refresh
Default = Auto.
The SIP UPDATE method (RFC 3311) allows a client to update parameters of a
session (such as the set of media streams and their codecs) but has no impact on
the state of a dialog. The options are:
• RE-INVITE Re-Invite messages are sent for session refresh.
• UPDATE UPDATE messages are sent for session refresh if the other end
indicates support for UPDATE in the allow header.
• Auto UPDATE messages are sent for session refresh if the other end indicates
support for UPDATE in the allow header. If UPDATE is not supported, REINVITE messages are sent.
Session Timer
Default = On Demand. Range = 90 to 64800
This field specifies the session expiry time. At the half way point of the expiry time,
a session refresh message is sent. Setting the field to On Demand disables the
session timer.
Media Connection
Preservation
Default = Disabled.
In Service
Default = On.
When enabled, allows established calls to continue despite brief network failures.
Call handling features are no longer available when a call is in a preserved state.
Preservation on public SIP trunks is not supported until tested with a specific
service provider.
When this field is not selected, the SIP trunk is unregistered and not available to
incoming and outgoing calls.
URI Type
Default = SIP.
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Configuration Mode Field Descriptions
Field
Description
When SIP or SIPS is selected in the drop-down box, the SIP URI format is used
(for example, name@example.com). When Tel is selected in the drop-down box,
the Tel URI format is used (for example, +1-425-555-4567). This affects the From
field of outgoing calls. The To field for outgoing calls will always use the format
specified by the short codes used for outgoing call routing. Recommendation:
When SIP Secured URI is required, the URI Type should be set to SIPS. SIPS can
be used only when Layer 4 Protocol is set to TLS.
Check OOS
Default = On.
If enabled, the system will regularly check if the trunk is in service using the
methods listed below. Checking that SIP trunks are in service ensures that
outgoing call routing is not delayed waiting for response on a SIP trunk that is not
currently usable.
For UDP and TCP trunks, OPTIONS message are regularly sent. If no reply to an
OPTIONS message is received the trunk is taken out of service.
For TCP trunks, if the TCP connection is disconnected the trunk will be taken out
of service.
For trunks using DNS, if the IP address is not resolved or the DNS resolution has
expired, the trunk is taken out of service.
Call Routing Method
Default = Request URI.
This field allows selection of which incoming SIP information should be used for
incoming number matching by the system's incoming call routes. The options are
to match either the Request URI or the To Header element provided with the
incoming call.
Originator number for
forwarded and twinning
calls
Name Priority
Default = Blank.
This field can be used to set a originator number for forwarded and twinned calls
when using any of the Send Caller ID options above other than None. If exported
or imported as part of a trunk template, this setting is not supported by Basic
Edition Quick mode systems.
Default = System Default.
For SIP trunks, the caller name displayed on an extension can either be that
supplied by the trunk or one obtained by checking for a number match in the
extension user's personal directory and the system directory. This setting
determines which method is used by the line. The options are:
• System Default: Use the system's Default Name Priority setting (System |
Telephony | Telephony).
• Favour Trunk: Display the name provided by the trunk. For example, the trunk
may be configured to provide the calling number or the name of the caller. The
system should display the caller information as it is provided by the trunk. If the
trunk does not provide a name, the system uses the Favour Directory method.
• Favour Directory: Search for a number match in the extension user's personal
directory and then in the system directory. The first match is used and overrides
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Line
Field
Description
the name provided by the SIP line. If no match is found, the name provided by
the line, if any, is used.
Caller ID From Header
Default = Off.
Incoming calls can include caller ID information in both the From field and in the
PAI fields. When this option is selected, the caller ID information in the From field
is used rather than that in the PAI fields.
Send From In Clear
Default = Off.
When selected, the user ID of the caller is included in the From field. This applies
even if the caller has selected to be or is configured to be anonymous, though their
anonymous state is honored in other fields used to display the caller identity.
User-Agent and Server
Headers
Default = Blank (Use system type and software level).
Service Busy Response
Default = 486 - Busy Here (503 - Service Unavailable for the France2 locale).
The value set in this field is used as the User-Agent and Server value included in
SIP request headers made by this line. If the field is blank, the type of IP Office
system and its software level used. Setting a unique value can be useful in call
diagnostics when the system has multiple SIP trunks.
For calls that result in a busy response from IP Office, this setting determines the
response code. The options are:
• 486 - Busy Here
• 503 - Service Unavailable
Action on CAC Location
Limit
Default = Allow Voicemail
When set to Allow Voicmail, the call is allowed to go to a user's voicemail when
the user's location call limit has been reached. When set to Reject Call, the call is
rejected with the failure response code configured in the Service Busy Response
field.
Related links
Line | SIP Line on page 354
Line | SIP Transport
Changing these settings requires a “merge with service disruption”. When the configuration file is
sent to the system, the SIP trunk is restarted.
Note that ITSP Proxy Address and Calls Route via Registrar are mergeable.
Deleting a SIP line requires a “merge with service disruption”.
Behavior during Service unavailable
A proxy server is considered Active once the system has received a response to an INVITE,
REGISTER or OPTIONS.
In the case of the proxy server responding with 503 - Service Unavailable, it should be considered
Active - In Maintenance. In this case, the following should occur:
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Configuration Mode Field Descriptions
If the response 503 - Service Unavailable was in response to an INVITE request:
• If calls are tied to registrations (Calls Route via Registrar enabled) and there are other
proxies available, the tied registrations should issue an Un-REGISTER and try to REGISTER
with a different proxy. The call should fail with cause = Temporary Fail.
• If calls are not tied, the INVITE should be immediately tried to a different proxy.
If the response 503 - Service Unavailable was in response to a REGISTER request:
If there are other proxies available, this registration only should issue an Un-REGISTER and try to
REGISTER with a different proxy.
If Explicit DNS Server(s) are configured, a DNS request should be sent out to see whether the
proxy server has disappeared from those being offered.
An Active-InMaintenance proxy server should not be used for a new transactions (INVITE or
REGISTER) until:
• There is a change in DNS responses indicating the proxy has become active.
• The configuration does not leave any better option available. In this case, there should be a
throttle so that no more than 5 failures (without successes) in 1 minute should be allowed.
• A config merge has occurred where the proxy string is changed.
• 10 minutes has expired.
Behavior during Not Responding
A proxy server that is not-responding (UDP) is indicated when 3 requests are sent and no replies
are received. This would normally occur during a single INVITE transaction.
Consideration should be given whether this is caused by a local network fault or is caused by the
Proxy being out of service. Since it is likely to be local, no action should be taken unless traffic is
received from an alternative proxy while this proxy is actually not responding. The state should be
"Possibly non responding".
If explicit DNS servers are configured, a DNS request should be sent out to see whether this Proxy
server has disappeared from those being offered.
If possible, an alternative proxy should be stimulated simultaneously with stimulating the suspect
server.
The server should be considered non-responding if it is persistently nonresponding while other
proxies are responding or if it is non-responding and has disappeared from the DNS advertisement.
While in the "possibly not responding" state, it would be better to send an INVITE to an alternative
proxy while simultaneously sending any appropriate message to this proxy. This will help to resolve
whether it is really not responding rather than there being local network problems. However, there is
no requirement to blacklist the proxy.
Once in the "definitely not responding" state:
• If there are other proxies available: this registration only issues an Un-REGISTER, and try to
REGISTER with a different proxy. Calls do not automatically clear.
• If a SIP message is received from it, the state should immediately go"Active".
• This proxy should be blacklisted unless there are no better options available. While blacklisted,
only one transaction per 10 minutes is allowed.
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Line
• Even if not blacklisted, there should be a throttle so that no more than 5 failures (without
successes) in 1 minute should be alowed.
Field
Description
ITSP Proxy Address
Default = Blank
This is the SIP Proxy address used for outgoing SIP calls. The address can be
specified in the following ways:
• If left blank, the ITSP Domain Name is used and is resolved by DNS resolution in
the same way as if a DNS address had been specified as below.
• An IP address.
• A list of up to 4 IP addresses, with each address separated by a comma or space.
- The addresses can include an indication of the relative call weighting of each
address compared to the others. This is done by adding a w N suffix to the
address where N is the weighting value. For example, in the list 213.74.81.102w3
213.74.81.100w2, the weighting values assigns 1.5 times the weight of calls to the
first address. The default weight if not specified is 1. A weight of 0 can be used to
disable an address. Weight is only applied to outgoing calls.
If there is more than one proxy defined, and no weight indication, then calls are
only sent to the first in the list until there is a failure at which point the next proxy is
used.
- If the Calls Route via Registrar setting below is enabled, the weighting is applied
to registrations rather than calls.
• A DNS address, for example sbc.example.com.
- The DNS response may return multiple proxy addresses (RFC 3263). If that is the
case, the system will resolve the address to use based on priority, TTL and
weighting information included with each address.
- A load balancing suffix can be added to specify that multiple proxy results should
be returned if possible, for example sbc.example.com(N). where N is the required
number of addresses from 1 to 4.
This field is mergeable. However, no more than 4 IP Addresses should be in use at
any time. So, if the combined new and old address settings exceed 4, the new
addresses are only phased into use as transactions in progress on the previous
addresses are completed.
Network Configuration
Layer 4 Protocol
Default = UDP.
The options are:
• TCP
• UDP
• TLS
• Auto
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Configuration Mode Field Descriptions
Field
Description
TLS connections support the following ciphers:
• TLS_RSA_WITH_AES_128_CBC_SHA
• TLS_RSA_WITH_AES_256_CBC_SHA
• TLS_DHE_RSA_WITH_AES_128_CBC_SHA
• TLS_DHE_RSA_WITH_AES_256_CBC_SHA
Use Network
Topology Info
Default = None.
This field associates the SIP line with the Network Topology settings (System |
LAN1 | Network Topology) settings of the LAN interface. It also applies the LAN
interfaces DiffServ Settings (System | LAN1 | VoIP) to the outgoing traffic on the
SIP line. If None is selected, STUN lookup is not applied and routing is determined by
the system's routing tables.
If no STUN server address is set for the interface, then the Binding Refresh Time
(System | LAN | Network Topology) is ignored by SIP Lines when calculating the
periodic OPTIONS timing unless the Firewall/NAT Type is set to Open Internet.
Send Port
When Network Configuration is set to TLS, the default setting is 5061. When Network
Configuration is set to TCP or UDP, the default setting is 5060.
Listen Port
When Network Configuration is set to TLS, the default setting is 5061. When Network
Configuration is set to TCP or UDP, the default setting is 5060.
Explicit DNS
Server(s)
Default = 0.0.0.0 (Off)
Calls Route via
Registrar
Default = On
Separate Registrar
Default = Blank
If specific DNS servers should be used for SIP trunk operation rather than the general
DNS server specified or obtained for the system, the server addresses can be
specified here. If exported or imported as part of a trunk template, this setting not
supported by Basic Edition Quick mode systems.
If selected, all calls are routed via the same proxy as used for registration. If multiple
ITSP proxy addresses have been specified, there is no load balancing of registrations.
This field allows the SIP registrar address to be specified if it is different from that of
the SIP proxy. The address can be specified as an IP address or DNS name.
Related links
SIP Line on page 353
Line | SIP URI
Having setup the SIP trunk to the SIP ITSP, the SIP URI's registered with that ITSP are entered on
this tab. A SIP URI (Uniform Resource Identifier) is similar to an internet email address and
represents the source or destination for SIP connection. The URI consists of two parts, the user part
(eg. name) and the host part (eg. example.com).
If the wildcard * is used in the SIP trunk's Local URI, Contact and Display fields, that SIP trunk will
accept any incoming SIP call. The incoming call routing is still performed by the system incoming
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Line
call routes based on matching the values received with the call or the URI's incoming group setting.
For outgoing calls using this SIP URI, all valid short code CLI manipulations are used (transforming
calling party number to ISDN will be ignored).
For the system, each SIP URI acts as a set of trunk channels. Outgoing calls can then be routed to
the required URI by short codes that match that URI's Outgoing Group setting. Incoming calls can
be routed by incoming call routes that match the URI's Incoming Group setting.
Note that the system only supports up to 150 URI records on a SIP line.
These settings are mergeable, with the exception of the Registration setting. Changing the
Registration setting requires a “merge with service disruption”. When the configuration file is sent to
the system, the SIP trunk is restarted and all calls on the line are dropped.
Deleting a SIP line requires a “merge with service disruption”.
Field
Description
Via
This field is for information only and cannot be edited. It shows the IP address of the
system LAN interface with which the SIP trunk is associated.
Local URI
Default = Use Authentication Name
This field sets the 'From' field for outgoing SIP calls using this URI. The value can
either be entered manually or one of the following options can be selected:
Contact
Default = Use Authentication Name
This field sets the 'Contact' field for SIP calls using this URI. The value can either be
entered manually or one of the following options can be selected:
Display Name
Default = Use Authentication Name This field sets the 'Name' value for SIP calls using
this URI. The value can either be entered manually or one of the following options can
be selected:
PAI
Default = None.
You can enable P-Asserted-Identity (PAI) headers to assert the identity of users in
outgoing SIP requests or response messages. Use this setting to select the source of
the user identity information or enter a value manually.
Note:
You can enter the wildcard character “*”. Entering this value populates the SIP
PAI header with the caller information available to IP Office.
The options are:
• None: When selected, the P-Preferred-Identity header is used instead of the PAsserted-Identity, in order to ensure compatibility with legacy networks.
• Use Credentials User Name: Use the User Name from the SIP Credentials record
being used for the call.
• Use Internal Data: Use the SIP Name value from the User | SIP tab of the user
making the call. The system can also use SIP URI information configured for a hunt
group (Hunt Group | SIP) or for the voicemail (System | Voicemail).
• Use Credentials Authentication Name: Use the Authentication Name from the
SIP Credentials record being used for the call.
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Configuration Mode Field Descriptions
Field
Description
• Use Credentials Contact: Use the Contact information from the SIP Credentials
record being used for the call.
Registration
Default = Primary
This field is used to select from a list of the account credentials configured on the line's
SIP Credentials tab.
Incoming Group ID
Default = 0, Range 0 to 99999.
The Incoming Group ID to which a line belongs is used to match it to incoming call
routes in the system configuration. The matching incoming call route is then used to
route incoming calls. The same ID can be used for multiple lines.
Outgoing Group ID
Default = 0, Range 0 to 99999.
Short codes that specify a number to dial also specify the line group to be used. The
system will then seize a line with a matching Outgoing Group ID.
For Basic Edition and Standard Edition deployments, the same ID can be used for
multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must also be
unique within the network, ie. the same ID cannot be used in the configuration of any
lines on another server system in the network.
Max Sessions
Reserved Group ID
numbers
Description
90000 - 99999
Reserved for system use (not enforced).
99999 and 99998
In a Server Edition network, reserved for the H.323 lines to the
Primary Server and Secondary Server respectively.
99901 to 99930
In a Server Edition network, reserved for the H.323 lines from
the Primary Server to each expansion system in the network.
0
In a Server Edition network, the ID 0 cannot be used.
98888
For IP Office deployed in an Enterprise Branch environment,
reserved for the SM line.
Default =10 This field sets the maximum number of simultaneous calls that can use
the URI before the system returns busy to any further calls. For Outbound Contact
Express deployments, the maximum value is 250.
Related links
SIP Line on page 353
Line | SIP VoIP
These settings have changed in release 9.1. To view the release 9.0 settings click here on
page 372.
This form is used to configure the VoIP settings applied to calls on the SIP trunk.
These settings are mergeable. Deleting a SIP line requires a “merge with service disruption”.
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Field
Description
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup. The available codecs
in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM, IP500
VCM V2 or IP500 Combo cards. For Server Edition, it is supported on Primary
Server, Secondary Server and Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
Fax Transport
Support
Default = Off.
This option is only available if Re-Invite Supported is selected. When enabled, the
system performs fax tone detection on calls routed via the line and, if fax tone is
detected, renegotiates the call codec as configured below. The SIP line provider must
support the selected fax method and Re-Invite. The system must have available VCM
resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo base card.
For systems in a network, fax relay is supported for fax calls between the systems.
The options are:
• None Select this option if fax is not supported by the line provider.
• G.711 G.711 is used for the sending and receiving of faxes.
• T38 T38 is used for the sending and receiving of faxes. This option is not
supported by Linux based systems.
• T38 Fallback When you enable this option, T38 is used for sending and receiving
faxes on a SIP line. If the called destination does not support T38, the system will
send a re-invite to change the transport method to G.711. This option is not
supported on Linux based systems.
DTMF Support
Default = RFC2833.
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Configuration Mode Field Descriptions
Field
Description
This setting is used to select the method by which DTMF key presses are signalled to
the remote end. The options are:
• In Band: Send DTMF digits as part of the audio path.
• RFC2833: Send DTMF digits using a separate audio stream from the voice path.
Note that use of RFC2833 is negotiated with the remote end of the call. If not agreed
or not supported, the line reverts to using in band signalling.
• Info: Send the DTMF digits in SIP INFO packets.
Media Security
Default = Disable.
These setting control whether SRTP is used for this line and the settings used for the
SRTP. The options are:
• Same As System: Use the same settings as the system setting configured on the
System | VoIP Security tab.
• Disable: Media security is not required. All media sessions (audio, video, and data)
will be enforced to use RTP only.
• Enforce: Media security is required. All media sessions (audio, video, and data) will
be enforced to use SRTP only.
Warning:
Selecting Enforce on a line or extension that does not support media security
will result in media setup failures.
• Best Effort: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
Advanced Media
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same setting as the system setting configured on the
System | VoIP Security tab.
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option
to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence
Suppression
Default = Off
Re-Invite Supported
Default = Off.
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunks between networked systems, the same
setting should be set at both ends.
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Field
Description
When enabled, Re-Invite can be used during a session to change the characteristics
of the session. For example when the target of an incoming call or a transfer does not
support the codec originally negotiated on the trunk. Requires the ITSP to also
support Re-Invite.
Codec Lockdown
Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response
to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP
answer that also lists multiple codecs. This means that the user agent may switch to
any of those codecs during the session without further negotiation. The system does
not support multiple concurrent codecs for a session, so loss of speech path will occur
if the codec is changed during the session. If codec lockdown is enabled, when the
system receives an SDP answer with more than one codec from the list of offered
codecs, it sends an extra re-INVITE using just a single codec from the list and
resubmits a new SDP offer with just the single chosen codec.
Allow Direct Media
Path
Default = Off.
This option is only available when Re-Invite Supported is enabled.
The settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure. If enabled, IP calls can take routes
other than through the system. This removes the need for a voice compression
channel. Both ends of the calls must support Direct Media and be using the same
protocol (H.323 or SIP). Enabling this option may cause some vendors problems with
changing the media path mid call.
Force direct media
with phones
PRACK/100rel
Supported
G.711 Fax ECAN
Default = Off.
The setting is only available when the trunk's Re-invite Supported and Allow Direct
Media Path settings are enabled and its DTMF Support option is set to RFC2833/
RFC4733. It also requires the H.323 IP extension involved in the call to also have
Allow Direct Media Path enabled. This feature is only supported with Avaya H.323 IP
telephones. For calls where the Avaya H.323 IP extension using the trunk is doing so
as a direct media call, this feature allows digits pressed on the extension to be
detected and the call changed to an indirect media call so that RFC2833 DTMF can
be sent. The call remains as an indirect media call for 15 seconds after the last digit
before reverting back to being a direct media call.
Default = Off.
When selected, supports Provisional Reliable Acknowledgement (PRACK) on SIP
trunks. Enable this parameter when you want to ensure that provisional responses,
such as announcement messages, have been delivered. Provisional responses
provide information on the progress of the request that is in process. For example,
while a cell phone call is being connected, there may be a delay while the cell phone
is located; an announcement such as “please wait while we attempt to reach the
subscriber” provides provisional information to the caller while the request is in
process. PRACK, which is defined in RFC 3262, provides a mechanism to ensure the
delivery of these provisional responses.
Default = Off (Grayed out for Server Edition Linux systems.)
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Configuration Mode Field Descriptions
Field
Description
This setting is only available when Fax Transport Support is set to G.711 or T.38
Fallback. When IP Office detects a fax call, the IP Office negotiates to G.711 (if not
already in G.711) and reconfigures the connection with echo cancellation (ECAN)
based on the 'G.711 Fax ECAN field. This can be used to avoid an ECAN mismatch
with the SIP trunk service provider. Also for fax calls, the connection’s NLP is
disabled, a fixed jitter buffer is set and silence suppression is disabled.
Related links
SIP Line on page 353
Line | SIP VoIP (Release 9.0) on page 372
Line | SIP VoIP (Release 9.0)
This form is used to configure the VoIP settings applied to calls on the SIP trunk.
These settings are mergeable.
Field
Description
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup. The available codecs
in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM, IP500
VCM V2 or IP500 Combo cards. For Server Edition, it is supported on Primary
Server, Secondary Server and Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
Fax Transport
Support:
February 2016
Default = Off.
This option is only available if Re-Invite Supported is selected. When enabled, the
system performs fax tone detection on calls routed via the line and, if fax tone is
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Field
Description
detected, renegotiates the call codec as configured below. The SIP line provider must
support the selected fax method and Re-Invite. The system must have available VCM
resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo base card. For
systems in a network, fax relay is supported for fax calls between the systems.The
options are:
• None Select this option if fax is not supported by the line provider.
• G.711 G.711 is used for the sending and receiving of faxes.
• T38 T38 is used for the sending and receiving of faxes.
• T38 Fallback When you enable this option, T38 is used for sending and receiving
faxes on a SIP line. If the called destination does not support T38, the system will
send a re-invite to change the transport method to G.711. This option is not
supported by Linux based systems.
Location
Default = Cloud.
Specify a location to associate the extension with a physical location. Associating an
extension with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined in the System |
Location form.
Call Initiation Timeout Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
DTMF Support
Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled to
the remote end. The options are:
• In Band
• RFC2833
• Info
VoIP Silence
Suppression
Default = Off
Allow Direct Media
Path
Default = Off.
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
This setting is only supported on Server Edition systems. This option is only available
when Re-Invite Supported is enabled.
The settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure. If enabled, IP calls can take routes
other than through the system. This removes the need for a voice compression
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Configuration Mode Field Descriptions
Field
Description
channel. Both ends of the calls must support Direct Media and be using the same
protocol (H.323 or SIP). Enabling this option may cause some vendors problems with
changing the media path mid call.
Re-Invite Supported
Default = Off.
When enabled, Re-Invite can be used during a session to change the characteristics
of the session. For example when the target of an incoming call or a transfer does not
support the codec originally negotiated on the trunk. Requires the ITSP to also
support Re-Invite.
Codec Lockdown
Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response
to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP
answer that also lists multiple codecs. This means that the user agent may switch to
any of those codecs during the session without further negotiation. The system does
not support multiple concurrent codecs for a session, so loss of speech path will occur
if the codec is changed during the session. If codec lockdown is enabled, when the
system receives an SDP answer with more than one codec from the list of offered
codecs, it sends an extra re-INVITE using just a single codec from the list and
resubmits a new SDP offer with just the single chosen codec.
PRACK/100rel
Supported
Default = Off.
Force direct media
with phones
Default = Off.
When selected, supports Provisional Reliable Acknowledgement (PRACK) on SIP
trunks. Enable this parameter when you want to ensure that provisional responses,
such as announcement messages, have been delivered. Provisional responses
provide information on the progress of the request that is in process. For example,
while a cell phone call is being connected, there may be a delay while the cell phone
is located; an announcement such as “please wait while we attempt to reach the
subscriber” provides provisional information to the caller while the request is in
process. PRACK, which is defined in RFC 3262, provides a mechanism to ensure the
delivery of these provisional responses.
The setting is only useable when the trunk's Re-invite Supported and Allow Direct
Media Path settings are enabled and its DTMF Support option is set to RFC2833/
RFC4733. It also requires the H.323 IP extension involved in the call to also have
Allow Direct Media Path enabled. This feature is only supported with Avaya H.323 IP
telephones. For calls where the Avaya H.323 IP extension using the trunk is doing so
as a direct media call, this feature allows digits presses on the extension to be
detected and the call changed to an indirect media call so that RFC2833 DTMF can
be sent. The call remains as an indirect media call for 15 seconds after the last digit
before reverting back to being a direct media call.
Related links
Line | SIP VoIP on page 368
Line | SIP T38 Fax
The settings on this tab are only accessible if Re-invite Supported and Fax Transport Support
are selected on the VoIP tab.
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Fax relay is only supported on IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or IP500
Combo cards. Fax relay is not supported on Server Edition.
Changing these settings requires a “merge with service disruption”. When the configuration file is
sent to the system, the SIP trunk is restarted and all calls on the line are dropped.
Deleting a SIP line requires a “merge with service disruption”.
Related links
SIP Line on page 353
Line | SIP Credentials
Used to enter the ITSP username and password for the SIP account with the ITSP. If you have
several SIP accounts going to the same ITSP IP address or domain name, you can enter up to 30
sets of ITSP account names and passwords on this tab.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
Deleting a SIP line requires a “merge with service disruption”.
Use the Add, Remove, and Edit buttons to manage the set of credentials for the SIP trunk
accounts. The settings for each account are listed below.
Field
Descriptions
Index
This number is assigned automatically and cannot be edited. If the From field on the
SIP URI being used for the call is set to Use Authentication Name , the registration
field of the SIP URI will indicate the index number of the SIP credentials to use for calls
by that SIP URI.
User Name
This name must be unique and is used to identify the trunk. The name can include the
domain if necessary.
Authentication Name
Default = Blank.
This field can be blank but must be completed if a Password is also specified. This
value is provided by the SIP ITSP. Depending on the settings on the Local URI tab
associated with the SIP call, it may also be used as the user part of the SIP URI. The
name can include the domain if necessary.
Contact
Default = Blank.
This field is used to enter a contact and can include the domain if necessary.
Password
Default = Blank.
This value is provided by the SIP ITSP. If a password is specified, the matching
Authentication Name must also be set.
Expiry
Default = 60 minutes.
This setting defines how often registration with the SIP ITSP is required following any
previous registration.
Registration Required
Default = On. I
If selected, the fields above above are used for registration when making calls. If
exported or imported as part of a trunk template, this setting is not supported by Basic
Edition Quick mode systems.
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Configuration Mode Field Descriptions
Related links
SIP Line on page 353
Line | SIP Advanced
These settings are mergable, with the exception of the Media Connection Preservation setting.
Changing the Media Connection Preservation setting requires a “merge with service disruption”.
When the configuration file is sent to the system, the SIP trunk is restarted and all calls on the line
are dropped.
Deleting a SIP line requires a “merge with service disruption”.
Field
Description
Addressing
Association Method
Default = By Source IP Address.
This setting sets the method by which a SIP line is associated with an incoming SIP
request.
The match criteria used for each line can be varied. The search for a line match for an
incoming request is done against each line in turn using each lines Association
Method. The order of line matching uses the configured Line Number settings until a
match occurs. If no match occurs the request is ignored. This method allows multiple
SIP lines with the same address settings. This may be necessary for scenarios where
it may be required to support multiple SIP lines to the same ITSP. For example when
the same ITSP supports different call plans on separate lines or where all outgoing
SIP lines are routed from the system via an additional on-site system. The options are:
• By Source IP Address: This option uses the source IP address and port of the
incoming request for association. The match is against the configured remote end of
the SIP line, using either an IP address/port or the resolution of a fully qualified
domain name.
• "From" header hostpart against ITSP domain: This option uses the host part of
the From header in the incoming SIP request for association. The match is against
the ITSP Domain Name above.
• R-URI hostpart against ITSP domain: This option uses the host part of the
Request-URI header in the incoming SIP request for association. The match is
against the ITSP Domain Name above.
• "To" header hostpart against ITSP domain: This option uses the host part of the
To header in the incoming SIP request for association. The match is against the
ITSP Domain Name above.
• "From" header hostpart against DNS-resolved ITSP domain: This option uses
the host part of the FROM header in the incoming SIP request for association. The
match is found by comparing the FROM header against a list of IP addresses
resulting from resolution of the ITSP Domain Name above or, if set, the ITSP Proxy
Address on the Transport tab.
• "Via" header hostpart against DNS-resolved ITSP domain: This option uses the
host part of the VIA header in the incoming SIP request for association. The match
is found by comparing the VIA header against a list of IP addresses resulting from
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Line
Field
Description
resolution of the ITSP Domain Name above or, if set, the ITSP Proxy Address on
the Transport tab.
• "From" header hostpart against ITSP proxy: This option uses the host part of the
“From” header in the incoming SIP request for association. The match is against the
ITSP Proxy Address on the Transport tab.
• "To" header hostpart against ITSP proxy: This option uses the host part of the
From header in the incoming SIP request for association. The match is against the
ITSP Proxy Address on the Transport tab.
• R-URI hostpart against ITSP proxy: This option uses the host part of the RequestURI in the incoming SIP request for association. The match is against the ITSP
Proxy Address on the Transport tab.
Call Routing Method
Default = Request URI.
This field allows selection of which incoming SIP information should be used for
incoming number matching by the system's incoming call routes. The options are to
match either the Request URI or the To Header element provided with the incoming
call.
Suppress DNS SRV
Lookups
Default = Off.
Controls whether to send SRV queries for this endpoint, or just NAPTR and A record
queries.
Identity
Use Phone Context
Default = Off.
When set to On, signals SIP enabled PBXs that the call routing identifier is a
telephone number.
Add user=phone
Default = Off.
This setting is available when Use Phone Context is set to On.
Use + for
International
Use PAI for Privacy
Default = Off.
When set to On, outgoing international calls use E.164/International format with a ‘+’
followed by the country code and then the directory number.
Default = Off.
When set to On, if the caller ID is withheld, the SIP message From: header is made
anonymous, and the caller’s identity, for admission control, billing, and emergency
services, is inserted into the P-Asserted-Identity header. This mechanism should only
be used in a trusted network and must be stripped out of the SIP message before it is
forwarded outside the trusted domain.
Use Domain for PAI
Default = Off.
When set to Off, the DNS resolved IP address of the ITSP Proxy is used for the host
part in the P-Asserted-Identity header. When set to On, the the Domain for PAI is
used.
Swap From and PAI
Default = Off.
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Configuration Mode Field Descriptions
Field
Description
When set to on, swaps the values in the From: and P-Asserted-Identity headers, as
required by some networks.
Caller ID from From
Header
Default = Off.
Send From In Clear
Default = Off.
Incoming calls can include caller ID information in both the From field and in the PAI
fields. When this option is selected, the caller ID information in the From field is used
rather than that in the PAI fields.
When selected, the user ID of the caller is included in the From field. This applies
even if the caller has selected to be or is configured to be anonymous, though their
anonymous state is honored in other fields used to display the caller identity.
Cache Auth
Credentials
Default = On.
User-Agent and
Server Headers
Default = Blank (Use system type and software level).
When set to On, allows the credentials challenge and response from a registration
transaction to be automatically inserted into later SIP messages without waiting for a
subsequent challenge.
The value set in this field is used as the User-Agent and Server value included in SIP
request headers made by this line. If the field is blank, the type of IP Office system
and its software level used. Setting a unique value can be useful in call diagnostics
when the system has multiple SIP trunks.
Media
Allow Empty INVITE
Default = Off.
When set to On, allows 3pcc devices to initiate calls to IP Office by sending an INVITE
without SDP.
Send Empty reINVITE
Default = Off.
This option is only available if Re-Invite Supported is selected on the VoIP tab.
If set to On, when connecting a call between two endpoints, IP Office sends an
INVITE without SDP in order to solicit the full media capabilities of both parties.
Allow To Tag Change
Default = Off.
When set to On, allows the IP Office to change media parameters when connecting a
call to a different party than that which was advertised in the media parameters of
provisional responses, such as 183 Session Progress.
P-Early-Media
Support
Default = None.
Send
SilenceSupp=off
Default = Off.
When set to on, allows the service provider and IP Office to supply several early
media responses, but indicate which should be used to supply media to the endpoint
before the call is answered.
Used for the G711 codec. When checked, the silence suppression off attribute is sent
in SDP on this trunk.
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Field
Description
Force Early Direct
Media
Default = Off.
Media Connection
Preservation
Default = Disabled.
When set to On, allows the direct connection of early media streams to IP endpoints
rather than anchoring it at the IP office.
When enabled, allows established calls to continue despite brief network failures. Call
handling features are no longer available when a call is in a preserved state.
Preservation on public SIP trunks is not supported until tested with a specific service
provider.
Call Control
Call Initiation Timeout Default = 4 seconds. Range = 1 to 99 seconds.
(s)
Sets how long the system should wait for a response to its attempt to initiate a call
before following the alternate routes set in an ARS form.
Call Queuing Timeout
(m)
Default = 5 minutes.
For incoming calls, how many minutes to wait before dropping a call that has been
queued waiting for a free VCM resource, or has remained in the unanswered state.
For an outgoing calls, how many minutes to wait for the call to be answered after
receiving a provisional response.
Service Busy
Response
Default = 486 - Busy Here (503 - Service Unavailable for the France2 locale).
For calls that result in a busy response from IP Office, this setting determines the
response code. The options are:
• 486 - Busy Here
• 503 - Service Unavailable
on No User
Responding Send
Default = 408-Request Timeout.
Action on CAC
Location Limit
Default = Allow Voicemail
Specifies whether to send a SIP response 408-Request Timeout or 480 Temporarily
Unavailable.
When set to Allow Voicemail, the call is allowed to go to a user's voicemail when the
user's location call limit has been reached. When set to Reject Call, the call is
rejected with the failure response code configured in the Service Busy Response
field.
Suppress Q.850
Reason Header
Default = Off.
Emulate NOTIFY for
REFER
Default = Off.
When SIP calls are released by sending BYE and CANCEL, a release reason header
is added to the message.
Use for SIP providers that do not send NOTIFY messages. After IP Office issues a
REFER, and the provider responds with 202 ACCEPTED, IP Office will assume the
transfer is complete and issue a BYE.
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Configuration Mode Field Descriptions
Field
Description
No REFER if using
Diversion
Default = Off.
When enabled, REFER is not sent on the trunk if the forwarding was done with 'Send
Caller ID = Diversion Header'. Applies to Forwards and Twinning.
Related links
SIP Line on page 353
Line | SIP Engineering
This page is used to enter values that apply special features to the SIP line. These are entered
using the Add, Edit and Remove buttons.
These settings are mergeable. Deleting a SIP line requires a “merge with service disruption”.
Related links
SIP Line on page 353
Line | SIP DECT Line
A SIP DECT line can be manually added. SIP DECT lines are used to manage D100 Base Station
operation.
Related links
Line on page 289
Line | SIP DECT Base on page 380
Line | SIP DECT VoIP on page 381
Line | SIP DECT Base
Currently, IP Office supports four D100 Base Stations.
These settings are not mergeable. Changes to these settings requires a reboot of the system.
Field
Description
Line Number
Default = Blank. A unique line number associated with the SIP DECT Base Station.
Associated Extensions are other extensions that can log into the base station.
Base Name
Default = Blank. Maximum 16 characters.
A name assigned to the base station. Each base station provisioned on the IP Office must
have a unique name. The field cannot be blank. The format is an alphanumeric string with
no special characters.
Base MAC
Address
Default = Blank.
The MAC Address of the base station. If only one base station is provisioned, the field can
remain at the default value. If multiple base stations are provisioned, the MAC address for
each base station must be entered.
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Line
Field
Description
Configure Base IP
Configure Base IP Default = Off.
Set to On to configure IP address attributes for the base station. When enabled, the
Configure Base IP settings are displayed.
DHCP Client
Default = On.
When enabled, specifies that the base station operates as a DHCP client. When enabled,
not other IP address attributes can be configured.
IP Address
Default = Blank.
The IP address of the base station. The IP address must be on the same subnet as one of
the LAN interfaces.
IP Mask
Default = Blank.
IP address mask.
IP Gateway
Default = Blank.
The default gateway address
Provisioning
Server
Default = IP Office interface address. The server address from where the Base Station
configuration files can be retrieved.
Description
Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
Related links
Line | SIP DECT Line on page 380
Line | SIP DECT VoIP
This form is used to configure the VoIP setting applied to calls on the SIP DECT line.
These settings are not mergeable. Changes to these settings requires a reboot of the system.
Field
Description
IP Address
Default = Blank.
The IP address of the SIP DECT extension.
Codec Selection
Default = Custom
This field defines the codec or codecs offered during call setup. The codecs available
to be used are set through the System Codec list (System | System Codec). The
Custom option allows specific configuration of the codec preferences to be different
from the system Default Selection list. When Custom is selected, the list can be
used to select which codecs are in the Unused list and in the Selected list and to
change the order of the selected codecs. The D100 Base Station supports only G711
codecs.
TDM | IP Gain
Default = Default (0dB). Range = -31dB to +31dB.
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Configuration Mode Field Descriptions
Field
Description
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
DTMF Support
Default =RFC2833
The D100 Base Station supports only RFC2833.
VoIP Silence
Suppression
Default = Off
Local Hold Music
Default = Off
Allow Direct Media
Path
Default = On
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
This settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure.
• If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
Media and be using the same protocol (H.323 or SIP). Enabling this option may
cause some vendors problems with changing the media path mid call.
• If disabled or not supported at on one end of the call, the call is routed via the
system. RTP relay support allows calls between devices using the same audio
codec to not require a voice compression channel.
RE-Invite Supported
Default = Off.
When enabled, Re-Invite can be used during a session to change the characteristics
of the session. For example when the target of an incoming call or a transfer does not
support the codec originally negotiated on the trunk. Requires the ITSP to also
support Re-Invite.
Related links
Line | SIP DECT Line on page 380
Line | SM Line
This type of line is used to create a SIP connection between an IP Office and an Avaya Aura®
Session Manager. The other end of the SIP connection must be configured on the Session Manager
as a SIP Entity Link.
An SM Line can only be added to IP Office system Standard Mode or Server Edition configurations.
It is typically used in IP Office Standard mode in Enterprise Branch deployments connected to the
Avaya Aura® network. For more details about IP Office Enterprise Branch deployments refer to
Deploying IP Office in an Avaya Aura® Branch Environment (18–603853).
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Line
An SM Line can also be used in IP Office Server Edition to connect to an Avaya Aura® Session
Manager. Through the SM Line, IP Office Server Edition supports interoperability with Avaya Aura®
Session Manager. It also supports interoperability, via the Avaya Aura® Session Manager, with
Avaya Aura® Communication Manager systems and with CS 1000 systems. Note that IP Office
Server Edition is not used as an enterprise branch product and does not support some of the IP
Office enterprise branch functionality, such as management by Avaya Aura® System Manager,
WebLM licensing, Centralized Users or voicemail over the SM Line.
If the Avaya Aura® network has multiple Avaya Aura® Session Managers to provide redundancy,
two SM lines can be added, one configured for each Avaya Aura® Session Manager.
Related links
Line on page 289
Line | Session Manager on page 383
Line | SM Line VoIP on page 385
SM Line T38 Fax on page 389
Line | Session Manager
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Changing the In Service setting to Disabled (out of service) requires a system reboot. However,
changing the In Service setting to Enabled is mergeable. Configuration changes made while the
line is out of service are also mergeable.
Field
Description
Line Number
Default = Automatically assigned.
This value is automatically assigned by the Enterprise Branch and should be
unique for each line added to the configuration.
• Session Manager line prioritization: Up to two Session Manager lines can be
configured. The two Session Manager lines are prioritized based on the line
number. The lower line number is considered the primary Session Manager line.
For example, if the first Session Manager line is configured as line number 17
and the second Session Manager line is configured as line 18, then line number
17 is considered the primary Session Manager line. If you want to designate the
second Session Manager line (line 18 in this example) as the primary Session
Manager line, you must change one or both of the line numbers so that the
second Session Manager line is configured with a lower number than the current
primary line.
• Session Manager line redundancy: Based on the priority of the Session
Manager lines designated by the line number, the active line to which theIP
Office> sends all calls will always be the highest priority Session Manager line in
service. That is, if the primary Session Manager line is in service, it will be the
active line for sending calls. If the connection to the primary Session Manager
line is lost, causing the IP Office to switch to the secondary Session Manager
line, then when the primary line comes back up later, the IP Office reverts back
to the primary Session Manager line.
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Configuration Mode Field Descriptions
Field
Description
In Service
Default = Enabled
This option can be used to administratively disable the SM Line. It does not reflect
the dynamic state of the line. If an SM Line is administratively disabled it is not
equivalent to being in the dynamic out of service state.
SM Domain Name
This should match a SIP domain defined in the Session Manager system's SIP
Domains table. Unless there are reasons to do otherwise, all the Enterprise Branch
systems in the Avaya Aura® network can share the same domain.
SM Address
Enter the IP address of the Session Manager the line should use in the Avaya
Aura network. The same Session Manager should be used for the matching Entity
Link record in the Avaya Aura® configuration.
Outgoing Group ID
Default = 98888
This value is not changeable. However note the value as it is used in Enterprise
Branch short codes used to route calls to the Session Manager.
Prefix
Default = Blank
This prefix will be added to any source number received with incoming calls.
Max Calls
Default = 10
Sets the number of simultaneous calls allowed between the Enterprise Branch and
Session Manager using this connection. Each call will use one of the available
licenses that are shared by all SIP trunks configured in the system.
Note:
You must configure the Maximum SIP Sessions setting on the System |
Telephony | Telephony page.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private.
The system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also
using any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Include location specific
information
URI Type
Default = Off.
Enabled when Network Type is set to Private. Set to On if the PBX on the other
end of the trunk is toll compliant.
Default = SIP.
When SIP or SIPS is selected in the drop-down box, the SIP URI format is used
(for example, name@example.com). This affects the From field of outgoing calls.
The To field for outgoing calls will always use the format specified by the short
codes used for outgoing call routing. Recommendation: When SIP Secured URI is
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Line
Field
Description
required, the URI Type should be set to SIPS. SIPS can be used only when Layer
4 Protocol is set to TLS.
Media Connection
Preservation
Default = Enabled.
When enabled, attempts to maintain established calls despite brief network
failures. Call handling features are no longer available when a call is in a
preserved state. When enabled, Media Connection Preservation applies to Avaya
H.323 phones that support connection preservation.
Network Configuration
TLS connections support the following ciphers:
• TLS_RSA_WITH_AES_128_CBC_SHA
• TLS_RSA_WITH_AES_256_CBC_SHA
• TLS_DHE_RSA_WITH_AES_128_CBC_SHA
• TLS_DHE_RSA_WITH_AES_256_CBC_SHA
Layer 4 Protocol
Default = TCP.
Send Port
When Network Configuration is set to TLS, the default setting is 5061. When
Network Configuration is set to TCP, the default setting is 5060.
Listen Port
When Network Configuration is set to TLS, the default setting is 5061. When
Network Configuration is set to TCP, the default setting is 5060.
Session Timer
Default = 1200. Range = 90 to 64800
This field specifies the session expiry time. At the half way point of the expiry time,
a session refresh message is sent. Setting the field to On Demand disables the
session timer.
Communication Manager supports SIP session refresh via UPDATE in
Communicaton Manger release 6.2 SP1 and later. If using an earlier release of
Communication Manager, then the Session Timer parameter must be set to On
Demand.
Description
Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
Related links
Line | SM Line on page 382
Line | SM Line VoIP
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Codec Selection
Default = System Default
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Configuration Mode Field Descriptions
Field
Description
This field defines the codec or codecs offered during call setup. The available
codecs in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition, it is supported on
Primary Server, Secondary Server and Expansion System (L) systems and on
Expansion System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500
Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to
change the order of the selected codecs.
Fax Transport Support
Default = Off.
This option is only available if Re-Invite Supported is selected. When enabled,
the system performs fax tone detection on calls routed via the line and, if fax tone
is detected, renegotiates the call codec as configured below. The SIP line provider
must support the selected fax method and Re-Invite. The system must have
available VCM resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo
base card.
For systems in a network, fax relay is supported for fax calls between the systems.
The options are:
• None Select this option if fax is not supported by the line provider.
• G.711 G.711 is used for the sending and receiving of faxes.
• T38 T38 is used for the sending and receiving of faxes. This option is not
supported by Linux based systems.
• T38 Fallback When you enable this option, T38 is used for sending and
receiving faxes on a SIP line. If the called destination does not support T38, the
system will send a re-invite to change the transport method to G.711. This option
is not supported on Linux based systems.
Location
Default = Cloud.
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Line
Field
Description
Specify a location to associate the extension with a physical location. Associating
an extension with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined in the Location
form.
Call Initiation Timeout
Default = 4 seconds. Range = 1 to 99 seconds.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
DTMF Support
Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled
to the remote end. The options are:
• In Band
• RFC2833
• Info
Media Security
Default = Disable.
These setting control whether SRTP is used for this line and the settings used for
the SRTP. The options are:
• Same As System: Use the same settings as the system setting configured on
System | VoIP Security.
• Disable: Media security is not required. All media sessions (audio, video, and
data) will be enforced to use RTP only.
• Enforce: Media security is required. All media sessions (audio, video, and data)
will be enforced to use SRTP only.
Warning:
Selecting Enforce on a line or extension that does not support media
security will result in media setup failures.
• Best Effort: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
Advanced Media
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same setting as the system setting configured on the
System | VoIP Security tab.
• Encryptions: Default = RTP This setting allows selection of which parts of a
media session should be protected using encryption. The default is to encrypt
just the RTP stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication.
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Configuration Mode Field Descriptions
Field
Description
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the
option to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence
Suppression
Default = Off.
Allow Direct Media Path
Default = On
When selected, this option will detect periods of silence on any call over the line
and will not send any data during those silent periods. This feature is not used on
IP lines using G.711 between systems. On trunk's between networked systems,
the same setting should be set at both ends.
This settings controls whether IP calls must be routed via the system or can be
routed alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes
the need for a voice compression channel. Both ends of the calls must support
Direct Media and be using the same protocol (H.323 or SIP). Enabling this option
may cause some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the
system. RTP relay support allows calls between devices using the same audio
codec to not require a voice compression channel.
Re-Invite Supported
Default = On.
When enabled, Re-Invite can be used during a session to change the
characteristics of the session. For example when the target of an incoming call or a
transfer does not support the codec originally negotiated on the trunk. Requires the
ITSP to also support Re-Invite.
Codec Lockdown
Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In
response to a SIP offer with a list of codecs supported, some SIP user agents
supply a SDP answer that also lists multiple codecs. This means that the user
agent may switch to any of those codecs during the session without further
negotiation. The system does not support multiple concurrent codecs for a
session, so loss of speech path will occur if the codec is changed during the
session. If codec lockdown is enabled, when the system receives an SDP answer
with more than one codec from the list of offered codecs, it sends an extra reINVITE using just a single codec from the list and resubmits a new SDP offer with
just the single chosen codec.
Force direct media with
phones
Default = Off.
G.711 Fax ECAN
Default = Off.
Related links
Line | SM Line on page 382
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Line
SM Line T38 Fax
The settings on this tab are only accessible if Re-invite Supported and Fax Transport Support
are selected on the VoIP tab.
Fax relay is only supported on IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or IP500
Combo cards. Fax relay is not supported on Server Edition.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Related links
Line | SM Line on page 382
Line | IP Office Line
This line type is used to connect two IP Office systems.
In previous releases, connecting two IP Office systems was achieved using H.323 Lines configured
with Supplementary Services set to IP Office SCN. In the current release, the IP Office line type is
used to connect IP Office systems. Separating out the IP Office line type from the H.323 line type
allows for the logical grouping of features and functions available when connecting two IP Office
systems, including IP Office systems connected through the cloud.
Note:
Setting an IP Office line with Transport Type = Proprietary and Networking Level = SCN will
interwork with a previous release system configured with an H.323 SCN line.
Related links
Line on page 289
Line | IP Office Line on page 389
Line | IP Office Line Short Codes on page 393
Line | IP Office Line VoIP on page 393
Line | IP Office Line
Field
Description
Line Number
Default = 0. Range = 1 to 349.
Enter the line number that you wish. Note that this must be unique.
Transport Type
Default = Proprietary.
The options are
• Proprietary: The default connection type when connecting two IP Office systems.
• WebSocket Client / Websocket Server: A WebSocket connection is an HTTP /
HTTPS initiated TCP pipe through which Call signalling and Network Signaling is
tunneled. This transport type is used to connect IP Office systems through the cloud.
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Configuration Mode Field Descriptions
Field
Description
Selecting one of the WebSocket options enables the Security field and the
Password fields.
Networking Level
Default = SCN.
The options are
• None: No supplementary services are supported.
• SCN: This option is used to link IP Office system within a multi-site network. The
systems within a multi-site network automatically exchange information about users
and extensions, allowing remote users to be called without any additional
configuration on the local system.
Security
Default = Unsecured.
The Security field is available when Transport Type is set to WebSocket Client or
WebSocket Server.
The options are
• Unsecured : The connection uses HTTP/TCP.
• Medium: The connection uses HTTPS/TLS.
• High: The connection uses HTTPS/TLS. The server certificate store must contain
the client identity certificate.
Network Type
Default = Public.
This option is available if System | Telephony | Telephony | Restrict Network
Interconnect is enabled. It allows the trunk to be set as either Public or Private. The
system will return number busy indication to any attempt to connect a call on a
Private trunk to a Public trunk or vice versa. This restriction includes transfers,
forwarding and conference calls.
Due to the nature of this feature, its use is not recommended on systems also using
any of the following other system features: multi-site networks, VPNremote,
application telecommuter mode.
Include location
specific information
Telephone Number
Default = Off.
Enabled when Network Type is set to Private. Set to On if the PBX on the other end
of the trunk is toll compliant.
Default = Blank.
Used to remember the telephone number of this line. For information only.
Prefix
Default = Blank.
The prefix is used in the following ways:
• For incoming calls The ISDN messaging tags indicates the call type (National,
International or Unknown). If the call type is unknown, then the number in the Prefix
field is added to the ICLID.
• For outgoing calls The prefix is not stripped, therefore any prefixes not suitable for
external line presentation should be stripped using short codes.
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Line
Field
Description
Outgoing Group ID
Default = 1. Range 0 to 99999.
Short codes that specify a number to dial also specify the line group to be used. The
system will then seize a line with a matching Outgoing Group ID. The same ID can be
used for multiple lines.
In a Server Edition network, the Outgoing Group ID used on a system must also be
unique within the network, ie. the same ID cannot be used in the configuration of any
lines on another server system in the network. The IDs 99999 and 99998 which are
reserved for the H.323 lines to the Primary Server and Secondary Server respectively.
The IDs 99901 to 99930 are reserved for the H.323 lines from the Primary Server to
each expansion system in the network. The ID 0 cannot be used in a Server Edition
network.
Number of Channels
Default = 20, Range 0 to 250.
Defines the number of operational channels that are available on this line.
Outgoing Channels
Default = 20, Range 0 to 250.
This defines the number of channels available, on this line, for outgoing calls. This
should normally be the same as Number of Channels field, but can be reduced to
ensure incoming calls cannot be blocked by outgoing calls.
Gateway
Address
Default = Blank.
Enter the IP address of the gateway device at the remote end. This address must not
be shared by any other IP line (H.323, SIP, SES or IP DECT).
Location
Default = Cloud.
Specify a location to associate the extension with a physical location. Associating an
extension with a location:
• Allows emergency services to identify the source of an emergency call.
• Allows you to configure call admission control settings for the location.
The drop down list contains all locations that have been defined on Location |
Location.
Password
Default = Blank.
Confirm Password
The Password field is enabled when Transport Type is set to WebSocket Server or
WebSocket Client.
WebSockets are bi-directional HTTP or HTTPS communication pipes initiated from a
client to a server. They permit clients behind local a firewall to traverse the internet to
a server by using well known ports and protocols. A matching password must be set
at each end of the line.
Port
When Transport Type is set to Proprietary, the default port is 1720 and cannot be
changed.
When Transport Type is set to WebSocket Client, the default port is 80.
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Configuration Mode Field Descriptions
Field
Description
The Port field is not available when Transport Type is set to WebSocket Server.
The HTTP and HTTPS receive ports are defined at the system level in the security
settings System Details tab.
SCN Backup Options
These options are only available when the Networking Level option is set to SCN. The intention of this feature
is to attempt to maintain a minimal level of operation while problems with the local system are resolved.
Supports Fallback
Default = Off.
The Supports Fallback fields are available when Networking Level is set to SCN.
When enabled, all three backup options are set to On.
Backs up my IP
Phones
Default = Off.
This option is supported on Server Edition systems for Avaya 1600 and 9600 series H.
323 phones. IP500 (v2) systems also support 4600 and 5600 series phones. When
selected, the local system share information about the registered phones and users
on those phones with the other system. If the local system is no longer visible to the
phones, the phones will reregister with the other system. When phones have
registered with the other system, they show an R on their display.
If the setting System | Telephony | Phone Failback is set to Automatic, and the
phone’s primary gatekeeper has been up for more than 10 minutes, the system
causes idle phones to perform a failback recovery to the original system.
If using resilience backup to support Avaya IP phones, Auto-create Extn and Autocreate User should not be left enabled after initial configuration or any subsequent
addition of new extensions and users. Leaving auto-create options enabled on a
system that is a failover target may cause duplicate extension/user records on the
multi-site network under multiple failure scenarios.
Backs up my Hunt
Groups
Default = Off.
When selected, any hunt groups the local system is advertising to the network are
advertised from the other system when fallback is required. The trigger for this
occurring is Avaya H.323 phones registered with the local system registering with the
other system, ie. Backs up my IP Phones above must also be enabled. In a Server
Edition network this option is only available on the H.323 trunk from the Primary
Server to the Secondary Server.
When used, the only hunt group members that will be available are as follows:
• If the group was a distributed hunt group, those members who were remote
members on other systems still visible within the network.
• Any local members who have hot desked to another system still visible within the
network.
When the local system becomes visible to the other system again, the groups will
return to be advertised from the local system.
Backs up my
Voicemail
February 2016
Default = Off.
This option can be used if the local system is hosting the Voicemail Pro server being
used by the network. If selected, when the local system is no longer visible to the
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Line
Field
Description
voicemail server, the other system will act as host for the voicemail server. In a Server
Edition network this option is only available on the H.323 trunk from the Primary
Server to the Secondary Server. It is assumed as being on an is automatically set by
the Resilience Administration tool.
The option requires the other system to have licenses for the Voicemail Pro features
that are required to operate during any fallback period.
Backs up my IP DECT Default = Off.
Phones
This option is only available when Transport Type is set to Proprietary.
This option is used for Avaya IP DECT phones registered with the system. When
selected, it will share information about the registered phones and users on those
phones with the other system.
If the local system is no longer visible to the phones, the phones will reregister with
the other system. The users who were currently on those phones will appear on the
other system as if they had hot desked. Note that when the local system is restored to
the network, the phones will not automatically re-register with it. A phone reset via
either a phone power cycle or using the System Status Application is required. When
phones have registered with the other system, they will show an R on their display.
Note:
Only one IP Office Line can have this configuration parameter set to On.
Description
Default = Blank. Maximum 31 characters.
Use this field to enter a description of this configuration.
Related links
Line | IP Office Line on page 389
Line | IP Office Line Short Codes
Incoming calls on IP Office Lines are not routed using Incoming Call Route settings.
Short codes can be added and edited using the Add, Remove and Edit buttons. Alternatively you
can right-click on the list of existing short code to add and edit short codes. These settings are not
mergeable. Changes to these settings will require a reboot of the system.
Related links
Line | IP Office Line on page 389
Line | IP Office Line VoIP
Field
Description
Code Selection
Default = System Default
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Configuration Mode Field Descriptions
Field
Description
This field defines the codec or codecs offered during call setup. The available codecs
in default preference order are:
• G.711 A-Law
• G.711 U-LAW
• G.729
• G.723.1
Note that the default order for G.711 codecs will vary to match the system's default
companding setting. G.723.1 is not supported on Linux based systems.
The G.722 64K codec is also supported on IP500 V2 systems with IP500 VCM, IP500
VCM V2 or IP500 Combo cards. For Server Edition, it is supported on Primary
Server, Secondary Server and Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available in this form are set through the codec list on System | VoIP.
The options are:
• System Default This is the default setting. When selected, the codec list below
matches the codecs set in the system wide list.
• Custom This option allows specific configuration of the codec preferences to be
different from the system list. When Custom is selected, the list can be used to
select which codecs are in the Unused list and in the Selected list and to change
the order of the selected codecs.
Call Initiation
Timeouts
Default = 4. Range = 1 to 99 seconds.
Media Security
Default = Same as System.
This option sets how long the system should wait for a response to its attempt to
initiate a call before following the alternate routes set in an ARS form.
Secure RTP (SRTP) can be used between IP Offices to add additional security. These
settings control whether SRTP is used for this line and the settings used for the SRTP.
The options are:
• Same As System: Use the same settings as the system setting configured on the
System | VoIP Security tab.
• Disable: Media security is not required. All media sessions (audio, video, and data)
will be enforced to use RTP only.
• Enforce: Media security is required. All media sessions (audio, video, and data) will
be enforced to use SRTP only.
Warning:
Selecting Enforce on a line or extension that does not support media security
will result in media setup failures.
• Best Effort: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
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Line
Field
Description
Advanced Media
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same setting as the system setting configured on the
System | VoIP Security tab.
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option
to select SRTP_AES_CM_128_SHA1_32.
VoIP Silence
Suppression
Fax Transport
Support
Out Of Band DTMF
Default = Off
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
Default = Off
This option is only supported on trunks with the Networking Level set to SCN. On IP
Office Lines, fax relay is supported across multi-site network lines with Fax Transport
Support selected. This will use 2 VCM channels in each of the systems. Fax relay is
only supported on IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or IP500
Combo cards. Fax relay is not supported on Server Edition Linux servers.
Default = On.
Out of Band DTMF is set to on and cannot be changed.
Allow Direct Media
Path
Default = On
This settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
Media and be using the same protocol (H.323 or SIP). Enabling this option may cause
some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the system.
RTP relay support allows calls between devices using the same audio codec to not
require a voice compression channel.
Related links
Line | IP Office Line on page 389
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Configuration Mode Field Descriptions
Control Unit | Control Unit
The Control Unit form gives details of the system and some devices connected to or within the
system. This includes some modules within the control nit as well as external expansion modules.
For Server Edition systems, for the Primary Server, Secondary Server and Expansion System (L) it
shows details for the physical server platform and details for the IP Office Media service being
hosted on that server. For the Expansion System (V2) it shows details of the IP500 V2 control unit
and the cards installed into the control unit.
The New and Delete actions on this form have special functions.
• New This action is used to added a WAN3 expansion module. If when a WAN3 is added to the
system, the WAN3 is not recognized following a system reboot, New on this form can be used
to scan for the WAN3 module.
• Delete This action can only be used with external expansion modules. This action can only be
used with external expansion modules attached to a system. The action should used with
caution as deleting a module will also delete any extensions or lines associated with the
module being deleted. If the module is physically present, default records are automatically
recreated following a reboot.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Device Number
This is automatically allocated by the system.
Unit Type
The name of the device.
Version
The version of software running on each unit.
Serial Number
This is the number the system uses to tie a physical Control Unit to a device
configuration (device number). For the control unit this is the MAC address. For a
device connected to an Expansion port, it is the Expansion port number plus 1.
Unit IP Address
This field shows the IP address for the LAN1.
Interconnect Number
For external expansion modules this is the control unit expansion port used for
connection. For other devices this is 0.
Module Number
For external expansion modules this is the control unit expansion port used for
connection. For internal devices in the control unit, Control Unit is displayed.
Operating Mode
This field is available when a DS16B or DS30B digital expansion module is selected
as the control unit. Select the operating mode based on the type of telephones
deployed.
• DS - 1400, 9500, 5400, 2400, T3, 4400 Series Phones
• BST - T7000, M7000 Series Phones
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Extension
Related links
Configuration Mode Field Descriptions on page 208
Extension
By default, each extension is normally associated with a user and uses that user's directory number
and other setting. Users with a log in code can move between extensions by logging in and out, so
the directory number is not a fixed property of the extension.
Non-IP Extensions
Physical extension ports are either integral to the control unit or added by the installation of an
analog or digital phone expansion module. Extension records are automatically created for each
physical extension port within the system. These ports cannot be added or deleted manually.For
Server Edition, non-IP extensions are only supported on Expansion System (V2) units.
Standard Telephone
A standard extension.
Quiet Headset
Used for analog extension devices that are permanently off-hook.
:
IVR Port
Used for analog ports connected to devices that require a specific disconnect clear signal at the end
of each call.
Paging Speaker
An analog extension port set to be used as a paging speaker connection.
FAX Machine
Indicates that the extension is connected to a FAX machine.
MOH Source
Indicates that the extension is being used as a music on hold source.
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Configuration Mode Field Descriptions
IP Extensions
These are used for IP phone devices and VoIP applications.
H.323 or SIP Extension
This icon indicates an IP extension. IP extensions are either added manually or by the automatic
detection of the phone being connected. IP extensions can also be added manually to support a
third-party IP phone device. Note that third-party IP phone devices require entry of an IP End-Points
license.
IP DECT or SIP DECT
An extension port manually added to match extensions within an Avaya IP DECT system connected
to the system via an IP DECT line.
Related links
Configuration Mode Field Descriptions on page 208
Extension | Extn on page 398
Extension | Analog on page 401
Extension | VoIP on page 403
T38 Fax on page 415
Extension | IP DECT on page 416
Extension | SIP DECT Base on page 417
Extension | Extn
Contains settings applicable to most types of extension.
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Extension ID
The physical ID of the extension port. Except for IP extensions, this settings is allocated
by the system and is not configurable.
Base Extension
Range = 2 to 9 digits.
This is the directory number of the extension's default associated user.
Following a restart, the system will attempt to log in the user with the same extension
number (if they are not already logged in elsewhere in the multi-site network). This does
not occur it that user is set to Force Login.
If another user logs onto an extension, when they log out, the extension returns to its
default associated user unless they have logged in elsewhere or are set to Force
Login.
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Extension
Field
Description
Extensions associated with IP phones should not be given extension numbers greater
than 7 digits.
Users for CBC and CCC should only use up to 4 digit extension numbers.
Phone Password
Default = Blank. Range = Up to 31 digits.
H.323 Extensions only. Does not apply to T3 series phones and DECT phones.
The password that must be entered as part of phone registration. This password is
used to secure registration of H.323 phones when there is no matching user to
authenticate against. Required if Media Security is enabled.
Caller Display Type
Default = On.
Controls the presentation of caller display information for analog extensions, see Caller
Display. For digital and IP extensions, this value is fixed as On. The table below lists
the supported options, all others are currently not used and default to matching UK.
Reset Volume after
Calls
Type
Description
Off
Disables caller display.
On
Enables caller display using the caller display type appropriate to
the System Locale, see IP Office™ Platform Locale Settings. If a
different setting is required it can be selected from the list of
supported options. For an analog extension connected to a fax
server or other device that requires the pass through of DTMF
tones, select DTMFF.
UK
FSK before the first ring conforming to BT SIN 227. Name and
number.
UK20
As per UK but with a maximum length of 20 characters. Name and
number.
DTMFA
Caller ID in the DTMF pattern A<caller ID>C. Number only.
DTMFB
Caller ID in DTMF after call connection. Number only.
DTMFC
Caller ID in the DTMF pattern A<caller ID>#. Number only.
DTMFF
Sends the called number in DTMF after call connection. Number
only. Used for fax servers. When calls are delivered via a hunt group
it is recommended that hunt group queuing is not used. If hunt group
queuing is being used, set the Queue Type to Assign Call on Agent
Alert.
DTMFD
Caller ID in the DTMF pattern D<caller ID>C. Number only.
FSKA
Variant of UK used for BT Relate 1100 phones. Name and number.
FSKB
ETSI specification with 0.25 second leading ring. Name and number.
FSKC
ETSI specification with 1.2 second leading ring. Name and number.
FSKD
Conforms to Belcore specification. Name and number.
Default = Off. Resets the phone's handset volume after each call. This option is
supported on Avaya 1400, 1600, 2400, 4400, 4600, 5400, 5600, 6400, 9500 and 9600
Series phones.
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Configuration Mode Field Descriptions
Field
Description
Device Type
This field indicates, the last known type of phone connected to the extension port.
• Analogue extension ports always report as Analog Handset since the presence or
absence of actual analog phone cannot be detected.
• Digital extension ports report the type of digital phone connected or Unknown digital
handset if no phone is detected.
• H.323 extensions report the type of IP phone registered or Unknown H.323 handset
if no phone is currently registered as that extension.
• SIP extensions report the type of SIP phone registered or Unknown SIP device if no
SIP device is currently registered as that extension.
For some types of phone, the phone can only report its general type to the system but
not the specific model. When that is the case, the field acts as a drop-drown to allow
selection of a specific model. The value selected here is also reported in other
applications such as the System Status Application, SNMP, etc.
Default Type
Possible Phone Models
T7100
M7100, M7100N, T7100, Audio Conferencing Unit.
T7208
M7208, M7208N, T7208.
M7310
M7310, M7310N, T7406, T7406E.
M7310BLF
M7310BLF, T7316.
M7324
M7324, M7324N.
Location
Specify a location to associate the extension with a physical location. Associating an
extension with a location allows emergency services to identify the source of an
emergency call. The drop down list contains all locations that have been defined in the
Location page.
Fallback as Remote
Worker
Default = Auto.
Determines what fallback address is used for Remote Worker phone resiliency.
The options are:
• Auto: Use the fallback address configured on the IP Office Line providing the service.
• No: Use the alternate gateway private address.
• Yes: Use the alternate gateway public address.
Module
This field indicates the external expansion module on which the port is located. BP
indicates an analog phone extension port on the base or control unit. BD indicates a
digital station (DS) port on the control unit. For an IP500 V2 control unit, BD and BP is
also followed by the slot number. VoIP extensions report as 0.
Port
This field indicates the port number on the Module indicated above. VoIP extensions
report as 0.
Disable
Speakerphone
Default = Off (Speakerphone enabled).
When selected, disables the fixed SPEAKER button if present on the phone using this
extension port. Only supported on Avaya DS, TCM and H.323 IP phones. An audible
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Extension
Field
Description
beep is sounded when a disabled SPEAKER button is pressed. Incoming calls such as
pages and intercom calls are still connected but the speech path is not audible until the
user goes off-hook using the handset or headset. Similarly calls made or answered
using other buttons on the phone are not audible unless the user goes off-hook using
the handset or headset. Currently connected calls are not affected by changes to this
setting.
Force Authorization
Default = On.
This setting is used with SIP extension devices.
Related links
Extension on page 397
Extension | Analog
This tab contains settings that are applicable to analog extensions. These extensions are provided
by ports marked as POT or PHONE on control units and expansion modules.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Equipment Classification:
Default = Standard Telephone.
Only available for analog extension ports. Note that changes to this setting are mergeable.
Quiet Headset
On extensions set to Quiet Headset, the audio path is disabled when the extension is
idle. Ringing is presented in the audio path. Caller ID is not supported on the phone.
This option can be used with analog extensions where the handset is replaced by a
headset since in such a scenario audio is only desired when a call is connected. Since
the audio path is disabled when idle, the Quiet Headset extension cannot dial digits to
make calls. Therefore to make and answer calls this option is typically used with the
user Offhook Station (User | Telephony | Call Settings) setting which allows the
extension user to make and answer calls using applications.
Paging Speaker
Used for analog ports connected to a paging amplifier. This extension will present busy
and cannot be called or be used to make calls. It can only be accessed using Dial
Paging features.
When using a UPAM connected to an analog extension port, the extension's Equipment
Classification (Extension | Analog) should be set to IVR Port and not Paging Speaker.
Standard Telephone
Use for normal analog phones.
Door Phone 1/Door
Phone 2
These two options are currently not used and so are grayed out.
IVR Port
Used for analog ports connected to devices that require a disconnect clear signal (ie. a
break in the loop current) at the end of each call. When selected the Disconnect Pulse
Width is used.
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Configuration Mode Field Descriptions
Field
Description
FAX Machine
If fax Relay is being used, this setting should be selected on any analog extension
connected to an analog fax machine. This setting can also be used with SIP trunks.
MOH Source
If selected, the port can be used as a music on hold source in the Tones and Music
settings. An extension set as a music on hold source cannot make or receive calls. The
audio input can be monitored through the extension music on hold controls. A suitable
interface device is required to provide the audio input to the extension port. It must look
to the system like an off-hook analog phone. For example a transformer with a 600
Ohm winding (such as a Bogen WMT1A) or a dedicated MoH device with a 600Ohm
output designed for connection to a PBX extension port which is providing loop current
can be used.
Flash Hook Pulse Width
The following options are only available for analog extension ports. They define the length of loop break that will
be considered a time break recall (TBR) signal.
Use System
Defaults
Default = Selected (On)
Minimum Width
Range = 0 to 2540 milliseconds.
Use the default values appropriate to the system's locale.
Minimum hook flash length used if Use System Defaults is not selected. Shorter
breaks are ignored a glitches.
Maximum Width
Range = 0 to 2550 milliseconds.
Maximum hook flash length used if Use System Defaults is not selected. Longer
breaks are treated as clearing.
Disconnect Pulse
Width
Default = 0ms. Range = 0 to 2550ms
Message Waiting
Lamp Indication
Type
Default = None
This setting is used with analog extensions where the Equipment Classification
above has been set to IVR Port. It sets the length of loop current break used to indicate
call clearing.
Allows the selection of the message waiting indication (MWI) mode for analog and IP
DECT extensions.
For control unit and Phone V1 module analog extensions, the options are:
• 101V
• 51V Stepped
• 81V
• Bellcore FSK
• Line Reversal A
• Line Reversal B
• None
• On
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Extension
Field
Description
For Phone V2 external module extensions and IP500 Phone base cards, the additional
option 101V is available.
For IP500 V2 systems, if the option Restrict Analog Extension Ringer Voltage is
selected (System | Telephony | Telephony), the MWI options are restricted to:
• Line Reversal A
• Line Reversal B
• None
Any extensions set to another option are forced to Line Reversal A.
On defaults the message waiting indication as follows using the system locale.
Locale
'On' =
Argentina, Australia, Brazil, Canada, Chile, 51V Stepped
China, Colombia, Japan, Korea, Mexico,
New Zealand, Peru, Russia, Saudi Arabia,
South Africa, Spain, United States,
Venezuela
Bahrain, Belgium, Denmark, Egypt,
Finland, France, Germany, Greece, Hong
Kong, Hungary, Iceland, Italy, India,
Kuwait, Morocco, Netherlands, Norway,
Oman, Pakistan, Poland, Portugal, Qatar,
Singapore, Sweden, Switzerland, Taiwan,
Turkey, United Arab Emirates, United
Kingdom
Hook Persistency:
On = 101V on Phone V2 modules and
IP500 Phone cards, otherwise 81V.
Default = 100ms. Range = 50 to 255ms.
Defines the time frame (in milliseconds) in which the system will wait before determining
that the phone is off-hook.
Related links
Extension on page 397
Extension | VoIP
This tab is only available for H.323 and SIP extensions. The settings available will vary depending
on the extension type.
Related links
Extension on page 397
H.323 Extension VoIP on page 404
SIP Extension VoIP on page 409
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403
Configuration Mode Field Descriptions
H.323 Extension VoIP
These settings have changed in release 9.1. View the 9.0 settings on page 406.
These settings are shown for a H.323 IP extension.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
IP Address
Default = 0.0.0.0
The IP address of the phone. The default setting accepts connection from any address.
For phones using DHCP, the field is not updated to show the IP address being used by
the phone.
For T3 IP phones installed using DHCP, the address obtained and being used by the
phone is displayed. If that address is from the same range as the DHCP pool being
supported by the IP Office system, Manager will indicate an error.
The IP Address field can be used to restrict the the source IP address that can used by
a Remote H.323 Extension. However, it should not used in the case where there is
more than one remote extension behind the domestic router.
MAC Address
Default = 0000000000000 (Grayed out)
This field is grayed out and not used.
Codec Selection
Default = System Default This field defines the codec or codecs offered during call
setup.
The available codecs in default preference order are: G.711 A-Law, G.711 U-Law, G.
729 and G.723.1. Note that the default order for G.711 codecs will vary to match the
system's default companding setting. G.723.1 is not supported on Linux based
systems.
The G.722 64K codec is also supported on IP500/IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition it is supported on Primary
Server, Secondary Serverand Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available to be used are set through the System Codec list (System |
System Codec). The options are:
• System Default: This is the default setting. When selected, the codec list below show
matches the codecs set in the system wide Default Selection list (System | Codecs).
• Custom: This option allows specific configuration of the codec preferences to be
different from the system Default Selection list. When Custom is selected, the list
can be used to select which codecs are in the Unused list and in the Selected list
and to change the order of the selected codecs.
TDM | IP Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain
Default = Default (0dB). Range = -31dB to +31dB.
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Extension
Field
Description
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
Supplementary
Services
Default = H450.
Media Security
Selects the supplementary service signaling method for use with non-Avaya IP devices.
Options are None, QSIG and H450. For H450, hold and transfer are supported. Note
that the selected method must be supported by the remote end.
Default = Disable.
These settings control whether SRTP is used for this extension and the settings used
for the SRTP. The options are:
• Disable: Media security is not required. All media sessions (audio, video, and data)
will be enforced to use RTP only.
• Enforce: Media security is required. All media sessions (audio, video, and data) will
be enforced to use SRTP only.
• Best Effort: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
Advanced Media
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same setting as the system setting configured on the
System | VoIP Security tab.
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option to
select SRTP_AES_CM_128_SHA1_32.
VoIP Silence
Suppression
Default = Off
Enable FastStart for
non-Avaya IP
Phones
Default = Off
Out of Band DTMF
Default = On
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
A fast connection procedure. Reduces the number of messages that need to be
exchanged before an audio channel is created.
When on, DTMF is sent as a separate signal ("Out of Band") rather than as part of the
encoded voice stream ("In Band"). The "Out of Band" signaling is inserted back into the
audio by the remote end. This is recommended for low bit-rate compression modes
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Configuration Mode Field Descriptions
Field
Description
such as G.729 and G.723 where DTMF in the voice stream can become distorted.
Switch off for T3 IP extensions.
For Avaya 1600, 4600, 5600 and 9600 Series phones, the system will enforce the
appropriate setting for the phone type.
For Avaya T3 IP phones, when Out-Of-Band is unchecked, the Allow Direct Media
Path option is ignored and calls are via the system in order to provide tones.
Local Tones
Default = Off
When selected, the H.323 phones generate their own tones.
Allow Direct Media
Path
Default = On
This settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
Media and be using the same protocol (H.323 or SIP). Enabling this option may cause
some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the system.
RTP relay support allows calls between devices using the same audio codec to not
require a voice compression channel.
T3 IP phones must be configured to 20ms packet size to use RTP relay. The phone
must have firmware T246 or higher.
Reserve License
Default = None. Each Avaya IP phones requires an Avaya IP Endpoint license. Each
non-Avaya IP phones requires an 3rd Party IP Endpoint license. Normally these
licenses are issued in the order that devices register. This option allows this extension
to be pre-licensed before the device has registered. This helps prevent a previously
licensed phone becoming unlicensed following a system restart if unlicensed devices
are also present. The options are:
• Reserve Avaya IP Endpoint License
• Reserve 3rd Party IP Endpoint License
• Both
• None
Note that when WebLM licensing is enabled, this field is automatically set to Reserve
Avaya IP Endpoint License. The Both and None options are not available.
Related links
Extension | VoIP on page 403
H.323 Extension VoIP (9.0) on page 406
H.323 Extension VoIP (9.0)
These settings are shown for a H.323 IP extension.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
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Extension
Field
Description
IP Address
Default = 0.0.0.0
The IP address of the phone. The default setting accepts connection from any address.
For phones using DHCP, the field is not updated to show the IP address being used by
the phone.
For T3 IP phones installed using DHCP, the address obtained and being used by the
phone is displayed. If that address is from the same range as the DHCP pool being
supported by the IP Office system, Manager will indicate an error.
The IP Address field can be used to restrict the the source IP address that can used by
a Remote H.323 Extension. However, it should not used in the case where there is
more than one remote extension behind the domestic router.
MAC Address
Default = 0000000000000 (Grayed out)
This field is grayed out and not used.
Codec Selection
Default = System Default This field defines the codec or codecs offered during call
setup.
The available codecs in default preference order are: G.711 A-Law, G.711 U-Law, G.
729 and G.723.1. Note that the default order for G.711 codecs will vary to match the
system's default companding setting. G.723.1 is not supported on Linux based
systems.
The G.722 64K codec is also supported on IP500/IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition it is supported on Primary
Server, Secondary Serverand Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available to be used are set through the System Codec list (System |
System Codec). The options are:
• System Default: This is the default setting. When selected, the codec list below show
matches the codecs set in the system wide Default Selection list (System | Codecs).
• Custom: This option allows specific configuration of the codec preferences to be
different from the system Default Selection list. When Custom is selected, the list
can be used to select which codecs are in the Unused list and in the Selected list
and to change the order of the selected codecs.
TDM | IP Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the IP connection to the system TDM
interface. This field is not shown on Linux based platforms.
Supplementary
Services
Default = H450.
Selects the supplementary service signaling method for use with non-Avaya IP devices.
Options are None, QSIG and H450. For H450, hold and transfer are supported. Note
that the selected method must be supported by the remote end.
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Configuration Mode Field Descriptions
Field
Description
VoIP Silence
Suppression
Default = Off
Enable FastStart for
non-Avaya IP
Phones
Default = Off
Out of Band DTMF
Default = On
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends.
A fast connection procedure. Reduces the number of messages that need to be
exchanged before an audio channel is created.
When on, DTMF is sent as a separate signal ("Out of Band") rather than as part of the
encoded voice stream ("In Band"). The "Out of Band" signaling is inserted back into the
audio by the remote end. This is recommended for low bit-rate compression modes
such as G.729 and G.723 where DTMF in the voice stream can become distorted.
Switch off for T3 IP extensions.
For Avaya 1600, 4600, 5600 and 9600 Series phones, the system will enforce the
appropriate setting for the phone type.
For Avaya T3 IP phones, when Out-Of-Band is unchecked, the Allow Direct Media
Path option is ignored and calls are via the system in order to provide tones.
Local Tones
Default = Off
When selected, the H.323 phones generate their own tones.
Allow Direct Media
Path
Default = On
This settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure.
If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
Media and be using the same protocol (H.323 or SIP). Enabling this option may cause
some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the system.
RTP relay support allows calls between devices using the same audio codec to not
require a voice compression channel.
T3 IP phones must be configured to 20ms packet size to use RTP relay. The phone
must have firmware T246 or higher.
Reserve License
Default = None. Each Avaya IP phones requires an Avaya IP Endpoint license. Each
non-Avaya IP phones requires an 3rd Party IP Endpoint license. Normally these
licenses are issued in the order that devices register. This option allows this extension
to be pre-licensed before the device has registered. This helps prevent a previously
licensed phone becoming unlicensed following a system restart if unlicensed devices
are also present. The options are:
• Reserve Avaya IP Endpoint License
• Reserve 3rd Party IP Endpoint License
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Extension
Field
Description
• Both
• None
Note that when WebLM licensing is enabled, this field is automatically set to Reserve
Avaya IP Endpoint License. The Both and None options are not available.
Related links
H.323 Extension VoIP on page 404
SIP Extension VoIP
These settings have changed in release 9.1. View the 9.0 settings on page 412.
These settings are shown for SIP IP extensions.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
IP Address
Default = 0.0.0.0
The IP address of the phone. The default setting accepts connection from any address.
If an address is entered, registration is only accepted from a device with that address.
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup.
The available codecs in default preference order are: G.711 A-Law, G.711 ULAW, G.
729 and G.723.1. Note that the default order for G.711 codecs will vary to match the
system's default companding setting. G.723.1 is not supported on Linux based
systems.
The G.722 64K codec is also supported on IP500/IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition it is supported on Primary
Server, Secondary Serverand Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo.
The codecs available to be used are set through the System Codec list (System |
System Codec). The options are:
• System Default: This is the default setting. When selected, the codec list below show
matches the codecs set in the system wide Default Selection list (System |
Codecs).
• Custom: This option allows specific configuration of the codec preferences to be
different from the system Default Selection list. When Custom is selected, the list
can be used to select which codecs are in the Unused list and in the Selected list
and to change the order of the selected codecs.
Fax Transport
Support:
Default = Off.
This option is only available if Re-Invite Supported is selected. When enabled, the
system performs fax tone detection on calls routed via the line and, if fax tone is
detected, renegotiates the call codec as configured below. The SIP line provider must
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Configuration Mode Field Descriptions
Field
Description
support the selected fax method and Re-Invite. The system must have available VCM
resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo base card.
For systems in a network, fax relay is supported for fax calls between the systems.
The options are:
• None Select this option if fax is not supported by the line provider.
• G.711 G.711 is used for the sending and receiving of faxes.
• T38 T38 is used for the sending and receiving of faxes. This option is not supported
by Linux based systems.
• T38 Fallback When you enable this option, T38 is used for sending and receiving
faxes on a SIP line. If the called destination does not support T38, the system will
send a re-invite to change the transport method to G.711. This option is not
supported on Linux based systems.
TDM | IP Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain
Default = Default (0dB). Range = -31dB to +31dB. Allows adjustment of the gain on
audio from the IP connection to the system TDM interface. This field is not shown on
Linux based platforms.
DTMF Support
Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled to
the remote end. The supported options are In Band, RFC2833 or Info.
3rd Party Auto
Answer
Default = None.
This setting applies to 3rd party standard SIP extensions. The options are:
• RFC 5373: Add an RFC 5373 auto answer header to the INVITE.
• answer-after: Add answer-after header.
• device auto answers: IP Office relies on the phone to auto answer calls.
Media Security
Default = Same as System.
These settings control whether SRTP is used for this extension and the settings used
for the SRTP. The options are:
• Same As System: Use the same settings as the system setting configured on the
System | VoIP Security tab.
• Disable: Media security is not required. All media sessions (audio, video, and data)
will be enforced to use RTP only.
• Enforce: Media security is required. All media sessions (audio, video, and data) will
be enforced to use SRTP only.
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Extension
Field
Description
Warning:
Selecting Enforce on a line or extension that does not support media security
will result in media setup failures.
• Best Effort: Media security is preferred. Attempt to use secure media first and if
unsuccessful, fall back to non-secure media.
Advanced Media
Security Options
Not displayed if Media Security is set to Disabled. The options are:
• Same as System: Use the same setting as the system setting configured on the
System | VoIP Security tab.
• Encryptions: Default = RTP This setting allows selection of which parts of a media
session should be protected using encryption. The default is to encrypt just the RTP
stream (the speech).
• Authentication: Default = RTP and RTCP This setting allows selection of which
parts of the media session should be protected using authentication.
• Replay Protection SRTP Window Size: Default = 64. Currently not adjustable.
• Crypto Suites: Default = SRTP_AES_CM_128_SHA1_80. There is also the option to
select SRTP_AES_CM_128_SHA1_32.
VoIP Silence
Suppression
Default = Off
Local Hold Music
Default = Off.
Allow Direct Media
Path
Default = On.
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends
This settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure
If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
Media and be using the same protocol (H.323 or SIP). Enabling this option may cause
some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the system.
RTP relay support allows calls between devices using the same audio codec to not
require a voice compression channel.
RE-Invite Supported
Default = On.
When enabled, Re-Invite can be used during a session to change the characteristics of
the session. For example when the target of an incoming call or a transfer does not
support the codec originally negotiated on the trunk. Requires the ITSP to also support
Re-Invite.
Codec Lockdown
Default = Off.
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Configuration Mode Field Descriptions
Field
Description
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response
to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP
answer that also lists multiple codecs. This means that the user agent may switch to
any of those codecs during the session without further negotiation. The system does
not support multiple concurrent codecs for a session, so loss of speech path will occur
if the codec is changed during the session. If codec lockdown is enabled, when the
system receives an SDP answer with more than one codec from the list of offered
codecs, it sends an extra re-INVITE using just a single codec from the list and
resubmits a new SDP offer with just the single chosen codec.
Reserve License
Default = None. Each Avaya IP phones requires an Avaya IP Endpoint license. Each
non-Avaya IP phones requires an 3rd Party IP Endpoint license. Normally these
licenses are issued in the order that devices register. This option allows this extension
to be pre-licensed before the device has registered. This helps prevent a previously
licensed phone becoming unlicensed following a system restart if unlicensed devices
are also present. The options are:
• Reserve Avaya IP Endpoint License
• Reserve 3rd Party IP Endpoint License
• Both
• None
Note the following:
• When WebLM licensing is enabled, this field is automatically set to Reserve Avaya
IP Endpoint License. The Both and None options are not available.
• When the Profile of the corresponding user is set to Centralized User, this field is
automatically set to Centralized Endpoint License and cannot be changed.
Related links
Extension | VoIP on page 403
Extension SIP VoIP (9.0) on page 412
Extension SIP VoIP (9.0)
These settings are shown for SIP IP extensions.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
IP Address
Default = 0.0.0.0
The IP address of the phone. The default setting accepts connection from any address.
If an address is entered, registration is only accepted from a device with that address.
Codec Selection
Default = System Default
This field defines the codec or codecs offered during call setup.
The available codecs in default preference order are: G.711 A-Law, G.711 ULAW, G.
729 and G.723.1. Note that the default order for G.711 codecs will vary to match the
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Extension
Field
Description
system's default companding setting. G.723.1 is not supported on Linux based
systems.
The G.722 64K codec is also supported on IP500/IP500 V2 systems with IP500 VCM,
IP500 VCM V2 or IP500 Combo cards. For Server Edition it is supported on Primary
Server, Secondary Serverand Expansion System (L) systems and on Expansion
System (V2) systems fitted with IP500 VCM, IP500 VCM V2 or IP500 Combo. It is not
supported on any systems that include any IP400 VCM cards.
The codecs available to be used are set through the System Codec list (System |
System Codec). The options are:
• System Default: This is the default setting. When selected, the codec list below show
matches the codecs set in the system wide Default Selection list (System | Codecs).
• Custom: This option allows specific configuration of the codec preferences to be
different from the system Default Selection list. When Custom is selected, the list
can be used to select which codecs are in the Unused list and in the Selected list
and to change the order of the selected codecs.
Fax Transport
Support:
Default = Off.
This option is only available if Re-Invite Supported is selected. When enabled, the
system performs fax tone detection on calls routed via the line and, if fax tone is
detected, renegotiates the call codec as configured below. The SIP line provider must
support the selected fax method and Re-Invite. The system must have available VCM
resources using an IP500 VCM, IP500 VCM V2 or IP500 Combo base card. For
systems in a network, fax relay is supported for fax calls between the systems.
The options are:
• None: Select this option if fax is not supported by the line provider.
• G.711: G.711 is used for the sending and receiving of faxes.
• T38: T38 is used for the sending and receiving of faxes.
• T38 Fallback: When you enable this option, T38 is used for sending and receiving
faxes on a SIP line. If the called destination does not support T38, the system will
send a re-invite to change the transport method to G.711. This option is not
supported by Linux based systems.
TDM | IP Gain
Default = Default (0dB). Range = -31dB to +31dB.
Allows adjustment of the gain on audio from the system TDM interface to the IP
connection. This field is not shown on Linux based platforms.
IP | TDM Gain
Default = Default (0dB). Range = -31dB to +31dB. Allows adjustment of the gain on
audio from the IP connection to the system TDM interface. This field is not shown on
Linux based platforms.
DTMF Support
Default = RFC2833.
This setting is used to select the method by which DTMF key presses are signalled to
the remote end. The supported options are In Band, RFC2833 or Info.
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413
Configuration Mode Field Descriptions
Field
Description
VoIP Silence
Suppression
Default = Off
Local Hold Music
Default = Off.
Allow Direct Media
Path
Default = On.
When selected, this option will detect periods of silence on any call over the line and
will not send any data during those silent periods. This feature is not used on IP lines
using G.711 between systems. On trunk's between networked systems, the same
setting should be set at both ends
This settings controls whether IP calls must be routed via the system or can be routed
alternately if possible within the network structure
If enabled, IP calls can take routes other than through the system. This removes the
need for a voice compression channel. Both ends of the calls must support Direct
Media and be using the same protocol (H.323 or SIP). Enabling this option may cause
some vendors problems with changing the media path mid call.
If disabled or not supported at on one end of the call, the call is routed via the system.
RTP relay support allows calls between devices using the same audio codec to not
require a voice compression channel.
RE-Invite Supported
Default = On.
When enabled, Re-Invite can be used during a session to change the characteristics of
the session. For example when the target of an incoming call or a transfer does not
support the codec originally negotiated on the trunk. Requires the ITSP to also support
Re-Invite.
Codec Lockdown
Default = Off.
Supports RFC 3264 Section 10.2 when RE-Invite Supported is enabled. In response
to a SIP offer with a list of codecs supported, some SIP user agents supply a SDP
answer that also lists multiple codecs. This means that the user agent may switch to
any of those codecs during the session without further negotiation. The system does
not support multiple concurrent codecs for a session, so loss of speech path will occur
if the codec is changed during the session. If codec lockdown is enabled, when the
system receives an SDP answer with more than one codec from the list of offered
codecs, it sends an extra re-INVITE using just a single codec from the list and
resubmits a new SDP offer with just the single chosen codec.
Reserve License
Default = None. Each Avaya IP phones requires an Avaya IP Endpoint license. Each
non-Avaya IP phones requires an 3rd Party IP Endpoint license. Normally these
licenses are issued in the order that devices register. This option allows this extension
to be pre-licensed before the device has registered. This helps prevent a previously
licensed phone becoming unlicensed following a system restart if unlicensed devices
are also present. The options are:
• Reserve Avaya IP Endpoint License
• Reserve 3rd Party IP Endpoint License
• Both
• None
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Extension
Field
Description
Note the following:
• When WebLM licensing is enabled, this field is automatically set to Reserve Avaya
IP Endpoint License. The Both and None options are not available.
• When the Profile of the corresponding user is set to Centralized User, this field is
automatically set to Centralized Endpoint License and cannot be changed.
Related links
SIP Extension VoIP on page 409
T38 Fax
The settings on this tab are only accessible if Re-invite Supported and Fax Transport Support
are selected on the VoIP tab.
Fax relay is only supported on IP500 V2 systems with IP500 VCM, IP500 VCM V2 and or IP500
Combo cards. Fax relay is not supported on Server Edition.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
Use Default Values
Default = On.
If selected, all the fields are set to their default values and greyed out.
T38 Fax Version
Default = 3.
The system can support Versions 0, 1, 2 and 3. During fax relay, the two gateways will
negotiate to use the highest version which they both support.
Transport
Default = UDPTL (fixed).
Currently only UDPTL is supported. TCP and RTP transport are not supported
For UDPTL, redundancy error correction is supported. Forward Error Correction (FEC)
is not supported.
Redundancy
Redundancy sends additional fax packets in order to increase the reliability. However increased redundancy
increases the bandwidth required for the fax transport.
Low Speed
Default = 0 (No redundancy). Range = 0 to 5. Sets the number of redundant T38 fax
packets that should be sent for low speed V.21 T.30 fax transmissions.
High Speed
Default = 0 (No redundancy). Range = 0 to 5. Sets the number of redundant T38 fax
packets that should be sent for V.17, V.27 and V.28 fax transmissions.
TCF Method
Default = Trans TCF.
TCF = Training Check Frame.
Max Bit Rate (bps)
Default = 14400. Lower rates can be selected if the current rate is not supported by the
fax equipment or is found to not be reliable.
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Configuration Mode Field Descriptions
Field
Description
EFlag Start Timer
(msecs)
Default = 2600.
EFlag Stop Timer
(msecs)
Default = 2300.
Tx Network Timeout
(secs)
Default = 150.
Scan Line Fix-up
Default = On.
TFOP Enhancement
Default = On.
Disable T30 ECM
Default = Off. When selected, disabled the T.30 Error Correction Mode used for fax
transmission.
Disable EFlags For
First DIS
Default = Off.
Disable T30 MR
Compression
Default = Off.
NSF Override
Default = Off. If selected, the NSF (Non-Standard Facility) information sent by the T38 device can be overridden
using the values in the fields below
Country Code
Default = 0.
Vendor Code
Default = 0.
Related links
Extension on page 397
Extension | IP DECT
IP DECT extensions are created manually after an IP DECT line has been added to the
configuration or added automatically as DECT handsets subscribe to the DECT system.
These settings are mergeable with the exception of the Reserve License setting. Changing the
Reserve License settings requires a reboot of the system.
Field
Description
DECT Line ID
Use the drop-down list to select the IP DECT line from the system to the Avaya IP
DECT system.
Message Waiting
Lamp Indication
Type
Default = On
Allows selection of the message waiting indication to use with the IP DECT extension.
The options are:
• None
• On
Reserve License
Default = None.
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Extension
Field
Description
Avaya IP phones require an Avaya IP Endpoint license in order to register with the
system. Normally licenses are issued in the order that devices register. This option
allows this extension to be pre-licensed before the device has registered. The options
are
• Reserve Avaya IP Endpoint License
• None
Note that when WebLM licensing is enabled, this field is automatically set to Reserve
Avaya IP Endpoint License and cannot be changed.
The additional fields below depend on whether the IP DECT line has Enable Provisioning
selected.
Field
Description
Enable Provisioning Not Selected
Handset Type
Default = Unknown
Correct selection of the handset type allows application of appropriate settings for the
handset display and buttons. Selectable handset types are 3720, 3725, 3740, 3749 or
Unknown.
Enable Provisioning Selected
IPEI
Default = 0
This field, if set to a value other than 0, sets the IPEI number of the handset that is able
to subscribe to the DECT R4 system using this extension number. The IPEI for each
DECT handset is unique.
Use Handset
Configuration
Default = Off.
If Use Handset Configuration. is selected, the handset user is able to set the phone
language and date/time format. If not selected, those settings will be driven by the
system or user locale settings in the system configuration.
Related links
Extension on page 397
Extension | SIP DECT Base
This tab is displayed for SIP DECT extensions.
These settings are not mergeable. Changes to these settings will require a reboot of the system.
Field
Description
DECT Line ID
Use the drop-down list to select the SIP DECT Line from the system to the base station.
Related links
Extension on page 397
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Configuration Mode Field Descriptions
User
Users are the people who use the system. They do not necessary have to be an extension user, for
example users are used for RAS dial in data access. In addition, more users can be created than
there are extensions, with users logging in to an extension when they want to receive calls.
By default, a user is automatically created to match each extension. They are numbered from 201
upwards and the first 16 are placed in the hunt group Main (200), which is the default destination for
incoming calls.
Standard User: A standard user.
Centralized User: A centralized user.
No User: Used to apply settings for extensions which currently have no associated user.
Remote Manager: Used as the default settings for dial in user connections.
Hot Desking User: Users with a Login Code can move between extensions by logging in and
off.
When a User is Deleted?
When a user is deleted, any calls in progress continue until completed. The ownership of the call is
shown as the NoUser user. Merging the deletion of a user causes all references to that deleted user
to be removed from the system.
Changing a User's Extension
Changing a user's extension number automatically logs the user in on the matching base extension
if available and the user doesn't have Forced Login enabled. If Forced Login is enabled, then the
user remains on the current extension being used until they log out and log in at the new extension.
Note that changing a user's extension number affects the user's ability to collect Voicemail
messages from their own extension. Each user's extension is set up as a "trusted location" under
the Source Numbers tab of the User configuration form. This "trusted location" allows the user to
dial *17 to collect Voicemail from his own extension. Therefore if the extension number is changed
so must the "trusted location".
The following related configuration items are automatically updated when a user extension is
changed:
•
•
•
•
User, Coverage and Bridged Appearance buttons associated with the user.
Hunt group membership (disabled membership state is maintained).
Forwards and Follow Me's set to the user as the destination.
Incoming call routes to this destination.
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User
• Dial in source numbers for access to the user's own voicemail.
• Direct call pickup buttons are updated.
• The extension number of an associated extension is updated.
Server Edition User Management
In a Server Edition network, individual users are still added to the configuration of a particular
server. Typically they are added to the configuration of the server that hosts the user's physical
extension or supports their main place of work. That server is treated as the host system for the
user. However, once a user is added to the configuration of a particular system, their settings can
also be accessed and edited under the Users grouping at the top of the navigation pane of
Manager.
Related links
Configuration Mode Field Descriptions on page 208
User | User on page 419
User | Voicemail on page 425
User | DND on page 429
User | Short Codes on page 430
Source Numbers on page 431
User | Telephony on page 435
User | Forwarding on page 444
User | Dial In on page 447
User | Voice Recording on page 447
User | Button Programming on page 449
User | Menu Programming on page 450
User | Mobility on page 452
User | Hunt Group Memberships on page 455
User | Announcements on page 455
User | SIP on page 457
User | Personal Directory on page 457
User | Web Self Administration on page 459
User | User
Users are the people who use the system or are Dial In users for data access. A system User may
or may not have an Extension Number that physical exists - this is useful if users do not require a
physical extension but wish to use system features, for example voicemail, forwarding, etc.
NoUser is used to apply settings to extensions which have no associated user. Remote Manager is
used as the default settings for dial in connections.
For Server Edition, this type of configuration record can be saved as a template and new records
created from a template. See Templates.
These settings are mergeable. Changes to these settings do not require a reboot of the system.
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Configuration Mode Field Descriptions
In Manager, symbol indicates that the setting can also be set and locked within a set of user
rights with which the user is associated using the Working Hour User Rights and Out of Hours User
Rights settings. The user rights applied can be controlled by a time profile selected as the user's
Working Hours Time Profile setting. The effect of the user rights can be displayed using the User
Rights View control.
Field
Description
Name
Range = Up to 15 characters.
This is the user's account name used for RAS Dial In, Caller Display and voicemail
mailbox. As the display on Caller Display telephones is normally only 16 digits long it is
useful to keep the name short. Only alphanumeric characters and space are supported in
this field. This field is case sensitive and must be unique.
Names should not start with a space. Do not use punctuation characters such as #, ?, /,
^, > and ,.
Voicemail uses the name to match a user to their mailbox. Changing a user's name will
route their voicemail calls to a new mailbox. Note however that Voicemail Pro is not case
sensitive and will treat names such as "Steve Smith", "steve smith" and "STEVE SMITH"
as being the same.
Do not provision a user with the Name "admin". The user name "admin" is a reserved
value on the one-X Portal Instant Message (IM) and Presence server. An IP Office
"admin" user will not have IM and presence services.
For Outbound Contact Express deployments, when an agent logs in to an extension, the
user name associated with the extension is changed to the agent ID.
Password
Default = Blan